From cbannister at slingshot.co.nz  Fri Aug  1 00:19:29 2014
From: cbannister at slingshot.co.nz (Chris Bannister)
Date: Fri, 1 Aug 2014 12:19:29 +1200
Subject: [LAU] JACK on Pi... almost working
In-Reply-To: <201407301101.11298.gerhard.zintel@web.de>
References: <20140726174251.GA3042@q400a.mobile.restivo.org>
 <53D3F70E.8010805@autostatic.com>
 <20140730075025.GE858@q400a.mobile.restivo.org>
 <201407301101.11298.gerhard.zintel@web.de>
Message-ID: <20140801001929.GB24535@tal>

On Wed, Jul 30, 2014 at 11:01:11AM +0200, Gerhard Zintel wrote:
> > 
> 
> use a Banana Pi, it is not nearly as crappy as Raspberry, much more
> powerfull (2 cores, 1 GHz, 1 GB), consumes less power and only a bit
> more expensive:
> 
> http://hardware-libre.fr/2014/06/raspberry-vs-banana-vs-a10-olinuxino-powering-and-sata-performance/

There's no actual figures! :( Not very helpful.


-- 
"If you're not careful, the newspapers will have you hating the people
who are being oppressed, and loving the people who are doing the 
oppressing." --- Malcolm X

From jeremy at autostatic.com  Fri Aug  1 05:42:06 2014
From: jeremy at autostatic.com (Jeremy Jongepier)
Date: Fri, 01 Aug 2014 07:42:06 +0200
Subject: [LAU] JACK on Pi... almost working
In-Reply-To: <201407301101.11298.gerhard.zintel@web.de>
References: <20140726174251.GA3042@q400a.mobile.restivo.org>
 <53D3F70E.8010805@autostatic.com>
 <20140730075025.GE858@q400a.mobile.restivo.org>
 <201407301101.11298.gerhard.zintel@web.de>
Message-ID: <53DB28AE.1080003@autostatic.com>

On 07/30/2014 11:01 AM, Gerhard Zintel wrote:
> use a Banana Pi, it is not nearly as crappy as Raspberry, much more powerfull (2 cores, 1 GHz, 1 GB), consumes less power and only a bit more expensive:
> 
> see here
> http://hardware-libre.fr/2014/06/raspberry-vs-banana-hardware-duel/
> and here
> http://hardware-libre.fr/2014/06/raspberry-vs-banana-vs-a10-olinuxino-powering-and-sata-performance/
> 
> Gerhard

What I like about the RPi is that it is afaik still the cheapest board
available. It also has a huge community and it's not being run by a
company but a foundation with a non-profit goal.
For real-time audio purposes it is certainly not the best choice, but
then, what do you expect for that price? And what do you expect from a
device that is made for educational purposes, not real-time,
low-latency, pro audio?

Jeremy

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From willgodfrey at musically.me.uk  Fri Aug  1 06:09:00 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Fri, 1 Aug 2014 07:09:00 +0100
Subject: [LAU] JACK on Pi... almost working
In-Reply-To: <53DB28AE.1080003@autostatic.com>
References: <20140726174251.GA3042@q400a.mobile.restivo.org>
 <53D3F70E.8010805@autostatic.com>
 <20140730075025.GE858@q400a.mobile.restivo.org>
 <201407301101.11298.gerhard.zintel@web.de>
 <53DB28AE.1080003@autostatic.com>
Message-ID: <20140801070900.16dcf622@debian>

On Fri, 01 Aug 2014 07:42:06 +0200
Jeremy Jongepier <jeremy at autostatic.com> wrote:

> On 07/30/2014 11:01 AM, Gerhard Zintel wrote:
> > use a Banana Pi, it is not nearly as crappy as Raspberry, much more powerfull (2 cores, 1 GHz, 1 GB), consumes less power and only a bit more expensive:
> > 
> > see here
> > http://hardware-libre.fr/2014/06/raspberry-vs-banana-hardware-duel/
> > and here
> > http://hardware-libre.fr/2014/06/raspberry-vs-banana-vs-a10-olinuxino-powering-and-sata-performance/
> > 
> > Gerhard
> 
> What I like about the RPi is that it is afaik still the cheapest board
> available. It also has a huge community and it's not being run by a
> company but a foundation with a non-profit goal.
> For real-time audio purposes it is certainly not the best choice, but
> then, what do you expect for that price? And what do you expect from a
> device that is made for educational purposes, not real-time,
> low-latency, pro audio?
> 
> Jeremy

Exactly!
This is something people seem to forget when they complain that it can't do
this, or that. They also forget that the founders sank their own personal money
into this project at the start, called on favours and twisted arms.

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From gerhard.zintel at web.de  Fri Aug  1 08:25:40 2014
From: gerhard.zintel at web.de (Gerhard Zintel)
Date: Fri, 1 Aug 2014 10:25:40 +0200
Subject: [LAU] JACK on Pi... almost working
In-Reply-To: <20140801001929.GB24535@tal>
References: <20140726174251.GA3042@q400a.mobile.restivo.org>
 <201407301101.11298.gerhard.zintel@web.de> <20140801001929.GB24535@tal>
Message-ID: <201408011025.40916.gerhard.zintel@web.de>

On Friday 01 August 2014, Chris Bannister wrote:
> On Wed, Jul 30, 2014 at 11:01:11AM +0200, Gerhard Zintel wrote:
> > > 
> > 
> > use a Banana Pi, it is not nearly as crappy as Raspberry, much more
> > powerfull (2 cores, 1 GHz, 1 GB), consumes less power and only a bit
> > more expensive:
> > 
> > http://hardware-libre.fr/2014/06/raspberry-vs-banana-vs-a10-olinuxino-powering-and-sata-performance/
> 
> There's no actual figures! :( Not very helpful.
> 
> 

Sorry to disappoint you. This link was mainly to show that power consumption of Banana-Pi is lower than for Raspberry Pi model B (because Ken wrote: "The dream 7 years ago was to have a portable, battery-powered, rock-solid, headless Linux synth "). I thought the figures (I read it as specifications?) are in the other link I gave (http://hardware-libre.fr/2014/06/raspberry-vs-banana-hardware-duel/).

For better specs see here:
http://www.bananapi.org/p/product.html

Gerhard

From kees.vanveen at gmail.com  Fri Aug  1 16:12:22 2014
From: kees.vanveen at gmail.com (Kees van Veen)
Date: Fri, 01 Aug 2014 18:12:22 +0200
Subject: [LAU] Linux Poetry: X-run blues
Message-ID: <53DBBC66.3090408@gmail.com>

Hi,

See below for my contribution to Linux audio poetry, reflections of 
personal struggle ;-)

Thanks to all Linux audio developers and supporters for all the great 
software, I hope no one feels offended or ridiculed by the lyrics.

Thanks,
Kees

X-run blues (battle hymn of Linux audio newbies)
----------------
I woke up early one morning, I got me the X-run blues
I woke up early one morning, I got me the X-run blues
Must've missed a couple of warnings, and I don't have any clues

I was all set to record, enter the X-run blues
I was really ready to record, enter the X-run blues
Could be the cpufreq, or them funny IRQs

I started googling for advice on lower latency
I started googling for advice on lower latency
It said 'Compile yourself a realtime kernel, son, and you just wait and 
see'

I switched to Ardour III, I paid my Ardour dues
I switched to Ardour III, I paid my Ardour dues
Tried all of the Jack flavors, can't lose the X-run blues

I developed 'X' paranoia, Xubuntu's where I'm at
I developed 'X' paranoia, the worst I ever had
Would switching to Fedora get me F-runs instead

Lately I don't care too much, though there's lots of things to check
Right now I don't care too much, lots of settings still to check
I like the vintage pops and crackles, just like a Robert Johnson track


From egor.sanin at gmail.com  Fri Aug  1 17:45:04 2014
From: egor.sanin at gmail.com (Egor Sanin)
Date: Fri, 1 Aug 2014 13:45:04 -0400
Subject: [LAU] Linux Poetry: X-run blues
In-Reply-To: <53DBBC66.3090408@gmail.com>
References: <53DBBC66.3090408@gmail.com>
Message-ID: <CALVwFW-r8m8hiUOes-JLfHiKDpvoPacT6KqdQ3gsnj=NyM7Cvw@mail.gmail.com>

On 8/1/14, Kees van Veen <kees.vanveen at gmail.com> wrote:
> Hi,
>
> See below for my contribution to Linux audio poetry, reflections of
> personal struggle ;-)
>
> Thanks to all Linux audio developers and supporters for all the great
> software, I hope no one feels offended or ridiculed by the lyrics.
>
> Thanks,
> Kees
>
> X-run blues (battle hymn of Linux audio newbies)
> ----------------
> I woke up early one morning, I got me the X-run blues
> I woke up early one morning, I got me the X-run blues
> Must've missed a couple of warnings, and I don't have any clues
>
> I was all set to record, enter the X-run blues
> I was really ready to record, enter the X-run blues
> Could be the cpufreq, or them funny IRQs
>
> I started googling for advice on lower latency
> I started googling for advice on lower latency
> It said 'Compile yourself a realtime kernel, son, and you just wait and
> see'
>
> I switched to Ardour III, I paid my Ardour dues
> I switched to Ardour III, I paid my Ardour dues
> Tried all of the Jack flavors, can't lose the X-run blues
>
> I developed 'X' paranoia, Xubuntu's where I'm at
> I developed 'X' paranoia, the worst I ever had
> Would switching to Fedora get me F-runs instead
>
> Lately I don't care too much, though there's lots of things to check
> Right now I don't care too much, lots of settings still to check
> I like the vintage pops and crackles, just like a Robert Johnson track

Excellent!
Totally waiting to hear this performed at next year's LAC sound night.

From ralf.mardorf at rocketmail.com  Fri Aug  1 19:23:56 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Fri, 01 Aug 2014 21:23:56 +0200
Subject: [LAU] [Bulk] Re:  Linux Poetry: X-run blues
In-Reply-To: <CALVwFW-r8m8hiUOes-JLfHiKDpvoPacT6KqdQ3gsnj=NyM7Cvw@mail.gmail.com>
References: <53DBBC66.3090408@gmail.com>
 <CALVwFW-r8m8hiUOes-JLfHiKDpvoPacT6KqdQ3gsnj=NyM7Cvw@mail.gmail.com>
Message-ID: <1406921036.14205.1.camel@rocketmail.com>

On Fri, 2014-08-01 at 13:45 -0400, Egor Sanin wrote:
> On 8/1/14, Kees van Veen <kees.vanveen at gmail.com> wrote:
[snip]
> Excellent!
> Totally waiting to hear this performed at next year's LAC sound night.

The lyrics seem to fit to the twelve-bar blues chord progressions and
the repeated parts of the lyrics seem to point to the typical whiny kind
of speech song, so the melody unlikely will be a surprise.

Hey Kees, where you goin' with that sledgehammer in your hands?
Hey Kees, I said where you goin' with that sledgehammer in your hands?


From ivan_521521 at yahoo.com  Sat Aug  2 00:02:04 2014
From: ivan_521521 at yahoo.com (Ivan K)
Date: Fri, 1 Aug 2014 17:02:04 -0700
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
Message-ID: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>

Well, I have the s/pdif output of my
USBDualTubePre microphone preamp plugged
into the s/pdif input of my M-Audio 2496,
though I have yet to be able to record?
any sound.

I am using Audacity.

I have used Audacity successfully in the past
to record though the analog inputs of the
M-Audio 2496 (to digitize sound from cassettes?
and vinyl) but Audacity's VU meters just?
don't register anything from the microphone?
going to the USBDualTubePre preamp to the?
M-Audio 2496's s/pdif input.

Does anyone have any suggestions?

I imagine I need to configure something
with one of the mixers, I have available?:

? alsamixer
? alsamixergui
? envy24control   (Envy24 Control Utility)
? xfce4-mixer


Thank you for your help.


P.S.  I am trying to only do a mono recording and so
I have my microphone plugged into the right channel
of the preamp.

From ken at restivo.org  Sat Aug  2 00:55:09 2014
From: ken at restivo.org (Ken Restivo)
Date: Fri, 1 Aug 2014 17:55:09 -0700
Subject: [LAU] JACK on Pi... almost working
In-Reply-To: <20140801070900.16dcf622@debian>
References: <20140726174251.GA3042@q400a.mobile.restivo.org>
 <53D3F70E.8010805@autostatic.com>
 <20140730075025.GE858@q400a.mobile.restivo.org>
 <201407301101.11298.gerhard.zintel@web.de>
 <53DB28AE.1080003@autostatic.com> <20140801070900.16dcf622@debian>
Message-ID: <20140802005509.GC8368@q400a.mobile.restivo.org>

On Fri, Aug 01, 2014 at 07:09:00AM +0100, Will Godfrey wrote:
> On Fri, 01 Aug 2014 07:42:06 +0200
> Jeremy Jongepier <jeremy at autostatic.com> wrote:
> 
> > On 07/30/2014 11:01 AM, Gerhard Zintel wrote:
> > > use a Banana Pi, it is not nearly as crappy as Raspberry, much more powerfull (2 cores, 1 GHz, 1 GB), consumes less power and only a bit more expensive:
> > > 
> > > see here
> > > http://hardware-libre.fr/2014/06/raspberry-vs-banana-hardware-duel/
> > > and here
> > > http://hardware-libre.fr/2014/06/raspberry-vs-banana-vs-a10-olinuxino-powering-and-sata-performance/
> > > 
> > > Gerhard
> > 
> > What I like about the RPi is that it is afaik still the cheapest board
> > available. It also has a huge community and it's not being run by a
> > company but a foundation with a non-profit goal.
> > For real-time audio purposes it is certainly not the best choice, but
> > then, what do you expect for that price? And what do you expect from a
> > device that is made for educational purposes, not real-time,
> > low-latency, pro audio?
> > 
> > Jeremy
> 
> Exactly!
> This is something people seem to forget when they complain that it can't do
> this, or that. They also forget that the founders sank their own personal money
> into this project at the start, called on favours and twisted arms.
> 

I still like the BeagleBone; it feels like a proper, well-engineered
development board.

The right tool for the synth job will eventually make itself known.

The Allwinner-based boards like the Banana and the Cubie and maybe
even the Udoo will be the right board for a synth; thanks for the tips.

The benchmark is my old EEE 1000. It had 2GB RAM and 1GHZ CPU. And
that was enough for me to gig with for years. So, as soon as a
1Ghz board with maybe 1GB RAM or 2GB RAM and well-designed dual-USB
connectivity comes down into the under-$100 range, there's the synth
I was hoping to build in 2007. I need only sit back and wait for
Moore's Law to do its magic. Eventually it will happen.

-ken

From looplog at gmail.com  Sat Aug  2 01:35:02 2014
From: looplog at gmail.com (michael noble)
Date: Sat, 2 Aug 2014 10:35:02 +0900
Subject: [LAU] JACK on Pi... almost working
In-Reply-To: <20140802005509.GC8368@q400a.mobile.restivo.org>
References: <20140726174251.GA3042@q400a.mobile.restivo.org>
 <53D3F70E.8010805@autostatic.com>
 <20140730075025.GE858@q400a.mobile.restivo.org>
 <201407301101.11298.gerhard.zintel@web.de>
 <53DB28AE.1080003@autostatic.com> <20140801070900.16dcf622@debian>
 <20140802005509.GC8368@q400a.mobile.restivo.org>
Message-ID: <CABf1g_xCsVffDs5V_2MyTKcoztUyfR7BBrtCK7=eAFmSDHPQVQ@mail.gmail.com>

On Sat, Aug 2, 2014 at 9:55 AM, Ken Restivo <ken at restivo.org> wrote:

> as soon as a
> 1Ghz board with maybe 1GB RAM or 2GB RAM and well-designed dual-USB
> connectivity comes down into the under-$100 range, there's the synth
> I was hoping to build in 2007
>

Not sure about USB performance, but you get a sub $100 board with a 1.7GHz
quad-core processor and 2GB RAM already. See here:

http://hardkernel.com/main/products/prdt_info.php?g_code=G138745696275

Their Pi compatible board is also interesting. For about $60 you can get
something smaller than the original Pi with a screen, battery
regulator/charger, extra USB ports, and additional ADC.

http://hardkernel.com/main/products/prdt_info.php?g_code=G140610189490
http://hardkernel.com/main/products/prdt_info.php?g_code=G140609436593
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From len at ovenwerks.net  Sat Aug  2 01:35:18 2014
From: len at ovenwerks.net (Len Ovens)
Date: Fri, 1 Aug 2014 18:35:18 -0700 (PDT)
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
Message-ID: <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>

On Fri, 1 Aug 2014, Ivan K wrote:

> Well, I have the s/pdif output of my
> USBDualTubePre microphone preamp plugged
> into the s/pdif input of my M-Audio 2496,
> though I have yet to be able to record?
> any sound.
>
> I am using Audacity.

ok.

> I have used Audacity successfully in the past
> to record though the analog inputs of the
> M-Audio 2496 (to digitize sound from cassettes?
> and vinyl) but Audacity's VU meters just?
> don't register anything from the microphone?
> going to the USBDualTubePre preamp to the?
> M-Audio 2496's s/pdif input.

:)

The s/pdif inputs are 9 and 10 normally (on the ice1712). Audacity will 
only record those if you record 10 tracks at a time. You can then split 
them into single tracks and then delete unused tracks... The mixer will 
not let you change this. If you are using pavucontrol, you can try 
setting the device to s/pdif only in the configuration tab... I have the 
devices turned off in pluse on my system so I can't check, I run pulse 
into jack all the time. Anyway look in the configuation tab for your 
device.

Or you can start Jackd first and then set audacity to talk to jack instead 
of alsa or pulse. You can use qjackctl to connect audacity to channel 9 
for recording. To see levels hit pause and record... infact you will have 
to do this before you connect ports with qjackctl.

This is why I would use pulse bridged to jackd so I can connect pulse to 
the right ports ahead of time. Then if audacity is using pulse it will 
"just work" (for me anyway).


--
Len Ovens
www.ovenwerks.net

From ivan_521521 at yahoo.com  Sat Aug  2 02:27:35 2014
From: ivan_521521 at yahoo.com (Ivan K)
Date: Fri, 1 Aug 2014 19:27:35 -0700
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
Message-ID: <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>

Len Ovens <len at ovenwerks.net> writes:
>?
> The s/pdif inputs are 9 and 10 normally (on the ice1712).

Damn good information to have.  Where is that sort
of information documented?

> Audacity will?
> only record those if you record 10 tracks at a time. [...]
??
> Or you can start Jackd first and then set audacity to talk to
> jack instead of alsa or pulse. You can use qjackctl to connect
> audacity to channel 9 for recording. To see levels hit pause
> and record... infact you will have to do this before you
> connect ports with qjackctl.

This seems most promising to me.

? I have audacity started up and in "pause/record mode".
? I have qjackctl started up. ?

But I don't really know where to go from here.  I have
hit the qjackctl "Connect" button, and see both the
"Audio" and "Alsa" tab.  Perhaps you direct me where to
go next or direct me to some documentation?

Thank you for your pointers!
??
> This is why I would use pulse bridged to jackd so I can
> connect pulse to the right ports ahead of time. Then if
> audacity is using pulse it will "just work" (for me anyway).

From len at ovenwerks.net  Sat Aug  2 04:58:46 2014
From: len at ovenwerks.net (Len Ovens)
Date: Fri, 1 Aug 2014 21:58:46 -0700 (PDT)
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
Message-ID: <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>

On Fri, 1 Aug 2014, Ivan K wrote:

> Len Ovens <len at ovenwerks.net> writes:
>> ?
>> The s/pdif inputs are 9 and 10 normally (on the ice1712).
>
> Damn good information to have.  Where is that sort
> of information documented?

Not really. But the ice1712 is the same chip used for the delta1010 which 
has 8 analog and one s/pdif pair. I just guessed... and kept trying inputs 
till I found it. I also have the USBDualTubePre and had to find out with 
my Delta66... I use inputs 1,2,3,4,9,10  ...

>> Or you can start Jackd first and then set audacity to talk to
>> jack instead of alsa or pulse. You can use qjackctl to connect
>> audacity to channel 9 for recording. To see levels hit pause
>> and record... infact you will have to do this before you
>> connect ports with qjackctl.
>
> This seems most promising to me.
>
> ? I have audacity started up and in "pause/record mode".
> ? I have qjackctl started up. ?
>
> But I don't really know where to go from here.  I have
> hit the qjackctl "Connect" button, and see both the
> "Audio" and "Alsa" tab.  Perhaps you direct me where to
> go next or direct me to some documentation?

Audacity uses the portaudio library to connect to Jackd.. so in the 
qjackctl connections window and the audio tab you should see PortAudio on 
the right side. Connect system/capture_9 to PortAudio/in_* (the * is a 
number that changes every time you hit record) and you should see the left 
meter showing audio. You may wish to disconnect whatever audacity auto 
connects first. The only documentation to look for portaudio is in the 
audacity setup screen where you set up jack it says:

Using: PortAudio V19 bla bla bla

If all you want to do is record, mhWaveEdit does play nicer with jack. 
You can set up which ports it connects to by default. The ports stay put 
from application start to finish. I don't know how the editing features 
are compared to Audacity though.



--
Len Ovens
www.ovenwerks.net

From gnome at hawaii.rr.com  Sat Aug  2 07:00:56 2014
From: gnome at hawaii.rr.com (david)
Date: Fri, 01 Aug 2014 21:00:56 -1000
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
Message-ID: <53DC8CA8.3030301@hawaii.rr.com>

On 08/01/2014 06:58 PM, Len Ovens wrote:
> On Fri, 1 Aug 2014, Ivan K wrote:
>
>> Len Ovens <len at ovenwerks.net> writes:
>>>
>>> The s/pdif inputs are 9 and 10 normally (on the ice1712).
>>
>> Damn good information to have.  Where is that sort
>> of information documented?
>
> Not really. But the ice1712 is the same chip used for the delta1010
> which has 8 analog and one s/pdif pair. I just guessed... and kept
> trying inputs till I found it. I also have the USBDualTubePre and had to
> find out with my Delta66... I use inputs 1,2,3,4,9,10  ...
>
>>> Or you can start Jackd first and then set audacity to talk to
>>> jack instead of alsa or pulse. You can use qjackctl to connect
>>> audacity to channel 9 for recording. To see levels hit pause
>>> and record... infact you will have to do this before you
>>> connect ports with qjackctl.
>>
>> This seems most promising to me.
>>
>>   I have audacity started up and in "pause/record mode".
>>   I have qjackctl started up.
>>
>> But I don't really know where to go from here.  I have
>> hit the qjackctl "Connect" button, and see both the
>> "Audio" and "Alsa" tab.  Perhaps you direct me where to
>> go next or direct me to some documentation?
>
> Audacity uses the portaudio library to connect to Jackd.. so in the
> qjackctl connections window and the audio tab you should see PortAudio
> on the right side. Connect system/capture_9 to PortAudio/in_* (the * is
> a number that changes every time you hit record) and you should see the
> left meter showing audio. You may wish to disconnect whatever audacity
> auto connects first. The only documentation to look for portaudio is in
> the audacity setup screen where you set up jack it says:
>
> Using: PortAudio V19 bla bla bla

My experience with recording via Audacity using JACK is that the 
PortAudio connection only appears in JACK while Audacity is actually 
using it (playing or recording). So I guess you start recording in 
Audacity, pause it, tweak the connections in JACK, then continue recording.

I record using a USB card that has only 2 inputs, so I've never had to 
tweak the connections JACK provides.

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From federicogalland at gmail.com  Sat Aug  2 08:56:37 2014
From: federicogalland at gmail.com (Fede)
Date: Sat, 2 Aug 2014 05:56:37 -0300
Subject: [LAU] Kxstudio RT kernel vs low latency
In-Reply-To: <CFFD392B.10CC9%hanaghan.osaudio@gmail.com>
References: <CABRv+93Fpy-QoNE1MH_-aejabjZGBJmCsufPryK12tift3ugqw@mail.gmail.com>
 <CFFD392B.10CC9%hanaghan.osaudio@gmail.com>
Message-ID: <20140802055637.7784c8a248c665560303bc63@gmail.com>

Hi all!

https://github.com/autch/pm_qos_req

I've just seen this. A kernel module to set the cpu_dma_latency to 0. I haven't tested it yet, though. It's probably more friendly than pmqos to set up in a musicmode script (that in my case removes some modules, ubinds usbs and bluetooth, and sets pci priorities).

It's for those of you with an older than sandybridge CPUs, newer should probably stick to the pstates hack mentioned by Joakim.

On a side note: Does anyone have any factual information on config_irq_time_accounting ? I noticed that it increases the cpu load, but am not sure if it benefits the audio/midi processing.
I'm working on a custom kernel config and trying to document how each parameter change affects audio and midi performance for later publishing.

Thanks!

From federicogalland at gmail.com  Sat Aug  2 09:20:28 2014
From: federicogalland at gmail.com (Fede)
Date: Sat, 2 Aug 2014 06:20:28 -0300
Subject: [LAU] drum synth
In-Reply-To: <20140719204028.786e8f5d@Scrapyard.lan>
References: <20140719002035.611de988@Scrapyard.lan>
 <53CA9053.8020108@autostatic.com>
 <20140719204028.786e8f5d@Scrapyard.lan>
Message-ID: <20140802062028.89b17dab42a32ed77dd62051@gmail.com>

I was looking for the best way to synthesize drums a few months ago, and while I tried various samplers and synths, I decided that my ultimate drum machine would be a tracker. The tracker interface cannot be beaten for the rhythmic purposes. Plus it has perfect timing since you don't depend on MIDI.

I'm currently using the hydrogen drumkit samples for that. Mainly the 909s which sound good enough.

Also, for the arrangements of my band I'm starting to use rosegarden
+linuxsampler, which I load GMaq's 4pc drumkit sf2.

Since the drum timbres don't usually change a lot during performance, this options plus some effects should be good enough (chibitracker comes with reverb and cheesetracker has built in ladspa).

If you want to make your own drum piece timbres, I'd recommend you to use audacity to draw your samples. You have access to all the LADSPA and nyquist plugins, and it's a really comfortable tool to work with short samples (I'm thinking of the envelope editor function which I love).

Good luck, and tell us the option you've taken.

From temps.jo at gmail.com  Sat Aug  2 12:13:58 2014
From: temps.jo at gmail.com (pierre jocelyn andre)
Date: Sat, 2 Aug 2014 14:13:58 +0200
Subject: [LAU] drum synth
In-Reply-To: <20140802062028.89b17dab42a32ed77dd62051@gmail.com>
References: <20140719002035.611de988@Scrapyard.lan>
 <53CA9053.8020108@autostatic.com>
 <20140719204028.786e8f5d@Scrapyard.lan>
 <20140802062028.89b17dab42a32ed77dd62051@gmail.com>
Message-ID: <CAJaXuwKCjkhqSFEJoEB1bHCWunU=tZmssfez=dm_byL7m=K0_g@mail.gmail.com>

Hello there,

again sorry for my poor English

my synthesizer creates sounds of drums
each sound weighs 16 bytes
we must forget audacity is too heavy, so much many bytes per sound
sounds produced by my synthesizer can not be encoded as encoding deforms,
as streaming deforms,

here you have some synthesizer sounds in wikimedia
http://commons.wikimedia.org/wiki/Special:Contributions/9temps
here you have video
https://www.youtube.com/watch?v=XwCeR5S8kHI
here you have some piano code
http://www.letime.net/vocale/lmmodel1jo.tar.gz

piano with drum
must be


I am currently working with the debian facile community to improve codes


best regards


2014-08-02 11:20 GMT+02:00 Fede <federicogalland at gmail.com>:

> I was looking for the best way to synthesize drums a few months ago, and
> while I tried various samplers and synths, I decided that my ultimate drum
> machine would be a tracker. The tracker interface cannot be beaten for the
> rhythmic purposes. Plus it has perfect timing since you don't depend on
> MIDI.
>
> I'm currently using the hydrogen drumkit samples for that. Mainly the 909s
> which sound good enough.
>
> Also, for the arrangements of my band I'm starting to use rosegarden
> +linuxsampler, which I load GMaq's 4pc drumkit sf2.
>
> Since the drum timbres don't usually change a lot during performance, this
> options plus some effects should be good enough (chibitracker comes with
> reverb and cheesetracker has built in ladspa).
>
> If you want to make your own drum piece timbres, I'd recommend you to use
> audacity to draw your samples. You have access to all the LADSPA and
> nyquist plugins, and it's a really comfortable tool to work with short
> samples (I'm thinking of the envelope editor function which I love).
>
> Good luck, and tell us the option you've taken.
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user
>
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From jonetsu at teksavvy.com  Sat Aug  2 14:26:43 2014
From: jonetsu at teksavvy.com (jonetsu at teksavvy.com)
Date: Sat, 2 Aug 2014 10:26:43 -0400
Subject: [LAU] Slowing down audio while keeping same pitch ?
In-Reply-To: <CAAAW3BWrX9BxL7c-Wou1D+W-Da50sWOpGnsZF31RFw_fdPL4EA@mail.gmail.com>
References: <20140729204713.07fcc456@mistral>
 <20140730071634.766508c9@shams.smbolton.com>
 <CAAAW3BWrX9BxL7c-Wou1D+W-Da50sWOpGnsZF31RFw_fdPL4EA@mail.gmail.com>
Message-ID: <20140802102643.4d3d9a7b@mistral>

On Wed, 30 Jul 2014 14:04:42 -0500,
Neil <djdualcore at gmail.com> wrote :

> Ardour
> Audacity
> Mixxx
> VLC Media Player
 
> Some of these won't do it directly on an MP3 but will happily do it
> on a copy in another format.

Ardour can play constant pitch slowed-down audio ?


From jonetsu at teksavvy.com  Sat Aug  2 14:33:05 2014
From: jonetsu at teksavvy.com (jonetsu at teksavvy.com)
Date: Sat, 2 Aug 2014 10:33:05 -0400
Subject: [LAU] Piecing together audio files
Message-ID: <20140802103305.20fcbb5c@mistral>

Hi !

  Thanks for the replies on slowing down audio.

  I would like to build an audio file using several other audio files,
preferably with some 2-4 seconds silence between them.  Actually this
is to upload to youtube a series of songs as one single file, that does
not need the use of a playlist.  The files are currently in wav (Ardour
export) and ogg.  Which tool can be used to construct such a file ?

  Very quickly I'd think of using Ardour to put all the files one after
another and then export the result, but that seems overkill, and
possibly taking more time to do. There has to be something more like a
'tool' to do that.  Suggestions ?

Cheers.

From jonetsu at teksavvy.com  Sat Aug  2 14:35:31 2014
From: jonetsu at teksavvy.com (jonetsu at teksavvy.com)
Date: Sat, 2 Aug 2014 10:35:31 -0400
Subject: [LAU] John Option: new song and new Adrour project
In-Reply-To: <1407301532080.16270@freeshell.de>
References: <53D8C430.1050109@fsfe.org>
	<1407301532080.16270@freeshell.de>
Message-ID: <20140802103531.2403a616@mistral>

On Wed, 30 Jul 2014 15:34:39 +0200 (CEST),
"F. Silvain" <silvain at freeshell.de> wrote :

> I love the sitar there. It adds a nice bit of spice. 

Entschuldigung, but this is funny.  A sitar, being from India, adding
spice !  :)))


From ralf.mardorf at rocketmail.com  Sat Aug  2 14:43:14 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sat, 02 Aug 2014 16:43:14 +0200
Subject: [LAU] [Bulk] Re: Slowing down audio while keeping same pitch ?
In-Reply-To: <20140802102643.4d3d9a7b@mistral>
References: <20140729204713.07fcc456@mistral>
 <20140730071634.766508c9@shams.smbolton.com>
 <CAAAW3BWrX9BxL7c-Wou1D+W-Da50sWOpGnsZF31RFw_fdPL4EA@mail.gmail.com>
 <20140802102643.4d3d9a7b@mistral>
Message-ID: <1406990594.23101.4.camel@rocketmail.com>

On Sat, 2014-08-02 at 10:26 -0400, jonetsu at teksavvy.com wrote:
> Ardour can play constant pitch slowed-down audio ?

[rocketmouse at archlinux ~]$ pacman -Qi ardour qtractor | grep Depends
Depends On     : [snip] soundtouch [snip]
Depends On     : [snip] rubberband [snip]

I never used one of it for track resize, but likely ardour, qtractor and
for sure other apps are able to do it, for sure by using much resources
and/or with lots of artifacts. I once tested rubberband independent of
resizing audio tracks, it needed much resources and add much artifacts.


From hodginson at gmail.com  Sat Aug  2 14:50:19 2014
From: hodginson at gmail.com (Chris Hogan)
Date: Sun, 3 Aug 2014 00:50:19 +1000
Subject: [LAU] Piecing together audio files
In-Reply-To: <20140802103305.20fcbb5c@mistral>
References: <20140802103305.20fcbb5c@mistral>
Message-ID: <CAP1x7Hk=4T99akooSPS+ux-a2yU-8rthfbzsU=ERy2Wgyjrtqg@mail.gmail.com>

The command line tool wavmerge will do this (although it only accepts .wav
files as input and they all need to be the same bitrate/frequency).

wavmerge -o output.wav input1.wav input2.wav input3.wav

There's no option to add silence between the merged files, so I usually
just make a .wav file containing the required amount of silence and merge
it in between the others:

wavmerge -o output.wav input1.wav silence.wav input2.wav silence.wav
input3.wav





On Sun, Aug 3, 2014 at 12:33 AM, jonetsu at teksavvy.com <jonetsu at teksavvy.com>
wrote:

> Hi !
>
>   Thanks for the replies on slowing down audio.
>
>   I would like to build an audio file using several other audio files,
> preferably with some 2-4 seconds silence between them.  Actually this
> is to upload to youtube a series of songs as one single file, that does
> not need the use of a playlist.  The files are currently in wav (Ardour
> export) and ogg.  Which tool can be used to construct such a file ?
>
>   Very quickly I'd think of using Ardour to put all the files one after
> another and then export the result, but that seems overkill, and
> possibly taking more time to do. There has to be something more like a
> 'tool' to do that.  Suggestions ?
>
> Cheers.
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user
>
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From idragosani at gmail.com  Sat Aug  2 14:50:05 2014
From: idragosani at gmail.com (Brett McCoy)
Date: Sat, 2 Aug 2014 10:50:05 -0400
Subject: [LAU] Piecing together audio files
In-Reply-To: <20140802103305.20fcbb5c@mistral>
References: <20140802103305.20fcbb5c@mistral>
Message-ID: <CAPNMgw4D0yL9AoX83uJ8L6ritaf_XjVnVhPvKp1sduPvVnjc-g@mail.gmail.com>

On Sat, Aug 2, 2014 at 10:33 AM, jonetsu at teksavvy.com
<jonetsu at teksavvy.com> wrote:

>   I would like to build an audio file using several other audio files,
> preferably with some 2-4 seconds silence between them.  Actually this
> is to upload to youtube a series of songs as one single file, that does
> not need the use of a playlist.  The files are currently in wav (Ardour
> export) and ogg.  Which tool can be used to construct such a file ?
>
>   Very quickly I'd think of using Ardour to put all the files one after
> another and then export the result, but that seems overkill, and
> possibly taking more time to do. There has to be something more like a
> 'tool' to do that.  Suggestions ?

If you are looking at a command-line tool for this, sox can concatenate files:

http://sox.sourceforge.net/Docs/Features

I believe there is a way you can insert silence at the beginning of a
file, but you'll have to read the man page on that.

On the other hand, Ardour is a perfectly fine way to do what you want
to do, especially if you need to do any last minute
leveling/compression or EQing to keep the tracks consistent

-- 
Brett W. McCoy -- http://www.brettwmccoy.com
------------------------------------------------------------------------
"In the rhythm of music a secret is hidden; If I were to divulge it,
it would overturn the world."
    -- Jelaleddin Rumi

From ralf.mardorf at rocketmail.com  Sat Aug  2 14:56:48 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sat, 02 Aug 2014 16:56:48 +0200
Subject: [LAU] [Bulk] Re: Slowing down audio while keeping same pitch ?
In-Reply-To: <1406990594.23101.4.camel@rocketmail.com>
References: <20140729204713.07fcc456@mistral>
 <20140730071634.766508c9@shams.smbolton.com>
 <CAAAW3BWrX9BxL7c-Wou1D+W-Da50sWOpGnsZF31RFw_fdPL4EA@mail.gmail.com>
 <20140802102643.4d3d9a7b@mistral> <1406990594.23101.4.camel@rocketmail.com>
Message-ID: <1406991408.23101.6.camel@rocketmail.com>

On Sat, 2014-08-02 at 16:43 +0200, Ralf Mardorf wrote:
> I once tested rubberband independent of resizing audio tracks, it
> needed much resources and add much artifacts.

I should explain that. I used rubberband as a plug in pitch shifter.
Track length wasn't resized and I bend the tuning that low, that the
tuning of the drums wasn't important anymore, IOW a few cents detuned
wasn't audible. The problem was less caused by the needed horsepower of
the computer, but more by some artefacts, that were very annoying
click-sounds, high volume spikes.


From ralf.mardorf at rocketmail.com  Sat Aug  2 15:04:07 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sat, 02 Aug 2014 17:04:07 +0200
Subject: [LAU] [Bulk] Re:  Piecing together audio files
In-Reply-To: <CAPNMgw4D0yL9AoX83uJ8L6ritaf_XjVnVhPvKp1sduPvVnjc-g@mail.gmail.com>
References: <20140802103305.20fcbb5c@mistral>
 <CAPNMgw4D0yL9AoX83uJ8L6ritaf_XjVnVhPvKp1sduPvVnjc-g@mail.gmail.com>
Message-ID: <1406991847.23101.8.camel@rocketmail.com>

On Sat, 2014-08-02 at 10:50 -0400, Brett McCoy wrote:
> On the other hand, Ardour is a perfectly fine

It's better to learn how to use one tool, even if it should be overkill
for some tasks, than to use several tools. As long as the user doesn't
need braille, I recommend to use a GUI based application, instead of
command line. Ardour 2 IMHO is a good tool for audio recording and
editing.



From federicogalland at gmail.com  Sat Aug  2 15:27:09 2014
From: federicogalland at gmail.com (Fede)
Date: Sat, 2 Aug 2014 12:27:09 -0300
Subject: [LAU] Kxstudio RT kernel vs low latency
In-Reply-To: <20140802055637.7784c8a248c665560303bc63@gmail.com>
References: <CABRv+93Fpy-QoNE1MH_-aejabjZGBJmCsufPryK12tift3ugqw@mail.gmail.com>
 <CFFD392B.10CC9%hanaghan.osaudio@gmail.com>
 <20140802055637.7784c8a248c665560303bc63@gmail.com>
Message-ID: <20140802122709.cdd0bdb92ade80d0a39d10b3@gmail.com>

> Hi all!
> 
> https://github.com/autch/pm_qos_req

Ok, already tried it a few hours. It allowed me to push my onboard hda_intel card with an IDT codec down to -p32 -n3 -r48000 with a youtube+alsa_in and amsynth test.
With everything open and producing output, the max dsp load was 14% on my T6400 (core2 2GHz).
Suspend, resume and hibernate worked fine and the module can be unloaded without freezing the laptop.

With my Fast Track Pro worked fine too in its ?hardware? lowest limit of -p64 -n2 -r48000 but I can't test suspend and resume since snd_usb_audio freezes the computer if the card is on when the computer goes to sleep (probably solveable through unloading the module before sleep with pm-utils).

MIDI tests coming.

What a great discovery.

Thanks to Joakim and everyone in this thread.

From ralf.mardorf at rocketmail.com  Sat Aug  2 15:25:22 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sat, 02 Aug 2014 17:25:22 +0200
Subject: [LAU] [Bulk] Re:  Kxstudio RT kernel vs low latency
In-Reply-To: <20140802122709.cdd0bdb92ade80d0a39d10b3@gmail.com>
References: <CABRv+93Fpy-QoNE1MH_-aejabjZGBJmCsufPryK12tift3ugqw@mail.gmail.com>
 <CFFD392B.10CC9%hanaghan.osaudio@gmail.com>
 <20140802055637.7784c8a248c665560303bc63@gmail.com>
 <20140802122709.cdd0bdb92ade80d0a39d10b3@gmail.com>
Message-ID: <1406993122.23101.10.camel@rocketmail.com>

On Sat, 2014-08-02 at 12:27 -0300, Fede wrote:
> I can't test suspend and resume

*?*

Don't suspend and resume real-time audio sessions using jack.



From len at ovenwerks.net  Sat Aug  2 16:16:56 2014
From: len at ovenwerks.net (Len Ovens)
Date: Sat, 2 Aug 2014 09:16:56 -0700 (PDT)
Subject: [LAU] Piecing together audio files
In-Reply-To: <20140802103305.20fcbb5c@mistral>
References: <20140802103305.20fcbb5c@mistral>
Message-ID: <alpine.DEB.2.02.1408020851000.23832@scott.cbbs.org>

On Sat, 2 Aug 2014, jonetsu at teksavvy.com wrote:

>  Very quickly I'd think of using Ardour to put all the files one after
> another and then export the result, but that seems overkill, and

Yes, it is designed to be that way. The idea being that it can also 
always be the right tool. There is something to be said for using a tool 
that is comfortable from being used a lot. On the other hand, the unix 
idea of using a string of tools that just do one thing well is still 
valid.

Having a machine with much more memory and power than needed makes the use 
of larger do-everything software valid if the user can spend less time 
learning and more time using. Using small tools can be more versatile 
though, because the user can think up new ways to connect things that the 
do-everything software may not be able to do.

--
Len Ovens
www.ovenwerks.net


From csanchezgs at gmail.com  Sat Aug  2 16:46:44 2014
From: csanchezgs at gmail.com (Carlos sanchiavedraz)
Date: Sat, 2 Aug 2014 18:46:44 +0200
Subject: [LAU] [Bulk] Re: Linux Poetry: X-run blues
In-Reply-To: <1406921036.14205.1.camel@rocketmail.com>
References: <53DBBC66.3090408@gmail.com>
 <CALVwFW-r8m8hiUOes-JLfHiKDpvoPacT6KqdQ3gsnj=NyM7Cvw@mail.gmail.com>
 <1406921036.14205.1.camel@rocketmail.com>
Message-ID: <CA+O6s5xU+qZpBVfXGvGdsSOmhqSUvEKQeob1fe72KoO5fDoJSA@mail.gmail.com>

El 01/08/2014 21:24, "Ralf Mardorf" <ralf.mardorf at rocketmail.com> escribi?:
>
> On Fri, 2014-08-01 at 13:45 -0400, Egor Sanin wrote:
> > On 8/1/14, Kees van Veen <kees.vanveen at gmail.com> wrote:
> [snip]
> > Excellent!
> > Totally waiting to hear this performed at next year's LAC sound night.
>
> The lyrics seem to fit to the twelve-bar blues chord progressions and
> the repeated parts of the lyrics seem to point to the typical whiny kind
> of speech song, so the melody unlikely will be a surprise.
>
> Hey Kees, where you goin' with that sledgehammer in your hands?
> Hey Kees, I said where you goin' with that sledgehammer in your hands?
>
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

LOL!, so good, Kees.

Waiting for that version on an good ol' style, with pops and cracks,
singing with a broke sad bluesy voice.
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From csanchezgs at gmail.com  Sat Aug  2 17:03:50 2014
From: csanchezgs at gmail.com (Carlos sanchiavedraz)
Date: Sat, 2 Aug 2014 19:03:50 +0200
Subject: [LAU] [Bulk] Re: music software for kids
In-Reply-To: <CAG+zz5mXJYz5BrR2sacZ13Xn_b4ssVg3xqvVuqvnQL7GOTBU3Q@mail.gmail.com>
References: <CAG+zz5nrr_+OmZFqtGw0Ye+=RYTrNNzQcVoSJODFUavSjnpxmQ@mail.gmail.com>
 <20140726090235.2691fcc9@eeyore.mozart.uni-klu.ac.at>
 <1406368461.5104.7.camel@rocketmail.com>
 <CAG+zz5mXJYz5BrR2sacZ13Xn_b4ssVg3xqvVuqvnQL7GOTBU3Q@mail.gmail.com>
Message-ID: <CA+O6s5w2LDpT8qk6oA=sh5=_L11J_EV=DRT0e0ee3MgnTWrmrQ@mail.gmail.com>

El 28/07/2014 16:43, "Brian Sorahan" <brian at gospacecraft.com> escribi?:
>
> Thanks for the suggestions everyone!
>
> @Ralf I halfway agree that you can learn the basics of more
fully-featured software by fiddling around. I also think that "babyish"
music software can be a lot of fun, and there is nothing wrong with a child
using it. I definitely plan on steering my 2 boys towards the powerful
programs (that I think are really fun), but if they have tons of fun making
cheesy little ditties with toy software (like I did when I was young), then
I'm all for it.
>
>
>
> On Sat, Jul 26, 2014 at 4:54 AM, Ralf Mardorf <ralf.mardorf at rocketmail.com>
wrote:
>>
>>
>> > Brian Sorahan <brian at gospacecraft.com> wrote:
>> > > Are there any music applications for linux that would be suitable for
>> > > a 10 year old
>>
>> I assume that the 10 year old child isn't retarded, so the best thing
>> IMO is to really learn how to make music, instead of using babyish music
>> software. Learning without teaching is possible, just by fiddling
>> around. A short explanation how to use Qtractor or a simple MIDI app,
>> adding a virtual keyboard or a real MIDI keyboard, using fluidsynth or
>> similar shouldn't expect too much of a 10 year old, even when there's
>> not much interest, just a little bit fiddling around is wanted. Reading
>> documentations isn't needed, I'm a dyslexic, so I know whereof I speak.
>> Btw. I worked a lot with children around that age, my last job working
>> with those children was from beginning of this year until the the
>> beginning of this month (school hols).
>>
>> 2 Cents,
>> Ralf
>>
>> _______________________________________________
>> Linux-audio-user mailing list
>> Linux-audio-user at lists.linuxaudio.org
>> http://lists.linuxaudio.org/listinfo/linux-audio-user
>
>
>
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user
>

Just to add a bit, more in the music programming side, FWIW, there's Sonic
PI:
http://www.cl.cam.ac.uk/projects/raspberrypi/sonicpi/
https://duckduckgo.com/lite/sonic+pi+children

Although I think it's better tinkering and experimenting, touch and feel
and hear, as it's already said.
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From federicogalland at gmail.com  Sat Aug  2 17:52:30 2014
From: federicogalland at gmail.com (Fede)
Date: Sat, 2 Aug 2014 14:52:30 -0300
Subject: [LAU] A short story: from zero to recording the drums in a
 budget
In-Reply-To: <20140722082321.GA6051@linuxaudio.org>
References: <CABVJJr-u3vrVrfQNRLM99FPBPEHCOOAHV7stB85z1LO5Ziqu1g@mail.gmail.com>
 <53CD6F72.20903@gareus.org>
 <CABVJJr8AbQeZc-2WyLwmU8FN5HnMZSc787jUtPYesiz3JM3qeQ@mail.gmail.com>
 <20140722082321.GA6051@linuxaudio.org>
Message-ID: <20140802145230.432a6bdfafa34581781b9634@gmail.com>

> I've got a similar setup: 512M, Arch + windowmaker,
> but can't get A3 to run on it in any usable way - it
> just eats too much memory. Any tricks ??
> 
> Ciao,
I feel awkward giving advice to a knowledged dsp programmer, but maybe this will help you.

I have an old desktop with a celeron 440 2GHz CPU, and 1GB of RAM (which run on 512MB for a month before the upgrade) which performs quite good after doing the following:

 Replace systemd with something lighter, like minirc https://github.com/hut/minirc It might be painful depending on what services you run on startup. If you stick to defaults, you're mostly covered.
Of course you can't load any extra services, but if you use http://dwm.suckless.org/ you're cutting a lot. I've once compiled X myself and it worked better and was softer on memory, but I don't have benchmarks. Also, disable unused ttys.

dwm is extremely handy to me. It's got 3 default view modes, monocle (apps cover whole screen), tile, and floating. All of this under 2000 lines of C code. With patches you get almost any thinkable split mode.
The key bindings are vi-based so it's easy to learn them if you used the editor. Everything is customizable before compiling.

I should mention that I also custom config my kernels and strip any foreign thing. That makes my atom N270 netbook use 90mb of ram before starting X (which I do through xinit with an .xinitrc).

I don't think one can go further than this without cutting needed functionality. But I'm sure qtractor would run on your P4 this way.

Bye!




From ralf.mardorf at rocketmail.com  Sat Aug  2 18:22:26 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sat, 02 Aug 2014 20:22:26 +0200
Subject: [LAU] [Bulk] Re: A short story: from zero to recording the
 drums in a budget
In-Reply-To: <20140802145230.432a6bdfafa34581781b9634@gmail.com>
References: <CABVJJr-u3vrVrfQNRLM99FPBPEHCOOAHV7stB85z1LO5Ziqu1g@mail.gmail.com>
 <53CD6F72.20903@gareus.org>
 <CABVJJr8AbQeZc-2WyLwmU8FN5HnMZSc787jUtPYesiz3JM3qeQ@mail.gmail.com>
 <20140722082321.GA6051@linuxaudio.org>
 <20140802145230.432a6bdfafa34581781b9634@gmail.com>
Message-ID: <1407003746.23101.12.camel@rocketmail.com>

On Sat, 2014-08-02 at 14:52 -0300, Fede wrote:
> if you use http://dwm.suckless.org/ you're cutting a lot.

I'm using JWM. There was a comparison between WMs and DEs that claimed
that JWM is the most light weights WM. I don't care about that claim,
I'm using it, just because I like it.

> I should mention that I also custom config my kernels and strip any foreign thing.

I run my own kernels too, but I don't care for a light weight kernel, so
I never tested  localmodconfig  , but it perhaps is interesting for
those who care about small kernels.


From nando at ccrma.Stanford.EDU  Sat Aug  2 18:25:50 2014
From: nando at ccrma.Stanford.EDU (Fernando Lopez-Lezcano)
Date: Sat, 02 Aug 2014 11:25:50 -0700
Subject: [LAU] Linux Poetry: X-run blues
In-Reply-To: <53DBBC66.3090408@gmail.com>
References: <53DBBC66.3090408@gmail.com>
Message-ID: <53DD2D2E.8020608@localhost>

On 08/01/2014 09:12 AM, Kees van Veen wrote:
> Hi,
>
> See below for my contribution to Linux audio poetry, reflections of
> personal struggle ;-)
>
> Thanks to all Linux audio developers and supporters for all the great
> software, I hope no one feels offended or ridiculed by the lyrics.

I hope we can soon hear a rendition of this song by Dave Philips and his 
guitar (with obvious audio dropouts and glitches caused by xruns, of 
course)...

:-)
-- Fernando


> X-run blues (battle hymn of Linux audio newbies)
> ----------------
> I woke up early one morning, I got me the X-run blues
> I woke up early one morning, I got me the X-run blues
> Must've missed a couple of warnings, and I don't have any clues
>
> I was all set to record, enter the X-run blues
> I was really ready to record, enter the X-run blues
> Could be the cpufreq, or them funny IRQs
>
> I started googling for advice on lower latency
> I started googling for advice on lower latency
> It said 'Compile yourself a realtime kernel, son, and you just wait and
> see'
>
> I switched to Ardour III, I paid my Ardour dues
> I switched to Ardour III, I paid my Ardour dues
> Tried all of the Jack flavors, can't lose the X-run blues
>
> I developed 'X' paranoia, Xubuntu's where I'm at
> I developed 'X' paranoia, the worst I ever had
> Would switching to Fedora get me F-runs instead
>
> Lately I don't care too much, though there's lots of things to check
> Right now I don't care too much, lots of settings still to check
> I like the vintage pops and crackles, just like a Robert Johnson track
>
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user


From ivan_521521 at yahoo.com  Sat Aug  2 20:55:15 2014
From: ivan_521521 at yahoo.com (Ivan K)
Date: Sat, 2 Aug 2014 13:55:15 -0700
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <53DC8CA8.3030301@hawaii.rr.com>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
Message-ID: <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>



Thanks to everyone whom responded.


Len Ovens <len at ovenwerks.net> writes:
>?
> Audacity uses the portaudio library to connect to Jackd.. so in the?
> qjackctl connections window and the audio tab you should see PortAudio on?
> the right side.

and

"David Jones" <gnome at hawaii.rr.com> writes:
>?
> My experience with recording via Audacity using JACK is that the?
> PortAudio connection only appears in JACK while Audacity is actually?
> using it (playing or recording). So I guess you start recording in?
> Audacity, pause it, tweak the connections in JACK, then continue recording.

Well, no matter if Audacity is recording or playing,
all I see in qjackctl, [Connect], "Audio" tab is:

== ==   Readable Clients/
== ==   Output Ports
== ==   - system
== == ? ? capture_1
== == ? ? capture_2

on the right, and then on the left, I see:

== ==   Writable Clients/
== ==   Input Ports
== ==   - system
== == ? ? playback_1
== == ? ? playback_2
== == ? ? playback_3
== == ? ? playback_4
== == ? ? playback_5
== == ? ? playback_6
== == ? ? playback_7
== == ? ? playback_8

I see don't see a mention of audacity or portaudio.
Also, does "system" refer to my M-Audio card or
the integrated audio on the motherboard?
If system does refer to my M-Audio card, I don't
see anything in the under Readable Clients/Output Ports
that distinguishes between the s/pdif input and the
ADC input.

On the other hand, when I run mhWaveEdit, as suggested by
Mr. Ovens, I do see the word "mhwe" listed both in?
the Readable Clients/Output Ports and Writable Clients/Input Ports.


At this point, I can continue to work with mhWaveEdit
as audacity appears to be problematic.
----

So running mhWaveEdit, I still cannot create any audio
files with actual sound, or see mhWaveEdit's VU meters
show any volume.

So, does "system" in the connect window of qjackctl
refer to my M-Audio card?

Thank you for your suggestions!

From james at jwm-art.net  Sat Aug  2 21:16:32 2014
From: james at jwm-art.net (James Morris)
Date: Sat, 2 Aug 2014 22:16:32 +0100
Subject: [LAU] drum synth
In-Reply-To: <CAJaXuwKCjkhqSFEJoEB1bHCWunU=tZmssfez=dm_byL7m=K0_g@mail.gmail.com>
References: <20140719002035.611de988@Scrapyard.lan>
 <53CA9053.8020108@autostatic.com>
 <20140719204028.786e8f5d@Scrapyard.lan>
 <20140802062028.89b17dab42a32ed77dd62051@gmail.com>
 <CAJaXuwKCjkhqSFEJoEB1bHCWunU=tZmssfez=dm_byL7m=K0_g@mail.gmail.com>
Message-ID: <20140802221632.1351df82@Scrapyard.lan>

On Sat, 2 Aug 2014 14:13:58 +0200
pierre jocelyn andre <temps.jo at gmail.com> wrote:

> Hello there,
> 
> again sorry for my poor English
> 
> my synthesizer creates sounds of drums
> each sound weighs 16 bytes
> we must forget audacity is too heavy, so much many bytes per sound
> sounds produced by my synthesizer can not be encoded as encoding
> deforms, as streaming deforms,
> 
> here you have some synthesizer sounds in wikimedia
> http://commons.wikimedia.org/wiki/Special:Contributions/9temps
> here you have video
> https://www.youtube.com/watch?v=XwCeR5S8kHI

I hear no drums in that video. I watched one of the others and found
the sounds interesting (as I commented there) but didn't hear any
drums, and it's not obvious what's going on, how can you change the
synthesis parameters for instance? Must the code be edited and
recompiled?


> here you have some piano code
> http://www.letime.net/vocale/lmmodel1jo.tar.gz

Segmentation faults as soon as I hit any button.


> 
> piano with drum
> must be
> 
> 
> I am currently working with the debian facile community to improve
> codes

I've just looked at MaFenetre.cpp and it's over 9000 lines!

You need to stop the lazy coding habit of copying and pasting the same
code over and over again with minor modifications. I see 1438 lines
beginning at line 6580 with the same code copied and pasted over and
over again, and then again another piece of code copied and pasted for
the last 1116 lines of the file.

You need to turn much of MaFenetre.cpp into functions to reduce the
redundancy.

Learn how to use structured programming.

Start setting minimal standards for your code and adhere to them.

But I'm only self taught too, and not a pro either, so what do I know.

james.


> 
> 
> best regards
> 
> 
> 2014-08-02 11:20 GMT+02:00 Fede <federicogalland at gmail.com>:
> 
> > I was looking for the best way to synthesize drums a few months
> > ago, and while I tried various samplers and synths, I decided that
> > my ultimate drum machine would be a tracker. The tracker interface
> > cannot be beaten for the rhythmic purposes. Plus it has perfect
> > timing since you don't depend on MIDI.
> >
> > I'm currently using the hydrogen drumkit samples for that. Mainly
> > the 909s which sound good enough.
> >
> > Also, for the arrangements of my band I'm starting to use rosegarden
> > +linuxsampler, which I load GMaq's 4pc drumkit sf2.
> >
> > Since the drum timbres don't usually change a lot during
> > performance, this options plus some effects should be good enough
> > (chibitracker comes with reverb and cheesetracker has built in
> > ladspa).
> >
> > If you want to make your own drum piece timbres, I'd recommend you
> > to use audacity to draw your samples. You have access to all the
> > LADSPA and nyquist plugins, and it's a really comfortable tool to
> > work with short samples (I'm thinking of the envelope editor
> > function which I love).
> >
> > Good luck, and tell us the option you've taken.
> > _______________________________________________
> > Linux-audio-user mailing list
> > Linux-audio-user at lists.linuxaudio.org
> > http://lists.linuxaudio.org/listinfo/linux-audio-user
> >


From james at jwm-art.net  Sat Aug  2 21:30:00 2014
From: james at jwm-art.net (James Morris)
Date: Sat, 2 Aug 2014 22:30:00 +0100
Subject: [LAU] drum synth
In-Reply-To: <20140802062028.89b17dab42a32ed77dd62051@gmail.com>
References: <20140719002035.611de988@Scrapyard.lan>
 <53CA9053.8020108@autostatic.com>
 <20140719204028.786e8f5d@Scrapyard.lan>
 <20140802062028.89b17dab42a32ed77dd62051@gmail.com>
Message-ID: <20140802223000.660c7528@Scrapyard.lan>

On Sat, 2 Aug 2014 06:20:28 -0300
Fede <federicogalland at gmail.com> wrote:

> I was looking for the best way to synthesize drums a few months ago,
> and while I tried various samplers and synths, I decided that my
> ultimate drum machine would be a tracker. The tracker interface
> cannot be beaten for the rhythmic purposes. Plus it has perfect
> timing since you don't depend on MIDI.
> 
> I'm currently using the hydrogen drumkit samples for that. Mainly the
> 909s which sound good enough.
> 
> Also, for the arrangements of my band I'm starting to use rosegarden
> +linuxsampler, which I load GMaq's 4pc drumkit sf2.
> 
> Since the drum timbres don't usually change a lot during performance,
> this options plus some effects should be good enough (chibitracker
> comes with reverb and cheesetracker has built in ladspa).
> 
> If you want to make your own drum piece timbres, I'd recommend you to
> use audacity to draw your samples. You have access to all the LADSPA
> and nyquist plugins, and it's a really comfortable tool to work with
> short samples (I'm thinking of the envelope editor function which I
> love).
> 
> Good luck, and tell us the option you've taken.

I resorted to downloading drum samples from Freesound directly into
Ardour and used that to create a two bar loop but never got any
further. Just don't know how I used to find the time... 

James.

From temps.jo at gmail.com  Sat Aug  2 21:47:47 2014
From: temps.jo at gmail.com (pierre jocelyn andre)
Date: Sat, 2 Aug 2014 23:47:47 +0200
Subject: [LAU] drum synth
In-Reply-To: <20140802221632.1351df82@Scrapyard.lan>
References: <20140719002035.611de988@Scrapyard.lan>
 <53CA9053.8020108@autostatic.com>
 <20140719204028.786e8f5d@Scrapyard.lan>
 <20140802062028.89b17dab42a32ed77dd62051@gmail.com>
 <CAJaXuwKCjkhqSFEJoEB1bHCWunU=tZmssfez=dm_byL7m=K0_g@mail.gmail.com>
 <20140802221632.1351df82@Scrapyard.lan>
Message-ID: <CAJaXuwJdO3EA4Xm--fuDj1_cYahxVgZcNVWDt3mvoZ987zJScw@mail.gmail.com>

Hi,
I mainly work on the laws of sound, not much on how to code,

but tux know coding and look at here
http://chezlefab.net/share/8e3b1648a0a683334186/lmmodeljo-fab.tar.gz
it-is the future, I just have to correct the wav converter

there are many drums
Try the code
A drum is simply a strong decrease in amplitude accompanied by a change in
length by 5 forehead (30 fronts enough)

"Segmentation faults , "that is if you try out linux

Best regard


2014-08-02 23:16 GMT+02:00 James Morris <james at jwm-art.net>:

> On Sat, 2 Aug 2014 14:13:58 +0200
> pierre jocelyn andre <temps.jo at gmail.com> wrote:
>
> > Hello there,
> >
> > again sorry for my poor English
> >
> > my synthesizer creates sounds of drums
> > each sound weighs 16 bytes
> > we must forget audacity is too heavy, so much many bytes per sound
> > sounds produced by my synthesizer can not be encoded as encoding
> > deforms, as streaming deforms,
> >
> > here you have some synthesizer sounds in wikimedia
> > http://commons.wikimedia.org/wiki/Special:Contributions/9temps
> > here you have video
> > https://www.youtube.com/watch?v=XwCeR5S8kHI
>
> I hear no drums in that video. I watched one of the others and found
> the sounds interesting (as I commented there) but didn't hear any
> drums, and it's not obvious what's going on, how can you change the
> synthesis parameters for instance? Must the code be edited and
> recompiled?
>
>
> > here you have some piano code
> > http://www.letime.net/vocale/lmmodel1jo.tar.gz
>
> Segmentation faults as soon as I hit any button.
>
>
> >
> > piano with drum
> > must be
> >
> >
> > I am currently working with the debian facile community to improve
> > codes
>
> I've just looked at MaFenetre.cpp and it's over 9000 lines!
>
> You need to stop the lazy coding habit of copying and pasting the same
> code over and over again with minor modifications. I see 1438 lines
> beginning at line 6580 with the same code copied and pasted over and
> over again, and then again another piece of code copied and pasted for
> the last 1116 lines of the file.
>
> You need to turn much of MaFenetre.cpp into functions to reduce the
> redundancy.
>
> Learn how to use structured programming.
>
> Start setting minimal standards for your code and adhere to them.
>
> But I'm only self taught too, and not a pro either, so what do I know.
>
> james.
>
>
> >
> >
> > best regards
> >
> >
> > 2014-08-02 11:20 GMT+02:00 Fede <federicogalland at gmail.com>:
> >
> > > I was looking for the best way to synthesize drums a few months
> > > ago, and while I tried various samplers and synths, I decided that
> > > my ultimate drum machine would be a tracker. The tracker interface
> > > cannot be beaten for the rhythmic purposes. Plus it has perfect
> > > timing since you don't depend on MIDI.
> > >
> > > I'm currently using the hydrogen drumkit samples for that. Mainly
> > > the 909s which sound good enough.
> > >
> > > Also, for the arrangements of my band I'm starting to use rosegarden
> > > +linuxsampler, which I load GMaq's 4pc drumkit sf2.
> > >
> > > Since the drum timbres don't usually change a lot during
> > > performance, this options plus some effects should be good enough
> > > (chibitracker comes with reverb and cheesetracker has built in
> > > ladspa).
> > >
> > > If you want to make your own drum piece timbres, I'd recommend you
> > > to use audacity to draw your samples. You have access to all the
> > > LADSPA and nyquist plugins, and it's a really comfortable tool to
> > > work with short samples (I'm thinking of the envelope editor
> > > function which I love).
> > >
> > > Good luck, and tell us the option you've taken.
> > > _______________________________________________
> > > Linux-audio-user mailing list
> > > Linux-audio-user at lists.linuxaudio.org
> > > http://lists.linuxaudio.org/listinfo/linux-audio-user
> > >
>
>
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From jonetsu at teksavvy.com  Sat Aug  2 21:59:15 2014
From: jonetsu at teksavvy.com (jonetsu at teksavvy.com)
Date: Sat, 2 Aug 2014 17:59:15 -0400
Subject: [LAU] Linux Poetry: X-run blues
In-Reply-To: <53DD2D2E.8020608@localhost>
References: <53DBBC66.3090408@gmail.com>
	<53DD2D2E.8020608@localhost>
Message-ID: <20140802175915.294a496b@mistral>

On Sat, 02 Aug 2014 11:25:50 -0700,
Fernando Lopez-Lezcano <nando at ccrma.Stanford.EDU> wrote :

> I hope we can soon hear a rendition of this song by Dave Philips and
> his guitar (with obvious audio dropouts and glitches caused by xruns,
> of course)...

And if there are none, they'll be simulated !

:)

From ralf.mardorf at rocketmail.com  Sat Aug  2 22:01:28 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 03 Aug 2014 00:01:28 +0200
Subject: [LAU] [Bulk] Re:  drum synth
In-Reply-To: <20140802223000.660c7528@Scrapyard.lan>
References: <20140719002035.611de988@Scrapyard.lan>
 <53CA9053.8020108@autostatic.com> <20140719204028.786e8f5d@Scrapyard.lan>
 <20140802062028.89b17dab42a32ed77dd62051@gmail.com>
 <20140802223000.660c7528@Scrapyard.lan>
Message-ID: <1407016888.23101.24.camel@rocketmail.com>

On Sat, 2014-08-02 at 22:30 +0100, James Morris wrote:
> I resorted to downloading drum samples from Freesound directly into
> Ardour and used that to create a two bar loop but never got any
> further. Just don't know how I used to find the time...

When I use CR-78 samples or similar, I visit a Wiki or another source,
e.g. https://en.wikipedia.org/wiki/Roland_CR-78 , then I use such an
example loop and copy the same rhythm by a MIDI file to play the samples
unison with the audio example loop. I tune the samples, correct loudness
and panning until it fits to the original drum machine. This is the
basic I need, most of the times the samples provided by soundfonts are
completely detuned. Anyway, once the samples are consistent, e.g. the
tuning of the kick fits to the snare, I detune the drums to fit to my
composition, I edit attack times etc., but I don't remember that I ever
have done this using Linux software ;). IMO a good basic for synth
sounds are _real_ Roland synth drum samples, TR series etc., but it's
important that the sounds are original, not detuned, not wrong loudness
relations, the samples need to be consistent, work with each other.
Sure, I will detune and edit attack and decay times etc. to fit to my
songs, but IMO the problem often is that there isn't a good basic to do
that. Most soundfonts that claim to provide CR, TR etc. are a PITA and
need much correction, before you can start to edit them to fit to your
needs.


From jonetsu at teksavvy.com  Sat Aug  2 22:09:30 2014
From: jonetsu at teksavvy.com (jonetsu at teksavvy.com)
Date: Sat, 2 Aug 2014 18:09:30 -0400
Subject: [LAU] Using .sfz files (was:Embertone Friedlander Violin on
 Linux)
In-Reply-To: <CABao80kiJf+JgQvv+8df4savNomQ2q6s7G9wYe1ruy=VyvjbFg@mail.gmail.com>
References: <CABao80=H5DCNoOoZ7ezqc1zP0_WPXvSW0B5gHYGsCYizxQpJAQ@mail.gmail.com>
 <539D5E65.3040003@web.de> <20140713203924.55a51f67@mistral>
 <CABao80kiJf+JgQvv+8df4savNomQ2q6s7G9wYe1ruy=VyvjbFg@mail.gmail.com>
Message-ID: <20140802180930.6d5cdb53@mistral>

On Sun, 13 Jul 2014 22:26:22 -0300,
Aiyumi Moriya <aiyumi.br at gmail.com> wrote :

> 2014-07-13 21:39 GMT-03:00, jonetsu at teksavvy.com
> <jonetsu at teksavvy.com>:
> > Is it possible to load this in qsynth ?  I tried acousbass.zip and
> > it only has (apart from wav files) a .sfz file while qsynth filters
> > on .sf2 or .SF2 files.  Renaming the .sfz to .sf2 won't do.
> 
> No. Qsynth is a frontend to Fluidsynth and accepts only SF2. I think
> you're confusing it with Qsampler, a frontend to Linuxsampler, which
> does read SFZ. That's probably the one you want.

Ah ! I do not use LinuxSampler, so far.

???????


From gnome at hawaii.rr.com  Sat Aug  2 22:13:54 2014
From: gnome at hawaii.rr.com (david)
Date: Sat, 02 Aug 2014 12:13:54 -1000
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
Message-ID: <53DD62A2.5070502@hawaii.rr.com>

On 08/02/2014 10:55 AM, Ivan K wrote:

> Thanks to everyone whom responded.
>
> Len Ovens <len at ovenwerks.net> writes:
>>
>> Audacity uses the portaudio library to connect to Jackd.. so in the
>> qjackctl connections window and the audio tab you should see PortAudio on
>> the right side.
>
> and
>
> "David Jones" <gnome at hawaii.rr.com> writes:
>>
>> My experience with recording via Audacity using JACK is that the
>> PortAudio connection only appears in JACK while Audacity is actually
>> using it (playing or recording). So I guess you start recording in
>> Audacity, pause it, tweak the connections in JACK, then continue recording.
>
> Well, no matter if Audacity is recording or playing,
> all I see in qjackctl, [Connect], "Audio" tab is:
>
> == ==   Readable Clients/
> == ==   Output Ports
> == ==   - system
> == ==     capture_1
> == ==     capture_2
>
> on the right, and then on the left, I see:
>
> == ==   Writable Clients/
> == ==   Input Ports
> == ==   - system
> == ==     playback_1
> == ==     playback_2
> == ==     playback_3
> == ==     playback_4
> == ==     playback_5
> == ==     playback_6
> == ==     playback_7
> == ==     playback_8
>
> I see don't see a mention of audacity or portaudio.

Hmmm, just on off-chance ... you do have Audacity set to use JACK? 
Sorry, maybe not being very helpful here.

> Also, does "system" refer to my M-Audio card or
> the integrated audio on the motherboard?

I believe that JACK's "system" refers to the audio card you've set JACK 
to use.

> If system does refer to my M-Audio card, I don't
> see anything in the under Readable Clients/Output Ports
> that distinguishes between the s/pdif input and the
> ADC input.

Didn't someone else in this thread already post about that? That SPDIF 
was 7 & 8?

> On the other hand, when I run mhWaveEdit, as suggested by
> Mr. Ovens, I do see the word "mhwe" listed both in
> the Readable Clients/Output Ports and Writable Clients/Input Ports.

A lot of audio programs are smarter about working with JACK than Audacity.

> At this point, I can continue to work with mhWaveEdit
> as audacity appears to be problematic.
> ----
>
> So running mhWaveEdit, I still cannot create any audio
> files with actual sound, or see mhWaveEdit's VU meters
> show any volume.
>
> So, does "system" in the connect window of qjackctl
> refer to my M-Audio card?
>
> Thank you for your suggestions!

Sorry, I've run out of clues. On my simple setup here, I started JACK, 
started Audacity, set Audacity to use JACK (it defaults to ALSA on my 
laptop); hit pause, then record, then unpaused. A PortAudio connection 
appeared in JACK's writable ports window. It was already connected to my 
in ports. I paused recording in Audacity and checked JACK; connection 
still there. When I hit Stop button in Audacity, the PortAudio 
connection disappeared from JACK.

I have jackdmp (JACK2) 1.9.10, Audacity 2.0.5.

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From jonetsu at teksavvy.com  Sat Aug  2 22:14:59 2014
From: jonetsu at teksavvy.com (jonetsu at teksavvy.com)
Date: Sat, 2 Aug 2014 18:14:59 -0400
Subject: [LAU] Hatsune Miku
Message-ID: <20140802181459.74ac6fc4@mistral>

Hi !

  Is it possible to use the Hatsune Miku/Megune Luka software in
Linux ?  Anyone doing this ?

Cheers.

From ralf.mardorf at rocketmail.com  Sat Aug  2 22:49:37 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 03 Aug 2014 00:49:37 +0200
Subject: [LAU] [Bulk] Re:  Record from M-Audio 2496 S/PDIF input
In-Reply-To: <53DD62A2.5070502@hawaii.rr.com>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
 <53DD62A2.5070502@hawaii.rr.com>
Message-ID: <1407019777.23101.38.camel@rocketmail.com>

On Sat, 2014-08-02 at 12:13 -1000, david wrote:
> Sorry, I've run out of clues.

The order to start the applications is very important.
First the OP needs to start jackd, e.g. by QjackCtl.
When jackd is running, the OP needs to start Audacity.
By Audacity's select box "Jack Audio Connector", not "ALSA" has to be
selected.
After pushing Pause and Record,the PortAudio input ports are visible.

alsamixer, envy24control or mudita24 is needed to rout the S/PDIF
signal. I remember that once a blind user had issues to get it working
for his Envy24 card, when using alsamixer, but using a GUI there
shouldn't be an issue.

I searched old mails to find the answer how the alsamixer issue was
solved ...

[rocketmouse at archlinux ~]$ top
  PID USER      PR  NI    VIRT    RES    SHR S  %CPU %MEM     TIME+ COMMAND                                      
23101 rocketm+  20   0 3165168 363776  74424 S 120.7  9.6  15:36.66 evolution

... opening a mail in Evolution's search folder might take a few hours. I'm not kidding.


From james at jwm-art.net  Sat Aug  2 23:07:23 2014
From: james at jwm-art.net (James Morris)
Date: Sun, 3 Aug 2014 00:07:23 +0100
Subject: [LAU] drum synth
In-Reply-To: <CAJaXuwJdO3EA4Xm--fuDj1_cYahxVgZcNVWDt3mvoZ987zJScw@mail.gmail.com>
References: <20140719002035.611de988@Scrapyard.lan>
 <53CA9053.8020108@autostatic.com>
 <20140719204028.786e8f5d@Scrapyard.lan>
 <20140802062028.89b17dab42a32ed77dd62051@gmail.com>
 <CAJaXuwKCjkhqSFEJoEB1bHCWunU=tZmssfez=dm_byL7m=K0_g@mail.gmail.com>
 <20140802221632.1351df82@Scrapyard.lan>
 <CAJaXuwJdO3EA4Xm--fuDj1_cYahxVgZcNVWDt3mvoZ987zJScw@mail.gmail.com>
Message-ID: <20140803000723.2a29673e@Scrapyard.lan>

On Sat, 2 Aug 2014 23:47:47 +0200
pierre jocelyn andre <temps.jo at gmail.com> wrote:

> Hi,
> I mainly work on the laws of sound, not much on how to code,
> 
> but tux know coding and look at here
> http://chezlefab.net/share/8e3b1648a0a683334186/lmmodeljo-fab.tar.gz
> it-is the future, I just have to correct the wav converter
> 
> there are many drums
> Try the code


I just tried this code. It does not seg-fault like the other code. I'm
not convinced it's the future as you mention. It doesn't sound like
you've worked that much on the laws of sound, each button or key press
produces the exactly same sound, no difference, the only way to produce
any difference is to press keys/buttons rapidly to layer the sound.


> A drum is simply a strong decrease in amplitude accompanied by a
> change in length by 5 forehead (30 fronts enough)

That theory could be applied to a pure DC signal but I'm not sure
if you might need another forehead or not.


> 
> "Segmentation faults , "that is if you try out linux
> 

Well I'm harldy going to rush out and buy Windows just so I can test
your program to see if it meets my needs am I?

And why does your program do horrible things to my terminal?

James.


> Best regard
> 
> 
> 2014-08-02 23:16 GMT+02:00 James Morris <james at jwm-art.net>:
> 
> > On Sat, 2 Aug 2014 14:13:58 +0200
> > pierre jocelyn andre <temps.jo at gmail.com> wrote:
> >
> > > Hello there,
> > >
> > > again sorry for my poor English
> > >
> > > my synthesizer creates sounds of drums
> > > each sound weighs 16 bytes
> > > we must forget audacity is too heavy, so much many bytes per sound
> > > sounds produced by my synthesizer can not be encoded as encoding
> > > deforms, as streaming deforms,
> > >
> > > here you have some synthesizer sounds in wikimedia
> > > http://commons.wikimedia.org/wiki/Special:Contributions/9temps
> > > here you have video
> > > https://www.youtube.com/watch?v=XwCeR5S8kHI
> >
> > I hear no drums in that video. I watched one of the others and found
> > the sounds interesting (as I commented there) but didn't hear any
> > drums, and it's not obvious what's going on, how can you change the
> > synthesis parameters for instance? Must the code be edited and
> > recompiled?
> >
> >
> > > here you have some piano code
> > > http://www.letime.net/vocale/lmmodel1jo.tar.gz
> >
> > Segmentation faults as soon as I hit any button.
> >
> >
> > >
> > > piano with drum
> > > must be
> > >
> > >
> > > I am currently working with the debian facile community to improve
> > > codes
> >
> > I've just looked at MaFenetre.cpp and it's over 9000 lines!
> >
> > You need to stop the lazy coding habit of copying and pasting the
> > same code over and over again with minor modifications. I see 1438
> > lines beginning at line 6580 with the same code copied and pasted
> > over and over again, and then again another piece of code copied
> > and pasted for the last 1116 lines of the file.
> >
> > You need to turn much of MaFenetre.cpp into functions to reduce the
> > redundancy.
> >
> > Learn how to use structured programming.
> >
> > Start setting minimal standards for your code and adhere to them.
> >
> > But I'm only self taught too, and not a pro either, so what do I
> > know.
> >
> > james.
> >
> >
> > >
> > >
> > > best regards
> > >
> > >
> > > 2014-08-02 11:20 GMT+02:00 Fede <federicogalland at gmail.com>:
> > >
> > > > I was looking for the best way to synthesize drums a few months
> > > > ago, and while I tried various samplers and synths, I decided
> > > > that my ultimate drum machine would be a tracker. The tracker
> > > > interface cannot be beaten for the rhythmic purposes. Plus it
> > > > has perfect timing since you don't depend on MIDI.
> > > >
> > > > I'm currently using the hydrogen drumkit samples for that.
> > > > Mainly the 909s which sound good enough.
> > > >
> > > > Also, for the arrangements of my band I'm starting to use
> > > > rosegarden +linuxsampler, which I load GMaq's 4pc drumkit sf2.
> > > >
> > > > Since the drum timbres don't usually change a lot during
> > > > performance, this options plus some effects should be good
> > > > enough (chibitracker comes with reverb and cheesetracker has
> > > > built in ladspa).
> > > >
> > > > If you want to make your own drum piece timbres, I'd recommend
> > > > you to use audacity to draw your samples. You have access to
> > > > all the LADSPA and nyquist plugins, and it's a really
> > > > comfortable tool to work with short samples (I'm thinking of
> > > > the envelope editor function which I love).
> > > >
> > > > Good luck, and tell us the option you've taken.
> > > > _______________________________________________
> > > > Linux-audio-user mailing list
> > > > Linux-audio-user at lists.linuxaudio.org
> > > > http://lists.linuxaudio.org/listinfo/linux-audio-user
> > > >
> >
> >


From ralf.mardorf at rocketmail.com  Sat Aug  2 23:10:49 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 03 Aug 2014 01:10:49 +0200
Subject: [LAU] [Bulk] Re:  Record from M-Audio 2496 S/PDIF input
In-Reply-To: <1407019777.23101.38.camel@rocketmail.com>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
 <53DD62A2.5070502@hawaii.rr.com> <1407019777.23101.38.camel@rocketmail.com>
Message-ID: <1407021049.23101.44.camel@rocketmail.com>

On Sun, 2014-08-03 at 00:49 +0200, Ralf Mardorf wrote:
> ... opening a mail in Evolution's search folder might take a few
> hours. I'm not kidding.

Here are the results:

http://lists.linuxaudio.org/pipermail/linux-audio-user/2012-December/088655.html
http://lists.linuxaudio.org/pipermail/linux-audio-user/2012-December/088656.html
http://lists.linuxaudio.org/pipermail/linux-audio-user/2012-December/088657.html
http://lists.linuxaudio.org/pipermail/linux-audio-user/2012-December/088658.html

Hth,
Ralf




From murks at tuxfamily.org  Sat Aug  2 23:32:22 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Sun, 3 Aug 2014 01:32:22 +0200
Subject: [LAU] Linux Poetry: X-run blues
In-Reply-To: <20140802175915.294a496b@mistral>
References: <53DBBC66.3090408@gmail.com> <53DD2D2E.8020608@localhost>
 <20140802175915.294a496b@mistral>
Message-ID: <20140803013222.1a284feb@eeyore.mozart.uni-klu.ac.at>

On Sat, 2 Aug 2014 17:59:15 -0400
"jonetsu at teksavvy.com" <jonetsu at teksavvy.com> wrote:

> On Sat, 02 Aug 2014 11:25:50 -0700,
> Fernando Lopez-Lezcano <nando at ccrma.Stanford.EDU> wrote :
> 
> > I hope we can soon hear a rendition of this song by Dave Philips and
> > his guitar (with obvious audio dropouts and glitches caused by
> > xruns, of course)...
> 
> And if there are none, they'll be simulated !
> 
> :)

I insist on real xruns, no fake crap, not just 'authentic', I want the
real deal.

Regards,
Philipp

From ralf.mardorf at rocketmail.com  Sat Aug  2 23:42:43 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 03 Aug 2014 01:42:43 +0200
Subject: [LAU] [Bulk] Re:  Linux Poetry: X-run blues
In-Reply-To: <20140803013222.1a284feb@eeyore.mozart.uni-klu.ac.at>
References: <53DBBC66.3090408@gmail.com> <53DD2D2E.8020608@localhost>
 <20140802175915.294a496b@mistral>
 <20140803013222.1a284feb@eeyore.mozart.uni-klu.ac.at>
Message-ID: <1407022963.29357.1.camel@rocketmail.com>

On Sun, 2014-08-03 at 01:32 +0200, Philipp ?berbacher wrote:
> I insist on real xruns, no fake crap, not just 'authentic', I want the
> real deal.

If the workstation is too good to produce xruns, I recommend to launch
several instances of jamin to increase DSP load.


From ken at restivo.org  Sat Aug  2 23:49:14 2014
From: ken at restivo.org (Ken Restivo)
Date: Sat, 2 Aug 2014 16:49:14 -0700
Subject: [LAU] Linux Poetry: X-run blues
In-Reply-To: <53DBBC66.3090408@gmail.com>
References: <53DBBC66.3090408@gmail.com>
Message-ID: <20140802234914.GA1241@tf101>

On Fri, Aug 01, 2014 at 06:12:22PM +0200, Kees van Veen wrote:
> Hi,
> 
> See below for my contribution to Linux audio poetry, reflections of
> personal struggle ;-)
> 
> Thanks to all Linux audio developers and supporters for all the
> great software, I hope no one feels offended or ridiculed by the
> lyrics.
> 

That's hilarious. Indeed, I nominate Dave Phillips to
blues that up.

BTW, I think it might be the second song named after Xruns:
http://www.restivo.org/blog/archives/xruns

> 
> X-run blues (battle hymn of Linux audio newbies)
> ----------------
> I woke up early one morning, I got me the X-run blues
> I woke up early one morning, I got me the X-run blues
> Must've missed a couple of warnings, and I don't have any clues
> 
> I was all set to record, enter the X-run blues
> I was really ready to record, enter the X-run blues
> Could be the cpufreq, or them funny IRQs
> 
> I started googling for advice on lower latency
> I started googling for advice on lower latency
> It said 'Compile yourself a realtime kernel, son, and you just wait
> and see'
> 
> I switched to Ardour III, I paid my Ardour dues
> I switched to Ardour III, I paid my Ardour dues
> Tried all of the Jack flavors, can't lose the X-run blues
> 
> I developed 'X' paranoia, Xubuntu's where I'm at
> I developed 'X' paranoia, the worst I ever had
> Would switching to Fedora get me F-runs instead
> 
> Lately I don't care too much, though there's lots of things to check
> Right now I don't care too much, lots of settings still to check
> I like the vintage pops and crackles, just like a Robert Johnson track
> 
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

From aiyumi.br at gmail.com  Sun Aug  3 01:20:53 2014
From: aiyumi.br at gmail.com (Aiyumi Moriya)
Date: Sat, 2 Aug 2014 22:20:53 -0300
Subject: [LAU] Hatsune Miku
In-Reply-To: <20140802181459.74ac6fc4@mistral>
References: <20140802181459.74ac6fc4@mistral>
Message-ID: <CABao80==WewrBjFUJs+Ko2-Hi0cgXj+OMKK65T=QVztUUKefBw@mail.gmail.com>

2014-08-02 19:14 GMT-03:00, jonetsu at teksavvy.com <jonetsu at teksavvy.com>:
> Hi !
>
>   Is it possible to use the Hatsune Miku/Megune Luka software in
> Linux ?  Anyone doing this ?
>
> Cheers.

I know that UTAU works on Wine. See:

http://utauarianna.altervista.org/tutorials/utau-on-linux-wine-how-to/

I already tried that, and it worked.

As for VOCALOID, I didn't try, but VOCALOID2 seems to work.

https://appdb.winehq.org/objectManager.php?sClass=application&iId=7554

I don't know about VOCALOID3.

You could test using one of the software trials, like Nekomura Iroha
for VOCALOID2

https://www.ah-soft.com/trial/v2-iroha.html

and Maika for VOCALOID3

http://www.voctro-vocaloid.com/maika

Besides Iroha, AHS has other trials, like Yuki and Kiyoteru for V2 and
Yukari and (recently) Zunko for V3.

https://www.ah-soft.com/trial/menu2.html


-- 
____________________

Blog: http://aiyumi.warpstar.net/

From ralf.mardorf at rocketmail.com  Sun Aug  3 01:40:16 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 03 Aug 2014 03:40:16 +0200
Subject: [LAU] S/PDIF - Was: Record from M-Audio 2496 S/PDIF input
Message-ID: <1407030016.29357.6.camel@rocketmail.com>

Hi,

since S/PDIF is consumer class people are used to connect S/PDIF devices
and I assume everything will work without the need to care about
something else.

S/PDIF usually isn't used for audio production, this might explain why
people tend to forget to set up the clock source. AFAIK there's no need
to care about it, when using stand alone consumer devices.

I suspect that sync very often is the culprit, when S/PDIF used with
prosumer or pro-audio devices doesn't work.

Since I'm not using S/PDIF I ask somebody using it, to write a Linux
audio Wiki that mentions to set up the master clock manually.

2 Cents,
Ralf


From simonzwise at gmail.com  Sun Aug  3 02:30:30 2014
From: simonzwise at gmail.com (Simon Wise)
Date: Sun, 03 Aug 2014 12:30:30 +1000
Subject: [LAU] JACK on Pi... almost working
In-Reply-To: <53DB28AE.1080003@autostatic.com>
References: <20140726174251.GA3042@q400a.mobile.restivo.org>
 <53D3F70E.8010805@autostatic.com>
 <20140730075025.GE858@q400a.mobile.restivo.org>
 <201407301101.11298.gerhard.zintel@web.de> <53DB28AE.1080003@autostatic.com>
Message-ID: <53DD9EC6.2080006@gmail.com>

On 01/08/14 15:42, Jeremy Jongepier wrote:

> What I like about the RPi is that it is afaik still the cheapest board
> available. It also has a huge community and it's not being run by a
> company but a foundation with a non-profit goal.
> For real-time audio purposes it is certainly not the best choice, but
> then, what do you expect for that price? And what do you expect from a
> device that is made for educational purposes, not real-time,
> low-latency, pro audio?

yes, and by doing it they have probably pushed others to try harder, and be more 
open (in fact sometimes a lot more open than Broadcom has allowed for the 
raspberries) ... which is also a good thing. But it is also important to let 
people know what they are capable of, and when something more powerful is required.

Simon

From ivan_521521 at yahoo.com  Sun Aug  3 03:37:22 2014
From: ivan_521521 at yahoo.com (Ivan K)
Date: Sat, 2 Aug 2014 20:37:22 -0700
Subject: [LAU] S/PDIF - Was: Record from M-Audio 2496 S/PDIF input
In-Reply-To: <1407030016.29357.6.camel@rocketmail.com>
References: <1407030016.29357.6.camel@rocketmail.com>
Message-ID: <1407037042.3411.YahooMailNeo@web122606.mail.ne1.yahoo.com>

"David Jones" <gnome at hawaii.rr.com> writes:
>
> Hmmm, just on off-chance ... you do have Audacity set
> to use JACK? ?Sorry, maybe not being very helpful here.

Oh, but you were.  Audacity has a "audio host" dropdown
box that was set to ALSA.  Once Jack is started, Jack is
available in?this box, and I have selected it.

> > Also, does "system" refer to my M-Audio card or
> > the integrated audio on the motherboard?
>?
> I believe that JACK's "system" refers to the audio card
> you've set JACK?to use.

My {$HOME}/.jackdrc  file reads:

? ?/usr/bin/jackd -dalsa -dhw:0 -r44100 -p16 -n2

aplay and arecord both list my M-Audio card as "card 0"

I can find no where in the qjackctl "Setup" menu
were this is set though.

> > If system does refer to my M-Audio card, I don't
> > see anything in the under Readable Clients/Output Ports
> > that distinguishes between the s/pdif input and the
> > ADC input.
>?
> Didn't someone else in this thread already post about
> that? That SPDIF?was 7 & 8?

That person said the it was 9 and 10.

> A lot of audio programs are smarter about working
> with JACK than Audacity.

Good to know.

> Sorry, I've run out of clues. On my simple setup here, I
> started JACK,?started Audacity, set Audacity to use JACK
> (it defaults to ALSA on my laptop); hit pause, then?
> record, then unpaused. A PortAudio connection?
> appeared in JACK's writable ports window.

And now, as I have audacity paused in record mode, the
PortAudio?connection now appears under the "Writable
Clients" part of the qjackctl "Connections" window.  The
number _under_ the "PortAudio" entry varies though.
Sometimes it reads "in_4", sometimes "in_5", etc.

I actually see a signal in Audacity's VU meter now, however,
the volume of this signal is pretty constant and has nothing
to do with?whether anyone is speaking in the microphone
or not.

And I have yet to record anything that is spoken into
the microphone.

Ralf Mardorf <ralf.mardorf at rocketmail.com> writes:
>
> alsamixer, envy24control or mudita24 is needed to rout the S/PDIF
> signal. I remember that once a blind user had issues to get
> it working?for his Envy24 card, when using alsamixer, but
> using a GUI there shouldn't be an issue.
[...]
>?
> http://lists.linuxaudio.org/pipermail/linux-audio-user/2012-December/088655.html
> http://lists.linuxaudio.org/pipermail/linux-audio-user/2012-December/088656.html
> http://lists.linuxaudio.org/pipermail/linux-audio-user/2012-December/088657.html
> http://lists.linuxaudio.org/pipermail/linux-audio-user/2012-December/088658.html

and thank you for this information.  I used envy24control to?
set "Master clock" to s/pdif (rather than a sample rate.

The VU meters of Envy24control now _do_ show a signal that
corresponds to when someone is actually speaking in the microphone.

However, the VU meters in __mhWaveEdit__ do not show a signal (I have
been toggling back and fourth between Audacity and mhWaveEdit).

I think I am close.

Maybe there is something I still need to set with Envy24Control?

Thanks again for the help.

From temps.jo at gmail.com  Sun Aug  3 05:11:35 2014
From: temps.jo at gmail.com (pierre jocelyn andre)
Date: Sun, 3 Aug 2014 07:11:35 +0200
Subject: [LAU] drum synth
In-Reply-To: <20140803000723.2a29673e@Scrapyard.lan>
References: <20140719002035.611de988@Scrapyard.lan>
 <53CA9053.8020108@autostatic.com>
 <20140719204028.786e8f5d@Scrapyard.lan>
 <20140802062028.89b17dab42a32ed77dd62051@gmail.com>
 <CAJaXuwKCjkhqSFEJoEB1bHCWunU=tZmssfez=dm_byL7m=K0_g@mail.gmail.com>
 <20140802221632.1351df82@Scrapyard.lan>
 <CAJaXuwJdO3EA4Xm--fuDj1_cYahxVgZcNVWDt3mvoZ987zJScw@mail.gmail.com>
 <20140803000723.2a29673e@Scrapyard.lan>
Message-ID: <CAJaXuwLDcPO7k1d8h2+YeThE6TWqBTamGktHhWOnvSzuSFABgw@mail.gmail.com>

Hi,
for the last code, I say :
in work,
we need to refine the movement in the tables




2014-08-03 1:07 GMT+02:00 James Morris <james at jwm-art.net>:

> On Sat, 2 Aug 2014 23:47:47 +0200
> pierre jocelyn andre <temps.jo at gmail.com> wrote:
>
> > Hi,
> > I mainly work on the laws of sound, not much on how to code,
> >
> > but tux know coding and look at here
> > http://chezlefab.net/share/8e3b1648a0a683334186/lmmodeljo-fab.tar.gz
> > it-is the future, I just have to correct the wav converter
> >
> > there are many drums
> > Try the code
>
>
> I just tried this code. It does not seg-fault like the other code. I'm
> not convinced it's the future as you mention. It doesn't sound like
> you've worked that much on the laws of sound, each button or key press
> produces the exactly same sound, no difference, the only way to produce
> any difference is to press keys/buttons rapidly to layer the sound.
>
>
> > A drum is simply a strong decrease in amplitude accompanied by a
> > change in length by 5 forehead (30 fronts enough)
>
> That theory could be applied to a pure DC signal but I'm not sure
> if you might need another forehead or not.
>
>
> >
> > "Segmentation faults , "that is if you try out linux
> >
>
> Well I'm harldy going to rush out and buy Windows just so I can test
> your program to see if it meets my needs am I?
>
> And why does your program do horrible things to my terminal?
>
> James.
>
>
> > Best regard
> >
> >
> > 2014-08-02 23:16 GMT+02:00 James Morris <james at jwm-art.net>:
> >
> > > On Sat, 2 Aug 2014 14:13:58 +0200
> > > pierre jocelyn andre <temps.jo at gmail.com> wrote:
> > >
> > > > Hello there,
> > > >
> > > > again sorry for my poor English
> > > >
> > > > my synthesizer creates sounds of drums
> > > > each sound weighs 16 bytes
> > > > we must forget audacity is too heavy, so much many bytes per sound
> > > > sounds produced by my synthesizer can not be encoded as encoding
> > > > deforms, as streaming deforms,
> > > >
> > > > here you have some synthesizer sounds in wikimedia
> > > > http://commons.wikimedia.org/wiki/Special:Contributions/9temps
> > > > here you have video
> > > > https://www.youtube.com/watch?v=XwCeR5S8kHI
> > >
> > > I hear no drums in that video. I watched one of the others and found
> > > the sounds interesting (as I commented there) but didn't hear any
> > > drums, and it's not obvious what's going on, how can you change the
> > > synthesis parameters for instance? Must the code be edited and
> > > recompiled?
> > >
> > >
> > > > here you have some piano code
> > > > http://www.letime.net/vocale/lmmodel1jo.tar.gz
> > >
> > > Segmentation faults as soon as I hit any button.
> > >
> > >
> > > >
> > > > piano with drum
> > > > must be
> > > >
> > > >
> > > > I am currently working with the debian facile community to improve
> > > > codes
> > >
> > > I've just looked at MaFenetre.cpp and it's over 9000 lines!
> > >
> > > You need to stop the lazy coding habit of copying and pasting the
> > > same code over and over again with minor modifications. I see 1438
> > > lines beginning at line 6580 with the same code copied and pasted
> > > over and over again, and then again another piece of code copied
> > > and pasted for the last 1116 lines of the file.
> > >
> > > You need to turn much of MaFenetre.cpp into functions to reduce the
> > > redundancy.
> > >
> > > Learn how to use structured programming.
> > >
> > > Start setting minimal standards for your code and adhere to them.
> > >
> > > But I'm only self taught too, and not a pro either, so what do I
> > > know.
> > >
> > > james.
> > >
> > >
> > > >
> > > >
> > > > best regards
> > > >
> > > >
> > > > 2014-08-02 11:20 GMT+02:00 Fede <federicogalland at gmail.com>:
> > > >
> > > > > I was looking for the best way to synthesize drums a few months
> > > > > ago, and while I tried various samplers and synths, I decided
> > > > > that my ultimate drum machine would be a tracker. The tracker
> > > > > interface cannot be beaten for the rhythmic purposes. Plus it
> > > > > has perfect timing since you don't depend on MIDI.
> > > > >
> > > > > I'm currently using the hydrogen drumkit samples for that.
> > > > > Mainly the 909s which sound good enough.
> > > > >
> > > > > Also, for the arrangements of my band I'm starting to use
> > > > > rosegarden +linuxsampler, which I load GMaq's 4pc drumkit sf2.
> > > > >
> > > > > Since the drum timbres don't usually change a lot during
> > > > > performance, this options plus some effects should be good
> > > > > enough (chibitracker comes with reverb and cheesetracker has
> > > > > built in ladspa).
> > > > >
> > > > > If you want to make your own drum piece timbres, I'd recommend
> > > > > you to use audacity to draw your samples. You have access to
> > > > > all the LADSPA and nyquist plugins, and it's a really
> > > > > comfortable tool to work with short samples (I'm thinking of
> > > > > the envelope editor function which I love).
> > > > >
> > > > > Good luck, and tell us the option you've taken.
> > > > > _______________________________________________
> > > > > Linux-audio-user mailing list
> > > > > Linux-audio-user at lists.linuxaudio.org
> > > > > http://lists.linuxaudio.org/listinfo/linux-audio-user
> > > > >
> > >
> > >
>
>
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From temps.jo at gmail.com  Sun Aug  3 05:12:41 2014
From: temps.jo at gmail.com (pierre jocelyn andre)
Date: Sun, 3 Aug 2014 07:12:41 +0200
Subject: [LAU] drum synth
In-Reply-To: <CAJaXuwLDcPO7k1d8h2+YeThE6TWqBTamGktHhWOnvSzuSFABgw@mail.gmail.com>
References: <20140719002035.611de988@Scrapyard.lan>
 <53CA9053.8020108@autostatic.com>
 <20140719204028.786e8f5d@Scrapyard.lan>
 <20140802062028.89b17dab42a32ed77dd62051@gmail.com>
 <CAJaXuwKCjkhqSFEJoEB1bHCWunU=tZmssfez=dm_byL7m=K0_g@mail.gmail.com>
 <20140802221632.1351df82@Scrapyard.lan>
 <CAJaXuwJdO3EA4Xm--fuDj1_cYahxVgZcNVWDt3mvoZ987zJScw@mail.gmail.com>
 <20140803000723.2a29673e@Scrapyard.lan>
 <CAJaXuwLDcPO7k1d8h2+YeThE6TWqBTamGktHhWOnvSzuSFABgw@mail.gmail.com>
Message-ID: <CAJaXuwJjWBt57AeAYxPwpAvWWSJVQNf7SSkg_kUhJynzc8LrXg@mail.gmail.com>

Maybe the code is not pretty. But who can write a note of music with 16
bytes? That an application has a few kilobytes notes that synthesizes musical
instrument?

What is the most beautiful car in the world if it does not roll, or music
request megabytes to work?

I prefer forward through the codes, those who want their children to play,
I said or he may find application at critical beliefs parfais a code, I do not
have to answer.

best regards


2014-08-03 7:11 GMT+02:00 pierre jocelyn andre <temps.jo at gmail.com>:

> Hi,
> for the last code, I say :
>  in work,
> we need to refine the movement in the tables
>
>
>
>
> 2014-08-03 1:07 GMT+02:00 James Morris <james at jwm-art.net>:
>
> On Sat, 2 Aug 2014 23:47:47 +0200
>> pierre jocelyn andre <temps.jo at gmail.com> wrote:
>>
>> > Hi,
>> > I mainly work on the laws of sound, not much on how to code,
>> >
>> > but tux know coding and look at here
>> > http://chezlefab.net/share/8e3b1648a0a683334186/lmmodeljo-fab.tar.gz
>> > it-is the future, I just have to correct the wav converter
>> >
>> > there are many drums
>> > Try the code
>>
>>
>> I just tried this code. It does not seg-fault like the other code. I'm
>> not convinced it's the future as you mention. It doesn't sound like
>> you've worked that much on the laws of sound, each button or key press
>> produces the exactly same sound, no difference, the only way to produce
>> any difference is to press keys/buttons rapidly to layer the sound.
>>
>>
>> > A drum is simply a strong decrease in amplitude accompanied by a
>> > change in length by 5 forehead (30 fronts enough)
>>
>> That theory could be applied to a pure DC signal but I'm not sure
>> if you might need another forehead or not.
>>
>>
>> >
>> > "Segmentation faults , "that is if you try out linux
>> >
>>
>> Well I'm harldy going to rush out and buy Windows just so I can test
>> your program to see if it meets my needs am I?
>>
>> And why does your program do horrible things to my terminal?
>>
>> James.
>>
>>
>> > Best regard
>> >
>> >
>> > 2014-08-02 23:16 GMT+02:00 James Morris <james at jwm-art.net>:
>> >
>> > > On Sat, 2 Aug 2014 14:13:58 +0200
>> > > pierre jocelyn andre <temps.jo at gmail.com> wrote:
>> > >
>> > > > Hello there,
>> > > >
>> > > > again sorry for my poor English
>> > > >
>> > > > my synthesizer creates sounds of drums
>> > > > each sound weighs 16 bytes
>> > > > we must forget audacity is too heavy, so much many bytes per sound
>> > > > sounds produced by my synthesizer can not be encoded as encoding
>> > > > deforms, as streaming deforms,
>> > > >
>> > > > here you have some synthesizer sounds in wikimedia
>> > > > http://commons.wikimedia.org/wiki/Special:Contributions/9temps
>> > > > here you have video
>> > > > https://www.youtube.com/watch?v=XwCeR5S8kHI
>> > >
>> > > I hear no drums in that video. I watched one of the others and found
>> > > the sounds interesting (as I commented there) but didn't hear any
>> > > drums, and it's not obvious what's going on, how can you change the
>> > > synthesis parameters for instance? Must the code be edited and
>> > > recompiled?
>> > >
>> > >
>> > > > here you have some piano code
>> > > > http://www.letime.net/vocale/lmmodel1jo.tar.gz
>> > >
>> > > Segmentation faults as soon as I hit any button.
>> > >
>> > >
>> > > >
>> > > > piano with drum
>> > > > must be
>> > > >
>> > > >
>> > > > I am currently working with the debian facile community to improve
>> > > > codes
>> > >
>> > > I've just looked at MaFenetre.cpp and it's over 9000 lines!
>> > >
>> > > You need to stop the lazy coding habit of copying and pasting the
>> > > same code over and over again with minor modifications. I see 1438
>> > > lines beginning at line 6580 with the same code copied and pasted
>> > > over and over again, and then again another piece of code copied
>> > > and pasted for the last 1116 lines of the file.
>> > >
>> > > You need to turn much of MaFenetre.cpp into functions to reduce the
>> > > redundancy.
>> > >
>> > > Learn how to use structured programming.
>> > >
>> > > Start setting minimal standards for your code and adhere to them.
>> > >
>> > > But I'm only self taught too, and not a pro either, so what do I
>> > > know.
>> > >
>> > > james.
>> > >
>> > >
>> > > >
>> > > >
>> > > > best regards
>> > > >
>> > > >
>> > > > 2014-08-02 11:20 GMT+02:00 Fede <federicogalland at gmail.com>:
>> > > >
>> > > > > I was looking for the best way to synthesize drums a few months
>> > > > > ago, and while I tried various samplers and synths, I decided
>> > > > > that my ultimate drum machine would be a tracker. The tracker
>> > > > > interface cannot be beaten for the rhythmic purposes. Plus it
>> > > > > has perfect timing since you don't depend on MIDI.
>> > > > >
>> > > > > I'm currently using the hydrogen drumkit samples for that.
>> > > > > Mainly the 909s which sound good enough.
>> > > > >
>> > > > > Also, for the arrangements of my band I'm starting to use
>> > > > > rosegarden +linuxsampler, which I load GMaq's 4pc drumkit sf2.
>> > > > >
>> > > > > Since the drum timbres don't usually change a lot during
>> > > > > performance, this options plus some effects should be good
>> > > > > enough (chibitracker comes with reverb and cheesetracker has
>> > > > > built in ladspa).
>> > > > >
>> > > > > If you want to make your own drum piece timbres, I'd recommend
>> > > > > you to use audacity to draw your samples. You have access to
>> > > > > all the LADSPA and nyquist plugins, and it's a really
>> > > > > comfortable tool to work with short samples (I'm thinking of
>> > > > > the envelope editor function which I love).
>> > > > >
>> > > > > Good luck, and tell us the option you've taken.
>> > > > > _______________________________________________
>> > > > > Linux-audio-user mailing list
>> > > > > Linux-audio-user at lists.linuxaudio.org
>> > > > > http://lists.linuxaudio.org/listinfo/linux-audio-user
>> > > > >
>> > >
>> > >
>>
>>
>
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From massimo at fsfe.org  Sun Aug  3 06:40:08 2014
From: massimo at fsfe.org (Massimo Barbieri)
Date: Sun, 03 Aug 2014 08:40:08 +0200
Subject: [LAU] John Option: new song and new Adrour project
In-Reply-To: <1407301532080.16270@freeshell.de>
References: <53D8C430.1050109@fsfe.org> <1407301532080.16270@freeshell.de>
Message-ID: <53DDD948.4030106@fsfe.org>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256

Il 30/07/2014 15:34, F. Silvain ha scritto:
> good song! I love the sitar there. It adds a nice bit of spice.
> Good mix as well, if a little sparse, but that sounds intentional. 
> Keep on sharing! I'll keep my ears open here for more.

Hi!
Thanks a lot for your comment. The sitar[1] comes from the soundfont
FluidR3_GM. It's amazing how many great free software, tools, plugins
and sounds we have on GNU/Linux!

Ciao,
Max-B

 1. https://soundcloud.com/john-option/bounce-john-option-hip-me-sitar

- -- 
IM: massimo at jabber.fsfe.org - OpenPGP Key-Id: 0x5D168FC1
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1
Comment: Using GnuPG with Icedove - http://www.enigmail.net/

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oUwBB5aX//UsrBYxfwUBAKF/SpJdO7GHffom25+f+iUwz0f957Kt9YNJ9x/fNVmt
=yggE
-----END PGP SIGNATURE-----

From len at ovenwerks.net  Sun Aug  3 06:40:48 2014
From: len at ovenwerks.net (Len Ovens)
Date: Sat, 2 Aug 2014 23:40:48 -0700 (PDT)
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
Message-ID: <alpine.DEB.2.02.1408022307460.23832@scott.cbbs.org>

On Sat, 2 Aug 2014, Ivan K wrote:

> Thanks to everyone whom responded.
We try....

> == ==   Readable Clients/
> == ==   Output Ports
> == ==   - system
> == == ? ? capture_1
> == == ? ? capture_2
>
> on the right, and then on the left, I see:
>
> == ==   Writable Clients/
> == ==   Input Ports
> == ==   - system
> == == ? ? playback_1
> == == ? ? playback_2
> == == ? ? playback_3
> == == ? ? playback_4
> == == ? ? playback_5
> == == ? ? playback_6
> == == ? ? playback_7
> == == ? ? playback_8

This does not look 2496 ish, it looks audiophile 192 ish... from my 
understanding and concidering I have not used either one of them. This to 
say that there are two m-audio audiophile models... that use two different 
kernel modules for two different chips inside.

The difference is that the 2496 uses the ice1712 which should show as 12 
inputs and 10 outputs... same as the delta44 delta66 (I have this) and 
delta1010. The 192 has an ice1724 which is 4 in and 8 out... but the 
analog in and the digital in are different devices. The analog in would be 
hw:0,0 and the s/pdif would be hw:0,1.

This page:
http://linuxmusicians.com/viewtopic.php?f=27&t=8898
seems to verify that much.

So, after you have jack running try running:
zita-a2j -d hw:0,1 -r 48000 -p 256 -n2 -c 2

This assumes your USBDualTubePre is set to 48000... and you have set Jack 
that way too :) Change the 48000 to 44100 otherwise. This would show up in 
qjackctl as zita-a2j capture_1 and 2 (does here)

if you don't have zita-a2j (you should download it) you can test with 
alsa_in instead.
alsa_in -j spdif -d hw:0,1 -r 48000 -p 256 -n 2 -c 2

This would show up in qjackctl as spdif capture_1 and 2

The -r value should be the same as your analog i/o and your pre s/pdiff 
rate (rear panel switch)

The -p value (with zita-a2j) can be one step less than you have jack set 
(I may get corrected :)
and -n should be the same.


--
Len Ovens
www.ovenwerks.net

From willgodfrey at musically.me.uk  Sun Aug  3 07:38:30 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Sun, 3 Aug 2014 08:38:30 +0100
Subject: [LAU] Linux Poetry: X-run blues
In-Reply-To: <20140803013222.1a284feb@eeyore.mozart.uni-klu.ac.at>
References: <53DBBC66.3090408@gmail.com> <53DD2D2E.8020608@localhost>
 <20140802175915.294a496b@mistral>
 <20140803013222.1a284feb@eeyore.mozart.uni-klu.ac.at>
Message-ID: <20140803083830.24f3f6d1@debian>

On Sun, 3 Aug 2014 01:32:22 +0200
Philipp ?berbacher <murks at tuxfamily.org> wrote:

> On Sat, 2 Aug 2014 17:59:15 -0400
> "jonetsu at teksavvy.com" <jonetsu at teksavvy.com> wrote:
> 
> > On Sat, 02 Aug 2014 11:25:50 -0700,
> > Fernando Lopez-Lezcano <nando at ccrma.Stanford.EDU> wrote :
> > 
> > > I hope we can soon hear a rendition of this song by Dave Philips and
> > > his guitar (with obvious audio dropouts and glitches caused by
> > > xruns, of course)...
> > 
> > And if there are none, they'll be simulated !
> > 
> > :)
> 
> I insist on real xruns, no fake crap, not just 'authentic', I want the
> real deal.
> 
> Regards,
> Philipp

Absolutely agree. What's more there must be documented proof!

Oh, er, nice bit of poetry :) 

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From willgodfrey at musically.me.uk  Sun Aug  3 07:47:19 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Sun, 3 Aug 2014 08:47:19 +0100
Subject: [LAU] John Option: new song and new Adrour project
In-Reply-To: <53DDD948.4030106@fsfe.org>
References: <53D8C430.1050109@fsfe.org> <1407301532080.16270@freeshell.de>
 <53DDD948.4030106@fsfe.org>
Message-ID: <20140803084719.31341c11@debian>

On Sun, 03 Aug 2014 08:40:08 +0200
Massimo Barbieri <massimo at fsfe.org> wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA256
> 
> Il 30/07/2014 15:34, F. Silvain ha scritto:
> > good song! I love the sitar there. It adds a nice bit of spice.
> > Good mix as well, if a little sparse, but that sounds intentional. 
> > Keep on sharing! I'll keep my ears open here for more.
> 
> Hi!
> Thanks a lot for your comment. The sitar[1] comes from the soundfont
> FluidR3_GM. It's amazing how many great free software, tools, plugins
> and sounds we have on GNU/Linux!
> 
> Ciao,
> Max-B

Very original song. Fun vid.

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From ralf.mardorf at rocketmail.com  Sun Aug  3 09:22:27 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 03 Aug 2014 11:22:27 +0200
Subject: [LAU] [Bulk] Re:  Record from M-Audio 2496 S/PDIF input
In-Reply-To: <alpine.DEB.2.02.1408022307460.23832@scott.cbbs.org>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408022307460.23832@scott.cbbs.org>
Message-ID: <1407057747.29357.8.camel@rocketmail.com>

On Sat, 2014-08-02 at 23:40 -0700, Len Ovens wrote:
> This assumes your USBDualTubePre is set to 48000... and you have set
> Jack that way too

and don't forget to chose 'S/PDIF in', _not_ 48000 for the sound card's
clock.



From fons at linuxaudio.org  Sun Aug  3 09:55:24 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Sun, 3 Aug 2014 09:55:24 +0000
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <alpine.DEB.2.02.1408022307460.23832@scott.cbbs.org>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408022307460.23832@scott.cbbs.org>
Message-ID: <20140803095524.GA19816@linuxaudio.org>

On Sat, Aug 02, 2014 at 11:40:48PM -0700, Len Ovens wrote:
 
> So, after you have jack running try running:
> zita-a2j -d hw:0,1 -r 48000 -p 256 -n2 -c 2
 
> This assumes your USBDualTubePre is set to 48000... and you have set
> Jack that way too :) Change the 48000 to 44100 otherwise. This would
> show up in qjackctl as zita-a2j capture_1 and 2 (does here)

Just use 

   zita-a2j -j spdif -d hw:0,1 -r 48000 -p 256 -n2 -c 2

to get correct port names.
 
> if you don't have zita-a2j (you should download it) you can test
> with alsa_in instead.

Wouldn't recommend that, and zita-a2j is included as an internal
client with recent Jack1 releases anyway. It can even be started
in one command together with Jack.

> The -p value (with zita-a2j) can be one step less than you have jack
> set (I may get corrected :) and -n should be the same.

There are no such restrictions.

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From massimo at fsfe.org  Sun Aug  3 11:19:08 2014
From: massimo at fsfe.org (Massimo Barbieri)
Date: Sun, 03 Aug 2014 13:19:08 +0200
Subject: [LAU] John Option: new song and new Adrour project
In-Reply-To: <20140803084719.31341c11@debian>
References: <53D8C430.1050109@fsfe.org> <1407301532080.16270@freeshell.de>
 <53DDD948.4030106@fsfe.org> <20140803084719.31341c11@debian>
Message-ID: <53DE1AAC.9030901@fsfe.org>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256

Il 03/08/2014 09:47, Will Godfrey ha scritto:
> Very original song. Fun vid.

Thank you so much!
Max-B

- -- 
IM: massimo at jabber.fsfe.org - OpenPGP Key-Id: 0x5D168FC1
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1
Comment: Using GnuPG with Icedove - http://www.enigmail.net/

iF4EAREIAAYFAlPeGqsACgkQnxH3+F0Wj8Ea8QEAxuMjzaDo37A3FzZZ6JzcDag0
BpywZSDYW5Ye0ZIWc2gA/3eopEr+vO0XHMYQHNrLWj2w12BxyhKvAxvAqzqANLkL
=Wz6o
-----END PGP SIGNATURE-----

From jbh at alchemy.lu  Sun Aug  3 11:36:54 2014
From: jbh at alchemy.lu (Joakim Hernberg)
Date: Sun, 3 Aug 2014 13:36:54 +0200
Subject: [LAU] Kxstudio RT kernel vs low latency
In-Reply-To: <CABRv+91PZU6u2HcrmAhCssp7b15a1HvFPrXV8GX2B+BvHthiTA@mail.gmail.com>
References: <2AEE2A87-78C8-4353-8DA7-D0A0ABC1CDA4@gmail.com>
 <20140725233704.739af974@tor.valhalla.alchemy.lu>
 <53D367C3.5010906@autostatic.com>
 <20140727120459.4170eaab@tor.valhalla.alchemy.lu>
 <CABRv+90jF2tjmv_QfViZQdaL97kvT_G9o7MyfWrfHnx5U21ADw@mail.gmail.com>
 <20140728140006.306dbcc5@tor.valhalla.alchemy.lu>
 <CABRv+91PZU6u2HcrmAhCssp7b15a1HvFPrXV8GX2B+BvHthiTA@mail.gmail.com>
Message-ID: <20140803133654.5f288058@tor.valhalla.alchemy.lu>

On Mon, 28 Jul 2014 23:22:48 +0100
James Stone <jamesmstone at gmail.com> wrote:

> I then removed this and had a poke around in the bios, and found that
> the main culprits for the xruns were C6 mode, and "AMD Power Now".
> Disabling these, and I now have an xrun-free experience with
> frames/period = 32/2 with pulseaudio/jack on my Scarlett 2i4, which is
> pretty amazing for a USB device IMO!

That's very amazing indeed.  But to be sure try to run hackbench too,
and see if it's xrun free under load too.

You can also run jack_iodelay and connect the output to the input on
your soundcard (turn speakers off as it might also produce feedback
depending on how the card is configured).  This ought to tell you the
realworld latency, which many times can be a lot bigger then that given
by buffersizes alone, due to "hidden" hardware buffers and
the delay introduced by the digital to analog conversion itself.

> There was also CPD mode and CState Pmin, which I disabled initially,
> but these don't seem to have any impact on xruns on my system. Cstate
> pmin seems to affect the reported DSP load - but otherwise doesn't
> affect xruns - so I think it's safe to keep on (and maybe is saving
> some power??). CPD mode doesn't seem to have any impact at all.

I suppose you could monitor the cpus temperature and if you see a
decrease in temperature, it most likely corresponds to lower power
consumption.

-- 

   Joakim

From jbh at alchemy.lu  Sun Aug  3 11:44:01 2014
From: jbh at alchemy.lu (Joakim Hernberg)
Date: Sun, 3 Aug 2014 13:44:01 +0200
Subject: [LAU] Kxstudio RT kernel vs low latency
In-Reply-To: <659AA6EE-91F7-42FE-87D7-389CDB848007@gmail.com>
References: <2AEE2A87-78C8-4353-8DA7-D0A0ABC1CDA4@gmail.com>
 <20140725233704.739af974@tor.valhalla.alchemy.lu>
 <53D367C3.5010906@autostatic.com>
 <20140727120459.4170eaab@tor.valhalla.alchemy.lu>
 <CABRv+90jF2tjmv_QfViZQdaL97kvT_G9o7MyfWrfHnx5U21ADw@mail.gmail.com>
 <20140728140006.306dbcc5@tor.valhalla.alchemy.lu>
 <CABRv+91PZU6u2HcrmAhCssp7b15a1HvFPrXV8GX2B+BvHthiTA@mail.gmail.com>
 <659AA6EE-91F7-42FE-87D7-389CDB848007@gmail.com>
Message-ID: <20140803134401.72b01d75@tor.valhalla.alchemy.lu>

On Mon, 28 Jul 2014 16:29:54 -0700
Russell Hanaghan <hanaghan.osaudio at gmail.com> wrote:

> I have many hiding spots to probe as it were. I do want to ask about
> older bios laptops tho. Running intel dual core 1.6ghz... So the
> application of these (ACPI? APM?) are handed off to OS software
> layer? Kernel module or whatever? If so (pls correct where I'm off)
> how to use equivalent control of settings from CL or tools? Given the
> sig change of stable state that Jame's refers to, it would be helpful
> to document this stuff somewhere under the "realtime Linux audio
> tweaks". The 32/2 with stability over USB2 would be a big deal in my
> case. I've not been able to come within screaming distance of that
> result. 

It might not be possible :)  Not that I know for sure, but I'd imagine
that the best you can do on older cpus is to use the performance
governor.  But do try the /dev/cpu_dma_latency trick, by running "sudo
cyclictest", maybe it's beneficial on your hardware too.

> I just dug the VAIO out of the gig rig so I can look at the FW TI
> chip and it's operating limitations.

You probably ought to give the interrupt handler servicing the FW
interface the highest priority, and then follow the same instructions I
gave in a previous post for setting up realtime priorities.

-- 

   Joakim

From jbh at alchemy.lu  Sun Aug  3 11:58:48 2014
From: jbh at alchemy.lu (Joakim Hernberg)
Date: Sun, 3 Aug 2014 13:58:48 +0200
Subject: [LAU] Kxstudio RT kernel vs low latency
In-Reply-To: <CABRv+93Fpy-QoNE1MH_-aejabjZGBJmCsufPryK12tift3ugqw@mail.gmail.com>
References: <2AEE2A87-78C8-4353-8DA7-D0A0ABC1CDA4@gmail.com>
 <20140725233704.739af974@tor.valhalla.alchemy.lu>
 <53D367C3.5010906@autostatic.com>
 <20140727120459.4170eaab@tor.valhalla.alchemy.lu>
 <CABRv+90jF2tjmv_QfViZQdaL97kvT_G9o7MyfWrfHnx5U21ADw@mail.gmail.com>
 <20140728140006.306dbcc5@tor.valhalla.alchemy.lu>
 <CABRv+91PZU6u2HcrmAhCssp7b15a1HvFPrXV8GX2B+BvHthiTA@mail.gmail.com>
 <659AA6EE-91F7-42FE-87D7-389CDB848007@gmail.com>
 <CABRv+93Fpy-QoNE1MH_-aejabjZGBJmCsufPryK12tift3ugqw@mail.gmail.com>
Message-ID: <20140803135848.4ea4599c@tor.valhalla.alchemy.lu>

On Tue, 29 Jul 2014 06:28:21 +0100
James Stone <jamesmstone at gmail.com> wrote:

> Anyway, easiest  way to test if it is that to run cyclictest with:
> "sudo cyclictest -m -n -Sp99 -i100 -d0" at the same time as running
> jackd and see if it makes any of the xruns go away. if it doesn't it
> is something else!

Probably best to just run "sudo cyclictest"  as it will enable
the /dev/cpu_dma_latency trick, if you run it will all the parameters
it will also use up execution time at a higher priority than your
audio, thus increasing the likelyhood of xruns.  The full command line
is more to ascertain that kernel scheduling is working well.

-- 

   Joakim

From jbh at alchemy.lu  Sun Aug  3 12:25:33 2014
From: jbh at alchemy.lu (Joakim Hernberg)
Date: Sun, 3 Aug 2014 14:25:33 +0200
Subject: [LAU] Kxstudio RT kernel vs low latency
In-Reply-To: <20140802055637.7784c8a248c665560303bc63@gmail.com>
References: <CABRv+93Fpy-QoNE1MH_-aejabjZGBJmCsufPryK12tift3ugqw@mail.gmail.com>
 <CFFD392B.10CC9%hanaghan.osaudio@gmail.com>
 <20140802055637.7784c8a248c665560303bc63@gmail.com>
Message-ID: <20140803142533.632c824d@tor.valhalla.alchemy.lu>

On Sat, 2 Aug 2014 05:56:37 -0300
Fede <federicogalland at gmail.com> wrote:

> On a side note: Does anyone have any factual information on
> config_irq_time_accounting ? I noticed that it increases the cpu
> load, but am not sure if it benefits the audio/midi processing. I'm
> working on a custom kernel config and trying to document how each
> parameter change affects audio and midi performance for later
> publishing.

AFAIK, it won't benefit audio/midi processing, it might even delay it
slightly, but note slightly...  It's purpose is to provide accounting
information, that is to say to help profile interrupt handlers.  I have
it turned off on the archlinux -rt kernel as I try to track the distro's
main kernel as closely as I can, and just enable or disable the few
options that are really relevant to getting good -rt performance.

-- 

   Joakim

From csanchezgs at gmail.com  Sun Aug  3 15:21:00 2014
From: csanchezgs at gmail.com (Carlos sanchiavedraz)
Date: Sun, 3 Aug 2014 17:21:00 +0200
Subject: [LAU] OT: new free soundset for DSI Prophet 12
In-Reply-To: <1407261250230.7801@freeshell.de>
References: <1407261250230.7801@freeshell.de>
Message-ID: <CA+O6s5xXSXVXBddaa2amWr0AbWWnGpN2ha-9Kc3nGCd1i6tK1A@mail.gmail.com>

2014-07-26 12:52 GMT+02:00 F. Silvain <silvain at freeshell.de>:
> Hey hey,
> not sure, if this is welcome, but I just uploaded a new free soundset for
> the DSI Prophet 12 hybrid synthesizer. It's a superb hardware instrument.
> Feel free to pass the links along!
> http://freeshell.de/~silvain/p12_fs1_soundset.zip
> http://freeshell.de/~silvain/p12_fs2_soundset.zip
>
> The first is a revision of the already existing soundset on my website. Both
> soundsets now upload to any of the 8 soundbanks of the Prophet 12.
>
> Enjoy and nice weekend!
>
> Ta-ta
> ----
> Ffanci
> * Internet: http://freeshell.de/~silvain
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

Listening now to Prophetic Jazz in your site. Nice stuff.

IINM, This soundsets are only to load on that specific synth, right?
Is there a way to load them in a soundfont or similar to use them?

Thanks for sharing.


-- 

C. sanchiavedraZ:
* NEW / NUEVO:     www.sanchiavedraZ.com
* Musix GNU+Linux: www.musix.es

From len at ovenwerks.net  Sun Aug  3 15:24:57 2014
From: len at ovenwerks.net (Len Ovens)
Date: Sun, 3 Aug 2014 08:24:57 -0700 (PDT)
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <20140803095524.GA19816@linuxaudio.org>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408022307460.23832@scott.cbbs.org>
 <20140803095524.GA19816@linuxaudio.org>
Message-ID: <alpine.DEB.2.02.1408030748440.23832@scott.cbbs.org>

On Sun, 3 Aug 2014, Fons Adriaensen wrote:

> On Sat, Aug 02, 2014 at 11:40:48PM -0700, Len Ovens wrote:
>
>   zita-a2j -j spdif -d hw:0,1 -r 48000 -p 256 -n2 -c 2
>
> to get correct port names.

I must have an older version, my man page doesn't show -j, good to know it 
is there, thank you.

> Wouldn't recommend that, and zita-a2j is included as an internal
> client with recent Jack1 releases anyway. It can even be started
> in one command together with Jack.

Hmm, I use 1.9.10. In the past I have used the features of jackdbus for a 
lot of things, but I am now probably to the point I don't have to. Jack is 
started at session start with jack_control but could be just command line. 
I don't use zita-a2j that much (testing only) as the old ensoniq card 
while quite quiet is "only" 16 bit and uses 1/8 plugs. I normally use no 
more than 2 inputs anyway. However, the debian/ubuntu repos seem to make 
it as hard as possible to install jack1... or rather hard to remove jack2 
or run jack1 along side. However, starting zita-a2j on the side is not a 
problem.


--
Len Ovens
www.ovenwerks.net


From len at ovenwerks.net  Sun Aug  3 15:34:03 2014
From: len at ovenwerks.net (Len Ovens)
Date: Sun, 3 Aug 2014 08:34:03 -0700 (PDT)
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <20140803095524.GA19816@linuxaudio.org>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408022307460.23832@scott.cbbs.org>
 <20140803095524.GA19816@linuxaudio.org>
Message-ID: <alpine.DEB.2.02.1408030831210.23832@scott.cbbs.org>

On Sun, 3 Aug 2014, Fons Adriaensen wrote:

> Wouldn't recommend that, and zita-a2j is included as an internal
> client with recent Jack1 releases anyway. It can even be started

I had heard that a2jmidid was going to get rolled in as well. It seems 
these things are not noted on:
http://trac.jackaudio.org/wiki/Q_differenc_jack1_jack2


--
Len Ovens
www.ovenwerks.net


From csanchezgs at gmail.com  Sun Aug  3 15:57:11 2014
From: csanchezgs at gmail.com (Carlos sanchiavedraz)
Date: Sun, 3 Aug 2014 17:57:11 +0200
Subject: [LAU] Beta testers required...
In-Reply-To: <20140723225643.GA19813@linuxaudio.org>
References: <20140723225643.GA19813@linuxaudio.org>
Message-ID: <CA+O6s5zbs1Udodw1o67sLda3TnrrDeYaiYX3aAt+N7sahkkzew@mail.gmail.com>

2014-07-24 0:56 GMT+02:00 Fons Adriaensen <fons at linuxaudio.org>:
> Anyone interested in beta-testing this please let me know.
>
>
> Zita-njbridge
> -------------
>
> Command line Jack clients to transmit full quality
> multichannel audio over a local IP network, with
> adaptive resampling at the receiver.
>
> Main features:
>
> * One-to-one (UDP) or one-to-many (multicast).
> * Sender and receiver(s) can each have their own
>   sample rate and period size.
> * Up to 64 channels, 16 or 24 bit or float samples.
> * Receiver(s) can select any combination of channels.
> * Low latency, optional additional buffering.
> * High quality jitter-free resampling.
> * Graceful handling of xruns, skipped cycles, lost
>   packets and freewheeling.
> * IP6 fully supported.
> * Requires zita-resampler, no other dependencies.
>
> Note that this version is meant for use on a *local*
> network. It may work or not on the wider internet if
> receiver(s) are configured for additional buffering,
> and if you are lucky. The current code will replace
> any gaps in the audio stream by silence, and does not
> attempt to re-insert packets that arrive out of order.
>
> You will need a fairly recent Jack version, as the
> code uses jack_get_cycle_times() and no fallback for
> that is provided.
>
> Ciao,
>
> --
> FA
>
> A world of exhaustive, reliable metadata would be an utopia.
> It's also a pipe-dream, founded on self-delusion, nerd hubris
> and hysterically inflated market opportunities. (Cory Doctorow)
>
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

Hello dear Fons.

This seems really attractive and useful to me. Maybe in some way I can
help, from the musicians side and also engineering/technical side.

Let me know if it's still required.

-- 

C. sanchiavedraZ:
* NEW / NUEVO:     www.sanchiavedraZ.com
* Musix GNU+Linux: www.musix.es

From fons at linuxaudio.org  Sun Aug  3 16:07:53 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Sun, 3 Aug 2014 16:07:53 +0000
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <alpine.DEB.2.02.1408030748440.23832@scott.cbbs.org>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408022307460.23832@scott.cbbs.org>
 <20140803095524.GA19816@linuxaudio.org>
 <alpine.DEB.2.02.1408030748440.23832@scott.cbbs.org>
Message-ID: <20140803160753.GA14272@linuxaudio.org>

On Sun, Aug 03, 2014 at 08:24:57AM -0700, Len Ovens wrote:
> On Sun, 3 Aug 2014, Fons Adriaensen wrote:
> 
> >On Sat, Aug 02, 2014 at 11:40:48PM -0700, Len Ovens wrote:
> >
> >  zita-a2j -j spdif -d hw:0,1 -r 48000 -p 256 -n2 -c 2
> >
> >to get correct port names.
> 
> I must have an older version, my man page doesn't show -j, good to
> know it is there, thank you.

It's shown by the command line help, but indeed not in the manpages.
Will be fixed.

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From csanchezgs at gmail.com  Sun Aug  3 16:22:53 2014
From: csanchezgs at gmail.com (Carlos sanchiavedraz)
Date: Sun, 3 Aug 2014 18:22:53 +0200
Subject: [LAU] Measuring the acoustical characteristics of my studio
 using FLOSS software?
In-Reply-To: <53D09521.6040704@localhost>
References: <CAL=sEa1zWvYAzqaYPUcKB70sSfgKdVH82sUD0j+eSZdvpAnk0g@mail.gmail.com>
 <20140723131439.GA27856@linuxaudio.org> <53D09521.6040704@localhost>
Message-ID: <CA+O6s5xJuouRk1pieBm6oOx1YLMnuabgSnpCxxfru17AYaRX2w@mail.gmail.com>

2014-07-24 7:09 GMT+02:00 Fernando Lopez-Lezcano <nando at ccrma.stanford.edu>:
> On 07/23/2014 06:14 AM, Fons Adriaensen wrote:
>>
>> On Wed, Jul 23, 2014 at 07:52:52AM +0200, Gabriel Nordeborn wrote:
>>
>>> So, my question is: How on earth do I do this?! Are there FLOSS tools
>>> available for this? Is it easy to do something as basic as this?
>>
>>
>> It's not basic and not easy.
>
> ...
>
>> If you want to correct your system after the room is finished, the
>> DRC software <http://drc-fir.sourceforge.net/> is the way to go.
>> Make sure to read the excellent documentation, all of it, so you
>> have and idea of what you're doing. It will explain some of the
>> hairy things I hinted at earlier. DRC will produce IR filters
>> which you can use with jconvolver (or with any other convolution
>> processor). The result will be much better than when using e.g.
>> a graphic EQ, because DRC does things that a simple equaliser
>> can't do.
>
>
> A good reference here:
> http://lac.linuxaudio.org/2008/download/papers/18.pdf
>
> ----
> AMBI at Home ? The search for extra?frontal intelligence
> Linux Audio Conference 2008, Cologne
> J?rn NETTINGSMEIER
> ----
>
> Although the paper talks about an Ambisonics rig at home, it also dwells on
> room correction and DRC (which is good for Ambisonics and everything else).
>
> I'm doing the same in my home studio and DRC does help (but of course it is
> not magic)...
> Good luck!
> -- Fernando
>
>
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

I'm not an acoustic expert, but maybe this helps.
It seems to me that the main critical points in a room are the ones
which produce early reflections, specially the areas at both sides of
your listening position (LP) where sound reflects and bounces back to
you (so it arrives at a different time than direct sound from
speakers). You can take in consideration the point right above the LP
as well.

There's a simple technique that may help to find those spots to have
in account for a potential acoustic treatment:

- Pick a mirror
- From your listening position perspective and moving the mirror (you
know, with its back to the wall), find the areas where you see the
image of your speakers reflected in the mirror. It's better to have
some helping hand so you can stay seated in your LP.

The point is: if you see the speakers/the light reflected from the
speakers that gets to your eyes bounced from the mirror, then the
sound will get to you bounced on those areas as well.

On this site you can get some sketches for a good placement of
acoustic elements and advice if you give the dimensions of your room:
http://www.auralex.com/ikc/


-- 

C. sanchiavedraZ:
* NEW / NUEVO:     www.sanchiavedraZ.com
* Musix GNU+Linux: www.musix.es

From fons at linuxaudio.org  Sun Aug  3 17:06:28 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Sun, 3 Aug 2014 17:06:28 +0000
Subject: [LAU] Measuring the acoustical characteristics of my studio
 using FLOSS software?
In-Reply-To: <CA+O6s5xJuouRk1pieBm6oOx1YLMnuabgSnpCxxfru17AYaRX2w@mail.gmail.com>
References: <CAL=sEa1zWvYAzqaYPUcKB70sSfgKdVH82sUD0j+eSZdvpAnk0g@mail.gmail.com>
 <20140723131439.GA27856@linuxaudio.org>
 <53D09521.6040704@localhost>
 <CA+O6s5xJuouRk1pieBm6oOx1YLMnuabgSnpCxxfru17AYaRX2w@mail.gmail.com>
Message-ID: <20140803170628.GB14272@linuxaudio.org>

On Sun, Aug 03, 2014 at 06:22:53PM +0200, Carlos sanchiavedraz wrote:

> - Pick a mirror
> - From your listening position perspective and moving the mirror (you
> know, with its back to the wall), find the areas where you see the
> image of your speakers reflected in the mirror. It's better to have
> some helping hand so you can stay seated in your LP.
> 
> The point is: if you see the speakers/the light reflected from the
> speakers that gets to your eyes bounced from the mirror, then the
> sound will get to you bounced on those areas as well.

Take that with some very big lumps of salt.

For at least half of the audible frequency range, the wavelenght of
sound is comparable or larger than the typical sizes of objects that
surround us. Which means that sound will not behave as light. This is
the main reason why so many people have a completely wrong idea of how
sound waves interact with objects or a room.


Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From ralf.mardorf at rocketmail.com  Sun Aug  3 17:18:37 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 03 Aug 2014 19:18:37 +0200
Subject: [LAU] [Bulk] Re: Measuring the acoustical characteristics of my
 studio using FLOSS software?
In-Reply-To: <CA+O6s5xJuouRk1pieBm6oOx1YLMnuabgSnpCxxfru17AYaRX2w@mail.gmail.com>
References: <CAL=sEa1zWvYAzqaYPUcKB70sSfgKdVH82sUD0j+eSZdvpAnk0g@mail.gmail.com>
 <20140723131439.GA27856@linuxaudio.org> <53D09521.6040704@localhost>
 <CA+O6s5xJuouRk1pieBm6oOx1YLMnuabgSnpCxxfru17AYaRX2w@mail.gmail.com>
Message-ID: <1407086317.778.3.camel@rocketmail.com>

On Sun, 2014-08-03 at 18:22 +0200, Carlos sanchiavedraz wrote:
> From your listening position perspective and moving the mirror (you
> know, with its back to the wall), find the areas where you see the
> image of your speakers reflected in the mirror.

For a variety of reasons I doubt that this is useful.



From ralf.mardorf at rocketmail.com  Sun Aug  3 17:30:11 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 03 Aug 2014 19:30:11 +0200
Subject: [LAU] [Bulk] Re: Measuring the acoustical characteristics of my
 studio using FLOSS software?
In-Reply-To: <20140803170628.GB14272@linuxaudio.org>
References: <CAL=sEa1zWvYAzqaYPUcKB70sSfgKdVH82sUD0j+eSZdvpAnk0g@mail.gmail.com>
 <20140723131439.GA27856@linuxaudio.org> <53D09521.6040704@localhost>
 <CA+O6s5xJuouRk1pieBm6oOx1YLMnuabgSnpCxxfru17AYaRX2w@mail.gmail.com>
 <20140803170628.GB14272@linuxaudio.org>
Message-ID: <1407087011.778.5.camel@rocketmail.com>

On Sun, 2014-08-03 at 17:06 +0000, Fons Adriaensen wrote:
> On Sun, Aug 03, 2014 at 06:22:53PM +0200, Carlos sanchiavedraz wrote:
> 
> > - Pick a mirror
> > - From your listening position perspective and moving the mirror (you
> > know, with its back to the wall), find the areas where you see the
> > image of your speakers reflected in the mirror. It's better to have
> > some helping hand so you can stay seated in your LP.
> > 
> > The point is: if you see the speakers/the light reflected from the
> > speakers that gets to your eyes bounced from the mirror, then the
> > sound will get to you bounced on those areas as well.
> 
> Take that with some very big lumps of salt.
> 
> For at least half of the audible frequency range, the wavelenght of
> sound is comparable or larger than the typical sizes of objects that
> surround us. Which means that sound will not behave as light. This is
> the main reason why so many people have a completely wrong idea of how
> sound waves interact with objects or a room.

Even if sound waves would behave like a ball thrown against the wall,
the nominal dispersion angle, the angles of the walls etc. pp. still
would be ignored by this mirror test, not to mention that we aren't
horses, with our eyes at the side of the head, were our ears usually
are.


From csanchezgs at gmail.com  Sun Aug  3 17:58:19 2014
From: csanchezgs at gmail.com (Carlos sanchiavedraz)
Date: Sun, 3 Aug 2014 19:58:19 +0200
Subject: [LAU] Measuring the acoustical characteristics of my studio
 using FLOSS software?
In-Reply-To: <20140803170628.GB14272@linuxaudio.org>
References: <CAL=sEa1zWvYAzqaYPUcKB70sSfgKdVH82sUD0j+eSZdvpAnk0g@mail.gmail.com>
 <20140723131439.GA27856@linuxaudio.org> <53D09521.6040704@localhost>
 <CA+O6s5xJuouRk1pieBm6oOx1YLMnuabgSnpCxxfru17AYaRX2w@mail.gmail.com>
 <20140803170628.GB14272@linuxaudio.org>
Message-ID: <CA+O6s5w5e=tZ6f0LxLYxaL5FRBsEzxOuLmV379cnU_t56aq-mA@mail.gmail.com>

2014-08-03 19:06 GMT+02:00 Fons Adriaensen <fons at linuxaudio.org>:
> On Sun, Aug 03, 2014 at 06:22:53PM +0200, Carlos sanchiavedraz wrote:
>
>> - Pick a mirror
>> - From your listening position perspective and moving the mirror (you
>> know, with its back to the wall), find the areas where you see the
>> image of your speakers reflected in the mirror. It's better to have
>> some helping hand so you can stay seated in your LP.
>>
>> The point is: if you see the speakers/the light reflected from the
>> speakers that gets to your eyes bounced from the mirror, then the
>> sound will get to you bounced on those areas as well.
>
> Take that with some very big lumps of salt.
>
> For at least half of the audible frequency range, the wavelenght of
> sound is comparable or larger than the typical sizes of objects that
> surround us. Which means that sound will not behave as light. This is
> the main reason why so many people have a completely wrong idea of how
> sound waves interact with objects or a room.
>
>
> Ciao,
>

Sure, Fons, put salt everywhere; I'm aware of some of that, but as I
pointed out I'm no expert, just wanted to add my two cents just to
begin making a blurred image of what's the problem to make it clearer
afterwards with more knowledge.
Thanks for clarifying anyway.

> --
> FA
>
> A world of exhaustive, reliable metadata would be an utopia.
> It's also a pipe-dream, founded on self-delusion, nerd hubris
> and hysterically inflated market opportunities. (Cory Doctorow)
>
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user



-- 

C. sanchiavedraZ:
* NEW / NUEVO:     www.sanchiavedraZ.com
* Musix GNU+Linux: www.musix.es

From csanchezgs at gmail.com  Sun Aug  3 18:02:09 2014
From: csanchezgs at gmail.com (Carlos sanchiavedraz)
Date: Sun, 3 Aug 2014 20:02:09 +0200
Subject: [LAU] [Bulk] Re: Measuring the acoustical characteristics of my
 studio using FLOSS software?
In-Reply-To: <1407087011.778.5.camel@rocketmail.com>
References: <CAL=sEa1zWvYAzqaYPUcKB70sSfgKdVH82sUD0j+eSZdvpAnk0g@mail.gmail.com>
 <20140723131439.GA27856@linuxaudio.org> <53D09521.6040704@localhost>
 <CA+O6s5xJuouRk1pieBm6oOx1YLMnuabgSnpCxxfru17AYaRX2w@mail.gmail.com>
 <20140803170628.GB14272@linuxaudio.org>
 <1407087011.778.5.camel@rocketmail.com>
Message-ID: <CA+O6s5y1E4ULbCgYOkSzJ7UnefZTuvAXtc8jyjKQYiHQpM7Fuw@mail.gmail.com>

2014-08-03 19:30 GMT+02:00 Ralf Mardorf <ralf.mardorf at rocketmail.com>:
> On Sun, 2014-08-03 at 17:06 +0000, Fons Adriaensen wrote:
>> On Sun, Aug 03, 2014 at 06:22:53PM +0200, Carlos sanchiavedraz wrote:
>>
>> > - Pick a mirror
>> > - From your listening position perspective and moving the mirror (you
>> > know, with its back to the wall), find the areas where you see the
>> > image of your speakers reflected in the mirror. It's better to have
>> > some helping hand so you can stay seated in your LP.
>> >
>> > The point is: if you see the speakers/the light reflected from the
>> > speakers that gets to your eyes bounced from the mirror, then the
>> > sound will get to you bounced on those areas as well.
>>
>> Take that with some very big lumps of salt.
>>
>> For at least half of the audible frequency range, the wavelenght of
>> sound is comparable or larger than the typical sizes of objects that
>> surround us. Which means that sound will not behave as light. This is
>> the main reason why so many people have a completely wrong idea of how
>> sound waves interact with objects or a room.
>
> Even if sound waves would behave like a ball thrown against the wall,
> the nominal dispersion angle, the angles of the walls etc. pp. still
> would be ignored by this mirror test, not to mention that we aren't
> horses, with our eyes at the side of the head, were our ears usually
> are.
>
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

Well, maybe next time I should just shut up :). Just for the record,
this information I learned from someone who was no newbie in this
matters.

-- 

C. sanchiavedraZ:
* NEW / NUEVO:     www.sanchiavedraZ.com
* Musix GNU+Linux: www.musix.es

From silvain at freeshell.de  Sun Aug  3 19:33:32 2014
From: silvain at freeshell.de (F. Silvain)
Date: Sun, 3 Aug 2014 21:33:32 +0200 (CEST)
Subject: [LAU] OT: new free soundset for DSI Prophet 12
In-Reply-To: <CA+O6s5xXSXVXBddaa2amWr0AbWWnGpN2ha-9Kc3nGCd1i6tK1A@mail.gmail.com>
References: <1407261250230.7801@freeshell.de>
 <CA+O6s5xXSXVXBddaa2amWr0AbWWnGpN2ha-9Kc3nGCd1i6tK1A@mail.gmail.com>
Message-ID: <1408032129170.22494@freeshell.de>

Carlos sanchiavedraz, Aug 3 2014:
...
> IINM, This soundsets are only to load on that specific synth, right?
> Is there a way to load them in a soundfont or similar to use them?

Hey Carlos,
sorry, no luck. No samples and no converters from Prophet 12 to softsynth. Only way to get them - right now - is to buy a Prophet 12 and the fun is expensive, yet worth every penny! :)
>
> Thanks for sharing.
>
>
> -- 
>
> C. sanchiavedraZ:
> * NEW / NUEVO:     www.sanchiavedraZ.com
> * Musix GNU+Linux: www.musix.es
>

Ta-ta
----
Ffanci
* Internet: http://freeshell.de/~silvain

From gabbe.nord at gmail.com  Sun Aug  3 20:39:42 2014
From: gabbe.nord at gmail.com (Gabriel Nordeborn)
Date: Sun, 3 Aug 2014 22:39:42 +0200
Subject: [LAU] Measuring the acoustical characteristics of my studio
 using FLOSS software?
In-Reply-To: <CA+O6s5w5e=tZ6f0LxLYxaL5FRBsEzxOuLmV379cnU_t56aq-mA@mail.gmail.com>
References: <CAL=sEa1zWvYAzqaYPUcKB70sSfgKdVH82sUD0j+eSZdvpAnk0g@mail.gmail.com>
 <20140723131439.GA27856@linuxaudio.org>
 <53D09521.6040704@localhost>
 <CA+O6s5xJuouRk1pieBm6oOx1YLMnuabgSnpCxxfru17AYaRX2w@mail.gmail.com>
 <20140803170628.GB14272@linuxaudio.org>
 <CA+O6s5w5e=tZ6f0LxLYxaL5FRBsEzxOuLmV379cnU_t56aq-mA@mail.gmail.com>
Message-ID: <CAL=sEa3p6Ro64Guw1SNn=WnpktBn_E-S9KnRvAmZ55z2egLsTg@mail.gmail.com>

Thanks for all your comments! I've built and mounted most of the planned
absorbers, with one exception; the absorber panels on each side of the LP,
which incidentally is what you're talking about ;-) thanks FOR bringing up
the mirror thing Carlos, that's what I've been thinking about doing, but
perhaps I shouldn't then given this conversation...

So, if the mirror trick doesn't work as intended, is there some other rule
of thumb I can use when placing my absorber panels on the sides of my LP?
Fons?

Thanks for all the help again all of you!
On Aug 3, 2014 7:58 PM, "Carlos sanchiavedraz" <csanchezgs at gmail.com> wrote:

> 2014-08-03 19:06 GMT+02:00 Fons Adriaensen <fons at linuxaudio.org>:
> > On Sun, Aug 03, 2014 at 06:22:53PM +0200, Carlos sanchiavedraz wrote:
> >
> >> - Pick a mirror
> >> - From your listening position perspective and moving the mirror (you
> >> know, with its back to the wall), find the areas where you see the
> >> image of your speakers reflected in the mirror. It's better to have
> >> some helping hand so you can stay seated in your LP.
> >>
> >> The point is: if you see the speakers/the light reflected from the
> >> speakers that gets to your eyes bounced from the mirror, then the
> >> sound will get to you bounced on those areas as well.
> >
> > Take that with some very big lumps of salt.
> >
> > For at least half of the audible frequency range, the wavelenght of
> > sound is comparable or larger than the typical sizes of objects that
> > surround us. Which means that sound will not behave as light. This is
> > the main reason why so many people have a completely wrong idea of how
> > sound waves interact with objects or a room.
> >
> >
> > Ciao,
> >
>
> Sure, Fons, put salt everywhere; I'm aware of some of that, but as I
> pointed out I'm no expert, just wanted to add my two cents just to
> begin making a blurred image of what's the problem to make it clearer
> afterwards with more knowledge.
> Thanks for clarifying anyway.
>
> > --
> > FA
> >
> > A world of exhaustive, reliable metadata would be an utopia.
> > It's also a pipe-dream, founded on self-delusion, nerd hubris
> > and hysterically inflated market opportunities. (Cory Doctorow)
> >
> > _______________________________________________
> > Linux-audio-user mailing list
> > Linux-audio-user at lists.linuxaudio.org
> > http://lists.linuxaudio.org/listinfo/linux-audio-user
>
>
>
> --
>
> C. sanchiavedraZ:
> * NEW / NUEVO:     www.sanchiavedraZ.com
> * Musix GNU+Linux: www.musix.es
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user
>
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From fons at linuxaudio.org  Sun Aug  3 21:54:56 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Sun, 3 Aug 2014 21:54:56 +0000
Subject: [LAU] Measuring the acoustical characteristics of my studio
 using FLOSS software?
In-Reply-To: <CAL=sEa3p6Ro64Guw1SNn=WnpktBn_E-S9KnRvAmZ55z2egLsTg@mail.gmail.com>
References: <CAL=sEa1zWvYAzqaYPUcKB70sSfgKdVH82sUD0j+eSZdvpAnk0g@mail.gmail.com>
 <20140723131439.GA27856@linuxaudio.org>
 <53D09521.6040704@localhost>
 <CA+O6s5xJuouRk1pieBm6oOx1YLMnuabgSnpCxxfru17AYaRX2w@mail.gmail.com>
 <20140803170628.GB14272@linuxaudio.org>
 <CA+O6s5w5e=tZ6f0LxLYxaL5FRBsEzxOuLmV379cnU_t56aq-mA@mail.gmail.com>
 <CAL=sEa3p6Ro64Guw1SNn=WnpktBn_E-S9KnRvAmZ55z2egLsTg@mail.gmail.com>
Message-ID: <20140803215456.GA12872@linuxaudio.org>

On Sun, Aug 03, 2014 at 10:39:42PM +0200, Gabriel Nordeborn wrote:
 
> So, if the mirror trick doesn't work as intended, is there some other rule
> of thumb I can use when placing my absorber panels on the sides of my LP?
> Fons?

The mirror trick will help you to remove the first reflection
of the side walls for mid and high frequencies. If that will
improve the sound remains to be seen. What is very clear is 
that it won't do much to improve a bad room sound because
that is not the result of just those first reflections.

Remember the last time you were standing close to a wall ?
Did that wall create a 'mirror image' of sounds arriving
from the other side ? I'm  pretty sure it didn't, one reason
being that your brain will reject such images - even if you
wanted to 'see' them you wouldn't be able to. Not until the
delays get a lot larger than what a typical room will produce,
and them we're talking about discrete echos. 

There are no 'rules of thumb' except some based on reliable
measurements of acoustic parameters.  

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From jostein at vait.se  Sun Aug  3 23:05:14 2014
From: jostein at vait.se (Jostein Chr. Andersen)
Date: Mon, 04 Aug 2014 01:05:14 +0200
Subject: [LAU] Measuring the acoustical characteristics of my studio
	using FLOSS software?
In-Reply-To: <CAL=sEa1zWvYAzqaYPUcKB70sSfgKdVH82sUD0j+eSZdvpAnk0g@mail.gmail.com>
References: <CAL=sEa1zWvYAzqaYPUcKB70sSfgKdVH82sUD0j+eSZdvpAnk0g@mail.gmail.com>
Message-ID: <4738547.fyS0WoeaMf@jca-studio>

On Wednesday, July 23, 2014 07:52:52 Gabriel Nordeborn wrote:
....
> 
> So, my question is: How on earth do I do this?! Are there FLOSS tools
> available for this? Is it easy to do something as basic as this?

Some of the FLOSS tools are nearer than you might think, it's your ears!   :-)  
Here is my thoughts on small control rooms/studios, and you can do a lot 
before starting to measure:


First of all, no matter if the room is perfect or not: When you mix, do it at 
an amplitude where you can speak normally.  That way, the room's impact is at 
a minimum and your ears will be very happy! Everything "sounds good" when you 
play loader. If it does not sound good when you play at a low level, then it's 
not finished yet.

And don't bother to make the room perfect, it's better to have a sweet  spot 
or two-three in the room where it's  good,  first of all  where you sit and mix 
and possible a spot or two where you can record vocals and acoustic 
instruments.

Measure tools and software are great, but you should also trust your ears. I 
think that is more important. I you from your listening position plays a sinus 
tone from a C1 note  (that''s around 33Hz) and walk chromatically up to C4 ( 
262Hz), then you will easily hear  that some notes probably will sound 
stronger and some weaker (phased out). When you identify a trouble note,  you 
can localize where the spot in  wall or ceiling or whatever that is making 
problems. Thankfully, not all room resonances will kick in from you listening 
position, but let's say that a low E (the one at approx 82 Hz,  ie the low 
guitar e-string) is stronger, then divide 172 with 82Hz and you will get 
approx. 2 meters as the answer.  Is there something with 2 meter of length 
that can create problems? It can be between two walls or even more likely the 
distance from your ears to where a wall and a ceiling meets. That sounds 
amplitude may be weaker if you move your head a little bit. You should be able 
to quickly identify the 3-4 worst problem areas in your listening position.

Apart from that, the measuring is of course very important, but most sound 
engineers (at least the ones I know) don't measure, they use their ears and 
have also learned their rooms and gear.

Jostein




From jonetsu at teksavvy.com  Sun Aug  3 23:15:18 2014
From: jonetsu at teksavvy.com (jonetsu at teksavvy.com)
Date: Sun, 3 Aug 2014 19:15:18 -0400
Subject: [LAU] Linux Poetry: X-run blues
In-Reply-To: <20140803083830.24f3f6d1@debian>
References: <53DBBC66.3090408@gmail.com> <53DD2D2E.8020608@localhost>
 <20140802175915.294a496b@mistral>
 <20140803013222.1a284feb@eeyore.mozart.uni-klu.ac.at>
 <20140803083830.24f3f6d1@debian>
Message-ID: <20140803191518.203c8fa0@mistral>

Le dimanche, 3 ao?t 2014 08:38:30 +0100,
Will Godfrey <willgodfrey at musically.me.uk> a ?crit :

> On Sun, 3 Aug 2014 01:32:22 +0200
> Philipp ?berbacher <murks at tuxfamily.org> wrote:
> 
> > On Sat, 2 Aug 2014 17:59:15 -0400
> > "jonetsu at teksavvy.com" <jonetsu at teksavvy.com> wrote:

> > > And if there are none, they'll be simulated !

> > I insist on real xruns, no fake crap, not just 'authentic', I want
> > the real deal.

> Absolutely agree. What's more there must be documented proof!

OK, so that's a start for the video that'll be going along the song.


From temps.jo at gmail.com  Mon Aug  4 00:24:00 2014
From: temps.jo at gmail.com (pierre jocelyn andre)
Date: Mon, 4 Aug 2014 02:24:00 +0200
Subject: [LAU] drum synth
In-Reply-To: <20140719002035.611de988@Scrapyard.lan>
References: <20140719002035.611de988@Scrapyard.lan>
Message-ID: <CAJaXuw+fsKNAnUynmkjCc4pDWdoN=JgjgU06zgezY+ze+s3O_w@mail.gmail.com>

Hello there,
I corrected the latest version, it works now.
I need to talk with other developers where I place the deb package

In French, we say,
la mis?re se subit,

a destitute is not guilty, that there have not light synthesizer free for
our children, they are not guilty
But a man who creates misery is a miserable

Please do not destroy wealth for reasons imagined on how to code

Best regard


2014-07-19 1:20 GMT+02:00 James Morris <james at jwm-art.net>:

> Hi,
>
> Are there any floss synths for linux dedicated to drum synthesis?
>
> James.
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user
>
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From ralf.mardorf at rocketmail.com  Mon Aug  4 05:33:34 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Mon, 04 Aug 2014 07:33:34 +0200
Subject: [LAU] [Bulk] Re: Measuring the acoustical characteristics of my
 studio using FLOSS software?
In-Reply-To: <4738547.fyS0WoeaMf@jca-studio>
References: <CAL=sEa1zWvYAzqaYPUcKB70sSfgKdVH82sUD0j+eSZdvpAnk0g@mail.gmail.com>
 <4738547.fyS0WoeaMf@jca-studio>
Message-ID: <1407130414.1852.3.camel@rocketmail.com>

On Mon, 2014-08-04 at 01:05 +0200, Jostein Chr. Andersen wrote:
> First of all, no matter if the room is perfect or not: When you mix,
> do it at an amplitude where you can speak normally.  That way, the
> room's impact is at a minimum

Somebody already mentioned that and in addition to use near-field
monitors. I will add that any speakers with less Watt, build to work at
low levels are ok. Some small speakers have good cases and produce good
bass sound.

> most sound engineers [...] have also learned their rooms and gear.

Also this already was mentioned. Perhaps it already was done by you?

Full ACK to both points. Btw. I experienced a lot of cheap control rooms
were it wasn't pleasant to stay. A home studio with a room that has got
a living room acoustic isn't the perfect control room, but OTOH the
acoustic of a living room is an ambience to feel well and people listen
to music while being in a living room, car, office, IOW sometimes
efforts to build a control room with less money and without the needed
knowledge doesn't result with a better room for mixing music. A coloured
wall, a window and some refelections for a room you're free to do what
you like to do could be better than a grey sound absorbing room without
daylight, but with a smoking, eating and drinking ban.

Assumed the room and speaker positions should be perfect, are the
monitors and amps perfect? It's an endless chain that can't be solved
for amateur and small professional studios. The people living above my
music room 24/7 run a circulating pump of an aircon, if the location
isn't perfect, there always will be similar ambient noise we can't get
rid of by deafen a wall. It's not only important what happens inside the
control room, but also to care about the things that come from outside
the room, such as subsonic noise, that will disturb your perception.

The most important IMO is to learn our rooms and gear.


From jostein at vait.se  Mon Aug  4 08:45:14 2014
From: jostein at vait.se (Jostein Chr. Andersen)
Date: Mon, 04 Aug 2014 10:45:14 +0200
Subject: [LAU] [Bulk] Re: Measuring the acoustical characteristics of my
	studio using FLOSS software?
In-Reply-To: <1407130414.1852.3.camel@rocketmail.com>
References: <CAL=sEa1zWvYAzqaYPUcKB70sSfgKdVH82sUD0j+eSZdvpAnk0g@mail.gmail.com>
 <4738547.fyS0WoeaMf@jca-studio> <1407130414.1852.3.camel@rocketmail.com>
Message-ID: <7527667.tv11Yxo9pa@jca-studio>

Hi Ralf,

On Monday, August 04, 2014 07:33:34 Ralf Mardorf wrote:
> On Mon, 2014-08-04 at 01:05 +0200, Jostein Chr. Andersen wrote:
...
> Also this already was mentioned. Perhaps it already was done by you?

It was not me that mentioned it previously, I have overlooked it. :-) I will 
try harder to read every posting next time!



From ralf.mardorf at rocketmail.com  Mon Aug  4 09:06:20 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Mon, 04 Aug 2014 11:06:20 +0200
Subject: [LAU] [Bulk] Re: [Bulk] Re: Measuring the acoustical
 characteristics of my studio using FLOSS software?
In-Reply-To: <7527667.tv11Yxo9pa@jca-studio>
References: <CAL=sEa1zWvYAzqaYPUcKB70sSfgKdVH82sUD0j+eSZdvpAnk0g@mail.gmail.com>
 <4738547.fyS0WoeaMf@jca-studio> <1407130414.1852.3.camel@rocketmail.com>
 <7527667.tv11Yxo9pa@jca-studio>
Message-ID: <1407143180.1852.7.camel@rocketmail.com>

On Mon, 2014-08-04 at 10:45 +0200, Jostein Chr. Andersen wrote:
> Hi Ralf,
> 
> On Monday, August 04, 2014 07:33:34 Ralf Mardorf wrote:
> > On Mon, 2014-08-04 at 01:05 +0200, Jostein Chr. Andersen wrote:
> ...
> > Also this already was mentioned. Perhaps it already was done by you?
> 
> It was not me that mentioned it previously, I have overlooked it. :-) I will 
> try harder to read every posting next time!

There's nothing wrong with repeating a few good points :).

Regards,
Ralf

PS: Perhaps I'm mistaken and it was mentioned in a similar thread on
another list :D.


From csanchezgs at gmail.com  Mon Aug  4 09:11:00 2014
From: csanchezgs at gmail.com (Carlos sanchiavedraz)
Date: Mon, 4 Aug 2014 11:11:00 +0200
Subject: [LAU] OT: new free soundset for DSI Prophet 12
In-Reply-To: <1408032129170.22494@freeshell.de>
References: <1407261250230.7801@freeshell.de>
 <CA+O6s5xXSXVXBddaa2amWr0AbWWnGpN2ha-9Kc3nGCd1i6tK1A@mail.gmail.com>
 <1408032129170.22494@freeshell.de>
Message-ID: <CA+O6s5wjLN2J4Dzm4JV8fost5AOGvFdSP1tAqW=_bBeZtC1iDg@mail.gmail.com>

2014-08-03 21:33 GMT+02:00 F. Silvain <silvain at freeshell.de>:
> Carlos sanchiavedraz, Aug 3 2014:
> ...
>
>> IINM, This soundsets are only to load on that specific synth, right?
>> Is there a way to load them in a soundfont or similar to use them?
>
>
> Hey Carlos,
> sorry, no luck. No samples and no converters from Prophet 12 to softsynth.
> Only way to get them - right now - is to buy a Prophet 12 and the fun is
> expensive, yet worth every penny! :)
>

Ok, I supposed that.
Keep on going with your music.

>>
>> Thanks for sharing.
>>
>>
>> --
>>
>> C. sanchiavedraZ:
>> * NEW / NUEVO:     www.sanchiavedraZ.com
>> * Musix GNU+Linux: www.musix.es
>>
>
> Ta-ta
> ----
> Ffanci
> * Internet: http://freeshell.de/~silvain



-- 

C. sanchiavedraZ:
* NEW / NUEVO:     www.sanchiavedraZ.com
* Musix GNU+Linux: www.musix.es

From brian at gospacecraft.com  Mon Aug  4 15:07:32 2014
From: brian at gospacecraft.com (Brian Sorahan)
Date: Mon, 4 Aug 2014 10:07:32 -0500
Subject: [LAU] [Bulk] Re: music software for kids
In-Reply-To: <CA+O6s5w2LDpT8qk6oA=sh5=_L11J_EV=DRT0e0ee3MgnTWrmrQ@mail.gmail.com>
References: <CAG+zz5nrr_+OmZFqtGw0Ye+=RYTrNNzQcVoSJODFUavSjnpxmQ@mail.gmail.com>
 <20140726090235.2691fcc9@eeyore.mozart.uni-klu.ac.at>
 <1406368461.5104.7.camel@rocketmail.com>
 <CAG+zz5mXJYz5BrR2sacZ13Xn_b4ssVg3xqvVuqvnQL7GOTBU3Q@mail.gmail.com>
 <CA+O6s5w2LDpT8qk6oA=sh5=_L11J_EV=DRT0e0ee3MgnTWrmrQ@mail.gmail.com>
Message-ID: <CAG+zz5mrLSd3kO9wTwbtgvnovptrWDq4T75pzv03kBV7dBsy7Q@mail.gmail.com>

On Sat, Aug 2, 2014 at 12:03 PM, Carlos sanchiavedraz <csanchezgs at gmail.com>
wrote:

>
> El 28/07/2014 16:43, "Brian Sorahan" <brian at gospacecraft.com> escribi?:
>
> >
> > Thanks for the suggestions everyone!
> >
> > @Ralf I halfway agree that you can learn the basics of more
> fully-featured software by fiddling around. I also think that "babyish"
> music software can be a lot of fun, and there is nothing wrong with a child
> using it. I definitely plan on steering my 2 boys towards the powerful
> programs (that I think are really fun), but if they have tons of fun making
> cheesy little ditties with toy software (like I did when I was young), then
> I'm all for it.
> >
> >
> >
> > On Sat, Jul 26, 2014 at 4:54 AM, Ralf Mardorf <
> ralf.mardorf at rocketmail.com> wrote:
> >>
> >>
> >> > Brian Sorahan <brian at gospacecraft.com> wrote:
> >> > > Are there any music applications for linux that would be suitable
> for
> >> > > a 10 year old
> >>
> >> I assume that the 10 year old child isn't retarded, so the best thing
> >> IMO is to really learn how to make music, instead of using babyish music
> >> software. Learning without teaching is possible, just by fiddling
> >> around. A short explanation how to use Qtractor or a simple MIDI app,
> >> adding a virtual keyboard or a real MIDI keyboard, using fluidsynth or
> >> similar shouldn't expect too much of a 10 year old, even when there's
> >> not much interest, just a little bit fiddling around is wanted. Reading
> >> documentations isn't needed, I'm a dyslexic, so I know whereof I speak.
> >> Btw. I worked a lot with children around that age, my last job working
> >> with those children was from beginning of this year until the the
> >> beginning of this month (school hols).
> >>
> >> 2 Cents,
> >> Ralf
> >>
> >> _______________________________________________
> >> Linux-audio-user mailing list
> >> Linux-audio-user at lists.linuxaudio.org
> >> http://lists.linuxaudio.org/listinfo/linux-audio-user
> >
> >
> >
> > _______________________________________________
> > Linux-audio-user mailing list
> > Linux-audio-user at lists.linuxaudio.org
> > http://lists.linuxaudio.org/listinfo/linux-audio-user
> >
>
> Just to add a bit, more in the music programming side, FWIW, there's Sonic
> PI:
> http://www.cl.cam.ac.uk/projects/raspberrypi/sonicpi/
> https://duckduckgo.com/lite/sonic+pi+children
>
> Although I think it's better tinkering and experimenting, touch and feel
> and hear, as it's already said.
>

Sonic Pi looks fun! I'll have to give that a try, thanks for the suggestion.
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From fons at linuxaudio.org  Mon Aug  4 15:20:11 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Mon, 4 Aug 2014 15:20:11 +0000
Subject: [LAU] First release of zita-njbridge
Message-ID: <20140804152011.GA2136@linuxaudio.org>

Zita-njbridge-0.1.0 is now available.

Zita-j2n and zita-n2j are command line Jack clients to
transmit full quality multichannel audio over a local IP
network, with adaptive resampling by the receiver(s).

Main features:

* One-to-one (UDP) or one-to-many (multicast).
* Sender and receiver(s) can each have their own
  sample rate and period size. No word clock sync
  is assumed.
* Up to 64 channels, 16 or 24 bit or float samples.
* Receiver(s) can select any combination of channels.
* Low latency, optional additional buffering.
* High quality jitter-free resampling.
* Graceful handling of xruns, skipped cycles, lost
  packets and freewheeling.
* IP6 fully supported.
* Requires zita-resampler, no other non-standard 
  dependencies.

Note that zita-njbridge is meant for use on a *local*
network providing more or less reliable delivery with
low to moderate delay. It may work or not on the wider
internet if receiver(s) are configured for additional
buffering, and if you are lucky. Performance on wire-
less networks is just a matter of chance.

You will need a fairly recent Jack version, as the
code uses jack_get_cycle_times() and no fallback for
that is provided.

Download from <http://kokkinizita.linuxaudio.org/linuxaudio/downloads/index.html>

See man zita-njbridge for more info.

-- 
FA

A world of exhaustive, reliable metadata would be a
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From ivan_521521 at yahoo.com  Mon Aug  4 18:57:42 2014
From: ivan_521521 at yahoo.com (Ivan K)
Date: Mon, 4 Aug 2014 11:57:42 -0700
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <20140803160753.GA14272@linuxaudio.org>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408022307460.23832@scott.cbbs.org>
 <20140803095524.GA19816@linuxaudio.org>
 <alpine.DEB.2.02.1408030748440.23832@scott.cbbs.org>
 <20140803160753.GA14272@linuxaudio.org>
Message-ID: <1407178662.24266.YahooMailNeo@web122601.mail.ne1.yahoo.com>



Thanks again to everyone who is helping me out.
I try to work with and digest everything that
people suggest which is why I do not respond
right away.

Let me fill you in on a few problems I have
had over the last day or more.

First: when I was trying to start Jack up
with qjackctl, I would get a window which read:

# [Window]
# Error - JACK Audio Connection Kit:
#
# Could not connect to JACK server as client.
# - Overall operation failed.
# - Unable to connect to server.
# Please check the messages window for more info.

The message window had a lot of text, some of
which was:

# Messages:
# 20:19:52.805 Patchbay deactivated.
# 20:19:52.817 Statistics reset.
# 20:19:52.870 ALSA connection change.
# Cannot connect to server socket err = No such file or directory


After using GOOGLE to try to locate an answer to this
question, I came across this solutions:

? delete {$HOME}/.jackdrc
? delete {$HOME}/.config/rncbc.org/QjackCtl.conf

? make the user that you are trying to run Jack under
? a member of the "audio" group. (I am running Fedora 20
? if that matters, maybe other distributions do not have
? an "audio" group.

With that, Jack was able to start again.

HOWEVER ... I have lost some of my M-Audio cards
functionality. ?

qjackctl's [Connect] window under the Audio tab
USE TO DISPLAY THIS:

On Sat, 2 Aug 2014, Ivan K wrote:
>?
> == ==   Readable Clients/Output Ports
> == ==   - system
> == == ? ? capture_1
> == == ? ? capture_2
>?
> on the right, and then on the left, I see:
>
> == ==   Writable Clients/Input Ports
> == ==   - system
> == == ? ? playback_1
> == == ? ? playback_2
> == == ? ? playback_3
> == == ? ? playback_4
> == == ? ? playback_5
> == == ? ? playback_6
> == == ? ? playback_7
> == == ? ? playback_8

BUT NOW, since my problem with starting JACK up,
the "Writable Clients/Input Ports" box is blank.
I lost my eight "writeable ports" and I am presently
working to get them back.  Any tips would be
appreciated.

Incidentally, what do these eight playback connections
represent.  The "M-Audio Audiophile 2496" only has
three outputs, left and right analog and S/PDIF.
And so do these playbacks 1 through 8 refer to some
sort of mixing channels in the firmware of the
Audiophile 2497?

To repeat, though, I have lost these playbacks 1 through
eight in the Writable Clients/Input Ports box.  How can
I get them back?


Responding directly to others,?
Len Ovens <len at ovenwerks.net> writes:


> This does not look 2496 ish, it looks audiophile 192 ish...?
> This to say that there are two m-audio audiophile models... that use
> two different kernel modules for two different chips inside.

Oh my card is definitely the "M-Audio Audiophile 2496".  It says
so on the box. And the commands aplay and arecord list the device as:

? card 0: M2496 [M Audio Audiophile 24/96], device 0: ICE1712 multi [ICE1712 multi]
? Subdevices: 1/1
? Subdevice #0: subdevice #0


> [...]
> zita-a2j -d hw:0,1 -r 48000 -p 256 -n2 -c 2
> alsa_in -j spdif -d hw:0,1 -r 48000 -p 256 -n 2 -c 2

I will report on my success and failure with the
zita-a2j/alsa_in commands soon.


Ralf Mardorf <ralf.mardorf at rocketmail.com> writes:
>?
> and don't forget to chose 'S/PDIF in', _not_ 48000 for the sound card's
> clock.

I will not loose that bit of information.



Thank you all for your continued help.

From lievenmoors at gmail.com  Mon Aug  4 19:16:16 2014
From: lievenmoors at gmail.com (Lieven Moors)
Date: Mon, 4 Aug 2014 21:16:16 +0200
Subject: [LAU] music software for kids
In-Reply-To: <20140725200153.42c6f38aceada1cd88c980be@gmail.com>
References: <CAG+zz5nrr_+OmZFqtGw0Ye+=RYTrNNzQcVoSJODFUavSjnpxmQ@mail.gmail.com>
 <20140725200153.42c6f38aceada1cd88c980be@gmail.com>
Message-ID: <20140804191616.GA452@satellite>

On Fri, Jul 25, 2014 at 08:01:53PM -0300, Fede wrote:
> > Hi all,
> > 
> > Are there any music applications for linux that would be suitable for a 10
> > year old to sit down and have fun without reading any documentation? (i.e.
> > something as simple as Mario Paint Composer, which I played with and loved
> > as a child)
> > I've scanned the list at http://wiki.linuxaudio.org/apps/start and googled
> > but nothing is jumping out at me.
> > 
> > Thanks!
> > Brian Sorahan


I liked hurtigmixer...

http://www.musikkverksted.no/

greetings,

lieven

From hanaghan.osaudio at gmail.com  Mon Aug  4 19:18:31 2014
From: hanaghan.osaudio at gmail.com (Russell Hanaghan)
Date: Mon, 4 Aug 2014 12:18:31 -0700
Subject: [LAU] First release of zita-njbridge
In-Reply-To: <20140804152011.GA2136@linuxaudio.org>
References: <20140804152011.GA2136@linuxaudio.org>
Message-ID: <877FDB46-E2B5-4B9D-AF80-0C1A8F4536B5@gmail.com>

Hello Mr Fons!

I have lots of time! lol 

I'm curious... This application excites me based on the following theoretical layout:

Budget studio with say at least 1 strong central DAW in a control room. Other satellite rooms tht can be linked with Gig Ethernet and smaller (cheaper) platforms in those rooms for maybe recording certain instruments (drums might be a bitch even at low latencies) an feeding those channels back to the master. Effectively replacing shielded audio cables (which run into real money) for a Cat5e or Cat6 cable and gig ports either side. Gig switches and cards have become cheap, especially compared to half decent shielded pair audio cable. 

Is this reasonable as it relates to the application? 

Thnx for your efforts!

~ Russell

> On Aug 4, 2014, at 8:20 AM, Fons Adriaensen <fons at linuxaudio.org> wrote:
> 
> Zita-njbridge-0.1.0 is now available.
> 
> Zita-j2n and zita-n2j are command line Jack clients to
> transmit full quality multichannel audio over a local IP
> network, with adaptive resampling by the receiver(s).
> 
> Main features:
> 
> * One-to-one (UDP) or one-to-many (multicast).
> * Sender and receiver(s) can each have their own
>  sample rate and period size. No word clock sync
>  is assumed.
> * Up to 64 channels, 16 or 24 bit or float samples.
> * Receiver(s) can select any combination of channels.
> * Low latency, optional additional buffering.
> * High quality jitter-free resampling.
> * Graceful handling of xruns, skipped cycles, lost
>  packets and freewheeling.
> * IP6 fully supported.
> * Requires zita-resampler, no other non-standard 
>  dependencies.
> 
> Note that zita-njbridge is meant for use on a *local*
> network providing more or less reliable delivery with
> low to moderate delay. It may work or not on the wider
> internet if receiver(s) are configured for additional
> buffering, and if you are lucky. Performance on wire-
> less networks is just a matter of chance.
> 
> You will need a fairly recent Jack version, as the
> code uses jack_get_cycle_times() and no fallback for
> that is provided.
> 
> Download from <http://kokkinizita.linuxaudio.org/linuxaudio/downloads/index.html>
> 
> See man zita-njbridge for more info.
> 
> -- 
> FA
> 
> A world of exhaustive, reliable metadata would be a
> It's also a pipe-dream, founded on self-delusion, nerd hubris
> and hysterically inflated market opportunities. (Cory Doctorow)
> 
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

From fons at linuxaudio.org  Mon Aug  4 19:39:20 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Mon, 4 Aug 2014 19:39:20 +0000
Subject: [LAU] First release of zita-njbridge
In-Reply-To: <877FDB46-E2B5-4B9D-AF80-0C1A8F4536B5@gmail.com>
References: <20140804152011.GA2136@linuxaudio.org>
 <877FDB46-E2B5-4B9D-AF80-0C1A8F4536B5@gmail.com>
Message-ID: <20140804193920.GD16288@linuxaudio.org>

On Mon, Aug 04, 2014 at 12:18:31PM -0700, Russell Hanaghan wrote:
 
> I'm curious... This application excites me based on the following theoretical layout:
> 
> Budget studio with say at least 1 strong central DAW in a control room. Other satellite
> rooms tht can be linked with Gig Ethernet and smaller (cheaper) platforms in those rooms
> for maybe recording certain instruments (drums might be a bitch even at low latencies)
> an feeding those channels back to the master. Effectively replacing shielded audio cables
> (which run into real money) for a Cat5e or Cat6 cable and gig ports either side. Gig
> switches and cards have become cheap, especially compared to half decent shielded pair
> audio cable. 
> 
> Is this reasonable as it relates to the application? 

I'm not entirely convinced by the cost argument, unless you'd wire
a lot of channels. You still need a PC and soundcard at the other
end.

But if you can live with the extra latency (which can be kept low 
in particular if you use dedicated NICs and wiring), yes this
could be a use case.

Another use case for a studio would be to make the studio output(s)
available everywhere in the building - offices, bar, etc. 

I originally developed this to be able to record concerts at the
concert hall of the CdM in the studio which is at the other end
of the building and on a different floor. Installing audio cables
or an optical fibre was out of the question, but there is network
wiring everywhere. 


Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From p8rpp at aol.com  Mon Aug  4 20:03:30 2014
From: p8rpp at aol.com (Peter P.)
Date: Mon, 4 Aug 2014 16:03:30 -0400
Subject: [LAU] First release of zita-njbridge
In-Reply-To: <877FDB46-E2B5-4B9D-AF80-0C1A8F4536B5@gmail.com>
References: <20140804152011.GA2136@linuxaudio.org>
 <877FDB46-E2B5-4B9D-AF80-0C1A8F4536B5@gmail.com>
Message-ID: <20140804193210.GA11589@aol.com>

* Russell Hanaghan <hanaghan.osaudio at gmail.com> [2014-08-04 15:18]:
> Hello Mr Fons!
> 
> I have lots of time! lol 
> 
> I'm curious... This application excites me based on the following theoretical layout:
> 
> Budget studio with say at least 1 strong central DAW in a control room. Other satellite rooms tht can be linked with Gig Ethernet and smaller (cheaper) platforms in those rooms for maybe recording certain instruments (drums might be a bitch even at low latencies) an feeding those channels back to the master. Effectively replacing shielded audio cables (which run into real money) for a Cat5e or Cat6 cable and gig ports either side. Gig switches and cards have become cheap, especially compared to half decent shielded pair audio cable. 
> 
> Is this reasonable as it relates to the application? 

Fons' software is amazing and would be the one application up to it I
assume. However the only caveat I can think of right now, is that when
you use headphone monitoring, latencies might be doubled when sending
let's say the Kick to the control room, and then back to another
satellite room. Might all be fine, or turn into a problem.

"2cents"

P

From len at ovenwerks.net  Mon Aug  4 19:59:51 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 4 Aug 2014 12:59:51 -0700 (PDT)
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <1407178662.24266.YahooMailNeo@web122601.mail.ne1.yahoo.com>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408022307460.23832@scott.cbbs.org>
 <20140803095524.GA19816@linuxaudio.org>
 <alpine.DEB.2.02.1408030748440.23832@scott.cbbs.org>
 <20140803160753.GA14272@linuxaudio.org>
 <1407178662.24266.YahooMailNeo@web122601.mail.ne1.yahoo.com>
Message-ID: <alpine.DEB.2.02.1408041202110.23832@scott.cbbs.org>

On Mon, 4 Aug 2014, Ivan K wrote:

> ? make the user that you are trying to run Jack under
> ? a member of the "audio" group. (I am running Fedora 20
> ? if that matters, maybe other distributions do not have
> ? an "audio" group.

Ubuntu does not add users to audio by default as some other distros do, so 
it has to be done manually  :P  Hopefully this will change.

> HOWEVER ... I have lost some of my M-Audio cards
> functionality. ?
>
> qjackctl's [Connect] window under the Audio tab
> USE TO DISPLAY THIS:
>
> On Sat, 2 Aug 2014, Ivan K wrote:
>> ?
>> == ==   Readable Clients/Output Ports
>> == ==   - system
>> == == ? ? capture_1
>> == == ? ? capture_2
>> ?
>> on the right, and then on the left, I see:
>>
>> == ==   Writable Clients/Input Ports
>> == ==   - system
>> == == ? ? playback_1
>> == == ? ? playback_2
>> == == ? ? playback_3
>> == == ? ? playback_4
>> == == ? ? playback_5
>> == == ? ? playback_6
>> == == ? ? playback_7
>> == == ? ? playback_8
>
> BUT NOW, since my problem with starting JACK up,
> the "Writable Clients/Input Ports" box is blank.
> I lost my eight "writeable ports" and I am presently
> working to get them back.  Any tips would be
> appreciated.

Have you tried to see if your s/pdif in works now? That looks like it 
"may" be hw:0,1 jack has been started with.

In qjackctl, setup screen, there is an "Interface:" box in the top(ish) 
right corner, beside that there is a ">". If you click on that, there 
should be a list of devices. Maybe give us a list of those. (full text)

> Incidentally, what do these eight playback connections
> represent.  The "M-Audio Audiophile 2496" only has
> three outputs, left and right analog and S/PDIF.
> And so do these playbacks 1 through 8 refer to some
> sort of mixing channels in the firmware of the
> Audiophile 2497?

The ice1712 has 8 analog outputs and 8 analog inputs as well as s/pdif in 
and out (and some other stuff) It does not monitor the outputs to see if a 
DAC is connected to them and so they all show up. The card manufacture 
chooses how much of this they will use. It seems there is enough 
communication from the ADCs that the chip can tell what is connected or 
not... or maybe the kernel module is more picky with the 2496 than with 
the d66. M-Audio may have added some glue logic that keeps this card from 
working just like the other delta series cards. It may be interesting to 
see if forcing the kernel module to see this as a delta1010 will give you 
full access. (including things that are not there.)

Normally with the ice1712 there should be 12 inputs (8 analog, s/pdif pair 
9 and 10 plus the multi-mixer outputs 11 and 12) and 10 outputs (8 analog 
and one s/pdif pair) In the case of the 2496, inputs 1,2,9,10 and outputs 
1,2,9,10 should be valid. However, it looks like the kernel driver is not 
set up that way. Channels 11 and 12 (inputs) should also be available as 
the output of the internal mixer and all outputs (10 of them) can be used 
as inputs to this intenal monitor mixer. (this sounds nice, but the 
reality is I have never used it  :)  as I have an external mixer already)

try adding a line to /etc/modprob.d/alsa-base.conf (this assumes a debian 
based distro... Ubuntu for example)

options snd_ice1712 delta1010
(or delta1010lt)
(from: 
https://www.kernel.org/doc/Documentation/sound/alsa/ALSA-Configuration.txt 
)

See if jack then shows 12/10 i/o and your s/pdif audio shows up on input 9 
or 10.

>
> Oh my card is definitely the "M-Audio Audiophile 2496".  It says
> so on the box. And the commands aplay and arecord list the device as:
>
> ? card 0: M2496 [M Audio Audiophile 24/96], device 0: ICE1712 multi [ICE1712 multi]
> ? Subdevices: 1/1
> ? Subdevice #0: subdevice #0
>

Then the below should not work:

>> [...]
>> zita-a2j -d hw:0,1 -r 48000 -p 256 -n2 -c 2
>> alsa_in -j spdif -d hw:0,1 -r 48000 -p 256 -n 2 -c 2


--
Len Ovens
www.ovenwerks.net

From len at ovenwerks.net  Mon Aug  4 20:58:49 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 4 Aug 2014 13:58:49 -0700 (PDT)
Subject: [LAU] First release of zita-njbridge
In-Reply-To: <20140804152011.GA2136@linuxaudio.org>
References: <20140804152011.GA2136@linuxaudio.org>
Message-ID: <alpine.DEB.2.02.1408041330000.23832@scott.cbbs.org>

On Mon, 4 Aug 2014, Fons Adriaensen wrote:

> Zita-njbridge-0.1.0 is now available.

Actually 0.1.1 since then.
>
> Zita-j2n and zita-n2j are command line Jack clients to
> transmit full quality multichannel audio over a local IP
> network, with adaptive resampling by the receiver(s).

This works very well with my limited testing.

In response to the query about studio routing use:

If you can afford the i/o cards this will work fine. The latency can be 
very low.

I did:
Audio file -> IF out
plus
Audio file (same jack port) -> zita-j2n -> zita-n2j -> IF out (same port 
as above)

Jack set to -p64.

There was (of course) some comb filtering but I suspect it was not enough 
delay to throw off any musicians if it was used for monitoring. I tried 
both --buff 0 and the default 10. The only difference being the frequency 
of the comb filtering. The only thing to watch is that the same audio 
doesn't get mixed with something that has gone through the link. Or mixing 
two mics, one local and one remote that share the same acoustic space.

Zita-njbridge does not handle on the fly latency changes in jack, so 
switching latency from really low for tracking to higher for mixdown will 
mean restarting the link(s). Though the link probably should not be needed 
for mixing anyway. It wouldn't take much to create a script that 
"respawed" the link if it was needed.

--
Len Ovens
www.ovenwerks.net


From ivan_521521 at yahoo.com  Mon Aug  4 22:34:14 2014
From: ivan_521521 at yahoo.com (Ivan K)
Date: Mon, 4 Aug 2014 15:34:14 -0700
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <1407178662.24266.YahooMailNeo@web122601.mail.ne1.yahoo.com>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408022307460.23832@scott.cbbs.org>
 <20140803095524.GA19816@linuxaudio.org>
 <alpine.DEB.2.02.1408030748440.23832@scott.cbbs.org>
 <20140803160753.GA14272@linuxaudio.org>
 <1407178662.24266.YahooMailNeo@web122601.mail.ne1.yahoo.com>
Message-ID: <1407191654.33899.YahooMailNeo@web122602.mail.ne1.yahoo.com>

I wrote:



> HOWEVER ... I have lost some of my M-Audio cards
> functionality. ?
>?
> qjackctl's [Connect] window under the Audio tab
> USE TO DISPLAY THIS:
>?
> On Sat, 2 Aug 2014, Ivan K wrote:
> >?
> > == ==   Readable Clients/Output Ports
> > == ==   - system
> > == == ? ? capture_1
> > == == ? ? capture_2
> >?
> > on the right, and then on the left, I see:
> >
> > == ==   Writable Clients/Input Ports
> > == ==   - system
> > == == ? ? playback_1
> > == == ? ? playback_2
> > == == ? ? playback_3
> > == == ? ? playback_4
> > == == ? ? playback_5
> > == == ? ? playback_6
> > == == ? ? playback_7
> > == == ? ? playback_8
>?
> BUT NOW, since my problem with starting JACK up,
> the "Writable Clients/Input Ports" box is blank.
> I lost my eight "writeable ports" and I am presently
> working to get them back.  Any tips would be
> appreciated.

I think I figured out why this is so.  Jack is currently configured
to look for "card 0" as displayed in /proc/asound/cards

Yesterday, "card 0" was my M-Audio 2496, and eight
"Writable Clients/Input Ports" were visible in qjackctl.

Today, "card 0" is my webcam and my M-Audio 2496
is "card 1".  My webcam has no playback capability
(just a microphone) and so qjackctl now displays no
"Writable Clients/Input Ports".

I imagine I changed the numbers on my audio devices
with all of the fiddling I have been doing.

Questions:

? How do I change the audio devices back, so my?
? M-Audio card is "card 0".

Or

? How to I get jack/qjackctl to look for "card 1"
? rather than "card 0"?

Thank you all again for your help and patience.

P.S.  I still am interested in the answer to this as well:


> Incidentally, what do these eight playback connections
> represent.  The "M-Audio Audiophile 2496" only has
> three outputs, left and right analog and S/PDIF.
> And so do these playbacks 1 through 8 refer to some
> sort of mixing channels in the firmware of the
> Audiophile 2497?

Thanks.

From gurusonic at gmail.com  Mon Aug  4 23:33:36 2014
From: gurusonic at gmail.com (Roger)
Date: Tue, 05 Aug 2014 09:33:36 +1000
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <1407191654.33899.YahooMailNeo@web122602.mail.ne1.yahoo.com>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408022307460.23832@scott.cbbs.org>
 <20140803095524.GA19816@linuxaudio.org>
 <alpine.DEB.2.02.1408030748440.23832@scott.cbbs.org>
 <20140803160753.GA14272@linuxaudio.org>
 <1407178662.24266.YahooMailNeo@web122601.mail.ne1.yahoo.com>
 <1407191654.33899.YahooMailNeo@web122602.mail.ne1.yahoo.com>
Message-ID: <53E01850.1060308@gmail.com>

On 05/08/14 08:34, Ivan K wrote:
> I think I figured out why this is so.  Jack is currently configured
> to look for "card 0" as displayed in /proc/asound/cards
>
> Yesterday, "card 0" was my M-Audio 2496, and eight
> "Writable Clients/Input Ports" were visible in qjackctl.
>
> Today, "card 0" is my webcam and my M-Audio 2496
> is "card 1".  My webcam has no playback capability
> (just a microphone) and so qjackctl now displays no
> "Writable Clients/Input Ports".
>
> I imagine I changed the numbers on my audio devices
> with all of the fiddling I have been doing.
>
> Questions:
>
>    How do I change the audio devices back, so my
>    M-Audio card is "card 0".
>
Put this in your alsa conf file in /etc/modprobe.d/ :

options snd slots=snd_ice1712,snd_hda_intel
options snd_ice1712 index=0
options snd_hda_intel index=1

This is from my system, obviously replace snd_hda_intel with the driver 
name of your webcam. You could also blacklist the webcam snd driver. 
Check with aplay -l

Cheers
Roger

From ivan_521521 at yahoo.com  Tue Aug  5 01:52:23 2014
From: ivan_521521 at yahoo.com (Ivan K)
Date: Mon, 4 Aug 2014 18:52:23 -0700
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <1407191654.33899.YahooMailNeo@web122602.mail.ne1.yahoo.com>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408022307460.23832@scott.cbbs.org>
 <20140803095524.GA19816@linuxaudio.org>
 <alpine.DEB.2.02.1408030748440.23832@scott.cbbs.org>
 <20140803160753.GA14272@linuxaudio.org>
 <1407178662.24266.YahooMailNeo@web122601.mail.ne1.yahoo.com>
 <1407191654.33899.YahooMailNeo@web122602.mail.ne1.yahoo.com>
Message-ID: <1407203543.32220.YahooMailNeo@web122604.mail.ne1.yahoo.com>



I have successfully recorded from my
USBDualTubePre microphone preamp using the S/PDIF
into my M-Audio 2496.  I used mhWaveEdit.

The issue was that qjackctl was  displaying the
Readable Clients/Output Ports  and
Writable Clients/Input Ports of my _motherboard_
audio, not my M-Audio 2496.

A recent off list reply coached me to set the
correct device in Qjackctl with "Setup | Interface | >".

Early on, Len Ovens informed me that the S/PDIF
inputs of the ice1712 are 9 and 10.

And of course, using envy24control to set "Master clock"
to s/pdif rather than a sample rate was required as well.

And so, thank you to all who responded!

From ken at restivo.org  Tue Aug  5 03:17:46 2014
From: ken at restivo.org (Ken Restivo)
Date: Mon, 4 Aug 2014 20:17:46 -0700
Subject: [LAU] First release of zita-njbridge
In-Reply-To: <20140804193920.GD16288@linuxaudio.org>
References: <20140804152011.GA2136@linuxaudio.org>
 <877FDB46-E2B5-4B9D-AF80-0C1A8F4536B5@gmail.com>
 <20140804193920.GD16288@linuxaudio.org>
Message-ID: <20140805031746.GA6971@tf101>

On Mon, Aug 04, 2014 at 07:39:20PM +0000, Fons Adriaensen wrote:
> On Mon, Aug 04, 2014 at 12:18:31PM -0700, Russell Hanaghan wrote:
>  
> > I'm curious... This application excites me based on the following theoretical layout:
> > 
> > Budget studio with say at least 1 strong central DAW in a control room. Other satellite
> > rooms tht can be linked with Gig Ethernet and smaller (cheaper) platforms in those rooms
> > for maybe recording certain instruments (drums might be a bitch even at low latencies)
> > an feeding those channels back to the master. Effectively replacing shielded audio cables
> > (which run into real money) for a Cat5e or Cat6 cable and gig ports either side. Gig
> > switches and cards have become cheap, especially compared to half decent shielded pair
> > audio cable. 
> > 
> > Is this reasonable as it relates to the application? 
> 
> I'm not entirely convinced by the cost argument, unless you'd wire
> a lot of channels. You still need a PC and soundcard at the other
> end.
> 
> But if you can live with the extra latency (which can be kept low 
> in particular if you use dedicated NICs and wiring), yes this
> could be a use case.
> 
> Another use case for a studio would be to make the studio output(s)
> available everywhere in the building - offices, bar, etc. 
> 
> I originally developed this to be able to record concerts at the
> concert hall of the CdM in the studio which is at the other end
> of the building and on a different floor. Installing audio cables
> or an optical fibre was out of the question, but there is network
> wiring everywhere. 
> 

Wouldn't really require this software, but a friend owns a studio
and I ran a wire for him from his studio to his garage door,
and put an FM transmitter there, so that he can listen to 
mixes in his car in the driveway, and adjust them on the fly. 
His WIFI reaches to the driveway and he has an iPad with screen 
sharing to control his DAW. He loves it; mix right there in the car!

-ken

From hanaghan.osaudio at gmail.com  Tue Aug  5 05:08:12 2014
From: hanaghan.osaudio at gmail.com (Russell Hanaghan)
Date: Mon, 4 Aug 2014 22:08:12 -0700
Subject: [LAU] First release of zita-njbridge
In-Reply-To: <alpine.DEB.2.02.1408041330000.23832@scott.cbbs.org>
References: <20140804152011.GA2136@linuxaudio.org>
 <alpine.DEB.2.02.1408041330000.23832@scott.cbbs.org>
Message-ID: <8A53EDA9-DE73-484B-9BA4-217370D90935@gmail.com>



~ Russell

> On Aug 4, 2014, at 1:58 PM, Len Ovens <len at ovenwerks.net> wrote:
> 
>> On Mon, 4 Aug 2014, Fons Adriaensen wrote:
>> 
>> Zita-njbridge-0.1.0 is now available.
> 
> Actually 0.1.1 since then.
>> 
>> Zita-j2n and zita-n2j are command line Jack clients to
>> transmit full quality multichannel audio over a local IP
>> network, with adaptive resampling by the receiver(s).
> 
> This works very well with my limited testing.
> 
> In response to the query about studio routing use:
> 
> If you can afford the i/o cards this will work fine. The latency can be very low.
> 
> I did:
> Audio file -> IF out
> plus
> Audio file (same jack port) -> zita-j2n -> zita-n2j -> IF out (same port as above)
> 
> Jack set to -p64.
> 
> There was (of course) some comb filtering but I suspect it was not enough delay to throw off any musicians if it was used for monitoring. I tried both --buff 0 and the default 10. The only difference being the frequency of the comb filtering. The only thing to watch is that the same audio doesn't get mixed with something that has gone through the link. Or mixing two mics, one local and one remote that share the same acoustic space.
> 
> Zita-njbridge does not handle on the fly latency changes in jack, so switching latency from really low for tracking to higher for mixdown will mean restarting the link(s). Though the link probably should not be needed for mixing anyway. It wouldn't take much to create a script that "respawed" the link if it was needed.
> 
> --
> 
Just out of curiosity, what kind o data throughput did that create? 100mb links? Switch and or router in the middle? Just curious to the interaction across the wire. 1024mb/s (GigE hardware is more modern, firmware typically closer to current) vs 100mb links. The common theory would be that gig 'pipes' only become relevant when data reaches capacity near or over 100mb. Bit like the 32bit vs 64bit OS argument in some ways. Perhaps the differences will be negligible in the end game. But it might be the difference of running that 64/2  link at 32/2... Too! 

From ralf.mardorf at rocketmail.com  Tue Aug  5 07:10:44 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 05 Aug 2014 09:10:44 +0200
Subject: [LAU] [Bulk] Re:  Record from M-Audio 2496 S/PDIF input
In-Reply-To: <53E01850.1060308@gmail.com>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408022307460.23832@scott.cbbs.org>
 <20140803095524.GA19816@linuxaudio.org>
 <alpine.DEB.2.02.1408030748440.23832@scott.cbbs.org>
 <20140803160753.GA14272@linuxaudio.org>
 <1407178662.24266.YahooMailNeo@web122601.mail.ne1.yahoo.com>
 <1407191654.33899.YahooMailNeo@web122602.mail.ne1.yahoo.com>
 <53E01850.1060308@gmail.com>
Message-ID: <1407222644.1341.1.camel@rocketmail.com>

On Tue, 2014-08-05 at 09:33 +1000, Roger wrote:
> options snd slots=snd_ice1712,snd_hda_intel
> options snd_ice1712 index=0
> options snd_hda_intel index=1

Hi Roger,

you're using the slot method, so the index lines aren't needed.

[rocketmouse at archlinux ~]$ cat /etc/modprobe.d/alsa-base.conf 
# ALSA module ordering
options snd slots=snd_hdspm,snd_ice1712,snd_ice1712

[rocketmouse at archlinux ~]$

I'm aware that some distros provide a default alsa-base.conf with many
lines that are unneeded. For a DAW it can't harm to tidy up audio
setting.

Regards,
Ralf


From gurusonic at gmail.com  Tue Aug  5 10:37:52 2014
From: gurusonic at gmail.com (Roger)
Date: Tue, 05 Aug 2014 20:37:52 +1000
Subject: [LAU] [Bulk] Re:  Record from M-Audio 2496 S/PDIF input
In-Reply-To: <1407222644.1341.1.camel@rocketmail.com>
References: <1406937724.76437.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408011811491.23832@scott.cbbs.org>
 <1406946455.15831.YahooMailNeo@web122604.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408012049070.23832@scott.cbbs.org>
 <53DC8CA8.3030301@hawaii.rr.com>
 <1407012915.59796.YahooMailNeo@web122606.mail.ne1.yahoo.com>
 <alpine.DEB.2.02.1408022307460.23832@scott.cbbs.org>
 <20140803095524.GA19816@linuxaudio.org>
 <alpine.DEB.2.02.1408030748440.23832@scott.cbbs.org>
 <20140803160753.GA14272@linuxaudio.org>
 <1407178662.24266.YahooMailNeo@web122601.mail.ne1.yahoo.com>
 <1407191654.33899.YahooMailNeo@web122602.mail.ne1.yahoo.com>
 <53E01850.1060308@gmail.com> <1407222644.1341.1.camel@rocketmail.com>
Message-ID: <53E0B400.7030107@gmail.com>

On 05/08/14 17:10, Ralf Mardorf wrote:
> On Tue, 2014-08-05 at 09:33 +1000, Roger wrote:
>> options snd slots=snd_ice1712,snd_hda_intel
>> options snd_ice1712 index=0
>> options snd_hda_intel index=1
> Hi Roger,
>
> you're using the slot method, so the index lines aren't needed.
Thanks. It works anyway. :) I will edit.

From raffaele.morelli at gmail.com  Tue Aug  5 12:37:53 2014
From: raffaele.morelli at gmail.com (Raffaele Morelli)
Date: Tue, 5 Aug 2014 14:37:53 +0200
Subject: [LAU] dynamic range analysis tool
Message-ID: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>

Hi,

is there any linux tool which can produce plots like this?
http://i46.tinypic.com/351iyqq.jpg

I am actually using a combination of audacity (export sample data) and R to
plot.

regards
/r
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From fons at linuxaudio.org  Tue Aug  5 13:13:05 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Tue, 5 Aug 2014 13:13:05 +0000
Subject: [LAU] dynamic range analysis tool
In-Reply-To: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
References: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
Message-ID: <20140805131305.GA11181@linuxaudio.org>

On Tue, Aug 05, 2014 at 02:37:53PM +0200, Raffaele Morelli wrote:
 
> is there any linux tool which can produce plots like this?
> http://i46.tinypic.com/351iyqq.jpg

ebur128 can do it, if you change one line:

Change line 218 int ebur128.cc to 

     fprintf (F, "%5.1lf %8.6lf %8.6lf %6d %6d\n", v, nm / km, ns / ks, hm [i], hs [i]);

recompile and install. Then

  ebur128 --prob --lufs somefile.wav

and then in gnuplot

  plot 'ebur128-prob' u 1:4 w i lt 3

or 

  plot 'ebur128-prob' u 1:5 w i lt 3

Ciao,


-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From ralf.mardorf at rocketmail.com  Tue Aug  5 13:16:35 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 05 Aug 2014 15:16:35 +0200
Subject: [LAU] [Bulk]  dynamic range analysis tool
In-Reply-To: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
References: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
Message-ID: <1407244595.25939.1.camel@rocketmail.com>


On Tue, 2014-08-05 at 14:37 +0200, Raffaele Morelli wrote:
> is there any linux tool which can produce plots like this?
> http://i46.tinypic.com/351iyqq.jpg

A plot that does show what ever kind of dB from -85 to -15 on one aches
and for the other aches it displays what ever from 0 to 100 at what ever
time for what ever frequencies?

> I am actually using a combination of audacity (export sample data) and
> R to plot.

I imported a wav and then searched for options to export and "R" the
sample data. I don't understand what you're doing.

Perhaps others are smarter and understand what you you're doing and what
you want to get, but I'm confused.



From ralf.mardorf at rocketmail.com  Tue Aug  5 13:24:21 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 05 Aug 2014 15:24:21 +0200
Subject: [LAU] [Bulk]  dynamic range analysis tool
In-Reply-To: <1407244595.25939.1.camel@rocketmail.com>
References: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
 <1407244595.25939.1.camel@rocketmail.com>
Message-ID: <1407245061.25939.3.camel@rocketmail.com>

On Tue, 2014-08-05 at 15:16 +0200, Ralf Mardorf wrote:
> On Tue, 2014-08-05 at 14:37 +0200, Raffaele Morelli wrote:
> > is there any linux tool which can produce plots like this?
> > http://i46.tinypic.com/351iyqq.jpg
> 
> A plot that does show what ever kind of dB from -85 to -15 on one aches
                                                                    axis
> and for the other aches it displays what ever from 0 to 100 at what ever
                   axis
> time for what ever frequencies?
> 
> > I am actually using a combination of audacity (export sample data) and
> > R to plot.
> 
> I imported a wav and then searched for options to export and "R" the
> sample data. I don't understand what you're doing.
> 
> Perhaps others are smarter and understand what you you're doing and what
> you want to get, but I'm confused.

:D



From raffaele.morelli at gmail.com  Tue Aug  5 13:27:26 2014
From: raffaele.morelli at gmail.com (Raffaele Morelli)
Date: Tue, 5 Aug 2014 15:27:26 +0200
Subject: [LAU] [Bulk] dynamic range analysis tool
In-Reply-To: <1407244595.25939.1.camel@rocketmail.com>
References: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
 <1407244595.25939.1.camel@rocketmail.com>
Message-ID: <CAD4guxO-Zp46mgK-RLascsqw9qiWPiHqM44nAhHPxG3Q91sY3Q@mail.gmail.com>

2014-08-05 15:16 GMT+02:00 Ralf Mardorf <ralf.mardorf at rocketmail.com>:

>
> On Tue, 2014-08-05 at 14:37 +0200, Raffaele Morelli wrote:
> > is there any linux tool which can produce plots like this?
> > http://i46.tinypic.com/351iyqq.jpg
>
> A plot that does show what ever kind of dB from -85 to -15 on one aches
> and for the other aches it displays what ever from 0 to 100 at what ever
> time for what ever frequencies?
>
> > I am actually using a combination of audacity (export sample data) and
> > R to plot.
>
> I imported a wav and then searched for options to export and "R" the
> sample data. I don't understand what you're doing.
>
> Perhaps others are smarter and understand what you you're doing and what
> you want to get, but I'm confused.
>

Ralph, ?R is a statistical analysis software not an audacity feature?
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From ralf.mardorf at rocketmail.com  Tue Aug  5 13:30:44 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 05 Aug 2014 15:30:44 +0200
Subject: [LAU] [Bulk] dynamic range analysis tool
In-Reply-To: <CAD4guxO-Zp46mgK-RLascsqw9qiWPiHqM44nAhHPxG3Q91sY3Q@mail.gmail.com>
References: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
 <1407244595.25939.1.camel@rocketmail.com>
 <CAD4guxO-Zp46mgK-RLascsqw9qiWPiHqM44nAhHPxG3Q91sY3Q@mail.gmail.com>
Message-ID: <1407245444.25939.4.camel@rocketmail.com>

On Tue, 2014-08-05 at 15:27 +0200, Raffaele Morelli wrote:
> Ralph, ?R is a statistical analysis software not an audacity feature?

My bad, Fons seems to have understood it correctly ;).



From raffaele.morelli at gmail.com  Tue Aug  5 13:33:49 2014
From: raffaele.morelli at gmail.com (Raffaele Morelli)
Date: Tue, 5 Aug 2014 15:33:49 +0200
Subject: [LAU] dynamic range analysis tool
In-Reply-To: <20140805131305.GA11181@linuxaudio.org>
References: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
 <20140805131305.GA11181@linuxaudio.org>
Message-ID: <CAD4guxMKBqOdCxNXvedE30-=WEKLn4mk+c_+d1JHg-JW3N8PoA@mail.gmail.com>

2014-08-05 15:13 GMT+02:00 Fons Adriaensen <fons at linuxaudio.org>:

> On Tue, Aug 05, 2014 at 02:37:53PM +0200, Raffaele Morelli wrote:
>
> > is there any linux tool which can produce plots like this?
> > http://i46.tinypic.com/351iyqq.jpg
>
> ebur128 can do it, if you change one line:
>
> Change line 218 int ebur128.cc to
>
>      fprintf (F, "%5.1lf %8.6lf %8.6lf %6d %6d\n", v, nm / km, ns / ks, hm
> [i], hs [i]);
>
> recompile and install. Then
>
>   ebur128 --prob --lufs somefile.wav
>
> and then in gnuplot
>
>   plot 'ebur128-prob' u 1:4 w i lt 3
>
> or
>
>   plot 'ebur128-prob' u 1:5 w i lt 3
>
> Ciao,
>

thank you Fons

/r
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From raffaele.morelli at gmail.com  Tue Aug  5 14:23:23 2014
From: raffaele.morelli at gmail.com (Raffaele Morelli)
Date: Tue, 5 Aug 2014 16:23:23 +0200
Subject: [LAU] dynamic range analysis tool
In-Reply-To: <20140805131305.GA11181@linuxaudio.org>
References: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
 <20140805131305.GA11181@linuxaudio.org>
Message-ID: <CAD4guxOw9OYj0Phric9CTpywxyisdnzcM6Zqd5CHhMW9bJ6W_A@mail.gmail.com>

2014-08-05 15:13 GMT+02:00 Fons Adriaensen <fons at linuxaudio.org>:

> On Tue, Aug 05, 2014 at 02:37:53PM +0200, Raffaele Morelli wrote:
>
> > is there any linux tool which can produce plots like this?
> > http://i46.tinypic.com/351iyqq.jpg
>
> ebur128 can do it, if you change one line:
>
> Change line 218 int ebur128.cc to
>
>      fprintf (F, "%5.1lf %8.6lf %8.6lf %6d %6d\n", v, nm / km, ns / ks, hm
> [i], hs [i]);
>
> recompile and install. Then
>
>   ebur128 --prob --lufs somefile.wav
>
> and then in gnuplot
>
>   plot 'ebur128-prob' u 1:4 w i lt 3
>
> or
>
>   plot 'ebur128-prob' u 1:5 w i lt 3
>
> Ciao,


?Ok, works but I have a couple of question.

I see only 751 records in the outfile. How ebur128 works? Is the file
scanned entirely and how?

/r

PS apologize but I can't dig into C code :-(
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From fons at linuxaudio.org  Tue Aug  5 14:49:59 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Tue, 5 Aug 2014 14:49:59 +0000
Subject: [LAU] dynamic range analysis tool
In-Reply-To: <CAD4guxOw9OYj0Phric9CTpywxyisdnzcM6Zqd5CHhMW9bJ6W_A@mail.gmail.com>
References: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
 <20140805131305.GA11181@linuxaudio.org>
 <CAD4guxOw9OYj0Phric9CTpywxyisdnzcM6Zqd5CHhMW9bJ6W_A@mail.gmail.com>
Message-ID: <20140805144959.GB8313@linuxaudio.org>

On Tue, Aug 05, 2014 at 04:23:23PM +0200, Raffaele Morelli wrote:
 
> Ok, works but I have a couple of question.
> 
> I see only 751 records in the outfile. How ebur128 works? Is the file
> scanned entirely and how?

Yes, the entire audio file is processed. Internally ebumeter and ebur128
compute two histograms, each consisting of 751 bins from -70 to +5 dB in
steps of 0.1 dB. 

The first histogram is the 'momentary' loudness, the average over
a sliding window of 400 ms. The second is the 'short-term' loudness,
computed using a sliding window of 3 seconds. 

The histograms are then used to compute average loudness and
loudness range according to the EBU-R128 standard. Ebumeter
will show them in real-time. 

For the details, see my LAC 2011 paper which you can find here:
<http://kokkinizita.linuxaudio.org/papers/index.html>.

The output file of ebur128 consists of five columns:

1. X-axis, level in dB
2. Cumulative probability of the momentary loudness, this is
   the histogram integrated and normalised to 1. In other words
   the probability that the level is less than the X-axis value.
3. Same for the short-term loudness.
4. The momentary loudness histogram, counts for each bin.
5. Same for the short-term loudnes.

The gnuplot commands I posted will plot 1 vs 4 or 1 vs 5.

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From chris at chriscaudle.org  Tue Aug  5 15:29:47 2014
From: chris at chriscaudle.org (Chris Caudle)
Date: Tue, 5 Aug 2014 10:29:47 -0500
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <mailman.12.1407242281.3355.linux-audio-user@lists.linuxaudio.org>
References: <mailman.12.1407242281.3355.linux-audio-user@lists.linuxaudio.org>
Message-ID: <da35909808ca3b955c09443b31ea9108.squirrel@email.powweb.com>

> From: Ivan K <ivan_521521 at yahoo.com>
>> HOWEVER ... I have lost some of my M-Audio cards
>> functionality. ?

> I think I figured out why this is so.  Jack is currently configured
> to look for "card 0" as displayed in /proc/asound/cards

The number changes if you have hot plug devices such as USB sound devices
(which includes webcams).  It could also change if the kernel or ALSA
changed the order of device enumeration between version upgrades.

You should be using the name and not number.  See this FAQ from
jackaudio.org:
http://jackaudio.org/faq/device_naming.html

For the Audiophile 2496 you would use hw:M2496 as the device instead of
hw:0 or hw:1.

-- 
Chris Caudle



From robin at gareus.org  Tue Aug  5 15:39:32 2014
From: robin at gareus.org (Robin Gareus)
Date: Tue, 05 Aug 2014 17:39:32 +0200
Subject: [LAU] dynamic range analysis tool
In-Reply-To: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
References: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
Message-ID: <53E0FAB4.40601@gareus.org>

On 08/05/2014 02:37 PM, Raffaele Morelli wrote:
> Hi,
> 
> is there any linux tool which can produce plots like this?
> http://i46.tinypic.com/351iyqq.jpg
> 

Not a plot exactly like the linked one, but the same information on a
circle can be produced live with meters.lv2. See the left image at
https://raw.githubusercontent.com/x42/meters.lv2/master/doc/LV2ebur128.png
for an example.

It uses Fons' ebur128 under the hood which he already explained.

You can get it from https://github.com/x42/meters.lv2 which is also
available on most GNU/Linux distributions part of the x42-plugins package.

Cheers!
robin

From kevinc at cosgroves.us  Tue Aug  5 23:22:46 2014
From: kevinc at cosgroves.us (Kevin Cosgrove)
Date: Tue, 05 Aug 2014 16:22:46 -0700
Subject: [LAU] [Bulk] Re: Measuring the acoustical characteristics of my
	studio using FLOSS software?
In-Reply-To: <CA+O6s5y1E4ULbCgYOkSzJ7UnefZTuvAXtc8jyjKQYiHQpM7Fuw@mail.gmail.com>
Message-ID: <20140805232246.99281BE06A@joseph.cosgroves.us>


I completely agree with Fons' point about wavelength and its 
relationship to objects within the room.  But, I've also seen
this mirror trick, "image method", in audio literature, for instance:

http://www.sgm-audio.com/research/rir/rir.html  See note [1]

https://www.cs.princeton.edu/courses/archive/fall00/cs426/lectures/acoustics/acoustics.pdf

I think I also saw it in Philip Newell's studio design book.

To Fons' point, none of what I've ever seen on this topic ever
included room contents, and this has always been presented as an
_efficient_ approximation.  For that efficiency some accuracy has
to be left out.

My sense is that this might have some usefulness, but not be the 
final word on the subject.

Cheerio...

--
Kevin



From kevinc at cosgroves.us  Tue Aug  5 23:28:40 2014
From: kevinc at cosgroves.us (Kevin Cosgrove)
Date: Tue, 05 Aug 2014 16:28:40 -0700
Subject: [LAU] Measuring the acoustical characteristics of my studio
	using FLOSS software?
In-Reply-To: <4738547.fyS0WoeaMf@jca-studio>
Message-ID: <20140805232841.0469DBE06A@joseph.cosgroves.us>


On 4 August 2014 at 1:05, "Jostein Chr. Andersen" <jostein at vait.se> wrote:

> First of all, no matter if the room is perfect or not: When you
> mix, do it at an amplitude where you can speak normally.  That
> way, the room's impact is at a minimum and your ears will be
> very happy! Everything "sounds good" when you play loader. If
> it does not sound good when you play at a low level, then it's
> not finished yet.

Don't forget about Fletcher-Munson, and more modern equal loudness 
curves, when choosing your mixing volume.  I choose to mix in the 
85-90dB range because of this.

Cheerio...

--
Kevin



From kevinc at cosgroves.us  Tue Aug  5 23:44:47 2014
From: kevinc at cosgroves.us (Kevin Cosgrove)
Date: Tue, 05 Aug 2014 16:44:47 -0700
Subject: [LAU] Record from M-Audio 2496 S/PDIF input
In-Reply-To: <1407203543.32220.YahooMailNeo@web122604.mail.ne1.yahoo.com>
Message-ID: <20140805234447.E69F4BE06A@joseph.cosgroves.us>


On 4 August 2014 at 18:52, Ivan K <ivan_521521 at yahoo.com> wrote:
 
> I have successfully recorded from my
> USBDualTubePre microphone preamp using the S/PDIF
> into my M-Audio 2496.  I used mhWaveEdit.
> 
> The issue was that qjackctl was  displaying the
> Readable Clients/Output Ports  and
> Writable Clients/Input Ports of my _motherboard_
> audio, not my M-Audio 2496.
> 
> A recent off list reply coached me to set the
> correct device in Qjackctl with "Setup | Interface | >".
> 
> Early on, Len Ovens informed me that the S/PDIF
> inputs of the ice1712 are 9 and 10.
> 
> And of course, using envy24control to set "Master clock"
> to s/pdif rather than a sample rate was required as well.
> 
> And so, thank you to all who responded!

Thanks for the discussion and your conclusions.  I'm about to try
to transfer the content of an audio DAT tape using a Panasonic
DAT deck via the S/PDIF i/o on my Delta 1010 audio interface.
You all may have just saved me some headaches.  FWIW, I'll be
attempting this using Ardour, which plays quite nicely with
JACK[12], and plays quite nicely on my system in general.  But,
S/PDIF is new to me.

Cheerio....

--
Kevin



From ralf.mardorf at rocketmail.com  Wed Aug  6 00:25:02 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 06 Aug 2014 02:25:02 +0200
Subject: [LAU] [Bulk] Re: Measuring the acoustical characteristics of my
 studio using FLOSS software?
In-Reply-To: <20140805232841.0469DBE06A@joseph.cosgroves.us>
References: <20140805232841.0469DBE06A@joseph.cosgroves.us>
Message-ID: <1407284702.27381.5.camel@rocketmail.com>

On Tue, 2014-08-05 at 16:28 -0700, Kevin Cosgrove wrote:
> Fletcher-Munson

In the end you need to control your mix at different volumes, but doing
the initial mix at household noise level is good not only regarding
reflections from the walls, it's the best you can do, assumed you want
to be an audio engineer for your whole life and not to become hearing
impaired in the way as most of the young humans under the age of 20
nowadays are.


From ralf.mardorf at rocketmail.com  Wed Aug  6 00:53:05 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 06 Aug 2014 02:53:05 +0200
Subject: [LAU] [Bulk] Re: Measuring the acoustical characteristics of my
 studio using FLOSS software?
In-Reply-To: <1407284702.27381.5.camel@rocketmail.com>
References: <20140805232841.0469DBE06A@joseph.cosgroves.us>
 <1407284702.27381.5.camel@rocketmail.com>
Message-ID: <1407286385.27381.7.camel@rocketmail.com>

On Wed, 2014-08-06 at 02:25 +0200, Ralf Mardorf wrote:
> On Tue, 2014-08-05 at 16:28 -0700, Kevin Cosgrove wrote:
> > Fletcher-Munson
> 
> In the end you need to control your mix at different volumes, but doing
> the initial mix at household noise level is good not only regarding
> reflections from the walls, it's the best you can do, assumed you want
> to be an audio engineer for your whole life and not to become hearing
> impaired in the way as most of the young humans under the age of 20
> nowadays are.

Btw. unfortunately I'm used to volumes louder than household noise level
even for the initial mix too. Not only our hearing needs louder volumes,
but also speakers and even headphones aren't that good at lower volumes.
Neither our hearing nor the electrical technology is good at low
volumes, OTOH household level should be the volume we should be used
too. If we need it louder, we already have a problem with our health.

We perhaps aren't hearing impaired, but we likely suffer from some kind
of tinnitus, high blood pressure, lack of concentration etc.. It seems
to be, that the louder the music is, the less affect is caused by the
differences between the hearing of our left and right ears to our
perception.



From ralf.mardorf at rocketmail.com  Wed Aug  6 01:54:36 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 06 Aug 2014 03:54:36 +0200
Subject: [LAU] Perception - Was: Measuring the acoustical characteristics of
 my studio using FLOSS software?
In-Reply-To: <1407286385.27381.7.camel@rocketmail.com>
References: <20140805232841.0469DBE06A@joseph.cosgroves.us>
 <1407284702.27381.5.camel@rocketmail.com>
 <1407286385.27381.7.camel@rocketmail.com>
Message-ID: <1407290076.27381.10.camel@rocketmail.com>

With my left ear I'm able to hear some frequencies or at least one
frequency at very low volumes I'm unable to hear with my right ear, so
technically it's my better ear, but with my right ear understanding the
content of speech and music is easier to do. Perhaps "content of music"
isn't a good phrasing, but similar to the content of speech, there's a
difference for the perception of music by comparing my left with my
right ear/hearing that seems to be independent of the technical ability
of hearing. Wit/reason is interacting different with perception from the
left and right ear.

JFTR my hearing with both ears is ok, there isn't a difference regarding
physical health of the ears. Regarding the brain, I'm right-hander, but
dyslexic. Being a left-hander or dyslexic seems to have impact to
artistically mind.

When mixing music I not only switch between stereo and mono, but I also
change the left and right channel.

I wonder how clearly the difference between left and right ear
perception/understanding is for others?!

Regards,
Ralf


From ralf.mardorf at rocketmail.com  Wed Aug  6 01:57:32 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 06 Aug 2014 03:57:32 +0200
Subject: [LAU] Perception - Was: Measuring the acoustical
 characteristics of my studio using FLOSS software?
In-Reply-To: <1407290076.27381.10.camel@rocketmail.com>
References: <20140805232841.0469DBE06A@joseph.cosgroves.us>
 <1407284702.27381.5.camel@rocketmail.com>
 <1407286385.27381.7.camel@rocketmail.com>
 <1407290076.27381.10.camel@rocketmail.com>
Message-ID: <1407290252.27381.12.camel@rocketmail.com>

PS: I see colours, when I listen to music, but I wouldn't call it real
synesthesia.


From ralf.mardorf at rocketmail.com  Wed Aug  6 02:17:55 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 06 Aug 2014 04:17:55 +0200
Subject: [LAU] Perception - Was: Measuring the acoustical
 characteristics of my studio using FLOSS software?
In-Reply-To: <1407291375.31541.1.camel@alice-dsl.net>
References: <20140805232841.0469DBE06A@joseph.cosgroves.us>
 <1407284702.27381.5.camel@rocketmail.com>
 <1407286385.27381.7.camel@rocketmail.com>
 <1407290076.27381.10.camel@rocketmail.com>
 <1407290252.27381.12.camel@rocketmail.com>
 <1407291375.31541.1.camel@alice-dsl.net>
Message-ID: <1407291475.31541.2.camel@rocketmail.com>

On Wed, 2014-08-06 at 03:57 +0200, Ralf Mardorf wrote:
> PS: I see colours, when I listen to music, but I wouldn't call it real
> synesthesia.

PPS: I don't see always the same colour for an oboe playing one tone,
but I see coloured films about the content. It's a pity that e.g. for
Qtractor it's impossible to select a wanted colour for tracks, it's only
possible to select a vague colour. For me, it would be very informing,
if I really would get a colour of choice.


From atte at youmail.dk  Wed Aug  6 22:01:35 2014
From: atte at youmail.dk (Atte)
Date: Thu, 07 Aug 2014 00:01:35 +0200
Subject: [LAU] Slowing down audio while keeping same pitch ?
In-Reply-To: <20140729204713.07fcc456@mistral>
References: <20140729204713.07fcc456@mistral>
Message-ID: <53E2A5BF.6030300@youmail.dk>

On 07/30/2014 02:47 AM, jonetsu at teksavvy.com wrote:
> Is it possible to slow down an audio file (mp3 for instance) while
> keeping the same pitch ?

I have to share a funny story from the beginning of time:

Years ago I was doing electronic music with a friend. We only dreamed
about time-stretching until we finally we able to get the (then new)
akai S1000, that could do time stretching. We never heard the effect but
had fantasized about how we could use it musically for years.

So first thing to do on the akai: Sample a break beat and stretch it.
Tiny display, no idea what parameters meant, so my friend dialed in some
numbers and hit "do it". I honestly think we waited for two hours for
the akai to process the break beat. Ancious to hear the result, hit a
note and it went wwwwhhhhhhhhiiiiizzzsssssssiiiiiiiiiizzzzzzzzzzuuuuu
(insert 30 seconds of insane digital
glitch...)-boom-boom-chack-boom-boom-chack (...followed by one bar of
super cool, normal tempo breakbeat.).

Turned out he asked the akai to stretch the first 100 samples or so by
several thousand percent :-)

-- 
Atte

http://atte.dk   http://modlys.dk

From czhenry at gmail.com  Thu Aug  7 02:00:43 2014
From: czhenry at gmail.com (Charles Z Henry)
Date: Wed, 6 Aug 2014 21:00:43 -0500
Subject: [LAU] Perception - Was: Measuring the acoustical
 characteristics of my studio using FLOSS software?
In-Reply-To: <1407290076.27381.10.camel@rocketmail.com>
References: <20140805232841.0469DBE06A@joseph.cosgroves.us>
 <1407284702.27381.5.camel@rocketmail.com>
 <1407286385.27381.7.camel@rocketmail.com>
 <1407290076.27381.10.camel@rocketmail.com>
Message-ID: <CAPfmNOGj9q6RYRNu2RSjpMs3vJ-=d817ynoZO2=PceA+04icpQ@mail.gmail.com>

On Tue, Aug 5, 2014 at 8:54 PM, Ralf Mardorf
<ralf.mardorf at rocketmail.com> wrote:
> With my left ear I'm able to hear some frequencies or at least one
> frequency at very low volumes I'm unable to hear with my right ear, so
> technically it's my better ear, but with my right ear understanding the
> content of speech and music is easier to do. Perhaps "content of music"
> isn't a good phrasing, but similar to the content of speech, there's a
> difference for the perception of music by comparing my left with my
> right ear/hearing that seems to be independent of the technical ability
> of hearing. Wit/reason is interacting different with perception from the
> left and right ear.
>
> JFTR my hearing with both ears is ok, there isn't a difference regarding
> physical health of the ears. Regarding the brain, I'm right-hander, but
> dyslexic. Being a left-hander or dyslexic seems to have impact to
> artistically mind.
>
> When mixing music I not only switch between stereo and mono, but I also
> change the left and right channel.
>
> I wonder how clearly the difference between left and right ear
> perception/understanding is for others?!
>
> Regards,
> Ralf

What you're describing is the right-ear advantage for perception of
speech.  This occurs among people with strongly left-hemisphere speech
function lateralization.  I think it's a relatively small effect size
(the variance is large by comparison and experiments require a large
number of subjects to get enough experimental power).

Not all people have left-hemisphere speech lateralization, although it
is the most common (and strongly wired from birth).  A fewer number of
people have right-hemisphere language dominance and some people have
no dominant hemisphere for language.  The well known Wernicke's area
is typically found in the left hemisphere planum temporale (secondary
auditory cortex adjacent to heschl's gyrus).

Language lateralization is strongly correlated with handedness.  There
are many studies of auditory attention in which the results of
left-handed people do not match with right-handed people.

There are ways to measure lateralization: my favorite is probably
trans-cranial doppler ultrasound sonography.  There is also the
amobarbitol test which is used prior to brain surgery to avoid
removing parts of the brain which are necessary for language.

I had written a paper about this subject back in 2006 (my final paper
in cognitive psychology).  My hypothesis in the paper was that
lateralization facilitates auditory attention.  Lesser degrees of
lateralization translate into less effective inhibition of
task-unrelated stimuli, according to the Triesman attention model, and
the corresponding ways in which auditory attention is allocated: by
spatial location, by spectrum, by content, and by temporal
characteristics.

I'd be happy to provide anyone with the studies and papers I have read
(or a copy of the paper I had written which includes the sources).
Disclaimer: I am not a current expert in the field, only a student,
and my knowledge may be a little dated.

From czhenry at gmail.com  Thu Aug  7 02:18:29 2014
From: czhenry at gmail.com (Charles Z Henry)
Date: Wed, 6 Aug 2014 21:18:29 -0500
Subject: [LAU] Perception - Was: Measuring the acoustical
 characteristics of my studio using FLOSS software?
In-Reply-To: <CAPfmNOGj9q6RYRNu2RSjpMs3vJ-=d817ynoZO2=PceA+04icpQ@mail.gmail.com>
References: <20140805232841.0469DBE06A@joseph.cosgroves.us>
 <1407284702.27381.5.camel@rocketmail.com>
 <1407286385.27381.7.camel@rocketmail.com>
 <1407290076.27381.10.camel@rocketmail.com>
 <CAPfmNOGj9q6RYRNu2RSjpMs3vJ-=d817ynoZO2=PceA+04icpQ@mail.gmail.com>
Message-ID: <CAPfmNOE+j_brOkRo0puvtu=MRk_M4zo56xY3-=neBtPf93xMYg@mail.gmail.com>

On Wed, Aug 6, 2014 at 9:00 PM, Charles Z Henry <czhenry at gmail.com> wrote:
> On Tue, Aug 5, 2014 at 8:54 PM, Ralf Mardorf
> <ralf.mardorf at rocketmail.com> wrote:
>> With my left ear I'm able to hear some frequencies or at least one
>> frequency at very low volumes I'm unable to hear with my right ear, so
>> technically it's my better ear, but with my right ear understanding the
>> content of speech and music is easier to do. Perhaps "content of music"
>> isn't a good phrasing, but similar to the content of speech, there's a
>> difference for the perception of music by comparing my left with my
>> right ear/hearing that seems to be independent of the technical ability
>> of hearing. Wit/reason is interacting different with perception from the
>> left and right ear.
>>
>> JFTR my hearing with both ears is ok, there isn't a difference regarding
>> physical health of the ears. Regarding the brain, I'm right-hander, but
>> dyslexic. Being a left-hander or dyslexic seems to have impact to
>> artistically mind.
>>
>> When mixing music I not only switch between stereo and mono, but I also
>> change the left and right channel.
>>
>> I wonder how clearly the difference between left and right ear
>> perception/understanding is for others?!
>>
>> Regards,
>> Ralf
>
> What you're describing is the right-ear advantage for perception of
> speech.  This occurs among people with strongly left-hemisphere speech
> function lateralization.  I think it's a relatively small effect size
> (the variance is large by comparison and experiments require a large
> number of subjects to get enough experimental power).
>
> Not all people have left-hemisphere speech lateralization, although it
> is the most common (and strongly wired from birth).  A fewer number of
> people have right-hemisphere language dominance and some people have
> no dominant hemisphere for language.  The well known Wernicke's area
> is typically found in the left hemisphere planum temporale (secondary
> auditory cortex adjacent to heschl's gyrus).
>
> Language lateralization is strongly correlated with handedness.  There
> are many studies of auditory attention in which the results of
> left-handed people do not match with right-handed people.
>
> There are ways to measure lateralization: my favorite is probably
> trans-cranial doppler ultrasound sonography.  There is also the
> amobarbitol test which is used prior to brain surgery to avoid
> removing parts of the brain which are necessary for language.
>
> I had written a paper about this subject back in 2006 (my final paper
> in cognitive psychology).  My hypothesis in the paper was that
> lateralization facilitates auditory attention.  Lesser degrees of
> lateralization translate into less effective inhibition of
> task-unrelated stimuli, according to the Triesman attention model, and
> the corresponding ways in which auditory attention is allocated: by
> spatial location, by spectrum, by content, and by temporal
> characteristics.
>
> I'd be happy to provide anyone with the studies and papers I have read
> (or a copy of the paper I had written which includes the sources).
> Disclaimer: I am not a current expert in the field, only a student,
> and my knowledge may be a little dated.

P.S.  There is also the matter of the auditory chiasm which provides
some sub-cortical processing in the auditory mid-brain.  Information
from each ear is in fact transmitted to both hemispheres, but it is
unclear how much information is sent ipsi-laterally and
contra-laterally and whether some lateralized processing occurs before
the hemispheres receive the information.  The main structures involved
in spatial localization are the olivary complex located in the pons
and the superior colliculus which analyzes differences in time of
arrival between the two ears.

The premise of the right-ear advantage for speech is that information
is strongly transmitted contra-laterally by the auditory mid-brain and
weakly ipsi-laterally.

From czhenry at gmail.com  Thu Aug  7 02:36:34 2014
From: czhenry at gmail.com (Charles Z Henry)
Date: Wed, 6 Aug 2014 21:36:34 -0500
Subject: [LAU] Perception - Was: Measuring the acoustical
 characteristics of my studio using FLOSS software?
In-Reply-To: <1407290076.27381.10.camel@rocketmail.com>
References: <20140805232841.0469DBE06A@joseph.cosgroves.us>
 <1407284702.27381.5.camel@rocketmail.com>
 <1407286385.27381.7.camel@rocketmail.com>
 <1407290076.27381.10.camel@rocketmail.com>
Message-ID: <CAPfmNOEAD+B5SiHWPK7bMTeTHvXZXF6GtuW1fDK7TujC25jW2w@mail.gmail.com>

On Tue, Aug 5, 2014 at 8:54 PM, Ralf Mardorf
<ralf.mardorf at rocketmail.com> wrote:
> With my left ear I'm able to hear some frequencies or at least one
> frequency at very low volumes I'm unable to hear with my right ear, so
> technically it's my better ear, but with my right ear understanding the
> content of speech and music is easier to do. Perhaps "content of music"
> isn't a good phrasing, but similar to the content of speech, there's a
> difference for the perception of music by comparing my left with my
> right ear/hearing that seems to be independent of the technical ability
> of hearing. Wit/reason is interacting different with perception from the
> left and right ear.
>
> JFTR my hearing with both ears is ok, there isn't a difference regarding
> physical health of the ears. Regarding the brain, I'm right-hander, but
> dyslexic. Being a left-hander or dyslexic seems to have impact to
> artistically mind.

Dyslexia is a often misunderstood condition, which has to do with
deficits in phonological processing, not lateralization.  There are
two primary "routes" to reading: one whole-word (holistic) and the
other phonological.  There are regions of the brain that respond
differently to words, pseudo-words (phonologically plausible strings),
and non-words (non-phonologically plausible strings).  For dyslexics,
there is an apparent disconnect between visually parsing strings and
retrieving the sounds of phonemes.  Readers with dyslexia often
retrieve the meaning of words as a whole rather than repeating the
sounds of words to themselves.

> When mixing music I not only switch between stereo and mono, but I also
> change the left and right channel.
>
> I wonder how clearly the difference between left and right ear
> perception/understanding is for others?!
>
> Regards,
> Ralf

From paul at linuxaudiosystems.com  Thu Aug  7 02:52:08 2014
From: paul at linuxaudiosystems.com (Paul Davis)
Date: Wed, 6 Aug 2014 22:52:08 -0400
Subject: [LAU] Slowing down audio while keeping same pitch ?
In-Reply-To: <20140802102643.4d3d9a7b@mistral>
References: <20140729204713.07fcc456@mistral>
 <20140730071634.766508c9@shams.smbolton.com>
 <CAAAW3BWrX9BxL7c-Wou1D+W-Da50sWOpGnsZF31RFw_fdPL4EA@mail.gmail.com>
 <20140802102643.4d3d9a7b@mistral>
Message-ID: <CAFa_cKmTcbMakY2fjtXO+iM2vZSArQ+ZkCMWQVD0X=HJAnHLTA@mail.gmail.com>

On Sat, Aug 2, 2014 at 10:26 AM, jonetsu at teksavvy.com <jonetsu at teksavvy.com>
wrote:

> On Wed, 30 Jul 2014 14:04:42 -0500,
> Neil <djdualcore at gmail.com> wrote :
>
> > Ardour
> > Audacity
> > Mixxx
> > VLC Media Player
>
> > Some of these won't do it directly on an MP3 but will happily do it
> > on a copy in another format.
>
> Ardour can play constant pitch slowed-down audio ?
>

switch to timefx tool ('t')
stretch region to new length
pitch remains unchanged
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From czhenry at gmail.com  Thu Aug  7 03:01:02 2014
From: czhenry at gmail.com (Charles Z Henry)
Date: Wed, 6 Aug 2014 22:01:02 -0500
Subject: [LAU] Perception - Was: Measuring the acoustical
 characteristics of my studio using FLOSS software?
In-Reply-To: <CAPfmNOE+j_brOkRo0puvtu=MRk_M4zo56xY3-=neBtPf93xMYg@mail.gmail.com>
References: <20140805232841.0469DBE06A@joseph.cosgroves.us>
 <1407284702.27381.5.camel@rocketmail.com>
 <1407286385.27381.7.camel@rocketmail.com>
 <1407290076.27381.10.camel@rocketmail.com>
 <CAPfmNOGj9q6RYRNu2RSjpMs3vJ-=d817ynoZO2=PceA+04icpQ@mail.gmail.com>
 <CAPfmNOE+j_brOkRo0puvtu=MRk_M4zo56xY3-=neBtPf93xMYg@mail.gmail.com>
Message-ID: <CAPfmNOF+-FsPfOncYwY86j3jR6vzyE3qBMkuCeGJ_EXcJCo8Ug@mail.gmail.com>

On Wed, Aug 6, 2014 at 9:18 PM, Charles Z Henry <czhenry at gmail.com> wrote:
> On Wed, Aug 6, 2014 at 9:00 PM, Charles Z Henry <czhenry at gmail.com> wrote:

> P.S.  There is also the matter of the auditory chiasm which provides
> some sub-cortical processing in the auditory mid-brain.  Information
> from each ear is in fact transmitted to both hemispheres, but it is
> unclear how much information is sent ipsi-laterally and
> contra-laterally and whether some lateralized processing occurs before
> the hemispheres receive the information.  The main structures involved
> in spatial localization are the olivary complex located in the pons
> and the superior colliculus which analyzes differences in time of
> arrival between the two ears.

| sed 's/superior colliculus/inferior colliculus/'

The superior colliculus is primarily implicated in visual processing
and has more to do with programming eye movements and attending to
visual objects.  Of course, the fact that they are right next to one
another probably means some information is shared between the two.

From ralf.mardorf at rocketmail.com  Thu Aug  7 15:13:06 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Thu, 07 Aug 2014 17:13:06 +0200
Subject: [LAU] Perception - Was: Measuring the acoustical
 characteristics of my studio using FLOSS software?
In-Reply-To: <CAPfmNOGj9q6RYRNu2RSjpMs3vJ-=d817ynoZO2=PceA+04icpQ@mail.gmail.com>
References: <20140805232841.0469DBE06A@joseph.cosgroves.us>
 <1407284702.27381.5.camel@rocketmail.com>
 <1407286385.27381.7.camel@rocketmail.com>
 <1407290076.27381.10.camel@rocketmail.com>
 <CAPfmNOGj9q6RYRNu2RSjpMs3vJ-=d817ynoZO2=PceA+04icpQ@mail.gmail.com>
Message-ID: <1407424386.3427.5.camel@rocketmail.com>

On Wed, 2014-08-06 at 21:00 -0500, Charles Z Henry wrote:
> I'd be happy to provide anyone with the studies and papers I have read
> (or a copy of the paper I had written which includes the sources).

An interesting read :). Chuck sent it off-list.



From ken at restivo.org  Fri Aug  8 01:24:41 2014
From: ken at restivo.org (Ken Restivo)
Date: Thu, 7 Aug 2014 18:24:41 -0700
Subject: [LAU] Why does PHASEX sound so damn good?
Message-ID: <20140808012441.GA9925@q400a.mobile.restivo.org>

Rolling by on randomize came a Me and My Cronies jam/joke from years ago:

http://www.restivo.org/blog/podpress_trac/web/558/0/Not_OK_Computer.ogg

And I was struck by how much PHASEX sounds like a real analog synth, like an ARP 2600 or similar, and so much more real than any other software synths I've used.

It sounds so... raw, uncontrolled, well, ANALOG. Most software simulations sound more or less authentic, but all so much more "tame", for want of a better term. But PHASEX always sounded to me (and felt, as I was playing with it) that at any moment it could do something crazy like throw a DC offset, to into an uncontrollable oscillation, or blow up my speakers, etc.

I don't like being at a loss for precise, engineering terms, or understanding WHY something is, so I'm asking any of the DSP'ers here who might also have looked at (and understood) PHASEX's source.

Any ideas what is so different about PHASEX, and what might be this quality of it's sound I could be trying to describe?

-ken

From ken at restivo.org  Fri Aug  8 01:36:25 2014
From: ken at restivo.org (Ken Restivo)
Date: Thu, 7 Aug 2014 18:36:25 -0700
Subject: [LAU] One more note on PHASEX and that track
Message-ID: <20140808013625.GB9925@q400a.mobile.restivo.org>

Annotations for that track (http://www.restivo.org/blog/podpress_trac/web/558/0/Not_OK_Computer.ogg)

PHASEX is present/prevalent from the intro through 05:59.
WhySynth is from 6:00 through 6:58
Nekostring is 07:48 through 07:55
AMS is 07:59 through 10:00

PHASEX sounds very much to me like a real analog synth. The rest sound very much like software simulations. I have no idea why the difference seems so striking.

-ken

From raffaele.morelli at gmail.com  Fri Aug  8 06:46:54 2014
From: raffaele.morelli at gmail.com (Raffaele Morelli)
Date: Fri, 8 Aug 2014 08:46:54 +0200
Subject: [LAU] dynamic range analysis tool
In-Reply-To: <20140805144959.GB8313@linuxaudio.org>
References: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
 <20140805131305.GA11181@linuxaudio.org>
 <CAD4guxOw9OYj0Phric9CTpywxyisdnzcM6Zqd5CHhMW9bJ6W_A@mail.gmail.com>
 <20140805144959.GB8313@linuxaudio.org>
Message-ID: <CAD4guxM_-WfyqAFDBkVLKWThm7RC23UE23+zh7axM_jY7hxpZA@mail.gmail.com>

2014-08-05 16:49 GMT+02:00 Fons Adriaensen <fons at linuxaudio.org>:

> On Tue, Aug 05, 2014 at 04:23:23PM +0200, Raffaele Morelli wrote:
>
> > Ok, works but I have a couple of question.
> >
> > I see only 751 records in the outfile. How ebur128 works? Is the file
> > scanned entirely and how?
>
> Yes, the entire audio file is processed. Internally ebumeter and ebur128
> compute two histograms, each consisting of 751 bins from -70 to +5 dB in
> steps of 0.1 dB.
>
> The first histogram is the 'momentary' loudness, the average over
> a sliding window of 400 ms. The second is the 'short-term' loudness,
> computed using a sliding window of 3 seconds.
>
> The histograms are then used to compute average loudness and
> loudness range according to the EBU-R128 standard. Ebumeter
> will show them in real-time.
>
> For the details, see my LAC 2011 paper which you can find here:
> <http://kokkinizita.linuxaudio.org/papers/index.html>.
>
> The output file of ebur128 consists of five columns:
>
> 1. X-axis, level in dB
> 2. Cumulative probability of the momentary loudness, this is
>    the histogram integrated and normalised to 1. In other words
>    the probability that the level is less than the X-axis value.
> 3. Same for the short-term loudness.
> 4. The momentary loudness histogram, counts for each bin.
> 5. Same for the short-term loudnes.
>
> The gnuplot commands I posted will plot 1 vs 4 or 1 vs 5.
>
> Ciao,
>
> --
> F
> ?A
>


?Great tool, I finally leaved it untouched as cumulative probability it's
enough for the scope*.

My 0.02?: I would suggest to add a cli option for output filename, leaving
'ebur128-prob' as default if none is specified

ciao
/r

* In the beginning of CD era it was common (I guess) to have album released
as they were on CD, lately we see more an more re-master editions of the
same material (plus something else that never made the original recording).
My curiosity is to "investigate" on the dynamic range of both releases.
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From ralf.mardorf at rocketmail.com  Fri Aug  8 06:59:49 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Fri, 08 Aug 2014 08:59:49 +0200
Subject: [LAU] Why does PHASEX sound so damn good?
In-Reply-To: <20140808012441.GA9925@q400a.mobile.restivo.org>
References: <20140808012441.GA9925@q400a.mobile.restivo.org>
Message-ID: <1407481189.16137.3.camel@rocketmail.com>

I agree that Phasex is a better virtual Linux synth, but it still is far
away from analog sound. IIRC another good virtual Linux synth is Calf
Monosynth. I compared the virtual Linux synth outputs using HDSPe AIO
outputs, ADA8000 outputs and EWX 24/96 outputs with an Oberheim
Matrix-1000 connected to the same analog mixer. The same for a virtual
digital synth, my DX7 does sound classes better than Hexter does. 

[rocketmouse at archlinux ~]$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: HDSPMx579bcc [RME AIO_579bcc], device 0: RME AIO [RME AIO]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 1: EWX2496 [TerraTec EWX24/96], device 0: ICE1712 multi [ICE1712 multi]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 2: EWX2496_1 [TerraTec EWX24/96], device 0: ICE1712 multi [ICE1712 multi]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

I guess Phasex and Yoshimi are the most used virtual Linux synth on my
machine, OTOH in more than 10 years using Linux only at home, I never
really finished an audio production at home, resp. I never released a
production. MIDI jitter, xruns, sound quality, a bad RME driver are
issues you don't experience with good stand alone gear.


From ralf.mardorf at rocketmail.com  Fri Aug  8 07:15:50 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Fri, 08 Aug 2014 09:15:50 +0200
Subject: [LAU] [Bulk] Re:  dynamic range analysis tool
In-Reply-To: <CAD4guxM_-WfyqAFDBkVLKWThm7RC23UE23+zh7axM_jY7hxpZA@mail.gmail.com>
References: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
 <20140805131305.GA11181@linuxaudio.org>
 <CAD4guxOw9OYj0Phric9CTpywxyisdnzcM6Zqd5CHhMW9bJ6W_A@mail.gmail.com>
 <20140805144959.GB8313@linuxaudio.org>
 <CAD4guxM_-WfyqAFDBkVLKWThm7RC23UE23+zh7axM_jY7hxpZA@mail.gmail.com>
Message-ID: <1407482150.16137.5.camel@rocketmail.com>

On Fri, 2014-08-08 at 08:46 +0200, Raffaele Morelli wrote:
> In the beginning of CD era it was common (I guess) to have album
> released as they were on CD, lately we see more an more re-master
> editions of the same material (plus something else that never made the
> original recording). 
> My curiosity is to "investigate" on the dynamic range of both
> releases.

Not to mention recordings of artists who died decades ago. You will hear
claims such as, the Hendrix family authorize original unreleased Jimi
Hendrix mixes, but when listening, you can hear that something with the
mixes is fishy.


From tito.01beta at gmail.com  Fri Aug  8 08:58:47 2014
From: tito.01beta at gmail.com (Tito Latini)
Date: Fri, 8 Aug 2014 10:58:47 +0200
Subject: [LAU] Why does PHASEX sound so damn good?
In-Reply-To: <20140808012441.GA9925@q400a.mobile.restivo.org>
References: <20140808012441.GA9925@q400a.mobile.restivo.org>
Message-ID: <20140808085846.GA1806@rhk.homenet.telecomitalia.it>

On Thu, Aug 07, 2014 at 06:24:41PM -0700, Ken Restivo wrote:
> Rolling by on randomize came a Me and My Cronies jam/joke from years ago:
> 
> http://www.restivo.org/blog/podpress_trac/web/558/0/Not_OK_Computer.ogg

nice!

> And I was struck by how much PHASEX sounds like a real analog synth, like an ARP 2600 or similar, and so much more real than any other software synths I've used.
> 
> It sounds so... raw, uncontrolled, well, ANALOG. Most software simulations sound more or less authentic, but all so much more "tame", for want of a better term. But PHASEX always sounded to me (and felt, as I was playing with it) that at any moment it could do something crazy like throw a DC offset, to into an uncontrollable oscillation, or blow up my speakers, etc.
> 
> I don't like being at a loss for precise, engineering terms, or understanding WHY something is, so I'm asking any of the DSP'ers here who might also have looked at (and understood) PHASEX's source.
> 
> Any ideas what is so different about PHASEX, and what might be this quality of it's sound I could be trying to describe?

Perhaps it is not about a particolar DSP but probably you are using
one of the follow wavetables for one or more oscillators:

  <https://github.com/williamweston/phasex/tree/v0.14.97/samples>

Of course they are usable with any sampler/tracker/etc.

Here is a simple test with SoX:

for smp in phasex/samples/*.raw; do
        play -t raw -r 48000 -c 1 -e float -b 32 "${smp}" \
             gain -9 speed 18 repeat 1000 fade h 0.05 1 0.2
done

From fons at linuxaudio.org  Fri Aug  8 10:22:36 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Fri, 8 Aug 2014 10:22:36 +0000
Subject: [LAU] Why does PHASEX sound so damn good?
In-Reply-To: <20140808012441.GA9925@q400a.mobile.restivo.org>
References: <20140808012441.GA9925@q400a.mobile.restivo.org>
Message-ID: <20140808102235.GA23816@linuxaudio.org>

On Thu, Aug 07, 2014 at 06:24:41PM -0700, Ken Restivo wrote:
 
> It sounds so... raw, uncontrolled, well, ANALOG. Most software simulations sound
> more or less authentic, but all so much more "tame", for want of a better term.
> But PHASEX always sounded to me (and felt, as I was playing with it) that at any
> moment it could do something crazy like throw a DC offset, to into an uncontrollable
> oscillation, or blow up my speakers, etc.

You have been lucky then. If your sound card and amp are DC coupled, 
Phasex *will* blow up your speakers. On my first test it produced a
DC offset 30 dB above the audio signal level. 

Whatever qualities it may have, this is crappy.

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From ralf.mardorf at rocketmail.com  Fri Aug  8 10:55:50 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Fri, 08 Aug 2014 12:55:50 +0200
Subject: [LAU] [Bulk] Re:  Why does PHASEX sound so damn good?
In-Reply-To: <20140808102235.GA23816@linuxaudio.org>
References: <20140808012441.GA9925@q400a.mobile.restivo.org>
 <20140808102235.GA23816@linuxaudio.org>
Message-ID: <1407495350.16137.72.camel@rocketmail.com>

On Fri, 2014-08-08 at 10:22 +0000, Fons Adriaensen wrote:
> On Thu, Aug 07, 2014 at 06:24:41PM -0700, Ken Restivo wrote:
>  
> > It sounds so... raw, uncontrolled, well, ANALOG. Most software simulations sound
> > more or less authentic, but all so much more "tame", for want of a better term.
> > But PHASEX always sounded to me (and felt, as I was playing with it) that at any
> > moment it could do something crazy like throw a DC offset, to into an uncontrollable
> > oscillation, or blow up my speakers, etc.
> 
> You have been lucky then. If your sound card and amp are DC coupled, 
> Phasex *will* blow up your speakers. On my first test it produced a
> DC offset 30 dB above the audio signal level. 
> 
> Whatever qualities it may have, this is crappy.

I can't remember such an issue, with the versions of Phasex I used.


From csanchezgs at gmail.com  Fri Aug  8 12:20:25 2014
From: csanchezgs at gmail.com (Carlos sanchiavedraz)
Date: Fri, 8 Aug 2014 14:20:25 +0200
Subject: [LAU] One more note on PHASEX and that track
In-Reply-To: <20140808013625.GB9925@q400a.mobile.restivo.org>
References: <20140808013625.GB9925@q400a.mobile.restivo.org>
Message-ID: <CA+O6s5wZguKjus0V6s0yHKo8N--8KOT2XWLdPwN3W==5AZ1+8g@mail.gmail.com>

2014-08-08 3:36 GMT+02:00 Ken Restivo <ken at restivo.org>:
> Annotations for that track (http://www.restivo.org/blog/podpress_trac/web/558/0/Not_OK_Computer.ogg)
>
> PHASEX is present/prevalent from the intro through 05:59.
> WhySynth is from 6:00 through 6:58
> Nekostring is 07:48 through 07:55
> AMS is 07:59 through 10:00
>
> PHASEX sounds very much to me like a real analog synth. The rest sound very much like software simulations. I have no idea why the difference seems so striking.
>
> -ken
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

Great, as always, Ken. Interesting that kind of VCS3/Nintendo loop
sound in context of a funky-rock band we've listened before.

And yes, Phasex sound really great.


-- 

C. sanchiavedraZ:
* NEW / NUEVO:     www.sanchiavedraZ.com
* Musix GNU+Linux: www.musix.es

From carlo.ratm at gmail.com  Fri Aug  8 13:54:45 2014
From: carlo.ratm at gmail.com (Carlo Ascani)
Date: Fri, 8 Aug 2014 15:54:45 +0200
Subject: [LAU] One more note on PHASEX and that track
In-Reply-To: <CA+O6s5wZguKjus0V6s0yHKo8N--8KOT2XWLdPwN3W==5AZ1+8g@mail.gmail.com>
References: <20140808013625.GB9925@q400a.mobile.restivo.org>
 <CA+O6s5wZguKjus0V6s0yHKo8N--8KOT2XWLdPwN3W==5AZ1+8g@mail.gmail.com>
Message-ID: <CABVJJr-uy=fCGu-5mHjxAB_W2uHLpHb=P=j2LTdu7x+=kNoL0Q@mail.gmail.com>

2014-08-08 14:20 GMT+02:00 Carlos sanchiavedraz <csanchezgs at gmail.com>:
> 2014-08-08 3:36 GMT+02:00 Ken Restivo <ken at restivo.org>:
>> Annotations for that track (http://www.restivo.org/blog/podpress_trac/web/558/0/Not_OK_Computer.ogg)
>>
This is just great!
I am always impressed by the drummer
I just love the songs and his style



-- 
Carlo Ascani | carlorat.me
skype: carloratm
irc: carloratm at freenode

From fons at linuxaudio.org  Fri Aug  8 17:42:04 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Fri, 8 Aug 2014 17:42:04 +0000
Subject: [LAU] dynamic range analysis tool
In-Reply-To: <CAD4guxM_-WfyqAFDBkVLKWThm7RC23UE23+zh7axM_jY7hxpZA@mail.gmail.com>
References: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
 <20140805131305.GA11181@linuxaudio.org>
 <CAD4guxOw9OYj0Phric9CTpywxyisdnzcM6Zqd5CHhMW9bJ6W_A@mail.gmail.com>
 <20140805144959.GB8313@linuxaudio.org>
 <CAD4guxM_-WfyqAFDBkVLKWThm7RC23UE23+zh7axM_jY7hxpZA@mail.gmail.com>
Message-ID: <20140808174204.GA21583@linuxaudio.org>

On Fri, Aug 08, 2014 at 08:46:54AM +0200, Raffaele Morelli wrote:
 
> Great tool, I finally leaved it untouched as cumulative probability it's
> enough for the scope*.
> 
> My 0.02?: I would suggest to add a cli option for output filename, leaving
> 'ebur128-prob' as default if none is specified

Yes, the cumulative plots look less sexy but will give a better idea of
dynamic range than a histogram. The 'loudness range' as displayed by
ebumeter and printed by ebur128 is actually computed from the (modified)
cumulative plots - details in the paper.

Next release will have the histograms as well, it's only a few chars extra
code, and the filename as you suggest.

Ciao,
 
-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From abonnements at revolwear.com  Fri Aug  8 18:11:09 2014
From: abonnements at revolwear.com (Max)
Date: Sat, 09 Aug 2014 03:11:09 +0900
Subject: [LAU] jack: interface selection has no effect
Message-ID: <53E512BD.8070107@revolwear.com>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

selecting a different interface in qjackctl has no effect.  i can't
use the external soundcard, because jack stays always with the
internal one.
any ideas what causes this symptom?

i am running pulse on top of jack like this:
http://trac.jackaudio.org/wiki/WalkThrough/User/PulseOnJack

m
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1

iEYEARECAAYFAlPlEr0ACgkQ3EB7kzgMM6IMsgCfXLjIAxmgzISXQeGRpF9QMoef
c0AAn3Hm+OvgLZtTWH/sKCSBhWTZpVrt
=PX5E
-----END PGP SIGNATURE-----

From ralf.mardorf at rocketmail.com  Fri Aug  8 19:23:21 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Fri, 08 Aug 2014 21:23:21 +0200
Subject: [LAU] [Bulk]  jack: interface selection has no effect
In-Reply-To: <53E512BD.8070107@revolwear.com>
References: <53E512BD.8070107@revolwear.com>
Message-ID: <1407525801.25731.1.camel@rocketmail.com>

On Sat, 2014-08-09 at 03:11 +0900, Max wrote:
> selecting a different interface in qjackctl has no effect.

You should post the messages.

On the left side, below QjackCtl's "Start" button, there's the
"Messages" button. Click it, select the "Messages" tab, click inside the
window, push Ctrl+A (select all), then Ctrl+C (copy), click into your
Thunderbird's editor window and push Ctrl+V (paste).

Regards,
Ralf


From ken at restivo.org  Fri Aug  8 23:47:26 2014
From: ken at restivo.org (Ken Restivo)
Date: Fri, 8 Aug 2014 16:47:26 -0700
Subject: [LAU] Why does PHASEX sound so damn good?
In-Reply-To: <20140808102235.GA23816@linuxaudio.org>
References: <20140808012441.GA9925@q400a.mobile.restivo.org>
 <20140808102235.GA23816@linuxaudio.org>
Message-ID: <20140808234726.GB22900@q400a.mobile.restivo.org>

On Fri, Aug 08, 2014 at 10:22:36AM +0000, Fons Adriaensen wrote:
> On Thu, Aug 07, 2014 at 06:24:41PM -0700, Ken Restivo wrote:
>  
> > It sounds so... raw, uncontrolled, well, ANALOG. Most software simulations sound
> > more or less authentic, but all so much more "tame", for want of a better term.
> > But PHASEX always sounded to me (and felt, as I was playing with it) that at any
> > moment it could do something crazy like throw a DC offset, to into an uncontrollable
> > oscillation, or blow up my speakers, etc.
> 
> You have been lucky then. If your sound card and amp are DC coupled, 
> Phasex *will* blow up your speakers. On my first test it produced a
> DC offset 30 dB above the audio signal level. 
> 

Ah, thanks, I'd forgotten that! I did this peice years ago in PHASEX,
and tried to correct the DC offset, but wasn't able to completely:
http://www.restivo.org/blog/podpress_trac/web/317/0/airbrush-0.1.ogg
(WARNING: might blow your speakers as described above)

It looks like this:
 http://bace.s3.amazonaws.com/dcoffset.jpg

> Whatever qualities it may have, this is crappy.

I guess so. But maybe it's dangerousness is what gives it its sound.

Will Alexander once described Keith Emerson's Moog as having "no
padded cell technology". Meaning, it was capable of destroying amps,
PA systems, expensive mixing boards, huge stadium-sized house sound
systems, etc. He treated the thing like a loaded weapon when plugging
it into stuff. Maybe this is what he meant by that.

-ken

From termtech at rogers.com  Sat Aug  9 00:04:30 2014
From: termtech at rogers.com (Tim E. Real)
Date: Fri, 08 Aug 2014 20:04:30 -0400
Subject: [LAU] [Bulk]  jack: interface selection has no effect
In-Reply-To: <53E512BD.8070107@revolwear.com>
References: <53E512BD.8070107@revolwear.com>
Message-ID: <3186626.5TZbtJ3EhK@col-desktop>

On August 9, 2014 03:11:09 AM Max wrote:
> selecting a different interface in qjackctl has no effect.  i can't
> use the external soundcard, because jack stays always with the
> internal one.
> any ideas what causes this symptom?
> 
> i am running pulse on top of jack like this:
> http://trac.jackaudio.org/wiki/WalkThrough/User/PulseOnJack
> 
> m

I have found that when you set it up that way, you'll find that 
 nothing you change in QJackCtl has any effect.

You must manually edit the file /home/.jackdrc
 and then restart pulse (or reboot).

But I've never quite understood how to properly restart pulse,
 it always seems to lead to a messy situation needing a reboot.

You'll see that changes in QJackCtl do not appear in .jackdrc -
 pulse seems to override it.

I don't quite understood why this is so.
Seems pulse uses internal jack server or something?
Even if you attempt to 'stop' jack from QJackCtl, it doesn't -
 pulse just keeps on running its jack server.
Simply restart QJackCtl and you'll see Jack is actually still running
 and has not stopped at all.

Can someone explain why, and how to properly restart pulse after
 editing the .jackdrc file, or if there's an easier way to allow 
 QJackCtl to truly alter the settings?

Thanks!
Tim.

From fons at linuxaudio.org  Sat Aug  9 09:46:03 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Sat, 9 Aug 2014 09:46:03 +0000
Subject: [LAU] Why does PHASEX sound so damn good?
In-Reply-To: <20140808234726.GB22900@q400a.mobile.restivo.org>
References: <20140808012441.GA9925@q400a.mobile.restivo.org>
 <20140808102235.GA23816@linuxaudio.org>
 <20140808234726.GB22900@q400a.mobile.restivo.org>
Message-ID: <20140809094603.GA6670@linuxaudio.org>

On Fri, Aug 08, 2014 at 04:47:26PM -0700, Ken Restivo wrote:
 
> It looks like this:
>  http://bace.s3.amazonaws.com/dcoffset.jpg

That's a very mild case, probably harmless. What I got, just by
selecting one of the presets ('classic pad' or something similar)
was a DC offset 30 times as big as the audio signal. In terms of
power that close to a 1000:1 ratio.
 
> > Whatever qualities it may have, this is crappy.
> 
> I guess so. But maybe it's dangerousness is what gives it its sound.

You can't hear DC. It may drive your speakers into some nice distortion
before the funny smell appears, but there are less dangerous ways to
achieve the same.

> Will Alexander once described Keith Emerson's Moog as having "no
> padded cell technology". Meaning, it was capable of destroying amps,
> PA systems, expensive mixing boards, huge stadium-sized house sound
> systems, etc. He treated the thing like a loaded weapon when plugging
> it into stuff. Maybe this is what he meant by that.

That can be said of any synth in fact. And I'm pretty sure that
ELP's PA system was not DC-coupled - the mixers they used had
transformer inputs. 

DC offsets occur naturally when using phase modulated oscillators
even if their basic waveform is DC-free. It's part of the way PM
synthesis works, and you want the DC in the modulation inputs. But
once the VCO outputs enter the other parts of the audio path it 
should be removed. It's trivially simple to do that.

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From gheskett at wdtv.com  Sat Aug  9 13:11:15 2014
From: gheskett at wdtv.com (Gene Heskett)
Date: Sat, 9 Aug 2014 09:11:15 -0400
Subject: [LAU] Why does PHASEX sound so damn good?
In-Reply-To: <20140809094603.GA6670@linuxaudio.org>
References: <20140808012441.GA9925@q400a.mobile.restivo.org>
 <20140808234726.GB22900@q400a.mobile.restivo.org>
 <20140809094603.GA6670@linuxaudio.org>
Message-ID: <201408090911.15285.gheskett@wdtv.com>

On Saturday 09 August 2014 05:46:03 Fons Adriaensen did opine
And Gene did reply:
> On Fri, Aug 08, 2014 at 04:47:26PM -0700, Ken Restivo wrote:
> > It looks like this:
> >  http://bace.s3.amazonaws.com/dcoffset.jpg
> 
> That's a very mild case, probably harmless. What I got, just by
> selecting one of the presets ('classic pad' or something similar)
> was a DC offset 30 times as big as the audio signal. In terms of
> power that close to a 1000:1 ratio.
> 
> > > Whatever qualities it may have, this is crappy.

You are being too kind, Ken.  Has this list gotten so genteel you can't 
use more descriptive terms?
> > 
> > I guess so. But maybe it's dangerousness is what gives it its sound.
> 
> You can't hear DC. It may drive your speakers into some nice distortion
> before the funny smell appears, but there are less dangerous ways to
> achieve the same.
> 
> > Will Alexander once described Keith Emerson's Moog as having "no
> > padded cell technology". Meaning, it was capable of destroying amps,
> > PA systems, expensive mixing boards, huge stadium-sized house sound
> > systems, etc. He treated the thing like a loaded weapon when plugging
> > it into stuff. Maybe this is what he meant by that.
> 
> That can be said of any synth in fact. And I'm pretty sure that
> ELP's PA system was not DC-coupled - the mixers they used had
> transformer inputs.

Whose iron can be quite well saturated when dealing with a DC level at 
their inputs.  That will have a sound all its own that I've never found to 
be at all pleasant due to the IM distortion it causes.
 
> DC offsets occur naturally when using phase modulated oscillators
> even if their basic waveform is DC-free. It's part of the way PM
> synthesis works, and you want the DC in the modulation inputs. But
> once the VCO outputs enter the other parts of the audio path it
> should be removed. It's trivially simple to do that.

The idea is simple, properly doing it in the presence of front office bean 
counters who don't understand why it should be done, is not.  But you knew 
that...

Cheers, Gene Heskett
-- 
"There are four boxes to be used in defense of liberty:
 soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
Genes Web page <http://geneslinuxbox.net:6309/gene>
US V Castleman, SCOTUS, Mar 2014 is grounds for Impeaching SCOTUS

From cannam at all-day-breakfast.com  Sat Aug  9 13:45:23 2014
From: cannam at all-day-breakfast.com (Chris Cannam)
Date: Sat, 09 Aug 2014 14:45:23 +0100
Subject: [LAU] Silvet Note Transcription Vamp plugin v1.0 released
Message-ID: <1407591923.874655.150881385.0CE6B148@webmail.messagingengine.com>

Silvet is a Vamp plugin for note transcription in polyphonic music.

   http://code.soundsoftware.ac.uk/projects/silvet

** What does it do?

Silvet listens to audio recordings of music and tries to work out what
notes are being played.

To use it, you need a Vamp plugin host (such as Sonic Visualiser).

How to use the plugin will depend on the host you use, but in the case
of Sonic Visualiser, you should load an audio file and then run Silvet
Note Transcription from the Transform menu.  This will add a note
layer to your session with the transcription in it, which you can
listen to or export as a MIDI file.

** How good is it?

Silvet performs well for some recordings, but the range of music that
works well is quite limited at this stage. Generally it works best
with piano or acoustic instruments in solo or small-ensemble music.

Silvet does not transcribe percussion and has a limited range of
instrument support. It does not technically support vocals, although
it will sometimes transcribe them anyway.

You can usually expect the output to be reasonably informative and to
bear some audible relationship to the actual notes, but you shouldn't
expect to get something that can be directly converted to a readable
score. For much rock/pop music in particular the results will be, at
best, recognisable.

To summarise: try it and see.

** Can it be used live?

In theory it can, because the plugin is causal: it emits notes as it
hears the audio. But it has to operate on long blocks of audio with a
latency of many seconds, so although it will work with non-seekable
streams, it isn't in practice responsive enough to use live.

** How does it work?

Silvet uses the method described in "A Shift-Invariant Latent Variable
Model for Automatic Music Transcription" by Emmanouil Benetos and
Simon Dixon (Computer Music Journal, 2012).

It uses probablistic latent-variable estimation to decompose a
Constant-Q time-frequency matrix into note activations using a set of
spectral templates learned from recordings of solo instruments.

For a formal evaluation, please refer to the 2012 edition of MIREX,
the Music Information Retrieval Evaluation Exchange, where the basic
method implemented in Silvet formed the BD1, BD2 and BD3 submissions
in the Multiple F0 Tracking task:

http://www.music-ir.org/mirex/wiki/2012:Multiple_Fundamental_Frequency_Estimation_%26_Tracking_Results


From fons at linuxaudio.org  Sat Aug  9 16:07:58 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Sat, 9 Aug 2014 16:07:58 +0000
Subject: [LAU] Why does PHASEX sound so damn good?
In-Reply-To: <201408090911.15285.gheskett@wdtv.com>
References: <20140808012441.GA9925@q400a.mobile.restivo.org>
 <20140808234726.GB22900@q400a.mobile.restivo.org>
 <20140809094603.GA6670@linuxaudio.org>
 <201408090911.15285.gheskett@wdtv.com>
Message-ID: <20140809160758.GA16380@linuxaudio.org>

On Sat, Aug 09, 2014 at 09:11:15AM -0400, Gene Heskett wrote:

> > > > Whatever qualities it may have, this is crappy.
> 
> You are being too kind, Ken.  Has this list gotten so genteel you can't 
> use more descriptive terms?

If you refer to the 'crappy', I wrote that, not Ken. 

To see what I mean:

<http://kokkinizita.linuxaudio.org/linuxaudio/phasex-mellow-pad.png>

All I needed to get that was loading the 'mellow-pad' patch from
sys-patches and hit a C-Maj chord. Output level as indicated by 
jkmeter was +9.5 dB FS. Which means that all that would remain
of this after DA conversion was full scale DC for a second or so.

Mellow, indeed.

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From willgodfrey at musically.me.uk  Sat Aug  9 17:38:39 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Sat, 9 Aug 2014 18:38:39 +0100
Subject: [LAU] Why does PHASEX sound so damn good?
In-Reply-To: <20140809160758.GA16380@linuxaudio.org>
References: <20140808012441.GA9925@q400a.mobile.restivo.org>
 <20140808234726.GB22900@q400a.mobile.restivo.org>
 <20140809094603.GA6670@linuxaudio.org>
 <201408090911.15285.gheskett@wdtv.com>
 <20140809160758.GA16380@linuxaudio.org>
Message-ID: <20140809183839.0c0d6e23@debian>

On Sat, 9 Aug 2014 16:07:58 +0000
Fons Adriaensen <fons at linuxaudio.org> wrote:

> On Sat, Aug 09, 2014 at 09:11:15AM -0400, Gene Heskett wrote:
> 
> > > > > Whatever qualities it may have, this is crappy.
> > 
> > You are being too kind, Ken.  Has this list gotten so genteel you can't 
> > use more descriptive terms?
> 
> If you refer to the 'crappy', I wrote that, not Ken. 
> 
> To see what I mean:
> 
> <http://kokkinizita.linuxaudio.org/linuxaudio/phasex-mellow-pad.png>
> 
> All I needed to get that was loading the 'mellow-pad' patch from
> sys-patches and hit a C-Maj chord. Output level as indicated by 
> jkmeter was +9.5 dB FS. Which means that all that would remain
> of this after DA conversion was full scale DC for a second or so.
> 
> Mellow, indeed.
> 
> Ciao,

Ouch!

I've had a few tries with phasex and it sounded interesting, but never got on
with it because of information overload - too many controls all visible at
once :(

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From len at ovenwerks.net  Sat Aug  9 18:53:49 2014
From: len at ovenwerks.net (Len Ovens)
Date: Sat, 9 Aug 2014 11:53:49 -0700 (PDT)
Subject: [LAU] jack: interface selection has no effect
In-Reply-To: <53E512BD.8070107@revolwear.com>
References: <53E512BD.8070107@revolwear.com>
Message-ID: <alpine.DEB.2.02.1408091147310.23832@scott.cbbs.org>

On Sat, 9 Aug 2014, Max wrote:

> selecting a different interface in qjackctl has no effect.  i can't
> use the external soundcard, because jack stays always with the
> internal one.
> any ideas what causes this symptom?
>
> i am running pulse on top of jack like this:
> http://trac.jackaudio.org/wiki/WalkThrough/User/PulseOnJack

That is the hardest way to do it. qjackctl expects (as installed) to use 
jackdbus. If you wish to use qjackctl with jackd you have to turn dbus off 
in the qjackctl settings. However, if jackd is started externally and not 
from qjackctl you will still not be able to stop jackd.

The only reason to do things this way is that you are using jackd1, if you 
are using jackd2, and you have dbus running in the session anyway, then 
run jackdbus. If you wish to run it from the command line use jack_control 
start. If you start it with qjackctl first with the settings you want, 
that will be enough, other wise you will have to learn how to set 
parameters with jack_control. There is no man page so try jack_control 
--help


--
Len Ovens
www.ovenwerks.net


From gheskett at wdtv.com  Sat Aug  9 20:46:53 2014
From: gheskett at wdtv.com (Gene Heskett)
Date: Sat, 9 Aug 2014 16:46:53 -0400
Subject: [LAU] Why does PHASEX sound so damn good?
In-Reply-To: <20140809160758.GA16380@linuxaudio.org>
References: <20140808012441.GA9925@q400a.mobile.restivo.org>
 <201408090911.15285.gheskett@wdtv.com>
 <20140809160758.GA16380@linuxaudio.org>
Message-ID: <201408091646.53681.gheskett@wdtv.com>

On Saturday 09 August 2014 12:07:58 Fons Adriaensen did opine
And Gene did reply:
> On Sat, Aug 09, 2014 at 09:11:15AM -0400, Gene Heskett wrote:
> > > > > Whatever qualities it may have, this is crappy.
> > 
> > You are being too kind, Ken.  Has this list gotten so genteel you
> > can't use more descriptive terms?
> 
> If you refer to the 'crappy', I wrote that, not Ken.

So I miss-read the quoting depth, again. :(
 
> To see what I mean:
> 
> <http://kokkinizita.linuxaudio.org/linuxaudio/phasex-mellow-pad.png>

Good grief, and that thing was allowed out the door?  It wouldn't have 
made it on my watch.
> 
> All I needed to get that was loading the 'mellow-pad' patch from
> sys-patches and hit a C-Maj chord. Output level as indicated by
> jkmeter was +9.5 dB FS. Which means that all that would remain
> of this after DA conversion was full scale DC for a second or so.
> 
> Mellow, indeed.
> 
> Ciao,


Cheers, Gene Heskett
-- 
"There are four boxes to be used in defense of liberty:
 soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
Genes Web page <http://geneslinuxbox.net:6309/gene>
US V Castleman, SCOTUS, Mar 2014 is grounds for Impeaching SCOTUS

From willgodfrey at musically.me.uk  Sat Aug  9 21:08:50 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Sat, 9 Aug 2014 22:08:50 +0100
Subject: [LAU] Why does PHASEX sound so damn good?
In-Reply-To: <201408091646.53681.gheskett@wdtv.com>
References: <20140808012441.GA9925@q400a.mobile.restivo.org>
 <201408090911.15285.gheskett@wdtv.com>
 <20140809160758.GA16380@linuxaudio.org>
 <201408091646.53681.gheskett@wdtv.com>
Message-ID: <20140809220850.409e3f3e@debian>

On Sat, 9 Aug 2014 16:46:53 -0400
Gene Heskett <gheskett at wdtv.com> wrote:

> On Saturday 09 August 2014 12:07:58 Fons Adriaensen did opine
> And Gene did reply:
> > On Sat, Aug 09, 2014 at 09:11:15AM -0400, Gene Heskett wrote:
> > > > > > Whatever qualities it may have, this is crappy.
> > > 
> > > You are being too kind, Ken.  Has this list gotten so genteel you
> > > can't use more descriptive terms?
> > 
> > If you refer to the 'crappy', I wrote that, not Ken.
> 
> So I miss-read the quoting depth, again. :(
>  
> > To see what I mean:
> > 
> > <http://kokkinizita.linuxaudio.org/linuxaudio/phasex-mellow-pad.png>
> 
> Good grief, and that thing was allowed out the door?  It wouldn't have 
> made it on my watch.
> > 
> > All I needed to get that was loading the 'mellow-pad' patch from
> > sys-patches and hit a C-Maj chord. Output level as indicated by
> > jkmeter was +9.5 dB FS. Which means that all that would remain
> > of this after DA conversion was full scale DC for a second or so.
> > 
> > Mellow, indeed.
> > 
> > Ciao,
> 
> 
> Cheers, Gene Heskett

Hmmm.
Just had a look on github no development for 2 years :(

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From paul at linuxaudiosystems.com  Sat Aug  9 23:23:40 2014
From: paul at linuxaudiosystems.com (Paul Davis)
Date: Sat, 9 Aug 2014 19:23:40 -0400
Subject: [LAU] Why does PHASEX sound so damn good?
In-Reply-To: <201408091646.53681.gheskett@wdtv.com>
References: <20140808012441.GA9925@q400a.mobile.restivo.org>
 <201408090911.15285.gheskett@wdtv.com>
 <20140809160758.GA16380@linuxaudio.org>
 <201408091646.53681.gheskett@wdtv.com>
Message-ID: <CAFa_cKkFB7Mxh04osXNQW6n3-xBBrFM13DHmf_Q87Y7BMqW7hQ@mail.gmail.com>

On Sat, Aug 9, 2014 at 4:46 PM, Gene Heskett <gheskett at wdtv.com> wrote:

> On Saturday 09 August 2014 12:07:58 Fons Adriaensen did opine
> And Gene did reply:
> > On Sat, Aug 09, 2014 at 09:11:15AM -0400, Gene Heskett wrote:
> > > > > > Whatever qualities it may have, this is crappy.
> > >
> > > You are being too kind, Ken.  Has this list gotten so genteel you
> > > can't use more descriptive terms?
> >
> > If you refer to the 'crappy', I wrote that, not Ken.
>
> So I miss-read the quoting depth, again. :(
>
> > To see what I mean:
> >
> > <http://kokkinizita.linuxaudio.org/linuxaudio/phasex-mellow-pad.png>
>
> Good grief, and that thing was allowed out the door?  It wouldn't have
> made it on my watch.
>

open source is well known for making do without windows; it tends to do
without doors also.
-------------- next part --------------
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From brent at keycorner.org  Sun Aug 10 00:28:35 2014
From: brent at keycorner.org (Brent Busby)
Date: Sat, 9 Aug 2014 19:28:35 -0500 (CDT)
Subject: [LAU] Ardour/Muse Jack tempo lock
Message-ID: <alpine.LNX.2.00.1408091922010.19090@village.keycorner.org>

I'm having some difficulty getting Ardour 3 and Muse to consistently 
tempo lock to eachother.  By consistently, I don't mean that they'll 
lose sync once I've got it working.  Rather, it seems to be a tricky 
business getting it to happen at all.

It seems to somewhat depend on the order they're started in, but also 
I've noticed that if I change Muse's sync settings, the tempo selection 
can become grayed out -- appropriate when it's a slave, but it never 
becomes active again when it is set as a master again.

Does anyone have any recommendations for this?  I'd prefer Ardour as the 
master, since it's the one handling actual sample data (with measure 
lines in its timeline for real audio waves) and not just sequencer 
playback, but at this point I'd almost take anything that would 
consistently work.  It seems that even if I setup Ardour as Jack 
timebase master and edit its tempo ruler to 140bpm, it will show measure 
lines on its timeline that are appropriate for 140bpm, but Jack programs 
like Muse and Hydrogen will still playback at their default 120bpm, as 
though they're receiving the transport control messages but not the 
tempo sync.  If I have one of them be the master, they disagree about 
what 140bpm is, which means they're still not really synced.  It's very 
frustrating...

-- 
+ Brent A. Busby	 + "We've all heard that a million monkeys
+ Sr. UNIX Systems Admin +  banging on a million typewriters will
+ University of Chicago	 +  eventually reproduce the entire works of
+ James Franck Institute +  Shakespeare.  Now, thanks to the Internet,
+ Materials Research Ctr +  we know this is not true." -Robert Wilensky

From brent at keycorner.org  Sun Aug 10 00:52:07 2014
From: brent at keycorner.org (Brent Busby)
Date: Sat, 9 Aug 2014 19:52:07 -0500 (CDT)
Subject: [LAU] Ardour/Muse Jack tempo lock
In-Reply-To: <alpine.LNX.2.00.1408091922010.19090@village.keycorner.org>
References: <alpine.LNX.2.00.1408091922010.19090@village.keycorner.org>
Message-ID: <alpine.LNX.2.00.1408091945230.19342@village.keycorner.org>

Yes, replying to my own message...

I found an XML tag in Muse's ".med" file for the sequence:

<master>0</master>

I toggled this on, and restarted Muse.  It now works as a master, and 
Ardour correctly tracks it.  Also, Muse's tempo setting control is no 
longer grayed out.

Before I edited the file, the settings in Muse preferences for Jack 
Transport and Jack Timebase Master both looked like they were turned on 
in the GUI, but apparently Master was turned off in the file, and Muse 
was very confused...looks like a bug in Muse.

I'd prefer to have Ardour as the master, but for now it's working so I'm 
just going to count my blessings and let Muse be master.

-- 
+ Brent A. Busby	 + "We've all heard that a million monkeys
+ Sr. UNIX Systems Admin +  banging on a million typewriters will
+ University of Chicago	 +  eventually reproduce the entire works of
+ James Franck Institute +  Shakespeare.  Now, thanks to the Internet,
+ Materials Research Ctr +  we know this is not true." -Robert Wilensky

From gheskett at wdtv.com  Sun Aug 10 02:47:06 2014
From: gheskett at wdtv.com (Gene Heskett)
Date: Sat, 9 Aug 2014 22:47:06 -0400
Subject: [LAU] Why does PHASEX sound so damn good?
In-Reply-To: <CAFa_cKkFB7Mxh04osXNQW6n3-xBBrFM13DHmf_Q87Y7BMqW7hQ@mail.gmail.com>
References: <20140808012441.GA9925@q400a.mobile.restivo.org>
 <201408091646.53681.gheskett@wdtv.com>
 <CAFa_cKkFB7Mxh04osXNQW6n3-xBBrFM13DHmf_Q87Y7BMqW7hQ@mail.gmail.com>
Message-ID: <201408092247.06705.gheskett@wdtv.com>

On Saturday 09 August 2014 19:23:40 Paul Davis did opine
And Gene did reply:
> On Sat, Aug 9, 2014 at 4:46 PM, Gene Heskett <gheskett at wdtv.com> wrote:
> > On Saturday 09 August 2014 12:07:58 Fons Adriaensen did opine
> > 
> > And Gene did reply:
> > > On Sat, Aug 09, 2014 at 09:11:15AM -0400, Gene Heskett wrote:
> > > > > > > Whatever qualities it may have, this is crappy.
> > > > 
> > > > You are being too kind, Ken.  Has this list gotten so genteel you
> > > > can't use more descriptive terms?
> > > 
> > > If you refer to the 'crappy', I wrote that, not Ken.
> > 
> > So I miss-read the quoting depth, again. :(
> > 
> > > To see what I mean:
> > > 
> > > <http://kokkinizita.linuxaudio.org/linuxaudio/phasex-mellow-pad.png
> > > >
> > 
> > Good grief, and that thing was allowed out the door?  It wouldn't
> > have made it on my watch.
> 
> open source is well known for making do without windows; it tends to do
> without doors also.

Chuckle.  Well said Paul.

Cheers, Gene Heskett
-- 
"There are four boxes to be used in defense of liberty:
 soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
Genes Web page <http://geneslinuxbox.net:6309/gene>
US V Castleman, SCOTUS, Mar 2014 is grounds for Impeaching SCOTUS

From termtech at rogers.com  Sun Aug 10 07:19:05 2014
From: termtech at rogers.com (Tim E. Real)
Date: Sun, 10 Aug 2014 03:19:05 -0400
Subject: [LAU] Ardour/Muse Jack tempo lock
In-Reply-To: <alpine.LNX.2.00.1408091922010.19090@village.keycorner.org>
References: <alpine.LNX.2.00.1408091922010.19090@village.keycorner.org>
Message-ID: <1860644.M2evTQpcQo@col-desktop>

On August 9, 2014 07:28:35 PM Brent Busby wrote:
> I'm having some difficulty getting Ardour 3 and Muse to consistently
> tempo lock to eachother.  By consistently, I don't mean that they'll
> lose sync once I've got it working.  Rather, it seems to be a tricky
> business getting it to happen at all.
> 
> It seems to somewhat depend on the order they're started in, but also
> I've noticed that if I change Muse's sync settings, the tempo selection
> can become grayed out -- appropriate when it's a slave, but it never
> becomes active again when it is set as a master again.

You need to open the MusE Transport Panel and turn 'Master'
 back on, or open the Graphical Master Track Editor and click 'Enable'.

It's because when you turn external sync on, either in the Midi Sync
 Dialog or with the Transport Panel 'Sync' button, 'Master' turns off -
 that is the the Master track is disabled - and it does NOT turn back
 on by itself once external sync is turned off again.
The reasons for 'Master' not automatically turning back on are
 obscure - I recall I tried to do it but ran into conceptual problems.

However, first simple thoughts upon reexamining it now are that it seems 
 we ought be able to store the last 'Master' setting and revert whenever 
 external sync is turned off again.
Or... maybe that's what I thought at the time but found there was
 a problem with that. I think there are some comments in the code 
 somewhere explaining why - I'll probably remember tomorrow after 
 rambling on here.

(Master Track is just a fancy term for tempo / time signature graph.)

> 
> Does anyone have any recommendations for this?  I'd prefer Ardour as the
> master, since it's the one handling actual sample data (with measure
> lines in its timeline for real audio waves) and not just sequencer
> playback, but at this point I'd almost take anything that would
> consistently work.  It seems that even if I setup Ardour as Jack
> timebase master and edit its tempo ruler to 140bpm, it will show measure
> lines on its timeline that are appropriate for 140bpm, but Jack programs
> like Muse and Hydrogen will still playback at their default 120bpm, as
> though they're receiving the transport control messages but not the
> tempo sync.  If I have one of them be the master, they disagree about
> what 140bpm is, which means they're still not really synced.  It's very
> frustrating...

Sorry about that.
Currently the only way to sync MusE with other apps is with Midi Clock.

(Don't confuse Sync Master with Jack Transport Master: Jack Transport info 
 is shared so it's always 'synced' in each app.)

You must enable MusE's External Midi Clock, either the Transport Panel's
 'Sync' button or the Midi Sync Dialog's  'Slave to External Sync' checkbox.
Then you have to tell MusE where that Midi Clock sync comes from (Ardour) 
 in the Midi Sync Dialog and make sure you also turn on accepting of
 realtime commands (that's start, stop, positional etc) in the 'rr' column.
You may also need to tell Ardour to turn on the sending of Midi Clock and 
 realtime commands. 

And... one more thing:
You'll have to duplicate any tempo map from Ardour to MusE.
Or MusE can record any external tempos and replace its current tempo map.

I spoke of this years ago: 
If MusE had a way of knowing when the other app's tempo map changes, 
 it could automatically change its tempo map and time line.
Even with Jack Transport this was not possible, only a 'linear' tempo stream.

But with the new Jack Metadata API, maybe it's now possible.


There is some experimental unfinished MusE code to let Jack Transport's 
 BarBeatTick info drive our midi engine instead of our own tempomap-based  
 frame-to-miditick converter with external clock support.

An idea that seemed straightforward and logical to me.
I wrote it a few years ago but got discouraged when I found inconsistent
 or non- full Jack Transport support in other apps.
I believe it is crucial that all the apps (MusE included) use the complete 
 full feature set of the Jack Transport API for the best accuracy.

There was also IIRC some kind of issue with the Jack Transport API where
 I think I found it was not always possible to have accurate time<>BBT info.
However I saw recently someone brought up an issue there and Paul
 committed some code for some of that time functionality.
Must take another look sometime.


Enough for now, hope that helps so far.
Tim.


From termtech at rogers.com  Sun Aug 10 07:42:46 2014
From: termtech at rogers.com (Tim E. Real)
Date: Sun, 10 Aug 2014 03:42:46 -0400
Subject: [LAU] [Bulk] Re:  Ardour/Muse Jack tempo lock
In-Reply-To: <1860644.M2evTQpcQo@col-desktop>
References: <alpine.LNX.2.00.1408091922010.19090@village.keycorner.org>
 <1860644.M2evTQpcQo@col-desktop>
Message-ID: <6232148.vTentDHj4N@col-desktop>

On August 10, 2014 03:19:05 AM Tim E. Real wrote:
> On August 9, 2014 07:28:35 PM Brent Busby wrote:
> > I'm having some difficulty getting Ardour 3 and Muse to consistently
> > tempo lock to eachother.  By consistently, I don't mean that they'll
> > lose sync once I've got it working.  Rather, it seems to be a tricky
> > business getting it to happen at all.
> > 
> > It seems to somewhat depend on the order they're started in, 

Addendum: 

Be aware if you start Ardour after MusE you must open MusE's Midi Settings 
 and manually add Adour as a new Jack or ALSA midi device. 
(Actually there will be several Adour devices, including one for CLOCK.)

Otherwise, if you start MusE after Ardour, MusE will automatically
 create all of its Ardour midi devices.

Tim.

> > but also
> > I've noticed that if I change Muse's sync settings, the tempo selection
> > can become grayed out -- appropriate when it's a slave, but it never
> > becomes active again when it is set as a master again.
> 
> You need to open the MusE Transport Panel and turn 'Master'
>  back on, or open the Graphical Master Track Editor and click 'Enable'.
> 
> It's because when you turn external sync on, either in the Midi Sync
>  Dialog or with the Transport Panel 'Sync' button, 'Master' turns off -
>  that is the the Master track is disabled - and it does NOT turn back
>  on by itself once external sync is turned off again.
> The reasons for 'Master' not automatically turning back on are
>  obscure - I recall I tried to do it but ran into conceptual problems.
> 
> However, first simple thoughts upon reexamining it now are that it seems
>  we ought be able to store the last 'Master' setting and revert whenever
>  external sync is turned off again.
> Or... maybe that's what I thought at the time but found there was
>  a problem with that. I think there are some comments in the code
>  somewhere explaining why - I'll probably remember tomorrow after
>  rambling on here.
> 
> (Master Track is just a fancy term for tempo / time signature graph.)
> 
> > Does anyone have any recommendations for this?  I'd prefer Ardour as the
> > master, since it's the one handling actual sample data (with measure
> > lines in its timeline for real audio waves) and not just sequencer
> > playback, but at this point I'd almost take anything that would
> > consistently work.  It seems that even if I setup Ardour as Jack
> > timebase master and edit its tempo ruler to 140bpm, it will show measure
> > lines on its timeline that are appropriate for 140bpm, but Jack programs
> > like Muse and Hydrogen will still playback at their default 120bpm, as
> > though they're receiving the transport control messages but not the
> > tempo sync.  If I have one of them be the master, they disagree about
> > what 140bpm is, which means they're still not really synced.  It's very
> > frustrating...
> 
> Sorry about that.
> Currently the only way to sync MusE with other apps is with Midi Clock.
> 
> (Don't confuse Sync Master with Jack Transport Master: Jack Transport info
>  is shared so it's always 'synced' in each app.)
> 
> You must enable MusE's External Midi Clock, either the Transport Panel's
>  'Sync' button or the Midi Sync Dialog's  'Slave to External Sync' checkbox.
> Then you have to tell MusE where that Midi Clock sync comes from (Ardour)
> in the Midi Sync Dialog and make sure you also turn on accepting of
> realtime commands (that's start, stop, positional etc) in the 'rr' column.
> You may also need to tell Ardour to turn on the sending of Midi Clock and
> realtime commands.
> 
> And... one more thing:
> You'll have to duplicate any tempo map from Ardour to MusE.
> Or MusE can record any external tempos and replace its current tempo map.
> 
> I spoke of this years ago:
> If MusE had a way of knowing when the other app's tempo map changes,
>  it could automatically change its tempo map and time line.
> Even with Jack Transport this was not possible, only a 'linear' tempo
> stream.
> 
> But with the new Jack Metadata API, maybe it's now possible.
> 
> 
> There is some experimental unfinished MusE code to let Jack Transport's
>  BarBeatTick info drive our midi engine instead of our own tempomap-based
>  frame-to-miditick converter with external clock support.
> 
> An idea that seemed straightforward and logical to me.
> I wrote it a few years ago but got discouraged when I found inconsistent
>  or non- full Jack Transport support in other apps.
> I believe it is crucial that all the apps (MusE included) use the complete
>  full feature set of the Jack Transport API for the best accuracy.
> 
> There was also IIRC some kind of issue with the Jack Transport API where
>  I think I found it was not always possible to have accurate time<>BBT info.
> However I saw recently someone brought up an issue there and Paul
>  committed some code for some of that time functionality.
> Must take another look sometime.
> 
> 
> Enough for now, hope that helps so far.
> Tim.
> 


From fons at linuxaudio.org  Sun Aug 10 09:44:29 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Sun, 10 Aug 2014 09:44:29 +0000
Subject: [LAU] Why does PHASEX sound so damn good?
In-Reply-To: <20140809183839.0c0d6e23@debian>
References: <20140808012441.GA9925@q400a.mobile.restivo.org>
 <20140808234726.GB22900@q400a.mobile.restivo.org>
 <20140809094603.GA6670@linuxaudio.org>
 <201408090911.15285.gheskett@wdtv.com>
 <20140809160758.GA16380@linuxaudio.org>
 <20140809183839.0c0d6e23@debian>
Message-ID: <20140810094429.GA30295@linuxaudio.org>

On Sat, Aug 09, 2014 at 06:38:39PM +0100, Will Godfrey wrote:
 
> I've had a few tries with phasex and it sounded interesting, but never got on
> with it because of information overload - too many controls all visible at
> once :(

The problem here is not the number of controls, but the lack of
visual hints normally provided by layout, sizes and colors. 
All the controls look the same and are in a single line in each
module so the only way to know which is which is to read the 
labels. It's a pity because obviously some work went into the
GUI - it was just not based on ergonomic principles.
Another strange thing is that signal flow is right to left.


Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From len at ovenwerks.net  Sun Aug 10 16:24:12 2014
From: len at ovenwerks.net (Len Ovens)
Date: Sun, 10 Aug 2014 09:24:12 -0700
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
Message-ID: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>

Tim E. Real wrote:
> I believe it is crucial that all the apps (MusE included) use the complete
> full feature set of the Jack Transport API for the best accuracy.

Is there a record enable in there? It seems to me it would be worthwhile
for recording audio in one application and MIDI in another or for punch in
out. The application would choose to see or ignore such a signal.

-- 
Len Ovens
www.OvenWerks.net


From robin at gareus.org  Sun Aug 10 17:11:24 2014
From: robin at gareus.org (Robin Gareus)
Date: Sun, 10 Aug 2014 19:11:24 +0200
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
Message-ID: <53E7A7BC.90008@gareus.org>

On 08/10/2014 06:24 PM, Len Ovens wrote:
> Tim E. Real wrote:
>> I believe it is crucial that all the apps (MusE included) use the complete
>> full feature set of the Jack Transport API for the best accuracy.

...and port latencies! Sadly, many jack applications ignore those which
leads to offsets, even though they share transport position.

> Is there a record enable in there? 

No, there isn't.

> It seems to me it would be worthwhile
> for recording audio in one application and MIDI in another or for punch in
> out. The application would choose to see or ignore such a signal.

It's not that easy.
Record enable is an application state not a position in time.
Rec-arm needs to allocate buffers, prepare files on disk etc etc. It's
not realtime-safe. For playback there's already a special state:
JackTransportStarting. Applications should acknowledge that they're
ready to roll before jack goes into "play". A lot of jack-clients forgo
this which is already trouble enough, though not directly harmful in
this case (just possibly out of sync playback).

Adding ranges (loop, punch), application state (rec-en), or even
varispeed support would likely increase the mess rather than solve
anything. All clients need to explicitly agree in order for that to work
properly.

Then, there's also the requirement: "We want to provide for ongoing
binary compatibility as the transport design evolves." [1] (IOW: "don't
break existing jack applications") which complicates the actual
implementation.

That being said, yes, it would be cool. Multiple independent transports
would be very nice as well, and I want a pony, too :)

ciao,
robin

[1] http://jackaudio.org/api/transport-design.html

From termtech at rogers.com  Sun Aug 10 17:31:12 2014
From: termtech at rogers.com (Tim E. Real)
Date: Sun, 10 Aug 2014 13:31:12 -0400
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
Message-ID: <2458841.xuWy5zZaIv@col-desktop>

On August 10, 2014 09:24:12 AM Len Ovens wrote:
> Tim E. Real wrote:
> > I believe it is crucial that all the apps (MusE included) use the complete
> > full feature set of the Jack Transport API for the best accuracy.
> 
> Is there a record enable in there? It seems to me it would be worthwhile
> for recording audio in one application and MIDI in another or for punch in
> out. The application would choose to see or ignore such a signal.

Mm, no such signal in the Jack Transport API that I know of.

But there are Midi commands for such things like play stop record etc, 
 some of which MusE does recognize, but some are waiting to be added.
Not sure about the full 'record' family of Midi commands, must check
 which have been added... I think basic record start and stop are supported.

Tim.

From termtech at rogers.com  Sun Aug 10 17:47:50 2014
From: termtech at rogers.com (Tim E. Real)
Date: Sun, 10 Aug 2014 13:47:50 -0400
Subject: [LAU] [Bulk] Re: Jack transport - was - Ardour/Muse Jack tempo
	lock
In-Reply-To: <53E7A7BC.90008@gareus.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <53E7A7BC.90008@gareus.org>
Message-ID: <2106311.W7MHbJi89h@col-desktop>

On August 10, 2014 07:11:24 PM Robin Gareus wrote:
> On 08/10/2014 06:24 PM, Len Ovens wrote:
> > Tim E. Real wrote:
> >> I believe it is crucial that all the apps (MusE included) use the
> >> complete
> >> full feature set of the Jack Transport API for the best accuracy.
> 
> ...and port latencies! Sadly, many jack applications ignore those which
> leads to offsets, even though they share transport position.

Yeah I know. Bummer. Recorded waves are incorrectly shifted etc.

I'm working very hard at the moment on adding automatic + manual 
 latency compensation to MusE. 

It is sooo crucial  - currently the top priority for me !

Man, it sure ain't easy, at least in MusE's case...


PS: While on the subject...

Regarding LADSPA and DSSI and VST DSSI plugins:

Many of them report a latency through a 'latency' output port.
But is there a standard? 
Do some report frames of latency while others report in milliseconds?

Tim.

> 
> > Is there a record enable in there?
> 
> No, there isn't.
> 
> > It seems to me it would be worthwhile
> > for recording audio in one application and MIDI in another or for punch in
> > out. The application would choose to see or ignore such a signal.
> 
> It's not that easy.
> Record enable is an application state not a position in time.
> Rec-arm needs to allocate buffers, prepare files on disk etc etc. It's
> not realtime-safe. For playback there's already a special state:
> JackTransportStarting. Applications should acknowledge that they're
> ready to roll before jack goes into "play". A lot of jack-clients forgo
> this which is already trouble enough, though not directly harmful in
> this case (just possibly out of sync playback).
> 
> Adding ranges (loop, punch), application state (rec-en), or even
> varispeed support would likely increase the mess rather than solve
> anything. All clients need to explicitly agree in order for that to work
> properly.
> 
> Then, there's also the requirement: "We want to provide for ongoing
> binary compatibility as the transport design evolves." [1] (IOW: "don't
> break existing jack applications") which complicates the actual
> implementation.
> 
> That being said, yes, it would be cool. Multiple independent transports
> would be very nice as well, and I want a pony, too :)
> 
> ciao,
> robin
> 
> [1] http://jackaudio.org/api/transport-design.html


From robin at gareus.org  Sun Aug 10 19:56:31 2014
From: robin at gareus.org (Robin Gareus)
Date: Sun, 10 Aug 2014 21:56:31 +0200
Subject: [LAU] [Bulk] Re: Jack transport - was - Ardour/Muse Jack tempo
 lock
In-Reply-To: <2106311.W7MHbJi89h@col-desktop>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <53E7A7BC.90008@gareus.org> <2106311.W7MHbJi89h@col-desktop>
Message-ID: <53E7CE6F.2070100@gareus.org>

On 08/10/2014 07:47 PM, Tim E. Real wrote:
> 
> PS: While on the subject...
> 
> Regarding LADSPA and DSSI and VST DSSI plugins:
> 
> Many of them report a latency through a 'latency' output port. But
> is there a standard? Do some report frames of latency while others
> report in milliseconds?

LV2 it is always in audio samples
http://lv2plug.in/ns/lv2core/#reportsLatency

With LADSPA, I don't know an official spec, but an output control port
with the untranslatable name "latency" (all lowercase) is assumed to
report the latency in audio-samples.

VST also uses samples, but for the API spec you'll have to check
yourself. I don't know for DSSI.

ciao,
robin

PS. latency need not be an integer number, there can be sub-sample
latencies.

From abonnements at revolwear.com  Mon Aug 11 00:10:22 2014
From: abonnements at revolwear.com (Max)
Date: Mon, 11 Aug 2014 09:10:22 +0900
Subject: [LAU] jack: interface selection has no effect
In-Reply-To: <alpine.DEB.2.02.1408091147310.23832@scott.cbbs.org>
References: <53E512BD.8070107@revolwear.com>
 <alpine.DEB.2.02.1408091147310.23832@scott.cbbs.org>
Message-ID: <53E809EE.70009@revolwear.com>

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On 08/10/2014 03:53 AM, Len Ovens wrote:
> On Sat, 9 Aug 2014, Max wrote:
> 
>> selecting a different interface in qjackctl has no effect.  i
>> can't use the external soundcard, because jack stays always with
>> the internal one. any ideas what causes this symptom?
>> 
>> i am running pulse on top of jack like this: 
>> http://trac.jackaudio.org/wiki/WalkThrough/User/PulseOnJack

Len, thanks for trying to help me out - appreciated. But I am more
confused than before.

> That is the hardest way to do it.

to do what?

> qjackctl expects (as installed) to use jackdbus.

i have no jackdbus package in my apt sources available, what's that?

> If you wish to use qjackctl with jackd you have to turn dbus off in
> the qjackctl settings. However, if jackd is started externally and
> not from qjackctl you will still not be able to stop jackd.

with dbus turned on or off I have the same symptoms: that changes in
jackctl have no effect.

> The only reason to do things this way is that you are using jackd1,
> if you are using jackd2,

jack2 is running

> and you have dbus running in the session anyway, then run
> jackdbus.

$ jackdbus
jackdbus should be auto-executed by D-Bus message bus daemon.
If you want to run it manually anyway, specify "auto" as only parameter
$ jackdbus auto

?

> If you wish to run it from the command line use jack_control start.
> If you start it with qjackctl first with the settings you want,
> that will be enough, other wise you will have to learn how to set
> parameters with jack_control. There is no man page so try
> jack_control --help

I wish that the settings in jackctl will have effect.

m.


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From len at ovenwerks.net  Mon Aug 11 05:23:21 2014
From: len at ovenwerks.net (Len Ovens)
Date: Sun, 10 Aug 2014 22:23:21 -0700
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <53E7A7BC.90008@gareus.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <53E7A7BC.90008@gareus.org>
Message-ID: <6c50fe5ef05fb7a22a2ef937f716c29d.squirrel@ssl.ovenwerks.net>


On Sun, August 10, 2014 10:11 am, Robin Gareus wrote:
> On 08/10/2014 06:24 PM, Len Ovens wrote:

>> Is there a record enable in there?
>
> No, there isn't.
>
>> It seems to me it would be worthwhile
>> for recording audio in one application and MIDI in another or for punch
>> in
>> out. The application would choose to see or ignore such a signal.
>
> It's not that easy.
> Record enable is an application state not a position in time.
> Rec-arm needs to allocate buffers, prepare files on disk etc etc. It's
> not realtime-safe.

I guess I thought that happened when the track was armed, but I do know
that some apps create new tracks if there are no armed tracks when the
master record is hit. your answer made me think...


> That being said, yes, it would be cool. Multiple independent transports
> would be very nice as well, and I want a pony, too :)

hope it comes with someone to muck...


-- 
Len Ovens
www.OvenWerks.net


From fons at linuxaudio.org  Mon Aug 11 10:05:59 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Mon, 11 Aug 2014 10:05:59 +0000
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <53E7A7BC.90008@gareus.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <53E7A7BC.90008@gareus.org>
Message-ID: <20140811100559.GA957@linuxaudio.org>

On Sun, Aug 10, 2014 at 07:11:24PM +0200, Robin Gareus wrote:
 
> It's not that easy.
> Record enable is an application state not a position in time.
> Rec-arm needs to allocate buffers, prepare files on disk etc etc.
> It's not realtime-safe.

That is a weak excuse, and it hides the real reason. Which is
that Jack transport does not require apps using it to remain
ready to run while stopped, nor provides any means for an app
to report 'not ready' until it's too late.

If you need to start 'on cue', which is a rather common thing
in audio engineering, the present system just fails.

Sure, repositioning, creating new tracks or arming some of them
for recording involves things that can't be done instantly and
that are not RT safe. But a few seconds after a well-designed
app is last repositioned or reconfigured it should be ready to
start or go into recording mode instantly. And to make this
useful at all the app should be able to report its readyness
to a shared transport control so that 'one or more not ready'
can be shown to the user somehow (typically done by flashing
the START button).

All that is required is

1. split the 'stopped' state in two: ready or not ready to
   run as configured,

2. require all apps, while stopped, to do whatever it takes
   to get ready ASAP when reconfigured and report 'not ready'
   meanwhile.

I've proposed this a number of times over the last years, it
was ignored each time. (1) is easy enough to implement, (2)
is for application authors, not for the Jack team.

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From len at ovenwerks.net  Mon Aug 11 19:41:46 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 11 Aug 2014 12:41:46 -0700 (PDT)
Subject: [LAU] jack: interface selection has no effect
In-Reply-To: <53E809EE.70009@revolwear.com>
References: <53E512BD.8070107@revolwear.com>
 <alpine.DEB.2.02.1408091147310.23832@scott.cbbs.org>
 <53E809EE.70009@revolwear.com>
Message-ID: <alpine.DEB.2.02.1408111153150.23832@scott.cbbs.org>

On Mon, 11 Aug 2014, Max wrote:
> On 08/10/2014 03:53 AM, Len Ovens wrote:
>> On Sat, 9 Aug 2014, Max wrote:
>>
>>> selecting a different interface in qjackctl has no effect.  i
>>> can't use the external soundcard, because jack stays always with
>>> the internal one. any ideas what causes this symptom?
>>>
>>> i am running pulse on top of jack like this:
>>> http://trac.jackaudio.org/wiki/WalkThrough/User/PulseOnJack
>
> Len, thanks for trying to help me out - appreciated. But I am more
> confused than before.
>
>> That is the hardest way to do it.
>
> to do what?

Pulse on jack.

>> qjackctl expects (as installed) to use jackdbus.
>
> i have no jackdbus package in my apt sources available, what's that?

It is a part of jackd2. Normally if you have jackd2 you also have 
jackdbus.

Then, if you use qjackctl to start jack first thing in the session before 
you run anything else or start jackd through a script. Then whatever 
device you have set up should get picked up by jackdbus. If you run 
something that needs jack first, chances are it will start jackd for you 
and once it is running, qjackctl has no control over it and can not stop 
and restart it. My personal feeling is that debian/ubuntu etc. should 
package jackd2 and jackdbus seperate so jackd is not available in a 
jackdbus system to cause problems. A script that takes the same 
commandline as jackd and then calls jack_control with the right parameters 
could be included instead.

Using jackdbus, and installing pulseaudio-module-jack, should besides 
installing module-jack-sink and module-jack-source also install 
module-jackdbus-detect which will auto-load the other two modules for you 
when jackdbus is detected as running.

Personally, I have a script like:

----------autojack----------8<---------
#! /bin/bash
# start jack and midi

DELAY=1
DRIVER=alsa
DEV=hw:0
RATE=48000
FRAME=2048
PERIOD=2

#sleep $DELAY

jack_control ds $DRIVER dps device $DEV dps rate $RATE dps period $FRAME \
         dps nperiods $PERIOD start

sleep $DELAY

a2jmidid -e &

sleep $DELAY

sleep $DELAY

pulseaudio -k

----------------------------8<------------------

The last line restarts pulseaudio because upstart seems to think it has to 
start pulse first :P  And I don't want my system too far off stock because 
I try to help those with a stock system. This gets started by:
~/.config/autostart/AutoJack.desktop
But I could have put it in /etc/xdg/autostart for system wide use.

---------AutoJack.desktop-------8<--------------
[Desktop Entry]
Encoding=UTF-8
Version=0.9.4
Type=Application
Name=AutoJack
Comment=Jackdbus starter
Exec=autojack
StartupNotify=false
Terminal=false
Hidden=false

-------------------------------8<---------------

jack_lsp shows:
system:capture_1
system:capture_2
... (the rest of my system listing skipped)
system:playback_9
system:playback_10
PulseAudio JACK Sink:front-left
PulseAudio JACK Sink:front-right
PulseAudio JACK Source:front-left
PulseAudio JACK Source:front-right
a2j:Midi Through [14] (capture): Midi Through Port-0
a2j:Midi Through [14] (playback): Midi Through Port-0
a2j:Ensoniq AudioPCI [16] (capture): ES1370
a2j:Ensoniq AudioPCI [16] (playback): ES1370

qjackctl is configured for the same card and start settings, the options 
tab has Execute script on startup: pulseaudio -k and Execute script after 
startup: a2jmidid -e &, but really I never use qjackctl to start and stop 
jackd as it runs from session start to stop. I use qjackctl for the 
patchbay (connections) and logging. In the Misc. tab I have qjackctl set 
for Enable D-Bus interface and I have Stop JACK audio server on 
application exit unchecked so that stopping qjackctl does not stop 
jackdbus.

I use this project:

http://www.ovenwerks.net/software/index.html

To control jack latency on the fly and to disconnect pulseaudio if I think 
it might interfere with what I am doing. It also sets the CPU governor and 
allows me to stop cron so there is no sudden net/disk activity in the 
middle of a take.

>From pavucontrol, on the Configuration tab, I set any audio card profiles 
to "Off". I probably should just set pulse not to use the alsa-sink/sourse 
modules, but so far I have been too lazy to do so.

In my opinion this is the proper way to get jack and pulse to play nice.

If one is using jackd1 Then everything changes because it is not dbus 
controlable (at least out of the box, I hear there is a patch)

Then the link you have above can be used and qjackctl can have the D-Bus 
interface turned off. However, there is more work to turn the pulse-jack 
bridge off and on on the fly as there are two modules to (un)load. Also, 
jackd has to be started with qjackctl if you wish to use that as the 
controler. Unless you want to add:
Execute script on Shutdown: killall -9 jackd
To your qjackctl setup.

I can not recommend jackd2 or jackd1 as one being better than the other. I 
think there are situations where one or the other shines. However, in the 
case where pulse and dbus are already in use, I would choose jackdbus as 
the most compatible version of jack to use. I do not know if jackd can 
safely be removed/-x/renamed or not when using jackdbus... that is 
something I should experiment with though  :)

--
Len Ovens
www.ovenwerks.net


From gianfranco at portalmod.com.br  Mon Aug 11 21:54:12 2014
From: gianfranco at portalmod.com.br (Gianfranco Ceccolini)
Date: Mon, 11 Aug 2014 18:54:12 -0300
Subject: [LAU] [ANN] MOD Duo's Kickstarter campaign to be launched in mid
	September
Message-ID: <CAOsA5s-p_=a_PZULiMqtKoFjSc0Z3tRPjpxYoYNaQejU+9NNaQ@mail.gmail.com>

Greetings Linux Audio Users and Developers !!!

I'm very happy to inform that we will launch the MOD Duo's Kickstarter
campaign in mid September.

The MOD Duo is our second model and we've been putting a lot of engineering
in it based on the feedback we had from the MOD Quadra experience.

We deeply hope it becomes a device that empowers the Linux Audio community,
bringing together developers and musicians.

A pre-campaign site was created to warm up the communication engines:

http://stepontothefuture.com.

Hope you all enjoy and spread the word

Kind regards

Gianfranco Ceccolini
The MOD Team
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From termtech at rogers.com  Tue Aug 12 00:01:40 2014
From: termtech at rogers.com (Tim E. Real)
Date: Mon, 11 Aug 2014 20:01:40 -0400
Subject: [LAU] [Bulk] Re: Jack transport - was - Ardour/Muse Jack tempo
	lock
In-Reply-To: <20140811100559.GA957@linuxaudio.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <53E7A7BC.90008@gareus.org> <20140811100559.GA957@linuxaudio.org>
Message-ID: <3967821.5Hi4RIW6Gm@col-desktop>

On August 11, 2014 10:05:59 AM Fons Adriaensen wrote:
> On Sun, Aug 10, 2014 at 07:11:24PM +0200, Robin Gareus wrote:
> > It's not that easy.
> > Record enable is an application state not a position in time.
> > Rec-arm needs to allocate buffers, prepare files on disk etc etc.
> > It's not realtime-safe.
> 
> That is a weak excuse, and it hides the real reason. Which is
> that Jack transport does not require apps using it to remain
> ready to run while stopped, nor provides any means for an app
> to report 'not ready' until it's too late.

Mm, do you know about the 'slow-sync' Jack callback function?
Or do you propose something slightly different?
As in, able to report readiness anytime all the time even in stop mode?

The 'slow-sync' Jack callback function is a great feature for
 holding up the transport until all the apps report they are 
 'ready to roll'.

But, the callback is not polled until a transport start is initiated,
 whereupon the transport enters a special state JackTransportStarting
 and doesn't roll until all the apps agree, or a timeout value has expired.

We already use the callback to hold off Jack Transport until we have updated
 our wave playback caches and so on - when everything is ready to roll.

I reckon this callback will play a role in my efforts to add automatic + manual 
 latency compensation to MusE:
'Live' stream sources such as Jack inputs, and routing paths which have 
 plugin latencies, and so on, can be aligned simply by adding artificial delay 
 (that's extra latency, but unavoidable) to the various paths so that 
 everything aligns.
But there is one stream source which has a trick up its sleeve:
Playback sources. As in wave file playback (MusE Wave Track for example).
These sources can be queued up ahead of time, and the only way I see
 to do that is to use the 'slow-sync' Jack callback.
I can let the audio play for a moment while I hold up the Jack Transport
 and then let it go.

There is one situation that is tough: when an external HW midi device
 initiates the play via midi play command for example.
There is no way to tell it to hold off for a moment.
Well, there might be, if the midi interface is bi-directional (in + out).
Also, one could delay all input coming from the external HW midi device
 provided all its outputs are fed through the computer. But if for example 
 its audio outputs are fed right into a mixer, well, out of luck there...

I'm not sure that knowing all the time if the apps are ready to roll 
 even in stop mode would be helpful.
I mean, if the likely intention is to eventually go into play mode,
 then the 'slow-sync' Jack callback does what we need. Right?

> If you need to start 'on cue', which is a rather common thing
> in audio engineering, the present system just fails.
> 
> Sure, repositioning, creating new tracks or arming some of them
> for recording involves things that can't be done instantly and
> that are not RT safe. But a few seconds after a well-designed
> app is last repositioned or reconfigured it should be ready to
> start or go into recording mode instantly. And to make this
> useful at all the app should be able to report its readyness
> to a shared transport control so that 'one or more not ready'
> can be shown to the user somehow (typically done by flashing
> the START button).
> 
> All that is required is
> 
> 1. split the 'stopped' state in two: ready or not ready to
>    run as configured,
> 
> 2. require all apps, while stopped, to do whatever it takes
>    to get ready ASAP when reconfigured and report 'not ready'
>    meanwhile.

But... not ready for what? The app cannot know what the
 next transport intention is until the actual command
 comes down the line. For example a transport repositioning.
So the 'slow-sync' Jack callback seems to be the only way 
 to deal with this: Broadcast the actual command and wait until
 all the apps are ready to roll.

Cheers.
Tim.

> 
> I've proposed this a number of times over the last years, it
> was ignored each time. (1) is easy enough to implement, (2)
> is for application authors, not for the Jack team.


From len at ovenwerks.net  Tue Aug 12 02:21:17 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 11 Aug 2014 19:21:17 -0700 (PDT)
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <2458841.xuWy5zZaIv@col-desktop>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <2458841.xuWy5zZaIv@col-desktop>
Message-ID: <alpine.DEB.2.02.1408111450200.23832@scott.cbbs.org>

On Sun, 10 Aug 2014, Tim E. Real wrote:

> On August 10, 2014 09:24:12 AM Len Ovens wrote:
>>
>> Is there a record enable in there? It seems to me it would be worthwhile
>
> Mm, no such signal in the Jack Transport API that I know of.
>
> But there are Midi commands for such things like play stop record etc,
> some of which MusE does recognize, but some are waiting to be added.
> Not sure about the full 'record' family of Midi commands, must check
> which have been added... I think basic record start and stop are supported.

Ya midi is there, one way or another. Most Applications that support 
recording "something" and jack sync/transport also support punch in/out 
via a midi controler.

MMC is supposed to be real time as well. I do not know the right answer. 
In my case I am making software for a midi control surface, but because 
the "midi" is generated after the signals get into the computer, I am also 
able to use the control surface to control jack transport directly (in 
fact that was easier than sending jack MIDI). From that point of view, 
having a record start would be an ideal global solution. Also, there is 
some software (the non-* group stands out in my mind) that does not have 
MIDI in for control purposes, but reacts to jack transport very nicely. 
One of the sub projects of this software is to split the computer keyboard 
so that the numeric keypad can be used as a midi/jacktransport control 
surface while the rest works as the system keyboard still, so I have an 
interest in a more complete jack transport from that POV. However, I am 
now using either and/or as the user sees fit. One way or another it will 
get done. It seems some controlers just send whatever midi note on is next 
on the table for record enable. However I am less worried about that 
because each key/switch can send a user set midi string/event. I will 
provide some preset ideas that seem useful to me.


--
Len Ovens
www.ovenwerks.net


From abonnements at revolwear.com  Tue Aug 12 02:34:20 2014
From: abonnements at revolwear.com (Max)
Date: Tue, 12 Aug 2014 11:34:20 +0900
Subject: [LAU] jack: interface selection has no effect
In-Reply-To: <alpine.DEB.2.02.1408111153150.23832@scott.cbbs.org>
References: <53E512BD.8070107@revolwear.com>
 <alpine.DEB.2.02.1408091147310.23832@scott.cbbs.org>
 <53E809EE.70009@revolwear.com>
 <alpine.DEB.2.02.1408111153150.23832@scott.cbbs.org>
Message-ID: <53E97D2C.4000605@revolwear.com>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 08/12/2014 04:41 AM, Len Ovens wrote:
> On Mon, 11 Aug 2014, Max wrote:
>> On 08/10/2014 03:53 AM, Len Ovens wrote:
>>> On Sat, 9 Aug 2014, Max wrote:
>>> 
>>>> selecting a different interface in qjackctl has no effect.
>>>> i can't use the external soundcard, because jack stays always
>>>> with the internal one. any ideas what causes this symptom?
>>>> 
>>>> i am running pulse on top of jack like this: 
>>>> http://trac.jackaudio.org/wiki/WalkThrough/User/PulseOnJack
>> 
>> Len, thanks for trying to help me out - appreciated. But I am
>> more confused than before.
>> 
>>> That is the hardest way to do it.
>> 
>> to do what?
> 
> Pulse on jack.

Would you suggest that your method is better and should be the
canonical way to do pulse on jack? If so, shall we write up an
alternative WalkThrough on the jackoudio wiki that helps people
setting that up?
Is that software you use (wrote?) something you want to share?  It
looks very useful.

m.

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From len at ovenwerks.net  Tue Aug 12 02:52:48 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 11 Aug 2014 19:52:48 -0700 (PDT)
Subject: [LAU] jack: interface selection has no effect
In-Reply-To: <53E97D2C.4000605@revolwear.com>
References: <53E512BD.8070107@revolwear.com>
 <alpine.DEB.2.02.1408091147310.23832@scott.cbbs.org>
 <53E809EE.70009@revolwear.com>
 <alpine.DEB.2.02.1408111153150.23832@scott.cbbs.org>
 <53E97D2C.4000605@revolwear.com>
Message-ID: <alpine.DEB.2.02.1408111941010.23832@scott.cbbs.org>

On Tue, 12 Aug 2014, Max wrote:

> On 08/12/2014 04:41 AM, Len Ovens wrote:
>> On Mon, 11 Aug 2014, Max wrote:
>>> On 08/10/2014 03:53 AM, Len Ovens wrote:
>>>> On Sat, 9 Aug 2014, Max wrote:
>>>>
>>>>> selecting a different interface in qjackctl has no effect.
>>>>> i can't use the external soundcard, because jack stays always
>>>>> with the internal one. any ideas what causes this symptom?
>>>>>
>>>>> i am running pulse on top of jack like this:
>>>>> http://trac.jackaudio.org/wiki/WalkThrough/User/PulseOnJack
>>>
>>> Len, thanks for trying to help me out - appreciated. But I am
>>> more confused than before.
>>>
>>>> That is the hardest way to do it.
>>>
>>> to do what?
>>
>> Pulse on jack.
>
> Would you suggest that your method is better and should be the
> canonical way to do pulse on jack? If so, shall we write up an
> alternative WalkThrough on the jackoudio wiki that helps people
> setting that up?

I certainly think using jackdbus when dbus is present is the better way, 
Not so sure (yet) if my exact setup is best. It has only been in use for a 
few months. It is also not quite finished. I need to package it at least 
for debian which I am reasonably familiar with and perhaps the old tar 
ball method too. I have not done a good look at the depends yet :)  and 
would like to add a config gui for autojack. It is written in pyqt and 
shell script.

> Is that software you use (wrote?) something you want to share?  It
> looks very useful.

In due time. I thought vacation would give me some time to finish it out 
so it was distributable.... alas, I have not had any time to myself, but 
camping was fun too :)

--
Len Ovens
www.ovenwerks.net


From termtech at rogers.com  Tue Aug 12 03:36:37 2014
From: termtech at rogers.com (Tim E. Real)
Date: Mon, 11 Aug 2014 23:36:37 -0400
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <alpine.DEB.2.02.1408111450200.23832@scott.cbbs.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <2458841.xuWy5zZaIv@col-desktop>
 <alpine.DEB.2.02.1408111450200.23832@scott.cbbs.org>
Message-ID: <3083133.S6eD1oE51i@col-desktop>

On August 11, 2014 07:21:17 PM you wrote:
> On Sun, 10 Aug 2014, Tim E. Real wrote:
> > On August 10, 2014 09:24:12 AM Len Ovens wrote:
> >> Is there a record enable in there? It seems to me it would be worthwhile
> > 
> > Mm, no such signal in the Jack Transport API that I know of.
> > 
> > But there are Midi commands for such things like play stop record etc,
> > some of which MusE does recognize, but some are waiting to be added.
> > Not sure about the full 'record' family of Midi commands, must check
> > which have been added... I think basic record start and stop are
> > supported.
> 
> Ya midi is there, one way or another. Most Applications that support
> recording "something" and jack sync/transport also support punch in/out
> via a midi controler.
> 
> MMC is supposed to be real time as well. I do not know the right answer.

MusE does MMC as well.
Partial MTC too, but no actual MTC syncing :-(

> In my case I am making software for a midi control surface, but because
> the "midi" is generated after the signals get into the computer, I am also
> able to use the control surface to control jack transport directly (in
> fact that was easier than sending jack MIDI). From that point of view,
> having a record start would be an ideal global solution. Also, there is
> some software (the non-* group stands out in my mind) that does not have
> MIDI in for control purposes, but reacts to jack transport very nicely.
> One of the sub projects of this software is to split the computer keyboard
> so that the numeric keypad can be used as a midi/jacktransport control
> surface while the rest works as the system keyboard still, so I have an
> interest in a more complete jack transport from that POV. 

Hmm ... 

The only way I could see this in terms of the Jack Transport 'states'
 is to add some new states:

For example:
JackTransportRecordArm (maybe to help prepare)
JackTransportRecordEnable (subsequent play mode actually means record mode).

But those do not qualify as 'states', they are just commands !

So we might propose something like:
JackTransportRecording (in actual recording state)

But then we might also need something like:
JackTransportRecordStarting (so that the app can differentiate between 
 play starting and record starting modes, and tell Jack Transport to wait)

But then, that's supposed to be the job of the slow-sync callback - holding
 up the transport if necessary. Maybe pass a flag to the callback or something.

And then there's the actual Jack commands to initiate all of this, such as
 jack_transport_record() and/or
 jack_transport_record_arm()

An argument against might be why stop there - let's throw in all kinds of 
 other states like punch in/out. But realistically these simple record states 
 might be OK. I mean, we have play state already. 
So record states seem a logical progression?

Apart from states, other methods of telling the app about record modes 
 might include a new 'record' callback or something...

Hope I can get some comments from folks who know more.

Obviously difficult to ensure compatibility with existing apps, I suppose...

Again, maybe the new Jack Metadata API is the answer.

> However, I am
> now using either and/or as the user sees fit. One way or another it will
> get done. It seems some controlers just send whatever midi note on is next
> on the table for record enable. However I am less worried about that
> because each key/switch can send a user set midi string/event. I will
> provide some preset ideas that seem useful to me.

In MusE you can map any midi note to any of the transport functions :-)
For example hit a high 'C' and it starts playing.
It has mappable Stop, Record, Goto Left Marker, Play, and Step functions.

Cheers.
Tim.

> 
> 
> --
> Len Ovens
> www.ovenwerks.net


From dj_kaza at hotmail.com  Tue Aug 12 05:22:55 2014
From: dj_kaza at hotmail.com (Kaza Kore)
Date: Tue, 12 Aug 2014 05:22:55 +0000
Subject: [LAU] Ubuntu Studio 14.04 Pulse removal question
Message-ID: <DUB113-W258174F3D395A5066FC6A28FEA0@phx.gbl>

All.

I have recently installed a fresh Ubuntu Studio 14.04, replacing my old 12.04 version I was running previously. In short; I believe I now have everything running directly through Jack, which is autostarted with a "pulseaudio -k" command and wonder whether it is worthwhile actually fully removing PA? And if so the best way to do this/exact packages to remove?

For those that are interested the steps I took were:
Compile and install libflashsupport-jack library following the instructions at http://jackaudio.org/faq/routing_flash.html (there was one more lib not listed I also had to install for <video.h>)
Install jack plugin for vlc, gstreamer and any media players I use.
Test audio is really going through Jack and not via the pulse-sync outputs. Test audio still present after running "pulseaudio -k".*
Remove autostart options: PulseAudio Sound System & XFCE Volume Deamon. **
Add "pulseaudio -k" to the startup command in Jack. And qjackctl as an autostart program.
Remap mediakeys to control level of amixer's Master. (eg (may be different on your setup) volume up 2dB: "amixer -c 0 sset 'Master',0 2dB+")

*Seems to be working nicely and when running "pulseaudio -k" the first time I instantly noticed the percentage time used by Jack approximately dropped by a half! (4% -2 % with just Audacious playing.) Seems to let me go maybe one setting lower. Not really tested yet though...

So thoughts on fully removing PA please :)

**These steps didn't seem to make as much difference as I thought they should do! Even though I thought I had stopped PA from running the pulse/jack sync still showed up in jack's connections and running "pulseaudio -k" still killed it, giving me the extra overhead in jack. This is why I added the command to the jack startup option. Surely these options should have stopped it from running in the first place, no??


Dale.
 		 	   		  
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From len at ovenwerks.net  Tue Aug 12 05:30:11 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 11 Aug 2014 22:30:11 -0700 (PDT)
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <3083133.S6eD1oE51i@col-desktop>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <2458841.xuWy5zZaIv@col-desktop>
 <alpine.DEB.2.02.1408111450200.23832@scott.cbbs.org>
 <3083133.S6eD1oE51i@col-desktop>
Message-ID: <alpine.DEB.2.02.1408112041270.23832@scott.cbbs.org>

On Mon, 11 Aug 2014, Tim E. Real wrote:

> On August 11, 2014 07:21:17 PM you wrote:
>> On Sun, 10 Aug 2014, Tim E. Real wrote:
>>> On August 10, 2014 09:24:12 AM Len Ovens wrote:
>>>> Is there a record enable in there? It seems to me it would be worthwhile
>> One of the sub projects of this software is to split the computer keyboard
>> so that the numeric keypad can be used as a midi/jacktransport control
>> surface while the rest works as the system keyboard still, so I have an
>> interest in a more complete jack transport from that POV.
>
> Hmm ...
>
> The only way I could see this in terms of the Jack Transport 'states'
> is to add some new states:
>
> For example:
> JackTransportRecordArm (maybe to help prepare)
> JackTransportRecordEnable (subsequent play mode actually means record mode).

MMC, from what I have read, seems to support two record enables:

  - arm tracks for recording. While these are in the realtime section of 
commands, I would think these would be the commands run before roll to 
give sw time to get things ready to record.
  - Then there is punchin/out or master recordenable which is meant to be 
realtime and should start the app recording with no transport stumble or 
wait.

Now maybe there are people who think using track-rec-enable for 
punchin/out is the way to go, but I can't see myself working that way. I 
also do not think it would be the right thing to do, to expect jack 
transport (or any other transport) to arm tracks of multiple apps... or 
even one app, because this would assume a static number of tracks. The 
punch in/out master rec-enable does make sense though. all the enabled 
tracks would record at the same time.

> But those do not qualify as 'states', they are just commands !

Yup.
>
> So we might propose something like:
> JackTransportRecording (in actual recording state)

Ok, I, in my limited POV, don't see a need for this unless one of the apps 
fails to start recording and wants to signal this somehow.
>
> But then we might also need something like:
> JackTransportRecordStarting (so that the app can differentiate between
> play starting and record starting modes, and tell Jack Transport to wait)

IMO, they are the same thing. When getting ready to play, anything that 
needs to be ready for record should be done too. The transport can not 
wait at the punch in point for sw to get ready or else the whole punch in 
concept is void. Any sw that has a track recenabled should be making sure 
it is ready to record on those tracks before playing. It _may_ be OK for 
an app to start recording some ms late provided the actual time of the 
record starting is saved as the punchin point for that app... this gets 
messy real quick though.

If there is to be any "get ready time" at the record strobe, the play 
state must not change once started just for recording sake, better a 
system wide record start delay of the punch in point.

I still feel the app should be ready to record anywhere from the play 
start with no delay.

> Obviously difficult to ensure compatibility with existing apps, I suppose...

I would expect an old app to ignore it (them, punch in and out) and lets 
get fancy and ask for the moon... jack transport would of course drop in a 
MIDI port for telling these old apps to start recording  :D

> In MusE you can map any midi note to any of the transport functions :-)
> For example hit a high 'C' and it starts playing.
> It has mappable Stop, Record, Goto Left Marker, Play, and Step functions.

With the variety of MIDI control surfaces around, one pretty much has to.

And now having gotten my feet wet with jack midi stuff... I am thinking of 
a MIDI control surface mapping app that takes MIDI in from a control 
surface and splits it for a number of applications. The idea being that a 
controller that has 8 channels and does banks, might send controls for 
bank 1 and 2 to one app and bank 3 to a second and maybe bank 4 to a 
third. While it may be possible to auto figure out how many banks each app 
has, the operator might be better for settng it up and knowing what is 
what. Of course as apps become more "do everything" the use for such a 
thing may be waning.

I have looked at OSC... but it has no standards (bad way of trying to 
make my point), each signal seems to have 
to be directed at a particular app/control. There seems to be no generic 
controls where a generic control surface can be used. Maybe that is 
because all the docs I have read are showing off it's flexablity. It seems 
even a native OSC control surface would need middleware to rename the 
signal so the destination was correctly chosen.

--
Len Ovens
www.ovenwerks.net


From len at ovenwerks.net  Tue Aug 12 05:36:39 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 11 Aug 2014 22:36:39 -0700 (PDT)
Subject: [LAU] Ubuntu Studio 14.04 Pulse removal question
In-Reply-To: <DUB113-W258174F3D395A5066FC6A28FEA0@phx.gbl>
References: <DUB113-W258174F3D395A5066FC6A28FEA0@phx.gbl>
Message-ID: <alpine.DEB.2.02.1408112232070.23832@scott.cbbs.org>

On Tue, 12 Aug 2014, Kaza Kore wrote:

> **These steps didn't seem to make as much difference as I thought they
> should do! Even though I thought I had stopped PA from running the
> pulse/jack sync still showed up in jack's connections and running
> "pulseaudio -k" still killed it, giving me the extra overhead in jack. This
> is why I added the command to the jack startup option. Surely these options
> should have stopped it from running in the first place, no??

pulseaudio -k should restart pulseaudio. Try: settings manager->session 
and startup->Application Autostart->uncheck pulseaudio. Then logout and 
back in.

--
Len Ovens
www.ovenwerks.net


From dj_kaza at hotmail.com  Tue Aug 12 06:07:09 2014
From: dj_kaza at hotmail.com (Kaza Kore)
Date: Tue, 12 Aug 2014 06:07:09 +0000
Subject: [LAU] Ubuntu Studio 14.04 Pulse removal question
In-Reply-To: <alpine.DEB.2.02.1408112232070.23832@scott.cbbs.org>
References: <DUB113-W258174F3D395A5066FC6A28FEA0@phx.gbl>,
 <alpine.DEB.2.02.1408112232070.23832@scott.cbbs.org>
Message-ID: <DUB113-W9ED54F040152113EA17F98FEA0@phx.gbl>



> Date: Mon, 11 Aug 2014 22:36:39 -0700
> From: len at ovenwerks.net
> To: dj_kaza at hotmail.com
> CC: linux-audio-user at lists.linuxaudio.org
> Subject: Re: [LAU] Ubuntu Studio 14.04 Pulse removal question
> 
> On Tue, 12 Aug 2014, Kaza Kore wrote:
> 
> > **These steps didn't seem to make as much difference as I thought they
> > should do! Even though I thought I had stopped PA from running the
> > pulse/jack sync still showed up in jack's connections and running
> > "pulseaudio -k" still killed it, giving me the extra overhead in jack. This
> > is why I added the command to the jack startup option. Surely these options
> > should have stopped it from running in the first place, no??
> 
> pulseaudio -k should restart pulseaudio. Try: settings manager->session 
> and startup->Application Autostart->uncheck pulseaudio. Then logout and 
> back in.
> 
> --
> Len Ovens
> www.ovenwerks.net
> 

As far as I know and can tell -k stands for Kill. if you run it twice it informs you PA is not running!

As I stated I had unchecked the all Pulse related options in the Autostart settings and it still starts on reboot!

Dale.
 		 	   		  
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From len at ovenwerks.net  Tue Aug 12 06:02:50 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 11 Aug 2014 23:02:50 -0700 (PDT)
Subject: [LAU] Ubuntu Studio 14.04 Pulse removal question
In-Reply-To: <alpine.DEB.2.02.1408112232070.23832@scott.cbbs.org>
References: <DUB113-W258174F3D395A5066FC6A28FEA0@phx.gbl>
 <alpine.DEB.2.02.1408112232070.23832@scott.cbbs.org>
Message-ID: <alpine.DEB.2.02.1408112250580.23832@scott.cbbs.org>

On Mon, 11 Aug 2014, Len Ovens wrote:

> On Tue, 12 Aug 2014, Kaza Kore wrote:
>
>> **These steps didn't seem to make as much difference as I thought they
>> should do! Even though I thought I had stopped PA from running the
>> pulse/jack sync still showed up in jack's connections and running
>> "pulseaudio -k" still killed it, giving me the extra overhead in jack. This
>> is why I added the command to the jack startup option. Surely these options
>> should have stopped it from running in the first place, no??
>
> pulseaudio -k should restart pulseaudio. Try: settings manager->session and 
> startup->Application Autostart->uncheck pulseaudio. Then logout and back in.

Forgot to mention... any app that tries to talk to pulse via D-bus will 
start pulse anyway... and it will auto respawn every time you kill it. I 
think the only thing that does this is pavucontrol (started by the sound 
icon in systray if you select sound settings.) For your use I would turn 
off respawn at least so you can kill it rather than restart. Edit 
/etc/pulse/client.conf There is a more correct file to edit/create in 
~/.config/pulse/ but I forget what it should be called.

It may say:
; autospawn = yes
add a line right under like
autospawn = no
(the ; is a comment mark)

--
Len Ovens
www.ovenwerks.net


From len at ovenwerks.net  Tue Aug 12 06:20:05 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 11 Aug 2014 23:20:05 -0700 (PDT)
Subject: [LAU] Ubuntu Studio 14.04 Pulse removal question
In-Reply-To: <DUB113-W9ED54F040152113EA17F98FEA0@phx.gbl>
References: <DUB113-W258174F3D395A5066FC6A28FEA0@phx.gbl>,
 <alpine.DEB.2.02.1408112232070.23832@scott.cbbs.org>
 <DUB113-W9ED54F040152113EA17F98FEA0@phx.gbl>
Message-ID: <alpine.DEB.2.02.1408112305150.23832@scott.cbbs.org>

On Tue, 12 Aug 2014, Kaza Kore wrote:

> As far as I know and can tell -k stands for Kill. if you run it twice it
> informs you PA is not running!

Yes but pulse is configured respawn by default, it does take some time to 
restart, but does so as soon as you kill it.
> 
> As I stated I had unchecked the all Pulse related options in the Autostart
> settings and it still starts on reboot!

That could be... all it takes is something asking via D-bus if pulse is 
there... and then it is.

My personal solution has been to unload module-jackdbus-detect when I do 
not want pulse connected to jack. pulse takes almost no cpu that way. The 
command to unload the module is:

pactl unload-module module-jackdbus-detect

The package pulseaudio-module-jack can be removed to make this more 
permanent, but if you will not use pulse maybe just:

sudo chmod -x /usr/bin/pulseaudio

There may be depends problems if you remove the pulseaudio package 
itself... for example you browser may be removed as well. Debian depends 
are there to protect dummys.

--
Len Ovens
www.ovenwerks.net


From dj_kaza at hotmail.com  Tue Aug 12 06:40:59 2014
From: dj_kaza at hotmail.com (Kaza Kore)
Date: Tue, 12 Aug 2014 06:40:59 +0000
Subject: [LAU] Ubuntu Studio 14.04 Pulse removal question
In-Reply-To: <alpine.DEB.2.02.1408112305150.23832@scott.cbbs.org>
References: <DUB113-W258174F3D395A5066FC6A28FEA0@phx.gbl>,
 <alpine.DEB.2.02.1408112232070.23832@scott.cbbs.org>
 <DUB113-W9ED54F040152113EA17F98FEA0@phx.gbl>,
 <alpine.DEB.2.02.1408112305150.23832@scott.cbbs.org>
Message-ID: <DUB113-W135524935A2907F33BD9F5A8FEA0@phx.gbl>



> Date: Mon, 11 Aug 2014 23:20:05 -0700
> From: len at ovenwerks.net
> To: dj_kaza at hotmail.com
> CC: linux-audio-user at lists.linuxaudio.org
> Subject: RE: [LAU] Ubuntu Studio 14.04 Pulse removal question
> 
> On Tue, 12 Aug 2014, Kaza Kore wrote:
> 
> > As far as I know and can tell -k stands for Kill. if you run it twice it
> > informs you PA is not running!
> 
> Yes but pulse is configured respawn by default, it does take some time to 
> restart, but does so as soon as you kill it.

Doesn't seem to be respawning now, definitely was still loading at startup  before I added the pulseaudio -k command to jack options but my laptop has been running all night, so up for maybe 16 hours since last reboot and this is my output from trying to kill pulseaudio in terminal.

$ pulseaudio -k
E: [pulseaudio] main.c: Failed to kill daemon: No such process

Clearly not respawned itself in all that time. But the fact it will do if any program requests it is quite a good hint for me, and hopefully I will see a change if it pops up again when I try and use something.

> > 
> > As I stated I had unchecked the all Pulse related options in the Autostart
> > settings and it still starts on reboot!
> 
> That could be... all it takes is something asking via D-bus if pulse is 
> there... and then it is.
> 
> My personal solution has been to unload module-jackdbus-detect when I do 
> not want pulse connected to jack. pulse takes almost no cpu that way. The 
> command to unload the module is:
> 
> pactl unload-module module-jackdbus-detect

Once I'm totally sure nothing is trying to access pulse any more I might add this line. Is there any difference between adding it to rc.local and adding it as an Autostart option, like I did with qjackctl?
> 
> The package pulseaudio-module-jack can be removed to make this more 
> permanent, but if you will not use pulse maybe just:
> 
> sudo chmod -x /usr/bin/pulseaudio

And then this when even more confident ;)
> 
> There may be depends problems if you remove the pulseaudio package 
> itself... for example you browser may be removed as well. Debian depends 
> are there to protect dummys.
> 

Yeah I worried and thought I had heard there could be problems with removing. So I aim to fully ensure it's never running and I can be happy with that :)

> --
> Len Ovens
> www.ovenwerks.net
> 

Dale.
 		 	   		  
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From dj_kaza at hotmail.com  Tue Aug 12 06:59:24 2014
From: dj_kaza at hotmail.com (Kaza Kore)
Date: Tue, 12 Aug 2014 06:59:24 +0000
Subject: [LAU] Ubuntu Studio 14.04 Pulse removal question
In-Reply-To: <DUB113-W135524935A2907F33BD9F5A8FEA0@phx.gbl>
References: <DUB113-W258174F3D395A5066FC6A28FEA0@phx.gbl>,
 <alpine.DEB.2.02.1408112232070.23832@scott.cbbs.org>,
 <DUB113-W9ED54F040152113EA17F98FEA0@phx.gbl>,
 <alpine.DEB.2.02.1408112305150.23832@scott.cbbs.org>,
 <DUB113-W135524935A2907F33BD9F5A8FEA0@phx.gbl>
Message-ID: <DUB113-W78A414DCC37A4E100583698FEA0@phx.gbl>

*(PA) definitely was still loading at startup*

I just had a thought as to the cause of this. I think I might have still had my Volume widget in my Panel showing the control from PA at that point and since changed it to display amixer's. From what you say this likely would have caused PA to start, even though I had disabled it in Autostart.

Hope you forgive me for not confirming this was the cause right now :)

Dale.

From: dj_kaza at hotmail.com
To: len at ovenwerks.net
CC: linux-audio-user at lists.linuxaudio.org
Subject: RE: [LAU] Ubuntu Studio 14.04 Pulse removal question
Date: Tue, 12 Aug 2014 06:40:59 +0000






> Date: Mon, 11 Aug 2014 23:20:05 -0700
> From: len at ovenwerks.net
> To: dj_kaza at hotmail.com
> CC: linux-audio-user at lists.linuxaudio.org
> Subject: RE: [LAU] Ubuntu Studio 14.04 Pulse removal question
> 
> On Tue, 12 Aug 2014, Kaza Kore wrote:
> 
> > As far as I know and can tell -k stands for Kill. if you run it twice it
> > informs you PA is not running!
> 
> Yes but pulse is configured respawn by default, it does take some time to 
> restart, but does so as soon as you kill it.

Doesn't seem to be respawning now, definitely was still loading at startup  before I added the pulseaudio -k command to jack options but my laptop has been running all night, so up for maybe 16 hours since last reboot and this is my output from trying to kill pulseaudio in terminal.

$ pulseaudio -k
E: [pulseaudio] main.c: Failed to kill daemon: No such process

Clearly not respawned itself in all that time. But the fact it will do if any program requests it is quite a good hint for me, and hopefully I will see a change if it pops up again when I try and use something.

> > 
> > As I stated I had unchecked the all Pulse related options in the Autostart
> > settings and it still starts on reboot!
> 
> That could be... all it takes is something asking via D-bus if pulse is 
> there... and then it is.
> 
> My personal solution has been to unload module-jackdbus-detect when I do 
> not want pulse connected to jack. pulse takes almost no cpu that way. The 
> command to unload the module is:
> 
> pactl unload-module module-jackdbus-detect

Once I'm totally sure nothing is trying to access pulse any more I might add this line. Is there any difference between adding it to rc.local and adding it as an Autostart option, like I did with qjackctl?
> 
> The package pulseaudio-module-jack can be removed to make this more 
> permanent, but if you will not use pulse maybe just:
> 
> sudo chmod -x /usr/bin/pulseaudio

And then this when even more confident ;)
> 
> There may be depends problems if you remove the pulseaudio package 
> itself... for example you browser may be removed as well. Debian depends 
> are there to protect dummys.
> 

Yeah I worried and thought I had heard there could be problems with removing. So I aim to fully ensure it's never running and I can be happy with that :)

> --
> Len Ovens
> www.ovenwerks.net
> 

Dale.
 		 	   		   		 	   		  
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From dj_kaza at hotmail.com  Tue Aug 12 07:23:03 2014
From: dj_kaza at hotmail.com (Kaza Kore)
Date: Tue, 12 Aug 2014 07:23:03 +0000
Subject: [LAU] Ubuntu Studio 14.04 Pulse removal question
In-Reply-To: <DUB113-W78A414DCC37A4E100583698FEA0@phx.gbl>
References: <DUB113-W258174F3D395A5066FC6A28FEA0@phx.gbl>,
 <alpine.DEB.2.02.1408112232070.23832@scott.cbbs.org>,
 <DUB113-W9ED54F040152113EA17F98FEA0@phx.gbl>,
 <alpine.DEB.2.02.1408112305150.23832@scott.cbbs.org>,
 <DUB113-W135524935A2907F33BD9F5A8FEA0@phx.gbl>,
 <DUB113-W78A414DCC37A4E100583698FEA0@phx.gbl>
Message-ID: <DUB113-W4016D33B75BB4D65C6958B8FEA0@phx.gbl>

Ok so even opening QasMixer will currently get PA to spawn when it wasn't previously running. I see Pulse is listed under Mixer Devices, plus it also appears very likely that is the mixer currently set as my Default from looking at the layout.

Is there some configuration to Alsa I should do to prevent this when not using PA?

Dale.

From: dj_kaza at hotmail.com
To: len at ovenwerks.net
CC: linux-audio-user at lists.linuxaudio.org
Subject: RE: [LAU] Ubuntu Studio 14.04 Pulse removal question
Date: Tue, 12 Aug 2014 06:59:24 +0000




*(PA) definitely was still loading at startup*

I just had a thought as to the cause of this. I think I might have still had my Volume widget in my Panel showing the control from PA at that point and since changed it to display amixer's. From what you say this likely would have caused PA to start, even though I had disabled it in Autostart.

Hope you forgive me for not confirming this was the cause right now :)

Dale.

From: dj_kaza at hotmail.com
To: len at ovenwerks.net
CC: linux-audio-user at lists.linuxaudio.org
Subject: RE: [LAU] Ubuntu Studio 14.04 Pulse removal question
Date: Tue, 12 Aug 2014 06:40:59 +0000






> Date: Mon, 11 Aug 2014 23:20:05 -0700
> From: len at ovenwerks.net
> To: dj_kaza at hotmail.com
> CC: linux-audio-user at lists.linuxaudio.org
> Subject: RE: [LAU] Ubuntu Studio 14.04 Pulse removal question
> 
> On Tue, 12 Aug 2014, Kaza Kore wrote:
> 
> > As far as I know and can tell -k stands for Kill. if you run it twice it
> > informs you PA is not running!
> 
> Yes but pulse is configured respawn by default, it does take some time to 
> restart, but does so as soon as you kill it.

Doesn't seem to be respawning now, definitely was still loading at startup  before I added the pulseaudio -k command to jack options but my laptop has been running all night, so up for maybe 16 hours since last reboot and this is my output from trying to kill pulseaudio in terminal.

$ pulseaudio -k
E: [pulseaudio] main.c: Failed to kill daemon: No such process

Clearly not respawned itself in all that time. But the fact it will do if any program requests it is quite a good hint for me, and hopefully I will see a change if it pops up again when I try and use something.

> > 
> > As I stated I had unchecked the all Pulse related options in the Autostart
> > settings and it still starts on reboot!
> 
> That could be... all it takes is something asking via D-bus if pulse is 
> there... and then it is.
> 
> My personal solution has been to unload module-jackdbus-detect when I do 
> not want pulse connected to jack. pulse takes almost no cpu that way. The 
> command to unload the module is:
> 
> pactl unload-module module-jackdbus-detect

Once I'm totally sure nothing is trying to access pulse any more I might add this line. Is there any difference between adding it to rc.local and adding it as an Autostart option, like I did with qjackctl?
> 
> The package pulseaudio-module-jack can be removed to make this more 
> permanent, but if you will not use pulse maybe just:
> 
> sudo chmod -x /usr/bin/pulseaudio

And then this when even more confident ;)
> 
> There may be depends problems if you remove the pulseaudio package 
> itself... for example you browser may be removed as well. Debian depends 
> are there to protect dummys.
> 

Yeah I worried and thought I had heard there could be problems with removing. So I aim to fully ensure it's never running and I can be happy with that :)

> --
> Len Ovens
> www.ovenwerks.net
> 

Dale.
 		 	   		   		 	   		   		 	   		  
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From fons at linuxaudio.org  Tue Aug 12 12:26:04 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Tue, 12 Aug 2014 12:26:04 +0000
Subject: [LAU] [Bulk] Re: Jack transport - was - Ardour/Muse Jack tempo
 lock
In-Reply-To: <3967821.5Hi4RIW6Gm@col-desktop>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <53E7A7BC.90008@gareus.org> <20140811100559.GA957@linuxaudio.org>
 <3967821.5Hi4RIW6Gm@col-desktop>
Message-ID: <20140812122604.GA13127@linuxaudio.org>

On Mon, Aug 11, 2014 at 08:01:40PM -0400, Tim E. Real wrote:
 
> Mm, do you know about the 'slow-sync' Jack callback function?

Of course I do. Do you really think I'd comment on anything without
knowing it inside out ? That very 'feature' is the problem.

> Or do you propose something slightly different?
> As in, able to report readiness anytime all the time even in stop mode?

Yes. Together with the requirement that any app using the transport API
should be ready to start running (playback and/or recording previously
armed tracks) instantly a reasonable time (at most a few seconds) after it
has been repositioned or reconfigured. Or at least have a state - PAUSE -
in which this is the case [1].

In fact any app used for audio recording or playback in a production
context should have that property even when used separately, without
Jack transport. All professional audio or video equipment I've ever
used could do that, even in the tape days [2]. It's a rather basic
requirement, and not at all difficult to implement. 

In other words, the not-ready state should be transient, it will
revert to ready ASAP, and it serves mainly as a warning to the user.
As soon as everybody reports ready, you know you can start instantly,
which is very reassuring in a live context.

> The 'slow-sync' Jack callback function is a great feature for
>  holding up the transport until all the apps report they are 
>  'ready to roll'.

To be able to start 'on cue' they must be ready when the start
command arrives and NOT delay it. Starting on cue is a very
common thing in broadcasting, theatre sound, live shows, audio
drama production,... to name just the few cases I've encountered
in my work. 

> But... not ready for what? The app cannot know what the
> next transport intention is until the actual command
> comes down the line.

Ready to do what would be required 'on cue', i.e. playback.
Recording on pre-armed tracks is no more difficult, so I'd
include that as well. All the rest (repositioning, preparing
to record) can take any time you want, nobody expects that
to be 'instant'.

I don't think that a global RECORD command or state should
be part of a shared transport control, it's local thing for
each of the controlled items. Those that have RECORD enabled
will record when started. Punch in/out can be controlled
locally, and today it will be preprogrammed anyway. [3]

Caio,


[1] Video tape recorders used to have a separate PAUSE state
since leaving the tape static against the rotating drum for 
too long would damage it. Audio tape recorders usually did
not have a separate PAUSE state, they could start instantly
from STOP.

[2] I know of one professional tape recorder that would spin down
the capstan when idle for more than a few minutes. But even that
one would warn you a few seconds before doing that, and hitting
STOP would override the spindown and keep the transport ready to
start instantly.

[3] In the 24-track tape days punch in/out could be done either
by pre-arming the required tracks and then using the global RECORD
at the right time(s), or by starting with RECORD enabled and then 
using the per-track controls to punch in and out. The latter would
be difficult in a software DAW, unless the tracks are prepared in
some way previously (which happens implicitly when using the first 
method). But it's not really required, the first method works OK,
even more so if things can be pre-programmed as in Ardour.

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From david at kenpro.com.au  Tue Aug 12 13:53:30 2014
From: david at kenpro.com.au (David)
Date: Tue, 12 Aug 2014 23:53:30 +1000
Subject: [LAU] Komplete Audio 6 and Jack
Message-ID: <53EA1C5A.8000702@kenpro.com.au>

I'm completely new to this, so be kind to me please.

Ubuntu studio 14.04
Komplete Audio 6
Rode condensor mics.

I've found this link which says it should "just work",

http://lists.linuxaudio.org/pipermail/linux-audio-user/2013-October/094733.html

but I can't seem to get response from the  inputs (microphones). I did 
manage to make Yoshimi talk to the KA6 via Jack but that isn't what I need.

Ultimately I want to be able to simply record stereo microphones into 
Ardour or Audacity. Preferably Audacity because I'm familiar with it.

I suspect that I'm not configuring Jack connections correctly but don't 
know enough about it to see what the problem is.

I've set the interface in qjackctl settings to the KA6.

The KA6 never shows up in the volume control panel in the way that a USB 
headset would. The KA6 does show up in both input and output sides of 
the qjackctl ALSA tab.


-- 
David McQuire
0418 310312


From willgodfrey at musically.me.uk  Tue Aug 12 14:58:48 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Tue, 12 Aug 2014 15:58:48 +0100
Subject: [LAU] Komplete Audio 6 and Jack
In-Reply-To: <53EA1C5A.8000702@kenpro.com.au>
References: <53EA1C5A.8000702@kenpro.com.au>
Message-ID: <20140812155848.033bbbe5@debian>

On Tue, 12 Aug 2014 23:53:30 +1000
David <david at kenpro.com.au> wrote:

> I'm completely new to this, so be kind to me please.
> 
> Ubuntu studio 14.04
> Komplete Audio 6
> Rode condensor mics.
> 
> I've found this link which says it should "just work",
> 
> http://lists.linuxaudio.org/pipermail/linux-audio-user/2013-October/094733.html
> 
> but I can't seem to get response from the  inputs (microphones). I did 
> manage to make Yoshimi talk to the KA6 via Jack but that isn't what I need.
> 
> Ultimately I want to be able to simply record stereo microphones into 
> Ardour or Audacity. Preferably Audacity because I'm familiar with it.
> 
> I suspect that I'm not configuring Jack connections correctly but don't 
> know enough about it to see what the problem is.
> 
> I've set the interface in qjackctl settings to the KA6.
> 
> The KA6 never shows up in the volume control panel in the way that a USB 
> headset would. The KA6 does show up in both input and output sides of 
> the qjackctl ALSA tab.
> 
> 
At the risk of stating the obvious, in qjackctl as well as the interface, have
you set both the Input Device and the Output Device.

Works fine for me on debian.

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From edogawa at aon.at  Tue Aug 12 15:19:11 2014
From: edogawa at aon.at (Edgar Aichinger)
Date: Tue, 12 Aug 2014 17:19:11 +0200
Subject: [LAU] Komplete Audio 6 and Jack
In-Reply-To: <20140812155848.033bbbe5@debian>
References: <53EA1C5A.8000702@kenpro.com.au> <20140812155848.033bbbe5@debian>
Message-ID: <2300445.0PWmiFZ6XB@edhp>

Am Dienstag, 12. August 2014, 15:58:48 schrieb Will Godfrey:
> On Tue, 12 Aug 2014 23:53:30 +1000
> David <david at kenpro.com.au> wrote:
> 
> > I'm completely new to this, so be kind to me please.
> > 
> > Ubuntu studio 14.04
> > Komplete Audio 6
> > Rode condensor mics.
> > 
> > I've found this link which says it should "just work",
> > 
> > http://lists.linuxaudio.org/pipermail/linux-audio-user/2013-October/094733.html
> > 
> > but I can't seem to get response from the  inputs (microphones). I did 
> > manage to make Yoshimi talk to the KA6 via Jack but that isn't what I need.
> > 
> > Ultimately I want to be able to simply record stereo microphones into 
> > Ardour or Audacity. Preferably Audacity because I'm familiar with it.
> > 
> > I suspect that I'm not configuring Jack connections correctly but don't 
> > know enough about it to see what the problem is.
> > 
> > I've set the interface in qjackctl settings to the KA6.
> > 
> > The KA6 never shows up in the volume control panel in the way that a USB 
> > headset would. The KA6 does show up in both input and output sides of 
> > the qjackctl ALSA tab.
> > 
> > 
> At the risk of stating the obvious, in qjackctl as well as the interface, have
> you set both the Input Device and the Output Device.
> 
> Works fine for me on debian.

AFAIK it's not recommended to use these except for special cases (don't even know which).

Better choose the right device using the Interface: box and the dropdown next to it, leave In/Output device at default and make sure duplex is selected...

2c, Edgar


From len at ovenwerks.net  Tue Aug 12 20:05:27 2014
From: len at ovenwerks.net (Len Ovens)
Date: Tue, 12 Aug 2014 13:05:27 -0700 (PDT)
Subject: [LAU] Komplete Audio 6 and Jack
In-Reply-To: <53EA1C5A.8000702@kenpro.com.au>
References: <53EA1C5A.8000702@kenpro.com.au>
Message-ID: <alpine.DEB.2.02.1408121218580.23832@scott.cbbs.org>

On Tue, 12 Aug 2014, David wrote:

> I'm completely new to this, so be kind to me please.
>
> Ubuntu studio 14.04
> Komplete Audio 6
> Rode condensor mics.

I am not sure so I may be asking obvious questions. Does the KA6 have 
phantom power turned on (48v switch on the back)? Does the KA6 show levels 
from the mic(s)? Are there level controls on the KA6 (yup)? Are they up?

Next: aplay -l and arecord -l says what? Ah someone has done that already:

The lines that matter:
card 1: K6 [Komplete Audio 6], device 0: USB Audio [USB Audio]
   Subdevices: 0/1
   Subdevice #0: subdevice #0
card 1: K6 [Komplete Audio 6], device 0: USB Audio [USB Audio]
   Subdevices: 1/1
   Subdevice #0: subdevice #0


> I've found this link which says it should "just work",
>
> http://lists.linuxaudio.org/pipermail/linux-audio-user/2013-October/094733.html

Well sort of. It does say "set the connections in qjackctl". I am not sure 
what that means.
In qjackctl in setup-Settings There is an interface box. The intuitive 
thing is to click the V beside and pick one... don't do this. Instead 
click on the > box and choose hw:K6. This should give 6 in and 6 out. If 
not... you will want to see if using Input Device and Output Device works 
better. I am not familiar with the K6 so I can't say. It should not be too 
hard to connect the input directly to the output to see if there is sound, 
but I prefer to start meterbridge and connect there so I know there is no 
hw monitor stuff giving me false info. Also, in a terminal, run alsamixer 
and make sure the capture levels are up. Or switches need to be set. There 
seems to be some capture switches.

> but I can't seem to get response from the  inputs (microphones). I did manage 
> to make Yoshimi talk to the KA6 via Jack but that isn't what I need.

So that is outputs.

> Ultimately I want to be able to simply record stereo microphones into Ardour 
> or Audacity. Preferably Audacity because I'm familiar with it.

Sounds like that should work.

> I suspect that I'm not configuring Jack connections correctly but don't know 
> enough about it to see what the problem is.
>
> I've set the interface in qjackctl settings to the KA6.
>
> The KA6 never shows up in the volume control panel in the way that a USB

One wonders which volume control pannel that is, and where within that.

> headset would. The KA6 does show up in both input and output sides of the 
> qjackctl ALSA tab.

The ALSA tab is MIDI i/o (yes there are two of them for two kinds of MIDI) 
You want to see i/o in the Audio tab. There should be system on both 
sides. and when expanded there should be 6 on each side.

Just sort of a splatter of ideas....

--
Len Ovens
www.ovenwerks.net


From david at kenpro.com.au  Wed Aug 13 04:08:46 2014
From: david at kenpro.com.au (David)
Date: Wed, 13 Aug 2014 14:08:46 +1000
Subject: [LAU] Komplete Audio 6 and Jack (solved with alsamixer switches)
In-Reply-To: <alpine.DEB.2.02.1408121218580.23832@scott.cbbs.org>
References: <53EA1C5A.8000702@kenpro.com.au>
 <alpine.DEB.2.02.1408121218580.23832@scott.cbbs.org>
Message-ID: <53EAE4CE.7040308@kenpro.com.au>

Thanks everyone. I've typed out everything I went through in case 
someone else stumbles on this with a similar problem

On 13/08/14 06:05, Len Ovens wrote:

> On Tue, 12 Aug 2014, David wrote:
>
>> I'm completely new to this, so be kind to me please.
>>
>> Ubuntu studio 14.04
>> Komplete Audio 6
>> Rode condensor mics.
>
> I am not sure so I may be asking obvious questions. 

Please ask obvious questions.. it's something obvious I am probably 
missing :(

> Does the KA6 have phantom power turned on (48v switch on the back)?

The Rode website says that this mic doesn't care either way about 
phantom power, but in any case I've tried it both on and off. Apparently 
some mics can be cooked with phantom power!

> Does the KA6 show levels from the mic(s)?

I get a "flashing light" level indicator on the KA6 when I test the mic.

> Are there level controls on the KA6 (yup)? Are they up?

both mic gain and master are set at 12 o'clock. (Edit: they don't seem 
to alter the record levels - odd?)

>
>
> Next: aplay -l and arecord -l says what? Ah someone has done that 
> already:
>
> The lines that matter:
> card 1: K6 [Komplete Audio 6], device 0: USB Audio [USB Audio]
>   Subdevices: 0/1
>   Subdevice #0: subdevice #0
> card 1: K6 [Komplete Audio 6], device 0: USB Audio [USB Audio]
>   Subdevices: 1/1
>   Subdevice #0: subdevice #0

That's exactly what I get (plus the built in sound card of course)

>
>
>> I've found this link which says it should "just work",
>>
>> http://lists.linuxaudio.org/pipermail/linux-audio-user/2013-October/094733.html 
>>
>
> Well sort of. It does say "set the connections in qjackctl". I am not 
> sure what that means.

neither do I. I've tried quite a few combinations with no luck so far.

> In qjackctl in setup-Settings There is an interface box. The intuitive 
> thing is to click the V beside and pick one... don't do this. Instead 
> click on the > box and choose hw:K6. This should give 6 in and 6 out.

In the Audio tab of QJackCtl Connections window, under "system" I get 6 
in and 6 out, so I guess that's it.

> If not... you will want to see if using Input Device and Output Device 
> works better. I am not familiar with the K6 so I can't say. It should 
> not be too hard to connect the input directly to the output to see if 
> there is sound, but I prefer to start meterbridge....

I had never heard of meterbridge. I did <$ jack_lsp> and it gave me 
"system:capture_1" etc etc, but when I started meterbridge I got no 
signal, even though the level light was flashing on the KA6 so there is 
still a problem.

> .... and connect there so I know there is no hw monitor stuff giving 
> me false info. Also, in a terminal, run alsamixer and make sure the 
> capture levels are up. Or switches need to be set. There seems to be 
> some capture switches.

Alsamixer proved interesting. I've never used it before. First I had to 
select the KA6 using <F6: Select Sound Card> and then I'm presented with 
an arcane screen. I messed around by tabbing through "monitor control 
front" "monitor control rear" and "monitor control 1" and typing <m> at 
each position. That changed the "MM" (mute?) in the box above each item 
to "00". This seemed to create some sort of magic because now I'm 
getting a vu-meter reading on <$ meterbridge system:capture_1>. I can't 
understand why altering the monitoring should make the input work, but 
apparently it does. Pressing <F4> in alsamixer tells me that "this 
device does not have any capture controls". Very odd.

This seems to be the answer to the problem.


>> Ultimately I want to be able to simply record stereo microphones into 
>> Ardour or Audacity. Preferably Audacity because I'm familiar with it.
>
> Sounds like that should work.
>
>>
>> The KA6 never shows up in the volume control panel ....
>
> One wonders which volume control pannel that is, and where within that.

I'm talking about the system settings volume control widget that lives 
on the taskbar.
After discovering alsamixer mute switches (thanks!), if I go to
volume control -> sound setting -> input device
the item labelled "Built-in Audio Analogue Stereo" is now responding - 
ie: the level meter moves when I test the mic. I can change the mic 
level there.

It's a bit obscure and doesn't refer to the KA6 at all - but hey, if it 
works I'm happy. Does this have something to do with 
pulseaudio-module-jack being installed??

>
>
> The ALSA tab is MIDI i/o (yes there are two of them for two kinds of 
> MIDI) You want to see i/o in the Audio tab. There should be system on 
> both sides. and when expanded there should be 6 on each side.

You are right.


>
> Just sort of a splatter of ideas....


exactly what I needed. Audacity now records sound from the KA6 with a 
mic plugged in. Next to try some actual recordings!

Where do I send the pizza?


>
> -- 
> Len Ovens
> www.ovenwerks.net
>

-- 
David McQuire
0418 310312


From termtech at rogers.com  Wed Aug 13 04:08:51 2014
From: termtech at rogers.com (Tim E. Real)
Date: Wed, 13 Aug 2014 00:08:51 -0400
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <20140812122604.GA13127@linuxaudio.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <3967821.5Hi4RIW6Gm@col-desktop> <20140812122604.GA13127@linuxaudio.org>
Message-ID: <3276416.mTp9TkTy7n@col-desktop>

On August 12, 2014 12:26:04 PM Fons Adriaensen wrote:
> On Mon, Aug 11, 2014 at 08:01:40PM -0400, Tim E. Real wrote:
> > Mm, do you know about the 'slow-sync' Jack callback function?
> 
> Of course I do. Do you really think I'd comment on anything without
> knowing it inside out ? 

Apologies, wasn't sure :-)

> That very 'feature' is the problem.

> > Or do you propose something slightly different?
> > As in, able to report readiness anytime all the time even in stop mode?
> 
> Yes. Together with the requirement that any app using the transport API
> should be ready to start running (playback and/or recording previously
> armed tracks) instantly a reasonable time (at most a few seconds) after it
> has been repositioned or reconfigured. Or at least have a state - PAUSE -
> in which this is the case [1].

Interesting... a pause state.

> 
> In fact any app used for audio recording or playback in a production
> context should have that property even when used separately, without
> Jack transport. All professional audio or video equipment I've ever
> used could do that, even in the tape days [2]. It's a rather basic
> requirement, and not at all difficult to implement.
> 
> In other words, the not-ready state should be transient, it will
> revert to ready ASAP, and it serves mainly as a warning to the user.
> As soon as everybody reports ready, you know you can start instantly,
> which is very reassuring in a live context.

OK I think I understand now:

With the current system, the user 'presses play button', but the button 
 then flashes informing the user it will be a few moments before we're 
 actually running, then it stops flashing and system runs.

With your proposal, the 'play' button flashes informing the user the 
 system is busy - BUT - at least when it stops flashing, he is /guaranteed/ 
 that the system will start immediately.

I would add: If play is flashing he shouldn't have to press it again when it 
 stops flashing, so let the system remember that he did in fact press play.

> > The 'slow-sync' Jack callback function is a great feature for
> > 
> >  holding up the transport until all the apps report they are
> >  'ready to roll'.
> 
> To be able to start 'on cue' they must be ready when the start
> command arrives and NOT delay it. Starting on cue is a very
> common thing in broadcasting, theatre sound, live shows, audio
> drama production,... to name just the few cases I've encountered
> in my work.

Agreed.

> > But... not ready for what? The app cannot know what the
> > next transport intention is until the actual command
> > comes down the line.
> 
> Ready to do what would be required 'on cue', i.e. playback.
> Recording on pre-armed tracks is no more difficult, so I'd
> include that as well. All the rest (repositioning, preparing
> to record) can take any time you want, nobody expects that
> to be 'instant'.
> 
> I don't think that a global RECORD command or state should
> be part of a shared transport control, it's local thing for
> each of the controlled items. Those that have RECORD enabled
> will record when started. 

Yeah. Traditionally play is a mechanical state (as in "forward ho!") and 
 record is not really a mechanical state but just a switch commanded 
 to turn on the record circuit and head etc.
That's why I tried to eek out some definitions yesterday but had trouble.

Still, to satisfy the OP request, there could be a global record ENABLE, 
 while still granting that record ARM is strictly a local thing.
To rephrase your last sentence could be:
Those that have local RECORD ARM enabled will record when started,
 provided global record ENABLE is on. 

> Punch in/out can be controlled
> locally, and today it will be preprogrammed anyway. [3]

Not sure about global punch in/out. Maybe.
But as with the record functions, each app would have to have a way of
 letting the user elect to participate in the global override(s) or not.
But, if the mechanism by which record functions could be added 
 could easily include other commands such as punch in/out, why not eh? 

> 
> Caio,
> 
> 

[Speaking my language here - Pro repair tech, 30+ years]:
> [1] Video tape recorders used to have a separate PAUSE state
> since leaving the tape static against the rotating drum for
> too long would damage it. Audio tape recorders usually did
> not have a separate PAUSE state, they could start instantly
> from STOP.
> 
> [2] I know of one professional tape recorder that would spin down
> the capstan when idle for more than a few minutes. But even that
> one would warn you a few seconds before doing that, and hitting
> STOP would override the spindown and keep the transport ready to
> start instantly.

How would these decks be told by software to come out of deep-pause mode?
To use my understanding of your proposal, paragraph above, the user's 
 (software) play button would flash indefinitely (meaning not ready)
 until he presses it, and it'll be a moment until the VCR can be ready 
 to play again.
>From the software POV, the play command would have to be sent and 
 then we wait for a signal from the deck that it's ready. 
Which is what the slow-sync callback does.
Seems both it and your proposal would be good to have?

Ah... But you will say: A system where one of the decks is in deep-pause 
 mode is not ready at all - it needs to be (manually) taken out or kept out 
 of that mode when anticipating that recording will take place soon, 
 to satisfy the instant run requirement. Right?

> 
> [3] In the 24-track tape days punch in/out could be done either
> by pre-arming the required tracks and then using the global RECORD
> at the right time(s), or by starting with RECORD enabled and then
> using the per-track controls to punch in and out. The latter would
> be difficult in a software DAW, unless the tracks are prepared in
> some way previously (which happens implicitly when using the first
> method). But it's not really required, the first method works OK,
> even more so if things can be pre-programmed as in Ardour.

MusE has pre-programmed punch in/out.
Can't recall if can be activated remotely.

Pleasure rappin' with ya.
Tim.

From robin at gareus.org  Wed Aug 13 04:48:12 2014
From: robin at gareus.org (Robin Gareus)
Date: Wed, 13 Aug 2014 06:48:12 +0200
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <3276416.mTp9TkTy7n@col-desktop>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <3967821.5Hi4RIW6Gm@col-desktop> <20140812122604.GA13127@linuxaudio.org>
 <3276416.mTp9TkTy7n@col-desktop>
Message-ID: <53EAEE0C.9070506@gareus.org>

On 08/13/2014 06:08 AM, Tim E. Real wrote:
[..]
> OK I think I understand now:
> 
> With the current system, the user 'presses play button', but the button 
>  then flashes informing the user it will be a few moments before we're 
>  actually running, then it stops flashing and system runs.
> 
> With your proposal, the 'play' button flashes informing the user the 
>  system is busy - BUT - at least when it stops flashing, he is /guaranteed/ 
>  that the system will start immediately.
> 

While the user-experience may improve, technically that mechanism only
makes sense if it is automated (eg a MMC slave or similar).

A user pressing a button and that event jumping all the fences and hoops
until it finally reaches jack has a random delay to begin with.

With a modern SSDs the seek and pre-buffer time until the transport is
ready to roll is likely shorter than the time it takes for a mouse-click
to be processed by the application on most systems.

2c,
robin

From len at ovenwerks.net  Wed Aug 13 05:46:37 2014
From: len at ovenwerks.net (Len Ovens)
Date: Tue, 12 Aug 2014 22:46:37 -0700 (PDT)
Subject: [LAU] Komplete Audio 6 and Jack (solved with alsamixer switches)
In-Reply-To: <53EAE4CE.7040308@kenpro.com.au>
References: <53EA1C5A.8000702@kenpro.com.au>
 <alpine.DEB.2.02.1408121218580.23832@scott.cbbs.org>
 <53EAE4CE.7040308@kenpro.com.au>
Message-ID: <alpine.DEB.2.10.1408122146280.2503@scott.cbbs.org>

On Wed, 13 Aug 2014, David wrote:

> Thanks everyone. I've typed out everything I went through in case someone 
> else stumbles on this with a similar problem

No problem. I learn too.
>> On Tue, 12 Aug 2014, David wrote:
>>> Rode condensor mics.
>> Does the KA6 have phantom power turned on (48v switch on the back)?
>
> The Rode website says that this mic doesn't care either way about phantom 
> power, but in any case I've tried it both on and off. Apparently some mics 
> can be cooked with phantom power!

Generally a condensor mic needs power, either batteries, phantom or a 
power supply of it's own. Ribbon mics are known to be sensitive if the 
cable happens to be miswired.

> In the Audio tab of QJackCtl Connections window, under "system" I get 6 in 
> and 6 out, so I guess that's it.

Right, two mics(1,2), two lines(3,4) and two from s/pdif(5,6). The outputs 
are similar. Perfect.

> I had never heard of meterbridge. I did <$ jack_lsp> and it gave me 
> "system:capture_1" etc etc, but when I started meterbridge I got no signal,

Ya meter bridge is kind of funny, the *.desktop file starts it with two 
meters that are supposed to be connected to alsa_pcm:playback_1 and 2 so 
on my system that means they are not connected to anything. But it is a 
handy signal indicator for trouble shooting when not using it for 
metering.

> Alsamixer proved interesting. I've never used it before. First I had to

It is something to remember as it deals only with the sound card where 
many other mixers deal with psuedo/soft controls.

> select the KA6 using <F6: Select Sound Card> and then I'm presented with an 
> arcane screen. I messed around by tabbing through "monitor control front" 
> "monitor control rear" and "monitor control 1" and typing <m> at each 
> position. That changed the "MM" (mute?) in the box above each item to "00".

Yes M is mute. I have gotten sound out of things using alsamix when 
nothing else works. It shows up interdependancies rather well too. (where 
moving one control affects another - I have one where muting one channel 
mutes three and unmuting only unmutes one)

> This seemed to create some sort of magic because now I'm getting a vu-meter 
> reading on <$ meterbridge system:capture_1>. I can't understand why altering 
> the monitoring should make the input work, but apparently it does. Pressing

Probably routing would be a better word than monitoring.

> <F4> in alsamixer tells me that "this device does not have any capture 
> controls". Very odd.

Quite common actually. You control the level with the physical controls on 
the box... although it seems with this one the volumes that look like 
output may affect the input. The manual was not that clear to me.

>> One wonders which volume control pannel that is, and where within that.
>
> I'm talking about the system settings volume control widget that lives on the 
> taskbar.

Ah, pavucontrol then, controls PA. Fine for desktop stuff like browsing 
though.

> After discovering alsamixer mute switches (thanks!), if I go to
> volume control -> sound setting -> input device
> the item labelled "Built-in Audio Analogue Stereo" is now responding - ie: 
> the level meter moves when I test the mic. I can change the mic level there.

I think it may be a soft level if alsamixer doesn't show it.

> It's a bit obscure and doesn't refer to the KA6 at all - but hey, if it works 
> I'm happy. Does this have something to do with pulseaudio-module-jack being 
> installed??

No, just pulse itself.

> exactly what I needed. Audacity now records sound from the KA6 with a mic 
> plugged in. Next to try some actual recordings!
>
> Where do I send the pizza?

It is just nice to see someone else enjoy linux audio and not go away 
frustrated.


--
Len Ovens
www.ovenwerks.net


From nescivi at gmail.com  Wed Aug 13 09:51:48 2014
From: nescivi at gmail.com (nescivi)
Date: Wed, 13 Aug 2014 11:51:48 +0200
Subject: [LAU] Creative Music Coding lab on August 26, STEIM, Amsterdam
Message-ID: <53EB3534.2000306@gmail.com>

Hi all,

On August 26 we welcome again all creative music coders at STEIM for an
evening of exchanging current work, problems and solutions - and music
together.

More information:
http://steim.org/event/creative-music-coding-lab-13/
Entrance is free.

And let us know if you plan to join (just to get an idea of how many
seats, and how much coffee and tea we should prepare)!

sincerely,
Marije

From pshirkey at boosthardware.com  Wed Aug 13 10:23:17 2014
From: pshirkey at boosthardware.com (Patrick Shirkey)
Date: Wed, 13 Aug 2014 20:23:17 +1000 (EST)
Subject: [LAU] JACKDUB
Message-ID: <59703.86.107.254.57.1407925397.squirrel@boosthardware.com>

Hi,

Some of you might be interested in a new project I have been working on:

http://jackdub.channellinux.com/

It is a fully realtime automated music playback system built with various
FLOSS tools. The system runs entirely in the cloud and doesn't even have a
sound card.

It's a work in progress so YMMV...



Cheers

--
Patrick Shirkey
Boost Hardware Ltd

From fons at linuxaudio.org  Wed Aug 13 10:59:44 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Wed, 13 Aug 2014 10:59:44 +0000
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <3276416.mTp9TkTy7n@col-desktop>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <3967821.5Hi4RIW6Gm@col-desktop>
 <20140812122604.GA13127@linuxaudio.org>
 <3276416.mTp9TkTy7n@col-desktop>
Message-ID: <20140813105944.GA18888@linuxaudio.org>

On Wed, Aug 13, 2014 at 12:08:51AM -0400, Tim E. Real wrote:
 
> Interesting... a pause state.

It's typical for video recorders (and DAT). Software apps wouldn't need one. 
 
> With the current system, the user 'presses play button', but the button 
>  then flashes informing the user it will be a few moments before we're 
>  actually running, then it stops flashing and system runs.
> 
> With your proposal, the 'play' button flashes informing the user the 
>  system is busy - BUT - at least when it stops flashing, he is /guaranteed/ 
>  that the system will start immediately.

How 'not ready' indicated is a side issue. It could be STOP flashing
when stopped and PLAY flashing if you try to start, or some separate
indication. 

> I would add: If play is flashing he shouldn't have to press it again when it 
>  stops flashing, so let the system remember that he did in fact press play.

The 'not ready' feedback exists to avoid the user trying to start
when it will fail, so this shouldn't happen. But when it does, yes
I'd agree that the start command should be remembered (as in fact
it is now).

Some of the tape machines I used could locate to some stored
position, and while they were doing this you could hit PLAY.
If the fast wind was forward they wouldn't even stop, just slow
down when approaching the locate position to arrive there at the
the correct speed and then go into PLAY. That's nice transport
design.

In software such things are almost trivial to implement, and they
make a difference. Ardour fails here when controlled via OSC. If
you send a 'locate' and then 'play' it will in many cases ignore
the 'play' command. You need to insert a delay between the two
to makes this sequence reliable. This hit me when I was writing
the automated systems used at the CdS, which led me to write a
dedicated player app and think a bit about reliable remote control
protocols.

> Still, to satisfy the OP request, there could be a global record ENABLE, 
>  while still granting that record ARM is strictly a local thing.
> To rephrase your last sentence could be:
> Those that have local RECORD ARM enabled will record when started,
>  provided global record ENABLE is on.

That assumes that all apps have per-track record enables. A simple
stereo recorder/player won't have them.

Anyway, to set up recording you'd have to access the local user
interface of that app, so having to enable record there is no big
deal. To me it also feels safer.
 
> > Punch in/out can be controlled
> > locally, and today it will be preprogrammed anyway. [3]
> 
> Not sure about global punch in/out. Maybe.

It's never a global thing. One app would do a punch in/out while
the others are just playing. And if not you're doing something
quite complicated, and this will need to be set up and checked 
locally on each app anyway.

> How would these decks be told by software to come out of deep-pause mode?

It would be local, or just by sending STOP. And most probably, when
put into remote control mode any 'deep sleep' mode would be disabled.
Anyway for SW apps such a state should not exist at all.

> From the software POV, the play command would have to be sent and 
> then we wait for a signal from the deck that it's ready. 
> Which is what the slow-sync callback does.
> Seems both it and your proposal would be good to have?

They way I've implemented this in some players uses four states:
all combinations of [STOP, RUN] and [READY, NOTREADY]. A state 
that includes NOTREADY is always transient, it will revert to the
corresponding READY one as soon as possible.

The combination RUN, NOTREADY is the one that Jack's 'slow sync'
implements. But it doesn't have STOP, NOTREADY. So that is what
I'd add, together with the requirement that NOTREADY must be
transient, i.e. an app is allowed to be not ready but must always
try to get ready ASAP, and not wait for e.g. a START command to
do so. Being ready of course includes that the audio path is or
can be set up, so things like only creating Jack ports when
started (Audacity) are a definite no-go.

> Pleasure rappin' with ya.
Same here :-)

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From fons at linuxaudio.org  Wed Aug 13 11:26:47 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Wed, 13 Aug 2014 11:26:47 +0000
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <53EAEE0C.9070506@gareus.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <3967821.5Hi4RIW6Gm@col-desktop>
 <20140812122604.GA13127@linuxaudio.org>
 <3276416.mTp9TkTy7n@col-desktop> <53EAEE0C.9070506@gareus.org>
Message-ID: <20140813112647.GB18888@linuxaudio.org>

On Wed, Aug 13, 2014 at 06:48:12AM +0200, Robin Gareus wrote:

> While the user-experience may improve, technically that mechanism only
> makes sense if it is automated (eg a MMC slave or similar).
> 
> A user pressing a button and that event jumping all the fences and hoops
> until it finally reaches jack has a random delay to begin with.

If we can use MIDI and stil play in sync with other musicians
then surely a START command can be propagated to wherever it
needs to arrive with acceptable delay. 

Pro tape decks could start in a few tens of ms at most. That
was perfectly acceptable. And surely we can do better than
that.


> With a modern SSDs the seek and pre-buffer time until the transport is
> ready to roll is likely shorter than the time it takes for a mouse-click
> to be processed by the application on most systems.

Only pre-filling play buffers when a START command arrives is against
all rules of sane RT design. Even when your disk systems are very fast.
And it's easy enough to do it right. 

And yes, some bloated GUI systems can be sluggish but if you want 
your app to be responsive (e.g. for visual editing) you'll have to
do something about that anyway. Nor should it be assumed that a
START command is always given by a mouse click.

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From ralf.mardorf at rocketmail.com  Wed Aug 13 12:20:19 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 13 Aug 2014 14:20:19 +0200
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <20140813105944.GA18888@linuxaudio.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <3967821.5Hi4RIW6Gm@col-desktop> <20140812122604.GA13127@linuxaudio.org>
 <3276416.mTp9TkTy7n@col-desktop> <20140813105944.GA18888@linuxaudio.org>
Message-ID: <1407932419.792.28.camel@rocketmail.com>

On Wed, 2014-08-13 at 10:59 +0000, Fons Adriaensen wrote:
> On Wed, Aug 13, 2014 at 12:08:51AM -0400, Tim E. Real wrote:
>  
> > Interesting... a pause state.
> 
> It's typical for video recorders (and DAT). Software apps wouldn't need one.

A crappy implementation, at least for cheap video and DAT. From pause to
record there usually (perhaps always) is a delay and after a while pause
automatically switches to stop, or at least should do it or do something
else to protect the material. To get everything in perfect sync _and_
ready for record/punch in, SMPTE sync usually is done by starting a few
seconds before the punch in. To get members of a band in sync is done in
a similar way, somebody e.g. knocks with drumsticks, IOW for recording
we humans also don't start with the recording directly after switching
from pause to record play, we listen for at least one bar (at 120 BPM
it's 2 seconds), before we start playing the instrument we want to
record.


From ralf.mardorf at rocketmail.com  Wed Aug 13 12:28:52 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 13 Aug 2014 14:28:52 +0200
Subject: [LAU] [Bulk] Re: Jack transport - was - Ardour/Muse Jack tempo
 lock
In-Reply-To: <20140813112647.GB18888@linuxaudio.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <3967821.5Hi4RIW6Gm@col-desktop> <20140812122604.GA13127@linuxaudio.org>
 <3276416.mTp9TkTy7n@col-desktop> <53EAEE0C.9070506@gareus.org>
 <20140813112647.GB18888@linuxaudio.org>
Message-ID: <1407932932.792.31.camel@rocketmail.com>

On Wed, 2014-08-13 at 11:26 +0000, Fons Adriaensen wrote:
> Nor should it be assumed that a START command is always given by a
> mouse click.

Indeed, there are several ways to do it, one way is to use MIDI gear and
MIDI gear is part of the production environment that other than a GUI,
should work in real-time. I guess a guitarist seldom does prefer a mouse
over a foot switch to start a recording, perhaps the guitarist's from
outer space with 3 arms like the mouse click.


From ralf.mardorf at rocketmail.com  Wed Aug 13 14:17:28 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 13 Aug 2014 16:17:28 +0200
Subject: [LAU] Header missing to build Zita AT 1
Message-ID: <1407939448.3069.4.camel@rocketmail.com>

Hi,

building
http://kokkinizita.linuxaudio.org/linuxaudio/downloads/zita-rev1-0.2.1.tar.bz2 did work, but building
http://kokkinizita.linuxaudio.org/linuxaudio/downloads/zita-at1-0.2.3.tar.bz2
failed.

[rocketmouse at archlinux source]$ make
g++  -O2 -ffast-math -Wall -MMD -MP -DVERSION=\"0.2.3\" -DSHARED=\"/usr/local/share/zita-at1\" -march=native -I/usr/X11R6/include `freetype-config --cflags`  -c -o zita-at1.o zita-at1.cc
In file included from jclient.h:28:0,
                 from zita-at1.cc:29:
retuner.h:27:28: fatal error: zita-resampler.h: No such file or
directory
 #include <zita-resampler.h>
                            ^
compilation terminated.
<builtin>: recipe for target 'zita-at1.o' failed
make: *** [zita-at1.o] Error 1

Regarding to the info given by the INSTALL text only clthreads and
clxclient are needed dependencies.

Even the headers provided by zita-resampler are missing zita-resampler.h

[rocketmouse at archlinux zita-at1-0.2.3]$ pacman -Ql zita-resampler | grep resampler.h
zita-resampler /usr/include/zita-resampler/resampler.h
zita-resampler /usr/include/zita-resampler/vresampler.h
zita-resampler /usr/share/doc/zita-resampler/resampler.html

The RME HDSPe AIO still doesn't provide all ADAT IOs as jack ports, so I
can't use a 19" reverb, that's why I will give Zita Rev 1 a shot, IOW I
already got what I need and since I'm in the middle of a home studio
production and I don't need AT 1, I won't trial by error how to build
AT 1, but it would be nice to test it.

How can I get rid of this fatal error?

Regards,
Ralf


From fons at linuxaudio.org  Wed Aug 13 14:45:14 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Wed, 13 Aug 2014 14:45:14 +0000
Subject: [LAU] Header missing to build Zita AT 1
In-Reply-To: <1407939448.3069.4.camel@rocketmail.com>
References: <1407939448.3069.4.camel@rocketmail.com>
Message-ID: <20140813144514.GF18888@linuxaudio.org>

On Wed, Aug 13, 2014 at 04:17:28PM +0200, Ralf Mardorf wrote:

>  #include <zita-resampler.h>

You need 

#include <zita-resampler/resampler.h>

instead. Package will be updated ASAP.  

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From ralf.mardorf at rocketmail.com  Wed Aug 13 16:26:04 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 13 Aug 2014 18:26:04 +0200
Subject: [LAU] Header missing to build Zita AT 1
In-Reply-To: <20140813144514.GF18888@linuxaudio.org>
References: <1407939448.3069.4.camel@rocketmail.com>
 <20140813144514.GF18888@linuxaudio.org>
Message-ID: <1407947164.4059.1.camel@rocketmail.com>

On Wed, 2014-08-13 at 14:45 +0000, Fons Adriaensen wrote:
> #include <zita-resampler/resampler.h>

After editing retuner.h it build without an error. Thank you :).

In QjackCtl there are only audio ports for AT1, there are no MIDI ports.
$ zita-at1 -h and the HTML doc don't inform how to connect MIDI.

Regards,
Ralf


From lmemsm at gmail.com  Wed Aug 13 16:27:10 2014
From: lmemsm at gmail.com (LM)
Date: Wed, 13 Aug 2014 12:27:10 -0400
Subject: [LAU] sound fonts/patch files
Message-ID: <CAFipMOE0nP2gpzm0Hz8PqGMVdP7Q0gXkofRtq8_0vvwnKrbd=g@mail.gmail.com>

I'm trying to find some sound fonts/patch file sets to use with
programs like Timidity and to figure out what licenses they're
available under.  Of course, there's the freepats collection (
http://freepats.zenvoid.org/ ).  freepats-20060219.tar.bz2 has GNU
GPLv2 license.  Am also looking for other collections with Free
licenses.  I ran across timidity_tiny at
http://edge.beemulated.net/utils/index.php which looks really
interesting because it's lightweight.  However, was unable to find a
license with it.  Tracked earlier copies of the files down to
KDEMultimedia's kmidi which says the license on kmidi is GPLv2 or
later.  With further checking, I found out many of the files were from
timidity_wow_samples.zip, timidity_drums_samples.zip,
timidity_base_samples.zip.  One sample was from propats 3 collection.
Not finding much on licensing for any of those files.  Does anyone
have suggestions for lightweight (or other) instrument
collections/patch files/soundfonts with Free licenses?  Thanks.

From len at ovenwerks.net  Wed Aug 13 16:41:00 2014
From: len at ovenwerks.net (Len Ovens)
Date: Wed, 13 Aug 2014 09:41:00 -0700 (PDT)
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <20140813105944.GA18888@linuxaudio.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <3967821.5Hi4RIW6Gm@col-desktop> <20140812122604.GA13127@linuxaudio.org>
 <3276416.mTp9TkTy7n@col-desktop> <20140813105944.GA18888@linuxaudio.org>
Message-ID: <alpine.DEB.2.10.1408130733260.2503@scott.cbbs.org>

On Wed, 13 Aug 2014, Fons Adriaensen wrote:

> On Wed, Aug 13, 2014 at 12:08:51AM -0400, Tim E. Real wrote:
>> Those that have local RECORD ARM enabled will record when started,
>>  provided global record ENABLE is on.
>
> That assumes that all apps have per-track record enables. A simple
> stereo recorder/player won't have them.
>
> Anyway, to set up recording you'd have to access the local user
> interface of that app, so having to enable record there is no big
> deal. To me it also feels safer.

To be honest, the reason I thought it would be nice is because I was 
looking at an app with no MIDI control in, but does have jack transport 
control. I was thinking that a master record enable was the one thing 
that traditional tape transports control clusters had that was missing. Of 
course just because it is traditional is not a great case for including 
it. On a tape machine all the controls are clustered for other reasons and 
the record arm sometimes does have more space around it or is sunken or 
has ridges on its sides.

Having said that, most control surfaces that include a transport section 
and for that matter, most sw that has a transport section laid out in the 
gui, has the record enable included in that group. So it is a tradition 
that is still in common use. What would be called a defacto standard or a 
standard by tradition. Again I don't know if it is a good standard or not 
but that is where the hand of a recording machine operator will look for 
the record button.

The next question is if a control surface should directly control jack 
transport. I think the answer is that enough people think is should that 
there are a number of standalone jack transport controlers around. Really 
the only thing missing is the record functionallity to be complete. 
Nudging and shuttling are already easy to do... I suppose scrubbing, or 
playing back at non-standard speed is not there, but I am not sure how 
that would be done and for now at least, this is not a universal thing 
that SW does, so most apps would ignore it anyway. A state that says make 
noise even if we are not in play, but we move the transport?

The whole thing comes down to sync and the purpose behind it. This is much 
more common in Linux than other places because we can. The purpose is to 
take a number of applications and perhaps physical playback/record devices 
and to use them all as if they were one machine. From that POV, a single 
transport control makes sense. From that POV, using one control surface 
for more than one application/device also makes sense. I would suggest, 
however, that this practice is becoming less common as applications add 
more functions and become more "all in one". It seems that audio recorders 
now edit MIDI (and have instrument plugins) and MIDI editors record and 
playback sound. Even video can be included. When I first started doing 
music on computers, recording audio on a computer was just a dream. MIDI 
was no problem (if you stayed away from windows) and so audio was recorded 
on a tape machine and the computer sequencer followed the timecode 
stripped on track 8 (permanently set to record at -10db for this purpose). 
Sync was the only way to get enough tracks for many things. Not any more.

Just to be complete... I have looked at OSC. OSC is very flexable. 
However, that flexability seems to be it's problem too. There is no way to 
create a generic control surface with /softwarename/track1/gain and then 
go "bank up/right" and have the mapping change to 
/softwarename/track9/gain within the software... no standards for this. To 
be fair, there was not in MIDI either till people started making control 
surfaces and the software had to deal with whatever they put out and that 
became the standard (The mackie control protcol for example). However, as 
a contol surface manufacture, why would I make an OSC product? What market 
share would that gain me? Really, in order to even create a middleware 
converter from MIDI controller to OSC would be a pain. I would have to do 
all the banking in that middleware. I would have to know all the strip 
names (takeing non-mixer as an example 
"/strip/stripname/pluginname/paramname param") and map them to a number... 
or name them in an unituitive manner in the first place. Ardour uses 
something totally different "/apname/route/function tracknumber param". 
The ardour method at least could be banked reliably. I suppose a patch bay 
kind of middleware that showed controler outputs and SW inputs for control 
could work, but it seems poor to have a user patch what would be normal 
routing for each project. OSC has a way to go, IMO, it is not ready to use 
with control surfaces.



--
Len Ovens
www.ovenwerks.net


From len at ovenwerks.net  Wed Aug 13 18:13:41 2014
From: len at ovenwerks.net (Len Ovens)
Date: Wed, 13 Aug 2014 11:13:41 -0700 (PDT)
Subject: [LAU] control surface design -  was - Jack transport
In-Reply-To: <alpine.DEB.2.10.1408130733260.2503@scott.cbbs.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <3967821.5Hi4RIW6Gm@col-desktop> <20140812122604.GA13127@linuxaudio.org>
 <3276416.mTp9TkTy7n@col-desktop> <20140813105944.GA18888@linuxaudio.org>
 <alpine.DEB.2.10.1408130733260.2503@scott.cbbs.org>
Message-ID: <alpine.DEB.2.10.1408130953300.2503@scott.cbbs.org>

On Wed, 13 Aug 2014, Len Ovens wrote:

> Just to be complete... I have looked at OSC. OSC is very flexable. However,

Another thought I had on this is that with a lot of control surfaces the 
information is a two way street, I move an ecoder, the softwre changed 
some parameter, the sw sends back a visual signal that matches that, the 
control surface changes the look of a light ring. A switch is pressed the 
sw lights the light, etc. The big one is bank change, which OSC could 
support but there is no support in the few pieces of sw I have looked at. 
I know MIDI can be transported over OSC, but why? I don't see applications 
that support it so there would again have to be middleware to convert, and 
there is, but each connection needs to be set up manually. OSC seems to 
have been designed from a SW point of view rather than HW where switches 
and controls are limited and their definition is static. I think OSC needs 
a static section that is well defined, probably starting with or bassed on 
MIDI. Any SW (that is adding OSC) that already has MIDI inputs should be 
able to take OSC input of /midi/whatever (or /static/midi/whatever so that 
static controls can be expanded) as a MIDI input to any of their midi 
ports and route a MIDI output back over OSC as well. I expect over time 
better static(standard) messages will replace MIDI messages (higher 
resolution, floating point or whatever) with messages that while more 
expansive might still be easily converted for old hw/sw. That is, that use 
the same naming for the same function at least.

OSC vs. MIDI
There is a lot of "OSC is better than MIDI because" stuff around. It seems 
a lot of it is not really true:
http://www.midi.org/aboutmidi/midi-osc.php

I would question the "MIDI supports multiple data formats" point in this 
link, while true, it is messy. OSC does do that better. But timing and 
transport are really the same in any practical sense (MIDI over net with 
timestamp as jack does). OSC lacks the one big thing MIDI has: "MIDI 
offers greater interoperability than OSC". It seems OSC is like Linux in 
1995 in this respect. Linux is as "big" as it is today because it "just 
works" in most cases. MIDI has this, OSC doesn't. OSC is great for 
experimental use and places where MIDI doesn't cover, but the time and 
knowledge have to be there. For most uses, MIDI just works.

--
Len Ovens
www.ovenwerks.net


From djbarney at djbarney.org  Wed Aug 13 19:56:49 2014
From: djbarney at djbarney.org (djbarney)
Date: Wed, 13 Aug 2014 19:56:49 +0000
Subject: [LAU] =?utf-8?q?File_managers_for_musicians=2C_waveform_thumbnail?=
 =?utf-8?q?s=2C_presentation_as_tightly_packed_spiral_a=27la_vinyl_=3F_=28?=
 =?utf-8?q?Caja=2C_MATE=29?=
Message-ID: <47e758090882c49435a0c52821059b47@djbarney.org>

Hi,

Thought I'd air this one to see if there's anyone already using this 
kind of thing out there.

I'd become frustrated by lack of support for musicians in Linux window 
manager file managers. No default waveform/spectrogram support. No MIDI 
notation previews. No BPM or key info, etc, etc.

I set up Caja to thumbnail WAV files using Sox ... see some screenshots 
on this forum thread ...

http://linuxmusicians.com/viewtopic.php?f=4&t=12514&p=55280#p55280

I'm looking at some other thumbnailers as listed on the thread.

I'm thinking of doing some development of the MATE Caja file browser to 
allow better presentation of audio file waveform thumbnails as currently 
they can only be square. Maybe LAU members already know of someone who 
has done this ? No point in reinventing the wheel ... otherwise I'll 
take this to LAD.

... and now to the BIG IDEA !

For long samples and tracks having a thumbnail becomes more problematic 
... it may just end up as a thin line, or something that fits the small 
thumbnail but is summing the entire track which is probably not very 
informative. One thing I realised is that I've been using visual 
recognition of tracks for years. I can pick tracks off my vinyl records 
based on the look of the grooves ... basically a vinyl representation of 
the audio waveform. One way of taking this into digital land would be to 
create a full length waveform, and then process it into a tightly packed 
spiral and thumbnail it. That might provide better recognition. So 
that's something I'm working on using Graphics Magick or some other way 
of processing it.

DJ Barney

-- 
~~~

http://djbarney.org

From fons at linuxaudio.org  Wed Aug 13 21:27:55 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Wed, 13 Aug 2014 21:27:55 +0000
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <alpine.DEB.2.10.1408130733260.2503@scott.cbbs.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <3967821.5Hi4RIW6Gm@col-desktop>
 <20140812122604.GA13127@linuxaudio.org>
 <3276416.mTp9TkTy7n@col-desktop>
 <20140813105944.GA18888@linuxaudio.org>
 <alpine.DEB.2.10.1408130733260.2503@scott.cbbs.org>
Message-ID: <20140813212755.GA7562@linuxaudio.org>

On Wed, Aug 13, 2014 at 09:41:00AM -0700, Len Ovens wrote:

> To be honest, the reason I thought it would be nice is because I was
> looking at an app with no MIDI control in, but does have jack
> transport control. I was thinking that a master record enable was
> the one thing that traditional tape transports control clusters had
> that was missing. Of course just because it is traditional is not a
> great case for including it. On a tape machine all the controls are
> clustered for other reasons and the record arm sometimes does have
> more space around it or is sunken or has ridges on its sides.

For good reasons...
 
> Having said that, most control surfaces that include a transport
> section and for that matter, most sw that has a transport section
> laid out in the gui, has the record enable included in that group.
> So it is a tradition that is still in common use. What would be
> called a defacto standard or a standard by tradition. Again I don't
> know if it is a good standard or not but that is where the hand of a
> recording machine operator will look for the record button.

There is one other thing to consider. In most cases the transport
section of a control surface will control a single device, it's 
just a remote control. In that case it certainly makes sense to
have a RECORD button there, even more so if the control surface
also provides access to individual tracks.

OTOH, Jack transoprt is an API designed to sync multiple devices,
which just happens to be usable as a remote control as well.

If it makes sense to have a RECORD function there is not so easy
to decide, I can't give a definite answer to that question. What
*is* clear is that if this is added then the semantics should be
well defined and not leave room for interpretation. Does a RECORD
button talking to Jack transport have the same effect as the other
buttons - if enabled it means enabling record mode on all controlled
apps ? In that case there is a problem with apps that don't have
per-track record enables, they would be put into record mode even
if that is not wanted. Or is it a 'third level' of control, above
the per-track and per-app record enables ? In that case it's less
useful than you may want - it's still necessary to use the per-app
ones.

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From fons at linuxaudio.org  Wed Aug 13 21:49:49 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Wed, 13 Aug 2014 21:49:49 +0000
Subject: [LAU] control surface design -  was - Jack transport
In-Reply-To: <alpine.DEB.2.10.1408130953300.2503@scott.cbbs.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <3967821.5Hi4RIW6Gm@col-desktop>
 <20140812122604.GA13127@linuxaudio.org>
 <3276416.mTp9TkTy7n@col-desktop>
 <20140813105944.GA18888@linuxaudio.org>
 <alpine.DEB.2.10.1408130733260.2503@scott.cbbs.org>
 <alpine.DEB.2.10.1408130953300.2503@scott.cbbs.org>
Message-ID: <20140813214949.GB7562@linuxaudio.org>

On Wed, Aug 13, 2014 at 11:13:41AM -0700, Len Ovens wrote:

> OSC vs. MIDI
> There is a lot of "OSC is better than MIDI because" stuff around.

One problem with OSC is that it is such a mixed beast.

On the lowest level is a data encoding format, no semantics.
But some of its features - structured paths and wildcards -
only have value at a semantic level. But for the rest that
level remains undefined. 

Ther result is that any control surface or similar device
needs to be programmable, and whatever is done with it will
be ad-hoc. 

It would be possible to define some standards, e.g. for
transport control. But unless they are 

*  very strictly defined, and those definitions are 
   enforced in some way,

*  and the standard is designed to be as universal as
   possible, without making assumptions or including
   things that are correct only 99% of the time,

any such standards are destined to fail.

Ciao,


-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From rennabh at gmail.com  Wed Aug 13 22:45:53 2014
From: rennabh at gmail.com (renato)
Date: Thu, 14 Aug 2014 00:45:53 +0200
Subject: [LAU] File managers for musicians, waveform thumbnails,
 presentation as tightly packed spiral a'la vinyl ? (Caja, MATE)
In-Reply-To: <47e758090882c49435a0c52821059b47@djbarney.org>
References: <47e758090882c49435a0c52821059b47@djbarney.org>
Message-ID: <20140814004553.664d1ec7@gmail.com>

On Wed, 13 Aug 2014 19:56:49 +0000
djbarney <djbarney at djbarney.org> wrote:

> Hi,
> 
> Thought I'd air this one to see if there's anyone already using this 
> kind of thing out there.
> 
> I'd become frustrated by lack of support for musicians in Linux
> window manager file managers. No default waveform/spectrogram
> support. No MIDI notation previews. No BPM or key info, etc, etc.
> 
> I set up Caja to thumbnail WAV files using Sox ... see some
> screenshots on this forum thread ...
> 
> http://linuxmusicians.com/viewtopic.php?f=4&t=12514&p=55280#p55280
> 
> I'm looking at some other thumbnailers as listed on the thread.
> 
> I'm thinking of doing some development of the MATE Caja file browser
> to allow better presentation of audio file waveform thumbnails as
> currently they can only be square. Maybe LAU members already know of
> someone who has done this ? No point in reinventing the wheel ...
> otherwise I'll take this to LAD.
> 

Hi, I think samplecat does some of the above. Also I've heard sox has
changed syntax between versions before, so it might break your program
in the future; others where recommending me ecasound instead... or
maybe libsndfile?

cheers,
renato

From rennabh at gmail.com  Wed Aug 13 22:52:09 2014
From: rennabh at gmail.com (renato)
Date: Thu, 14 Aug 2014 00:52:09 +0200
Subject: [LAU] tagging sample libraries: tmsu
Message-ID: <20140814005209.57b3e8fa@gmail.com>

Hi, since I've seen this problem mentioned here before, I just wanted to
give a heads up regarding this program I've recently discovered, tmsu.
It lets you tag files, and you can later use these tags in any program,
since it will create a virtual filesystem (using fuse), with directories
for tags and queries (like "show me all field recordings of stormy
weather done in 2013", provided of course you tagged the files with the
relevant information), all on the fly (to create a query you just
create a directory). It seems very nice for the task, very KISS, though
I have just played shortly with it

http://tmsu.org/

have fun,
renato

From danni.coy at gmail.com  Thu Aug 14 01:23:00 2014
From: danni.coy at gmail.com (Danni Coy)
Date: Thu, 14 Aug 2014 11:23:00 +1000
Subject: [LAU] File managers for musicians, waveform thumbnails,
 presentation as tightly packed spiral a'la vinyl ? (Caja, MATE)
In-Reply-To: <20140814004553.664d1ec7@gmail.com>
References: <47e758090882c49435a0c52821059b47@djbarney.org>
 <20140814004553.664d1ec7@gmail.com>
Message-ID: <CAGR1oLDmEvqXq3=R0jVaiU6h7ug5BRFN-5GSBJCt2Y77Enz_og@mail.gmail.com>

May I make two suggestions
1) mood files -> some audio players generate these - most notably
Amarok with the right plugins installed. These break up the audio
files into 3 bands lows/mids/highs and display the level for each
band.
2) the circular wave displays in freewheeling... These would fit into
an icon preview very easily.

I have some experience doing preview plugins for Dolphin if you think
that would be helpful.

On Thu, Aug 14, 2014 at 8:45 AM, renato <rennabh at gmail.com> wrote:
> On Wed, 13 Aug 2014 19:56:49 +0000
> djbarney <djbarney at djbarney.org> wrote:
>
>> Hi,
>>
>> Thought I'd air this one to see if there's anyone already using this
>> kind of thing out there.
>>
>> I'd become frustrated by lack of support for musicians in Linux
>> window manager file managers. No default waveform/spectrogram
>> support. No MIDI notation previews. No BPM or key info, etc, etc.
>>
>> I set up Caja to thumbnail WAV files using Sox ... see some
>> screenshots on this forum thread ...
>>
>> http://linuxmusicians.com/viewtopic.php?f=4&t=12514&p=55280#p55280
>>
>> I'm looking at some other thumbnailers as listed on the thread.
>>
>> I'm thinking of doing some development of the MATE Caja file browser
>> to allow better presentation of audio file waveform thumbnails as
>> currently they can only be square. Maybe LAU members already know of
>> someone who has done this ? No point in reinventing the wheel ...
>> otherwise I'll take this to LAD.
>>
>
> Hi, I think samplecat does some of the above. Also I've heard sox has
> changed syntax between versions before, so it might break your program
> in the future; others where recommending me ecasound instead... or
> maybe libsndfile?
>
> cheers,
> renato
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

From len at ovenwerks.net  Thu Aug 14 02:16:52 2014
From: len at ovenwerks.net (Len Ovens)
Date: Wed, 13 Aug 2014 19:16:52 -0700 (PDT)
Subject: [LAU] Ardour MIDI tracer
Message-ID: <alpine.DEB.2.10.1408131728020.2503@scott.cbbs.org>


Is it just me? Has anyone else looked at pitch bend events on the Ardour 
MIDI Tracer? Quick test:

================================================
  - edit->Preferences->Control surfaces.
  - select both enabled and feedback for generic MIDI.
  - double click on it and select bcf2000 with mackie protocol
  - open an external midi monitor (using qmidiroute here) and connect it to
 	Ardours MIDI control out.
  - also open Ardours MIDI tracer window and connect it to the same output.
  (Now qmidiroute will be in decimal and MIDI Tracer is hex..)
  - use the mouse to move the gain up and down on channel one with an audio track.
=================================================

I am seeing numbers that make sense on qmidiroute -8013 to 8177, but midi 
Tracer only shows two hex digits ever, shouldn't there be four (maybe 
three to handle 10bits)? It looks like I am seeing only the LSB (I am not 
sure about this as the lowest nyble should be 0). My about box says 
3.5.308 so maybe this has been fixed since then. 3.5.380 seems to be were 
things are at now. I have looked through the change logs here: 
https://ardour.org/whatsnew.html for 3.5.380, (this doesn't seem to be 
included in the source package for 3.5.380) and 
https://ardour.org/news/3.5.older.html which is included with the source 
package. And I don't see any mention of a fix.

BTW, I am having lots of fun with my "MIDI" controler project. Built 
around a $3 USB computer keyboard from dollar store. I use two columns for 
each strip for 10 strips. Strip one for example is 1->z and 2->x. 1 is 
channel select (does nothing right now), 2 is record, q&w are pan control, 
a&z are the fader, s is solo and x is mute. up and down changes banks

Ah, banks. I have bank at 5 and am using 8 channels to test. It starts on 
channel 1 to 5, bank right does 6 to 8 (good so far), then another bank 
right does channel 8. I don't know if this is right or wrong, but my 
expectation is that once my controller has even one strip outside the 
range of channels that I would no longer be able to bank right. I would 
also expect the GUI to make sure all the channels my controler can control 
to be visible. My thought is that in the same way a selected track gets a 
red border (in the mixer) perhaps a green border around the group of 
tracks in the current bank. The BCF2000, for example, does not give much 
indication visually of where the banks are at. Mine of course gives none, 
but I could make a GUI across the bottom of the screen the controler sits 
in front of that does that... it just seems redundant when the mixer is 
already on the screen just above. I do realize that the controler code at 
least for the MCP stuff is not complete and will get done when it gets 
done. What I have already is a wonderful step up.

The transport is on the numeric pad.

All of this could easily change as I add some encoders for pan/faders



--
Len Ovens
www.ovenwerks.net


From len at ovenwerks.net  Thu Aug 14 04:12:54 2014
From: len at ovenwerks.net (Len Ovens)
Date: Wed, 13 Aug 2014 21:12:54 -0700 (PDT)
Subject: [LAU] control surface design -  was - Jack transport
In-Reply-To: <20140813214949.GB7562@linuxaudio.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <3967821.5Hi4RIW6Gm@col-desktop> <20140812122604.GA13127@linuxaudio.org>
 <3276416.mTp9TkTy7n@col-desktop> <20140813105944.GA18888@linuxaudio.org>
 <alpine.DEB.2.10.1408130733260.2503@scott.cbbs.org>
 <alpine.DEB.2.10.1408130953300.2503@scott.cbbs.org>
 <20140813214949.GB7562@linuxaudio.org>
Message-ID: <alpine.DEB.2.10.1408132039370.2503@scott.cbbs.org>

On Wed, 13 Aug 2014, Fons Adriaensen wrote:

> On Wed, Aug 13, 2014 at 11:13:41AM -0700, Len Ovens wrote:
>
>> OSC vs. MIDI
> Ther result is that any control surface or similar device
> needs to be programmable, and whatever is done with it will
> be ad-hoc.

That is what I am seeing.

> It would be possible to define some standards, e.g. for
> transport control. But unless they are
>
> *  very strictly defined, and those definitions are
>   enforced in some way,
>
> *  and the standard is designed to be as universal as
>   possible, without making assumptions or including
>   things that are correct only 99% of the time,

That would be MIDI.

> any such standards are destined to fail.

Concidering OSC has been around for 12 years(v1.0, 17 years since first 
implementation), it may have already. The specification is the most 
non-specific thing I have ever seen. From the home page it seems to have 
not moved at all from 2009 (waiting for funding so 1.1 can be released). 
It would seem almost the same thing could be done with an ssh session 
using arbitrary strings.

--
Len Ovens
www.ovenwerks.net


From willgodfrey at musically.me.uk  Thu Aug 14 06:44:58 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Thu, 14 Aug 2014 07:44:58 +0100
Subject: [LAU] control surface design -  was - Jack transport
In-Reply-To: <alpine.DEB.2.10.1408132039370.2503@scott.cbbs.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <3967821.5Hi4RIW6Gm@col-desktop>
 <20140812122604.GA13127@linuxaudio.org>
 <3276416.mTp9TkTy7n@col-desktop>
 <20140813105944.GA18888@linuxaudio.org>
 <alpine.DEB.2.10.1408130733260.2503@scott.cbbs.org>
 <alpine.DEB.2.10.1408130953300.2503@scott.cbbs.org>
 <20140813214949.GB7562@linuxaudio.org>
 <alpine.DEB.2.10.1408132039370.2503@scott.cbbs.org>
Message-ID: <20140814074458.2f61820c@debian>

On Wed, 13 Aug 2014 21:12:54 -0700 (PDT)
Len Ovens <len at ovenwerks.net> wrote:

> Concidering OSC has been around for 12 years(v1.0, 17 years since first 
> implementation), it may have already. The specification is the most 
> non-specific thing I have ever seen. From the home page it seems to have 
> not moved at all from 2009 (waiting for funding so 1.1 can be released). 
> It would seem almost the same thing could be done with an ssh session 
> using arbitrary strings.

Wow! I hadn't realised it had been around for so long.

In these posts, you described my own concerns pretty accurately. It could
easily become a nightmare of incompatibility. If various bits of kit want to
talk to each other, don't  they all need a translation layer with separate
lists for every device they know about?

Also, while I can see that OSC fits very well for setup messages and general
housekeeping, I wonder at the overhead incurred for time-critical messages.

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From jeremy at autostatic.com  Thu Aug 14 10:48:50 2014
From: jeremy at autostatic.com (Jeremy Jongepier)
Date: Thu, 14 Aug 2014 12:48:50 +0200
Subject: [LAU] Header missing to build Zita AT 1
In-Reply-To: <1407947164.4059.1.camel@rocketmail.com>
References: <1407939448.3069.4.camel@rocketmail.com>
 <20140813144514.GF18888@linuxaudio.org>
 <1407947164.4059.1.camel@rocketmail.com>
Message-ID: <53EC9412.4090004@autostatic.com>

On 08/13/2014 06:26 PM, Ralf Mardorf wrote:
> On Wed, 2014-08-13 at 14:45 +0000, Fons Adriaensen wrote:
>> #include <zita-resampler/resampler.h>
> 
> After editing retuner.h it build without an error. Thank you :).
> 
> In QjackCtl there are only audio ports for AT1, there are no MIDI ports.
> $ zita-at1 -h and the HTML doc don't inform how to connect MIDI.
> 
> Regards,
> Ralf
> 

$ jack_lsp -t | grep pitch -A1
zita-at1:pitch
	8 bit raw midi

So it should be a matter of connecting a MIDI output port to the
zita-at1:pitch JACK MIDI input port.

Jeremy


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From ralf.mardorf at rocketmail.com  Thu Aug 14 11:18:28 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Thu, 14 Aug 2014 13:18:28 +0200
Subject: [LAU] Header missing to build Zita AT 1
In-Reply-To: <53EC9412.4090004@autostatic.com>
References: <1407939448.3069.4.camel@rocketmail.com>
 <20140813144514.GF18888@linuxaudio.org>
 <1407947164.4059.1.camel@rocketmail.com> <53EC9412.4090004@autostatic.com>
Message-ID: <1408015108.778.33.camel@rocketmail.com>

On Thu, 2014-08-14 at 12:48 +0200, Jeremy Jongepier wrote:
> On 08/13/2014 06:26 PM, Ralf Mardorf wrote:
> > In QjackCtl there are only audio ports for AT1, there are no MIDI ports.
> > $ zita-at1 -h and the HTML doc don't inform how to connect MIDI.
> $ jack_lsp -t | grep pitch -A1
> zita-at1:pitch
> 	8 bit raw midi
> 
> So it should be a matter of connecting a MIDI output port to the
> zita-at1:pitch JACK MIDI input port.

Thank you :).

Btw. it's too funny, after compiling I started jackd by QjackCtl [1] and
then AT1 and expected to see a port.

If I start the audio session I currently do and then AT1, the port is
shown by QjackCtl [2] ;).

My bad :).

Regards,
Ralf

[1]
[rocketmouse at archlinux arch2014.1]$ cat ~/.jackdrc 
/usr/bin/jackd -dalsa -dhw:0 -r48000 -p256 -n2

[2]
[rocketmouse at archlinux arch2014.1]$ cat jack start.*
[snip]
roxterm --tab -n "? jackd"       -e "jackd --sync -Xalsarawmidi -dalsa -r48000 -p256"        ; sleep 2
roxterm --tab -n "? a2jmidid"    -e "a2jmidid -e"                                            ; sleep 2
[snip]


From ralf.mardorf at rocketmail.com  Thu Aug 14 11:31:31 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Thu, 14 Aug 2014 13:31:31 +0200
Subject: [LAU] Header missing to build Zita AT 1
In-Reply-To: <1408015108.778.33.camel@rocketmail.com>
References: <1407939448.3069.4.camel@rocketmail.com>
 <20140813144514.GF18888@linuxaudio.org>
 <1407947164.4059.1.camel@rocketmail.com> <53EC9412.4090004@autostatic.com>
 <1408015108.778.33.camel@rocketmail.com>
Message-ID: <1408015891.778.39.camel@rocketmail.com>

PS: The only issue now seems to be, that Qtractor doesn't support
individual MIDI IOs or that I don't know what to do, to get individual
ports. This already is an annoyance for me, when not using AT1. A quick
test, Qtractor MIDI out connected to AT1 MIDI in (AT1 audio wasn't
connected) showed, that AT1 receives in omni mode.

:(

So I at least need to add a MIDI filter between Qtractor and AT1. Is
there a lightweight command line filter that can filter out the MIDI
events of a single channel?


From falktx at gmail.com  Thu Aug 14 11:36:18 2014
From: falktx at gmail.com (Filipe Coelho)
Date: Thu, 14 Aug 2014 12:36:18 +0100
Subject: [LAU] Header missing to build Zita AT 1
In-Reply-To: <1408015891.778.39.camel@rocketmail.com>
References: <1407939448.3069.4.camel@rocketmail.com>
 <20140813144514.GF18888@linuxaudio.org>
 <1407947164.4059.1.camel@rocketmail.com> <53EC9412.4090004@autostatic.com>
 <1408015108.778.33.camel@rocketmail.com>
 <1408015891.778.39.camel@rocketmail.com>
Message-ID: <53EC9F32.2090700@gmail.com>

On 08/14/2014 12:31 PM, Ralf Mardorf wrote:
> PS: The only issue now seems to be, that Qtractor doesn't support
> individual MIDI IOs or that I don't know what to do, to get individual
> ports.
You can use jackass plugin within qtractor.
See https://github.com/falkTX/JackAss/ for more details.

From ralf.mardorf at rocketmail.com  Thu Aug 14 13:29:53 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Thu, 14 Aug 2014 15:29:53 +0200
Subject: [LAU] [Bulk] Re:  Header missing to build Zita AT 1
In-Reply-To: <53EC9F32.2090700@gmail.com>
References: <1407939448.3069.4.camel@rocketmail.com>
 <20140813144514.GF18888@linuxaudio.org>
 <1407947164.4059.1.camel@rocketmail.com> <53EC9412.4090004@autostatic.com>
 <1408015108.778.33.camel@rocketmail.com>
 <1408015891.778.39.camel@rocketmail.com> <53EC9F32.2090700@gmail.com>
Message-ID: <1408022993.778.44.camel@rocketmail.com>

On Thu, 2014-08-14 at 12:36 +0100, Filipe Coelho wrote:
> On 08/14/2014 12:31 PM, Ralf Mardorf wrote:
> > PS: The only issue now seems to be, that Qtractor doesn't support
> > individual MIDI IOs or that I don't know what to do, to get
> individual
> > ports.
> You can use jackass plugin within qtractor.
> See https://github.com/falkTX/JackAss/ for more details.

JackAss.cpp:29:52: fatal error:
public.sdk/source/vst2.x/audioeffect.cpp: No such file or directory
 #include "public.sdk/source/vst2.x/audioeffect.cpp"

So I tried to find it in one of my old installs, but I only found

[rocketmouse at archlinux JackAss]$ ls /mnt/suse11.2/usr/local/include/a*
/mnt/suse11.2/usr/local/include/aeffect.h  /mnt/suse11.2/usr/local/include/aeffectx.h

Steinberg claimed that there is no account for an address I used and I
couldn't create a new account using this address, that seemingly already
is used for an account, so I needed to create a new account with another
mail address. Ok, now I have an account to login, but I can't see a
download for VST SDK 2.x, there seems to be a download link for Version
3.6.0 only.


From ralf.mardorf at rocketmail.com  Thu Aug 14 15:08:17 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Thu, 14 Aug 2014 17:08:17 +0200
Subject: [LAU] Linux mixer
Message-ID: <1408028897.778.55.camel@rocketmail.com>

Hi,

regarding to some mixer issues with Qtractor, discussed at other lists,
I searched the Internet for a sane virtual mixer, but didn't find one
for Linux.

I need stereo or mono channels, it doesn't matter when stereo channels
only provide balance, it's ok, real panning would be better.

An insert (audio out/in) isn't needed, I could add an "insert" before I
connect to the mixer. Inserting LV2 and LADSPA effects would be useful,
but aren't too important, with one exception, each channel needs an EQ
(Fons' parametric EQ or a similar EQ) and I need at least one _post_
fader aux send to a sub group or at least to an aux return pot for the
stereo master sum. Pre fader aux would be completely useless. A mute
and/or solo switch would be useful too, not needed, but it's important,
that the mixer settings can be stored and restored.

Regards,
Ralf


From jeremy at autostatic.com  Thu Aug 14 15:45:14 2014
From: jeremy at autostatic.com (Jeremy Jongepier)
Date: Thu, 14 Aug 2014 17:45:14 +0200
Subject: [LAU] Header missing to build Zita AT 1
In-Reply-To: <1408015891.778.39.camel@rocketmail.com>
References: <1407939448.3069.4.camel@rocketmail.com>
 <20140813144514.GF18888@linuxaudio.org>
 <1407947164.4059.1.camel@rocketmail.com> <53EC9412.4090004@autostatic.com>
 <1408015108.778.33.camel@rocketmail.com>
 <1408015891.778.39.camel@rocketmail.com>
Message-ID: <53ECD98A.2080800@autostatic.com>

On 08/14/2014 01:31 PM, Ralf Mardorf wrote:
> PS: The only issue now seems to be, that Qtractor doesn't support
> individual MIDI IOs or that I don't know what to do, to get individual
> ports.

Create a new MIDI bus in Qtractor: View - Buses - select MIDI Master bus
- change the name of the bus - click Create.

Jeremy

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From len at ovenwerks.net  Thu Aug 14 15:55:49 2014
From: len at ovenwerks.net (Len Ovens)
Date: Thu, 14 Aug 2014 08:55:49 -0700 (PDT)
Subject: [LAU] Linux mixer
In-Reply-To: <1408028897.778.55.camel@rocketmail.com>
References: <1408028897.778.55.camel@rocketmail.com>
Message-ID: <alpine.DEB.2.10.1408140851150.2503@scott.cbbs.org>

On Thu, 14 Aug 2014, Ralf Mardorf wrote:

> regarding to some mixer issues with Qtractor, discussed at other lists,
> I searched the Internet for a sane virtual mixer, but didn't find one
> for Linux.
>
> I need stereo or mono channels, it doesn't matter when stereo channels
> only provide balance, it's ok, real panning would be better.
>
> An insert (audio out/in) isn't needed, I could add an "insert" before I
> connect to the mixer. Inserting LV2 and LADSPA effects would be useful,
> but aren't too important, with one exception, each channel needs an EQ
> (Fons' parametric EQ or a similar EQ) and I need at least one _post_
> fader aux send to a sub group or at least to an aux return pot for the
> stereo master sum. Pre fader aux would be completely useless. A mute
> and/or solo switch would be useful too, not needed, but it's important,
> that the mixer settings can be stored and restored.

Non-mixer seems to have all that.
Using jackrack or lv2rack and filling with whatever plugins create a mixer 
would work too.
non-mixer does save projects though. I do not know about the racks.
Ardour could be used, but is probably overkill and may use more resources 
than you wish.


--
Len Ovens
www.ovenwerks.net


From ralf.mardorf at rocketmail.com  Thu Aug 14 15:58:31 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Thu, 14 Aug 2014 17:58:31 +0200
Subject: [LAU] [solved] Header missing to build Zita AT 1
In-Reply-To: <1408022993.778.44.camel@rocketmail.com>
References: <1407939448.3069.4.camel@rocketmail.com>
 <20140813144514.GF18888@linuxaudio.org>
 <1407947164.4059.1.camel@rocketmail.com> <53EC9412.4090004@autostatic.com>
 <1408015108.778.33.camel@rocketmail.com>
 <1408015891.778.39.camel@rocketmail.com> <53EC9F32.2090700@gmail.com>
 <1408022993.778.44.camel@rocketmail.com>
Message-ID: <1408031911.778.59.camel@rocketmail.com>

On Thu, 2014-08-14 at 15:29 +0200, Ralf Mardorf wrote:
> On Thu, 2014-08-14 at 12:36 +0100, Filipe Coelho wrote:
> > On 08/14/2014 12:31 PM, Ralf Mardorf wrote:
> > > PS: The only issue now seems to be, that Qtractor doesn't support
> > > individual MIDI IOs or that I don't know what to do, to get
> > individual
> > > ports.
> > You can use jackass plugin within qtractor.
> > See https://github.com/falkTX/JackAss/ for more details.
> 
> JackAss.cpp:29:52: fatal error:
> public.sdk/source/vst2.x/audioeffect.cpp: No such file or directory
>  #include "public.sdk/source/vst2.x/audioeffect.cpp"
> [snip] Ok, now I have an account to login, but I can't see a
> download for VST SDK 2.x, there seems to be a download link for Version
> 3.6.0 only.

"Select one of the buses of the type of bus that you want to create
(MIDI or Audio) and it's properties will be shown. Once you change the
                                                   ^^^^^^^^^^^^^^^^^^^
name, the Update and Create buttons will be available." - http://sourceforge.net/p/qtractor/wiki/How%20To%20-%20Create%20Individual%20MIDI%20and%20Audio%20Buses-Ports/
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^

So there's no need for JackAss anymore :), the usage of the GUI just is
strange, but it's possible to get the individual outputs :).

There still is to test, if Qtractor's audio track aux sends will be post
fader, but that's an issue, that's not related to this thread ;).


From peter at peterlutek.com  Thu Aug 14 16:03:27 2014
From: peter at peterlutek.com (Peter Lutek)
Date: Thu, 14 Aug 2014 12:03:27 -0400
Subject: [LAU] Linux mixer
In-Reply-To: <alpine.DEB.2.10.1408140851150.2503@scott.cbbs.org>
References: <1408028897.778.55.camel@rocketmail.com>
 <alpine.DEB.2.10.1408140851150.2503@scott.cbbs.org>
Message-ID: <036ff5d4783d733f51fca3040281afa6@peterlutek.com>

On 2014-08-14 11:55, Len Ovens wrote:
> On Thu, 14 Aug 2014, Ralf Mardorf wrote:
>
>> regarding to some mixer issues with Qtractor, discussed at other 
>> lists,
>> I searched the Internet for a sane virtual mixer, but didn't find 
>> one
>> for Linux.
>>
>> I need stereo or mono channels, it doesn't matter when stereo 
>> channels
>> only provide balance, it's ok, real panning would be better.
>>
>> An insert (audio out/in) isn't needed, I could add an "insert" 
>> before I
>> connect to the mixer. Inserting LV2 and LADSPA effects would be 
>> useful,
>> but aren't too important, with one exception, each channel needs an 
>> EQ
>> (Fons' parametric EQ or a similar EQ) and I need at least one _post_
>> fader aux send to a sub group or at least to an aux return pot for 
>> the
>> stereo master sum. Pre fader aux would be completely useless. A mute
>> and/or solo switch would be useful too, not needed, but it's 
>> important,
>> that the mixer settings can be stored and restored.
>
> Non-mixer seems to have all that.

unfortunately, Non-mixer only hosts LADSPA, not LV2, i believe... but i 
guess you could do an insert to jalv2 for that, right?

cheers!
.pltk.

-- 
Peter Lutek
improvising musician in Toronto, Canada
http://peterlutek.com

From peter at peterlutek.com  Thu Aug 14 16:05:58 2014
From: peter at peterlutek.com (Peter Lutek)
Date: Thu, 14 Aug 2014 12:05:58 -0400
Subject: [LAU] Linux mixer
In-Reply-To: <036ff5d4783d733f51fca3040281afa6@peterlutek.com>
References: <1408028897.778.55.camel@rocketmail.com>
 <alpine.DEB.2.10.1408140851150.2503@scott.cbbs.org>
 <036ff5d4783d733f51fca3040281afa6@peterlutek.com>
Message-ID: <8f19bd5a34e38598228b08611d5f4c7b@peterlutek.com>

On 2014-08-14 12:03, Peter Lutek wrote:
> On 2014-08-14 11:55, Len Ovens wrote:
>> On Thu, 14 Aug 2014, Ralf Mardorf wrote:
>>
>>> regarding to some mixer issues with Qtractor, discussed at other 
>>> lists,
>>> I searched the Internet for a sane virtual mixer, but didn't find 
>>> one
>>> for Linux.
>>>
>>> I need stereo or mono channels, it doesn't matter when stereo 
>>> channels
>>> only provide balance, it's ok, real panning would be better.
>>>
>>> An insert (audio out/in) isn't needed, I could add an "insert" 
>>> before I
>>> connect to the mixer. Inserting LV2 and LADSPA effects would be 
>>> useful,
>>> but aren't too important, with one exception, each channel needs an 
>>> EQ
>>> (Fons' parametric EQ or a similar EQ) and I need at least one 
>>> _post_
>>> fader aux send to a sub group or at least to an aux return pot for 
>>> the
>>> stereo master sum. Pre fader aux would be completely useless. A 
>>> mute
>>> and/or solo switch would be useful too, not needed, but it's 
>>> important,
>>> that the mixer settings can be stored and restored.
>>
>> Non-mixer seems to have all that.
>
> unfortunately, Non-mixer only hosts LADSPA, not LV2, i believe... but
> i guess you could do an insert to jalv2 for that, right?
>

ooops.... meant "jalv", of course!  :)
.pltk.

From ralf.mardorf at rocketmail.com  Thu Aug 14 16:12:53 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Thu, 14 Aug 2014 18:12:53 +0200
Subject: [LAU] [Bulk] Re:  Linux mixer
In-Reply-To: <alpine.DEB.2.10.1408140851150.2503@scott.cbbs.org>
References: <1408028897.778.55.camel@rocketmail.com>
 <alpine.DEB.2.10.1408140851150.2503@scott.cbbs.org>
Message-ID: <1408032773.778.64.camel@rocketmail.com>

On Thu, 2014-08-14 at 08:55 -0700, Len Ovens wrote:

> Non-mixer seems to have all that.

The GUI doesn't look very user friendly.

> Using jackrack or lv2rack and filling with whatever plugins create a mixer 
> would work too.

True, but that's an annoying not user friendly workaround and I planned
to do it that way ot to test non-mixer.

> Ardour could be used, but is probably overkill and may use more resources 
> than you wish.

Indeed, I could launch Ardour just to use it as a mixer. Ardour's mixer
is ok, but I don't want to use Ardour, anyway, I could test how much
resources Ardour2 (or 3) needs and use Qtractor with Ardour's mixer. I
didn't think about using Ardour as a mixer only! A good idea :)! Thank
you.

Now that I found out, how to add a bus to Qtractor, at least for audio
of the MIDI plugins I could use the aux send plugins. Qtrators mixer
seems to send MIDI CC to the synth plugins, the fader seems not to
control the audio signal, so any aux send plug automatically is post
fader, I just fear that for audio tracks those aux send plugins will be
pre fader.



From ralf.mardorf at rocketmail.com  Thu Aug 14 16:15:17 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Thu, 14 Aug 2014 18:15:17 +0200
Subject: [LAU] [Bulk] Re:  Header missing to build Zita AT 1
In-Reply-To: <53ECD98A.2080800@autostatic.com>
References: <1407939448.3069.4.camel@rocketmail.com>
 <20140813144514.GF18888@linuxaudio.org>
 <1407947164.4059.1.camel@rocketmail.com> <53EC9412.4090004@autostatic.com>
 <1408015108.778.33.camel@rocketmail.com>
 <1408015891.778.39.camel@rocketmail.com> <53ECD98A.2080800@autostatic.com>
Message-ID: <1408032917.778.66.camel@rocketmail.com>

On Thu, 2014-08-14 at 17:45 +0200, Jeremy Jongepier wrote:
> - change the name of the bus - click Create.
    ^^^^^^^^^^^^^^^^^^^^^^^^^

Yes, that was the culprit :).

"I suspect that this one "gotcha" - the fact that you have to modify the
name before you can create a new bus - may be the least user-friendly
aspect of an otherwise fairly discoverable interface." -
http://sourceforge.net/p/qtractor/wiki/How%20To%20-%20Create%20Individual%20MIDI%20and%20Audio%20Buses-Ports/



From ralf.mardorf at rocketmail.com  Thu Aug 14 16:35:25 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Thu, 14 Aug 2014 18:35:25 +0200
Subject: [LAU] [Bulk] Re:  Linux mixer
In-Reply-To: <036ff5d4783d733f51fca3040281afa6@peterlutek.com>
References: <1408028897.778.55.camel@rocketmail.com>
 <alpine.DEB.2.10.1408140851150.2503@scott.cbbs.org>
 <036ff5d4783d733f51fca3040281afa6@peterlutek.com>
Message-ID: <1408034125.32198.1.camel@rocketmail.com>

On Thu, 2014-08-14 at 12:03 -0400, Peter Lutek wrote:
> On 2014-08-14 11:55, Len Ovens wrote:
> > Non-mixer seems to have all that.
> 
> unfortunately, Non-mixer only hosts LADSPA, not LV2, i believe... but i 
> guess you could do an insert to jalv2 for that, right?

All I need regarding to effect plugins is LADAPA.

[rocketmouse at archlinux ~]$ pacman -Ql fil-plugins
[snip]
fil-plugins /usr/lib/ladspa/filters.so

I'm an old school engineer. 

I need an EQ for each channel, I usually don't use compressors etc. for
individual tracks, I make a transparent mix giving the audio signal it's
own place in the "frequency room" instead of killing the dynamic.

Sometimes a flanger, phaser or what ever else effect is nice, but analog
mixers also don't provide those special effects and I also don't need
them for the workflow using a virtual mixer. If such an effect is
needed, then it's less work to use an external virtual host (I anyway
prefer hardware effects), but very important for the workflow is a good
EQ and a few post fader aux sends, especially for reverb. Btw.
automation also isn't needed, even in studios flying faders usually are
used to restore a session, seldom for wild automated mixing during a
production.

Btw. for the stereo sum I like to use multi band compression.

IMO it's a pity that so few Linux mixers provide "normal" workflow and
it's funny that users tend to use that much fader automation, while they
at the same time add tons of limiters and compressors.

Resume: Linux mixing techniques seems to be for another generation of
engineers, but less good for dinos. It's not my way to mix music.


From falktx at gmail.com  Thu Aug 14 16:47:34 2014
From: falktx at gmail.com (Filipe Coelho)
Date: Thu, 14 Aug 2014 17:47:34 +0100
Subject: [LAU] [Bulk] Re:  Header missing to build Zita AT 1
In-Reply-To: <1408022993.778.44.camel@rocketmail.com>
References: <1407939448.3069.4.camel@rocketmail.com>
 <20140813144514.GF18888@linuxaudio.org>
 <1407947164.4059.1.camel@rocketmail.com> <53EC9412.4090004@autostatic.com>
 <1408015108.778.33.camel@rocketmail.com>
 <1408015891.778.39.camel@rocketmail.com> <53EC9F32.2090700@gmail.com>
 <1408022993.778.44.camel@rocketmail.com>
Message-ID: <53ECE826.1020807@gmail.com>

On 08/14/2014 02:29 PM, Ralf Mardorf wrote:
> On Thu, 2014-08-14 at 12:36 +0100, Filipe Coelho wrote:
>> On 08/14/2014 12:31 PM, Ralf Mardorf wrote:
>>> PS: The only issue now seems to be, that Qtractor doesn't support
>>> individual MIDI IOs or that I don't know what to do, to get
>> individual
>>> ports.
>> You can use jackass plugin within qtractor.
>> See https://github.com/falkTX/JackAss/ for more details.
> JackAss.cpp:29:52: fatal error:
> public.sdk/source/vst2.x/audioeffect.cpp: No such file or directory
>   #include "public.sdk/source/vst2.x/audioeffect.cpp"
>
> So I tried to find it in one of my old installs, but I only found
>
> [rocketmouse at archlinux JackAss]$ ls /mnt/suse11.2/usr/local/include/a*
> /mnt/suse11.2/usr/local/include/aeffect.h  /mnt/suse11.2/usr/local/include/aeffectx.h
>
> Steinberg claimed that there is no account for an address I used and I
> couldn't create a new account using this address, that seemingly already
> is used for an account, so I needed to create a new account with another
> mail address. Ok, now I have an account to login, but I can't see a
> download for VST SDK 2.x, there seems to be a download link for Version
> 3.6.0 only.
Yeah, the VST SDK is a tricky thing.

But there are pre-compiled binaries in the git repo.


From jeremy at autostatic.com  Thu Aug 14 18:09:39 2014
From: jeremy at autostatic.com (Jeremy Jongepier)
Date: Thu, 14 Aug 2014 20:09:39 +0200
Subject: [LAU] [Bulk] Re:  Header missing to build Zita AT 1
In-Reply-To: <53ECE826.1020807@gmail.com>
References: <1407939448.3069.4.camel@rocketmail.com>
 <20140813144514.GF18888@linuxaudio.org>
 <1407947164.4059.1.camel@rocketmail.com> <53EC9412.4090004@autostatic.com>
 <1408015108.778.33.camel@rocketmail.com>
 <1408015891.778.39.camel@rocketmail.com> <53EC9F32.2090700@gmail.com>
 <1408022993.778.44.camel@rocketmail.com> <53ECE826.1020807@gmail.com>
Message-ID: <53ECFB63.3080302@autostatic.com>

On 08/14/2014 06:47 PM, Filipe Coelho wrote:
> On 08/14/2014 02:29 PM, Ralf Mardorf wrote:
>> On Thu, 2014-08-14 at 12:36 +0100, Filipe Coelho wrote:
>>> On 08/14/2014 12:31 PM, Ralf Mardorf wrote:
>>>> PS: The only issue now seems to be, that Qtractor doesn't support
>>>> individual MIDI IOs or that I don't know what to do, to get
>>> individual
>>>> ports.
>>> You can use jackass plugin within qtractor.
>>> See https://github.com/falkTX/JackAss/ for more details.
>> JackAss.cpp:29:52: fatal error:
>> public.sdk/source/vst2.x/audioeffect.cpp: No such file or directory
>>   #include "public.sdk/source/vst2.x/audioeffect.cpp"
>>
>> So I tried to find it in one of my old installs, but I only found
>>
>> [rocketmouse at archlinux JackAss]$ ls /mnt/suse11.2/usr/local/include/a*
>> /mnt/suse11.2/usr/local/include/aeffect.h 
>> /mnt/suse11.2/usr/local/include/aeffectx.h
>>
>> Steinberg claimed that there is no account for an address I used and I
>> couldn't create a new account using this address, that seemingly already
>> is used for an account, so I needed to create a new account with another
>> mail address. Ok, now I have an account to login, but I can't see a
>> download for VST SDK 2.x, there seems to be a download link for Version
>> 3.6.0 only.
> Yeah, the VST SDK is a tricky thing.
> 
> But there are pre-compiled binaries in the git repo.

https://raw.githubusercontent.com/AutoStatic/scripts/ubuntu/vst_sdk_installer

Just add your Steinberg user name and password and run the script as
root. Patched header files will be installed to /usr/include/vst by default.

Jeremy

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From ralf.mardorf at rocketmail.com  Thu Aug 14 20:46:07 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Thu, 14 Aug 2014 22:46:07 +0200
Subject: [LAU] [Bulk] Re: [Bulk] Re: Header missing to build Zita AT 1
In-Reply-To: <53ECFB63.3080302@autostatic.com>
References: <1407939448.3069.4.camel@rocketmail.com>
 <20140813144514.GF18888@linuxaudio.org>
 <1407947164.4059.1.camel@rocketmail.com> <53EC9412.4090004@autostatic.com>
 <1408015108.778.33.camel@rocketmail.com>
 <1408015891.778.39.camel@rocketmail.com> <53EC9F32.2090700@gmail.com>
 <1408022993.778.44.camel@rocketmail.com> <53ECE826.1020807@gmail.com>
 <53ECFB63.3080302@autostatic.com>
Message-ID: <1408049167.32198.7.camel@rocketmail.com>


> >> JackAss.cpp:29:52: fatal error:
> >> public.sdk/source/vst2.x/audioeffect.cpp: No such file or directory
> >>   #include "public.sdk/source/vst2.x/audioeffect.cpp"
> >>
> >> So I tried to find it in one of my old installs, but I only found
> >>
> >> [rocketmouse at archlinux JackAss]$ ls /mnt/suse11.2/usr/local/include/a*
> >> /mnt/suse11.2/usr/local/include/aeffect.h 
> >> /mnt/suse11.2/usr/local/include/aeffectx.h


> https://raw.githubusercontent.com/AutoStatic/scripts/ubuntu/vst_sdk_installer

Hi Jeremy,

if I understand correctly, the script does provide the two headers I
own.

"unzip -j -o $sdk_zip "*aeffect*" -d $sdk_dir"
"sed -i '69s/__cdecl/\/\/__cdecl/' $sdk_dir/aeffect.h"

But it doesn't provide audioeffect.cpp. It seems to be available here:

https://raw.githubusercontent.com/CharlesHolbrow/ChuckDelay/master/public.sdk/source/vst2.x/audioeffect.cpp

But what's about the headers:
#include "audioeffect.h"
#include "aeffeditor.h"

They don't fit to "*aeffect*".

Regards,
Ralf


From len at ovenwerks.net  Fri Aug 15 00:13:11 2014
From: len at ovenwerks.net (Len Ovens)
Date: Thu, 14 Aug 2014 17:13:11 -0700 (PDT)
Subject: [LAU] Linux mixer
In-Reply-To: <036ff5d4783d733f51fca3040281afa6@peterlutek.com>
References: <1408028897.778.55.camel@rocketmail.com>
 <alpine.DEB.2.10.1408140851150.2503@scott.cbbs.org>
 <036ff5d4783d733f51fca3040281afa6@peterlutek.com>
Message-ID: <alpine.DEB.2.10.1408141710480.2503@scott.cbbs.org>

On Thu, 14 Aug 2014, Peter Lutek wrote:

> On 2014-08-14 11:55, Len Ovens wrote:
>> On Thu, 14 Aug 2014, Ralf Mardorf wrote:
>> 
>>> regarding to some mixer issues with Qtractor, discussed at other lists,
>>> I searched the Internet for a sane virtual mixer, but didn't find one
>>> for Linux.
>>> 
>>> I need stereo or mono channels, it doesn't matter when stereo channels
>>> only provide balance, it's ok, real panning would be better.
>>> 
>>> An insert (audio out/in) isn't needed, I could add an "insert" before I
>>> connect to the mixer. Inserting LV2 and LADSPA effects would be useful,
>>> but aren't too important, with one exception, each channel needs an EQ
>>> (Fons' parametric EQ or a similar EQ) and I need at least one _post_
>>> fader aux send to a sub group or at least to an aux return pot for the
>>> stereo master sum. Pre fader aux would be completely useless. A mute
>>> and/or solo switch would be useful too, not needed, but it's important,
>>> that the mixer settings can be stored and restored.
>> 
>> Non-mixer seems to have all that.
>
> unfortunately, Non-mixer only hosts LADSPA, not LV2, i believe... but i guess 
> you could do an insert to jalv2 for that, right?

For the plugins he asked for, they are available in LADSPA.

--
Len Ovens
www.ovenwerks.net


From janina at rednote.net  Fri Aug 15 07:45:37 2014
From: janina at rednote.net (Janina Sajka)
Date: Fri, 15 Aug 2014 03:45:37 -0400
Subject: [LAU] Slowing down audio while keeping same pitch ?
In-Reply-To: <20140729204713.07fcc456@mistral>
References: <20140729204713.07fcc456@mistral>
Message-ID: <20140815074537.GD4459@concerto.rednote.net>

The technical term for this is "time shifting."

You're probably not a Linux user, but Linux has a wonderful command line
utility for doing this called rubberband
Janina

jonetsu at teksavvy.com writes:
> Is it possible to slow down an audio file (mp3 for instance) while
> keeping the same pitch ?  Does it even make sense ?  The software would
> have to insert more of the same audio 'bits' to make it last longer.
> Is it possible ?
> 
> Cheers.
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

-- 

Janina Sajka,	Phone:	+1.443.300.2200
			sip:janina at asterisk.rednote.net
		Email:	janina at rednote.net

Linux Foundation Fellow
Executive Chair, Accessibility Workgroup:	http://a11y.org

The World Wide Web Consortium (W3C), Web Accessibility Initiative (WAI)
Chair,	Protocols & Formats	http://www.w3.org/wai/pf
	Indie UI			http://www.w3.org/WAI/IndieUI/


From ralf.mardorf at rocketmail.com  Fri Aug 15 08:06:17 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Fri, 15 Aug 2014 10:06:17 +0200
Subject: [LAU] [Bulk] Re: Slowing down audio while keeping same pitch ?
In-Reply-To: <20140815074537.GD4459@concerto.rednote.net>
References: <20140729204713.07fcc456@mistral>
 <20140815074537.GD4459@concerto.rednote.net>
Message-ID: <1408089977.2680.4.camel@rocketmail.com>

On Fri, 2014-08-15 at 03:45 -0400, Janina Sajka wrote:
> The technical term for this is "time shifting."

"Time stretching is the process of changing the speed or duration of an
audio signal without affecting its pitch. Pitch scaling or pitch
shifting is the opposite: the process of changing the pitch without
affecting the speed." -
https://en.wikipedia.org/wiki/Audio_timescale-pitch_modification

OTOH this short explanation is very vague. A pitch shifter usually
transforms the audio signal by adding a Micky Mouse vocal effect, so for
pitch shifting without this effect, a time stretching software is
needed.


From janina at rednote.net  Fri Aug 15 08:06:21 2014
From: janina at rednote.net (Janina Sajka)
Date: Fri, 15 Aug 2014 04:06:21 -0400
Subject: [LAU] Slowing down audio while keeping same pitch ?
In-Reply-To: <20140815074537.GD4459@concerto.rednote.net>
References: <20140729204713.07fcc456@mistral>
 <20140815074537.GD4459@concerto.rednote.net>
Message-ID: <20140815080621.GE4459@concerto.rednote.net>

Argh, it's too late, and I should be sleeping ...

The correct technical term is "time scale modification." Here's a
Wikipedia page on the subject:

https://en.wikipedia.org/wiki/Audio_timescale-pitch_modification

Sorry for the mindless comment below.

Janina

Janina Sajka writes:
> The technical term for this is "time shifting."
> 
> You're probably not a Linux user, but Linux has a wonderful command line
> utility for doing this called rubberband
> Janina
> 
> jonetsu at teksavvy.com writes:
> > Is it possible to slow down an audio file (mp3 for instance) while
> > keeping the same pitch ?  Does it even make sense ?  The software would
> > have to insert more of the same audio 'bits' to make it last longer.
> > Is it possible ?
> > 
> > Cheers.
> > _______________________________________________
> > Linux-audio-user mailing list
> > Linux-audio-user at lists.linuxaudio.org
> > http://lists.linuxaudio.org/listinfo/linux-audio-user
> 
> -- 
> 
> Janina Sajka,	Phone:	+1.443.300.2200
> 			sip:janina at asterisk.rednote.net
> 		Email:	janina at rednote.net
> 
> Linux Foundation Fellow
> Executive Chair, Accessibility Workgroup:	http://a11y.org
> 
> The World Wide Web Consortium (W3C), Web Accessibility Initiative (WAI)
> Chair,	Protocols & Formats	http://www.w3.org/wai/pf
> 	Indie UI			http://www.w3.org/WAI/IndieUI/
> 
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

-- 

Janina Sajka,	Phone:	+1.443.300.2200
			sip:janina at asterisk.rednote.net
		Email:	janina at rednote.net

Linux Foundation Fellow
Executive Chair, Accessibility Workgroup:	http://a11y.org

The World Wide Web Consortium (W3C), Web Accessibility Initiative (WAI)
Chair,	Protocols & Formats	http://www.w3.org/wai/pf
	Indie UI			http://www.w3.org/WAI/IndieUI/


From fons at linuxaudio.org  Fri Aug 15 15:05:00 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Fri, 15 Aug 2014 15:05:00 +0000
Subject: [LAU] Update of zita-at1
Message-ID: <20140815150500.GB4669@linuxaudio.org>

Zita-at1-0.4.0 is now avaliable at the usual place:

<http://kokkinizita.linuxaudio.org/linuxaudio/downloads/index.html>

Bugfixes, MIDI channel selection added.

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From ralf.mardorf at rocketmail.com  Fri Aug 15 17:06:18 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Fri, 15 Aug 2014 19:06:18 +0200
Subject: [LAU] [Bulk]  Update of zita-at1
In-Reply-To: <20140815150500.GB4669@linuxaudio.org>
References: <20140815150500.GB4669@linuxaudio.org>
Message-ID: <1408122378.838.1.camel@rocketmail.com>

On Fri, 2014-08-15 at 15:05 +0000, Fons Adriaensen wrote:
> Zita-at1-0.4.0 is now avaliable at the usual place:
> 
> <http://kokkinizita.linuxaudio.org/linuxaudio/downloads/index.html>
> 
> Bugfixes, MIDI channel selection added.

Btw. I use two instances of AT1 to add strange phasing/panning to stereo
percussion and ambient synth noise. By mixing those strange signals with
the original audio stereo signals and checking the result with a phase
correlator I ensure that the result is ok and not completely insane. For
my usage it's wanted that the left and right channel aren't doing the
same, but perhaps it's useful to provide pitch detection by one channel,
to process two channels in the same way. I guess that usually vocals and
instruments are recorded mono when using an auto tuner is intended, but
perhaps sometimes people want to fix a stereo recording.

For AT1 and Rev1 I'm missing a config file, so I ended with using the gx
plugin for restoring the settings. Fortunately I can use AT1 with the
default settings (audio only, no MIDI). Funnily enough, I could use Rev1
more or less with the default settings, most important is setting it to
100% effect signal, instead of the effect/dry mix.

REV1 IMO is a very good reverb, but IMO it only provides one character,
good quality, very usable, but different characters would be nice.

Did anybody try to use AT1 to manipulate a synth sound's pitch? IOW on
purpose playing wrong notes, to get an auto tune effect. I guess I'll
test it with a very hard sawtooth. Perhaps an e. guitar could sound a
little bit like an old guitar synth?


From brummer- at web.de  Fri Aug 15 18:37:59 2014
From: brummer- at web.de (hermann meyer)
Date: Fri, 15 Aug 2014 20:37:59 +0200
Subject: [LAU] [Bulk]  Update of zita-at1
In-Reply-To: <1408122378.838.1.camel@rocketmail.com>
References: <20140815150500.GB4669@linuxaudio.org>
 <1408122378.838.1.camel@rocketmail.com>
Message-ID: <53EE5387.6010404@web.de>


Am 15.08.2014 19:06, schrieb Ralf Mardorf:
> For AT1 and Rev1 I'm missing a config file, so I ended with using the gx
> plugin for restoring the settings.
For the record,
there is no version of AT1 as gx plugin, nor as LV2 or Ladspa.
Please note, that gx plugin is a own api in itself.
http://sourceforge.net/p/guitarix/git/ci/master/tree/trunk/src/headers/gx_plugin.h
The gx version of Rev1 is the faust implementation of (zita) Rev1, in  
both cases, as gx plugin and as LV2 plug.

regards
hermann

From ralf.mardorf at rocketmail.com  Fri Aug 15 19:34:00 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Fri, 15 Aug 2014 21:34:00 +0200
Subject: [LAU] [Bulk]  Update of zita-at1
In-Reply-To: <53EE5387.6010404@web.de>
References: <20140815150500.GB4669@linuxaudio.org>
 <1408122378.838.1.camel@rocketmail.com> <53EE5387.6010404@web.de>
Message-ID: <1408131240.838.3.camel@rocketmail.com>

On Fri, 2014-08-15 at 20:37 +0200, hermann meyer wrote:
> Am 15.08.2014 19:06, schrieb Ralf Mardorf:
> > For AT1 and Rev1 I'm missing a config file, so I ended with using the gx
> > plugin for restoring the settings.
> For the record,
> there is no version of AT1 as gx plugin, nor as LV2 or Ladspa.
> Please note, that gx plugin is a own api in itself.
> http://sourceforge.net/p/guitarix/git/ci/master/tree/trunk/src/headers/gx_plugin.h
> The gx version of Rev1 is the faust implementation of (zita) Rev1, in  
> both cases, as gx plugin and as LV2 plug.

Sure, I'm using the gx version of Rev1. I started with compiling
zita-rev1 and zita-at1 and used both in combination. The original test
was to find out, if it's possible to get talking drums, by using conga
samples in combination with AT1 and to cloud side effects of AT1 by the
reverb.


From paul at linuxaudiosystems.com  Fri Aug 15 21:26:26 2014
From: paul at linuxaudiosystems.com (Paul Davis)
Date: Fri, 15 Aug 2014 17:26:26 -0400
Subject: [LAU] Ardour MIDI tracer
In-Reply-To: <alpine.DEB.2.10.1408131728020.2503@scott.cbbs.org>
References: <alpine.DEB.2.10.1408131728020.2503@scott.cbbs.org>
Message-ID: <CAFa_cKkV0Zj3ubyTSHJm9AhOLOYc9=qMatyenbbVJbUozTdYeQ@mail.gmail.com>

On Wed, Aug 13, 2014 at 10:16 PM, Len Ovens <len at ovenwerks.net> wrote:

>
> Is it just me? Has anyone else looked at pitch bend events on the Ardour
> MIDI Tracer? Quick test:
>
> ================================================
>  - edit->Preferences->Control surfaces.
>  - select both enabled and feedback for generic MIDI.
>  - double click on it and select bcf2000 with mackie protocol
>  - open an external midi monitor (using qmidiroute here) and connect it to
>         Ardours MIDI control out.
>  - also open Ardours MIDI tracer window and connect it to the same output.
>  (Now qmidiroute will be in decimal and MIDI Tracer is hex..)
>  - use the mouse to move the gain up and down on channel one with an audio
> track.
> =================================================
>
> I am seeing numbers that make sense on qmidiroute -8013 to 8177, but midi
> Tracer only shows two hex digits ever, shouldn't there be four (maybe three
> to handle 10bits)? It looks like I am seeing only the LSB (I am not sure
> about this as the lowest nyble should be 0). My about box says 3.5.308 so
> maybe this has been fixed since then. 3.5.380 seems to be were things are
> at now. I have looked through the change logs here:
> https://ardour.org/whatsnew.html for 3.5.380, (this doesn't seem to be
> included in the source package for 3.5.380) and
> https://ardour.org/news/3.5.older.html which is included with the source
> package. And I don't see any mention of a fix.
>
> BTW, I am having lots of fun with my "MIDI" controler project. Built
> around a $3 USB computer keyboard from dollar store. I use two columns for
> each strip for 10 strips. Strip one for example is 1->z and 2->x. 1 is
> channel select (does nothing right now), 2 is record, q&w are pan control,
> a&z are the fader, s is solo and x is mute. up and down changes banks
>
> Ah, banks. I have bank at 5 and am using 8 channels to test. It starts on
> channel 1 to 5, bank right does 6 to 8 (good so far), then another bank
> right does channel 8. I don't know if this is right or wrong, but my
> expectation is that once my controller has even one strip outside the range
> of channels that I would no longer be able to bank right. I would also
> expect the GUI to make sure all the channels my controler can control to be
> visible. My thought is that in the same way a selected track gets a red
> border (in the mixer) perhaps a green border around the group of tracks in
> the current bank. The BCF2000, for example, does not give much indication
> visually of where the banks are at. Mine of course gives none, but I could
> make a GUI across the bottom of the screen the controler sits in front of
> that does that... it just seems redundant when the mixer is already on the
> screen just above. I do realize that the controler code at least for the
> MCP stuff is not complete and will get done when it gets done. What I have
> already is a wonderful step up.
>
> The transport is on the numeric pad.
>
> All of this could easily change as I add some encoders for pan/faders
>


 the trace shows MIDI messages. there are no 10 or 14 bit MIDI messages.
only "controllers" with 14 bit state. 14 bit state is sent as two messages
(when necessary). the tracer shows each individual message.

>
>
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From fons at linuxaudio.org  Fri Aug 15 22:07:42 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Fri, 15 Aug 2014 22:07:42 +0000
Subject: [LAU] [Bulk]  Update of zita-at1
In-Reply-To: <1408122378.838.1.camel@rocketmail.com>
References: <20140815150500.GB4669@linuxaudio.org>
 <1408122378.838.1.camel@rocketmail.com>
Message-ID: <20140815220742.GA9487@linuxaudio.org>

On Fri, Aug 15, 2014 at 07:06:18PM +0200, Ralf Mardorf wrote:

> For AT1 and Rev1 I'm missing a config file,

A preset system for zita-*** is in the works.

> most important is setting it to 100% effect signal, instead of
> the effect/dry mix.

That will be the default in future releases.
 
> REV1 IMO is a very good reverb, but IMO it only provides one character,
> good quality, very usable, but different characters would be nice.

It is good because it doesn't try to force the algorithm used
into something it can't do well.
 
-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From fons at linuxaudio.org  Fri Aug 15 22:12:56 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Fri, 15 Aug 2014 22:12:56 +0000
Subject: [LAU] [Bulk]  Update of zita-at1
In-Reply-To: <1408131240.838.3.camel@rocketmail.com>
References: <20140815150500.GB4669@linuxaudio.org>
 <1408122378.838.1.camel@rocketmail.com> <53EE5387.6010404@web.de>
 <1408131240.838.3.camel@rocketmail.com>
Message-ID: <20140815221256.GB9487@linuxaudio.org>

On Fri, Aug 15, 2014 at 09:34:00PM +0200, Ralf Mardorf wrote:
 
> Sure, I'm using the gx version of Rev1. I started with compiling
> zita-rev1 and zita-at1 and used both in combination. The original test
> was to find out, if it's possible to get talking drums, by using conga
> samples in combination with AT1 and to cloud side effects of AT1 by the
> reverb.

I don't think you will get very far with trying to use at1 to 
create 'speaking drums'. It is *not* a vocoder (some autotuners
are). You could use it retune some percussive sounds that have
a definite pitch, but that's not the same thing.

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From ralf.mardorf at rocketmail.com  Fri Aug 15 22:52:59 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sat, 16 Aug 2014 00:52:59 +0200
Subject: [LAU] [Bulk] Re:  [Bulk]  Update of zita-at1
In-Reply-To: <20140815220742.GA9487@linuxaudio.org>
References: <20140815150500.GB4669@linuxaudio.org>
 <1408122378.838.1.camel@rocketmail.com>
 <20140815220742.GA9487@linuxaudio.org>
Message-ID: <1408143179.28013.4.camel@rocketmail.com>

On Fri, 2014-08-15 at 22:07 +0000, Fons Adriaensen wrote:
> On Fri, Aug 15, 2014 at 07:06:18PM +0200, Ralf Mardorf wrote:
> 
> > For AT1 and Rev1 I'm missing a config file,
> 
> A preset system for zita-*** is in the works

That's good news :).

> It is good because it doesn't try to force the algorithm used
> into something it can't do well.

Ok, it's better to have one character that is good, than to have several
characters that sound disgusting.

On Fri, 2014-08-15 at 22:12 +0000, Fons Adriaensen wrote:
I don't think you will get very far with trying to use at1 to 
> create 'speaking drums'. It is *not* a vocoder (some autotuners
> are). You could use it retune some percussive sounds that have
> a definite pitch, but that's not the same thing.

I only tried to imitate the instrument "talking drum",
https://en.wikipedia.org/wiki/Talking_drum in a way, when it's played by
just changing the pitch a little bit, when the tension of the drumhead
is only changed minimally. Using a conga sample and a pitch bender wheel
doesn't sound good. AT1 can't do it either, but it sounds better than
using the pitch bender wheel. I didn't use MIDI to control the AT1, just
audio with a conga and bongo rhythm and let AT1 auto correct the tuning,
so sometimes it was audible when the pitch was corrected.


From len at ovenwerks.net  Sat Aug 16 00:24:51 2014
From: len at ovenwerks.net (Len Ovens)
Date: Fri, 15 Aug 2014 17:24:51 -0700 (PDT)
Subject: [LAU] Ardour MIDI tracer
In-Reply-To: <CAFa_cKkV0Zj3ubyTSHJm9AhOLOYc9=qMatyenbbVJbUozTdYeQ@mail.gmail.com>
References: <alpine.DEB.2.10.1408131728020.2503@scott.cbbs.org>
 <CAFa_cKkV0Zj3ubyTSHJm9AhOLOYc9=qMatyenbbVJbUozTdYeQ@mail.gmail.com>
Message-ID: <alpine.DEB.2.10.1408151454100.2503@scott.cbbs.org>

On Fri, 15 Aug 2014, Paul Davis wrote:

> On Wed, Aug 13, 2014 at 10:16 PM, Len Ovens <len at ovenwerks.net> wrote:
>
>       Is it just me? Has anyone else looked at pitch bend events on the
>       Ardour MIDI Tracer? Quick test:
>
>       ================================================
>       ?- edit->Preferences->Control surfaces.
>       ?- select both enabled and feedback for generic MIDI.
>       ?- double click on it and select bcf2000 with mackie protocol
>       ?- open an external midi monitor (using qmidiroute here) and connect
>       it to
>       ? ? ? ? Ardours MIDI control out.
>       ?- also open Ardours MIDI tracer window and connect it to the same
>       output.
>       ?(Now qmidiroute will be in decimal and MIDI Tracer is hex..)
>       ?- use the mouse to move the gain up and down on channel one with an
>       audio track.
>       =================================================
> 
> ?the trace shows MIDI messages. there are no 10 or 14 bit MIDI messages. only
> "controllers" with 14 bit state. 14 bit state is sent as two messages (when
> necessary). the tracer shows each individual message.

Pitch bend, which the mackie faders use, is specified in the MIDI standard 
and in the MCP spec as: Ex yy zz in one event. WHere x = channel, yy = LSB 
(7 bits) and zz = MSB (7 bits). This is one event. This the way I send 
fader info to Ardour and It is also the way Ardour sends info back to me 
when I use a mouse to move a fader. Lets compare the output of the tracer 
with qmidiroute: (using fader5 as happens)

tracer			db	qmidiroute			midi
===========================================================================
Pitch Bend chn  5 1a	-0.5	Ch  5, Pitch  4378 (111a)	e4 1a 42
crt/alt/mousewheel down
Pitch Bend chn  5 09	-0.5	Ch  5, Pitch  4361 (1109)	e4 09 42
Pitch Bend chn  5 79	-0.5	Ch  5, Pitch  4345 (10f9)	e4 79 41
There is no second midi event with the msb, this info is missing.

I am well aware that controllers are 7 bits in an event. The MIDI standard 
does double up some of them for 14 bit, but I am not aware of anyone who 
uses them.

Speaking of which... I was looking at the MIDI codes for some more upscale 
Control surfaces:
Allen & Heath iLive control surfaces
Yamaha CL series mixers

Both of these use NRPN (Non-Registered Parameter Number) for some of their 
messages. The A&H uses this because they have run out of controllers (I 
would guess) and still only get 7 bits for their faders. They use three 
events, the first one has the mixer channel (as well as the midi channel), 
the second has a code for what in the channel it controls (0x17 for fader) 
and the third is the data. Yamaha uses four events, the first two carrying 
the controller number (they have not bothered to group them the way A&H 
has) and the last two are MSB and then LSB. The wiki for NRPN says the 
controller number for the first two bytes should be 0x63 and 0x62, 
however, Yamaha has these listed backwards. As there are other typos in 
the same page, this may just be a typo too.

I do not know if the Ardour midimap binding msg="byte byte byte" would 
handle this or not. I know Jack would consider this 4 events and probably 
not accept just the first status byte with 8 running status data bytes 
following as a single event.

--
Len Ovens
www.ovenwerks.net

From tim at quitte.de  Sat Aug 16 06:45:58 2014
From: tim at quitte.de (Tim Goetze)
Date: Sat, 16 Aug 2014 08:45:58 +0200 (CEST)
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <20140813105944.GA18888@linuxaudio.org>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <3967821.5Hi4RIW6Gm@col-desktop> <20140812122604.GA13127@linuxaudio.org>
 <3276416.mTp9TkTy7n@col-desktop> <20140813105944.GA18888@linuxaudio.org>
Message-ID: <Pine.LNX.4.64.1408160835410.22241@summer.q>

[Fons Adriaensen]
>They way I've implemented this in some players uses four states:
>all combinations of [STOP, RUN] and [READY, NOTREADY]. A state 
>that includes NOTREADY is always transient, it will revert to the
>corresponding READY one as soon as possible.

I have to agree that this is a far more useful approach than jack's
current "slow-sync" and I think that for jack to become a truly
professional tool the adoption of this state model is inevitable.

Tim

From halfbeinghalfthing at gmail.com  Sat Aug 16 12:52:39 2014
From: halfbeinghalfthing at gmail.com (Robert Persson)
Date: Sat, 16 Aug 2014 14:52:39 +0200
Subject: [LAU] Trouble with Alsa MIDI connections
Message-ID: <CA+jhH7Hf+bm+NtawJaPDFh2vSm6gNdFUt+TFqJ2zopuAfvXVZQ@mail.gmail.com>

Hello,

I have two machines running the latest UbuntuStudio. On one of them MIDI is
working fine, but on the other I am having problems.

On the problem machine I can't connect either of my external MIDI keyboards
to another MIDI port while Jack is running. If I highlight the keyboard and
the other port in the QJackCtl Alsa graph and click "connect", nothing
happens. I can however connect them while Jack is inactive, and that
connection will continue working even after Jack has started again.

I am also having a problem with an application running in Wine (Reaper).
Reaper's MIDI port does not appear in the QJackCtl Alsa graph, but Reaper
shows the other MIDI ports available in Alsa in its own preferences.
However if I try to connect it to either of my external keyboards it fails,
and it does so regardless of whether Jack is running. It will however
connect to MIDI Through (which gave me a reasonable workaround when my
routing requirements were simpler than they are now).

Does anyone know what I need to do to get MIDI connections working properly?

Many thanks
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From jeremy at autostatic.com  Sat Aug 16 19:34:42 2014
From: jeremy at autostatic.com (Jeremy Jongepier)
Date: Sat, 16 Aug 2014 21:34:42 +0200
Subject: [LAU] [Bulk] Re: [Bulk] Re: Header missing to build Zita AT 1
In-Reply-To: <1408049167.32198.7.camel@rocketmail.com>
References: <1407939448.3069.4.camel@rocketmail.com>
 <20140813144514.GF18888@linuxaudio.org>
 <1407947164.4059.1.camel@rocketmail.com> <53EC9412.4090004@autostatic.com>
 <1408015108.778.33.camel@rocketmail.com>
 <1408015891.778.39.camel@rocketmail.com> <53EC9F32.2090700@gmail.com>
 <1408022993.778.44.camel@rocketmail.com> <53ECE826.1020807@gmail.com>
 <53ECFB63.3080302@autostatic.com> <1408049167.32198.7.camel@rocketmail.com>
Message-ID: <53EFB252.6060707@autostatic.com>

On 08/14/2014 10:46 PM, Ralf Mardorf wrote:
> Hi Jeremy,
> 
> if I understand correctly, the script does provide the two headers I
> own.
> 
> "unzip -j -o $sdk_zip "*aeffect*" -d $sdk_dir"
> "sed -i '69s/__cdecl/\/\/__cdecl/' $sdk_dir/aeffect.h"
> 
> But it doesn't provide audioeffect.cpp. It seems to be available here:
> 
> https://raw.githubusercontent.com/CharlesHolbrow/ChuckDelay/master/public.sdk/source/vst2.x/audioeffect.cpp
> 
> But what's about the headers:
> #include "audioeffect.h"
> #include "aeffeditor.h"
> 
> They don't fit to "*aeffect*".
> 
> Regards,
> Ralf

Hello Ralf,

The script downloads the headers I need to build Qtractor with native
Linux VST support. That's all I need.

Jeremy

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From paul at linuxaudiosystems.com  Sat Aug 16 19:53:40 2014
From: paul at linuxaudiosystems.com (Paul Davis)
Date: Sat, 16 Aug 2014 15:53:40 -0400
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <Pine.LNX.4.64.1408160835410.22241@summer.q>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <3967821.5Hi4RIW6Gm@col-desktop>
 <20140812122604.GA13127@linuxaudio.org>
 <3276416.mTp9TkTy7n@col-desktop>
 <20140813105944.GA18888@linuxaudio.org>
 <Pine.LNX.4.64.1408160835410.22241@summer.q>
Message-ID: <CAFa_cKmLkGUCjdnWExVTWMwisqymWh97x_5yE_uAMQyBfdZ8_Q@mail.gmail.com>

On Sat, Aug 16, 2014 at 2:45 AM, Tim Goetze <tim at quitte.de> wrote:

> [I have to agree that this is a far more useful approach than jack's
> current "slow-sync" and I think that for jack to become a truly
> professional tool the adoption of this state model is inevitable.
>

ignoring any details of the transport API, there isn't anything that will
make JACK a "truly professional tool".
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From tim at quitte.de  Sat Aug 16 20:49:35 2014
From: tim at quitte.de (Tim Goetze)
Date: Sat, 16 Aug 2014 22:49:35 +0200 (CEST)
Subject: [LAU] Jack transport - was - Ardour/Muse Jack tempo lock
In-Reply-To: <CAFa_cKmLkGUCjdnWExVTWMwisqymWh97x_5yE_uAMQyBfdZ8_Q@mail.gmail.com>
References: <88806505c84414d739ef85acb790d42e.squirrel@ssl.ovenwerks.net>
 <3967821.5Hi4RIW6Gm@col-desktop> <20140812122604.GA13127@linuxaudio.org>
 <3276416.mTp9TkTy7n@col-desktop> <20140813105944.GA18888@linuxaudio.org>
 <Pine.LNX.4.64.1408160835410.22241@summer.q>
 <CAFa_cKmLkGUCjdnWExVTWMwisqymWh97x_5yE_uAMQyBfdZ8_Q@mail.gmail.com>
Message-ID: <Pine.LNX.4.64.1408162232200.22241@summer.q>

[Paul Davis]
>On Sat, Aug 16, 2014 at 2:45 AM, Tim Goetze <tim at quitte.de> wrote:
>> [I have to agree that this is a far more useful approach than jack's
>> current "slow-sync" and I think that for jack to become a truly
>> professional tool the adoption of this state model is inevitable.
>
>ignoring any details of the transport API, there isn't anything that will
>make JACK a "truly professional tool".

I beg your forgiveness for my very poor choice of words, but still I
am hopeful, even certain, that this mistake of mine will not be able
to taint your habitual power of objective consideration telling you
that the current approach is inferior to Fons' proposal in a number of
ways.

Cheers, Tim

From ralf.mardorf at rocketmail.com  Sun Aug 17 02:26:58 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 17 Aug 2014 04:26:58 +0200
Subject: [LAU] [Bulk] Re: [Bulk] Re: [Bulk] Re: Header missing to build
 Zita AT 1
In-Reply-To: <53EFB252.6060707@autostatic.com>
References: <1407939448.3069.4.camel@rocketmail.com>
 <20140813144514.GF18888@linuxaudio.org>
 <1407947164.4059.1.camel@rocketmail.com> <53EC9412.4090004@autostatic.com>
 <1408015108.778.33.camel@rocketmail.com>
 <1408015891.778.39.camel@rocketmail.com> <53EC9F32.2090700@gmail.com>
 <1408022993.778.44.camel@rocketmail.com> <53ECE826.1020807@gmail.com>
 <53ECFB63.3080302@autostatic.com> <1408049167.32198.7.camel@rocketmail.com>
 <53EFB252.6060707@autostatic.com>
Message-ID: <1408242418.1011.5.camel@rocketmail.com>

On Sat, 2014-08-16 at 21:34 +0200, Jeremy Jongepier wrote:
> Hello Ralf,
> 
> The script downloads the headers I need to build Qtractor with native
> Linux VST support. That's all I need.
> 
> Jeremy

Ok, I still own the both headers. I downloaded them years ago, when I
build Qtractor from source too. The recommended VST plugin needs
something else. Fortunately the plugin isn't needed. It was a user
error, now I know how to get additional MIDI outputs for Qtractor and
more important, Fons added MIDI channel support to AT1 (for the song I'm
making now I stay with the old version of AT1).

:)


From jamesmstone at gmail.com  Sun Aug 17 07:59:49 2014
From: jamesmstone at gmail.com (James Stone)
Date: Sun, 17 Aug 2014 08:59:49 +0100
Subject: [LAU] vsts and wine startup times
In-Reply-To: <CABRv+93y_RAogmWyx080g48aYpOMD-znVM1gsm4kfgkFqJG=vg@mail.gmail.com>
References: <CABRv+93y_RAogmWyx080g48aYpOMD-znVM1gsm4kfgkFqJG=vg@mail.gmail.com>
Message-ID: <CABRv+924z_HgDscUuYW7TSpaPS9FCw6SXcf7-PZV5TgChP07nA@mail.gmail.com>

I have been playing around with a new program that does win->linux vsts
called airwave. It seems pretty nice as it does xembed properly so all
menus etc. appear correctly, and the vst performance seems acceptable
(fewer xruns than vst-bridge).

I hit a problem when first running it which turned out that it was due to
the host not waiting long enough for the vst to initialise. It hard coded a
maximum wait of 3 seconds.. Whereas on my system, wine takes around 24
seconds to start first time or around 4 seconds on subsequent times. I have
been in discussion with the dev about increasing the timeout. He changed it
to 8s which works OK apart from first run on my system.

So my questions: 1) what is the longest startup time for wine that would be
reasonable to assume?
2) is 24 seconds abnormal for a reasonably recent (last 3years - AMD
A6-3670 ) desktop system?

James
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From gnome at hawaii.rr.com  Sun Aug 17 08:04:03 2014
From: gnome at hawaii.rr.com (david)
Date: Sat, 16 Aug 2014 22:04:03 -1000
Subject: [LAU] JACK not working with Audiophile 2496 anymore
Message-ID: <53F061F3.9000008@hawaii.rr.com>

I get these messages from JACK when I start it using QJackCtl, trying to 
use my AudioPhile 2496. This is running on Debian Sid, uname -a reports 
"3.14-2-amd64 #1 SMP Debian 3.14.15-2 (2014-08-09) x86_64 GNU/Linux" 
(but I was getting the same error on kernel 3.02.4 before that.) I am a 
member of the audio group.

JACK starts and runs fine if I pick the "default" interface, but that 
doesn't play any audio through the Audiophile. (I have no idea what it's 
playing through.)

While parts of Pulse are installed, according to "killall pulse" Pulse 
audio is not running. Trying to remove the installed Pulse packages 
wanted to also remove a bunch of other apps, such as csound, GIMP, some 
KDE4 apps, etc.

I found an old thread on a forum where the poster said installing 
"dbus-python" fixed it, but the closest package I could find similar to 
that name is "python-dbus" and it's installed.

This is a fresh Debian Sid install and my first experience with Debian's 
move to systemd instead of good old reliable init, so could that be 
messing things up? Or that it's using PAM (my prior Debian experience 
didn't include PAM) and htop shows a number of (sd-pam) processes running)?

jackdmp 1.9.10
Copyright 2001-2005 Paul Davis and others.
Copyright 2004-2014 Grame.
jackdmp comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
JACK server starting in realtime mode with priority 10
self-connect-mode is "Don't restrict self connect requests"
audio_reservation_init
Acquire audio card Audio2
creating alsa driver ... 
hw:M2496|hw:M2496|512|2|48000|0|0|nomon|swmeter|-|32bit
configuring for 48000Hz, period = 512 frames (10.7 ms), buffer = 2 periods
ALSA: final selected sample format for capture: 32bit integer little-endian
ALSA: use 2 periods for capture
ALSA: final selected sample format for playback: 32bit integer little-endian
ALSA: use 2 periods for playback
21:08:30.707 Could not connect to JACK server as client. - Overall 
operation failed. - Server communication error. Please check the 
messages window for more info.
JackPosixProcessSync::LockedTimedWait error usec = 5000000 err = 
Connection timed out
Driver is not running
Cannot create new client
Cannot read socket fd = 14 err = Success
CheckRes error
JackSocketClientChannel read fail
Cannot open qjackctl client
ALSA: poll time out, polled for 15999022 usecs
JackAudioDriver::ProcessAsync: read error, stopping...
21:10:14.445 JACK is stopping...
Jack main caught signal 15
Released audio card Audio2
audio_reservation_finish
21:10:14.463 JACK was stopped successfully.
21:10:14.463 Post-shutdown script...
21:10:14.463 killall jackd
jackd: no process found
21:10:14.879 Post-shutdown script terminated with exit status=256.

aplay -l lists the following audio devices:
**** List of PLAYBACK Hardware Devices ****
card 0: SB [HDA ATI SB], device 0: VT1828S Analog [VT1828S Analog]
   Subdevices: 1/1
   Subdevice #0: subdevice #0
card 0: SB [HDA ATI SB], device 1: VT1828S Digital [VT1828S Digital]
   Subdevices: 1/1
   Subdevice #0: subdevice #0
card 0: SB [HDA ATI SB], device 2: VT1828S Alt Analog [VT1828S Alt Analog]
   Subdevices: 1/1
   Subdevice #0: subdevice #0
card 1: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0]
   Subdevices: 1/1
   Subdevice #0: subdevice #0
card 2: M2496 [M Audio Audiophile 24/96], device 0: ICE1712 multi 
[ICE1712 multi]
   Subdevices: 1/1
   Subdevice #0: subdevice #0

These problems originally began when I had to reset the BIOS, which 
re-enabled the motherboard audio. I tried disabling the motherboard 
audio, but then ALSA wouldn't load at all.

My replacement desktop system is too modern to support such old 
equipment as the Audiophile, it would be nice to have it working in the 
old desktop system where it worked fine for many years, but now no 
longer works.

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From ralf.mardorf at rocketmail.com  Sun Aug 17 10:25:10 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 17 Aug 2014 12:25:10 +0200
Subject: [LAU] [Bulk]  JACK not working with Audiophile 2496 anymore
In-Reply-To: <53F061F3.9000008@hawaii.rr.com>
References: <53F061F3.9000008@hawaii.rr.com>
Message-ID: <1408271110.1011.14.camel@rocketmail.com>

At first I would clean the card configurations

sudo -i
mv /etc/asound.conf /root/
mv ~/.asoundrc /root/
mv /etc/modprobe.d/alsa-base.conf /root/

and after that add a

/etc/modprobe.d/alsa-base.conf

with this content only:

options snd slots=snd_ice1712

Reboot and then start jackd again, using hw:0.


From ralf.mardorf at rocketmail.com  Sun Aug 17 10:34:03 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 17 Aug 2014 12:34:03 +0200
Subject: [LAU] [Bulk]  JACK not working with Audiophile 2496 anymore
In-Reply-To: <53F061F3.9000008@hawaii.rr.com>
References: <53F061F3.9000008@hawaii.rr.com>
Message-ID: <1408271643.1011.16.camel@rocketmail.com>

On Sat, 2014-08-16 at 22:04 -1000, david wrote:
> Trying to remove the installed Pulse packages wanted to also remove a
> bunch of other apps, such as csound, GIMP, some 
> KDE4 apps, etc.

Build an empty dummy package for the pulseaudio package only, _not_ for
libpulse and other packages.

With equivs building an empty dummy package is easy to do.
http://www.debian.org/doc/manuals/apt-howto/ch-helpers.en.html

> self-connect-mode is "Don't restrict self connect requests"

What does this message mean? Perhaps you should start jackd by command
line instead of QjackCtl.

jackd -dalsa -dhw:0 -r48000 -p256 -n2


From ralf.mardorf at rocketmail.com  Sun Aug 17 10:38:34 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 17 Aug 2014 12:38:34 +0200
Subject: [LAU] [Bulk]  JACK not working with Audiophile 2496 anymore
In-Reply-To: <53F061F3.9000008@hawaii.rr.com>
References: <53F061F3.9000008@hawaii.rr.com>
Message-ID: <1408271914.1011.18.camel@rocketmail.com>

Sorry for the pps:

In addition you should unload the driver for the unwanted card:

sudo modprobe -r snd_what_ever_the_driver_is named


From ralf.mardorf at rocketmail.com  Sun Aug 17 10:43:36 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 17 Aug 2014 12:43:36 +0200
Subject: [LAU] [Bulk]  JACK not working with Audiophile 2496 anymore
In-Reply-To: <53F061F3.9000008@hawaii.rr.com>
References: <53F061F3.9000008@hawaii.rr.com>
Message-ID: <1408272216.1011.20.camel@rocketmail.com>

Four mails aren't allowed [1] :(, but perhaps you missed to uncheck
something in the QjackCtl Setup... > Misc tab >
[ ] Enable D-Bus interface

;)


From willgodfrey at musically.me.uk  Sun Aug 17 11:12:33 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Sun, 17 Aug 2014 12:12:33 +0100
Subject: [LAU] JACK not working with Audiophile 2496 anymore
In-Reply-To: <53F061F3.9000008@hawaii.rr.com>
References: <53F061F3.9000008@hawaii.rr.com>
Message-ID: <20140817121233.43325cf8@debian>

On Sat, 16 Aug 2014 22:04:03 -1000
david <gnome at hawaii.rr.com> wrote:

> My replacement desktop system is too modern to support such old 
> equipment as the Audiophile, it would be nice to have it working in the 
> old desktop system where it worked fine for many years, but now no 
> longer works.

This is really bad news. I've been using a 2496 for ages and found it extremely
reliable. No more updates for me then :(

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From ralf.mardorf at rocketmail.com  Sun Aug 17 12:06:44 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 17 Aug 2014 14:06:44 +0200
Subject: [LAU] [Bulk] Re: JACK not working with Audiophile 2496 anymore
In-Reply-To: <20140817121233.43325cf8@debian>
References: <53F061F3.9000008@hawaii.rr.com> <20140817121233.43325cf8@debian>
Message-ID: <1408277204.4447.1.camel@rocketmail.com>

On Sun, 2014-08-17 at 12:12 +0100, Will Godfrey wrote:
> On Sat, 16 Aug 2014 22:04:03 -1000
> david <gnome at hawaii.rr.com> wrote:
> 
> > My replacement desktop system is too modern to support such old 
> > equipment as the Audiophile, it would be nice to have it working in the 
> > old desktop system where it worked fine for many years, but now no 
> > longer works.
> 
> This is really bad news. I've been using a 2496 for ages and found it extremely
> reliable. No more updates for me then :(

There still are mobos with PCI available. I own a mobo with PCI and
PCIe. There are no issues with my two _PCI_ envy24 cards from TerraTec,
but many problems with my RME HDSPe AIO _PCIe_ card. "HDSPe AIO is the
newly developed PCI Express version of the HDSP 9632." - RME

I should have bought the less expensive HDSP 9632 instead ;). For
testing purpose I installed Windows XP, my RME sound card isn't broken,
it definitively is a Linux driver issue and perhaps PCIe support on
Linux too, but maybe PCIe support is not an issue.

IOW upgrading Linux software shouldn't cause issues for a Envy24 audio
cards, just buying a new mobo without PCI makes old cards unusable.


From alan.mckay at gmail.com  Sun Aug 17 14:07:13 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Sun, 17 Aug 2014 10:07:13 -0400
Subject: [LAU] Audacity on Ubuntu 14.04 is REALLY unstable
Message-ID: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>

Hi folks,

I took Audacity out of apt and wow is it unstable.  Constantly
hanging.  At least it is able to recover most of the time after a
restart - except for once last night where I lost my LP rip.  It was
bearable before then but now I want a better solution.

Is anyone else having this problem?
Any solution in mind?
Should I pull down one of the daily builds from Audacity and try it?
Or build from source or something?

My Ubuntu is fully updated.

I was also thinking of trying CentOS7 to see how that does.

thanks,
-Alan

-- 
"Don't eat anything you've ever seen advertised on TV"
         - Michael Pollan, author of "In Defense of Food"

From willgodfrey at musically.me.uk  Sun Aug 17 14:22:28 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Sun, 17 Aug 2014 15:22:28 +0100
Subject: [LAU] Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
Message-ID: <20140817152228.0f16a0a6@debian>

On Sun, 17 Aug 2014 10:07:13 -0400
Alan McKay <alan.mckay at gmail.com> wrote:

> Hi folks,
> 
> I took Audacity out of apt and wow is it unstable.  Constantly
> hanging.  At least it is able to recover most of the time after a
> restart - except for once last night where I lost my LP rip.  It was
> bearable before then but now I want a better solution.
> 
> Is anyone else having this problem?
> Any solution in mind?
> Should I pull down one of the daily builds from Audacity and try it?
> Or build from source or something?
> 
> My Ubuntu is fully updated.
> 
> I was also thinking of trying CentOS7 to see how that does.
> 
> thanks,
> -Alan
> 
No problem with V 2.05 here. What version are you using?

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From alan.mckay at gmail.com  Sun Aug 17 14:27:01 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Sun, 17 Aug 2014 10:27:01 -0400
Subject: [LAU] Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <20140817152228.0f16a0a6@debian>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <20140817152228.0f16a0a6@debian>
Message-ID: <CAH8ZPGkXWuhGDoV_jTudyEM9g2g8zc==GuEBJu0YUPKr3e2QFw@mail.gmail.com>

On Sun, Aug 17, 2014 at 10:22 AM, Will Godfrey
<willgodfrey at musically.me.uk> wrote:
> No problem with V 2.05 here. What version are you using?

2.0.5.
And you have Ubuntu 14.04?
Crap that means my PC is suspect.  I just bought this thing a couple
of months ago due to hardware issues on my other one.

I've got 8Gigs RAM and originally found it even less stable when
running other apps especially web browsers, so I've been running it
without any of those things and still have issues (though not as many)

So maybe a HD issue?  I'll have to do a drive scan ...


-- 
"Don't eat anything you've ever seen advertised on TV"
         - Michael Pollan, author of "In Defense of Food"

From willgodfrey at musically.me.uk  Sun Aug 17 14:53:47 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Sun, 17 Aug 2014 15:53:47 +0100
Subject: [LAU] Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGkXWuhGDoV_jTudyEM9g2g8zc==GuEBJu0YUPKr3e2QFw@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <20140817152228.0f16a0a6@debian>
 <CAH8ZPGkXWuhGDoV_jTudyEM9g2g8zc==GuEBJu0YUPKr3e2QFw@mail.gmail.com>
Message-ID: <20140817155347.43bea82e@debian>

On Sun, 17 Aug 2014 10:27:01 -0400
Alan McKay <alan.mckay at gmail.com> wrote:

> On Sun, Aug 17, 2014 at 10:22 AM, Will Godfrey
> <willgodfrey at musically.me.uk> wrote:
> > No problem with V 2.05 here. What version are you using?
> 
> 2.0.5.
> And you have Ubuntu 14.04?
> Crap that means my PC is suspect.  I just bought this thing a couple
> of months ago due to hardware issues on my other one.
> 
> I've got 8Gigs RAM and originally found it even less stable when
> running other apps especially web browsers, so I've been running it
> without any of those things and still have issues (though not as many)
> 
> So maybe a HD issue?  I'll have to do a drive scan ...

Not using Ubuntu, but debian testing, which is probably a pretty good match in
this respect.

If you're having problems with other apps as well it does look like a computer
issue. Install memtest and give it a long run from boot up and see if you
shows anything. 

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From alan.mckay at gmail.com  Sun Aug 17 14:57:03 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Sun, 17 Aug 2014 10:57:03 -0400
Subject: [LAU] Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <20140817155347.43bea82e@debian>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <20140817152228.0f16a0a6@debian>
 <CAH8ZPGkXWuhGDoV_jTudyEM9g2g8zc==GuEBJu0YUPKr3e2QFw@mail.gmail.com>
 <20140817155347.43bea82e@debian>
Message-ID: <CAH8ZPGkgN-D8YQWyXmWZP_r5yu-aZhTPEVYVcVYNQgcPyCv4Rg@mail.gmail.com>

On Sun, Aug 17, 2014 at 10:53 AM, Will Godfrey
<willgodfrey at musically.me.uk> wrote:
> If you're having problems with other apps as well it does look like a computer
> issue.

Well the thing is I am not having issues with anything else.
The freeze-ups do look like issues when writing to bad sectors on the
HD - I've seen those symptoms with other apps in the past.  But
Audacity is the only thing I have trouble with and I've been running
this box for 6 months maybe now.

I might just shoot it over to another HD for the heck of it and see
what happens.


-- 
"Don't eat anything you've ever seen advertised on TV"
         - Michael Pollan, author of "In Defense of Food"

From eviltwin69 at cableone.net  Sun Aug 17 15:19:33 2014
From: eviltwin69 at cableone.net (Jan Depner)
Date: Sun, 17 Aug 2014 10:19:33 -0500
Subject: [LAU] Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
Message-ID: <1408288773.18721.2.camel@eviltwin>

On Sun, 2014-08-17 at 10:07 -0400, Alan McKay wrote:
> Hi folks,
> 
> I took Audacity out of apt and wow is it unstable.  Constantly
> hanging.  At least it is able to recover most of the time after a
> restart - except for once last night where I lost my LP rip.  It was
> bearable before then but now I want a better solution.
> 
> Is anyone else having this problem?
> Any solution in mind?
> Should I pull down one of the daily builds from Audacity and try it?
> Or build from source or something?
> 
> My Ubuntu is fully updated.
> 
> I was also thinking of trying CentOS7 to see how that does.
> 
> thanks,
> -Alan
> 

I'm running Ubuntu 14.04 (fully updated) and Audacity 2.0.5 here.  No
problems at all.

Cheers,
Jan




From alan.mckay at gmail.com  Sun Aug 17 22:26:17 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Sun, 17 Aug 2014 18:26:17 -0400
Subject: [LAU] Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <1408288773.18721.2.camel@eviltwin>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
Message-ID: <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>

Nope, replacing the HD did not fix it.

So what's next?

From willgodfrey at musically.me.uk  Sun Aug 17 23:11:14 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Mon, 18 Aug 2014 00:11:14 +0100
Subject: [LAU] Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
Message-ID: <20140818001114.2e566efd@debian>

On Sun, 17 Aug 2014 18:26:17 -0400
Alan McKay <alan.mckay at gmail.com> wrote:

> Nope, replacing the HD did not fix it.
> 
> So what's next?

As I said, use memtest. This sound much more like a memory issue than anything
else.

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From alan.mckay at gmail.com  Sun Aug 17 23:13:26 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Sun, 17 Aug 2014 19:13:26 -0400
Subject: [LAU] Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <20140818001114.2e566efd@debian>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
Message-ID: <CAH8ZPGm_Kz4wov4u3bcmFARwOs+S4TgXJvyJLf6SsShrpzWksQ@mail.gmail.com>

On Sun, Aug 17, 2014 at 7:11 PM, Will Godfrey
<willgodfrey at musically.me.uk> wrote:
> As I said, use memtest. This sound much more like a memory issue than anything
> else.

OK I will - but why would I see it only with Audacity and nowhere else?


-- 
"Don't eat anything you've ever seen advertised on TV"
         - Michael Pollan, author of "In Defense of Food"

From gurusonic at gmail.com  Mon Aug 18 00:56:37 2014
From: gurusonic at gmail.com (Roger)
Date: Mon, 18 Aug 2014 10:56:37 +1000
Subject: [LAU] JACK not working with Audiophile 2496 anymore
In-Reply-To: <53F061F3.9000008@hawaii.rr.com>
References: <53F061F3.9000008@hawaii.rr.com>
Message-ID: <53F14F45.5070002@gmail.com>

On 17/08/14 18:04, david wrote:
> I get these messages from JACK when I start it using QJackCtl, trying 
> to use my AudioPhile 2496. This is running on Debian Sid, uname -a 
> reports "3.14-2-amd64 #1 SMP Debian 3.14.15-2 (2014-08-09) x86_64 
> GNU/Linux" (but I was getting the same error on kernel 3.02.4 before 
> that.) I am a member of the audio group.
>
> JACK starts and runs fine if I pick the "default" interface, but that 
> doesn't play any audio through the Audiophile. (I have no idea what 
> it's playing through.)
I had a similar issue on my siduction audio workstation install. I'm 
using Cadence rather than QJackCtl but that shouldn't make any 
difference. I think Jack was trying to use hdmi audio which was card0. 
You need to make the 2496 card0 instead. I made an alsa conf file in 
/etc/modprobe.d/ with these lines:
options snd_ice1712 index=0
options snd_hda_intel index=1

Now it works perfectly. Kernel is currently 3.16-1.towo-siduction-amd64 
#1 SMP PREEMPT but 3.15 worked well also.
siduction so far seems to be excellent for audio. I started building 
this dedicated to audio after several crashes with Mixbus in KXStudio 
which most likely related to Ubuntu-specific window management.

Roger

From abonnements at revolwear.com  Mon Aug 18 05:03:32 2014
From: abonnements at revolwear.com (Max)
Date: Mon, 18 Aug 2014 14:03:32 +0900
Subject: [LAU] jack: interface selection has no effect
In-Reply-To: <alpine.DEB.2.02.1408111941010.23832@scott.cbbs.org>
References: <53E512BD.8070107@revolwear.com>
 <alpine.DEB.2.02.1408091147310.23832@scott.cbbs.org>
 <53E809EE.70009@revolwear.com>
 <alpine.DEB.2.02.1408111153150.23832@scott.cbbs.org>
 <53E97D2C.4000605@revolwear.com>
 <alpine.DEB.2.02.1408111941010.23832@scott.cbbs.org>
Message-ID: <53F18924.4090109@revolwear.com>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

I figured it out.
purge qjackctl
install Cadence from the kxstudio repository.

That qjackctl is oblivious about that is has actually no control over
jack any more puzzles me. I will not miss it.


On 08/12/2014 11:52 AM, Len Ovens wrote:
> On Tue, 12 Aug 2014, Max wrote:
> 
>> On 08/12/2014 04:41 AM, Len Ovens wrote:
>>> On Mon, 11 Aug 2014, Max wrote:
>>>> On 08/10/2014 03:53 AM, Len Ovens wrote:
>>>>> On Sat, 9 Aug 2014, Max wrote:
>>>>> 
>>>>>> selecting a different interface in qjackctl has no
>>>>>> effect. i can't use the external soundcard, because jack
>>>>>> stays always with the internal one. any ideas what causes
>>>>>> this symptom?
>>>>>> 
>>>>>> i am running pulse on top of jack like this: 
>>>>>> http://trac.jackaudio.org/wiki/WalkThrough/User/PulseOnJack
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1

iEYEARECAAYFAlPxiSQACgkQ3EB7kzgMM6JIaACfYYkXj/PtqXF3ggcvCRVM7zXL
PysAnRNHvUYDKkAkxiyLyJhebXBuEfSa
=3Hnl
-----END PGP SIGNATURE-----

From murks at tuxfamily.org  Mon Aug 18 06:19:45 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Mon, 18 Aug 2014 08:19:45 +0200
Subject: [LAU] Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
Message-ID: <20140818081945.5c355f19@eeyore.mozart.uni-klu.ac.at>

On Sun, 17 Aug 2014 10:07:13 -0400
Alan McKay <alan.mckay at gmail.com> wrote:

> Hi folks,
> 
> I took Audacity out of apt and wow is it unstable.  Constantly
> hanging.  At least it is able to recover most of the time after a
> restart - except for once last night where I lost my LP rip.  It was
> bearable before then but now I want a better solution.
> 
> Is anyone else having this problem?
> Any solution in mind?
> Should I pull down one of the daily builds from Audacity and try it?
> Or build from source or something?
> 
> My Ubuntu is fully updated.
> 
> I was also thinking of trying CentOS7 to see how that does.
> 
> thanks,
> -Alan

Hi Alan,

are the crashes reproducible? Is there any pattern to the crashes?
What is your configuration (record/play through PA, ALSA, jack)?
What sort of work do you do? What sort of operations? Do you have
massive file sizes?

Regards,
Philipp

From len at ovenwerks.net  Mon Aug 18 06:56:14 2014
From: len at ovenwerks.net (Len Ovens)
Date: Sun, 17 Aug 2014 23:56:14 -0700 (PDT)
Subject: [LAU] jack: interface selection has no effect
In-Reply-To: <53F18924.4090109@revolwear.com>
References: <53E512BD.8070107@revolwear.com>
 <alpine.DEB.2.02.1408091147310.23832@scott.cbbs.org>
 <53E809EE.70009@revolwear.com>
 <alpine.DEB.2.02.1408111153150.23832@scott.cbbs.org>
 <53E97D2C.4000605@revolwear.com>
 <alpine.DEB.2.02.1408111941010.23832@scott.cbbs.org>
 <53F18924.4090109@revolwear.com>
Message-ID: <alpine.DEB.2.10.1408172351500.2503@scott.cbbs.org>

On Mon, 18 Aug 2014, Max wrote:

> I figured it out.
> purge qjackctl
> install Cadence from the kxstudio repository.
>
> That qjackctl is oblivious about that is has actually no control over
> jack any more puzzles me. I will not miss it.

I found that chmod -x jackd worked better. The debian/ubuntu jackd2 
package is broken: it ships both jackd and jackdbus in the same package. 
The package should be split.

--
Len Ovens
www.ovenwerks.net


From david at kenpro.com.au  Mon Aug 18 07:16:39 2014
From: david at kenpro.com.au (David)
Date: Mon, 18 Aug 2014 17:16:39 +1000
Subject: [LAU] Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <20140818081945.5c355f19@eeyore.mozart.uni-klu.ac.at>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <20140818081945.5c355f19@eeyore.mozart.uni-klu.ac.at>
Message-ID: <53F1A857.1070505@kenpro.com.au>


On 18/08/14 16:19, Philipp ?berbacher wrote:
> On Sun, 17 Aug 2014 10:07:13 -0400
> Alan McKay <alan.mckay at gmail.com> wrote:
>
>> Hi folks,
>>
>> I took Audacity out of apt and wow is it unstable.  Constantly
>> hanging.  At least it is able to recover most of the time after a
>> restart - except for once last night where I lost my LP rip.  It was
>> bearable before then but now I want a better solution.


I've used Audacity a little on Studio Ubuntu 14.04 with all updates. 
Nothing problematic there, but I don't use it very hard. Just a little 
recording from Microphone and general editing. I used to get occasional 
hangs on 12.04 but they always recovered on restart. Sorry I can't be 
more help. Studio is slightly different from standard Ubuntu - which one 
are you using.


>>
>> Is anyone else having this problem?
>> Any solution in mind?
>> Should I pull down one of the daily builds from Audacity and try it?
>> Or build from source or something?
>>
>> My Ubuntu is fully updated.
>>
>> I was also thinking of trying CentOS7 to see how that does.
>>
>> thanks,
>> -Alan
> Hi Alan,
>
> are the crashes reproducible? Is there any pattern to the crashes?
> What is your configuration (record/play through PA, ALSA, jack)?
> What sort of work do you do? What sort of operations? Do you have
> massive file sizes?
>
> Regards,
> Philipp
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

-- 
David McQuire
0418 310312


From ralf.mardorf at rocketmail.com  Mon Aug 18 08:22:54 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Mon, 18 Aug 2014 10:22:54 +0200
Subject: [LAU] [Bulk] Re:  Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGkXWuhGDoV_jTudyEM9g2g8zc==GuEBJu0YUPKr3e2QFw@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <20140817152228.0f16a0a6@debian>
 <CAH8ZPGkXWuhGDoV_jTudyEM9g2g8zc==GuEBJu0YUPKr3e2QFw@mail.gmail.com>
Message-ID: <1408350174.2320.1.camel@rocketmail.com>

On Sun, 2014-08-17 at 10:27 -0400, Alan McKay wrote:
> I've got 8Gigs RAM and originally found it even less stable when
> running other apps especially web browsers

Look out for BIOS updates, or what ever the BIOS is named nowadays (EFI,
UEFI). Build latest stable vanilla kernel, use a real Linux distro
instead of this evil spyware distro
( https://www.gnu.org/philosophy/ubuntu-spyware.html ).

My mobo years ago needed a few BIOS and kernel upgrades.

-- 
http://iknowwhereyourcatlives.com/


From ralf.mardorf at rocketmail.com  Mon Aug 18 08:42:16 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Mon, 18 Aug 2014 10:42:16 +0200
Subject: [LAU] [Bulk] Re:  Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <20140818001114.2e566efd@debian>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
Message-ID: <1408351336.2320.6.camel@rocketmail.com>

On Mon, 2014-08-18 at 00:11 +0100, Will Godfrey wrote:
> On Sun, 17 Aug 2014 18:26:17 -0400
> Alan McKay <alan.mckay at gmail.com> wrote:
> 
> > Nope, replacing the HD did not fix it.
> > 
> > So what's next?
> 
> As I said, use memtest. This sound much more like a memory issue than anything
> else.

Use memtest from a memtest live media, I often experienced issues when
using memtest from Ubuntu and Debian installs.

-- 
http://iknowwhereyourcatlives.com/


From gnome at hawaii.rr.com  Mon Aug 18 08:46:59 2014
From: gnome at hawaii.rr.com (david)
Date: Sun, 17 Aug 2014 22:46:59 -1000
Subject: [LAU] JACK not working with Audiophile 2496 anymore
In-Reply-To: <53F14F45.5070002@gmail.com>
References: <53F061F3.9000008@hawaii.rr.com> <53F14F45.5070002@gmail.com>
Message-ID: <53F1BD83.4090702@hawaii.rr.com>

On 08/17/2014 02:56 PM, Roger wrote:
> On 17/08/14 18:04, david wrote:
>> I get these messages from JACK when I start it using QJackCtl,
>> trying to use my AudioPhile 2496. This is running on Debian Sid,
>> uname -a reports "3.14-2-amd64 #1 SMP Debian 3.14.15-2 (2014-08-09)
>> x86_64 GNU/Linux" (but I was getting the same error on kernel
>> 3.02.4 before that.) I am a member of the audio group.
>>
>> JACK starts and runs fine if I pick the "default" interface, but
>> that doesn't play any audio through the Audiophile. (I have no idea
>> what it's playing through.)
> I had a similar issue on my siduction audio workstation install. I'm
>  using Cadence rather than QJackCtl but that shouldn't make any
> difference. I think Jack was trying to use hdmi audio which was
> card0. You need to make the 2496 card0 instead. I made an alsa conf
> file in /etc/modprobe.d/ with these lines: options snd_ice1712
> index=0 options snd_hda_intel index=1
>
> Now it works perfectly. Kernel is currently
> 3.16-1.towo-siduction-amd64 #1 SMP PREEMPT but 3.15 worked well
> also. siduction so far seems to be excellent for audio. I started
> building this dedicated to audio after several crashes with Mixbus in
> KXStudio which most likely related to Ubuntu-specific window
> management.

Thanks, that puts the cards in the right order ... will see if it
addresses my problem. Specifically selecting the Audiophile in QJackCtl
doesn't make it work, so I don't think it's because JACK is trying to
use audio device 0.

Anyway, it didn't fix the problem.

I tried this suggestion from Ralf:

> Four mails aren't allowed [1] , but perhaps you missed to uncheck
> something in the QjackCtl Setup... > Misc tab > [ ] Enable D-Bus
> interface

Enabling that option and restarting QJackCtl informs me:
ERROR: Driver is not running.
ERROR: Cannot open client name = dbusapi
ERROR: Failed to create dbusapi jack client

Selecting the HW:M2496 option from QJackCtl's dropdown and starting JACK
gave me this message:

DBUS: Jack server could not be started.

I never had to have DBUS enabled under QJackCtl's Misc tab to make the
2496 work before.

Also from Ralf:

>> self-connect-mode is "Don't restrict self connect requests"
>
> What does this message mean?

Haven't a clue. Don't particularly recall noticing it before.

> Perhaps you should start jackd by command line instead of QjackCtl.
>
> jackd -dalsa -dhw:0 -r48000 -p256 -n2

Tried that. The terminal window hangs a moment, then reports
"JackAudioDriver::ProcessAsync: read error, stopping..." Trying to start
Yoshimi with that window open only adds "Driver is not running, cannot
create new client." Changing the -dhw:0 to -d:hw:M2496 gives me the same
results.

Now let's try Ralf's other suggestion:

> At first I would clean the card configurations
>
> sudo -i mv /etc/asound.conf /root/ mv ~/.asoundrc /root/ mv
> /etc/modprobe.d/alsa-base.conf /root/
>
> and after that add a
>
> /etc/modprobe.d/alsa-base.conf
>
> with this content only:
>
> options snd slots=snd_ice1712
>
> Reboot and then start jackd again, using hw:0.

Can't find any "/etc/asound.conf"; wrong file name?

There's no .asoundrc in my home directory.

There's no alsa-base.conf file in /etc/modprobe.d. There's only the
alsa.conf created per Roger's instructions earlier. Let's try renaming
that to alsa-base.conf ... well, that was interesting. Doing that leaves
me with only the built-in devices: HDMI (card 0) and HDA ATI SB (card
1). Let's go back to what it was called before, at least the Audiophile
appeared on the list of options.

Now let's try Ralf's version of alsa-base.conf. That brought back the
Audiophile as card 0, but the other audio devices are still listed. And
JACK still reports "read error, stopping..."

Finally, a private email from James:

> You need to change your alsa configuration to force ICE1712 driver to
> card 1. I would also disable on board sound then troubleshoot the no
> sound issue. It only complicates things to have 2 cards..

Someone mentioned an alternative to QJackCtl, Cadence? Can't find that 
in Debian Sid repository.

I see a qdbus and a qdbus-qt5 package in the repository. qdbus is 
installed. Do I need to add the qdbus-qt5 package? The libqd5dbus5 
package IS installed ... Trying to remove that lib offers to remove a 
bunch of other QT5 libs, along with Luminance HDR graphics app.

Lets add the qdbus-qt5 package ... well, JACK still won't start from the 
command line when I specify the Audiophile, which aplay -l lists as card 0.

When I had the onboard sound disabled before (via BIOS), ALSA would not 
start at all. Let's try that again, just in case it simplifies things. 
OK, now that aplay shows me the Audiophile as card 0 and HDMI as card 1. 
Still doesn't work. Tried with default, Audiophile and ice_1712. They 
all report the same problem.

For further test, booted an old Ubuntu Studio 12.10 CD I have here and 
the Audiophile worked just fine. So I copied the 
/etc/modprobe.d/alsa-base.conf file from the running US12 environment to 
my system partition, rebooted, and QJackCtl starts jack with no problems 
and Yoshimi is currently making lovely sounds through the Audiophile. 
The only audio devices that appear are the Audiophile (card 0) and HDMI 
(card 1).

So we can call this one fixed. Thanks for the info and help!

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From dj_kaza at hotmail.com  Mon Aug 18 09:42:11 2014
From: dj_kaza at hotmail.com (Kaza Kore)
Date: Mon, 18 Aug 2014 09:42:11 +0000
Subject: [LAU] JACK not working with Audiophile 2496 anymore
In-Reply-To: <53F1BD83.4090702@hawaii.rr.com>
References: <53F061F3.9000008@hawaii.rr.com>
 <53F14F45.5070002@gmail.com>,<53F1BD83.4090702@hawaii.rr.com>
Message-ID: <DUB113-W4442670A1DBD8A2F61CA3D8FD40@phx.gbl>

...> So I copied the 
> /etc/modprobe.d/alsa-base.conf file from the running US12 environment to 
> my system partition, rebooted, and QJackCtl starts jack with no problems 
> and Yoshimi is currently making lovely sounds through the Audiophile. 
> The only audio devices that appear are the Audiophile (card 0) and HDMI 
> (card 1).
> 


Maybe paste your current, working contents of /etc/modprobe.d/alsa-base.conf here so if anybody else is having a similar problem they can see what worked for you :)

Dale.
 		 	   		  
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From alan.mckay at gmail.com  Mon Aug 18 10:14:42 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Mon, 18 Aug 2014 06:14:42 -0400
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <1408351336.2320.6.camel@rocketmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
Message-ID: <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>

memtest ran overnight with no issues. 4 full passes.

i'll send another email later today with more details that some are
asking for ... getting myself underway right now

From ralf.mardorf at rocketmail.com  Mon Aug 18 10:30:47 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Mon, 18 Aug 2014 12:30:47 +0200
Subject: [LAU] [Bulk] Re: JACK not working with Audiophile 2496 anymore
In-Reply-To: <DUB113-W4442670A1DBD8A2F61CA3D8FD40@phx.gbl>
References: <53F061F3.9000008@hawaii.rr.com> <53F14F45.5070002@gmail.com>
 ,<53F1BD83.4090702@hawaii.rr.com>
 <DUB113-W4442670A1DBD8A2F61CA3D8FD40@phx.gbl>
Message-ID: <1408357847.28874.0.camel@rocketmail.com>

On Mon, 2014-08-18 at 09:42 +0000, Kaza Kore wrote:
> Maybe paste your current, working contents
> of /etc/modprobe.d/alsa-base.conf here so if anybody else is having a
> similar problem they can see what worked for you :)

And/or add a   [solved]   to the subject.

-- 
http://iknowwhereyourcatlives.com/


From alan.mckay at gmail.com  Mon Aug 18 10:54:45 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Mon, 18 Aug 2014 06:54:45 -0400
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
Message-ID: <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>

OK here is more detail on what I do.

All I am doing is ripping some old LPs.   Very simple stuff.  Then
splitting it up into MP3s

Whenever I am ripping if I am using a web browser at the same time,
what happens is audacity stops recording as though someone had hit the
"stop" button.  It does not hand or anything, and I can hit "record"
again and it goes (but it still screws up my rip of course)

Then when I am done of that an processing, a really simple one.  I
zoom into the area between songs then I want to play that short clip
to see exactly where one song ends and the other begins, and maybe
figure out whether there is a few seconds of silence I can cut out.
So I hit "play" and then when I hit "stop" at the other end, it hangs
solid and eventually gives me the option to kill or wait.  I kill it.
That one can sometimes happen the very first thing into audacity
before I've done any other editing.

As for settings - I think all defaults but I'll go through that later.

I am using regular Ubuntu not studio.

No massive files - just the length of an LP.  Usually 45 minutes but
it can happen at the 2 minute mark.

From murks at tuxfamily.org  Mon Aug 18 12:21:56 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Mon, 18 Aug 2014 14:21:56 +0200
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
Message-ID: <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>

On Mon, 18 Aug 2014 06:54:45 -0400
Alan McKay <alan.mckay at gmail.com> wrote:

> OK here is more detail on what I do.
> 
> All I am doing is ripping some old LPs.   Very simple stuff.  Then
> splitting it up into MP3s
> 
> Whenever I am ripping if I am using a web browser at the same time,
> what happens is audacity stops recording as though someone had hit the
> "stop" button.  It does not hand or anything, and I can hit "record"
> again and it goes (but it still screws up my rip of course)
> 
> Then when I am done of that an processing, a really simple one.  I
> zoom into the area between songs then I want to play that short clip
> to see exactly where one song ends and the other begins, and maybe
> figure out whether there is a few seconds of silence I can cut out.
> So I hit "play" and then when I hit "stop" at the other end, it hangs
> solid and eventually gives me the option to kill or wait.  I kill it.
> That one can sometimes happen the very first thing into audacity
> before I've done any other editing.
> 
> As for settings - I think all defaults but I'll go through that later.
> 
> I am using regular Ubuntu not studio.
> 
> No massive files - just the length of an LP.  Usually 45 minutes but
> it can happen at the 2 minute mark.

My guess is that it has to do with the audio backend. I had problems
with another program that also uses portaudio. That one would not work
properly with ALSA but only with pulseaudio.

If this is with pulse, turn it off, use plain alsa, see whether this
still happens and vice versa.
You can also try jack. Audacity uses jack in a very weird way, as soon
as you roll audacity autoconnects to the first outputs it finds and
disconnects once you stop rolling, but if that is compatible with your
general setup than that's OK. The benefit of running jack in this case
would be that nothing else can access this soundcard and interfere.

Regards,
Philipp

From nabob_cd at yahoo.com  Mon Aug 18 11:32:49 2014
From: nabob_cd at yahoo.com (Menno)
Date: Mon, 18 Aug 2014 04:32:49 -0700 (PDT)
Subject: [LAU] Delta1010: S/PDIF out has sync but no audio
Message-ID: <1408361569528-92201.post@n7.nabble.com>

Hi.

i use Kubuntu 14.04. (KXStudio in fact)
i have a M-Audio Delta 1010 soundcard. I see that my digital amp gets a
syncsignal, when i use VLC for example, but i hear no sound.

Is there a solution for this?

thanks!



--
View this message in context: http://linux-audio.4202.n7.nabble.com/Delta1010-S-PDIF-out-has-sync-but-no-audio-tp92201.html
Sent from the linux-audio-user mailing list archive at Nabble.com.

From alan.mckay at gmail.com  Mon Aug 18 12:44:31 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Mon, 18 Aug 2014 08:44:31 -0400
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
Message-ID: <CAH8ZPGmRGHN2kt+fYmyjVcNvN0x__Kei+V68XtHC31oR0qzB=g@mail.gmail.com>

It is the sound card built into my Gigabyte mobo and all defaults from
there.   I do not fully understand all that other stuff you said but I will
look into it when I am back home
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From len at ovenwerks.net  Mon Aug 18 13:58:33 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 18 Aug 2014 06:58:33 -0700 (PDT)
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian> <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
Message-ID: <alpine.DEB.2.10.1408180646130.2503@scott.cbbs.org>

On Mon, 18 Aug 2014, Alan McKay wrote:

> Whenever I am ripping if I am using a web browser at the same time,
> what happens is audacity stops recording as though someone had hit the
> "stop" button.  It does not hand or anything, and I can hit "record"
> again and it goes (but it still screws up my rip of course)

Control problem.

> Then when I am done of that an processing, a really simple one.  I
> zoom into the area between songs then I want to play that short clip
> to see exactly where one song ends and the other begins, and maybe
> figure out whether there is a few seconds of silence I can cut out.
> So I hit "play" and then when I hit "stop" at the other end, it hangs
> solid and eventually gives me the option to kill or wait.  I kill it.
> That one can sometimes happen the very first thing into audacity
> before I've done any other editing.

I have done that and not had a problem. Again this is a control problem - 
something bad happens in the control of the program.

> I am using regular Ubuntu not studio.

So that would be the new (still experimental IMO) unity desktop with it's 
ever changing interface that is slowly moving from X to MIR. You don't 
have to use studio, but have you tried xubuntu or kubuntu? Both of these 
desktops seem to have no problems with audacity. Studio, which I am using 
uses XFCE and kubunu, which my wife uses, is KDE. My wife uses audacity as 
her "DAW"  :)  so she has done lots of starts and stops with no problem.

Also look for the next release of audacity in the next month or so.

--
Len Ovens
www.ovenwerks.net


From alan.mckay at gmail.com  Mon Aug 18 14:18:13 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Mon, 18 Aug 2014 10:18:13 -0400
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <alpine.DEB.2.10.1408180646130.2503@scott.cbbs.org>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <alpine.DEB.2.10.1408180646130.2503@scott.cbbs.org>
Message-ID: <CAH8ZPGm0bnVf2E7t5q_8rH3w+uOK+F1hes1EfcN4W5GWgJKoXA@mail.gmail.com>

On Mon, Aug 18, 2014 at 9:58 AM, Len Ovens <len at ovenwerks.net> wrote:
> I have done that and not had a problem. Again this is a control problem -
> something bad happens in the control of the program.

Not sure what you are saying ... that I'm screwing something up in how
I'm using it?

> So that would be the new (still experimental IMO) unity desktop with it's
> ever changing interface that is slowly moving from X to MIR.

Yeah I should install the old desktop and try it - I'll do that.  I
had been doing that for some time with Ubuntu but finally gave in and
started using the new one. I use it here at work too.  It really is
not that bad.  Though at home I get an error every time I log in so it
does not like something about my hardware I guess.

I'll install the traditional desktop and try it.



-- 
"Don't eat anything you've ever seen advertised on TV"
         - Michael Pollan, author of "In Defense of Food"

From fons at linuxaudio.org  Mon Aug 18 14:34:58 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Mon, 18 Aug 2014 14:34:58 +0000
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
Message-ID: <20140818143458.GA26987@linuxaudio.org>

On Mon, Aug 18, 2014 at 02:21:56PM +0200, Philipp ?berbacher wrote:

> You can also try jack. Audacity uses jack in a very weird way, as soon
> as you roll audacity autoconnects to the first outputs it finds and
> disconnects once you stop rolling,

That's only one of the many apps that claim to support Jack but
get it completely wrong. In many cases, but not always, portaudio
is to blame. 

All that after ten years of Jack. IMHO it's time for some experiments
in ballistics, using rotten tomatos. 

The wole point of using Jack is to allow the user to make arbitrary
connections. Any way of 'supporting' Jack that does not allow this
gets it wrong. Even if connecting to the first two physical outputs
or something similar covers 99% of use cases. Because to do that
you don't need Jack at all. If users want Jack it's because their
requirements are in the remaining 1%. 

No doubt someone will reply 'just submit a patch'. That's easier
said than done. In almost all cases I've investigated it's not
just the 'audio interface' part of an app that needs changes, not
even if that is well abstracted as it usually is. In many cases the
logic that gets it wrong is hidden deeply inside the app, and fixing
it will affect more or less everything, including the user interface.
Any attempt to do that would require knowing the entire app inside
out. Only the original authors can do that in reasonable time. Even
if I were to submit a patch to e.g. Audacity, VLC or fldigi that would
really fix the way those broken apps handle Jack, there is zero chance
that it would be accepted, because it would be very invasive.

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From alan.mckay at gmail.com  Mon Aug 18 14:40:53 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Mon, 18 Aug 2014 10:40:53 -0400
Subject: [LAU] Ubuntu Studio (was: Audacity unstable)
Message-ID: <CAH8ZPGm4vw5hyxo1QF7Pi-gxwjhtdhPCnLU-DZ4aOVrteOyK0Q@mail.gmail.com>

Hey folks, I'm thinking I might just install Ubuntu Studio tonight and
run with that.

The website does not seem to tell me the essential differences from Ubuntu.

Which desktop is used?
Are there any other major differences I'll notice aside from this?

thanks,
-Alan

-- 
"Don't eat anything you've ever seen advertised on TV"
         - Michael Pollan, author of "In Defense of Food"

From len at ovenwerks.net  Mon Aug 18 14:43:21 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 18 Aug 2014 07:43:21 -0700 (PDT)
Subject: [LAU] Delta1010: S/PDIF out has sync but no audio
In-Reply-To: <1408361569528-92201.post@n7.nabble.com>
References: <1408361569528-92201.post@n7.nabble.com>
Message-ID: <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>

On Mon, 18 Aug 2014, Menno wrote:

> Hi.
>
> i use Kubuntu 14.04. (KXStudio in fact)
> i have a M-Audio Delta 1010 soundcard. I see that my digital amp gets a
> syncsignal, when i use VLC for example, but i hear no sound.
>
> Is there a solution for this?

This is one of those we have to ask a bunch of dumb questions things  :) 
Things I expect you are doing:
- you are using jack to connect to the 1010.
- You are sending your audio out to playback_9 and playback_10
- In Mudita24 (or envy24control) you have set your Hardware settings to be 
compatible with your digital amp. Professional for AES3 and consumer for 
s/pdif.
- if consumer, the copyright part is as open as possible: Copy permitted 
and original.
- If AES3, Audio is selected.
- You are sending a sample rate your digital amp can deal with (48k is 
standard).

I do use s/pdif in with no problem, but not the out as I have nothing to 
send it to.

--
Len Ovens
www.ovenwerks.net


From jamesmstone at gmail.com  Mon Aug 18 14:45:21 2014
From: jamesmstone at gmail.com (James Stone)
Date: Mon, 18 Aug 2014 15:45:21 +0100
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <20140818143458.GA26987@linuxaudio.org>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
Message-ID: <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>

On Mon, Aug 18, 2014 at 3:34 PM, Fons Adriaensen <fons at linuxaudio.org> wrote:
> On Mon, Aug 18, 2014 at 02:21:56PM +0200, Philipp ?berbacher wrote:
>
>> You can also try jack. Audacity uses jack in a very weird way, as soon
>> as you roll audacity autoconnects to the first outputs it finds and
>> disconnects once you stop rolling,
>
> That's only one of the many apps that claim to support Jack but
> get it completely wrong. In many cases, but not always, portaudio
> is to blame.
>

Personally I think the way Audacity handles audio on linux is very bad
- doesn't manage to do Alsa, Jack or Pulseaudio right as far as I can
see (if I don't run jack it endlessly changes the sample rate on my
card - making lots of clicks and pops as it takes over 1 minute to
start up!). I tried discussing problems on their forums but to no
avail.

It's a shame there isnt a good alternative sample editing program (to
my knowledge).

James

From fons at linuxaudio.org  Mon Aug 18 14:57:08 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Mon, 18 Aug 2014 14:57:08 +0000
Subject: [LAU] Delta1010: S/PDIF out has sync but no audio
In-Reply-To: <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
References: <1408361569528-92201.post@n7.nabble.com>
 <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
Message-ID: <20140818145708.GA5867@linuxaudio.org>

On Mon, Aug 18, 2014 at 07:43:21AM -0700, Len Ovens wrote:

> - You are sending your audio out to playback_9 and playback_10

If you know how to make vlc connect to those, please tell me...

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From ats at offog.org  Mon Aug 18 14:57:38 2014
From: ats at offog.org (Adam Sampson)
Date: Mon, 18 Aug 2014 15:57:38 +0100
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 (James Stone's message of "Mon, 18 Aug 2014 15:45:21 +0100")
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
Message-ID: <y2aoavhc0bh.fsf@cartman.at.offog.org>

James Stone <jamesmstone at gmail.com> writes:

> It's a shame there isnt a good alternative sample editing program (to
> my knowledge).

I guess it depends on what you want it to do. I used to use Audacity for
normalising and splitting up large audio files (e.g. lecture recordings,
digitised cassettes), but I switched to mhWaveEdit a couple of years ago
for this and have been very happy with it.

-- 
Adam Sampson <ats at offog.org>                         <http://offog.org/>

From len at ovenwerks.net  Mon Aug 18 15:03:06 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 18 Aug 2014 08:03:06 -0700 (PDT)
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <20140818143458.GA26987@linuxaudio.org>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian> <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
Message-ID: <alpine.DEB.2.10.1408180747180.2503@scott.cbbs.org>

On Mon, 18 Aug 2014, Fons Adriaensen wrote:

> On Mon, Aug 18, 2014 at 02:21:56PM +0200, Philipp ?berbacher wrote:
>
>> You can also try jack. Audacity uses jack in a very weird way, as soon
>> as you roll audacity autoconnects to the first outputs it finds and
>> disconnects once you stop rolling,
>
> That's only one of the many apps that claim to support Jack but
> get it completely wrong. In many cases, but not always, portaudio
> is to blame.

If using jack I recommend mhWaveEdit, the gui is not as nice, but jack 
works right. It is interesting, that in talking to devs in Audacity, they 
will answer about anything, but the idea of jack being open from 
application start gets ignored... not a no we won't do that or it's too 
hard, just silence. There is a note on one of their pages that says "we 
welcome feedback on jack implementation" but it is really bogus. The 
recording part of the app is not properly designed for any audio back 
end... pulse, windows, OSx, whatever. I think one needs to realize 
audacity, despite all that has been said, does not do jack. Record 
somewhere else, then use it to edit.

> No doubt someone will reply 'just submit a patch'. That's easier

This is one of those cases where Audacity->pulse->jack and 
jack->pulse->Audacity is what works best. Really, this is meant to be 
stand alone and connect direct to ALSA.

> if I were to submit a patch to e.g. Audacity, VLC or fldigi that would
> really fix the way those broken apps handle Jack, there is zero chance
> that it would be accepted, because it would be very invasive.

In my talking with the devs, I would agree.

--
Len Ovens
www.ovenwerks.net

From len at ovenwerks.net  Mon Aug 18 15:06:22 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 18 Aug 2014 08:06:22 -0700 (PDT)
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGm0bnVf2E7t5q_8rH3w+uOK+F1hes1EfcN4W5GWgJKoXA@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian> <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <alpine.DEB.2.10.1408180646130.2503@scott.cbbs.org>
 <CAH8ZPGm0bnVf2E7t5q_8rH3w+uOK+F1hes1EfcN4W5GWgJKoXA@mail.gmail.com>
Message-ID: <alpine.DEB.2.10.1408180804330.2503@scott.cbbs.org>

On Mon, 18 Aug 2014, Alan McKay wrote:

> On Mon, Aug 18, 2014 at 9:58 AM, Len Ovens <len at ovenwerks.net> wrote:
>> I have done that and not had a problem. Again this is a control problem -
>> something bad happens in the control of the program.
>
> Not sure what you are saying ... that I'm screwing something up in how
> I'm using it?

No, I am saying that Unity is messing things up. I have seen more problems 
with people trying to do audio work with unity that go away when unity 
goes away than I would care to share.

--
Len Ovens
www.ovenwerks.net


From lbracci at gmail.com  Mon Aug 18 15:07:20 2014
From: lbracci at gmail.com (Luigino Bracci)
Date: Mon, 18 Aug 2014 10:37:20 -0430
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian> <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
Message-ID: <CAOx8cMwijrDwPNW7_o5mnDB0b=VYcGeQRJc6LQssjzbssU6b9w@mail.gmail.com>

I suggest to try another audio backend with Audacity. In the past, we had
many troubles, hangups and crashes when using Audacity with Pulseaudio (the
default sound server in Ubuntu); the problems stopped when we switched to
Jackd. Unfortunely, it is not easy to use Jack in Ubuntu, because you must
disable Pulseaudio or install and configure pulseaudio-module-jack.

I prefer to use KXStudio, an Ubuntu-based distribution, because it
integrates by default Pulseaudio and Jack working together, using a custom
module called "Cadence". I work in a radio station, we have 12 workstations
with KXstudio and Audacity is full stable as a rock.

Bye.


2014-08-18 6:24 GMT-04:30 Alan McKay <alan.mckay at gmail.com>:

> OK here is more detail on what I do.
>
> All I am doing is ripping some old LPs.   Very simple stuff.  Then
> splitting it up into MP3s
>
> Whenever I am ripping if I am using a web browser at the same time,
> what happens is audacity stops recording as though someone had hit the
> "stop" button.  It does not hand or anything, and I can hit "record"
> again and it goes (but it still screws up my rip of course)
>
> Then when I am done of that an processing, a really simple one.  I
> zoom into the area between songs then I want to play that short clip
> to see exactly where one song ends and the other begins, and maybe
> figure out whether there is a few seconds of silence I can cut out.
> So I hit "play" and then when I hit "stop" at the other end, it hangs
> solid and eventually gives me the option to kill or wait.  I kill it.
> That one can sometimes happen the very first thing into audacity
> before I've done any other editing.
>
> As for settings - I think all defaults but I'll go through that later.
>
> I am using regular Ubuntu not studio.
>
> No massive files - just the length of an LP.  Usually 45 minutes but
> it can happen at the 2 minute mark.
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user
>
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From len at ovenwerks.net  Mon Aug 18 15:17:40 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 18 Aug 2014 08:17:40 -0700 (PDT)
Subject: [LAU] Ubuntu Studio (was: Audacity unstable)
In-Reply-To: <CAH8ZPGm4vw5hyxo1QF7Pi-gxwjhtdhPCnLU-DZ4aOVrteOyK0Q@mail.gmail.com>
References: <CAH8ZPGm4vw5hyxo1QF7Pi-gxwjhtdhPCnLU-DZ4aOVrteOyK0Q@mail.gmail.com>
Message-ID: <alpine.DEB.2.10.1408180806410.2503@scott.cbbs.org>

On Mon, 18 Aug 2014, Alan McKay wrote:

> Hey folks, I'm thinking I might just install Ubuntu Studio tonight and
> run with that.
>
> The website does not seem to tell me the essential differences from Ubuntu.

It is ubuntu. The only thing that is unique is the kernel is lowlatency 
(not RT) and jack is already set up to access the RT parts of the system.

> Which desktop is used?

XFCE. ubuntustudio uses the desktop from xubuntu with relatively few 
changes. The menu is customized to organize the audio apps so that when 
you open it there is not a list of apps as long as your arm.

The ubuntstudio metas will work on any ubuntu flavour (though as you have 
found unity causes problems with the apps themselves... though not 
always). My wife's computer has the studio kernel and audio apps and runs 
jackdbus as it's audio backend over kubuntu. Jack does have to have one 
file renamed and the user added to the audio group so it has RT access.


--
Len Ovens
www.ovenwerks.net


From len at ovenwerks.net  Mon Aug 18 15:45:06 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 18 Aug 2014 08:45:06 -0700 (PDT)
Subject: [LAU] Delta1010: S/PDIF out has sync but no audio
In-Reply-To: <20140818145708.GA5867@linuxaudio.org>
References: <1408361569528-92201.post@n7.nabble.com>
 <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
 <20140818145708.GA5867@linuxaudio.org>
Message-ID: <alpine.DEB.2.10.1408180831250.2503@scott.cbbs.org>

On Mon, 18 Aug 2014, Fons Adriaensen wrote:

> On Mon, Aug 18, 2014 at 07:43:21AM -0700, Len Ovens wrote:
>
>> - You are sending your audio out to playback_9 and playback_10
>
> If you know how to make vlc connect to those, please tell me...

VLC->pulse->jack or VLC->Pulse. Pulse does have an option to set the d1010 
to "Digital Stereo Duplex" (s/pdif in and out) or "Digital Stereo Output + 
Analog Stereo Input"

The pulse->jack bridge can be started with a custom number of channels 
rather than two and with no auto-connect. VLC is a desktop app, use 
desktop sound... which for most distros is pulse.

--
Len Ovens
www.ovenwerks.net


From fons at linuxaudio.org  Mon Aug 18 16:32:45 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Mon, 18 Aug 2014 16:32:45 +0000
Subject: [LAU] Delta1010: S/PDIF out has sync but no audio
In-Reply-To: <alpine.DEB.2.10.1408180831250.2503@scott.cbbs.org>
References: <1408361569528-92201.post@n7.nabble.com>
 <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
 <20140818145708.GA5867@linuxaudio.org>
 <alpine.DEB.2.10.1408180831250.2503@scott.cbbs.org>
Message-ID: <20140818163245.GA6965@linuxaudio.org>

On Mon, Aug 18, 2014 at 08:45:06AM -0700, Len Ovens wrote:
> On Mon, 18 Aug 2014, Fons Adriaensen wrote:
> 
> >On Mon, Aug 18, 2014 at 07:43:21AM -0700, Len Ovens wrote:
> >
> >>- You are sending your audio out to playback_9 and playback_10
> >
> >If you know how to make vlc connect to those, please tell me...
> 
> VLC->pulse->jack or VLC->Pulse. Pulse does have an option to set the
> d1010 to "Digital Stereo Duplex" (s/pdif in and out) or "Digital
> Stereo Output + Analog Stereo Input"

No need for PA, ALSA's Jack plugin can do the same.

But the question was how to make those connections using VLC's native
Jack support which it claims to have. 

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From gnome at hawaii.rr.com  Mon Aug 18 18:02:01 2014
From: gnome at hawaii.rr.com (david)
Date: Mon, 18 Aug 2014 08:02:01 -1000
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <alpine.DEB.2.10.1408180804330.2503@scott.cbbs.org>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian> <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <alpine.DEB.2.10.1408180646130.2503@scott.cbbs.org>
 <CAH8ZPGm0bnVf2E7t5q_8rH3w+uOK+F1hes1EfcN4W5GWgJKoXA@mail.gmail.com>
 <alpine.DEB.2.10.1408180804330.2503@scott.cbbs.org>
Message-ID: <53F23F99.3010606@hawaii.rr.com>

On 08/18/2014 05:06 AM, Len Ovens wrote:
> On Mon, 18 Aug 2014, Alan McKay wrote:
>
>> On Mon, Aug 18, 2014 at 9:58 AM, Len Ovens <len at ovenwerks.net> wrote:
>>> I have done that and not had a problem. Again this is a control
>>> problem -
>>> something bad happens in the control of the program.
>>
>> Not sure what you are saying ... that I'm screwing something up in how
>> I'm using it?
>
> No, I am saying that Unity is messing things up. I have seen more
> problems with people trying to do audio work with unity that go away
> when unity goes away than I would care to share.

Unity affects others apps badly, too. My wife used JPilot with her Palm 
PDA, but under Unity JPilot would freeze. Under ANY OTHER DESKTOP 
ENVIRONMENT, it worked fine.

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From gnome at hawaii.rr.com  Mon Aug 18 18:07:49 2014
From: gnome at hawaii.rr.com (david)
Date: Mon, 18 Aug 2014 08:07:49 -1000
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian> <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
Message-ID: <53F240F5.8060608@hawaii.rr.com>

On 08/18/2014 04:45 AM, James Stone wrote:
> On Mon, Aug 18, 2014 at 3:34 PM, Fons Adriaensen <fons at linuxaudio.org> wrote:
>> On Mon, Aug 18, 2014 at 02:21:56PM +0200, Philipp ?berbacher wrote:
>>
>>> You can also try jack. Audacity uses jack in a very weird way, as soon
>>> as you roll audacity autoconnects to the first outputs it finds and
>>> disconnects once you stop rolling,
>>
>> That's only one of the many apps that claim to support Jack but
>> get it completely wrong. In many cases, but not always, portaudio
>> is to blame.
>>
>
> Personally I think the way Audacity handles audio on linux is very bad
> - doesn't manage to do Alsa, Jack or Pulseaudio right as far as I can
> see (if I don't run jack it endlessly changes the sample rate on my
> card - making lots of clicks and pops as it takes over 1 minute to
> start up!). I tried discussing problems on their forums but to no
> avail.

I use Audacity on 2 different machines, both 64-bit Debian Sid, with a 
UCA-202 USB sound card on the laptop and the now-working-again (YAY!) 
Audiophile on the desktop. With or without JACK, Audacity never changes 
the sample rate as you mention above. Doesn't take a minute to start up, 
either.

I think there's some more fundamental problem with your system setup 
than Audacity. Maybe Audacity's difficulties handling audio just make it 
more sensitive to the fundamental problem than other apps.

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From willgodfrey at musically.me.uk  Mon Aug 18 18:17:17 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Mon, 18 Aug 2014 19:17:17 +0100
Subject: [LAU] Play Time
Message-ID: <20140818191717.332bb894@debian>

Amongst all the 'stuff' I still sometimes get the chance to actually produce
some music so thought I'd share a fairly recent one.

I describe it as 'Chord progressions, inversions and melodic conversations' -
hence:

http://www.musically.me.uk/music/Inversations_C.ogg

Or if you want the 'other' format change the extension to mp3

This is produced by one instance of Yoshimi, with Rosegarden and me playing on
a dumb MIDI keyboard.

I first set up a simple 1 bar loop as a click track and played the chords. Then
played the bass line to this. When doing this I always play the base an octave
up so I can hear it more clearly, then shift it down.

Disposing of the click I then played the lead parts, and finally the counter
melodies.

After a little bit of shuffling instruments I recorded the whole lot in one
pass.

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From abram.hindle at ualberta.ca  Mon Aug 18 18:45:09 2014
From: abram.hindle at ualberta.ca (Abram Hindle)
Date: Mon, 18 Aug 2014 12:45:09 -0600
Subject: [LAU] Seeking computer musicians for a survey on how do computer
 musicians develop computer musical instruments or applications?
Message-ID: <53F249B5.2050908@ualberta.ca>

Hello,

I apologize for the intrusion but the linux audio users list hosts
exactly the people we are looking for: computer musicians who code.
We're trying to figure out how software engineering research can help
computer musicians or if computer musicians need any help.

Are you a computer musician? Do you code in languages geared towards
music (e.g., Pure Data, Max MSP, Chuck, SuperCollider, etc.) or make
music in other programming languages like C, Java, C++, Javascript, etc.?

If so, we would like to hear from you! We the PIs, Gregory Burlet and
Abram Hindle, are from the Department of Computing Science at the
University of Alberta and are conducting a survey of computer musicians
to investigate how this demographic of software developers program
musical instruments or applications.

Please visit the survey invitation website
<http://webdocs.cs.ualberta.ca/%7Egburlet/musiccoders_survey.html> and
click the "I consent, take me to the survey" button to complete the
survey (if you consent). The survey will take 5 to 10 minutes.

http://webdocs.cs.ualberta.ca/%7Egburlet/musiccoders_survey.html

Thanks for your time!

Gregory Burlet and Abram Hindle
Graduate Student & Assistant Professor
Department of Computing Science
University of Alberta
CANADA



From diego.simak at gmail.com  Mon Aug 18 18:49:43 2014
From: diego.simak at gmail.com (Diego Simak)
Date: Mon, 18 Aug 2014 15:49:43 -0300
Subject: [LAU] Play Time
In-Reply-To: <20140818191717.332bb894@debian>
References: <20140818191717.332bb894@debian>
Message-ID: <CAKC-MPoQRuA-KaYYkqEePDxedzyWcCrmYq271bGti341hNPLLQ@mail.gmail.com>

Hi Will, I really liked it, beautiful mood and instrument selection.
Thank you very much for sharing it





2014-08-18 15:17 GMT-03:00 Will Godfrey <willgodfrey at musically.me.uk>:

> Amongst all the 'stuff' I still sometimes get the chance to actually
> produce
> some music so thought I'd share a fairly recent one.
>
> I describe it as 'Chord progressions, inversions and melodic
> conversations' -
> hence:
>
> http://www.musically.me.uk/music/Inversations_C.ogg
>
> Or if you want the 'other' format change the extension to mp3
>
> This is produced by one instance of Yoshimi, with Rosegarden and me
> playing on
> a dumb MIDI keyboard.
>
> I first set up a simple 1 bar loop as a click track and played the chords.
> Then
> played the bass line to this. When doing this I always play the base an
> octave
> up so I can hear it more clearly, then shift it down.
>
> Disposing of the click I then played the lead parts, and finally the
> counter
> melodies.
>
> After a little bit of shuffling instruments I recorded the whole lot in one
> pass.
>
> --
> Will J Godfrey
> http://www.musically.me.uk
> Say you have a poem and I have a tune.
> Exchange them and we can both have a poem, a tune, and a song.
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user
>
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From silvain at freeshell.de  Mon Aug 18 19:23:54 2014
From: silvain at freeshell.de (F. Silvain)
Date: Mon, 18 Aug 2014 21:23:54 +0200 (CEST)
Subject: [LAU] Play Time
In-Reply-To: <20140818191717.332bb894@debian>
References: <20140818191717.332bb894@debian>
Message-ID: <1408182119340.14718@freeshell.de>

Will,
this is a charming tune. I like the first lead and am impressed by a couple of 
the other sounds. Your command of Yoshimi is extensive.

It's good to hear, that you always find the time to make some more music and 
share it. Thank you.

Ta-ta
----
Ffanci
* Internet: http://freeshell.de/~silvain

From alan.mckay at gmail.com  Tue Aug 19 00:45:41 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Mon, 18 Aug 2014 20:45:41 -0400
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <53F240F5.8060608@hawaii.rr.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
Message-ID: <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>

OK I installed the class desktop and am now ripping while I type this
in my browser :-)

Will let you know how it goes.  If I can facebook for 5 minutes
without a crash then we are good ...

Thanks all!

From alan.mckay at gmail.com  Tue Aug 19 00:47:41 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Mon, 18 Aug 2014 20:47:41 -0400
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
Message-ID: <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>

Well that did not take long ...  it just stopped my recording.  Though
this time I got an error dialog that I've never seen before - maybe I
just need the low latency kernel?  How can I install that from regular
Ubuntu :

Latency Correction setting has caused the recorded audio to be hidden
before zero.
Audacity has brought it back to start at zero.
You may have to use the Time Shift Tool (<---> or F5) to drag the
track to the right place.

From alan.mckay at gmail.com  Tue Aug 19 01:32:26 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Mon, 18 Aug 2014 21:32:26 -0400
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
 <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
Message-ID: <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>

OK, low-latency kernel is in ... we'll see how it goes.

Now that we are messing with kernels I actually know what I am doing :-)

From alan.mckay at gmail.com  Tue Aug 19 01:46:37 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Mon, 18 Aug 2014 21:46:37 -0400
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
 <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
 <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
Message-ID: <CAH8ZPG=A4AMKNCXCKC2mOcFtvDofrBey2+mWWrHeXJb5qJV7jg@mail.gmail.com>

Well I've been ripping for 13 minutes now and heavily using my browser
... I think we have a winner on the low latency kernel.

Will let you know if I have another crash but it is looking pretty
good right now.

From alan.mckay at gmail.com  Tue Aug 19 02:08:24 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Mon, 18 Aug 2014 22:08:24 -0400
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPG=A4AMKNCXCKC2mOcFtvDofrBey2+mWWrHeXJb5qJV7jg@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
 <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
 <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
 <CAH8ZPG=A4AMKNCXCKC2mOcFtvDofrBey2+mWWrHeXJb5qJV7jg@mail.gmail.com>
Message-ID: <CAH8ZPGmFooY4eJUmrCocgTLpPEV+9mceTAyWsgUPexfjTCXyMw@mail.gmail.com>

Interesting.

So I was doing some googling to try to find out some technical details
on what is involved in the low latency kernel , and I came across this
to start with.

http://askubuntu.com/questions/126664/why-to-choose-low-latency-kernel-over-generic-or-realtime-ones

As I was reading I was curious as to my CPU power, so I opened a shell
to run "lshw".  As soon as I hit "enter" my rip stopped in the usual
fashion, but then everything on my desktop slowed right down.  My
mouse was moving in slow-motion.  I tried typing this and it was
missing letters (since rebooted).

So it seems basically that my system is underpowered, I guess.  I have
4 cores, and this is from lshw.  Maybe I need a real sound card too?


         description: CPU
          product: AMD A8-5600K APU with Radeon(tm) HD Graphics
          vendor: Advanced Micro Devices [AMD]
          physical id: 35
          bus info: cpu at 0
          version: AMD A8-5600K APU with Radeon(tm) HD Graphics
          slot: P0
          size: 1400MHz
          capacity: 3600MHz
          width: 64 bits
          clock: 100MHz

From countfuzzball at gmail.com  Tue Aug 19 04:03:21 2014
From: countfuzzball at gmail.com (Andrew C)
Date: Tue, 19 Aug 2014 05:03:21 +0100
Subject: [LAU]  Internal speaker set as default playback,
	not my usb soundcard.
Message-ID: <CAGD0fJG3zXUoDv1EBwQYD-jd7zjfSaSOWCWri6on2opT4OgQ-Q@mail.gmail.com>

Hello all,

Recently did an upgrade to Ubuntu 14.04 from 12.04. All went well except
I've encountered a very weird problem.

My usb audio is no longer the default sound card. I had to some some
trickery (made snd-usb-audio=-1 and snd-hda-intel=-2) in
/etc/modprobe.d/alsa-base.conf to get the usb card as #0 as listed by
/proc/asound/cards and then I got all sound going out through my usb card.
Fun fun fun.

Now with Ubuntu 14.04, my usb card is still listed as #0 in asound/cards,
but mplayer defaults to my internal speakers when issued 'mplayer
sound.wav'.

Also tried:
mplayer -Dalsa=hw=0.0 sound.wav
Plays through my usb soundcard

mplayer -Dalsa=hw=1.0 sound.wav
Plays through my internal speakers

mplayer -Dalsa=plughw sound.wav
Plays through my usb soundcard

mplayer -Dalsa=default sound.wav
Plays through my internal speakers.

Also I tried opening 'pavucontrol' fwiw and disabling the internal speaker.
Playing a sound file without any other options via mplayer sends the sound
through my usb soundcard.

This is really strange and perhaps I'm not entirely understanding all the
"wonderfulness" that is alsa.

Cheers,

Andrew.
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From willgodfrey at musically.me.uk  Tue Aug 19 07:06:16 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Tue, 19 Aug 2014 08:06:16 +0100
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGmFooY4eJUmrCocgTLpPEV+9mceTAyWsgUPexfjTCXyMw@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
 <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
 <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
 <CAH8ZPG=A4AMKNCXCKC2mOcFtvDofrBey2+mWWrHeXJb5qJV7jg@mail.gmail.com>
 <CAH8ZPGmFooY4eJUmrCocgTLpPEV+9mceTAyWsgUPexfjTCXyMw@mail.gmail.com>
Message-ID: <20140819080616.651b3c9b@debian>

On Mon, 18 Aug 2014 22:08:24 -0400
Alan McKay <alan.mckay at gmail.com> wrote:

> Interesting.
> 
> So I was doing some googling to try to find out some technical details
> on what is involved in the low latency kernel , and I came across this
> to start with.
> 
> http://askubuntu.com/questions/126664/why-to-choose-low-latency-kernel-over-generic-or-realtime-ones
> 
> As I was reading I was curious as to my CPU power, so I opened a shell
> to run "lshw".  As soon as I hit "enter" my rip stopped in the usual
> fashion, but then everything on my desktop slowed right down.  My
> mouse was moving in slow-motion.  I tried typing this and it was
> missing letters (since rebooted).
> 
> So it seems basically that my system is underpowered, I guess.  I have
> 4 cores, and this is from lshw.  Maybe I need a real sound card too?
> 
> 
>          description: CPU
>           product: AMD A8-5600K APU with Radeon(tm) HD Graphics
>           vendor: Advanced Micro Devices [AMD]
>           physical id: 35
>           bus info: cpu at 0
>           version: AMD A8-5600K APU with Radeon(tm) HD Graphics
>           slot: P0
>           size: 1400MHz
>           capacity: 3600MHz
>           width: 64 bits
>           clock: 100MHz

Something is really, really weird about your entire setup. You shouldn't need
to jump through all those hoops. I've never had issues like that, not even on
an old single core 32  bit athlon.

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From WillGodfrey at musically.me.uk  Tue Aug 19 07:07:00 2014
From: WillGodfrey at musically.me.uk (Will J Godfrey)
Date: Tue, 19 Aug 2014 08:07:00 +0100
Subject: [LAU] Play Time
In-Reply-To: <1408182119340.14718@freeshell.de>
References: <20140818191717.332bb894@debian> <1408182119340.14718@freeshell.de>
Message-ID: <20140819080700.1b777fe8@debian>

On Mon, 18 Aug 2014 21:23:54 +0200 (CEST)
"F. Silvain" <silvain at freeshell.de> wrote:

> Will,
> this is a charming tune. I like the first lead and am impressed by a couple of 
> the other sounds. Your command of Yoshimi is extensive.
> 
> It's good to hear, that you always find the time to make some more music and 
> share it. Thank you.
> 
> Ta-ta
> ----
> Ffanci
> * Internet: http://freeshell.de/~silvain

Thanks guys. Glad you like it :)
-- 
It wasn't me! (Well actually, it probably was)

... the hard part is not dodging what life throws at you,
but trying to catch the good bits.

From jamesmstone at gmail.com  Tue Aug 19 07:23:23 2014
From: jamesmstone at gmail.com (James Stone)
Date: Tue, 19 Aug 2014 08:23:23 +0100
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGmFooY4eJUmrCocgTLpPEV+9mceTAyWsgUPexfjTCXyMw@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
 <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
 <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
 <CAH8ZPG=A4AMKNCXCKC2mOcFtvDofrBey2+mWWrHeXJb5qJV7jg@mail.gmail.com>
 <CAH8ZPGmFooY4eJUmrCocgTLpPEV+9mceTAyWsgUPexfjTCXyMw@mail.gmail.com>
Message-ID: <CABRv+90uv1D59QH9iGYKWWfLaHjruncdVCJ_FHSzh+dLf=ZoKg@mail.gmail.com>

Have you seen this bug report:

https://bugs.launchpad.net/ubuntu/+source/audacity/+bug/355846

Seems quite similar to your problem. Have you set latency correction to 0?

I think you may be better off making long recordings with time machine.

J
On 19 Aug 2014 03:08, "Alan McKay" <alan.mckay at gmail.com> wrote:

> Interesting.
>
> So I was doing some googling to try to find out some technical details
> on what is involved in the low latency kernel , and I came across this
> to start with.
>
>
> http://askubuntu.com/questions/126664/why-to-choose-low-latency-kernel-over-generic-or-realtime-ones
>
> As I was reading I was curious as to my CPU power, so I opened a shell
> to run "lshw".  As soon as I hit "enter" my rip stopped in the usual
> fashion, but then everything on my desktop slowed right down.  My
> mouse was moving in slow-motion.  I tried typing this and it was
> missing letters (since rebooted).
>
> So it seems basically that my system is underpowered, I guess.  I have
> 4 cores, and this is from lshw.  Maybe I need a real sound card too?
>
>
>          description: CPU
>           product: AMD A8-5600K APU with Radeon(tm) HD Graphics
>           vendor: Advanced Micro Devices [AMD]
>           physical id: 35
>           bus info: cpu at 0
>           version: AMD A8-5600K APU with Radeon(tm) HD Graphics
>           slot: P0
>           size: 1400MHz
>           capacity: 3600MHz
>           width: 64 bits
>           clock: 100MHz
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user
>
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From gnome at hawaii.rr.com  Tue Aug 19 07:52:04 2014
From: gnome at hawaii.rr.com (david)
Date: Mon, 18 Aug 2014 21:52:04 -1000
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <0230eb1f-5a62-412a-82ca-e2c899dd880f@email.android.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian> <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <0230eb1f-5a62-412a-82ca-e2c899dd880f@email.android.com>
Message-ID: <53F30224.2000909@hawaii.rr.com>

Well, I've heard that Gnome 3's default configuration makes it a major 
resource hog - particularly for memory. Like KDE4. I avoid Gnome 3. Both 
it and KDE4 launch a bunch of system services that I think can really 
impact RT use.

Sorry, I don't know anything about how Fedora might be configured 
differently from Debian.

On 08/18/2014 12:12 PM, Sam Tuke wrote:
> For what its worth, audacity is unstable on Fedora with Gnome 3 as well.
> Routine crashes and corrupted recovery files make it a real headache to
> use. Maybe its the versions of the packages we're using?
>
> Sam.
>
> On 18 August 2014 20:07:49 CEST, david <gnome at hawaii.rr.com> wrote:
>
>     On 08/18/2014 04:45 AM, James Stone wrote:
>
>         On Mon, Aug 18, 2014 at 3:34 PM, Fons Adriaensen
>
>             On Mon, Aug 18, 2014 at 02:21:56PM +0200, Philipp ?berbacher
>             wrote:
>
>                 You can also try jack. Audacity uses jack in a very
>                 weird way, as soon
>                 as you roll audacity autoconnects to the first outputs
>                 it finds and
>                 disconnects once you stop rolling,
>
>
>             That's only one of the many apps that claim to support Jack but
>             get it completely wrong. In many cases, but not always,
>             portaudio
>             is to blame.
>
>         Perso nally I think the way Audacity handles audio on linux is
>         very bad
>         - doesn't manage to do Alsa, Jack or Pulseaudio right as far as
>         I can
>         see (if I don't run jack it endlessly changes the sample rate on my
>         card - making lots of clicks and pops as it takes over 1 minute to
>         start up!). I tried discussing problems on their forums but to no
>         avail.
>
>
>     I use Audacity on 2 different machines, both 64-bit Debian Sid, with a
>     UCA-202 USB sound card on the laptop and the now-working-again (YAY!)
>     Audiophile on the desktop. With or without JACK, Audacity never changes
>     the sample rate as you mention above. Doesn't take a minute to start up,
>     either.
>
>     I think there's some more fundamental problem with your system setup
>     than Audacity. Maybe Audacity's difficulties handling audio just make it
>     more sensitive to the fundamental problem than other apps.


-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From murks at tuxfamily.org  Tue Aug 19 07:57:12 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Tue, 19 Aug 2014 09:57:12 +0200
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGmFooY4eJUmrCocgTLpPEV+9mceTAyWsgUPexfjTCXyMw@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
 <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
 <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
 <CAH8ZPG=A4AMKNCXCKC2mOcFtvDofrBey2+mWWrHeXJb5qJV7jg@mail.gmail.com>
 <CAH8ZPGmFooY4eJUmrCocgTLpPEV+9mceTAyWsgUPexfjTCXyMw@mail.gmail.com>
Message-ID: <20140819095712.14f9b3bd@eeyore.mozart.uni-klu.ac.at>

On Mon, 18 Aug 2014 22:08:24 -0400
Alan McKay <alan.mckay at gmail.com> wrote:

> Interesting.
> 
> So I was doing some googling to try to find out some technical details
> on what is involved in the low latency kernel , and I came across this
> to start with.
> 
> http://askubuntu.com/questions/126664/why-to-choose-low-latency-kernel-over-generic-or-realtime-ones
> 
> As I was reading I was curious as to my CPU power, so I opened a shell
> to run "lshw".  As soon as I hit "enter" my rip stopped in the usual
> fashion, but then everything on my desktop slowed right down.  My
> mouse was moving in slow-motion.  I tried typing this and it was
> missing letters (since rebooted).
> 
> So it seems basically that my system is underpowered, I guess.  I have
> 4 cores, and this is from lshw.  Maybe I need a real sound card too?
> 
> 
>          description: CPU
>           product: AMD A8-5600K APU with Radeon(tm) HD Graphics
>           vendor: Advanced Micro Devices [AMD]
>           physical id: 35
>           bus info: cpu at 0
>           version: AMD A8-5600K APU with Radeon(tm) HD Graphics
>           slot: P0
>           size: 1400MHz
>           capacity: 3600MHz
>           width: 64 bits
>           clock: 100MHz

A low-latency or RT-kernel is absolutely not necessary to use audacity,
and I think you system is far more powerful than mine (i3). For what
you're doing any sound card should be sufficient, so that shouldn't
be it either. When your whole system slows down something is very wrong.

I still think that it could be related to your audio setup. I'm not on
Ubuntu but Audacity 2.0.5 can misbehave for me too. For example with
some audio devices I could get audacity to roll and it completely hung
as soon as I pressed stop. It seems like it does not handle errors of
that kind well.

I guess you use audacity with Pulse Audio, but you could try it with
ALSA and also with ALSA while making sure that Pulse Audio is disabled.
Also make sure you have the correct playback and recording devices set.
You can do that stuff in the preferences. There is also a 'devices
toolbar' where you can do the same basic setup.

In the preferences you can also find a buffer setting which you could
increase (default=100ms), but even if that helps there still is a
problem with your setup.

Regards,
Philipp

From raffaele.morelli at gmail.com  Tue Aug 19 07:59:35 2014
From: raffaele.morelli at gmail.com (Raffaele Morelli)
Date: Tue, 19 Aug 2014 09:59:35 +0200
Subject: [LAU] dynamic range analysis tool
In-Reply-To: <20140808174204.GA21583@linuxaudio.org>
References: <CAD4guxPW-Jms2FR=vUhO+pfkMXKb+ga_0niT_iL5UdMotxB06w@mail.gmail.com>
 <20140805131305.GA11181@linuxaudio.org>
 <CAD4guxOw9OYj0Phric9CTpywxyisdnzcM6Zqd5CHhMW9bJ6W_A@mail.gmail.com>
 <20140805144959.GB8313@linuxaudio.org>
 <CAD4guxM_-WfyqAFDBkVLKWThm7RC23UE23+zh7axM_jY7hxpZA@mail.gmail.com>
 <20140808174204.GA21583@linuxaudio.org>
Message-ID: <CAD4guxOcxxWRpsACktoFyGc872O7u=zxF3EZGBy6BhjndH4Vgw@mail.gmail.com>

2014-08-08 19:42 GMT+02:00 Fons Adriaensen <fons at linuxaudio.org>:

> On Fri, Aug 08, 2014 at 08:46:54AM +0200, Raffaele Morelli wrote:
>
> > Great tool, I finally leaved it untouched as cumulative probability it's
> > enough for the scope*.
> >
> > My 0.02?: I would suggest to add a cli option for output filename,
> leaving
> > 'ebur128-prob' as default if none is specified
>
> Yes, the cumulative plots look less sexy but will give a better idea of
> dynamic range than a histogram. The 'loudness range' as displayed by
> ebumeter and printed by ebur128 is actually computed from the (modified)
> cumulative plots - details in the paper.
>

?cumulative plot is way better to graphically compare two (or a bunch of)
audio files dynamic.

statistician here ;-)
here's a few R lines to generate "sexy" plots, the script is supposed to be
ran in the same dir where prob files are

?
require(ggplot2)
require(reshape2)

# read prob files, a shell script is being used to create and rename prob
files as outputted from ebur128
files <- list.files(".", "*.prob$")

# load prob files as tables in a list
data.list <- lapply(files, read.table)

# create an empty matrix
s<-matrix(nrow=751, ncol=length(files))

# loads cumulative probabilities into the matrix
for( i in 1:length(files)) {
    s[,i]<- data.list[[i]]$V2
}

# matrix to dataframe and proper columns name
s2 <- as.data.frame(s)
colnames(s2) <- files
s2$dbs <- data.list[[1]]$V1 # add X-axis variable (dB)

# melt the data to a long format
df2 <- melt(data = s2, id.vars = "dbs")

# plot, using the aesthetics argument 'colour'
g<-ggplot(data = df2, aes(x = dbs, y = value, colour = variable)) +
geom_line()
plot(g)
?

Next release will have the histograms as well, it's only a few chars extra
> code, and the filename as you suggest.
>

?great
/r
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From gnome at hawaii.rr.com  Tue Aug 19 08:04:48 2014
From: gnome at hawaii.rr.com (david)
Date: Mon, 18 Aug 2014 22:04:48 -1000
Subject: [LAU] [SOLVED] Re:  JACK not working with Audiophile 2496 anymore
In-Reply-To: <53F1BD83.4090702@hawaii.rr.com>
References: <53F061F3.9000008@hawaii.rr.com> <53F14F45.5070002@gmail.com>
 <53F1BD83.4090702@hawaii.rr.com>
Message-ID: <53F30520.6020008@hawaii.rr.com>

On 08/17/2014 10:46 PM, david wrote:
> On 08/17/2014 02:56 PM, Roger wrote:
>> On 17/08/14 18:04, david wrote:
>>> I get these messages from JACK when I start it using QJackCtl,
>>> trying to use my AudioPhile 2496. This is running on Debian Sid,
>>> uname -a reports "3.14-2-amd64 #1 SMP Debian 3.14.15-2 (2014-08-09)
>>> x86_64 GNU/Linux" (but I was getting the same error on kernel
>>> 3.02.4 before that.) I am a member of the audio group.
>>>
>>> JACK starts and runs fine if I pick the "default" interface, but
>>> that doesn't play any audio through the Audiophile. (I have no idea
>>> what it's playing through.)

> For further test, booted an old Ubuntu Studio 12.10 CD I have here and
> the Audiophile worked just fine. So I copied the
> /etc/modprobe.d/alsa-base.conf file from the running US12 environment to
> my system partition, rebooted, and QJackCtl starts jack with no problems
> and Yoshimi is currently making lovely sounds through the Audiophile.
> The only audio devices that appear are the Audiophile (card 0) and HDMI
> (card 1).

As others on the list suggested, here's the alsa-base.conf file I copied 
as-is from Ubuntu Studio 12.10 live CD. I didn't add or remove anything 
from it; I'm sure there's stuff in there that doesn't need to be there 
anymore! (Also, watch out for wordwraps!)

# autoloader aliases
install sound-slot-0 /sbin/modprobe snd-card-0
install sound-slot-1 /sbin/modprobe snd-card-1
install sound-slot-2 /sbin/modprobe snd-card-2
install sound-slot-3 /sbin/modprobe snd-card-3
install sound-slot-4 /sbin/modprobe snd-card-4
install sound-slot-5 /sbin/modprobe snd-card-5
install sound-slot-6 /sbin/modprobe snd-card-6
install sound-slot-7 /sbin/modprobe snd-card-7

# Cause optional modules to be loaded above generic modules
install snd /sbin/modprobe --ignore-install snd $CMDLINE_OPTS && { 
/sbin/modprobe --quiet --use-blacklist snd-ioctl32 ; /sbin/modprobe 
--quiet --use-blacklist snd-seq ; }
#
# Workaround at bug #499695 (reverted in Ubuntu see LP #319505)
install snd-pcm /sbin/modprobe --ignore-install snd-pcm $CMDLINE_OPTS && 
{ /sbin/modprobe --quiet --use-blacklist snd-pcm-oss ; : ; }
install snd-mixer /sbin/modprobe --ignore-install snd-mixer 
$CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-mixer-oss 
; : ; }
install snd-seq /sbin/modprobe --ignore-install snd-seq $CMDLINE_OPTS && 
{ /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; /sbin/modprobe 
--quiet --use-blacklist snd-seq-oss ; : ; }
#
install snd-rawmidi /sbin/modprobe --ignore-install snd-rawmidi 
$CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; 
: ; }
# Cause optional modules to be loaded above sound card driver modules
install snd-emu10k1 /sbin/modprobe --ignore-install snd-emu10k1 
$CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist 
snd-emu10k1-synth ; }
install snd-via82xx /sbin/modprobe --ignore-install snd-via82xx 
$CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq ; }

# Load saa7134-alsa instead of saa7134 (which gets dragged in by it anyway)
install saa7134 /sbin/modprobe --ignore-install saa7134 $CMDLINE_OPTS && 
{ /sbin/modprobe --quiet --use-blacklist saa7134-alsa ; : ; }
# Prevent abnormal drivers from grabbing index 0
options bt87x index=-2
options cx88_alsa index=-2
options saa7134-alsa index=-2
options snd-atiixp-modem index=-2
options snd-intel8x0m index=-2
options snd-via82xx-modem index=-2
options snd-usb-audio index=-2
options snd-usb-caiaq index=-2
options snd-usb-ua101 index=-2
options snd-usb-us122l index=-2
options snd-usb-usx2y index=-2
# Ubuntu #62691, enable MPU for snd-cmipci
options snd-cmipci mpu_port=0x330 fm_port=0x388
# Keep snd-pcsp from being loaded as first soundcard
options snd-pcsp index=-2
# Keep snd-usb-audio from beeing loaded as first soundcard
options snd-usb-audio index=-2


-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From murks at tuxfamily.org  Tue Aug 19 08:07:19 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Tue, 19 Aug 2014 10:07:19 +0200
Subject: [LAU] Internal speaker set as default playback,
 not my usb soundcard.
In-Reply-To: <CAGD0fJG3zXUoDv1EBwQYD-jd7zjfSaSOWCWri6on2opT4OgQ-Q@mail.gmail.com>
References: <CAGD0fJG3zXUoDv1EBwQYD-jd7zjfSaSOWCWri6on2opT4OgQ-Q@mail.gmail.com>
Message-ID: <20140819100719.322322d5@eeyore.mozart.uni-klu.ac.at>

On Tue, 19 Aug 2014 05:03:21 +0100
Andrew C <countfuzzball at gmail.com> wrote:

> Hello all,
> 
> Recently did an upgrade to Ubuntu 14.04 from 12.04. All went well
> except I've encountered a very weird problem.
> 
> My usb audio is no longer the default sound card. I had to some some
> trickery (made snd-usb-audio=-1 and snd-hda-intel=-2) in
> /etc/modprobe.d/alsa-base.conf to get the usb card as #0 as listed by
> /proc/asound/cards and then I got all sound going out through my usb
> card. Fun fun fun.
> 
> Now with Ubuntu 14.04, my usb card is still listed as #0 in
> asound/cards, but mplayer defaults to my internal speakers when
> issued 'mplayer sound.wav'.
> 
> Also tried:
> mplayer -Dalsa=hw=0.0 sound.wav
> Plays through my usb soundcard
> 
> mplayer -Dalsa=hw=1.0 sound.wav
> Plays through my internal speakers
> 
> mplayer -Dalsa=plughw sound.wav
> Plays through my usb soundcard
> 
> mplayer -Dalsa=default sound.wav
> Plays through my internal speakers.
> 
> Also I tried opening 'pavucontrol' fwiw and disabling the internal
> speaker. Playing a sound file without any other options via mplayer
> sends the sound through my usb soundcard.
> 
> This is really strange and perhaps I'm not entirely understanding all
> the "wonderfulness" that is alsa.
> 
> Cheers,
> 
> Andrew.

Hi Andrew,
I had lots of fun with that stuff too, but mostly because I wanted the
internal interface to be default when no usb interface is connected and
the usb to be the default when it is connected. It took me a while
to find a decent solution and what I use now is this:
https://wiki.archlinux.org/index.php/Alsa#Hot-plugging_a_USB_sound_card

I guess this approach or another on this page might work for you. Note
that this is for alsa, PA will do its own thing and consequently what
you set in pavucontrol should only affect PA. Not quite true,
PulseAudio also has a tendency to modify the alsa mixer in really stupid
ways (e.g. with my internal interface when the mic volume is
higher than 25% it starts to turn up mic boost which leads to horrible
distortion and you can set the alsamixer however you like, PA overrides
it). /rant


Regards,
Philipp

From ralf.mardorf at rocketmail.com  Tue Aug 19 08:50:58 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 19 Aug 2014 10:50:58 +0200
Subject: [LAU] [Bulk] Re: [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY
 unstable
In-Reply-To: <CAH8ZPGm0bnVf2E7t5q_8rH3w+uOK+F1hes1EfcN4W5GWgJKoXA@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian> <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <alpine.DEB.2.10.1408180646130.2503@scott.cbbs.org>
 <CAH8ZPGm0bnVf2E7t5q_8rH3w+uOK+F1hes1EfcN4W5GWgJKoXA@mail.gmail.com>
Message-ID: <1408438258.9742.1.camel@rocketmail.com>

On Mon, 2014-08-18 at 10:18 -0400, Alan McKay wrote:
> On Mon, Aug 18, 2014 at 9:58 AM, Len Ovens <len at ovenwerks.net> wrote:
> > So that would be the new (still experimental IMO) unity desktop with it's
> > ever changing interface that is slowly moving from X to MIR.
> 
> I'll install the traditional desktop and try it.

Make a new install of Ubuntu Studio (advantage for you, Len is member of
the developers team ;), Xubuntu or Kubuntu or any other *buntu. AFAIK
non of them includes the Ubuntu spyware, so this seems to be a better
way, than just installing Xfce4, KDE or other packages. Better than
using another *buntu IMO is to use another distro with a WM instead of a
DE.

-- 
http://iknowwhereyourcatlives.com/


From ralf.mardorf at rocketmail.com  Tue Aug 19 08:59:27 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 19 Aug 2014 10:59:27 +0200
Subject: [LAU] [Bulk]  Ubuntu Studio (was: Audacity unstable)
In-Reply-To: <CAH8ZPGm4vw5hyxo1QF7Pi-gxwjhtdhPCnLU-DZ4aOVrteOyK0Q@mail.gmail.com>
References: <CAH8ZPGm4vw5hyxo1QF7Pi-gxwjhtdhPCnLU-DZ4aOVrteOyK0Q@mail.gmail.com>
Message-ID: <1408438767.9742.4.camel@rocketmail.com>

On Mon, 2014-08-18 at 10:40 -0400, Alan McKay wrote:
> Hey folks, I'm thinking I might just install Ubuntu Studio tonight and
> run with that.

https://lists.ubuntu.com/mailman/listinfo/ubuntu-studio-users

> The website does not seem to tell me the essential differences from Ubuntu.

Some are: - a customized menu
          - the lowlatency kernel
            (PREEMPT, resp. IIRC full PREEMPT), not PREEMPT RT

> Which desktop is used?

Xfce4

> Are there any other major differences I'll notice aside from this?

Yesno, make a new install, assumed there should be issues with the
installer, install Xubuntu and add the Ubuntu Studio meta packages.

Stay with LAU, but also subscribe to Ubuntu Studio Users.

-- 
http://iknowwhereyourcatlives.com/


From ralf.mardorf at rocketmail.com  Tue Aug 19 09:23:04 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 19 Aug 2014 11:23:04 +0200
Subject: [LAU] [Bulk] Re: [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY
 unstable
In-Reply-To: <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian> <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
 <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
 <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
Message-ID: <1408440184.9742.6.camel@rocketmail.com>

On Mon, 2014-08-18 at 21:32 -0400, Alan McKay wrote:
> OK, low-latency kernel is in ... we'll see how it goes.
> 
> Now that we are messing with kernels I actually know what I am doing :-)

So you didn't forget to add the user to the group "audio" ("realtime" or
whatever it's called for Ubuntu Studio and you also set up PAM?

-- 
http://iknowwhereyourcatlives.com/


From nabob_cd at yahoo.com  Tue Aug 19 09:25:51 2014
From: nabob_cd at yahoo.com (Menno)
Date: Tue, 19 Aug 2014 02:25:51 -0700 (PDT)
Subject: [LAU] Delta1010: S/PDIF out has sync but no audio
In-Reply-To: <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
References: <1408361569528-92201.post@n7.nabble.com>
 <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
Message-ID: <1408440351607-92247.post@n7.nabble.com>

Hi Lev, Fons,

there is something very wrong here with my Delta1010 and KXStudio setup.

I checked with WindowsXP on the same machine and there i have:
- stereo sound analog
- stereo sound via S/PDIF
this means that the card works...

In KXStudio, with Mudita in the same settings as the Windows MAudio mixer:
- only channel 1 out
- a sync signal via S/PDIF but no sound

I have no Jack running yet - i think that first these problems should be
solved before i can build upon it any further.
The final goal is to get Jack running, so i can connect anything with
anything :)

I hope you can help me out.




--
View this message in context: http://linux-audio.4202.n7.nabble.com/Delta1010-S-PDIF-out-has-sync-but-no-audio-tp92201p92247.html
Sent from the linux-audio-user mailing list archive at Nabble.com.

From alan.mckay at gmail.com  Tue Aug 19 10:37:49 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Tue, 19 Aug 2014 06:37:49 -0400
Subject: [LAU] [Bulk] Re: [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY
	unstable
In-Reply-To: <1408440184.9742.6.camel@rocketmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
 <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
 <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
 <1408440184.9742.6.camel@rocketmail.com>
Message-ID: <CAH8ZPGka=jpfFq_yowEuDQXU2UDZVbbifG8immEGAjvjAiLBog@mail.gmail.com>

On Tue, Aug 19, 2014 at 5:23 AM, Ralf Mardorf
<ralf.mardorf at rocketmail.com> wrote:
> So you didn't forget to add the user to the group "audio" ("realtime" or
> whatever it's called for Ubuntu Studio and you also set up PAM?

Hmmm, neither.   Adding me to group audio makes sense, but what's with PAM?
What do I have to do there?  I can't find anything obvious with
google.  I see a reference
to it here :

https://bbs.archlinux.org/viewtopic.php?id=85334

Nothing using the search tool on the audacity site.


-- 
"Don't eat anything you've ever seen advertised on TV"
         - Michael Pollan, author of "In Defense of Food"

From djbarney at djbarney.org  Tue Aug 19 10:43:34 2014
From: djbarney at djbarney.org (djbarney)
Date: Tue, 19 Aug 2014 10:43:34 +0000
Subject: [LAU]
 =?utf-8?q?File_managers_for_musicians=2C_waveform_thumbnail?=
 =?utf-8?q?s=2C_presentation_as_tightly_packed_spiral_a=27la_vinyl_=3F_=28?=
 =?utf-8?q?Caja=2C_MATE=29?=
In-Reply-To: <CAGR1oLDmEvqXq3=R0jVaiU6h7ug5BRFN-5GSBJCt2Y77Enz_og@mail.gmail.com>
References: <47e758090882c49435a0c52821059b47@djbarney.org>
 <20140814004553.664d1ec7@gmail.com>
 <CAGR1oLDmEvqXq3=R0jVaiU6h7ug5BRFN-5GSBJCt2Y77Enz_og@mail.gmail.com>
Message-ID: <acd7b7de0e388556dd0ee81d4de04c9b@djbarney.org>

On 2014-08-14 01:23, Danni Coy wrote:
> May I make two suggestions
> 1) mood files -> some audio players generate these - most notably
> Amarok with the right plugins installed. These break up the audio
> files into 3 bands lows/mids/highs and display the level for each
> band.
> 2) the circular wave displays in freewheeling... These would fit into
> an icon preview very easily.
> 
> I have some experience doing preview plugins for Dolphin if you think
> that would be helpful.

Hi. Yes that could be set by the user if they want mood file previews. 
Yes, freewheeling thumbs are circular but you can't fit much in a simply 
wrapped around (polar coordinates) thumb like that. Think how much 
visual information there is on the surface of a vinyl record.

Dolphin plugins would be great, but I want to keep this as easily 
adjustable by the user as possible ... setting their won CLI tools 
through text settings files. Now a Dolphin plugin that would allow that 
?

I'm looking at creating my own interface with Tk/TCL or might just hack 
the Caja code.

DJ Barney

> 
> On Thu, Aug 14, 2014 at 8:45 AM, renato <rennabh at gmail.com> wrote:
>> On Wed, 13 Aug 2014 19:56:49 +0000
>> djbarney <djbarney at djbarney.org> wrote:
>> 
>>> Hi,
>>> 
>>> Thought I'd air this one to see if there's anyone already using this
>>> kind of thing out there.
>>> 
>>> I'd become frustrated by lack of support for musicians in Linux
>>> window manager file managers. No default waveform/spectrogram
>>> support. No MIDI notation previews. No BPM or key info, etc, etc.
>>> 
>>> I set up Caja to thumbnail WAV files using Sox ... see some
>>> screenshots on this forum thread ...
>>> 
>>> http://linuxmusicians.com/viewtopic.php?f=4&t=12514&p=55280#p55280
>>> 
>>> I'm looking at some other thumbnailers as listed on the thread.
>>> 
>>> I'm thinking of doing some development of the MATE Caja file browser
>>> to allow better presentation of audio file waveform thumbnails as
>>> currently they can only be square. Maybe LAU members already know of
>>> someone who has done this ? No point in reinventing the wheel ...
>>> otherwise I'll take this to LAD.
>>> 
>> 
>> Hi, I think samplecat does some of the above. Also I've heard sox has
>> changed syntax between versions before, so it might break your program
>> in the future; others where recommending me ecasound instead... or
>> maybe libsndfile?
>> 
>> cheers,
>> renato
>> _______________________________________________
>> Linux-audio-user mailing list
>> Linux-audio-user at lists.linuxaudio.org
>> http://lists.linuxaudio.org/listinfo/linux-audio-user

-- 
~~~

http://djbarney.org

From djbarney at djbarney.org  Tue Aug 19 10:43:54 2014
From: djbarney at djbarney.org (djbarney)
Date: Tue, 19 Aug 2014 10:43:54 +0000
Subject: [LAU]
 =?utf-8?q?File_managers_for_musicians=2C_waveform_thumbnail?=
 =?utf-8?q?s=2C_presentation_as_tightly_packed_spiral_a=27la_vinyl_=3F_=28?=
 =?utf-8?q?Caja=2C_MATE=29?=
In-Reply-To: <20140814004553.664d1ec7@gmail.com>
References: <47e758090882c49435a0c52821059b47@djbarney.org>
 <20140814004553.664d1ec7@gmail.com>
Message-ID: <b63f7230ea6976766ca8fd5ed51f5eb4@djbarney.org>

On 2014-08-13 22:45, renato wrote:
> On Wed, 13 Aug 2014 19:56:49 +0000
> djbarney <djbarney at djbarney.org> wrote:
> 
>> Hi,
>> 
>> Thought I'd air this one to see if there's anyone already using this
>> kind of thing out there.
>> 
>> I'd become frustrated by lack of support for musicians in Linux
>> window manager file managers. No default waveform/spectrogram
>> support. No MIDI notation previews. No BPM or key info, etc, etc.
>> 
>> I set up Caja to thumbnail WAV files using Sox ... see some
>> screenshots on this forum thread ...
>> 
>> http://linuxmusicians.com/viewtopic.php?f=4&t=12514&p=55280#p55280
>> 
>> I'm looking at some other thumbnailers as listed on the thread.
>> 
>> I'm thinking of doing some development of the MATE Caja file browser
>> to allow better presentation of audio file waveform thumbnails as
>> currently they can only be square. Maybe LAU members already know of
>> someone who has done this ? No point in reinventing the wheel ...
>> otherwise I'll take this to LAD.
>> 
> 
> Hi, I think samplecat does some of the above. Also I've heard sox has
> changed syntax between versions before, so it might break your program
> in the future; others where recommending me ecasound instead... or
> maybe libsndfile?
> 
> cheers,
> renato

Hi, yes saw Samplecat but could not get it to compile. Good effort but 
to be honest the interface needs some work.

So syntax changing is not a massive problem. I want to keep text 
settings files and CLI tools as, in my opinion, writing in C++ all the 
time and using libraries takes power away from the user (of course there 
are appropriate uses for coding).

DJ Barney

-- 
~~~

http://djbarney.org

From fons at linuxaudio.org  Tue Aug 19 11:05:46 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Tue, 19 Aug 2014 11:05:46 +0000
Subject: [LAU] Delta1010: S/PDIF out has sync but no audio
In-Reply-To: <1408440351607-92247.post@n7.nabble.com>
References: <1408361569528-92201.post@n7.nabble.com>
 <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
 <1408440351607-92247.post@n7.nabble.com>
Message-ID: <20140819110546.GB14822@linuxaudio.org>

On Tue, Aug 19, 2014 at 02:25:51AM -0700, Menno wrote:
 
> I have no Jack running yet - i think that first these problems should be
> solved before i can build upon it any further.
> The final goal is to get Jack running, so i can connect anything with
> anything :)

It could be easier if you get Jack running on that card. SPDIF outputs
are 9 and 10. Use any other Jack app to send some signal to those.

If you configure vlc to use jack it will probably use the wrong
outputs (1 and 2) but you can change the connections manually
using qjackctl. It's not a final solution but it allows you to
test things.

Ciao,


-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From ats at offog.org  Tue Aug 19 11:45:25 2014
From: ats at offog.org (Adam Sampson)
Date: Tue, 19 Aug 2014 12:45:25 +0100
Subject: [LAU] Delta1010: S/PDIF out has sync but no audio
In-Reply-To: <1408440351607-92247.post@n7.nabble.com> (Menno's message of
 "Tue, 19 Aug 2014 02:25:51 -0700 (PDT)")
References: <1408361569528-92201.post@n7.nabble.com>
 <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
 <1408440351607-92247.post@n7.nabble.com>
Message-ID: <y2ak364bt4a.fsf@cartman.at.offog.org>

Menno <nabob_cd at yahoo.com> writes:

> In KXStudio, with Mudita in the same settings as the Windows MAudio mixer:
> - only channel 1 out
> - a sync signal via S/PDIF but no sound

Have you tried using envy24control? It's a mixer specifically for
ICE1712 cards like the Delta 1010; it gives you better control over
input/output routing than the generic mixers, and shows you the current
audio levels on all the inputs and outputs. It would be worth checking
that the SPDIF output is fed from the digital mix output (which is
probably what you want) and is actually receiving a signal.

It looks like it's in the "alsa-utils" package on Ubuntu...

-- 
Adam Sampson <ats at offog.org>                         <http://offog.org/>

From alan.mckay at gmail.com  Tue Aug 19 11:56:05 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Tue, 19 Aug 2014 07:56:05 -0400
Subject: [LAU] [Bulk] Re: [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY
	unstable
In-Reply-To: <CAH8ZPGka=jpfFq_yowEuDQXU2UDZVbbifG8immEGAjvjAiLBog@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
 <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
 <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
 <1408440184.9742.6.camel@rocketmail.com>
 <CAH8ZPGka=jpfFq_yowEuDQXU2UDZVbbifG8immEGAjvjAiLBog@mail.gmail.com>
Message-ID: <CAH8ZPGkQAGusC5rCjx6iO35JUS5XwFGbLRY5mSQ1PXhOZbxmhg@mail.gmail.com>

After putting myself in the audio group I do the "lshw" test again
while ripping and all seems fine.

From alan.mckay at gmail.com  Tue Aug 19 11:56:37 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Tue, 19 Aug 2014 07:56:37 -0400
Subject: [LAU] [Bulk] Re: [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY
	unstable
In-Reply-To: <CAH8ZPGkQAGusC5rCjx6iO35JUS5XwFGbLRY5mSQ1PXhOZbxmhg@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
 <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
 <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
 <1408440184.9742.6.camel@rocketmail.com>
 <CAH8ZPGka=jpfFq_yowEuDQXU2UDZVbbifG8immEGAjvjAiLBog@mail.gmail.com>
 <CAH8ZPGkQAGusC5rCjx6iO35JUS5XwFGbLRY5mSQ1PXhOZbxmhg@mail.gmail.com>
Message-ID: <CAH8ZPGk63f64=o14LEZ6jqiQiYTG3aOtDu+Q2+GytB8RyOX_Sg@mail.gmail.com>

Oh wait that was 2 changes - I also put my latency to 0 - I should
undo one of them and try again.

From nabob_cd at yahoo.com  Tue Aug 19 12:02:05 2014
From: nabob_cd at yahoo.com (Menno)
Date: Tue, 19 Aug 2014 05:02:05 -0700 (PDT)
Subject: [LAU] Delta1010: S/PDIF out has sync but no audio
In-Reply-To: <y2ak364bt4a.fsf@cartman.at.offog.org>
References: <1408361569528-92201.post@n7.nabble.com>
 <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
 <1408440351607-92247.post@n7.nabble.com>
 <y2ak364bt4a.fsf@cartman.at.offog.org>
Message-ID: <1408449725804-92255.post@n7.nabble.com>

i worked with Ubuntu12.04 with Enlightenment and the packages from KXStudio
before i made a clean install of KXSudio14.04 (with KDE).
With 12.04 everything worked all right; i had Jack running with the same
MAudio Delta 1010, and had a nice low latency.
What i don't understand why things are so different with 14.04 on the same
system.
I was hoping that someone could help me in debugging this. Where to start?



--
View this message in context: http://linux-audio.4202.n7.nabble.com/Delta1010-S-PDIF-out-has-sync-but-no-audio-tp92201p92255.html
Sent from the linux-audio-user mailing list archive at Nabble.com.

From countfuzzball at gmail.com  Tue Aug 19 12:48:22 2014
From: countfuzzball at gmail.com (Andrew C)
Date: Tue, 19 Aug 2014 13:48:22 +0100
Subject: [LAU] Internal speaker set as default playback,
	not my usb soundcard.
In-Reply-To: <20140819100719.322322d5@eeyore.mozart.uni-klu.ac.at>
References: <CAGD0fJG3zXUoDv1EBwQYD-jd7zjfSaSOWCWri6on2opT4OgQ-Q@mail.gmail.com>
 <20140819100719.322322d5@eeyore.mozart.uni-klu.ac.at>
Message-ID: <CAGD0fJHa7eEH7MQrGW2OJeOwqTss5CGm91UhzLZDceWs9VCQag@mail.gmail.com>

Speaking of .asoundrc, I had set up my like so (which predictably no longer
works, even though my usb card is still at index 0):

pcm.!default {
    type            plug
    slave.pcm       "softvol"   #make use of softvol
}

pcm.softvol {
    type            softvol
    slave {
        pcm         "dmix"      #redirect the output to dmix (instead of
"hw:0,0")
    }
    control {
        name        "PCM"       #override the PCM slider to set the softvol
volume level globally
        card        0
    }
}

I doubt this would cause a mess up (even so, even *that* is no longer
working, my soundcard ignores any changes made by alsamixer).

At this point, would I guess there's some pulseaudio weirdness happening?
Any ideas how I can use alsa straight up, unless I've somehow been using a
pulseaudio-alsa wrapper this entire time.

To my mind, it makes absolutely no sense why stuff would be playing out of
Card 1, and not Card 0. Or at least why that has suddenly changed to be the
default with this upgrade.

Andrew.
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From ralf.mardorf at rocketmail.com  Tue Aug 19 14:25:15 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 19 Aug 2014 16:25:15 +0200
Subject: [LAU] [Bulk] Re: [Bulk] Re: [Bulk] Re: Audacity on Ubuntu 14.04
 is REALLY unstable
In-Reply-To: <CAH8ZPGka=jpfFq_yowEuDQXU2UDZVbbifG8immEGAjvjAiLBog@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian> <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
 <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
 <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
 <1408440184.9742.6.camel@rocketmail.com>
 <CAH8ZPGka=jpfFq_yowEuDQXU2UDZVbbifG8immEGAjvjAiLBog@mail.gmail.com>
Message-ID: <1408458315.7754.1.camel@rocketmail.com>

On Tue, 2014-08-19 at 06:37 -0400, Alan McKay wrote:
> On Tue, Aug 19, 2014 at 5:23 AM, Ralf Mardorf
> <ralf.mardorf at rocketmail.com> wrote:
> > So you didn't forget to add the user to the group "audio" ("realtime" or
> > whatever it's called for Ubuntu Studio and you also set up PAM?
> 
> Hmmm, neither.   Adding me to group audio makes sense, but what's with PAM?
> What do I have to do there?  I can't find anything obvious with
> google.

To /etc/security/limits.conf or /etc/security/limits.d/audio.conf add:

@audio 	- rtprio 	99
@audio 	- memlock 	unlimited

some prefer to give less rtprio, perhaps you'll prefer

@audio   -  rtprio     95
@audio   -  memlock    unlimited

IIRC on older machines graphics, mouse, keyboard could become
unresponsive with prio set to 99. "nice" values are useless for rt.

You should use the rtirq script too. OTOH, somebody already mentioned,
that for using Audacity there's no need for a PREEMPT or PREEMPT RT
kernel.

http://www.rncbc.org/jack/

Be careful when using a rtirq package from official Ubuntu repositories,
nobody seems to test if the script fits to the provided kernel, IOW
sometimes the provided version doesn't work, simply run the script with
the status option, then you'll see if it works and don't forget to set
up the config for the script, OTOH the default likely will be ok for
your setup.

rtc0 should get the highest prio, followed by your sound device.

-- 
"I Know Where Your Cat Lives visualizes public photos of cats on a world
map using coordinates embedded in their metadata.
[snip]
"I Know Where Your Cat Lives" iknowwhereyourcatlives.com is a data
experiment that visualizes a sample of 1 million public pics of cats on
a world map, locating them by the latitude and longitude coordinates
embedded in their metadata. The cats were accessed via publicly
available APIs provided by popular photo sharing websites. The photos
were then run through various clustering algorithms using a
supercomputer at Florida State University in order to represent the
enormity of the data source.
[snip]
" -
https://www.kickstarter.com/projects/1910822604/i-know-where-your-cat-lives



From len at ovenwerks.net  Tue Aug 19 15:02:49 2014
From: len at ovenwerks.net (Len Ovens)
Date: Tue, 19 Aug 2014 08:02:49 -0700 (PDT)
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <53F30224.2000909@hawaii.rr.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian> <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <0230eb1f-5a62-412a-82ca-e2c899dd880f@email.android.com>
 <53F30224.2000909@hawaii.rr.com>
Message-ID: <alpine.DEB.2.10.1408190757050.2503@scott.cbbs.org>

On Mon, 18 Aug 2014, david wrote:

> Well, I've heard that Gnome 3's default configuration makes it a major 
> resource hog - particularly for memory. Like KDE4. I avoid Gnome 3. Both 
> it and KDE4 launch a bunch of system services that I think can really 
> impact RT use.

Major difference between KDE and Gnome 3 from what I can tell. My 10 year 
old P4 with 1G ram had no problems with KDE, but was unusable with Gnome3. 
Kde does have lots of "how much cpu would you like to use" settings, 
though if you do turn off all the eye candy one wonders why use KDE... The 
answer seems to be stability. I have in this house machines running both 
xfce and KDE. There seem to be some applications (mostly video) that run 
on any machine running KDE and only some of the machines when running 
xfce.


--
Len Ovens
www.ovenwerks.net


From len at ovenwerks.net  Tue Aug 19 15:17:35 2014
From: len at ovenwerks.net (Len Ovens)
Date: Tue, 19 Aug 2014 08:17:35 -0700 (PDT)
Subject: [LAU] [Bulk] Re: [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY
 unstable
In-Reply-To: <CAH8ZPGka=jpfFq_yowEuDQXU2UDZVbbifG8immEGAjvjAiLBog@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian> <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
 <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
 <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
 <1408440184.9742.6.camel@rocketmail.com>
 <CAH8ZPGka=jpfFq_yowEuDQXU2UDZVbbifG8immEGAjvjAiLBog@mail.gmail.com>
Message-ID: <alpine.DEB.2.10.1408190807480.2503@scott.cbbs.org>

On Tue, 19 Aug 2014, Alan McKay wrote:

> On Tue, Aug 19, 2014 at 5:23 AM, Ralf Mardorf
> <ralf.mardorf at rocketmail.com> wrote:
>> So you didn't forget to add the user to the group "audio" ("realtime" or
>> whatever it's called for Ubuntu Studio and you also set up PAM?
>
> Hmmm, neither.   Adding me to group audio makes sense, but what's with PAM?
> What do I have to do there?  I can't find anything obvious with
> google.  I see a reference
> to it here :

In a terminal enter:
groups

you should get a list of groups the current user is a member of. One of 
them should be audio. If not:
sudo adduser <username> audio


ls /etc/security/limits.d/
should show:
audio.conf
if jack was installed correctly. If instead you have:
audio.conf.disabled

Then:
sudo mv /etc/security/limits.d/audio.conf.disabled /etc/security/limits.d/audio.conf

This will make sure jack is actually using what the lowlatency kernel 
offers.


--
Len Ovens
www.ovenwerks.net


From countfuzzball at gmail.com  Tue Aug 19 15:49:44 2014
From: countfuzzball at gmail.com (Andrew C)
Date: Tue, 19 Aug 2014 16:49:44 +0100
Subject: [LAU] Internal speaker set as default playback,
	not my usb soundcard.
In-Reply-To: <20140819154752.028b1220@eeyore.mozart.uni-klu.ac.at>
References: <CAGD0fJG3zXUoDv1EBwQYD-jd7zjfSaSOWCWri6on2opT4OgQ-Q@mail.gmail.com>
 <20140819100719.322322d5@eeyore.mozart.uni-klu.ac.at>
 <CAGD0fJHa7eEH7MQrGW2OJeOwqTss5CGm91UhzLZDceWs9VCQag@mail.gmail.com>
 <20140819154752.028b1220@eeyore.mozart.uni-klu.ac.at>
Message-ID: <CAGD0fJE8SuGXEzX3_wdNn3n41KxcVePCVu_HGMdP+sh9CRv=Bg@mail.gmail.com>

Excellent, thanks for that tip, Phillip! Got it fixed now.

Put "autospawn = no" into $HOME/.config/pulse/client.conf, killed PA as you
suggested and everything worked as expected, alsamixer controls the volume
(.asoundrc works perfectly again) and mplayer/flash/MOCP plays through my
usb card.

Andrew.


On Tue, Aug 19, 2014 at 2:47 PM, Philipp ?berbacher <murks at tuxfamily.org>
wrote:

> On Tue, 19 Aug 2014 13:48:22 +0100
> Andrew C <countfuzzball at gmail.com> wrote:
>
> > Speaking of .asoundrc, I had set up my like so (which predictably no
> > longer works, even though my usb card is still at index 0):
> >
> > pcm.!default {
> >     type            plug
> >     slave.pcm       "softvol"   #make use of softvol
> > }
> >
> > pcm.softvol {
> >     type            softvol
> >     slave {
> >         pcm         "dmix"      #redirect the output to dmix (instead
> > of "hw:0,0")
> >     }
> >     control {
> >         name        "PCM"       #override the PCM slider to set the
> > softvol volume level globally
> >         card        0
> >     }
> > }
> >
> > I doubt this would cause a mess up (even so, even *that* is no longer
> > working, my soundcard ignores any changes made by alsamixer).
> >
> > At this point, would I guess there's some pulseaudio weirdness
> > happening? Any ideas how I can use alsa straight up, unless I've
> > somehow been using a pulseaudio-alsa wrapper this entire time.
> >
> > To my mind, it makes absolutely no sense why stuff would be playing
> > out of Card 1, and not Card 0. Or at least why that has suddenly
> > changed to be the default with this upgrade.
> >
> > Andrew.
>
> I'm not sure whether you get direct alsa access while PA is running.
>
> You could disable pulseaudio and try again. I have configured PA
> so that it is only ever started manually by me and I simply use
> 'pulseaudio --start' and 'pulseaudio --kill'. However, depending on
> your distributions setup it might be a PITA to stop PA and keep it
> stopped. You can use 'pulseaudio --check' to see whether it runs, it
> returns 0 if it runs, 1 otherwise. Or just use 'pulseaudio --kill' and
> see whether it throws an error or succeeds because something started PA
> again. This can happen because PA is set to respawn or because some
> program started PA on startup. That's a PITA I don't want to deal with,
> hence my entirely manual setup.
>
> Regards,
> Philipp
>
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From len at ovenwerks.net  Tue Aug 19 15:50:25 2014
From: len at ovenwerks.net (Len Ovens)
Date: Tue, 19 Aug 2014 08:50:25 -0700 (PDT)
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <CAH8ZPGmFooY4eJUmrCocgTLpPEV+9mceTAyWsgUPexfjTCXyMw@mail.gmail.com>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian> <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
 <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
 <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
 <CAH8ZPG=A4AMKNCXCKC2mOcFtvDofrBey2+mWWrHeXJb5qJV7jg@mail.gmail.com>
 <CAH8ZPGmFooY4eJUmrCocgTLpPEV+9mceTAyWsgUPexfjTCXyMw@mail.gmail.com>
Message-ID: <alpine.DEB.2.10.1408190822220.2503@scott.cbbs.org>

On Mon, 18 Aug 2014, Alan McKay wrote:

> As I was reading I was curious as to my CPU power, so I opened a shell
> to run "lshw".  As soon as I hit "enter" my rip stopped in the usual
> fashion, but then everything on my desktop slowed right down.  My
> mouse was moving in slow-motion.  I tried typing this and it was
> missing letters (since rebooted).
>
> So it seems basically that my system is underpowered, I guess.  I have
> 4 cores, and this is from lshw.  Maybe I need a real sound card too?
>
>
>         description: CPU
>          product: AMD A8-5600K APU with Radeon(tm) HD Graphics
>          vendor: Advanced Micro Devices [AMD]

My first thought at seeing this is: I wonder if this is one of those AMD 
products where the GPU steals CPU cycles to operate. I know one of the AMD 
lines of CPUs that includes the GPU has this problem. The GPU steals CPU 
cycles that the OS has no control over at all. This page:
http://wiki.linuxcnc.org/cgi-bin/wiki.pl?Latency-Test
Seems to think it is ok though. Do check all the things they disabled in 
BIOS.

For normal running of audacity at a reasonable latency it shouldn't 
matter. Audacity is not a multitracker (there are some people who have 
used it that way, but Ardour is so much easier for this) it's best use is 
just recording so a higher latency is better. (50ms plus) As others have 
said, mhWaveEdit is a better recorder and is the most similar to Audacity 
in feel.


--
Len Ovens
www.ovenwerks.net


From alan.mckay at gmail.com  Tue Aug 19 23:55:19 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Tue, 19 Aug 2014 19:55:19 -0400
Subject: [LAU] [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY unstable
In-Reply-To: <alpine.DEB.2.10.1408190822220.2503@scott.cbbs.org>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
 <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
 <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
 <CAH8ZPG=A4AMKNCXCKC2mOcFtvDofrBey2+mWWrHeXJb5qJV7jg@mail.gmail.com>
 <CAH8ZPGmFooY4eJUmrCocgTLpPEV+9mceTAyWsgUPexfjTCXyMw@mail.gmail.com>
 <alpine.DEB.2.10.1408190822220.2503@scott.cbbs.org>
Message-ID: <CAH8ZPGk6DWmP6d1Si82JnYerRBpMk8nGngX8LmTdj=xFgTYGHQ@mail.gmail.com>

Well I switched back to generic kernel now that I am in the audio group

Ripped for almost 45 minutes before it hit "stop" on me.  Which is
pretty outstanding.

At this point I think I give up.  I have another computer I was going
to dedicate to media PC anyway, and it is closer to the turntable.  So
I'm just going to set that up and see what happens.

Well, I'll keep this going if you guys will ... off to reply to a few
more of your emails.

From alan.mckay at gmail.com  Tue Aug 19 23:57:20 2014
From: alan.mckay at gmail.com (Alan McKay)
Date: Tue, 19 Aug 2014 19:57:20 -0400
Subject: [LAU] [Bulk] Re: [Bulk] Re: Audacity on Ubuntu 14.04 is REALLY
	unstable
In-Reply-To: <alpine.DEB.2.10.1408190807480.2503@scott.cbbs.org>
References: <CAH8ZPGnAbHu+R_mPo0Qd3OTQPWOgAmZUmbxK2cGEZdvEqpxY0w@mail.gmail.com>
 <1408288773.18721.2.camel@eviltwin>
 <CAH8ZPGkkSspDoQwQu66Bjh7PqnTRNHzEd0pADBbKj=wMga_CVA@mail.gmail.com>
 <20140818001114.2e566efd@debian>
 <1408351336.2320.6.camel@rocketmail.com>
 <CAH8ZPGma=kgG5R3xZr=9iQaxqiz1rj0SvBeVuQ0UCq+0O_zGtA@mail.gmail.com>
 <CAH8ZPGnjRX9i_aXdoMbgeAC4QWkEMKh8vOhB2UJtULEqk1eF6g@mail.gmail.com>
 <20140818142156.5f3261de@eeyore.mozart.uni-klu.ac.at>
 <20140818143458.GA26987@linuxaudio.org>
 <CABRv+92AQ7qyk4BOZ_q49VS+2-_Zu1TudUZxpyO00DmNobmgDQ@mail.gmail.com>
 <53F240F5.8060608@hawaii.rr.com>
 <CAH8ZPGmGK3e+4Q=XiUeExVwWsZ+K1pBJVqemjG07UCha_JFGAA@mail.gmail.com>
 <CAH8ZPG=cunx-cKQXWCZE1xA_Wq-7wXCJ6tmyV3MqcKzOztBsMA@mail.gmail.com>
 <CAH8ZPG=t=+kvUv4ZV0to95WH+ctixZ1BmxqDhEqvMoO5=Br9bQ@mail.gmail.com>
 <1408440184.9742.6.camel@rocketmail.com>
 <CAH8ZPGka=jpfFq_yowEuDQXU2UDZVbbifG8immEGAjvjAiLBog@mail.gmail.com>
 <alpine.DEB.2.10.1408190807480.2503@scott.cbbs.org>
Message-ID: <CAH8ZPG=C+85e44ZaXF3REnGooaJwq07irdz2akZiO301xJn9-Q@mail.gmail.com>

On Tue, Aug 19, 2014 at 11:17 AM, Len Ovens <len at ovenwerks.net> wrote:
> In a terminal enter:
> groups

Yup, I am there.   Added it yesterday

> ls /etc/security/limits.d/
> should show:
> audio.conf
> if jack was installed correctly. If instead you have:
> audio.conf.disabled

audio.conf is there

But now I'm back to the generic kernel.  It seems just adding myself
to the audio group was a massive help.




-- 
"Don't eat anything you've ever seen advertised on TV"
         - Michael Pollan, author of "In Defense of Food"

From egor.sanin at gmail.com  Thu Aug 21 15:38:23 2014
From: egor.sanin at gmail.com (Egor Sanin)
Date: Thu, 21 Aug 2014 11:38:23 -0400
Subject: [LAU] [LAD] [LAA] hrec
Message-ID: <CALVwFW_021TH63Dkq6eYQnoNBU=L8mS_k5nS8_YRDQ1KpO3ctA@mail.gmail.com>

Hi folks,

I've been working on and off for the last little while on a command
line recording utility.  It's really basic and the code is pretty
ugly.  It exists partly because in my lazy search I couldn't find
anything that satisfied my particular need, and also because it was
interesting.  I just made a first release, so you could try it if you
like.

The program is called hrec, it's basically a curses front end to a
very limited subset of ecasound's functions.

Code is here:
https://sourceforge.net/projects/hrec/

AUR package is here:
https://aur.archlinux.org/packages/hrec/

I'm planning to do change it soon, but the basics will remain the
same.  Mostly I plan to remove the playback functionality, as it's
almost useless -- plenty of other software to do it much better.  Also
I need to migrate the code to Python 3, but I'm getting stuck on the
python ecasound bindings.  In the couple of (admittedly not very
thorough) attemps I've made I ran into problems with the current pyeca
module.  Any advice would be appreciated.  Anyway I thought I might as
well release it now and get some feedback.

I hope someone finds hrec useful!

Criticism and insults are also welcome.

Thanks!

From silvain at freeshell.de  Thu Aug 21 16:03:12 2014
From: silvain at freeshell.de (F. Silvain)
Date: Thu, 21 Aug 2014 18:03:12 +0200 (CEST)
Subject: [LAU] [LAD] [LAA] hrec
In-Reply-To: <CALVwFW_021TH63Dkq6eYQnoNBU=L8mS_k5nS8_YRDQ1KpO3ctA@mail.gmail.com>
References: <CALVwFW_021TH63Dkq6eYQnoNBU=L8mS_k5nS8_YRDQ1KpO3ctA@mail.gmail.com>
Message-ID: <1408211758540.8578@freeshell.de>

Egor Sanin, Aug 21 2014:

> Hi folks,
>
> I've been working on and off for the last little while on a command
> line recording utility.  It's really basic and the code is pretty
> ugly.  It exists partly because in my lazy search I couldn't find
> anything that satisfied my particular need, and also because it was
> interesting.  I just made a first release, so you could try it if you
> like.
...
> Criticism and insults are also welcome.

Hey egor,
thanks for the new tool. I just installed and ran into this error:
*** cut ***
Traceback (most recent call last):
   File "/usr/local/bin/hrec", line 19, in <module>
     from pyeca import *
   File "/usr/lib/python3.1/lib-dynload/pyeca.py", line 46, in <module>
     from ecacontrol import *
   File "/usr/lib/python3.1/lib-dynload/ecacontrol.py", line 77
     print 'c=' + I._cmd
              ^
SyntaxError: invalid syntax
*** end ***

I just updated ecasound from git and made sure it compiled with python3.1 .

Ta-ta
----
Ffanci
* Internet: http://freeshell.de/~silvain

From egor.sanin at gmail.com  Thu Aug 21 16:12:09 2014
From: egor.sanin at gmail.com (Egor Sanin)
Date: Thu, 21 Aug 2014 12:12:09 -0400
Subject: [LAU] [LAD] [LAA] hrec
In-Reply-To: <1408211758540.8578@freeshell.de>
References: <CALVwFW_021TH63Dkq6eYQnoNBU=L8mS_k5nS8_YRDQ1KpO3ctA@mail.gmail.com>
 <1408211758540.8578@freeshell.de>
Message-ID: <CALVwFW8B8rspd5e1OA0J32zC0h39tn5_ebA_ZVyKsEbxdRRPwA@mail.gmail.com>

Hi Silvain

Thanks for trying it out!

On 8/21/14, F. Silvain <silvain at freeshell.de> wrote:
> Hey egor,
> thanks for the new tool. I just installed and ran into this error:
> *** cut ***
> Traceback (most recent call last):
>    File "/usr/local/bin/hrec", line 19, in <module>
>      from pyeca import *
>    File "/usr/lib/python3.1/lib-dynload/pyeca.py", line 46, in <module>
>      from ecacontrol import *
>    File "/usr/lib/python3.1/lib-dynload/ecacontrol.py", line 77
>      print 'c=' + I._cmd
>               ^
> SyntaxError: invalid syntax
> *** end ***
>
> I just updated ecasound from git and made sure it compiled with python3.1 .

Yeah at the moment I can only confirm that hrec works when run with
Python2 explicitly.  As I mentioned I had problems running it with
Python3.  As far as I know, the pyeca python module, which I use as a
bridge between hrec and ecasound, doesn't play well with Python3.

Try installing hrec with python2 explicitly, on Arch Linux it would be
like this:

$ python2 setup.py install

From silvain at freeshell.de  Thu Aug 21 16:50:17 2014
From: silvain at freeshell.de (F. Silvain)
Date: Thu, 21 Aug 2014 18:50:17 +0200 (CEST)
Subject: [LAU] [LAD] [LAA] hrec
In-Reply-To: <CALVwFW8B8rspd5e1OA0J32zC0h39tn5_ebA_ZVyKsEbxdRRPwA@mail.gmail.com>
References: <CALVwFW_021TH63Dkq6eYQnoNBU=L8mS_k5nS8_YRDQ1KpO3ctA@mail.gmail.com>
 <1408211758540.8578@freeshell.de>
 <CALVwFW8B8rspd5e1OA0J32zC0h39tn5_ebA_ZVyKsEbxdRRPwA@mail.gmail.com>
Message-ID: <1408211849110.18110@freeshell.de>

Hey Egor,
sorry, I misread your mail and then my output. I tried with python2 first. But 
the error there wasn't about Ecasound at all. It's in subprocess, having no 
attribute checkout. Have to investigate. Later though...

Thanks for the quick heads-up!

Ta-ta
----
Ffanci
* Internet: http://freeshell.de/~silvain

From gerald.mwangi at gmx.de  Thu Aug 21 19:27:49 2014
From: gerald.mwangi at gmx.de (Gerald Mwangi)
Date: Thu, 21 Aug 2014 21:27:49 +0200
Subject: [LAU] *** GMX Spamverdacht *** Re: File managers for musicians,
 waveform thumbnails,
 presentation as tightly packed spiral a'la vinyl ? (Caja, MATE)
In-Reply-To: <b63f7230ea6976766ca8fd5ed51f5eb4@djbarney.org>
References: <47e758090882c49435a0c52821059b47@djbarney.org>
 <20140814004553.664d1ec7@gmail.com>
 <b63f7230ea6976766ca8fd5ed51f5eb4@djbarney.org>
Message-ID: <53F64835.5020205@gmx.de>

Hi,
while I don't know much about sox, I think this problem is a subset the 
audio labeling/Segmentation class of problem. The guys over at the CLAM 
project (http://clam-project.org/) have been dealing with this. They 
also developed a set of c++/qt widgets for the visual representation of 
musical data.
I also don't think that a waveform is suitable for thumbnail: the y-axis 
is used for the amplitude which completely useless given the small 
sizeof a thumbnail. A better representation is the frequency/time image 
(frequency on the y-axis, timeon the x-axis). The amplitude is given by 
the color/intensity at time x, frequency y. The benefit is that one has 
more info (frequency) in the thumbnail.
regards,
Gerald
On 19.08.2014 12:43, djbarney wrote:
> On 2014-08-13 22:45, renato wrote:
>> On Wed, 13 Aug 2014 19:56:49 +0000
>> djbarney <djbarney at djbarney.org> wrote:
>>
>>> Hi,
>>>
>>> Thought I'd air this one to see if there's anyone already using this
>>> kind of thing out there.
>>>
>>> I'd become frustrated by lack of support for musicians in Linux
>>> window manager file managers. No default waveform/spectrogram
>>> support. No MIDI notation previews. No BPM or key info, etc, etc.
>>>
>>> I set up Caja to thumbnail WAV files using Sox ... see some
>>> screenshots on this forum thread ...
>>>
>>> http://linuxmusicians.com/viewtopic.php?f=4&t=12514&p=55280#p55280
>>>
>>> I'm looking at some other thumbnailers as listed on the thread.
>>>
>>> I'm thinking of doing some development of the MATE Caja file browser
>>> to allow better presentation of audio file waveform thumbnails as
>>> currently they can only be square. Maybe LAU members already know of
>>> someone who has done this ? No point in reinventing the wheel ...
>>> otherwise I'll take this to LAD.
>>>
>>
>> Hi, I think samplecat does some of the above. Also I've heard sox has
>> changed syntax between versions before, so it might break your program
>> in the future; others where recommending me ecasound instead... or
>> maybe libsndfile?
>>
>> cheers,
>> renato
>
> Hi, yes saw Samplecat but could not get it to compile. Good effort but 
> to be honest the interface needs some work.
>
> So syntax changing is not a massive problem. I want to keep text 
> settings files and CLI tools as, in my opinion, writing in C++ all the 
> time and using libraries takes power away from the user (of course 
> there are appropriate uses for coding).
>
> DJ Barney
>


From robin at gareus.org  Fri Aug 22 12:33:50 2014
From: robin at gareus.org (Robin Gareus)
Date: Fri, 22 Aug 2014 14:33:50 +0200
Subject: [LAU] [ANN] xjadeo 0.8.0
Message-ID: <53F738AE.9090709@gareus.org>


Xjadeo is a video player that displays a video-clip in synchronized to
an external time source (MTC, LTC, JACK-transport).

http://xjadeo.sf.net/

-=-

Greetings Soundtrack Designers and fellow Multimedia Artists,

Xjadeo version 0.8.0 just came out and brings a lot of significant
changes. Most notably:

  * openGL display
  * video-frame indexing
  * built-in UI / context menu

With openGL, video-scaling is now performed in hardware and playback
synchronized to the screen's vertical refresh (if the hardware permits
that; most graphics cards do). This is the new default display and
supersedes prior platform-specific video outputs (XVideo, X11/imlib2,
SDL, quartz, which are still available via the --vo option and also used
a fallback).

Video files are now scanned and indexed on load which provides for
reliable seeking to video frames for a wide variety of codecs where
frame-accurate seeking was not possible with earlier versions of xjadeo.
This also acts as a guard to detect and refuse broken video files early on.

User interaction has been overhauled, most notably by adding a menu that
facilitates discovering key-bindings. This deprecates the external
control application qjadeo which previously came with xjadeo.


There have been over 200 changes since the last release, the complete
changelog is available at https://github.com/x42/xjadeo
Other highlights include:

  * separate On-Screen-Display for Sync-Source and Video Timecode
  * self-documenting OSC API
  * disable screensaver
  * 64 bit timeline
  * new website

Note that various command line options have changed. The seek-related
-K, -k parameters are no longer needed due to the change to indexing.
Letterbox is enabled by default, and it is now also possible to start
xjadeo without an initial file. In short, a lot of defaults have been
updated to make xjadeo more topical (despite that fact the the menu for
the X11 variant is plain old toolkit-less Xlib :)


Statically linked binaries are available for GNU/Linux, OSX and Windows
from http://xjadeo.sourceforge.net/download.html as is the source code
in terms of the GPLv2.

xjadeo is developed and has been tested for accuracy with ffmpeg-2.2.5
it may or may not work properly with different versions, but compiles
with any version of ffmpeg >= 1.0 to date.

Many thanks to Chris Goddard who provided valuable feedback and spent
several weeks on quality assurance and polishing user interaction. We're
far from done on the quest to 1.0, yet 0.8.0 marks a major milestone in
the life of xjadeo.


Cheers!
robin

From jonetsu at teksavvy.com  Sat Aug 23 20:07:33 2014
From: jonetsu at teksavvy.com (jonetsu at teksavvy.com)
Date: Sat, 23 Aug 2014 16:07:33 -0400
Subject: [LAU] How to kick Hydrogen off jack ?
Message-ID: <20140823160733.343b9dac@mistral>

Hello,

 I try to use Hydrogen's (0.9.6) tap tempo feature but it sticks to the
jack-defined bpm no matter what.  I made Ardour be the jack master and
changed the tempo to 130 and Hydrogen started at 130 after the tap
tempo count-in.  A slow tap tempo at that. 

Hydrogen does not display the two jack buttons that are to be under the
midi-in/CPU user interface widgets.  I haven't found any option in the
menus to tell Hydrogen to forget about jack.  What to do ?

Cheers.

From emailgrant at gmail.com  Sun Aug 24 13:08:32 2014
From: emailgrant at gmail.com (Grant)
Date: Sun, 24 Aug 2014 06:08:32 -0700
Subject: [LAU] ALSA: always use samplerate_best
Message-ID: <CAN0CFw2XLSLGA6ZLQrpkX4J7Vu99=nbcmsr2AuCtYmVU3NhkzQ@mail.gmail.com>

I have a USB DAC that can only handle 16/44.1 as input and output. I
think ALSA will resample everything to 16/44.1 automatically, but I'd
like that to happen with the highest quality resampler which I think
is samplerate_best. I use xfce4 and I don't want to install
pulseaudio. I've added the following to /etc/asound.conf:

defaults.pcm.rate_converter "samplerate_best"

Should that do it? How can I verify that it's working?

- Grant

From clemens at ladisch.de  Sun Aug 24 14:04:17 2014
From: clemens at ladisch.de (Clemens Ladisch)
Date: Sun, 24 Aug 2014 16:04:17 +0200
Subject: [LAU] ALSA: always use samplerate_best
In-Reply-To: <CAN0CFw2XLSLGA6ZLQrpkX4J7Vu99=nbcmsr2AuCtYmVU3NhkzQ@mail.gmail.com>
References: <CAN0CFw2XLSLGA6ZLQrpkX4J7Vu99=nbcmsr2AuCtYmVU3NhkzQ@mail.gmail.com>
Message-ID: <53F9F0E1.6040606@ladisch.de>

Grant wrote:
> I've added the following to /etc/asound.conf:
>
> defaults.pcm.rate_converter "samplerate_best"
>
> Should that do it?

Yes.  (If that converter is installed.)

> How can I verify that it's working?

Run aplay with -v:

  $ aplay -v something.wav
  Playing WAVE 'something.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
  Plug PCM: Rate conversion PCM (48000, sformat=S32_LE)
  Converter: libspeex (builtin)
  Protocol version: 10002
  Its setup is:
    ...


Regards,
Clemens

From len at ovenwerks.net  Sun Aug 24 14:43:53 2014
From: len at ovenwerks.net (Len Ovens)
Date: Sun, 24 Aug 2014 07:43:53 -0700 (PDT)
Subject: [LAU] ALSA: always use samplerate_best
In-Reply-To: <CAN0CFw2XLSLGA6ZLQrpkX4J7Vu99=nbcmsr2AuCtYmVU3NhkzQ@mail.gmail.com>
References: <CAN0CFw2XLSLGA6ZLQrpkX4J7Vu99=nbcmsr2AuCtYmVU3NhkzQ@mail.gmail.com>
Message-ID: <alpine.DEB.2.10.1408240724321.2503@scott.cbbs.org>

On Sun, 24 Aug 2014, Grant wrote:

> I have a USB DAC that can only handle 16/44.1 as input and output. I
> think ALSA will resample everything to 16/44.1 automatically, but I'd

Normally, the application connecting to ALSA looks at the port to find out 
what sample rates it can do and adjusts accordingly. Any recording 
application should lock to the interface rate with no resampleing. MP3s, 
Oggs and other compressed/lossy formats do internal resampling/filtering 
to match the desired output sampling rate anyway, but most of them are 
44.1k to begin with. Wav files and flac and other no lossy formates are 
the only ones where resampling is needed if they are not already 44.1k. In 
general any wav files will be ones you recorded and already be the right sample 
rate.

I think what I am saying is that for most cases the sample rate of your 
audio IF doesn't matter. So adding resampling to everything doesn't make 
sense... maybe try without first.

> like that to happen with the highest quality resampler which I think
> is samplerate_best. I use xfce4 and I don't want to install
> pulseaudio. I've added the following to /etc/asound.conf:
>
> defaults.pcm.rate_converter "samplerate_best"
>
> Should that do it? How can I verify that it's working?

Playback a 48K wav file. if there are no extra clicks or pops it is fine.

--
Len Ovens
www.ovenwerks.net


From dpchrist at holgerdanske.com  Sun Aug 24 17:26:12 2014
From: dpchrist at holgerdanske.com (David Christensen)
Date: Sun, 24 Aug 2014 10:26:12 -0700
Subject: [LAU] META how to search list archives?
Message-ID: <53FA2034.30803@holgerdanske.com>

linux-audio-user:

How do I search the archives for this mailing list?

     http://lists.linuxaudio.org/pipermail/linux-audio-user/



TIA,

David

From stephen.doonan at gmail.com  Sun Aug 24 17:33:27 2014
From: stephen.doonan at gmail.com (Stephen Doonan)
Date: Sun, 24 Aug 2014 11:33:27 -0600
Subject: [LAU] Play Time
In-Reply-To: <20140818191717.332bb894@debian>
References: <20140818191717.332bb894@debian>
Message-ID: <53FA21E7.7000709@gmail.com>

On 08/18/2014 12:17 PM, Will Godfrey wrote:

> I first set up a simple 1 bar loop as a click track and played the chords. Then
> played the bass line to this. When doing this I always play the base an octave
> up so I can hear it more clearly, then shift it down.
>
> Disposing of the click I then played the lead parts, and finally the counter
> melodies.


Interesting recording method. Thank you for sharing that. And very nice 
piece. :-)



From emailgrant at gmail.com  Sun Aug 24 17:38:53 2014
From: emailgrant at gmail.com (Grant)
Date: Sun, 24 Aug 2014 10:38:53 -0700
Subject: [LAU] ALSA: always use samplerate_best
In-Reply-To: <53F9F0E1.6040606@ladisch.de>
References: <CAN0CFw2XLSLGA6ZLQrpkX4J7Vu99=nbcmsr2AuCtYmVU3NhkzQ@mail.gmail.com>
 <53F9F0E1.6040606@ladisch.de>
Message-ID: <CAN0CFw02r=krAfOpw92DgkZRfJ2axNikkEtmXy29vfWAmJpqMw@mail.gmail.com>

>> I've added the following to /etc/asound.conf:
>>
>> defaults.pcm.rate_converter "samplerate_best"
>>
>> Should that do it?
>
> Yes.  (If that converter is installed.)
>
>> How can I verify that it's working?
>
> Run aplay with -v:
>
>   $ aplay -v something.wav
>   Playing WAVE 'something.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
>   Plug PCM: Rate conversion PCM (48000, sformat=S32_LE)
>   Converter: libspeex (builtin)
>   Protocol version: 10002
>   Its setup is:
>     ...


Thank you Clemens, I just needed to install alsa-utils (with
USE=libsamplerate) on Gentoo.

- Grant

From emailgrant at gmail.com  Sun Aug 24 17:48:34 2014
From: emailgrant at gmail.com (Grant)
Date: Sun, 24 Aug 2014 10:48:34 -0700
Subject: [LAU] ALSA: always use samplerate_best
In-Reply-To: <alpine.DEB.2.10.1408240724321.2503@scott.cbbs.org>
References: <CAN0CFw2XLSLGA6ZLQrpkX4J7Vu99=nbcmsr2AuCtYmVU3NhkzQ@mail.gmail.com>
 <alpine.DEB.2.10.1408240724321.2503@scott.cbbs.org>
Message-ID: <CAN0CFw2w99DRVExGtWHD96Uwyu-2aFiVyCAc97zbyKU-w_m5zA@mail.gmail.com>

>> I have a USB DAC that can only handle 16/44.1 as input and output. I
>> think ALSA will resample everything to 16/44.1 automatically, but I'd
>
>
> Normally, the application connecting to ALSA looks at the port to find out
> what sample rates it can do and adjusts accordingly. Any recording
> application should lock to the interface rate with no resampleing. MP3s,
> Oggs and other compressed/lossy formats do internal resampling/filtering to
> match the desired output sampling rate anyway, but most of them are 44.1k to
> begin with. Wav files and flac and other no lossy formates are the only ones
> where resampling is needed if they are not already 44.1k. In general any wav
> files will be ones you recorded and already be the right sample rate.
>
> I think what I am saying is that for most cases the sample rate of your
> audio IF doesn't matter. So adding resampling to everything doesn't make
> sense... maybe try without first.


I think you're saying that the ALSA resampler won't be used if the
upstream application does the resampling itself.  Is that correct?
How can I find out if ALSA is the one resampling in a particular
scenario?

- Grant

From ralf.mardorf at rocketmail.com  Sun Aug 24 17:49:00 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 24 Aug 2014 19:49:00 +0200
Subject: [LAU] [Bulk]  META how to search list archives?
In-Reply-To: <53FA2034.30803@holgerdanske.com>
References: <53FA2034.30803@holgerdanske.com>
Message-ID: <1408902540.26332.3.camel@rocketmail.com>

On Sun, 2014-08-24 at 10:26 -0700, David Christensen wrote:
> linux-audio-user:
> 
> How do I search the archives for this mailing list?
> 
>      http://lists.linuxaudio.org/pipermail/linux-audio-user/

This is hard to do. Perhaps I can help, since I've got much of it in my
MUA's folders. You could try search terms using https://startpage.com/
and when startpage guess Linux is porn, what usually happens after a
while, use http://www.google.com/ .


From harryhaaren at gmail.com  Sun Aug 24 17:59:33 2014
From: harryhaaren at gmail.com (Harry van Haaren)
Date: Sun, 24 Aug 2014 18:59:33 +0100
Subject: [LAU] META how to search list archives?
In-Reply-To: <53FA2034.30803@holgerdanske.com>
References: <53FA2034.30803@holgerdanske.com>
Message-ID: <CAKudYbMDwmT==7jWbnAXD7faA4jxvQY-hs+1OnhoSg42KKY5gA@mail.gmail.com>

The following are two ways to search:

http://lalists.stanford.edu/
http://linux-audio.4202.n7.nabble.com/


HTH, -Harry

On Sun, Aug 24, 2014 at 6:26 PM, David Christensen
<dpchrist at holgerdanske.com> wrote:
> linux-audio-user:
>
> How do I search the archives for this mailing list?
>
>     http://lists.linuxaudio.org/pipermail/linux-audio-user/
>
>
>
> TIA,
>
> David
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

From ralf.mardorf at rocketmail.com  Sun Aug 24 18:08:41 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 24 Aug 2014 20:08:41 +0200
Subject: [LAU] [Bulk] Re:  META how to search list archives?
In-Reply-To: <CAKudYbMDwmT==7jWbnAXD7faA4jxvQY-hs+1OnhoSg42KKY5gA@mail.gmail.com>
References: <53FA2034.30803@holgerdanske.com>
 <CAKudYbMDwmT==7jWbnAXD7faA4jxvQY-hs+1OnhoSg42KKY5gA@mail.gmail.com>
Message-ID: <1408903721.26332.5.camel@rocketmail.com>

On Sun, 2014-08-24 at 18:59 +0100, Harry van Haaren wrote:
> The following are two ways to search:
> 
> http://lalists.stanford.edu/
Using Google, but that's ok, there's no other way. I would try this one.

> http://linux-audio.4202.n7.nabble.com/
Beside the Google thingies, this side comes with tons of advertising
trackers. https://www.ghostery.com/en/ 



From jeremy at autostatic.com  Sun Aug 24 19:28:49 2014
From: jeremy at autostatic.com (Jeremy Jongepier)
Date: Sun, 24 Aug 2014 21:28:49 +0200
Subject: [LAU] META how to search list archives?
In-Reply-To: <53FA2034.30803@holgerdanske.com>
References: <53FA2034.30803@holgerdanske.com>
Message-ID: <53FA3CF1.5080006@autostatic.com>

On 08/24/2014 07:26 PM, David Christensen wrote:
> linux-audio-user:
> 
> How do I search the archives for this mailing list?
> 
>     http://lists.linuxaudio.org/pipermail/linux-audio-user/
> 
> 
> 
> TIA,
> 
> David

Google:
search term site:lists.linuxaudio.org/pipermail/linux-audio-user/

So for say Qtractor it would be:
qtractor site:lists.linuxaudio.org/pipermail/linux-audio-user/

Which yields:
https://www.google.nl/search?q=qtractor+site%3Alists.linuxaudio.org%2Fpipermail%2Flinux-audio-user%2F

Jeremy

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From dpchrist at holgerdanske.com  Sun Aug 24 20:42:48 2014
From: dpchrist at holgerdanske.com (David Christensen)
Date: Sun, 24 Aug 2014 13:42:48 -0700
Subject: [LAU] META how to search list archives?
In-Reply-To: <CAKudYbMDwmT==7jWbnAXD7faA4jxvQY-hs+1OnhoSg42KKY5gA@mail.gmail.com>
References: <53FA2034.30803@holgerdanske.com>
 <CAKudYbMDwmT==7jWbnAXD7faA4jxvQY-hs+1OnhoSg42KKY5gA@mail.gmail.com>
Message-ID: <53FA4E48.8090302@holgerdanske.com>

On 08/24/2014 10:59 AM, Harry van Haaren wrote:
> The following are two ways to search:
> http://lalists.stanford.edu/

The search box requires that the user leave in "site:lalists.stanford.edu".


On 08/24/2014 12:28 PM, Jeremy Jongepier wrote:
> Google:
> search term site:lists.linuxaudio.org/pipermail/linux-audio-user/

Yes, that's the old standby; same as above.


On 08/24/2014 10:59 AM, Harry van Haaren wrote:
> http://linux-audio.4202.n7.nabble.com/

Much better.  :-)


David


From dj_kaza at hotmail.com  Mon Aug 25 01:57:03 2014
From: dj_kaza at hotmail.com (Kaza Kore)
Date: Mon, 25 Aug 2014 01:57:03 +0000
Subject: [LAU] ALSA: always use samplerate_best
In-Reply-To: <alpine.DEB.2.10.1408240724321.2503@scott.cbbs.org>
References: <CAN0CFw2XLSLGA6ZLQrpkX4J7Vu99=nbcmsr2AuCtYmVU3NhkzQ@mail.gmail.com>,
 <alpine.DEB.2.10.1408240724321.2503@scott.cbbs.org>
Message-ID: <DUB113-W11956DC3221ABC6211049068FDF0@phx.gbl>



> Date: Sun, 24 Aug 2014 07:43:53 -0700
> From: len at ovenwerks.net
> To: emailgrant at gmail.com
> CC: linux-audio-user at lists.linuxaudio.org
> Subject: Re: [LAU] ALSA: always use samplerate_best
MP3s, 
> Oggs and other compressed/lossy formats do internal resampling/filtering 
> to match the desired output sampling rate anyway


What are you going on about? Data files do not do any internal resampling!!


On that note I've noticed my 48kHz mp3s do not play correctly on Rhythmbox, which uses GStreamer, but are fine in Audacious which doesn't. Rhythmbox is using the GStreamer Jack plugin as I have fully disabled PA though and wonder if this may be the issue... Anyway don't mean to derail the thread. ;)

Dale.
 		 	   		  
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From murks at tuxfamily.org  Mon Aug 25 10:16:24 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Mon, 25 Aug 2014 12:16:24 +0200
Subject: [LAU] gxtuner trouble and alternatives
Message-ID: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>

Hi there,

I'm looking for nice an simple tuning program for tuning some Ukuleles.
I stumbled upon gxtuner, which seems very nice, except that it doesn't
work for some reason.
I use the thing with jack. The mic definitely works, I can hear
everything trough the speakers if I make the appropriate connection.
However, gxtuner doesn't seem to be processing any audio, the needle
doesn't move and it doesn't show any notes. Any idea what could be
wrong? I'm on Arch Linux, gxtuner 2.1.

Any alternatives you can recommend?

Regards,
Philipp

From jmckernon at gmail.com  Mon Aug 25 10:41:13 2014
From: jmckernon at gmail.com (James Mckernon)
Date: Mon, 25 Aug 2014 11:41:13 +0100
Subject: [LAU] yoshimi --alsa-midi autoconnect?
Message-ID: <CAEByO6gDuxQoq-dzfBjKq_5q+WjKi9y6oHSLJSUYOrZabXH8eA@mail.gmail.com>

Hi all,

Yoshimi seems be able to autoconnect to a MIDI input device (e.g. a
keyboard) using its --alsa-midi (short form -a) option. But I can't seem to
get it to work. I'm not sure what string I should pass to identify

For example, running:

yoshimi --alsa-midi="microKONTROL MIDI 1"

(i.e. using the name which aconnect -o gives for the device) gives the
error:

ALSA lib seq.c:935:(snd_seq_open_noupdate) Unknown SEQ microKONTROL MIDI 1
Error, failed to open alsa midi device: microKONTROL MIDI 1

Any ideas how I should identify my keyboard? (OR does this feature simply
not work at the moment?)

Thanks!
J
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From ralf.mardorf at rocketmail.com  Mon Aug 25 11:23:56 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Mon, 25 Aug 2014 13:23:56 +0200
Subject: [LAU] gxtuner trouble and alternatives
In-Reply-To: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
Message-ID: <1408965836.29920.32.camel@rocketmail.com>

On Mon, 2014-08-25 at 12:16 +0200, Philipp ?berbacher wrote:
> I'm on Arch Linux, gxtuner 2.1.

[rocketmouse at archlinux ~]$ pacman -Q gxtuner
gxtuner 2.1-1
[rocketmouse at archlinux ~]$ pacman -Q jack2
jack2 1.9.10-1

Here gxtuner does recognize my sine waves (voice init) from a DX7, but
high frequencies are ignored and btw. for the frequencies that are
recognised, gxtuner or my DX7 is very inaccurate.

My Boss TU-12H is able to notice all frequencies from the DX7 and claims
that all frequencies are accurate. My hearing claims that the Boss
tuner's claim is ok and that gxtuner is mistake.

Some years ago I tested several tuners. The Boss TU-12H is one of the
best I know, while many other tuners were completely unusable. One Linux
tuner I tested was very good too, IIRC it was fmit but it doesn't work
anymore since years, perhaps it was another Linux tuner. I bought and
tested many tuners, the Boss TU-12H and another one from Korg (not
available anymore) are the only tuners that fit to my taste ... yes, one
Linux tuner, perhaps fmit fit to my needs too, but it stopped working
years ago. I don't test tuners all the times, I simply stay with the
tuner that fits to my needs. You should test any instrument tuner
available for Linux, some might be crap and others might do what you
want.


From willgodfrey at musically.me.uk  Mon Aug 25 11:44:03 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Mon, 25 Aug 2014 12:44:03 +0100
Subject: [LAU] yoshimi --alsa-midi autoconnect?
In-Reply-To: <CAEByO6gDuxQoq-dzfBjKq_5q+WjKi9y6oHSLJSUYOrZabXH8eA@mail.gmail.com>
References: <CAEByO6gDuxQoq-dzfBjKq_5q+WjKi9y6oHSLJSUYOrZabXH8eA@mail.gmail.com>
Message-ID: <20140825124403.2b28b37f@debian>

On Mon, 25 Aug 2014 11:41:13 +0100
James Mckernon <jmckernon at gmail.com> wrote:

> Hi all,
> 
> Yoshimi seems be able to autoconnect to a MIDI input device (e.g. a
> keyboard) using its --alsa-midi (short form -a) option. But I can't seem to
> get it to work. I'm not sure what string I should pass to identify
> 
> For example, running:
> 
> yoshimi --alsa-midi="microKONTROL MIDI 1"
> 
> (i.e. using the name which aconnect -o gives for the device) gives the
> error:
> 
> ALSA lib seq.c:935:(snd_seq_open_noupdate) Unknown SEQ microKONTROL MIDI 1
> Error, failed to open alsa midi device: microKONTROL MIDI 1
> 
> Any ideas how I should identify my keyboard? (OR does this feature simply
> not work at the moment?)
> 
> Thanks!
> J

You've found a bug :(

I've always used qjackctl so hadn't noticed it. It seems the actual error is
being generated by ALSA itself, but I don't know if it's getting a badly formed
call or what.

I'll try and investigate ASAP.

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From murks at tuxfamily.org  Mon Aug 25 12:25:39 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Mon, 25 Aug 2014 14:25:39 +0200
Subject: [LAU] gxtuner trouble and alternatives
In-Reply-To: <1408965836.29920.32.camel@rocketmail.com>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
 <1408965836.29920.32.camel@rocketmail.com>
Message-ID: <20140825142539.021197c9@eeyore.mozart.uni-klu.ac.at>

On Mon, 25 Aug 2014 13:23:56 +0200
Ralf Mardorf <ralf.mardorf at rocketmail.com> wrote:

> On Mon, 2014-08-25 at 12:16 +0200, Philipp ?berbacher wrote:
> > I'm on Arch Linux, gxtuner 2.1.
> 
> [rocketmouse at archlinux ~]$ pacman -Q gxtuner
> gxtuner 2.1-1
> [rocketmouse at archlinux ~]$ pacman -Q jack2
> jack2 1.9.10-1
> 
> Here gxtuner does recognize my sine waves (voice init) from a DX7, but
> high frequencies are ignored and btw. for the frequencies that are
> recognised, gxtuner or my DX7 is very inaccurate.
> 
> My Boss TU-12H is able to notice all frequencies from the DX7 and
> claims that all frequencies are accurate. My hearing claims that the
> Boss tuner's claim is ok and that gxtuner is mistake.
> 
> Some years ago I tested several tuners. The Boss TU-12H is one of the
> best I know, while many other tuners were completely unusable. One
> Linux tuner I tested was very good too, IIRC it was fmit but it
> doesn't work anymore since years, perhaps it was another Linux tuner.
> I bought and tested many tuners, the Boss TU-12H and another one from
> Korg (not available anymore) are the only tuners that fit to my
> taste ... yes, one Linux tuner, perhaps fmit fit to my needs too, but
> it stopped working years ago. I don't test tuners all the times, I
> simply stay with the tuner that fits to my needs. You should test any
> instrument tuner available for Linux, some might be crap and others
> might do what you want.

Thanks Ralf, I tried lingot and it seems to be OK. Not as fancy looking
but it nicely shows notes, cents and frequency. No problem with jack
either. I still wonder what is wrong with gxtuner though.

Regards,
Philipp

From ralf.mardorf at rocketmail.com  Mon Aug 25 12:35:49 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Mon, 25 Aug 2014 14:35:49 +0200
Subject: [LAU] [Bulk]  gxtuner trouble and alternatives
In-Reply-To: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
Message-ID: <1408970149.1780.1.camel@rocketmail.com>

On Mon, 2014-08-25 at 12:16 +0200, Philipp ?berbacher wrote:
> Any alternatives you can recommend?

Recommendations are hard to do ;). I love the TU-12H temperature, I love
atonal music, based on the temperature similar or equal to this tuner's
temperature. I dislike to tune an instrument without a tuner, it's hard
to do and takes much time. Some instruments can't be tuned by tuners,
using a tuner is impossible to tune e.g. a (grand) piano ... you need to
mute strings during tuning ... "hell". When I was a guitarist some years
ago, tuning a guitar wasn't pleasant ;). Gxtuner seemingly isn't made to
tune the high frequency instruments. To find a tuner that works is hard
to do. Tune the instrument by hearing, assumed you don't want to sync
synth. When using natural instruments and synthetic instruments, then
the tuning is one of the important parts of mixing the composition ...
only trust your hearing and spend much time to fit the temperatures of
one instrument to another. It's painful, hard to do, the opposite of
fun :D.

I hate tuning instruments, I'm not good in doing this, but people always
are surprised that the MIDI gear does fit that good to the tuning of the
natural instruments of my recordings. Amen. In the early days I tuned
guitars by standard tuning (Guitar at 440 Hz only) using a Korg GT-6J (I
still own it and it still should work), later until today I'm using the
Boss TU-12H for all kinds of instruments with the limited pitch range
from A = 440 to 445.

There's no perfect tuning, but a temperature that fit close to the
feeling and even if you find a tuner that fits to your feeling regarding
the temperature, it might not be able to provide the wanted tuning < or
> 440 Hz.


From ralf.mardorf at rocketmail.com  Mon Aug 25 12:41:58 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Mon, 25 Aug 2014 14:41:58 +0200
Subject: [LAU] [Bulk] Re:  gxtuner trouble and alternatives
In-Reply-To: <20140825142539.021197c9@eeyore.mozart.uni-klu.ac.at>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
 <1408965836.29920.32.camel@rocketmail.com>
 <20140825142539.021197c9@eeyore.mozart.uni-klu.ac.at>
Message-ID: <1408970518.1780.3.camel@rocketmail.com>

On Mon, 2014-08-25 at 14:25 +0200, Philipp ?berbacher wrote:
> Thanks Ralf, I tried lingot and it seems to be OK. Not as fancy
> looking but it nicely shows notes, cents and frequency. No problem
> with jack either. I still wonder what is wrong with gxtuner though.

Likely nothing is wrong with this tuner, tuners are problematic. Thank
you, I'll test lingot soon. It's not pleasant to insert the Boss tuner I
prefer all the times.



From clemens at ladisch.de  Mon Aug 25 12:57:09 2014
From: clemens at ladisch.de (Clemens Ladisch)
Date: Mon, 25 Aug 2014 14:57:09 +0200
Subject: [LAU] yoshimi --alsa-midi autoconnect?
In-Reply-To: <20140825124403.2b28b37f@debian>
References: <CAEByO6gDuxQoq-dzfBjKq_5q+WjKi9y6oHSLJSUYOrZabXH8eA@mail.gmail.com>
 <20140825124403.2b28b37f@debian>
Message-ID: <53FB32A5.7030005@ladisch.de>

Will Godfrey wrote:
> James Mckernon <jmckernon at gmail.com> wrote:
>> yoshimi --alsa-midi="microKONTROL MIDI 1"
>>
>> ALSA lib seq.c:935:(snd_seq_open_noupdate) Unknown SEQ microKONTROL MIDI 1

The device name for snd_seq_open() must always be "default".

This option should change the client/port name.
(There is snd_seq_parse_address() for that.)


Regards,
Clemens

From ralf.mardorf at rocketmail.com  Mon Aug 25 13:09:11 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Mon, 25 Aug 2014 15:09:11 +0200
Subject: [LAU] [Bulk] Re:  gxtuner trouble and alternatives
In-Reply-To: <20140825142539.021197c9@eeyore.mozart.uni-klu.ac.at>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
 <1408965836.29920.32.camel@rocketmail.com>
 <20140825142539.021197c9@eeyore.mozart.uni-klu.ac.at>
Message-ID: <1408972151.1780.11.camel@rocketmail.com>

"Ukulele" ;). For playing chords or scales? I never used an ukulele, but
tuning for chords is more important, than for playing melodies. Octave
uncleanness of the instrument is less important for tunes, but a PITA
for chord comping. Fortunately I prefer power chords over suspended
chords, IMO tuning already is a PITA when playing punk-rock, but at
least an issue for jazz comping.


From willgodfrey at musically.me.uk  Mon Aug 25 14:12:55 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Mon, 25 Aug 2014 15:12:55 +0100
Subject: [LAU] yoshimi --alsa-midi autoconnect?
In-Reply-To: <53FB32A5.7030005@ladisch.de>
References: <CAEByO6gDuxQoq-dzfBjKq_5q+WjKi9y6oHSLJSUYOrZabXH8eA@mail.gmail.com>
 <20140825124403.2b28b37f@debian> <53FB32A5.7030005@ladisch.de>
Message-ID: <20140825151255.45e4e37b@debian>

On Mon, 25 Aug 2014 14:57:09 +0200
Clemens Ladisch <clemens at ladisch.de> wrote:

> Will Godfrey wrote:
> > James Mckernon <jmckernon at gmail.com> wrote:
> >> yoshimi --alsa-midi="microKONTROL MIDI 1"
> >>
> >> ALSA lib seq.c:935:(snd_seq_open_noupdate) Unknown SEQ microKONTROL MIDI 1
> 
> The device name for snd_seq_open() must always be "default".
> 
> This option should change the client/port name.
> (There is snd_seq_parse_address() for that.)
> 
> 
> Regards,
> Clemens

Thanks. You've probably saved me  a lot of time :)

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From willgodfrey at musically.me.uk  Mon Aug 25 14:13:40 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Mon, 25 Aug 2014 15:13:40 +0100
Subject: [LAU] Play Time
In-Reply-To: <53FA21E7.7000709@gmail.com>
References: <20140818191717.332bb894@debian>
	<53FA21E7.7000709@gmail.com>
Message-ID: <20140825151340.5da4f019@debian>

On Sun, 24 Aug 2014 11:33:27 -0600
Stephen Doonan <stephen.doonan at gmail.com> wrote:

> On 08/18/2014 12:17 PM, Will Godfrey wrote:
> 
> > I first set up a simple 1 bar loop as a click track and played the chords. Then
> > played the bass line to this. When doing this I always play the base an octave
> > up so I can hear it more clearly, then shift it down.
> >
> > Disposing of the click I then played the lead parts, and finally the counter
> > melodies.
> 
> 
> Interesting recording method. Thank you for sharing that. And very nice 
> piece. :-)

Pleasantly surprised to get another reply to this :) 

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From brummer- at web.de  Mon Aug 25 14:51:15 2014
From: brummer- at web.de (Hermann Meyer)
Date: Mon, 25 Aug 2014 16:51:15 +0200
Subject: [LAU] gxtuner trouble and alternatives
In-Reply-To: <20140825142539.021197c9@eeyore.mozart.uni-klu.ac.at>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
 <1408965836.29920.32.camel@rocketmail.com>
 <20140825142539.021197c9@eeyore.mozart.uni-klu.ac.at>
Message-ID: <ba1ccb79-2f3e-4c4c-9cb6-7162c830ddbe@email.android.com>

You need to run jack in realtime mode to use gx_tuner. ?f jack is already rt, try a higher level. Gx_tuner use a lower rt-level then jack, if jack use a to low level, the pitchtracker thread wont be finished in time.
-- 
Diese Nachricht wurde von meinem Android Mobiltelefon mit WEB.DE Mail gesendet.



"Philipp ?berbacher" <murks at tuxfamily.org> schrieb:

On Mon, 25 Aug 2014 13:23:56 +0200
Ralf Mardorf <ralf.mardorf at rocketmail.com> wrote:

> On Mon, 2014-08-25 at 12:16 +0200, Philipp ?berbacher wrote:
> > I'm on Arch Linux, gxtuner 2.1.
> 
> [rocketmouse at archlinux ~]$ pacman -Q gxtuner
> gxtuner 2.1-1
> [rocketmouse at archlinux ~]$ pacman -Q jack2
> jack2 1.9.10-1
> 
> Here gxtuner does recognize my sine waves (voice init) from a DX7, but
> high frequencies are ignored and btw. for the frequencies that are
> recognised, gxtuner or my DX7 is very inaccurate.
> 
> My Boss TU-12H is able to notice all frequencies from the DX7 and
> claims that all frequencies are accurate. My hearing claims that the
> Boss tuner's claim is ok and that gxtuner is mistake.
> 
> Some years ago I tested several tuners. The Boss TU-12H is one of the
> best I know, while many other tuners were completely unusable. One
> Linux tuner I tested was very good too, IIRC it was fmit but it
> doesn't work anymore since years, perhaps it was another Linux tuner.
> I bought and tested many tuners, the Boss TU-12H and another one from
> Korg (not available anymore) are the only tuners that fit to my
> taste ... yes, one Linux tuner, perhaps fmit fit to my needs too, but
> it stopped working years ago. I don't test tuners all the times, I
> simply stay with the tuner that fits to my needs. You should test any
> instrument tuner available for Linux, some might be crap and others
> might do what you want.

Thanks Ralf, I tried lingot and it seems to be OK. Not as fancy looking
but it nicely shows notes, cents and frequency. No problem with jack
either. I still wonder what is wrong with gxtuner though.

Regards,
Philipp
_____________________________________________

Linux-audio-user mailing list
Linux-audio-user at lists.linuxaudio.org
http://lists.linuxaudio.org/listinfo/linux-audio-user

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From len at ovenwerks.net  Mon Aug 25 15:25:01 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 25 Aug 2014 08:25:01 -0700 (PDT)
Subject: [LAU] gxtuner trouble and alternatives
In-Reply-To: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
Message-ID: <alpine.DEB.2.10.1408250758300.2503@scott.cbbs.org>

On Mon, 25 Aug 2014, Philipp ?berbacher wrote:

> I'm looking for nice an simple tuning program for tuning some Ukuleles.
> I stumbled upon gxtuner, which seems very nice, except that it doesn't
> work for some reason.

Generally I use guitarix, but that would be overkill for what you are 
doing, So I installed some tuners:

lingot: I like it, the meter is very responsive without being too jittery. 
Works with jack. I will probably keep this.

gxtuner: this works for me too, a bit more needle jitter than lingot, but 
still very usable. Works with jack. No menus or settings besides the two 
in the bottom corners.

gtkguitune: I could not get this to work. (I did not try very hard) It did 
not create a jack port nor does it have any way of choosing an audio 
device. For me this is useless as I run Jack all the time and do not want 
to stop jack just to tune. I do recognize the GUI from times past and seem 
to recall it was hard to use anyway, being too sensitive and jittery to 
see what I was doing.

You may prefer lingot if only because you have visual feedback that there 
is signal present even if there is not enough information for tuning 
purposes, but gxtuner should work with jack just fine.

I am finding out why my guitar is so hard to play.... The intonation is 
really bad... E is flat while G (two frets up) is sharp :P  This guitar 
uses a fret for the nut so this may be hard to fix.



--
Len Ovens
www.ovenwerks.net

From harryhaaren at gmail.com  Mon Aug 25 15:30:41 2014
From: harryhaaren at gmail.com (Harry van Haaren)
Date: Mon, 25 Aug 2014 16:30:41 +0100
Subject: [LAU] gxtuner trouble and alternatives
In-Reply-To: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
Message-ID: <CAKudYbO6JyYh0SPOrfg=hOoh=D6-Yj9HknrADtwukFiW6-ZSTw@mail.gmail.com>

On Mon, Aug 25, 2014 at 11:16 AM, Philipp ?berbacher
<murks at tuxfamily.org> wrote:
> Any alternatives you can recommend?

Robin Gareus' Tuna.lv2:
https://github.com/x42/tuna.lv2

PS: Thanks Robin, its awesome :)

From brummer- at web.de  Mon Aug 25 18:36:54 2014
From: brummer- at web.de (hermann meyer)
Date: Mon, 25 Aug 2014 20:36:54 +0200
Subject: [LAU] gxtuner trouble and alternatives
In-Reply-To: <20140825142539.021197c9@eeyore.mozart.uni-klu.ac.at>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
 <1408965836.29920.32.camel@rocketmail.com>
 <20140825142539.021197c9@eeyore.mozart.uni-klu.ac.at>
Message-ID: <53FB8246.4070003@web.de>


Am 25.08.2014 14:25, schrieb Philipp ?berbacher:
> I still wonder what is wrong with gxtuner though.
Gxtuner use a lower priory then jack, to avoid conflicts with really 
real-time apps. So, if you use jack with a to low rt-priory, gxtuner 
wouldn't work, because the pitch tracker thread won't finish before the 
deadline.

Gxtuner work well for stringed instruments.

However, wouldn't comment on the rest of the thread.

by
hermann

From nabob_cd at yahoo.com  Mon Aug 25 19:13:25 2014
From: nabob_cd at yahoo.com (Menno)
Date: Mon, 25 Aug 2014 12:13:25 -0700 (PDT)
Subject: [LAU] Delta1010: S/PDIF out has sync but no audio
In-Reply-To: <1408449725804-92255.post@n7.nabble.com>
References: <1408361569528-92201.post@n7.nabble.com>
 <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
 <1408440351607-92247.post@n7.nabble.com>
 <y2ak364bt4a.fsf@cartman.at.offog.org>
 <1408449725804-92255.post@n7.nabble.com>
Message-ID: <1408994005915-92386.post@n7.nabble.com>

i made progress with my KXStudio 14.04 system.
i removed jack1, replaced it with jack2 (thanks faltx!). Much better!

Build and installed Mudita24 1.1.0 (that is not in the repo yet)

I think my system runs solid now. There is only one weird thing left :P
-i have a sync signal on the Delta's 1010 S/PDIF out, but there is still no
sound.

The weird thing is, that when i run jack and make a connection to the S/PDIF
Hardware playback 9 and 10, i see the meters going up and down - but i there
is no sound, only sync signal.
And:
the analog outputs playback 1 and 2 have the music instead, even if i have
cut off the connection to Hardware 1 and 2. 

Could it be that this is a software bug? Has hardware playback 9 and 10 by
accident become playback 1 and 2 by a simple copy and paste command in the
software?







--
View this message in context: http://linux-audio.4202.n7.nabble.com/Delta1010-S-PDIF-out-has-sync-but-no-audio-tp92201p92386.html
Sent from the linux-audio-user mailing list archive at Nabble.com.

From brummer- at web.de  Mon Aug 25 20:28:30 2014
From: brummer- at web.de (hermann meyer)
Date: Mon, 25 Aug 2014 22:28:30 +0200
Subject: [LAU] gxtuner trouble and alternatives
In-Reply-To: <20140825142539.021197c9@eeyore.mozart.uni-klu.ac.at>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
 <1408965836.29920.32.camel@rocketmail.com>
 <20140825142539.021197c9@eeyore.mozart.uni-klu.ac.at>
Message-ID: <53FB9C6E.3070404@web.de>


Am 25.08.2014 14:25, schrieb Philipp ?berbacher:
>   I still wonder what is wrong with gxtuner though.


You properly need to increase the jack rt-priory. gxtuner use a lower 
priory then jack, to avoid any trouble with other jack clients, a tuner 
should be the first to step back. If jack runs with a to low rt-priory 
(or without), gxtuner wouldn't work, because the pitch-tracker thread 
wouldn't finished before timeout.

Regarding incorrectness of the tuner, it is ~2 cents.

best
hermann


From brent at keycorner.org  Mon Aug 25 21:43:24 2014
From: brent at keycorner.org (Brent Busby)
Date: Mon, 25 Aug 2014 16:43:24 -0500 (CDT)
Subject: [LAU] gxtuner trouble and alternatives
In-Reply-To: <alpine.DEB.2.10.1408250758300.2503@scott.cbbs.org>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
 <alpine.DEB.2.10.1408250758300.2503@scott.cbbs.org>
Message-ID: <alpine.LNX.2.00.1408251637340.7405@village.keycorner.org>

On Mon, 25 Aug 2014, Len Ovens wrote:

> On Mon, 25 Aug 2014, Philipp ?berbacher wrote:
>
> lingot: I like it, the meter is very responsive without being too 
> jittery. Works with jack. I will probably keep this.

This is my favorite one, except for its bugs.  I like that it's so good 
at figuring out the fundamental frequency even when there are a *lot* of 
harmonics.  Also, it's very fast.  If music with 16th-notes is played 
into it, it will literally figure out the scale notes as the 16th-notes 
go by without moving the meter.  I've seen hardware tuners that couldn't 
do that.

However, there's a noticeable CPU load even when it's just sitting there 
with no Jack connection sent to it, and it actually causes xruns when 
there are any Jack connections to it.  (I will disconnect it when I'm 
not actively tuning just to keep it from causing an xrun.)

I'd probably keep it running all the time to keep my Moog's wandering 
pitch tracking in check if it weren't for the xrun/cpu issues.

-- 
+ Brent A. Busby	 + "We've all heard that a million monkeys
+ Sr. UNIX Systems Admin +  banging on a million typewriters will
+ University of Chicago	 +  eventually reproduce the entire works of
+ James Franck Institute +  Shakespeare.  Now, thanks to the Internet,
+ Materials Research Ctr +  we know this is not true." -Robert Wilensky

From ralf.mardorf at rocketmail.com  Mon Aug 25 23:21:56 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 01:21:56 +0200
Subject: [LAU] [Bulk] Re:  gxtuner trouble and alternatives
In-Reply-To: <alpine.LNX.2.00.1408251637340.7405@village.keycorner.org>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
 <alpine.DEB.2.10.1408250758300.2503@scott.cbbs.org>
 <alpine.LNX.2.00.1408251637340.7405@village.keycorner.org>
Message-ID: <1409008916.1780.13.camel@rocketmail.com>

On Mon, 2014-08-25 at 16:43 -0500, Brent Busby wrote:
> I'd probably keep it running all the time to keep my Moog's wandering 
> pitch tracking in check if it weren't for the xrun/cpu issues.

AT1 isn't resource hungry. Assumed the Moog is just a little bit out of
tune, AT1 might correct it without side effects.

http://kokkinizita.linuxaudio.org/linuxaudio/zita-at1-doc/quickguide.html




From ralf.mardorf at rocketmail.com  Mon Aug 25 23:26:40 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 01:26:40 +0200
Subject: [LAU] [Bulk] Re:  [Bulk] Re:  gxtuner trouble and alternatives
In-Reply-To: <1409008916.1780.13.camel@rocketmail.com>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
 <alpine.DEB.2.10.1408250758300.2503@scott.cbbs.org>
 <alpine.LNX.2.00.1408251637340.7405@village.keycorner.org>
 <1409008916.1780.13.camel@rocketmail.com>
Message-ID: <1409009200.1780.14.camel@rocketmail.com>

On Tue, 2014-08-26 at 01:21 +0200, Ralf Mardorf wrote:
> On Mon, 2014-08-25 at 16:43 -0500, Brent Busby wrote:
> > I'd probably keep it running all the time to keep my Moog's wandering 
> > pitch tracking in check if it weren't for the xrun/cpu issues.
> 
> AT1 isn't resource hungry. Assumed the Moog is just a little bit out of
> tune, AT1 might correct it without side effects.
> 
> http://kokkinizita.linuxaudio.org/linuxaudio/zita-at1-doc/quickguide.html

It might conflict with portamento ;).


From len at ovenwerks.net  Tue Aug 26 00:37:02 2014
From: len at ovenwerks.net (Len Ovens)
Date: Mon, 25 Aug 2014 17:37:02 -0700 (PDT)
Subject: [LAU] Delta1010: S/PDIF out has sync but no audio
In-Reply-To: <1408994005915-92386.post@n7.nabble.com>
References: <1408361569528-92201.post@n7.nabble.com>
 <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
 <1408440351607-92247.post@n7.nabble.com>
 <y2ak364bt4a.fsf@cartman.at.offog.org>
 <1408449725804-92255.post@n7.nabble.com>
 <1408994005915-92386.post@n7.nabble.com>
Message-ID: <alpine.DEB.2.10.1408251259100.2503@scott.cbbs.org>

On Mon, 25 Aug 2014, Menno wrote:

> I think my system runs solid now. There is only one weird thing left :P
> -i have a sync signal on the Delta's 1010 S/PDIF out, but there is still no
> sound.
>
> The weird thing is, that when i run jack and make a connection to the S/PDIF
> Hardware playback 9 and 10, i see the meters going up and down - but i there
> is no sound, only sync signal.
> And:
> the analog outputs playback 1 and 2 have the music instead, even if i have
> cut off the connection to Hardware 1 and 2.
>
> Could it be that this is a software bug? Has hardware playback 9 and 10 by
> accident become playback 1 and 2 by a simple copy and paste command in the
> software?

I wish I could say, but I have no s/pdif in amplifier. That is no way to 
monitor the s/pdif out.

Looking at the Patchbay/Router in mudita24 (1.0.4) the last two columns 
are s/pdif out and on mine they are set to s/pdif out. The fist way to 
test, is to change this so that they are both set to H/W In 1 and put some 
audio into channel one (no jack needed for this test, but a good, high 
level yes) and see if you then get audio out from s/pdif. If you don't:

  - mudita sw bug
  - alsa sw bug
  - d1010 fw bug
  - d1010 internal hw problem.
  - d1010 Hardware settings mistake.

Are the things that would make sense. The Hardware Settings tab in mudita 
has S/pdif Output settings. Try both Professional and Consumer settings. 
Make sure Data mode is Audio. In consumer set copy permitted and original. 
I am not sure what your amplifier needs as input... it may only look at 
Copyrighted/Original or some odd thing. I don't know if I can tell 
anything from scoping mine as my scope is plain analog no digital 
functions so it may be difficult to sync a zero DC signal. (I have a d66 
with no word clock)

SW/FW bugs are unlikely as I would expect others would have mentioned that 
already, there are a lot of d1010s around. (lots of delta series boxes in 
general) I would expect you are not the only person who has tried to use 
s/pdif out.

--
Len Ovens
www.ovenwerks.net


From ralf.mardorf at rocketmail.com  Tue Aug 26 00:50:28 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 02:50:28 +0200
Subject: [LAU] [Bulk] Re:  gxtuner trouble and alternatives
In-Reply-To: <alpine.DEB.2.10.1408250758300.2503@scott.cbbs.org>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
 <alpine.DEB.2.10.1408250758300.2503@scott.cbbs.org>
Message-ID: <1409014228.1780.20.camel@rocketmail.com>

On Mon, 2014-08-25 at 08:25 -0700, Len Ovens wrote:
> lingot: I like it, the meter is very responsive without being too
> jittery.

When tuning guitars with a meter that is too jittery, a flageolet above
the 12th fret (octave harmonic) does the trick. "flageolet" in
translation seems to be fishy: don't push the string on the fret, just
touch it with the finger, without contact to the fret. Anyway, you need
a jitter free tuner to adjust your guitar's octave cleanness. Many e.
guitars of amateur guitarists suffer from disgusting unclean octaves.
It's easy to adjust an e. guitar, unfortunately adjusting a classical
guitar isn't that easy, you need to kill an elephant and hone down
several ivory insets for the bridge or something like this ;) ... I
heard they sell inserts made of plastics to adjust unclean octaves ;).


From ralf.mardorf at rocketmail.com  Tue Aug 26 01:20:42 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 03:20:42 +0200
Subject: [LAU] [Bulk] Re:  gxtuner trouble and alternatives
In-Reply-To: <alpine.DEB.2.10.1408250758300.2503@scott.cbbs.org>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
 <alpine.DEB.2.10.1408250758300.2503@scott.cbbs.org>
Message-ID: <1409016042.1780.24.camel@rocketmail.com>

Btw. take a look to the bridge of photos from Jimi Hendrix Stratocaster
and forward/backward the thingies on your e. guitars bridge as shown by
the photo. I suspect Jimi Hendrix live tuning not only suffered from the
kind of whammy bar he used ;).


From ralf.mardorf at rocketmail.com  Tue Aug 26 02:04:52 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 04:04:52 +0200
Subject: [LAU] [Bulk] Re:  Delta1010: S/PDIF out has sync but no audio
In-Reply-To: <alpine.DEB.2.10.1408251259100.2503@scott.cbbs.org>
References: <1408361569528-92201.post@n7.nabble.com>
 <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
 <1408440351607-92247.post@n7.nabble.com>
 <y2ak364bt4a.fsf@cartman.at.offog.org>
 <1408449725804-92255.post@n7.nabble.com>
 <1408994005915-92386.post@n7.nabble.com>
 <alpine.DEB.2.10.1408251259100.2503@scott.cbbs.org>
Message-ID: <1409018692.1780.26.camel@rocketmail.com>

On Mon, 2014-08-25 at 17:37 -0700, Len Ovens wrote:
> I wish I could say, but I have no s/pdif in amplifier. That is no way to 
> monitor the s/pdif out.

I own two TerraTec Envy24 cards and I could test S/PDIF out connected to
a DAT recorder's S/PDIF in. Not right now, but in a few weeks. Sorry! 


From ralf.mardorf at rocketmail.com  Tue Aug 26 02:35:56 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 04:35:56 +0200
Subject: [LAU] [Bulk] Re:  gxtuner trouble and alternatives
In-Reply-To: <alpine.DEB.2.10.1408250758300.2503@scott.cbbs.org>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
 <alpine.DEB.2.10.1408250758300.2503@scott.cbbs.org>
Message-ID: <1409020556.1780.28.camel@rocketmail.com>

On Mon, 2014-08-25 at 08:25 -0700, Len Ovens wrote:
> Generally I use guitarix, but that would be overkill

Generally I use analog gear for my guitar, but I tested several computer
based effects and IMO rakarrack provides a much easier to use GUI. I
can't say much about the quality of the effects or about the tuner's
quality, but usage of rakarrack is idiot prove, while using guitarix
seems to be for Linux experts only.


From brummer- at web.de  Tue Aug 26 07:55:23 2014
From: brummer- at web.de (Hermann Meyer)
Date: Tue, 26 Aug 2014 09:55:23 +0200
Subject: [LAU] [Bulk] Re:  gxtuner trouble and alternatives
In-Reply-To: <1409020556.1780.28.camel@rocketmail.com>
References: <20140825121624.60171c2a@eeyore.mozart.uni-klu.ac.at>
 <alpine.DEB.2.10.1408250758300.2503@scott.cbbs.org>
 <1409020556.1780.28.camel@rocketmail.com>
Message-ID: <7ad9dfcc-abf9-4b03-82b4-468533b89444@email.android.com>

http://libremusicproduction.com/articles/ultimate-guide-getting-started-guitarix

-- 
Diese Nachricht wurde von meinem Android Mobiltelefon mit WEB.DE Mail gesendet.



Ralf Mardorf <ralf.mardorf at rocketmail.com> schrieb:

On Mon, 2014-08-25 at 08:25 -0700, Len Ovens wrote:
> Generally I use guitarix, but that would be overkill

Generally I use analog gear for my guitar, but I tested several computer
based effects and IMO rakarrack provides a much easier to use GUI. I
can't say much about the quality of the effects or about the tuner's
quality, but usage of rakarrack is idiot prove, while using guitarix
seems to be for Linux experts only.

_____________________________________________

Linux-audio-user mailing list
Linux-audio-user at lists.linuxaudio.org
http://lists.linuxaudio.org/listinfo/linux-audio-user

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From nescivi at gmail.com  Tue Aug 26 08:15:25 2014
From: nescivi at gmail.com (nescivi)
Date: Tue, 26 Aug 2014 10:15:25 +0200
Subject: [LAU] Creative Music Coding lab on August 26, STEIM, Amsterdam
In-Reply-To: <53EB3534.2000306@gmail.com>
References: <53EB3534.2000306@gmail.com>
Message-ID: <53FC421D.80906@gmail.com>

Hiho,

this is tonight!

sincerely,
Marije

On 13-08-14 11:51, nescivi wrote:
> Hi all,
> 
> On August 26 we welcome again all creative music coders at STEIM for an
> evening of exchanging current work, problems and solutions - and music
> together.
> 
> More information:
> http://steim.org/event/creative-music-coding-lab-13/
> Entrance is free.
> 
> And let us know if you plan to join (just to get an idea of how many
> seats, and how much coffee and tea we should prepare)!
> 
> sincerely,
> Marije
> 


From ralf.mardorf at rocketmail.com  Tue Aug 26 11:24:04 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 13:24:04 +0200
Subject: [LAU] [Bulk] Re:  Delta1010: S/PDIF out has sync but no audio
In-Reply-To: <alpine.DEB.2.10.1408251259100.2503@scott.cbbs.org>
References: <1408361569528-92201.post@n7.nabble.com>
 <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
 <1408440351607-92247.post@n7.nabble.com>
 <y2ak364bt4a.fsf@cartman.at.offog.org>
 <1408449725804-92255.post@n7.nabble.com>
 <1408994005915-92386.post@n7.nabble.com>
 <alpine.DEB.2.10.1408251259100.2503@scott.cbbs.org>
Message-ID: <1409052244.1170.17.camel@rocketmail.com>

The good news, I had the time to test S/PDIF out today with one of my
Envy24 cards, the bad news, I wasn't able to get S/PDIF out working.

With a TerraTec EWX24/96 I tested
Envy24 S/PDIF optical out -> Sony DTC-670 S/PDIF optical in
With and without
Sony DTC-670 S/PDIF optical out -> Envy24 S/PDIF optical in


mhwe outL -> jkmeter in-1, playback_1, 3, 5, 7, 9
mhwe outR -> jkmeter in-2, playback_2, 4, 6, 8, 10

$ mudita24 -v -c1
$ envy24control -v -c1

Digital in works, analog out works, but I don't know what to do, to get
digital out working.

JFTR I don't know if the DAT recorder's optical S/PDIF in is ok. If it
should be possible with Linux to set my RME HDSPe AIO ADAT IOs to become
S/PDIF IOs (the manual says it's possible when using Windows), I could
test if the Sony in is ok, or simply connect an Envy card to the RME
card.

Mudita version 1.1.0
envy24control version 0.6.0
$ uname -rm
3.16.1-1-ARCH x86_64


From murks at tuxfamily.org  Tue Aug 26 12:09:26 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Tue, 26 Aug 2014 14:09:26 +0200
Subject: [LAU] Sequencer suggestions?
Message-ID: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>

Hi there,

I'm playing with all the new toys you brought me and have quite some
fun. Just two or three years ago I knew everything out there, now
there are a whole lot of new things to play with.

I have a few questions though.
Is there any new midi sequencer? I know the traditional ones, seq24,
rosegarden, muse, qtractor and so on and briefly tried ardour3 as well
as luppp but could not make heads or tails of it yet. What do you use?

Any idea what happened to the calf stuff? I tried them from git but
the, as far as I remember, wonderful wavetable synth crashes the hosts.
There seems to be a fork of the calf instruments and plugins by falktx
but the wavetable synth and the GUIs seem to be missing.

My goal at the moment is to create a single, simple and probably quite
horrible song electronic music song. That's why I'm looking for a nice
sequencer and some instruments.

Regards,
Philipp

From ralf.mardorf at rocketmail.com  Tue Aug 26 12:40:18 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 14:40:18 +0200
Subject: [LAU] [Bulk]  Sequencer suggestions?
In-Reply-To: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
Message-ID: <1409056818.7093.3.camel@rocketmail.com>

On Tue, 2014-08-26 at 14:09 +0200, Philipp ?berbacher wrote:
> What do you use?

I'm using Qtractor, but I'm dissatisfied.

> Any idea what happened to the calf stuff? I tried them from git but
> the, as far as I remember, wonderful wavetable synth crashes the hosts.

Regarding to this issues I replaced calf monosynth with synthv1. Some
time ago, perhaps one or two years, Rui released some new instruments,
synthv1 is the first I'm testing at the moment, but it seems to be
hugely satisfying. I didn't compare it with real synth, but at least for
a virtual synth it sounds nice.


From emailgrant at gmail.com  Tue Aug 26 12:45:20 2014
From: emailgrant at gmail.com (Grant)
Date: Tue, 26 Aug 2014 05:45:20 -0700
Subject: [LAU] ALSA: always use samplerate_best
In-Reply-To: <CAN0CFw2w99DRVExGtWHD96Uwyu-2aFiVyCAc97zbyKU-w_m5zA@mail.gmail.com>
References: <CAN0CFw2XLSLGA6ZLQrpkX4J7Vu99=nbcmsr2AuCtYmVU3NhkzQ@mail.gmail.com>
 <alpine.DEB.2.10.1408240724321.2503@scott.cbbs.org>
 <CAN0CFw2w99DRVExGtWHD96Uwyu-2aFiVyCAc97zbyKU-w_m5zA@mail.gmail.com>
Message-ID: <CAN0CFw3O5OkbNV+dLXerin4=o2qF9_ESKXu0f1ENSt_r4rRAsg@mail.gmail.com>

>>> I have a USB DAC that can only handle 16/44.1 as input and output. I
>>> think ALSA will resample everything to 16/44.1 automatically, but I'd
>>
>> I think what I am saying is that for most cases the sample rate of your
>> audio IF doesn't matter. So adding resampling to everything doesn't make
>> sense... maybe try without first.
>
> I think you're saying that the ALSA resampler won't be used if the
> upstream application does the resampling itself.  Is that correct?
> How can I find out if ALSA is the one resampling in a particular
> scenario?


Is there any way to find out if this is happening?  Do some
applications really do this?

- Grant

From ralf.mardorf at rocketmail.com  Tue Aug 26 12:48:07 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 14:48:07 +0200
Subject: [LAU] [Bulk] Re:  Delta1010: S/PDIF out has sync but no audio
In-Reply-To: <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
References: <1408361569528-92201.post@n7.nabble.com>
 <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
Message-ID: <1409057287.7093.6.camel@rocketmail.com>

> I see that my digital amp gets a syncsignal

My DAT recorder claims 44.1 KHz, when jackd is set to 44.1 KHz, but also
when set to 48 KHz and what ever I select by mudita24 or envy24control,
but I tested it using another Envy24 sound card, I don't have a
Delta1010.


From murks at tuxfamily.org  Tue Aug 26 13:20:22 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Tue, 26 Aug 2014 15:20:22 +0200
Subject: [LAU] [Bulk]  Sequencer suggestions?
In-Reply-To: <1409056818.7093.3.camel@rocketmail.com>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
 <1409056818.7093.3.camel@rocketmail.com>
Message-ID: <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>

On Tue, 26 Aug 2014 14:40:18 +0200
Ralf Mardorf <ralf.mardorf at rocketmail.com> wrote:

> On Tue, 2014-08-26 at 14:09 +0200, Philipp ?berbacher wrote:
> > What do you use?
> 
> I'm using Qtractor, but I'm dissatisfied.
> 
> > Any idea what happened to the calf stuff? I tried them from git but
> > the, as far as I remember, wonderful wavetable synth crashes the
> > hosts.
> 
> Regarding to this issues I replaced calf monosynth with synthv1. Some
> time ago, perhaps one or two years, Rui released some new instruments,
> synthv1 is the first I'm testing at the moment, but it seems to be
> hugely satisfying. I didn't compare it with real synth, but at least
> for a virtual synth it sounds nice.

Thaks Ralf,
the monosynth is nice in its own right and it still works. I think
I liked the wavetable synth better.
I briefly tried fabla which is also a wavetable synth but couldn't
quite get sounds that I liked out of it, some of its presets do better
though, so I probably just haven't figured out how to do it yet. I also
used to like phasex, but since the recent thread about it I'm a bit
scared to use it.

I guess I'll give non-sequencer another go, it looks a bit different
from what I remember. First I'll have to make myself familiar with NSM
though, back and the day there was only lash and ladish.

I'll try to not get caught up in too many technical details this time
around and make some sound instead.

Regards,
Philipp

From ralf.mardorf at rocketmail.com  Tue Aug 26 13:40:54 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 15:40:54 +0200
Subject: [LAU] [Bulk] Re:  [Bulk]  Sequencer suggestions?
In-Reply-To: <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
 <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
Message-ID: <1409060454.7093.10.camel@rocketmail.com>

On Tue, 2014-08-26 at 15:20 +0200, Philipp ?berbacher wrote:
> I also used to like phasex, but since the recent thread about it I'm a
> bit scared to use it.

I'm also scared. I claimed that I didn't notice this issue, but later I
remembered there were issues. I don't know if it were dc offsets.

"2.23  DC Offset Remover (dcRemove, 1207)

Simply removes the DC (0 Hz) component from an audio signal, uses a high
pass filter, so has some side effects, but they should be minimal." -
http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html#tth_sEc2.23

Would this protect against phasex loudspeaker-heater feature? 




From ralf.mardorf at rocketmail.com  Tue Aug 26 13:46:59 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 15:46:59 +0200
Subject: [LAU] [Bulk] Re:  [Bulk]  Sequencer suggestions?
In-Reply-To: <1409060454.7093.10.camel@rocketmail.com>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
 <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
Message-ID: <1409060819.7093.12.camel@rocketmail.com>

On Tue, 2014-08-26 at 15:40 +0200, Ralf Mardorf wrote:
> On Tue, 2014-08-26 at 15:20 +0200, Philipp ?berbacher wrote:
> > I also used to like phasex, but since the recent thread about it I'm a
> > bit scared to use it.
> 
> I'm also scared. I claimed that I didn't notice this issue, but later I
> remembered there were issues. I don't know if it were dc offsets.
> 
> "2.23  DC Offset Remover (dcRemove, 1207)
> 
> Simply removes the DC (0 Hz) component from an audio signal, uses a high
> pass filter, so has some side effects, but they should be minimal." -
> http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html#tth_sEc2.23
> 
> Would this protect against phasex loudspeaker-heater feature? 

It's said it's not audible, IIRC I sometimes heard a loud, very loud
single crackle. Could this be caused by a dc offset? 



From fons at linuxaudio.org  Tue Aug 26 14:08:20 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Tue, 26 Aug 2014 14:08:20 +0000
Subject: [LAU] [Bulk] Re:  [Bulk]  Sequencer suggestions?
In-Reply-To: <1409060819.7093.12.camel@rocketmail.com>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
 <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
Message-ID: <20140826140820.GA16859@linuxaudio.org>

On Tue, Aug 26, 2014 at 03:46:59PM +0200, Ralf Mardorf wrote:

> > "2.23  DC Offset Remover (dcRemove, 1207)
> > 
> > Simply removes the DC (0 Hz) component from an audio signal, uses a high
> > pass filter, so has some side effects, but they should be minimal." -
> > http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html#tth_sEc2.23
> > 
> > Would this protect against phasex loudspeaker-heater feature? 
> 
> It's said it's not audible, IIRC I sometimes heard a loud, very loud
> single crackle. Could this be caused by a dc offset? 

Yes. The problem here is that the DC offset originates in the
oscillators and is multiplied by the envelope. Hence the result
is not really constant. A highpass filter will remove the constant
or slowly changing part during a note, but not the 'thumps' at the
beginning and end. Depending on the cutoff frequency and the rise
and fall times of the envelope these could still be much larger than
the signal. They won't probably fry you speaker but could result in
clipping and hence crackles. 

The only real solution is to remove the DC offset at the output
of the oscillators, before it enters the rest of the audio chain.

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From dlphillips at woh.rr.com  Tue Aug 26 14:13:04 2014
From: dlphillips at woh.rr.com (Dave Phillips)
Date: Tue, 26 Aug 2014 10:13:04 -0400
Subject: [LAU] Sequencer suggestions?
In-Reply-To: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
Message-ID: <53FC95F0.4080204@woh.rr.com>


On 08/26/2014 08:09 AM, Philipp ?berbacher wrote:
> Hi there,
>
> ...
> Is there any new midi sequencer?

If you're not averse to spending money, check out Bitwig. I like its 
MIDI capabilities.

Best,

dp


From ralf.mardorf at rocketmail.com  Tue Aug 26 14:40:38 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 16:40:38 +0200
Subject: [LAU] Sequencer suggestions?
In-Reply-To: <20140826140820.GA16859@linuxaudio.org>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
 <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
Message-ID: <1409064038.7093.22.camel@rocketmail.com>

On Tue, 2014-08-26 at 14:08 +0000, Fons Adriaensen wrote:
> On Tue, Aug 26, 2014 at 03:46:59PM +0200, Ralf Mardorf wrote:
> 
> > > "2.23  DC Offset Remover (dcRemove, 1207)
> > > 
> > > Simply removes the DC (0 Hz) component from an audio signal, uses a high
> > > pass filter, so has some side effects, but they should be minimal." -
> > > http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html#tth_sEc2.23
> > > 
> > > Would this protect against phasex loudspeaker-heater feature? 
> > 
> > It's said it's not audible, IIRC I sometimes heard a loud, very loud
> > single crackle. Could this be caused by a dc offset? 
> 
> Yes. The problem here is that the DC offset originates in the
> oscillators and is multiplied by the envelope. Hence the result
> is not really constant. A highpass filter will remove the constant
> or slowly changing part during a note, but not the 'thumps' at the
> beginning and end. Depending on the cutoff frequency and the rise
> and fall times of the envelope these could still be much larger than
> the signal. They won't probably fry you speaker but could result in
> clipping and hence crackles. 
> 
> The only real solution is to remove the DC offset at the output
> of the oscillators, before it enters the rest of the audio chain.

Thank you Fons,

I reported the issue.

https://github.com/williamweston/phasex/issues/10#issue-41179099

I've got no doubts anymore, that I experienced the DC offsets too.

Regards,
Ralf


From fons at linuxaudio.org  Tue Aug 26 15:13:37 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Tue, 26 Aug 2014 15:13:37 +0000
Subject: [LAU] Sequencer suggestions?
In-Reply-To: <1409064038.7093.22.camel@rocketmail.com>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
 <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
Message-ID: <20140826151337.GB16859@linuxaudio.org>

On Tue, Aug 26, 2014 at 04:40:38PM +0200, Ralf Mardorf wrote:

> > The only real solution is to remove the DC offset at the output
> > of the oscillators, before it enters the rest of the audio chain.
> 
> Thank you Fons,
> 
> I reported the issue.
> 
> https://github.com/williamweston/phasex/issues/10#issue-41179099
> 
> I've got no doubts anymore, that I experienced the DC offsets too.

The strange thing is that there is a DC blocker in the code,
but it's not enabled (at least not in the AUR package), and
it's in the wrong place. It sort of works but it far from ideal.
I'm currently patching the code to have a DC blocker in each
voice and right after the oscillators are mixed.

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From murks at tuxfamily.org  Tue Aug 26 15:20:24 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Tue, 26 Aug 2014 17:20:24 +0200
Subject: [LAU] Sequencer suggestions?
In-Reply-To: <20140826151337.GB16859@linuxaudio.org>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
 <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
Message-ID: <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>

On Tue, 26 Aug 2014 15:13:37 +0000
Fons Adriaensen <fons at linuxaudio.org> wrote:

> On Tue, Aug 26, 2014 at 04:40:38PM +0200, Ralf Mardorf wrote:
> 
> > > The only real solution is to remove the DC offset at the output
> > > of the oscillators, before it enters the rest of the audio chain.
> > 
> > Thank you Fons,
> > 
> > I reported the issue.
> > 
> > https://github.com/williamweston/phasex/issues/10#issue-41179099
> > 
> > I've got no doubts anymore, that I experienced the DC offsets too.
> 
> The strange thing is that there is a DC blocker in the code,
> but it's not enabled (at least not in the AUR package), and
> it's in the wrong place. It sort of works but it far from ideal.
> I'm currently patching the code to have a DC blocker in each
> voice and right after the oscillators are mixed.

Thanks Fons, much appreciated. It seems the original author didn't
touch the code in more than 1 1/2 years, so the bug report would likely
had no effect.

Regards,
Philipp

From ralf.mardorf at rocketmail.com  Tue Aug 26 15:27:19 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 17:27:19 +0200
Subject: [LAU] [Bulk] Re:  Sequencer suggestions?
In-Reply-To: <20140826151337.GB16859@linuxaudio.org>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
 <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
Message-ID: <1409066839.7093.25.camel@rocketmail.com>

On Tue, 2014-08-26 at 15:13 +0000, Fons Adriaensen wrote:
> On Tue, Aug 26, 2014 at 04:40:38PM +0200, Ralf Mardorf wrote:
> 
> > > The only real solution is to remove the DC offset at the output
> > > of the oscillators, before it enters the rest of the audio chain.
> > 
> > Thank you Fons,
> > 
> > I reported the issue.
> > 
> > https://github.com/williamweston/phasex/issues/10#issue-41179099
> > 
> > I've got no doubts anymore, that I experienced the DC offsets too.
> 
> The strange thing is that there is a DC blocker in the code,
> but it's not enabled (at least not in the AUR package), and
> it's in the wrong place. It sort of works but it far from ideal.
> I'm currently patching the code to have a DC blocker in each
> voice and right after the oscillators are mixed.

Sometimes I dislike you, but often you provide very useful things :). My
apologies for my weaknesses and thank your for your effort :).

Fixing phasex is very cool! IMO some sounds can compare with CEM
(Curtis) based analog synth.


From looplog at gmail.com  Tue Aug 26 15:27:56 2014
From: looplog at gmail.com (michael noble)
Date: Wed, 27 Aug 2014 00:27:56 +0900
Subject: [LAU] Sequencer suggestions?
In-Reply-To: <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
 <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
 <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
Message-ID: <CABf1g_yUsqBWF=soUeCrQ79P1RzLTB3tNLWC8raT=Jn2jSGPjQ@mail.gmail.com>

On Wed, Aug 27, 2014 at 12:20 AM, Philipp ?berbacher <murks at tuxfamily.org>
wrote:

> It seems the original author didn't
> touch the code in more than 1 1/2 years, so the bug report would likely
> had no effect.
>

The original author seems to have been missing in action for quite some
time, though it's not the first time. He disappeared for a long time before
the last exciting, though flawed update, which led to a github fork at the
time. When he re-appeared to much fanfare he communicated great intent to
maintain Phasex, and then promptly disappeared again. Attempts by multiple
people to contact the author through numerous channels have been
unsuccessful so far as I know.

It would be a shame for Phasex to go by the wayside, as it really is a very
cool sounding synth, problems aside.
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From ralf.mardorf at rocketmail.com  Tue Aug 26 16:02:01 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 18:02:01 +0200
Subject: [LAU] [Bulk] Re:  Sequencer suggestions?
In-Reply-To: <CABf1g_yUsqBWF=soUeCrQ79P1RzLTB3tNLWC8raT=Jn2jSGPjQ@mail.gmail.com>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
 <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
 <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
 <CABf1g_yUsqBWF=soUeCrQ79P1RzLTB3tNLWC8raT=Jn2jSGPjQ@mail.gmail.com>
Message-ID: <1409068921.7093.31.camel@rocketmail.com>

On Wed, 2014-08-27 at 00:27 +0900, michael noble wrote:
> 
> On Wed, Aug 27, 2014 at 12:20 AM, Philipp ?berbacher
> <murks at tuxfamily.org> wrote:
>         It seems the original author didn't
>         touch the code in more than 1 1/2 years, so the bug report
>         would likely
>         had no effect.
>         
> 
> The original author seems to have been missing in action for quite
> some time, though it's not the first time. He disappeared for a long
> time before the last exciting, though flawed update, which led to a
> github fork at the time. When he re-appeared to much fanfare he
> communicated great intent to maintain Phasex, and then promptly
> disappeared again. Attempts by multiple people to contact the author
> through numerous channels have been unsuccessful so far as I know. 
> 
> 
> It would be a shame for Phasex to go by the wayside, as it really is a
> very cool sounding synth, problems aside.

Actually it's the only virtual synth that IMO relatively often can
compare with some CEM (Curtis) based analog synth-sounds. The "Moog's
wandering pitch tracking" sounds sometimes exceed even CEM (Curtis)
synth, however, even some Moog synth used CEM (Curtis). Dave Smith
likely is the Steve Wozniak of vintage analog synth maintaining. It's a
pity that he nowadays has got the monopoly for good synth chips :(. JFTR
I don't know how to repair my digital age TG33 either ;) ... chips die
out.



From fons at linuxaudio.org  Tue Aug 26 16:06:30 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Tue, 26 Aug 2014 16:06:30 +0000
Subject: [LAU] Sequencer suggestions?
In-Reply-To: <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
 <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
 <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
Message-ID: <20140826160630.GC16859@linuxaudio.org>

On Tue, Aug 26, 2014 at 05:20:24PM +0200, Philipp ?berbacher wrote:

> Thanks Fons, much appreciated. It seems the original author didn't
> touch the code in more than 1 1/2 years, so the bug report would likely
> had no effect.


The following patches seem to fix the problem. They remove the 
(inoperative) per patch DC blocker and add one per voice, just
after the oscillators. More testing may be required.


--- engine.h.orig   2013-01-13 08:18:49.000000000 +0100
+++ engine.h    2014-08-26 17:18:13.514909324 +0200
@@ -84,9 +84,7 @@
    sample_t    bps;                        /* beats per second */
    sample_t    out1;                       /* output sample 2 */
    sample_t    out2;                       /* output sample 1 */
-#ifdef ENABLE_DC_REJECTION_FILTER
-   sample_t    dcR_const;
-#endif
+   sample_t    wdcf;
 } GLOBAL;
 
 
@@ -105,6 +103,8 @@
    int         portamento_sample;          /* sample number within portamento */
    int         portamento_samples;         /* portamento time in samples */
    int         age;                        /* voice age, in samples */
+        sample_t    dcf1;                       /* DC filter state 1 */
+        sample_t    dcf2;                       /* DC filter state 2 */
    sample_t    out1;                       /* output sample 1 */
    sample_t    out2;                       /* output sample 2 */
    sample_t    amp_env;                    /* smoothed final output of env generator */





--- engine.c.orig   2013-01-13 08:18:49.000000000 +0100
+++ engine.c    2014-08-26 17:54:29.301481175 +0200
@@ -193,9 +193,7 @@
 
    /* (-3dB @ 20Hz) DC blocking filter */
    /* 1.0 - (M_PI * 2 * freq / (sample_t) f_sample_rate) */
-#ifdef ENABLE_DC_REJECTION_FILTER
-   global.dcR_const = 1.0 - (125.6 / (sample_t) f_sample_rate);
-#endif
+   global.wdcf = 125.7 / (sample_t) f_sample_rate;
 }
 
 
@@ -487,6 +485,10 @@
            voice->midi_key   = -1;
            voice->keypressed = -1;
 
+           /* init dc filters */
+           voice->dcf1 = 0.0;
+           voice->dcf2 = 0.0;
+
            /* initialize moog filters */
            voice->filter_y1_1    = 0.0;
            voice->filter_y1_2    = 0.0;
@@ -780,10 +782,6 @@
 run_part(PART *part, PATCH_STATE *state, unsigned int part_num)
 {
    unsigned int    osc;
-#ifdef ENABLE_DC_REJECTION_FILTER
-   sample_t        tmp1;
-   sample_t        tmp2;
-#endif
 
    /* generate amplitude envelopes for all voices */
    run_voice_envelopes(part, state, part_num);
@@ -842,19 +840,6 @@
    if (state->delay_mix_cc) {
        run_delay(get_delay(part_num), part, state);
    }
-
-   /* output this sample to the buffer */
-#ifdef ENABLE_DC_REJECTION_FILTER
-   tmp1 = part->out1;
-   part->out1 = part->out1 - part->dcR_in1 + global.dcR_const * part->dcR_out1;
-   part->dcR_in1  = tmp1;
-   part->dcR_out1 = part->out1;
-
-   tmp2 = part->out2;
-   part->out2 = part->out2 - part->dcR_in2 + global.dcR_const * part->dcR_out2;
-   part->dcR_in2  = tmp2;
-   part->dcR_out2 = part->out2;
-#endif
 }
 
 
@@ -1300,6 +1285,12 @@
        run_osc(voice, part, state, osc);
    }
 
+   /* apply DC filters */
+   voice->dcf1 += global.wdcf * (voice->out1 - voice->dcf1);
+   voice->out1 -= voice->dcf1;
+   voice->dcf2 += global.wdcf * (voice->out2 - voice->dcf2);
+   voice->out2 -= voice->dcf2;
+
    /* oscs are mixed.  now apply AM oscs. */
    for (osc = 0; osc < NUM_OSCS; osc++) {
        if (state->osc_modulation[osc] == MOD_TYPE_AM) {



-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From ralf.mardorf at rocketmail.com  Tue Aug 26 16:16:31 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 18:16:31 +0200
Subject: [LAU] [Bulk] Re:  Sequencer suggestions?
In-Reply-To: <20140826160630.GC16859@linuxaudio.org>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
 <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
 <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
 <20140826160630.GC16859@linuxaudio.org>
Message-ID: <1409069791.7093.33.camel@rocketmail.com>

On Tue, 2014-08-26 at 16:06 +0000, Fons Adriaensen wrote:
> The following patches seem to fix the problem.

Thank you very much Fons :),

just to be on the safe side, please don't add the patch inline the
email, send an attachment. IIRC you know a pad sound that caused an DC
offset, so please send the settings that caused a DC offset, it might be
a good base to start editing sounds to test your patch.

Regards,
Ralf

PS: Feel free to provide an Arch PKGBUILD :p.


From fons at linuxaudio.org  Tue Aug 26 16:23:53 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Tue, 26 Aug 2014 16:23:53 +0000
Subject: [LAU] [Bulk] Re:  Sequencer suggestions?
In-Reply-To: <1409069791.7093.33.camel@rocketmail.com>
References: <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
 <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
 <20140826160630.GC16859@linuxaudio.org>
 <1409069791.7093.33.camel@rocketmail.com>
Message-ID: <20140826162353.GD16859@linuxaudio.org>

On Tue, Aug 26, 2014 at 06:16:31PM +0200, Ralf Mardorf wrote:
> On Tue, 2014-08-26 at 16:06 +0000, Fons Adriaensen wrote:
> > The following patches seem to fix the problem.
> 
> Thank you very much Fons :),
> 
> just to be on the safe side, please don't add the patch inline the
> email, send an attachment. IIRC you know a pad sound that caused an DC
> offset, so please send the settings that caused a DC offset, it might be
> a good base to start editing sounds to test your patch.

Not sure if attachments make it to the list. 

The patch that gave the horrible DC offset in my test a few
weeks ago was 'mellow-pad' from sys-patches, I used the same
to test today.

I'm not the maintainer of the AUR package :-) Whoever that is
may be lurking here and pick up the fix.

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From ralf.mardorf at rocketmail.com  Tue Aug 26 16:36:30 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 18:36:30 +0200
Subject: [LAU] [Bulk] Re:  [Bulk] Re:  Sequencer suggestions?
In-Reply-To: <20140826162353.GD16859@linuxaudio.org>
References: <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
 <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
 <20140826160630.GC16859@linuxaudio.org>
 <1409069791.7093.33.camel@rocketmail.com>
 <20140826162353.GD16859@linuxaudio.org>
Message-ID: <1409070990.7093.37.camel@rocketmail.com>



On Tue, 2014-08-26 at 16:23 +0000, Fons Adriaensen wrote:
> On Tue, Aug 26, 2014 at 06:16:31PM +0200, Ralf Mardorf wrote:
> > On Tue, 2014-08-26 at 16:06 +0000, Fons Adriaensen wrote:
> > > The following patches seem to fix the problem.
> > 
> > Thank you very much Fons :),
> > 
> > just to be on the safe side, please don't add the patch inline the
> > email, send an attachment. IIRC you know a pad sound that caused an DC
> > offset, so please send the settings that caused a DC offset, it might be
> > a good base to start editing sounds to test your patch.
> 
> Not sure if attachments make it to the list.

No :(. It doesn't matter, I will copy and paste and if needed correct
them tomorrow or the day after tomorrow.

> The patch that gave the horrible DC offset in my test a few
> weeks ago was 'mellow-pad' from sys-patches, I used the same
> to test today.

Ok.

> I'm not the maintainer of the AUR package :-) Whoever that is
> may be lurking here and pick up the fix.

:D

[rocketmouse at archlinux ~]$ pacman -Qi phasex-git
Name           : phasex-git
Version        : 20130331-1
Description    : An experimental MIDI softsynth with flexible phase modulation and oscillator/LFO sources
Architecture   : x86_64
URL            : https://github.com/williamweston/phasex
Packager       : Unknown Packager
Build Date     : Sun 07 Apr 2013 01:29:56 PM CEST
Install Date   : Sun 07 Apr 2013 01:30:50 PM CEST
Install Reason : Explicitly installed
Install Script : Yes
Validated By   : None

Ouch!

https://aur.archlinux.org/packages/phasex-git/ :

Submitter:	rtfreedman
Maintainer:	rtfreedman

FirstSubmitted:	2013-03-31 20:22
Last Updated:	2013-04-10 15:33

;)

I'll forwarded this mail to archaudio-discuss at archaudio.org , but I feel
responsible to do the work on my own.




From murks at tuxfamily.org  Tue Aug 26 16:46:37 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Tue, 26 Aug 2014 18:46:37 +0200
Subject: [LAU] [Bulk] Re:  [Bulk] Re:  Sequencer suggestions?
In-Reply-To: <1409070990.7093.37.camel@rocketmail.com>
References: <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
 <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
 <20140826160630.GC16859@linuxaudio.org>
 <1409069791.7093.33.camel@rocketmail.com>
 <20140826162353.GD16859@linuxaudio.org>
 <1409070990.7093.37.camel@rocketmail.com>
Message-ID: <20140826184637.22ffebca@eeyore.mozart.uni-klu.ac.at>

On Tue, 26 Aug 2014 18:36:30 +0200
Ralf Mardorf <ralf.mardorf at rocketmail.com> wrote:

> 
> 
> On Tue, 2014-08-26 at 16:23 +0000, Fons Adriaensen wrote:
> > On Tue, Aug 26, 2014 at 06:16:31PM +0200, Ralf Mardorf wrote:
> > > On Tue, 2014-08-26 at 16:06 +0000, Fons Adriaensen wrote:
> > > > The following patches seem to fix the problem.
> > > 
> > > Thank you very much Fons :),
> > > 
> > > just to be on the safe side, please don't add the patch inline the
> > > email, send an attachment. IIRC you know a pad sound that caused
> > > an DC offset, so please send the settings that caused a DC
> > > offset, it might be a good base to start editing sounds to test
> > > your patch.
> > 
> > Not sure if attachments make it to the list.
> 
> No :(. It doesn't matter, I will copy and paste and if needed correct
> them tomorrow or the day after tomorrow.
> 
> > The patch that gave the horrible DC offset in my test a few
> > weeks ago was 'mellow-pad' from sys-patches, I used the same
> > to test today.
> 
> Ok.
> 
> > I'm not the maintainer of the AUR package :-) Whoever that is
> > may be lurking here and pick up the fix.
> 
> :D
> 
> [rocketmouse at archlinux ~]$ pacman -Qi phasex-git
> Name           : phasex-git
> Version        : 20130331-1
> Description    : An experimental MIDI softsynth with flexible phase
> modulation and oscillator/LFO sources Architecture   : x86_64
> URL            : https://github.com/williamweston/phasex
> Packager       : Unknown Packager
> Build Date     : Sun 07 Apr 2013 01:29:56 PM CEST
> Install Date   : Sun 07 Apr 2013 01:30:50 PM CEST
> Install Reason : Explicitly installed
> Install Script : Yes
> Validated By   : None
> 
> Ouch!
> 
> https://aur.archlinux.org/packages/phasex-git/ :
> 
> Submitter:	rtfreedman
> Maintainer:	rtfreedman
> 
> FirstSubmitted:	2013-03-31 20:22
> Last Updated:	2013-04-10 15:33
> 
> ;)
> 
> I'll forwarded this mail to archaudio-discuss at archaudio.org , but I
> feel responsible to do the work on my own.

And with the non-git package you can basically forget to get the patch
in, it is 'maintained' by speps who basically just sits on tons of
audio packages without doing anything. This is one 'trusted user' I
do not trust to close his own zippers. Better just create another
package, it will be faster by years.

I just tried to patch the git package but something goes very wrong
here. I get no error but it looks like the patch isn't applied either
and I can't get rid of the sed error either. I haven't done this sort of
thing in years.

Regards,
Philipp

From willgodfrey at musically.me.uk  Tue Aug 26 16:48:12 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Tue, 26 Aug 2014 17:48:12 +0100
Subject: [LAU] Sequencer suggestions?
In-Reply-To: <53FC95F0.4080204@woh.rr.com>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
 <53FC95F0.4080204@woh.rr.com>
Message-ID: <20140826174812.6a7c1a12@debian>

On Tue, 26 Aug 2014 10:13:04 -0400
Dave Phillips <dlphillips at woh.rr.com> wrote:

> 
> On 08/26/2014 08:09 AM, Philipp ?berbacher wrote:
> > Hi there,
> >
> > ...
> > Is there any new midi sequencer?
> 
> If you're not averse to spending money, check out Bitwig. I like its 
> MIDI capabilities.
> 
> Best,
> 
> dp

Still using Rosegarden. It has it's problems (like all things) but it's proved
to be pretty reliable for all these years.

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From ralf.mardorf at rocketmail.com  Tue Aug 26 16:59:04 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 18:59:04 +0200
Subject: [LAU] Sequencer suggestions?
In-Reply-To: <20140826184637.22ffebca@eeyore.mozart.uni-klu.ac.at>
References: <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
 <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
 <20140826160630.GC16859@linuxaudio.org>
 <1409069791.7093.33.camel@rocketmail.com>
 <20140826162353.GD16859@linuxaudio.org>
 <1409070990.7093.37.camel@rocketmail.com>
 <20140826184637.22ffebca@eeyore.mozart.uni-klu.ac.at>
Message-ID: <1409072344.13214.3.camel@rocketmail.com>

On Tue, 2014-08-26 at 18:46 +0200, Philipp ?berbacher wrote:
> I just tried to patch the git package but something goes very wrong
> here. I get no error but it looks like the patch isn't applied either
> and I can't get rid of the sed error either. I haven't done this sort of
> thing in years.

Perhaps a typo? You should post the output. For what version are Fon's
patches and what version did you use? I'm to lazy to test it today, I'll
continue within the next two days.

Philipp are you on Arch Linux too?

IIRC Fons' compiles much software, instead of using PKGBUILDS. I use
much official and many AUR PKGBUILDS, very seldom I build software on my
own.

There's some latitude. 	We might have different source codes installed. 


From ralf.mardorf at rocketmail.com  Tue Aug 26 17:08:52 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Tue, 26 Aug 2014 19:08:52 +0200
Subject: [LAU] Sequencer suggestions?
In-Reply-To: <1409072344.13214.3.camel@rocketmail.com>
References: <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
 <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
 <20140826160630.GC16859@linuxaudio.org>
 <1409069791.7093.33.camel@rocketmail.com>
 <20140826162353.GD16859@linuxaudio.org>
 <1409070990.7093.37.camel@rocketmail.com>
 <20140826184637.22ffebca@eeyore.mozart.uni-klu.ac.at>
 <1409072344.13214.3.camel@rocketmail.com>
Message-ID: <1409072932.13214.5.camel@rocketmail.com>


> I just tried to patch the git package

So you used the AUR package?! Remove it and checkout from git without
using a PKGBUILD ... or wait until tomorrow, I will test it too.



From nabob_cd at yahoo.com  Tue Aug 26 17:30:50 2014
From: nabob_cd at yahoo.com (Menno)
Date: Tue, 26 Aug 2014 10:30:50 -0700 (PDT)
Subject: [LAU] Delta1010: S/PDIF out has sync but no audio
In-Reply-To: <1409057287.7093.6.camel@rocketmail.com>
References: <1408361569528-92201.post@n7.nabble.com>
 <alpine.DEB.2.10.1408180730360.2503@scott.cbbs.org>
 <1409057287.7093.6.camel@rocketmail.com>
Message-ID: <1409074250944-92425.post@n7.nabble.com>

Lev, Ralf, thanks for your efforts to try to help here, much appreciated!

Nice idea to make a connection with jack from a hardware input to the S/PDIF
directly with a rythm and test it. I've set the mixer to the corresponding
hardware input.
I do see the meters going up and down and i do have the sync signal on my
amp. But no sound.
Even when the hardware playback is NOT connected, i have the analog sound on
the digital mixer output.
Can this indicate this an internal routing bug?
And:
Because the S/PDIF output works in Windows XP, i suspect that there is a
software bug somewhere, perhaps alsa??




--
View this message in context: http://linux-audio.4202.n7.nabble.com/Delta1010-S-PDIF-out-has-sync-but-no-audio-tp92201p92425.html
Sent from the linux-audio-user mailing list archive at Nabble.com.

From fons at linuxaudio.org  Tue Aug 26 19:12:04 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Tue, 26 Aug 2014 19:12:04 +0000
Subject: [LAU] [Bulk] Re:  [Bulk] Re:  Sequencer suggestions?
In-Reply-To: <20140826184637.22ffebca@eeyore.mozart.uni-klu.ac.at>
References: <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
 <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
 <20140826160630.GC16859@linuxaudio.org>
 <1409069791.7093.33.camel@rocketmail.com>
 <20140826162353.GD16859@linuxaudio.org>
 <1409070990.7093.37.camel@rocketmail.com>
 <20140826184637.22ffebca@eeyore.mozart.uni-klu.ac.at>
Message-ID: <20140826191204.GA11173@linuxaudio.org>

On Tue, Aug 26, 2014 at 06:46:37PM +0200, Philipp ?berbacher wrote:
 
> And with the non-git package you can basically forget to get the patch
> in, it is 'maintained' by speps who basically just sits on tons of
> audio packages without doing anything. This is one 'trusted user' I
> do not trust to close his own zippers. Better just create another
> package, it will be faster by years.
> 
> I just tried to patch the git package but something goes very wrong
> here. I get no error but it looks like the patch isn't applied either
> and I can't get rid of the sed error either. I haven't done this sort of
> thing in years.

The patch is against the AUR phasex package 0.14.97 maintained 
by William Weston.  

If you apply the patch to sources obtained via AUR you need

 makepkg -e -f

without the -e the original sources will be extracted and
overwrite the patch.

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From willgodfrey at musically.me.uk  Tue Aug 26 20:24:39 2014
From: willgodfrey at musically.me.uk (Will Godfrey)
Date: Tue, 26 Aug 2014 21:24:39 +0100
Subject: [LAU] Noo 'pooter :)
Message-ID: <20140826212439.352361f6@debian>

In the next couple of days I should be getting a new fanless dual core machine
with a 64bit intel cpu running at 3.1G. This is really intended to replace my
ageing 'office' machine, but I thought I might as well set it up for decent
audio too.

As this will be a clean install, I'm wondering what people might suggest as
for best distro to make full use of it - all my other machines have had a
progression of debian upgrades so are probably full of crud.

-- 
Will J Godfrey
http://www.musically.me.uk
Say you have a poem and I have a tune.
Exchange them and we can both have a poem, a tune, and a song.

From randallw at ornl.gov  Tue Aug 26 20:29:09 2014
From: randallw at ornl.gov (Randall Wetherington)
Date: Tue, 26 Aug 2014 13:29:09 -0700 (PDT)
Subject: [LAU] Roland Octa-Capture
In-Reply-To: <F2B7D5C6-C5D2-4E2B-97F0-5E06696D10EC@frii.com>
References: <0C7991E8-C23C-4D83-AF59-0A3527E41C62@frii.com>
 <5195DD19.40509@ladisch.de> <519733F0.4010502@ladisch.de>
 <F2B7D5C6-C5D2-4E2B-97F0-5E06696D10EC@frii.com>
Message-ID: <1409084949335-92433.post@n7.nabble.com>

Hi Kevin,

I am wondering if there has been any status change on this. I have a
octa-capture box and would like to get it running with Angstrom Linux. How
did you do with your Linux challenge? 



--
View this message in context: http://linux-audio.4202.n7.nabble.com/Roland-Octa-Capture-tp85177p92433.html
Sent from the linux-audio-user mailing list archive at Nabble.com.

From david.jo.adler at gmail.com  Tue Aug 26 21:26:04 2014
From: david.jo.adler at gmail.com (David Adler)
Date: Tue, 26 Aug 2014 23:26:04 +0200
Subject: [LAU]  Noo 'pooter :)
In-Reply-To: <20140826212439.352361f6@debian>
References: <20140826212439.352361f6@debian>
Message-ID: <20140826212604.GB587@digit.localdomain>

On Tue, Aug 26, 2014 at 09:24:39PM +0100, Will Godfrey wrote:

> As this will be a clean install, I'm wondering what people might suggest as
> for best distro to make full use of it - all my other machines have had a
> progression of debian upgrades so are probably full of crud.

Use Arch. It might sound counter-intuitive but despite (or because
of(?)) the rolling release model it requires very little maintenance. 

The regular glimpse on the homepage's news feed is recommended but
it's been a long time since anything popped up there that actually
required manual intervention. If this happens, the instructions have
proven to be adequate. Other than that, occasionally configuration files 
suffixed *.pacnew/*.pacsave need to be merged and voil?, you have a
crud-free up-to-date system that won't send you to dependency hell when 
attempting to install recent software.

The above might sound a bit like over-optimistic marketing speak but it
reflects my experience and from what I've heard it's not just me. 

That said, Debian testing didn't exactly give me headaches -- it'd be my
second choice for audio -- but my experiences with Arch (quite a few
years now, no re-installation) are plainly positive.

https://wiki.archlinux.org/index.php/Arch_Linux_system_maintenance


greetz,
 -d


From harryhaaren at gmail.com  Tue Aug 26 22:24:38 2014
From: harryhaaren at gmail.com (Harry van Haaren)
Date: Tue, 26 Aug 2014 23:24:38 +0100
Subject: [LAU] [Bulk] Sequencer suggestions?
In-Reply-To: <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
 <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
Message-ID: <CAKudYbORu9RQxouVotma6MvZojWgxq7GTQTtKu_cEm-zXVz23w@mail.gmail.com>

On Tue, Aug 26, 2014 at 2:20 PM, Philipp ?berbacher <murks at tuxfamily.org> wrote:
> I briefly tried fabla which is also a wavetable synth but couldn't
> quite get sounds that I liked out of it, some of its presets do better
> though, so I probably just haven't figured out how to do it yet.

I presume you mean Sorcer not Fabla, http://www.openavproductions.com/sorcer ?
Sorcer is aimed at "dubstep basslines", and is not a general purpose synth.

Good news is that OpenAV is working on a multi-purpose wavetable synth, however
it is not yet ready for announcement. Development is ongoing and progressing,
but this is a very large and complex instrument, please be patient! For the
curious, there's some info about Ninja here:
http://openavproductions.com/ninja

Cheers, -Harry

From waynedpj at in-giro.org  Tue Aug 26 22:26:18 2014
From: waynedpj at in-giro.org (Wayne DePrince Jr.)
Date: Tue, 26 Aug 2014 18:26:18 -0400
Subject: [LAU] Noo 'pooter :)
In-Reply-To: <20140826212604.GB587@digit.localdomain>
References: <20140826212439.352361f6@debian>
 <20140826212604.GB587@digit.localdomain>
Message-ID: <1409091978.8599.179.camel@localhost.localdomain>

On mar, 2014-08-26 at 23:26 +0200, David Adler wrote: 

> On Tue, Aug 26, 2014 at 09:24:39PM +0100, Will Godfrey wrote:
> 
> > As this will be a clean install, I'm wondering what people might suggest as
> > for best distro to make full use of it - all my other machines have had a
> > progression of debian upgrades so are probably full of crud.
> 
> Use Arch. It might sound counter-intuitive but despite (or because
> of(?)) the rolling release model it requires very little maintenance. 
> 
> The regular glimpse on the homepage's news feed is recommended but
> it's been a long time since anything popped up there that actually
> required manual intervention. If this happens, the instructions have
> proven to be adequate. Other than that, occasionally configuration files 
> suffixed *.pacnew/*.pacsave need to be merged and voil?, you have a
> crud-free up-to-date system that won't send you to dependency hell when 
> attempting to install recent software.
> 
> The above might sound a bit like over-optimistic marketing speak but it
> reflects my experience and from what I've heard it's not just me. 
> 
> That said, Debian testing didn't exactly give me headaches -- it'd be my
> second choice for audio -- but my experiences with Arch (quite a few
> years now, no re-installation) are plainly positive.
> 
> https://wiki.archlinux.org/index.php/Arch_Linux_system_maintenance
> 
> 
> greetz,
>  -d
> 


    in the same vein, and for similar reasons, i can heartily recommend
Gentoo, or better yet Sabayon (no compiling, binary packages).  along
with the excellent Pro Audio Overlay
http://proaudio.tuxfamily.org/wiki/index.php?title=Main_Page i have been
happily making music with Gentoo/Sabayon for 6+ years.  the big win is
the avoidance of major upgrades with the rolling release.

    also, your message on Arch has me thinking i should check it out as
well ;)

peace, w


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From harryhaaren at gmail.com  Tue Aug 26 22:30:31 2014
From: harryhaaren at gmail.com (Harry van Haaren)
Date: Tue, 26 Aug 2014 23:30:31 +0100
Subject: [LAU] Sequencer suggestions?
In-Reply-To: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
References: <20140826140926.10b100f5@eeyore.mozart.uni-klu.ac.at>
Message-ID: <CAKudYbPJdEr1CEHnOeEuVbBkePmP-mzTH=mZ3PWAPMEXPiMjkQ@mail.gmail.com>

On Tue, Aug 26, 2014 at 1:09 PM, Philipp ?berbacher <murks at tuxfamily.org> wrote:
> and briefly luppp but could not make heads or tails of it yet.
Luppp does not (at time of writing) support MIDI. There is ongoing
development in
this area too, but note this is not yet available or stable for testing.

> the, as far as I remember, wonderful wavetable synth crashes the hosts.
Full disclosure, last I checked, the wavetable synth had to be compiled with
--build-experimental-plugins   or similar flag. If those are built into distro
packages, its not the Calf devs fault..

> There seems to be a fork of the calf instruments and plugins by falktx
> but the wavetable synth and the GUIs seem to be missing.
Re wavetable synth missing: good. Re GUI's, I don't know.

> My goal at the moment is to create a single, simple and probably quite
> horrible song electronic music song. That's why I'm looking for a nice
> sequencer and some instruments.
I will advise QTractor, and some LV2 instruents. I use Fabla for drum beats,
Sorcer for some dirty basslines, LMMS (exported to audio) for some strange
things, ZynAddSubFX for pads / bells / general, and various LinuxSampler /
QSynth patches for the rest. Its quite a toolset to get to grips with,
but it does work.

QTractor has good support for automation (although its a little
"hidden" in the UI at first,
its actually fine to work with once its learned how-to :)

HTH, -Harry

From falktx at gmail.com  Tue Aug 26 22:34:02 2014
From: falktx at gmail.com (Filipe Coelho)
Date: Tue, 26 Aug 2014 23:34:02 +0100
Subject: [LAU] DISTRHO: New plugins and minor fixing (2014-08-26)
Message-ID: <53FD0B5A.3070502@gmail.com>

In this release we bring 4 new Linux plugin ports:
- EasySSP
- LUFS Meter
- Luftikus
- Stereo Source Separator
(Go to http://distrho.sourceforge.net/ports to see the current list of 
Linux ports.)

The DPF-based plugins also had some minor fixes:
- 3BandEQ/Splitter had its sliders inverted, now fixed
- ProM now has pre-compiled linux binaries; UI can be resized by using - 
and + keys
- MVerb knobs order has been fixed
- Allow to open UI in LV2 hosts that don't support options feature (Ingen)
- Workaround for some VST hosts that don't set sample rate during init 
(Ardour3 and energyXT)

------------------------------------------------------------------------------------------------------------------------

Users of KXStudio repositories already have the latest release.
All distrho plugins and ports are part of the KXStudio meta-packages, 
but in case you don't want that you can install them manually:

    |sudo apt-get install distrho-mini-series distrho-mverb
    distrho-nekobi distrho-prom distrho-plugin-ports
    sudo apt-get install arctican-plugins dexed drowaudio-plugins
    juced-plugins klangfalter obxd pitcheddelay tal-plugins wolpertinger|
    |sudo apt-get install easyssp lufsmeter luftikus|

If you need to download debs manually, go here 
https://launchpad.net/~kxstudio-debian/+archive/ubuntu/plugins/+packages 
<https://launchpad.net/%7Ekxstudio-debian/+archive/ubuntu/plugins/+packages>

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From gnome at hawaii.rr.com  Tue Aug 26 22:37:17 2014
From: gnome at hawaii.rr.com (gnome at hawaii.rr.com)
Date: Tue, 26 Aug 2014 22:37:17 +0000
Subject: [LAU] Noo 'pooter :)
In-Reply-To: <20140826212604.GB587@digit.localdomain>
Message-ID: <20140826223717.KU0EI.103947.root@dnvrco-web18>

---- David Adler <david.jo.adler at gmail.com> wrote: 
> On Tue, Aug 26, 2014 at 09:24:39PM +0100, Will Godfrey wrote:
> 
> > As this will be a clean install, I'm wondering what people might suggest as
> > for best distro to make full use of it - all my other machines have had a
> > progression of debian upgrades so are probably full of crud.
> 
> Use Arch. It might sound counter-intuitive but despite (or because
> of(?)) the rolling release model it requires very little maintenance. 
> 
> The regular glimpse on the homepage's news feed is recommended but
> it's been a long time since anything popped up there that actually
> required manual intervention. If this happens, the instructions have
> proven to be adequate. Other than that, occasionally configuration files 
> suffixed *.pacnew/*.pacsave need to be merged and voil?, you have a
> crud-free up-to-date system that won't send you to dependency hell when 
> attempting to install recent software.
> 
> The above might sound a bit like over-optimistic marketing speak but it
> reflects my experience and from what I've heard it's not just me. 
> 
> That said, Debian testing didn't exactly give me headaches -- it'd be my
> second choice for audio -- but my experiences with Arch (quite a few
> years now, no re-installation) are plainly positive.
> 
> https://wiki.archlinux.org/index.php/Arch_Linux_system_maintenance

I've been running my systems on Aptosid (Debian Sid) that does a pretty good job (generally) of smoothing out the quirks of Sid.

My only problem with Debian releases is that sometimes they're way behind on application versions.

Right now, Sid is kind of in between when it comes to switching from init to systemd. That might or might not cause problems.

David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com



From gnome at hawaii.rr.com  Tue Aug 26 22:42:50 2014
From: gnome at hawaii.rr.com (gnome at hawaii.rr.com)
Date: Tue, 26 Aug 2014 22:42:50 +0000
Subject: [LAU] Noo 'pooter :)
In-Reply-To: <20140826212439.352361f6@debian>
Message-ID: <20140826224250.LAWK6.104014.root@dnvrco-web18>


---- Will Godfrey <willgodfrey at musically.me.uk> wrote: 
> In the next couple of days I should be getting a new fanless dual core machine
> with a 64bit intel cpu running at 3.1G. This is really intended to replace my
> ageing 'office' machine, but I thought I might as well set it up for decent
> audio too.
> 
> As this will be a clean install, I'm wondering what people might suggest as
> for best distro to make full use of it - all my other machines have had a
> progression of debian upgrades so are probably full of crud.

I just replaced an unstable AMD Phenom box with an Intel Pentium (64-bit, dual core, 3GHz) box. A fresh install of Debian Sid (sans Aptosid) is running happily on it. I tried simply booting the old machine's installed Aptosid on it, but apparently the jump from Phenom to 4th gen Haswell Intel was too much and it wouldn't work.

I miss my Audiophile :( but the built-in audio works just fine.

David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com


From murks at tuxfamily.org  Wed Aug 27 00:30:07 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Wed, 27 Aug 2014 02:30:07 +0200
Subject: [LAU] [Bulk] Re:  [Bulk] Re:  Sequencer suggestions?
In-Reply-To: <20140826191204.GA11173@linuxaudio.org>
References: <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
 <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
 <20140826160630.GC16859@linuxaudio.org>
 <1409069791.7093.33.camel@rocketmail.com>
 <20140826162353.GD16859@linuxaudio.org>
 <1409070990.7093.37.camel@rocketmail.com>
 <20140826184637.22ffebca@eeyore.mozart.uni-klu.ac.at>
 <20140826191204.GA11173@linuxaudio.org>
Message-ID: <20140827023007.47139273@eeyore.mozart.uni-klu.ac.at>

On Tue, 26 Aug 2014 19:12:04 +0000
Fons Adriaensen <fons at linuxaudio.org> wrote:

> On Tue, Aug 26, 2014 at 06:46:37PM +0200, Philipp ?berbacher wrote:
>  
> > And with the non-git package you can basically forget to get the
> > patch in, it is 'maintained' by speps who basically just sits on
> > tons of audio packages without doing anything. This is one 'trusted
> > user' I do not trust to close his own zippers. Better just create
> > another package, it will be faster by years.
> > 
> > I just tried to patch the git package but something goes very wrong
> > here. I get no error but it looks like the patch isn't applied
> > either and I can't get rid of the sed error either. I haven't done
> > this sort of thing in years.
> 
> The patch is against the AUR phasex package 0.14.97 maintained 
> by William Weston.  
> 
> If you apply the patch to sources obtained via AUR you need
> 
>  makepkg -e -f
> 
> without the -e the original sources will be extracted and
> overwrite the patch.
> 
> Ciao,

Ok, that clarifies where the patch is against, thanks. I tried to patch
via modifying the PKGBUILD but didn't manage to. Guess I should try it
'manually' first, thanks for the hint with -e. I'll try tomorrow with a
clear head.

Philipp

From murks at tuxfamily.org  Wed Aug 27 01:12:01 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Wed, 27 Aug 2014 03:12:01 +0200
Subject: [LAU] Noo 'pooter :)
In-Reply-To: <20140826212604.GB587@digit.localdomain>
References: <20140826212439.352361f6@debian>
 <20140826212604.GB587@digit.localdomain>
Message-ID: <20140827031201.3b4496d5@eeyore.mozart.uni-klu.ac.at>

On Tue, 26 Aug 2014 23:26:04 +0200
David Adler <david.jo.adler at gmail.com> wrote:

> On Tue, Aug 26, 2014 at 09:24:39PM +0100, Will Godfrey wrote:
> 
> > As this will be a clean install, I'm wondering what people might
> > suggest as for best distro to make full use of it - all my other
> > machines have had a progression of debian upgrades so are probably
> > full of crud.
> 
> Use Arch. It might sound counter-intuitive but despite (or because
> of(?)) the rolling release model it requires very little maintenance. 

Hi David. I'm alive! ;)

> The regular glimpse on the homepage's news feed is recommended but
> it's been a long time since anything popped up there that actually
> required manual intervention. If this happens, the instructions have
> proven to be adequate. Other than that, occasionally configuration
> files suffixed *.pacnew/*.pacsave need to be merged and voil?, you
> have a crud-free up-to-date system that won't send you to dependency
> hell when attempting to install recent software.

Agreed, there is the occasional message on the front page, but it's
about once a month or even more seldom. Usually I notice that something
changed after I run 'pacman -Syu' but before actually going through
with it. Renaming of packages and the likes are a good indicator for
bigger changes. No real problems with that so far. I update very
frequently though, so if there is a problem, which happens but happens
rarely, I know where it comes from and can fix it right away.
I don't completely agree on crud-free. If you install software, run it
and it creates files in your home directory, then delete the software
the crud in your home will stick around, but I guess this is the same
with almost every distro. There might be some system level crud over
the years, I'm not sure, it didn't cause any problems yet.
Merging the occasional .pacnew file takes maybe five minutes and the
most frequent candidate is /etc/pacman.d/mirrorlist with a hand ful of
changes, so not terribly important.

> The above might sound a bit like over-optimistic marketing speak but
> it reflects my experience and from what I've heard it's not just me. 
> 
> That said, Debian testing didn't exactly give me headaches -- it'd be
> my second choice for audio -- but my experiences with Arch (quite a
> few years now, no re-installation) are plainly positive.
> 
> https://wiki.archlinux.org/index.php/Arch_Linux_system_maintenance
> 
> 
> greetz,
>  -d

AUR could use a going-over though, there's quite some audio stuff there
that's not building, no longer available or whatever. Even if
someone takes the time it's still somewhat difficult though since speps
still sits on most audio packages like a hen.

Regards,
Philipp

From ralf.mardorf at rocketmail.com  Wed Aug 27 01:56:37 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 27 Aug 2014 03:56:37 +0200
Subject: [LAU] [Bulk]  Noo 'pooter :)
In-Reply-To: <20140826212439.352361f6@debian>
References: <20140826212439.352361f6@debian>
Message-ID: <1409104597.13214.85.camel@rocketmail.com>

On Tue, 2014-08-26 at 21:24 +0100, Will Godfrey wrote:
> I'm wondering what people might suggest as for best distro to make
> full use of it

I don't think there is something like the best distro, it's a matter of
taste.

https://www.archlinux.org/

The advantage of Arch Linux are
- it follows stable versions from upstream
- it doesn't split software from upstream into strange packages (you
  might see it as an disadvantage, when the packages aren't split so
  that the headers are installed without a dev package)
- The official repositories for packages are provided in a BSD port like
  way, called ABS
- packages not available by the official repositories are provided by
  trusted users on the AUR page, so the amount of available packages is
  similar to Debian
- the package management is idiot prove (catchwords: pacman, yaourt,
  PKGBUILD)
- MOST IMPORTANT: Arch Linux installs are comparable to the Debian
  expert install, but the Arch install really doesn't install unneeded
  software, you need to do everything on your own and end up without
  unneeded services, you will be aware how to set up things, the Arch
  Wikis are amazing good.
  Arch Linux is one of the distros used by several LAU and LAD
  subscribers.


From ralf.mardorf at rocketmail.com  Wed Aug 27 02:13:24 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 27 Aug 2014 04:13:24 +0200
Subject: [LAU] [Bulk] Re:  Noo 'pooter :)
In-Reply-To: <20140827031201.3b4496d5@eeyore.mozart.uni-klu.ac.at>
References: <20140826212439.352361f6@debian>
 <20140826212604.GB587@digit.localdomain>
 <20140827031201.3b4496d5@eeyore.mozart.uni-klu.ac.at>
Message-ID: <1409105604.13214.87.camel@rocketmail.com>

On Wed, 2014-08-27 at 03:12 +0200, Philipp ?berbacher wrote:
> Merging the occasional .pacnew file takes maybe five minutes and the
> most frequent candidate is /etc/pacman.d/mirrorlist with a hand ful of
> changes, so not terribly important.

Most of the times I ignore .pacnews ;).

JFTR there are also a few audio folks

$ cat /etc/pacman.conf
# [snip]
# ArchAudio Production
[archaudio-production]
Server = http://repos.archaudio.org/$repo/$arch
# [snip]

Btw. the pacman.conf's IgnorePkg and NoExtract options could be very
useful.

http://archaudio.org/



From ralf.mardorf at rocketmail.com  Wed Aug 27 05:25:41 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 27 Aug 2014 07:25:41 +0200
Subject: [LAU] Lightweight WM that can be used without any issue with audio
 applications
Message-ID: <1409117141.13214.101.camel@rocketmail.com>

I switched from Xfce4 to JWM. JWM does work with all apps excepted of
QjackCtl. Often I need to kill QjackCtl, because the Connect window
always is above all other windows and it's impossible to close it. I
don't want to use a tiling window manager or a WM that is nearly as
bloated, as DEs such as Xfce4. What WMs do you use?

A lightweight DE would be ok too, as long as it isn't unstable and has
got no dependencies to GNOME software. Bad coded stuff like
Enlightenment is a complete no-go. I want to handle windows with the
mouse, they should be free to overlap and I don't want to experience
issues, even not issues that happens seldom, workspaces should be
editable. JWM is nearly perfect, excepted of the QjackCTL issue.


From email+music at blaise.ca  Wed Aug 27 05:53:01 2014
From: email+music at blaise.ca (Blaise Alleyne)
Date: Wed, 27 Aug 2014 01:53:01 -0400
Subject: [LAU] Noo 'pooter :)
In-Reply-To: <20140826212439.352361f6@debian>
References: <20140826212439.352361f6@debian>
Message-ID: <53FD723D.7050004@blaise.ca>

On 26/08/14 04:24 PM, Will Godfrey wrote:
> In the next couple of days I should be getting a new fanless dual core machine
> with a 64bit intel cpu running at 3.1G. This is really intended to replace my
> ageing 'office' machine, but I thought I might as well set it up for decent
> audio too.
> 
> As this will be a clean install, I'm wondering what people might suggest as
> for best distro to make full use of it - all my other machines have had a
> progression of debian upgrades so are probably full of crud.
> 

I read this article the other day recommending KXStudio or AVLinux, though I
haven't tried either yet myself:

http://www.libremusicproduction.com/articles/advantages-choosing-audio-orientated-linux-distribution


Seems like the recommendations from others are better if you want to configure
your box and have more control over it, but it seems that maybe distros like
KXStudio or AVLinux might be worth considering if you're looking for something
that comes with some sensible defaults for audio production "out of the box".
*shrugs*


From arve.barsnes at gmail.com  Wed Aug 27 06:01:57 2014
From: arve.barsnes at gmail.com (Arve Barsnes)
Date: Wed, 27 Aug 2014 08:01:57 +0200
Subject: [LAU] Lightweight WM that can be used without any issue with
 audio applications
In-Reply-To: <1409117141.13214.101.camel@rocketmail.com>
References: <1409117141.13214.101.camel@rocketmail.com>
Message-ID: <CAJjrzcVGqMGS0zttmhONEQXiz_y3G4cvSyOr=jt70k3q4upfDA@mail.gmail.com>

On 27 August 2014 07:25, Ralf Mardorf <ralf.mardorf at rocketmail.com> wrote:
> I switched from Xfce4 to JWM. JWM does work with all apps excepted of
> QjackCtl. Often I need to kill QjackCtl, because the Connect window
> always is above all other windows and it's impossible to close it.
Sounds weird. Why is it above other windows? Does not happen here. If
you don't have window decorations to close it, doesn't another click
on the Connect button in the main qjackctl window close it?

> What WMs do you use?
Personally, I use openbox. Never had any trouble.

> Bad coded stuff like
> Enlightenment is a complete no-go.
Wondering why you say this though. I thought Enlightenment was pretty
good stuff, and I wanted to try it out sometime.

Arve

From gnome at hawaii.rr.com  Wed Aug 27 06:45:52 2014
From: gnome at hawaii.rr.com (david)
Date: Tue, 26 Aug 2014 20:45:52 -1000
Subject: [LAU] Lightweight WM that can be used without any issue with
 audio applications
In-Reply-To: <1409117141.13214.101.camel@rocketmail.com>
References: <1409117141.13214.101.camel@rocketmail.com>
Message-ID: <53FD7EA0.30008@hawaii.rr.com>

On 08/26/2014 07:25 PM, Ralf Mardorf wrote:
> I switched from Xfce4 to JWM. JWM does work with all apps excepted of
> QjackCtl. Often I need to kill QjackCtl, because the Connect window
> always is above all other windows and it's impossible to close it. I
> don't want to use a tiling window manager or a WM that is nearly as
> bloated, as DEs such as Xfce4. What WMs do you use?
>
> A lightweight DE would be ok too, as long as it isn't unstable and has
> got no dependencies to GNOME software. Bad coded stuff like
> Enlightenment is a complete no-go. I want to handle windows with the
> mouse, they should be free to overlap and I don't want to experience
> issues, even not issues that happens seldom, workspaces should be
> editable. JWM is nearly perfect, excepted of the QjackCTL issue.

Hmm, what're your issues with "bloat" in Xfce4? Has worked here for many 
years on some pretty limited (old) hardware.

Only times I've had to go to something less "bloated" I like OpenBox. 
Only had to do that for memory-intensive graphics processing, so I don't 
recall encountering any problems with QJackCtl.

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From jeremy at autostatic.com  Wed Aug 27 07:47:12 2014
From: jeremy at autostatic.com (Jeremy Jongepier)
Date: Wed, 27 Aug 2014 09:47:12 +0200
Subject: [LAU] Lightweight WM that can be used without any issue with
 audio applications
In-Reply-To: <CAJjrzcVGqMGS0zttmhONEQXiz_y3G4cvSyOr=jt70k3q4upfDA@mail.gmail.com>
References: <1409117141.13214.101.camel@rocketmail.com>
 <CAJjrzcVGqMGS0zttmhONEQXiz_y3G4cvSyOr=jt70k3q4upfDA@mail.gmail.com>
Message-ID: <53FD8D00.9040202@autostatic.com>

On 08/27/2014 08:01 AM, Arve Barsnes wrote:
> On 27 August 2014 07:25, Ralf Mardorf <ralf.mardorf at rocketmail.com> wrote:
>> I switched from Xfce4 to JWM. JWM does work with all apps excepted of
>> QjackCtl. Often I need to kill QjackCtl, because the Connect window
>> always is above all other windows and it's impossible to close it.

Untick "Keep child windos always on top" in Setup - Misc.

> Sounds weird. Why is it above other windows? Does not happen here. If
> you don't have window decorations to close it, doesn't another click
> on the Connect button in the main qjackctl window close it?
> 
>> What WMs do you use?
> Personally, I use openbox. Never had any trouble.

Same here. In the past I had issues with Java applications but with the
version I'm using now (3.5.0) I don't recall having any issues.

Jeremy


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From gnome at hawaii.rr.com  Wed Aug 27 09:05:46 2014
From: gnome at hawaii.rr.com (david)
Date: Tue, 26 Aug 2014 23:05:46 -1000
Subject: [LAU] KXStudio reminded me of why I don't like ZynAddSubFX
Message-ID: <53FD9F6A.30805@hawaii.rr.com>

Booted live KXStudio 14.04 DVD. Cadence started JACK, pointed uselessly 
at the onboard audio. I pointed it at my USB sound card with the same 
settings I use normally. Started Zyn. It visibly made sound (Zyn's 
indicator), but no sound came out. (I found nothing in Cadence to 
connect Zyn to JACK. But then I don't know Cadence.) I ran KMixer and 
audio volume was up for the USB card. So I said, Oh, well, guess it 
doesn't work with Zyn. (Not the first time I've had problems with Zyn 
and JACK, I much prefer Yoshimi.) So I close Zyn. After which Cadence 
reports that JACK server is stopped. Restarting JACK or even force 
restarting a JACK restart using Cadence gives me no audio devices to 
pick from.

KXStudio isn't ready for this user yet, I guess.

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From murks at tuxfamily.org  Wed Aug 27 09:24:23 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Wed, 27 Aug 2014 11:24:23 +0200
Subject: [LAU] Lightweight WM that can be used without any issue with
 audio applications
In-Reply-To: <1409117141.13214.101.camel@rocketmail.com>
References: <1409117141.13214.101.camel@rocketmail.com>
Message-ID: <20140827112423.6e7f760a@eeyore.mozart.uni-klu.ac.at>

On Wed, 27 Aug 2014 07:25:41 +0200
Ralf Mardorf <ralf.mardorf at rocketmail.com> wrote:

> I switched from Xfce4 to JWM. JWM does work with all apps excepted of
> QjackCtl. Often I need to kill QjackCtl, because the Connect window
> always is above all other windows and it's impossible to close it. I
> don't want to use a tiling window manager or a WM that is nearly as
> bloated, as DEs such as Xfce4. What WMs do you use?

I used to experiment with WMs but I use Openbox since quite a while. It
does all the common stuff just fine, the only thing is that you get
into the intricacies of menu building. I ended up managing my menus
manually, which can be annoying, but at least they are categorised as I
want them too and not full of crud. There are helper programs for that
but they could be better (I'm thinking of obmenu). The other thing is
that the configurations are all XML and for advanced stuff it may be
necessary to edit the XML files by hand, something I do not enjoy.
Since a while ago I even have simple manual tiling working, without any
additional software.


> A lightweight DE would be ok too, as long as it isn't unstable and has
> got no dependencies to GNOME software. Bad coded stuff like
> Enlightenment is a complete no-go.

I actually like enlightenment (e17) for some reason. When I tested it
in the past it was a bit buggy though, but almost worse was that all
the themes had a lot of *bling*, meaning artificial gloss and fancy
animations and other stuff I can't stand. Of course they wanted to show
off that they can do that with low resource usage and without
compositing, which is cool, but they should have included at least one
really boring simple theme. Maybe I'll give it another spin some time.

Regards,
Philipp

From david.jo.adler at gmail.com  Wed Aug 27 09:42:45 2014
From: david.jo.adler at gmail.com (David Adler)
Date: Wed, 27 Aug 2014 11:42:45 +0200
Subject: [LAU]  Noo 'pooter :)
In-Reply-To: <20140827031201.3b4496d5@eeyore.mozart.uni-klu.ac.at>
References: <20140826212439.352361f6@debian>
 <20140826212604.GB587@digit.localdomain>
 <20140827031201.3b4496d5@eeyore.mozart.uni-klu.ac.at>
Message-ID: <20140827094245.GA8513@digit.localdomain>

On Wed, Aug 27, 2014 at 03:12:01AM +0200, Philipp ?berbacher wrote:
> On Tue, 26 Aug 2014 23:26:04 +0200 David Adler wrote:
> > On Tue, Aug 26, 2014 at 09:24:39PM +0100, Will Godfrey wrote:
> > 
> > > As this will be a clean install, I'm wondering what people might
> > > suggest as for best distro to make full use of it - all my other
> > > machines have had a progression of debian upgrades so are probably
> > > full of crud.
> > 
> > Use Arch. It might sound counter-intuitive but despite (or because
> > of(?)) the rolling release model it requires very little maintenance. 
> 
> Hi David. I'm alive! ;)

Same here. ;)

> > The regular glimpse on the homepage's news feed is recommended but
> > it's been a long time since anything popped up there that actually
> > required manual intervention. If this happens, the instructions have
> > proven to be adequate. Other than that, occasionally configuration
> > files suffixed *.pacnew/*.pacsave need to be merged and voil?, you
> > have a crud-free up-to-date system that won't send you to dependency
> > hell when attempting to install recent software.
>
...
> I don't completely agree on crud-free. If you install software, run it
> and it creates files in your home directory, then delete the software
> the crud in your home will stick around, but I guess this is the same
> with almost every distro. There might be some system level crud over
> the years, I'm not sure, it didn't cause any problems yet.

Admittedly, I might have overindulged in marketing-speak here. But
package managers not touching stuff in $HOME is a feature, not a bug.
And it's not the distro's fault if some nanny-software unsolicitedly
clutters it with directories like "Audio Projects" or "Video Projects"
(yes, including the white space). Orphaned configuration files in home 
are usually tiny, with just 16 gig of HD I would have noticed if 
otherwise.

I haven't seen system level crud accumulating over the years but maybe I
haven't been looking diligently enough. Whenever "accidentally"
installing bloated stuff like a Java VM, 'packman -Rsn' will elegantly 
solve this -- the distro is not to blame. :)

> AUR could use a going-over though, there's quite some audio stuff there
> that's not building, no longer available or whatever. Even if
> someone takes the time it's still somewhat difficult though since speps
> still sits on most audio packages like a hen.

Agreed, there you touched a disadvantage of Arch compared to Debian.
Many packages that would be installable via apt are only found in AUR
and are of mixed quality. The speps-phenomenon is awkward indeed,
especially considering (s)he's filed as "trusted user". In many cases, 
the AUR really is more of a starting point for DIY than a serious 
repository.


regards,
 -david


From fons at linuxaudio.org  Wed Aug 27 09:50:49 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Wed, 27 Aug 2014 09:50:49 +0000
Subject: [LAU] [Bulk]  Noo 'pooter :)
In-Reply-To: <1409104597.13214.85.camel@rocketmail.com>
References: <20140826212439.352361f6@debian>
 <1409104597.13214.85.camel@rocketmail.com>
Message-ID: <20140827095049.GA21559@linuxaudio.org>

On Wed, Aug 27, 2014 at 03:56:37AM +0200, Ralf Mardorf wrote:

> On Tue, 2014-08-26 at 21:24 +0100, Will Godfrey wrote:

> > I'm wondering what people might suggest as for best distro to make
> > full use of it
> 
> I don't think there is something like the best distro, it's a matter of
> taste.
> 
> https://www.archlinux.org/

My own machines and all those at the CdM and CdS have been running
Archlinux for the last four or five years.

It's the right distro if you want to stay in control, but that means
you have to invest some time and learn how to set up things. For the
first install you'll have to do some things manually (e.g. partition
disks, install a boot loader etc.), but the wiki explaining all this
is very good. Package management via pacman is excellent and things
get updated rater rapidly.

Support on the arch-general list is OK, but note that the Arch devs
can be on the arrogant side if you ask questions but fail to read
essential documentation first, or if you question their wisdom. 

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From falktx at gmail.com  Wed Aug 27 10:06:07 2014
From: falktx at gmail.com (Filipe Coelho)
Date: Wed, 27 Aug 2014 11:06:07 +0100
Subject: [LAU] KXStudio reminded me of why I don't like ZynAddSubFX
In-Reply-To: <53FD9F6A.30805@hawaii.rr.com>
References: <53FD9F6A.30805@hawaii.rr.com>
Message-ID: <53FDAD8F.8080608@gmail.com>

On 08/27/2014 10:05 AM, david wrote:
> Booted live KXStudio 14.04 DVD. Cadence started JACK, pointed 
> uselessly at the onboard audio. I pointed it at my USB sound card with 
> the same settings I use normally. Started Zyn. It visibly made sound 
> (Zyn's indicator), but no sound came out. (I found nothing in Cadence 
> to connect Zyn to JACK. But then I don't know Cadence.) I ran KMixer 
> and audio volume was up for the USB card. So I said, Oh, well, guess 
> it doesn't work with Zyn. (Not the first time I've had problems with 
> Zyn and JACK, I much prefer Yoshimi.) So I close Zyn. After which 
> Cadence reports that JACK server is stopped. Restarting JACK or even 
> force restarting a JACK restart using Cadence gives me no audio 
> devices to pick from.
>
> KXStudio isn't ready for this user yet, I guess.
>
I don't remember getting a bug report about this, unless this counts as one?

In any case, the Zyn in the kxstudio repos does not auto-connect to 
system outputs.
This is so it can be safely used in a session manager environment.

The 14.04 ISO is a bit old now.
There has been some serious Cadence improvements and a new Zyn release 
since then.
A new ISO will be released in a few days.

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From ralf.mardorf at rocketmail.com  Wed Aug 27 10:52:40 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 27 Aug 2014 12:52:40 +0200
Subject: [LAU] [Bulk] Re: Lightweight WM that can be used without any
 issue with audio applications
In-Reply-To: <20140827112423.6e7f760a@eeyore.mozart.uni-klu.ac.at>
References: <1409117141.13214.101.camel@rocketmail.com>
 <20140827112423.6e7f760a@eeyore.mozart.uni-klu.ac.at>
Message-ID: <1409136760.13214.124.camel@rocketmail.com>

I don't know why the QjackCtl Window sometimes is above the other
windows when using JWM. JWM provides to chose the behavior, but if I
would chose Layer > Above, then at least the Window buttons would be
available. I guess I should test openbox, editing xml files isn't an
issue, I needed to edit JWM's xml config. A small menu is what I want,
for JWM the menu only provides what I often need and by
Alt+F3 xfce4-appfinder is launched. I use JWM with a few Xfce4 apps, but
I replaced many Xfce4 apps by other apps. Xfce4 is stable, RAM and
computer's horsepower aren't issues, but I dislike some Xfce4 stuff. I
don't want crappy GNOME software, I want a perfect terminal emulation,
etc. this is possible with Xfce4, but you need to care about it, remove
gvfs, install roxterm, xfe, rodent or similar, when the Xfce4 defaults
are not exactly what you want. JWM doesn't suffer from
https://en.wikipedia.org/wiki/Stacking_window_manager#Limitations . Does
openbox suffer from those issues?

On Wed, 2014-08-27 at 11:24 +0200, Philipp ?berbacher wrote:
> I actually like enlightenment (e17) for some reason. When I tested it
> in the past it was a bit buggy though, but almost worse was that all
> the themes had a lot of *bling*, meaning artificial gloss and fancy
> animations and other stuff I can't stand. Of course they wanted to show
> off that they can do that with low resource usage and without
> compositing, which is cool, but they should have included at least one
> really boring simple theme. Maybe I'll give it another spin some time.

All the times I tested it, when people claimed that it shouldn't have
bugs anymore, it still was buggy and I also can't stand the themes. They
were ok many years ago, for those who wanted themes in keeping with the
period.

I want themes that are close to GUIs I prefer.

Fit's to my taste:
http://www.alonsoruibal.com/wp-content/uploads/2009/12/rakarrack.png

Doesn't fit to my taste:
http://guitarix.sourceforge.net/slideshow//gx1.png

I don't care about the design, I can live with gloss and fancy, but the
workflow is important. Not only applications nowadays provide a workflow
I dislike, but also DEs, e.g. Unity and GNOME. I learned audio
engineering in the 80th, so it just might be that I'm addicted to
self-explaining, pragmatic workflows, the new playful toy appeal might
have it's advantages too, but it's not the way I want to go.



From ralf.mardorf at rocketmail.com  Wed Aug 27 10:52:48 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 27 Aug 2014 12:52:48 +0200
Subject: [LAU] [Bulk] Re:  Noo 'pooter :)
In-Reply-To: <53FD723D.7050004@blaise.ca>
References: <20140826212439.352361f6@debian> <53FD723D.7050004@blaise.ca>
Message-ID: <1409136768.13214.125.camel@rocketmail.com>

On Wed, 2014-08-27 at 01:53 -0400, Blaise Alleyne wrote:
> Seems like the recommendations from others are better if you want to
> configure your box and have more control over it, but it seems that
> maybe distros like KXStudio or AVLinux might be worth considering if
> you're looking for something that comes with some sensible defaults
> for audio production "out of the box".

The OP is an experienced Debian user, KXStudio, AVLinux and Ubuntu
Studio are ok, but perhaps the OP wants to test another approach. An
issue with AVLinux might be the architecture. My favorite from those
distros is Ubuntu Studio. Keep in mind that Arch Linux is one of the
distros that made the transition to systemd a long time ago, Debian and
Ubuntu will do it soon, that likely could cause issues you don't want to
experience for an "office" machine. Replacing X (Ubuntu) also could
cause issues.



From murks at tuxfamily.org  Wed Aug 27 11:18:10 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Wed, 27 Aug 2014 13:18:10 +0200
Subject: [LAU] [Bulk] Re: Lightweight WM that can be used without any
 issue with audio applications
In-Reply-To: <1409136760.13214.124.camel@rocketmail.com>
References: <1409117141.13214.101.camel@rocketmail.com>
 <20140827112423.6e7f760a@eeyore.mozart.uni-klu.ac.at>
 <1409136760.13214.124.camel@rocketmail.com>
Message-ID: <20140827131810.66ad6bec@eeyore.mozart.uni-klu.ac.at>

On Wed, 27 Aug 2014 12:52:40 +0200
Ralf Mardorf <ralf.mardorf at rocketmail.com> wrote:

> I don't know why the QjackCtl Window sometimes is above the other
> windows when using JWM. JWM provides to chose the behavior, but if I
> would chose Layer > Above, then at least the Window buttons would be
> available. I guess I should test openbox, editing xml files isn't an
> issue, I needed to edit JWM's xml config. A small menu is what I want,
> for JWM the menu only provides what I often need and by
> Alt+F3 xfce4-appfinder is launched. I use JWM with a few Xfce4 apps,
> but I replaced many Xfce4 apps by other apps. Xfce4 is stable, RAM and
> computer's horsepower aren't issues, but I dislike some Xfce4 stuff. I
> don't want crappy GNOME software, I want a perfect terminal emulation,
> etc. this is possible with Xfce4, but you need to care about it,
> remove gvfs, install roxterm, xfe, rodent or similar, when the Xfce4
> defaults are not exactly what you want. JWM doesn't suffer from
> https://en.wikipedia.org/wiki/Stacking_window_manager#Limitations .
> Does openbox suffer from those issues?

Openbox comes pretty much bare bones, so you can use whatever you want
for things like panels, clipboard, terminals and so on, it's just a
window manager with a menu system (and you can fill the menu however
you like).

Editing the XML files is rarely needed, there are programs like obconf,
obmenu and obkey that can do that for you most of the time.

I think it does suffer from that, but what is described there is just
something that happens when a program freezes. That happens rarely
enough not to be a problem.

Regards,
Philipp

From jeremy at autostatic.com  Wed Aug 27 12:10:48 2014
From: jeremy at autostatic.com (Jeremy Jongepier)
Date: Wed, 27 Aug 2014 14:10:48 +0200
Subject: [LAU] [Bulk] Re: Lightweight WM that can be used without any
 issue with audio applications
In-Reply-To: <1409136760.13214.124.camel@rocketmail.com>
References: <1409117141.13214.101.camel@rocketmail.com>
 <20140827112423.6e7f760a@eeyore.mozart.uni-klu.ac.at>
 <1409136760.13214.124.camel@rocketmail.com>
Message-ID: <53FDCAC8.40302@autostatic.com>

On 08/27/2014 12:52 PM, Ralf Mardorf wrote:
> I don't know why the QjackCtl Window sometimes is above the other
> windows when using JWM. 

Did you already check the "Keep child windows on top" setting in
QjackCtl's Setup - Misc window?

> I want themes that are close to GUIs I prefer.
> 
> Fit's to my taste:
> http://www.alonsoruibal.com/wp-content/uploads/2009/12/rakarrack.png
> 
> Doesn't fit to my taste:
> http://guitarix.sourceforge.net/slideshow//gx1.png
> 

First one is a Clearlooks-like theme, that's available for Openbox.
Second is a default Openbox theme. No need to edit any files, just use
obconf to switch themes.

Jeremy

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From ralf.mardorf at rocketmail.com  Wed Aug 27 12:33:29 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 27 Aug 2014 14:33:29 +0200
Subject: [LAU] Lightweight WM that can be used without any issue with
 audio applications
In-Reply-To: <53FDCAC8.40302@autostatic.com>
References: <1409117141.13214.101.camel@rocketmail.com>
 <20140827112423.6e7f760a@eeyore.mozart.uni-klu.ac.at>
 <1409136760.13214.124.camel@rocketmail.com> <53FDCAC8.40302@autostatic.com>
Message-ID: <1409142809.13214.130.camel@rocketmail.com>

On Wed, 2014-08-27 at 14:10 +0200, Jeremy Jongepier wrote:
> Did you already check the "Keep child windows on top" setting in
> QjackCtl's Setup - Misc window?

Yes it's disabled. It's definitively a bug, the Window not only is above
all other Windows and there's no way to change this, often it's also on
all virtual desktops. This doesn't happen always, just sometimes and
only for QjackCtl. It never happened on Xfce4, just with JWM. The Window
buttons might be there, but under JWMs panel. The only thing I can do is
Ctrl+Alt+F2 (tty) and kill QjackCtl. Fortunately jackd and all music
apps can continue to run and only QjackCtl needs to be launched again.


From ralf.mardorf at rocketmail.com  Wed Aug 27 12:40:46 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 27 Aug 2014 14:40:46 +0200
Subject: [LAU] Lightweight WM that can be used without any issue with
 audio applications
In-Reply-To: <1409142809.13214.130.camel@rocketmail.com>
References: <1409117141.13214.101.camel@rocketmail.com>
 <20140827112423.6e7f760a@eeyore.mozart.uni-klu.ac.at>
 <1409136760.13214.124.camel@rocketmail.com> <53FDCAC8.40302@autostatic.com>
 <1409142809.13214.130.camel@rocketmail.com>
Message-ID: <1409143246.13214.131.camel@rocketmail.com>

On Wed, 2014-08-27 at 14:33 +0200, Ralf Mardorf wrote:
> On Wed, 2014-08-27 at 14:10 +0200, Jeremy Jongepier wrote:
> > Did you already check the "Keep child windows on top" setting in
> > QjackCtl's Setup - Misc window?
> 
> Yes it's disabled. It's definitively a bug, the Window not only is above
> all other Windows and there's no way to change this, often it's also on
> all virtual desktops. This doesn't happen always, just sometimes and
> only for QjackCtl. It never happened on Xfce4, just with JWM. The Window
> buttons might be there, but under JWMs panel. The only thing I can do is
> Ctrl+Alt+F2 (tty) and kill QjackCtl. Fortunately jackd and all music
> apps can continue to run and only QjackCtl needs to be launched again.

Next time I will not run  "killall  qjackctl"  but test  "jwm -restart".



From ralf.mardorf at rocketmail.com  Wed Aug 27 12:42:48 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Wed, 27 Aug 2014 14:42:48 +0200
Subject: [LAU] Lightweight WM that can be used without any issue with
 audio applications
In-Reply-To: <1409143246.13214.131.camel@rocketmail.com>
References: <1409117141.13214.101.camel@rocketmail.com>
 <20140827112423.6e7f760a@eeyore.mozart.uni-klu.ac.at>
 <1409136760.13214.124.camel@rocketmail.com> <53FDCAC8.40302@autostatic.com>
 <1409142809.13214.130.camel@rocketmail.com>
 <1409143246.13214.131.camel@rocketmail.com>
Message-ID: <1409143368.13214.132.camel@rocketmail.com>

On Wed, 2014-08-27 at 14:40 +0200, Ralf Mardorf wrote:
> On Wed, 2014-08-27 at 14:33 +0200, Ralf Mardorf wrote:
> > On Wed, 2014-08-27 at 14:10 +0200, Jeremy Jongepier wrote:
> > > Did you already check the "Keep child windows on top" setting in
> > > QjackCtl's Setup - Misc window?
> > 
> > Yes it's disabled. It's definitively a bug, the Window not only is above
> > all other Windows and there's no way to change this, often it's also on
> > all virtual desktops. This doesn't happen always, just sometimes and
> > only for QjackCtl. It never happened on Xfce4, just with JWM. The Window
> > buttons might be there, but under JWMs panel. The only thing I can do is
> > Ctrl+Alt+F2 (tty) and kill QjackCtl. Fortunately jackd and all music
> > apps can continue to run and only QjackCtl needs to be launched again.
> 
> Next time I will not run  "killall  qjackctl"  but test  "jwm -restart".

Can't be done from tty ;D.



From ve4per at outlook.com  Wed Aug 27 12:57:26 2014
From: ve4per at outlook.com (VE4PER / Andy)
Date: Wed, 27 Aug 2014 07:57:26 -0500
Subject: [LAU] KXStudio reminded me of why I don't like ZynAddSubFX
In-Reply-To: <53FD9F6A.30805@hawaii.rr.com>
References: <53FD9F6A.30805@hawaii.rr.com>
Message-ID: <BLU436-SMTP43FD7FA09C95032D4823BB9CDD0@phx.gbl>

Have you checked to ensure jack plugins for your app, if available are 
installed and enabled? Also many pgms don't show in jack patchbay until 
they are up and running and they disappear when audio stops.

FWIW
On 14-08-27 04:05 AM, david wrote:
> Booted live KXStudio 14.04 DVD. Cadence started JACK, pointed 
> uselessly at the onboard audio. I pointed it at my USB sound card with 
> the same settings I use normally. Started Zyn. It visibly made sound 
> (Zyn's indicator), but no sound came out. (I found nothing in Cadence 
> to connect Zyn to JACK. But then I don't know Cadence.) I ran KMixer 
> and audio volume was up for the USB card. So I said, Oh, well, guess 
> it doesn't work with Zyn. (Not the first time I've had problems with 
> Zyn and JACK, I much prefer Yoshimi.) So I close Zyn. After which 
> Cadence reports that JACK server is stopped. Restarting JACK or even 
> force restarting a JACK restart using Cadence gives me no audio 
> devices to pick from.
>
> KXStudio isn't ready for this user yet, I guess.
>


From len at ovenwerks.net  Wed Aug 27 13:11:08 2014
From: len at ovenwerks.net (Len Ovens)
Date: Wed, 27 Aug 2014 06:11:08 -0700 (PDT)
Subject: [LAU] [Bulk] Re:  Noo 'pooter :)
In-Reply-To: <1409136768.13214.125.camel@rocketmail.com>
References: <20140826212439.352361f6@debian> <53FD723D.7050004@blaise.ca>
 <1409136768.13214.125.camel@rocketmail.com>
Message-ID: <alpine.DEB.2.10.1408270604000.2503@scott.cbbs.org>

On Wed, 27 Aug 2014, Ralf Mardorf wrote:

> distros is Ubuntu Studio. Keep in mind that Arch Linux is one of the
> distros that made the transition to systemd a long time ago, Debian and
> Ubuntu will do it soon, that likely could cause issues you don't want to
> experience for an "office" machine. Replacing X (Ubuntu) also could
> cause issues.

The Ubuntu LTS is upstart and will not change. Two years till 1604 which 
will be systemd. Considering where the init -> upsart -> systemd thing has 
gone, I wouldn't want to be working on MIR. It seems in the end Ubuntu 
follows Debian... So MIR (X replacement) may also come and go. Kubuntu it 
seems will keep X alive and well and xubuntu and ubuntustudio may continue 
to use X as well.

--
Len Ovens
www.ovenwerks.net


From jeremy at autostatic.com  Wed Aug 27 13:19:58 2014
From: jeremy at autostatic.com (Jeremy Jongepier)
Date: Wed, 27 Aug 2014 15:19:58 +0200
Subject: [LAU] Lightweight WM that can be used without any issue with
 audio applications
In-Reply-To: <1409143368.13214.132.camel@rocketmail.com>
References: <1409117141.13214.101.camel@rocketmail.com>
 <20140827112423.6e7f760a@eeyore.mozart.uni-klu.ac.at>
 <1409136760.13214.124.camel@rocketmail.com> <53FDCAC8.40302@autostatic.com>
 <1409142809.13214.130.camel@rocketmail.com>
 <1409143246.13214.131.camel@rocketmail.com>
 <1409143368.13214.132.camel@rocketmail.com>
Message-ID: <53FDDAFE.1010200@autostatic.com>

On 08/27/2014 02:42 PM, Ralf Mardorf wrote:
>> > Next time I will not run  "killall  qjackctl"  but test  "jwm -restart".
> Can't be done from tty ;D.

In most cases prepending your command with DISPLAY=:0 should work.
Otherwise check what the value of the $DISPLAY variable is in X.

Jeremy

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From jh at brainiac.com  Wed Aug 27 14:09:23 2014
From: jh at brainiac.com (Joe Hartley)
Date: Wed, 27 Aug 2014 10:09:23 -0400
Subject: [LAU] Noo 'pooter :)
In-Reply-To: <20140826212604.GB587@digit.localdomain>
References: <20140826212439.352361f6@debian>
 <20140826212604.GB587@digit.localdomain>
Message-ID: <20140827100923.123de6c63b9cfb2cb5d34604@brainiac.com>

On Tue, 26 Aug 2014 23:26:04 +0200
David Adler <david.jo.adler at gmail.com> wrote:
> On Tue, Aug 26, 2014 at 09:24:39PM +0100, Will Godfrey wrote:
> 
> > As this will be a clean install, I'm wondering what people might suggest as
> > for best distro to make full use of it - all my other machines have had a
> > progression of debian upgrades so are probably full of crud.
> 
> Use Arch. It might sound counter-intuitive but despite (or because
> of(?)) the rolling release model it requires very little maintenance. 

Another Archer here.  I go way back with Linux, to the days of compiling
Yggdrasil in the early 90s, and even further back than that with SysV and
SunOS Unixes, so I like the amount of control over the OS Arch gives me.

I also like the fact that it will only install what I want in the OS.
I've never had PulseAudio(ptui!) on the machine which has saved me a lot
of aggravation that I'd had with Fedora-based installations.

The wikis for installation are quite good, but the process can be picky.
If at all possible, I recommend starting with a fresh system drive where
you don't care if you muck it up a time or two.  Once it's running, it's 
pretty easy to maintain.  As others have mentioned, every now and then 
something changes that requires some thought before installing (I'm
reminded of the changeover to systemd) but reading the Arch website every
now and then will ket you know of possible stumbling blocks.

Good luck!

-- 
======================================================================
       Joe Hartley - UNIX/network Consultant - jh at brainiac.com
 Without deviation from the norm, "progress" is not possible. - FZappa

From jh at brainiac.com  Wed Aug 27 14:13:08 2014
From: jh at brainiac.com (Joe Hartley)
Date: Wed, 27 Aug 2014 10:13:08 -0400
Subject: [LAU] Lightweight WM that can be used without any issue with
 audio applications
In-Reply-To: <1409117141.13214.101.camel@rocketmail.com>
References: <1409117141.13214.101.camel@rocketmail.com>
Message-ID: <20140827101308.491ee767926c7d97a7106a1b@brainiac.com>

On Wed, 27 Aug 2014 07:25:41 +0200
Ralf Mardorf <ralf.mardorf at rocketmail.com> wrote:

> I switched from Xfce4 to JWM. JWM does work with all apps excepted of
> QjackCtl. Often I need to kill QjackCtl, because the Connect window
> always is above all other windows and it's impossible to close it. I
> don't want to use a tiling window manager or a WM that is nearly as
> bloated, as DEs such as Xfce4. What WMs do you use?

I really like FluxBox, a derivative of BlackBox.

-- 
======================================================================
       Joe Hartley - UNIX/network Consultant - jh at brainiac.com
 Without deviation from the norm, "progress" is not possible. - FZappa

From p8rpp at aol.com  Wed Aug 27 14:38:31 2014
From: p8rpp at aol.com (Peter P.)
Date: Wed, 27 Aug 2014 10:38:31 -0400
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
Message-ID: <20140827143830.GA7320@aol.com>

Dear Community,

I was wondering if anyone had suggestions for audio interfaces that
could be venerable successors to an RME HDSP+Multiface. I am using
mine since over ten years, and hence prefer to stick to laptops with
ExpressCard slots, but am curious in general if this is still one of
the few pro mobile interface for Linux users.

The following features would be required:

-Operation on laptops, either through PCMCIA, ExpressCard, USB, Firewire
-Stable low-latency operation under Linux
-16-18 channels of I/O each, eg half of them analogue, the others digital.

thank you for any hints!

best, Peter

From peter at peterlutek.com  Wed Aug 27 15:17:50 2014
From: peter at peterlutek.com (Peter Lutek)
Date: Wed, 27 Aug 2014 11:17:50 -0400
Subject: [LAU]
 =?utf-8?q?Successor/replacement_for_RME_HDSP+Multiface=3F?=
In-Reply-To: <20140827143830.GA7320@aol.com>
References: <20140827143830.GA7320@aol.com>
Message-ID: <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>

On 2014-08-27 10:38, Peter P. wrote:
> Dear Community,
>
> I was wondering if anyone had suggestions for audio interfaces that
> could be venerable successors to an RME HDSP+Multiface.

an excellent question, peter! i believe there are a number of us 
wondering if our trusty multifaces might eventually give up the ghost!

cheers!
.pltk.

-- 
Peter Lutek
improvising musician in Toronto, Canada
http://peterlutek.com

From idragosani at gmail.com  Wed Aug 27 15:23:32 2014
From: idragosani at gmail.com (Brett McCoy)
Date: Wed, 27 Aug 2014 11:23:32 -0400
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
Message-ID: <CAPNMgw4zf7Z_G6f6UKetSyMkNumRosnWpcey-NTEFSx2iq_Eiw@mail.gmail.com>

On Wed, Aug 27, 2014 at 11:17 AM, Peter Lutek <peter at peterlutek.com> wrote:

> On 2014-08-27 10:38, Peter P. wrote:
>
>> Dear Community,
>>
>> I was wondering if anyone had suggestions for audio interfaces that
>> could be venerable successors to an RME HDSP+Multiface.
>>
>
> an excellent question, peter! i believe there are a number of us wondering
> if our trusty multifaces might eventually give up the ghost!
>

I hope not! I'm not ready to give up on my Multiface II yet.

I imagine the firewire based interfaces are the worthy successors?

-- 
Brett W. McCoy -- http://www.brettwmccoy.com
------------------------------------------------------------------------
"In the rhythm of music a secret is hidden; If I were to divulge it, it
would overturn the world."
    -- Jelaleddin Rumi
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From peter at peterlutek.com  Wed Aug 27 15:27:08 2014
From: peter at peterlutek.com (Peter Lutek)
Date: Wed, 27 Aug 2014 11:27:08 -0400
Subject: [LAU]
 =?utf-8?q?Successor/replacement_for_RME_HDSP+Multiface=3F?=
In-Reply-To: <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
Message-ID: <daa3b263e38f9bd369249cab0792f342@peterlutek.com>

On 2014-08-27 11:17, Peter Lutek wrote:
> On 2014-08-27 10:38, Peter P. wrote:
>> Dear Community,
>>
>> I was wondering if anyone had suggestions for audio interfaces that
>> could be venerable successors to an RME HDSP+Multiface.
>
> an excellent question, peter! i believe there are a number of us
> wondering if our trusty multifaces might eventually give up the 
> ghost!
>

are you using the original multiface, as i am? if so, wouldn't the 
multiface II be an option?

cheers!
.pltk.

-- 
Peter Lutek
improvising musician in Toronto, Canada
http://peterlutek.com

From p8rpp at aol.com  Wed Aug 27 15:45:20 2014
From: p8rpp at aol.com (Peter P.)
Date: Wed, 27 Aug 2014 11:45:20 -0400
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
Message-ID: <20140827154520.GB7320@aol.com>

* Peter Lutek <peter at peterlutek.com> [2014-08-27 11:27]:
> On 2014-08-27 11:17, Peter Lutek wrote:
> >On 2014-08-27 10:38, Peter P. wrote:
> >>Dear Community,
> >>
> >>I was wondering if anyone had suggestions for audio interfaces that
> >>could be venerable successors to an RME HDSP+Multiface.
> >
> >an excellent question, peter! i believe there are a number of us
> >wondering if our trusty multifaces might eventually give up the
> >ghost!
> >
> 
> are you using the original multiface, as i am? if so, wouldn't the
> multiface II be an option?
I understand the Multiface II to be basically the same thing as the
Multiface I except for the analog headphone volume pot which will only
suffer damage from transport and will start to crackle after one year
of (not) using it.

From peter at peterlutek.com  Wed Aug 27 15:52:04 2014
From: peter at peterlutek.com (Peter Lutek)
Date: Wed, 27 Aug 2014 11:52:04 -0400
Subject: [LAU]
 =?utf-8?q?Successor/replacement_for_RME_HDSP+Multiface=3F?=
In-Reply-To: <20140827154520.GB7320@aol.com>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
Message-ID: <c4020a697c44790248a2f9eb17e5ddfe@peterlutek.com>

On 2014-08-27 11:45, Peter P. wrote:
> * Peter Lutek <peter at peterlutek.com> [2014-08-27 11:27]:
>> On 2014-08-27 11:17, Peter Lutek wrote:
>> >On 2014-08-27 10:38, Peter P. wrote:
>> >>Dear Community,
>> >>
>> >>I was wondering if anyone had suggestions for audio interfaces 
>> that
>> >>could be venerable successors to an RME HDSP+Multiface.
>> >
>> >an excellent question, peter! i believe there are a number of us
>> >wondering if our trusty multifaces might eventually give up the
>> >ghost!
>> >
>>
>> are you using the original multiface, as i am? if so, wouldn't the
>> multiface II be an option?
> I understand the Multiface II to be basically the same thing as the
> Multiface I except for the analog headphone volume pot which will 
> only
> suffer damage from transport and will start to crackle after one year
> of (not) using it.

interesting point! perhaps one could remove that pot? still, i look 
forward to suggestions of other replacements...

cheers!
.pltk.

-- 
Peter Lutek
improvising musician in Toronto, Canada
http://peterlutek.com

From brent at keycorner.org  Wed Aug 27 16:08:08 2014
From: brent at keycorner.org (Brent Busby)
Date: Wed, 27 Aug 2014 11:08:08 -0500 (CDT)
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <20140827154520.GB7320@aol.com>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
Message-ID: <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>

On Wed, 27 Aug 2014, Peter P. wrote:

> * Peter Lutek <peter at peterlutek.com> [2014-08-27 11:27]:
>> are you using the original multiface, as i am? if so, wouldn't the
>> multiface II be an option?

> I understand the Multiface II to be basically the same thing as the
> Multiface I except for the analog headphone volume pot which will only
> suffer damage from transport and will start to crackle after one year
> of (not) using it.

Aren't they still making the Multiface II?  I have it (with the PCI-E 
card), and it's worked fantastically for me for seven years.  I 
certainly hope they're not thinking of discontinuing it.

-- 
+ Brent A. Busby	 + "We've all heard that a million monkeys
+ Sr. UNIX Systems Admin +  banging on a million typewriters will
+ University of Chicago	 +  eventually reproduce the entire works of
+ James Franck Institute +  Shakespeare.  Now, thanks to the Internet,
+ Materials Research Ctr +  we know this is not true." -Robert Wilensky

From moshwe at gmail.com  Wed Aug 27 16:27:20 2014
From: moshwe at gmail.com (Moshe Werner)
Date: Wed, 27 Aug 2014 19:27:20 +0300
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
Message-ID: <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>

I think there is a pcmcia mad I interface now. Should work with Linux.
Also there are the fireface 800 and 400 devices that are supported by
ffado. They should have what you're looking for. Another option would be
the rme babyface, now that it is class complient.

I would love to have some recent devices supported under Linux, I believe
all of us do.
For now I'm sticking with rme stuff. I have the hdsp9652 which is working
flawless.

Cheers
Moshe
?????? 27 ???? 2014 19:08, "Brent Busby" <brent at keycorner.org> ???:

> On Wed, 27 Aug 2014, Peter P. wrote:
>
>  * Peter Lutek <peter at peterlutek.com> [2014-08-27 11:27]:
>>
>>> are you using the original multiface, as i am? if so, wouldn't the
>>> multiface II be an option?
>>>
>>
>  I understand the Multiface II to be basically the same thing as the
>> Multiface I except for the analog headphone volume pot which will only
>> suffer damage from transport and will start to crackle after one year
>> of (not) using it.
>>
>
> Aren't they still making the Multiface II?  I have it (with the PCI-E
> card), and it's worked fantastically for me for seven years.  I certainly
> hope they're not thinking of discontinuing it.
>
> --
> + Brent A. Busby         + "We've all heard that a million monkeys
> + Sr. UNIX Systems Admin +  banging on a million typewriters will
> + University of Chicago  +  eventually reproduce the entire works of
> + James Franck Institute +  Shakespeare.  Now, thanks to the Internet,
> + Materials Research Ctr +  we know this is not true." -Robert Wilensky
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user
>
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From paul at linuxaudiosystems.com  Wed Aug 27 17:02:29 2014
From: paul at linuxaudiosystems.com (Paul Davis)
Date: Wed, 27 Aug 2014 13:02:29 -0400
Subject: [LAU] ALSA: always use samplerate_best
In-Reply-To: <CAN0CFw3O5OkbNV+dLXerin4=o2qF9_ESKXu0f1ENSt_r4rRAsg@mail.gmail.com>
References: <CAN0CFw2XLSLGA6ZLQrpkX4J7Vu99=nbcmsr2AuCtYmVU3NhkzQ@mail.gmail.com>
 <alpine.DEB.2.10.1408240724321.2503@scott.cbbs.org>
 <CAN0CFw2w99DRVExGtWHD96Uwyu-2aFiVyCAc97zbyKU-w_m5zA@mail.gmail.com>
 <CAN0CFw3O5OkbNV+dLXerin4=o2qF9_ESKXu0f1ENSt_r4rRAsg@mail.gmail.com>
Message-ID: <CAFa_cKmWkVsXx_AfoVTm1yZsBzfUifPG5eZDONKGJPxA1W2Fug@mail.gmail.com>

On Tue, Aug 26, 2014 at 8:45 AM, Grant <emailgrant at gmail.com> wrote:

> >>> I have a USB DAC that can only handle 16/44.1 as input and output. I
> >>> think ALSA will resample everything to 16/44.1 automatically, but I'd
> >>
> >> I think what I am saying is that for most cases the sample rate of your
> >> audio IF doesn't matter. So adding resampling to everything doesn't make
> >> sense... maybe try without first.
> >
> > I think you're saying that the ALSA resampler won't be used if the
> > upstream application does the resampling itself.  Is that correct?
> > How can I find out if ALSA is the one resampling in a particular
> > scenario?
>
>
> Is there any way to find out if this is happening?  Do some
> applications really do this?
>

depends on the device the application opens.

If you tell it to open a device with a name like "hw:N" or "hw:CARDNAME"
then ALSA will offer only the capabilities present in the audio hardware.
If you open it with a name like "plughw:N" or "plughw:CARD" then ALSA will
do everything it can to honor the format, sampling rate etc. asked for by
the application (including resampling, channel multiplexing etc.).
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From clemens at ladisch.de  Wed Aug 27 17:09:34 2014
From: clemens at ladisch.de (Clemens Ladisch)
Date: Wed, 27 Aug 2014 19:09:34 +0200
Subject: [LAU] ALSA: always use samplerate_best
In-Reply-To: <CAN0CFw3O5OkbNV+dLXerin4=o2qF9_ESKXu0f1ENSt_r4rRAsg@mail.gmail.com>
References: <CAN0CFw2XLSLGA6ZLQrpkX4J7Vu99=nbcmsr2AuCtYmVU3NhkzQ@mail.gmail.com>
 <alpine.DEB.2.10.1408240724321.2503@scott.cbbs.org>
 <CAN0CFw2w99DRVExGtWHD96Uwyu-2aFiVyCAc97zbyKU-w_m5zA@mail.gmail.com>
 <CAN0CFw3O5OkbNV+dLXerin4=o2qF9_ESKXu0f1ENSt_r4rRAsg@mail.gmail.com>
Message-ID: <53FE10CE.8030705@ladisch.de>

Grant wrote:
>> How can I find out if ALSA is the one resampling in a particular
>> scenario?
>
> Is there any way to find out if this is happening?

No, unless the application tells you.

> Do some applications really do this?

For example, mplayer uses its own resampler.


Regards,
Clemens

From hanaghan.osaudio at gmail.com  Wed Aug 27 17:34:04 2014
From: hanaghan.osaudio at gmail.com (Russell Hanaghan)
Date: Wed, 27 Aug 2014 10:34:04 -0700
Subject: [LAU] Noo 'pooter :)
In-Reply-To: <53FD723D.7050004@blaise.ca>
References: <20140826212439.352361f6@debian> <53FD723D.7050004@blaise.ca>
Message-ID: <1406900A-AC9B-4B38-8F4A-6FF3A319A8AB@gmail.com>



~ Russell

> On Aug 26, 2014, at 10:53 PM, Blaise Alleyne <email+music at blaise.ca> wrote:
> 
>> On 26/08/14 04:24 PM, Will Godfrey wrote:
>> In the next couple of days I should be getting a new fanless dual core machine
>> with a 64bit intel cpu running at 3.1G. This is really intended to replace my
>> ageing 'office' machine, but I thought I might as well set it up for decent
>> audio too.
>> 
>> As this will be a clean install, I'm wondering what people might suggest as
>> for best distro to make full use of it - all my other machines have had a
>> progression of debian upgrades so are probably full of crud.
> 
> I read this article the other day recommending KXStudio or AVLinux, though I
> haven't tried either yet myself:
> 
> http://www.libremusicproduction.com/articles/advantages-choosing-audio-orientated-linux-distribution
> 
> 
> Seems like the recommendations from others are better if you want to configure
> your box and have more control over it, but it seems that maybe distros like
> KXStudio or AVLinux might be worth considering if you're looking for something
> that comes with some sensible defaults for audio production "out of the box".
> *shrugs*
> 

I have used KXstudio for sometime, including studio sessions with paying customers. While it may not suit the 'super user', it provides a good, useable platform OOTB. The exclusive tools (cadence, Carla, etc) are quite useful IMHO. The only thing I find is that it needs a pretty beefy hardware platform. KDE is a hungry resource beast. I don't have any issues at all on the i5 w/ 8gb ram box but it's too heavy for older laptops ( my Vaio dual core 1.6 for example). I've only found low latency kernels within Debian stuff, no rt or realtime. 

I'm just a basic user, even after all these years. Been trying to find other OS' that have rt kernels and lighter front ends and tht are not Debian based. In my search thus far, this leaves me with Arch or Gentoo. They are more old school and require reading, and learning new things! :) it may be time for me to do just that... For the new musicians just looking to alternatives away from M$ & Osx, I think the boxed distros are great. It's a bit much to ask a musician to learn arch or gentoo terminal geek speak right out of the gate AFAIC. 


> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

From hanswil at notam02.no  Wed Aug 27 18:55:58 2014
From: hanswil at notam02.no (Hans Wilmers)
Date: Wed, 27 Aug 2014 20:55:58 +0200
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <20140827143830.GA7320@aol.com>
References: <20140827143830.GA7320@aol.com>
Message-ID: <53FE29BE.7080809@notam02.no>

On 08/27/2014 04:38 PM, Peter P. wrote:
> Dear Community,
>
> I was wondering if anyone had suggestions for audio interfaces that
> could be venerable successors to an RME HDSP+Multiface. I am using
> mine since over ten years, and hence prefer to stick to laptops with
> ExpressCard slots, but am curious in general if this is still one of
> the few pro mobile interface for Linux users.
>
> The following features would be required:
>
> -Operation on laptops, either through PCMCIA, ExpressCard, USB, Firewire
> -Stable low-latency operation under Linux
> -16-18 channels of I/O each, eg half of them analogue, the others digital.
>
> thank you for any hints!
>
Both Fireface UCX and UFX can run in class compliant USB2 mode.

I tested both briefly a while ago. Worked fine and stable, but I did not 
test low latency.

/ Hans




---
Hans Wilmers
NOTAM
Sandakerveien 24 D, bygg F3
N-0473 Oslo Norway
tlf.: +47 22358065
mob.: +47 92459361
http://www.notam02.no

From fons at linuxaudio.org  Wed Aug 27 19:11:06 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Wed, 27 Aug 2014 19:11:06 +0000
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <c4020a697c44790248a2f9eb17e5ddfe@peterlutek.com>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <c4020a697c44790248a2f9eb17e5ddfe@peterlutek.com>
Message-ID: <20140827191106.GA9666@linuxaudio.org>

On Wed, Aug 27, 2014 at 11:52:04AM -0400, Peter Lutek wrote:

> >I understand the Multiface II to be basically the same thing as the
> >Multiface I except for the analog headphone volume pot which will
> >only
> >suffer damage from transport and will start to crackle after one year
> >of (not) using it.
> 
> interesting point! perhaps one could remove that pot? still, i look
> forward to suggestions of other replacements...

I've been using a Multiface II for some years, and the headphone
pot has never been a problem. But something else has: the silly
ADAT connectors wich have a shutter instead of a dummy plug to
protect them when not used. The shutter hinges (0.5 mm of plastic)
will break and render the connector useless. Not typical for the
Multiface but for all equipment using those crappy connectors.
 
Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From p8rpp at aol.com  Wed Aug 27 19:22:48 2014
From: p8rpp at aol.com (Peter P.)
Date: Wed, 27 Aug 2014 15:22:48 -0400
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>
Message-ID: <20140827192248.GC7320@aol.com>

* Moshe Werner <moshwe at gmail.com> [2014-08-27 12:27]:
> I think there is a pcmcia mad I interface now. Should work with Linux.
The Madiface with its ExpressCard work fine under Linux and can be
considered the best mobile card with 64 channels I/O each.
> Also there are the fireface 800 and 400 devices that are supported by
> ffado. They should have what you're looking for. 
It would be interesting what the current status of ffado and the
FF400/800 is. Last time I spoke to someone about it, it was more like
"8 channels, and no mixer yet".
> the rme babyface, now that it is class complient.
But with its limited I/O is targeted at other users.
 
> I would love to have some recent devices supported under Linux, I believe
> all of us do.
> For now I'm sticking with rme stuff. I have the hdsp9652 which is working
> flawless.
I can only second that! What a nice card.

best, P

From p8rpp at aol.com  Wed Aug 27 19:23:54 2014
From: p8rpp at aol.com (Peter P.)
Date: Wed, 27 Aug 2014 15:23:54 -0400
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <53FE29BE.7080809@notam02.no>
References: <20140827143830.GA7320@aol.com>
 <53FE29BE.7080809@notam02.no>
Message-ID: <20140827192354.GD7320@aol.com>

* Hans Wilmers <hanswil at notam02.no> [2014-08-27 14:56]:
> On 08/27/2014 04:38 PM, Peter P. wrote:
> >Dear Community,
> >
> >I was wondering if anyone had suggestions for audio interfaces that
> >could be venerable successors to an RME HDSP+Multiface. I am using
> >mine since over ten years, and hence prefer to stick to laptops with
> >ExpressCard slots, but am curious in general if this is still one of
> >the few pro mobile interface for Linux users.
> >
> >The following features would be required:
> >
> >-Operation on laptops, either through PCMCIA, ExpressCard, USB, Firewire
> >-Stable low-latency operation under Linux
> >-16-18 channels of I/O each, eg half of them analogue, the others digital.
> >
> >thank you for any hints!
> >
> Both Fireface UCX and UFX can run in class compliant USB2 mode.
> 
> I tested both briefly a while ago. Worked fine and stable, but I did
> not test low latency.
Thanks Hans. How many channels would that mean then? Is there a mixer
GUI available under Linux?

From hanswil at notam02.no  Wed Aug 27 19:58:28 2014
From: hanswil at notam02.no (Hans Wilmers)
Date: Wed, 27 Aug 2014 21:58:28 +0200
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <20140827192354.GD7320@aol.com>
References: <20140827143830.GA7320@aol.com> <53FE29BE.7080809@notam02.no>
 <20140827192354.GD7320@aol.com>
Message-ID: <53FE3864.3090503@notam02.no>

On 08/27/2014 09:23 PM, Peter P. wrote:
> * Hans Wilmers <hanswil at notam02.no> [2014-08-27 14:56]:
 >>
>> Both Fireface UCX and UFX can run in class compliant USB2 mode.
>>
>> I tested both briefly a while ago. Worked fine and stable, but I did
>> not test low latency.
> Thanks Hans. How many channels would that mean then? Is there a mixer
> GUI available under Linux?
>

If I recall correctly, for UCX all the channels are available as in:
http://www.rme-audio.de/en_products_fireface_ucx.php

I can check it out next week with both UCX and UFX.

I am not aware of a mixer GUI for Linux for this card, but I'd guess the 
mixing matrix is not available in class complaint mode anyway. (Does 
anybody know better?)

/ Hans

From jhernberg at alchemy.lu  Wed Aug 27 21:12:30 2014
From: jhernberg at alchemy.lu (Joakim Hernberg)
Date: Wed, 27 Aug 2014 23:12:30 +0200
Subject: [LAU] [Bulk] Re:  Noo 'pooter :)
In-Reply-To: <1409105604.13214.87.camel@rocketmail.com>
References: <20140826212439.352361f6@debian>
 <20140826212604.GB587@digit.localdomain>
 <20140827031201.3b4496d5@eeyore.mozart.uni-klu.ac.at>
 <1409105604.13214.87.camel@rocketmail.com>
Message-ID: <20140827231230.5a4f762c@alchemy.lu>

On Wed, 27 Aug 2014 04:13:24 +0200
Ralf Mardorf <ralf.mardorf at rocketmail.com> wrote:

> On Wed, 2014-08-27 at 03:12 +0200, Philipp ?berbacher wrote:

> Most of the times I ignore .pacnews ;)

Don't know if that's something to be proud of.  One day it's bound to
bite you the proverbial @ss...

The problem is also if you put it off, then once you have to do it,
you're gonna have alot of work instead of just a minute here and there.

--

   Joakim

From termtech at rogers.com  Wed Aug 27 23:24:19 2014
From: termtech at rogers.com (Tim E. Real)
Date: Wed, 27 Aug 2014 19:24:19 -0400
Subject: [LAU] testing... ignore
Message-ID: <12472785.uMzDJX2D7F@col-desktop>

test

From termtech at rogers.com  Wed Aug 27 23:50:08 2014
From: termtech at rogers.com (Tim E. Real)
Date: Wed, 27 Aug 2014 19:50:08 -0400
Subject: [LAU] testing... ignore
Message-ID: <45459679.VKZYm7D8u9@col-desktop>

testing

From ralf.mardorf at rocketmail.com  Thu Aug 28 02:51:38 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Thu, 28 Aug 2014 04:51:38 +0200
Subject: [LAU] [Bulk] Re:  [Bulk] Re:  Noo 'pooter :)
In-Reply-To: <20140827231230.5a4f762c@alchemy.lu>
References: <20140826212439.352361f6@debian>
 <20140826212604.GB587@digit.localdomain>
 <20140827031201.3b4496d5@eeyore.mozart.uni-klu.ac.at>
 <1409105604.13214.87.camel@rocketmail.com>
 <20140827231230.5a4f762c@alchemy.lu>
Message-ID: <1409194298.27627.4.camel@rocketmail.com>

On Wed, 2014-08-27 at 23:12 +0200, Joakim Hernberg wrote:
> On Wed, 27 Aug 2014 04:13:24 +0200
> Ralf Mardorf <ralf.mardorf at rocketmail.com> wrote:
> 
> > On Wed, 2014-08-27 at 03:12 +0200, Philipp ?berbacher wrote:
> 
> > Most of the times I ignore .pacnews ;)
> 
> Don't know if that's something to be proud of.  One day it's bound to
> bite you the proverbial @ss...

Ignore perhaps is the wrong word. I don't merge or replace files, when
the diff just provides a new commented out option, mirror etc. I don't
need. Usually .pacnews exactly do that, they not really provide
something new that is needed.



From dpchrist at holgerdanske.com  Thu Aug 28 03:44:16 2014
From: dpchrist at holgerdanske.com (David Christensen)
Date: Wed, 27 Aug 2014 20:44:16 -0700
Subject: [LAU] band organs
Message-ID: <53FEA590.60306@holgerdanske.com>

linux-audio-user:

I'm a Linux user would like to play (sequence) band organ MIDI files 
using soundfont synthesis.  I have set up a Core Duo laptop with Debian 
Wheezy, recompiled the kernel with the realtime patch, and installed 
various music software packages (RoseGarden, FluidSynth, etc.).


STFW I am able to find free/ open source software (FOSS) MIDI files of 
band organ music rolls, but I am unable to find any FOSS band organ 
soundfont files.


I am also unable to find technical information on band organs, notably 
music roll encoding/ decoding -- e.g. which organ functions and 
instruments/ pitches are on which music roll channels -- and how this is 
dealt with when the rolls are converted to MIDI and played using 
something other than the make and model band organ intended by the arranger.


Any suggestions?


TIA,

David

From ralf.mardorf at rocketmail.com  Thu Aug 28 04:01:48 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Thu, 28 Aug 2014 06:01:48 +0200
Subject: [LAU] Connectors - Was: Successor/replacement for RME
	HDSP+Multiface?
In-Reply-To: <20140827191106.GA9666@linuxaudio.org>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <c4020a697c44790248a2f9eb17e5ddfe@peterlutek.com>
 <20140827191106.GA9666@linuxaudio.org>
Message-ID: <1409198508.8439.1.camel@rocketmail.com>

On Wed, 2014-08-27 at 19:11 +0000, Fons Adriaensen wrote:
> the silly ADAT connectors wich have a shutter instead of a dummy plug
> to protect them when not used. The shutter hinges (0.5 mm of plastic)
> will break and render the connector useless.

The Neutrik and Rean 6.35mm, 1?4" jacks neither mono/unbalance nor
"stereo"/balanced fit to the jack sockets of some gear. I've got very
bad experiences with Behringer gear and btw. the Behringer gear's female
XLR sockets don't provide a locking mechanism. Mating cycles and mating
reliability seems to be out of style.

If somebody should be aware what 6.35mm, 1?4" jacks can be used for the
ADA8000 please let me know. Neutrick and Rean are a PITA, at least for
the sockets of my ADA8000.

SATA connectors also suck, even hot gluing old SATA connectors doesn't
help much.



From gnome at hawaii.rr.com  Thu Aug 28 07:37:36 2014
From: gnome at hawaii.rr.com (david)
Date: Wed, 27 Aug 2014 21:37:36 -1000
Subject: [LAU] KXStudio reminded me of why I don't like ZynAddSubFX
In-Reply-To: <BLU436-SMTP43FD7FA09C95032D4823BB9CDD0@phx.gbl>
References: <53FD9F6A.30805@hawaii.rr.com>
 <BLU436-SMTP43FD7FA09C95032D4823BB9CDD0@phx.gbl>
Message-ID: <53FEDC40.5040402@hawaii.rr.com>

With Cadence, I see no patchbay or any other way to connect anything. On 
my Debian system, with QTJackCtl, Zynn shows up in the QJackCtl 
Connections window the moment I start it, I connect it, whether or not 
it's making any sound. Yoshimi works likewise. When I close Zynn, Jack 
stays running with no problems.

Maybe the problem is Cadence?

On 08/27/2014 02:57 AM, VE4PER / Andy wrote:
> Have you checked to ensure jack plugins for your app, if available are
> installed and enabled? Also many pgms don't show in jack patchbay until
> they are up and running and they disappear when audio stops.
>
> FWIW
> On 14-08-27 04:05 AM, david wrote:
>> Booted live KXStudio 14.04 DVD. Cadence started JACK, pointed
>> uselessly at the onboard audio. I pointed it at my USB sound card with
>> the same settings I use normally. Started Zyn. It visibly made sound
>> (Zyn's indicator), but no sound came out. (I found nothing in Cadence
>> to connect Zyn to JACK. But then I don't know Cadence.) I ran KMixer
>> and audio volume was up for the USB card. So I said, Oh, well, guess
>> it doesn't work with Zyn. (Not the first time I've had problems with
>> Zyn and JACK, I much prefer Yoshimi.) So I close Zyn. After which
>> Cadence reports that JACK server is stopped. Restarting JACK or even
>> force restarting a JACK restart using Cadence gives me no audio
>> devices to pick from.
>>
>> KXStudio isn't ready for this user yet, I guess.

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From gnome at hawaii.rr.com  Thu Aug 28 07:41:34 2014
From: gnome at hawaii.rr.com (david)
Date: Wed, 27 Aug 2014 21:41:34 -1000
Subject: [LAU] KXStudio reminded me of why I don't like ZynAddSubFX
In-Reply-To: <53FDAD8F.8080608@gmail.com>
References: <53FD9F6A.30805@hawaii.rr.com> <53FDAD8F.8080608@gmail.com>
Message-ID: <53FEDD2E.9030309@hawaii.rr.com>

On 08/27/2014 12:06 AM, Filipe Coelho wrote:
> On 08/27/2014 10:05 AM, david wrote:
>> Booted live KXStudio 14.04 DVD. Cadence started JACK, pointed
>> uselessly at the onboard audio. I pointed it at my USB sound card with
>> the same settings I use normally. Started Zyn. It visibly made sound
>> (Zyn's indicator), but no sound came out. (I found nothing in Cadence
>> to connect Zyn to JACK. But then I don't know Cadence.) I ran KMixer
>> and audio volume was up for the USB card. So I said, Oh, well, guess
>> it doesn't work with Zyn. (Not the first time I've had problems with
>> Zyn and JACK, I much prefer Yoshimi.) So I close Zyn. After which
>> Cadence reports that JACK server is stopped. Restarting JACK or even
>> force restarting a JACK restart using Cadence gives me no audio
>> devices to pick from.
>>
>> KXStudio isn't ready for this user yet, I guess.
>>
> I don't remember getting a bug report about this, unless this counts as one?
>
> In any case, the Zyn in the kxstudio repos does not auto-connect to
> system outputs.

Zynn in Debian autoconnects to JACK, but I have to connect it manually 
(that's the way I like it) within JACK.

> This is so it can be safely used in a session manager environment.

OK, makes sense. So how DO you connect Zynn using JACK in KXStudio to 
your desired audio outputs?

> The 14.04 ISO is a bit old now.
> There has been some serious Cadence improvements and a new Zyn release
> since then.
> A new ISO will be released in a few days.

OK. The one I downloaded was 14.04b.

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From falktx at gmail.com  Thu Aug 28 07:58:17 2014
From: falktx at gmail.com (Filipe Coelho)
Date: Thu, 28 Aug 2014 08:58:17 +0100
Subject: [LAU] KXStudio reminded me of why I don't like ZynAddSubFX
In-Reply-To: <53FEDD2E.9030309@hawaii.rr.com>
References: <53FD9F6A.30805@hawaii.rr.com> <53FDAD8F.8080608@gmail.com>
 <53FEDD2E.9030309@hawaii.rr.com>
Message-ID: <53FEE119.9090909@gmail.com>

On 08/28/2014 08:41 AM, david wrote:
> On 08/27/2014 12:06 AM, Filipe Coelho wrote:
>> In any case, the Zyn in the kxstudio repos does not auto-connect to
>> system outputs.
> Zynn in Debian autoconnects to JACK, but I have to connect it manually 
> (that's the way I like it) within JACK.
>> This is so it can be safely used in a session manager environment.
> OK, makes sense. So how DO you connect Zynn using JACK in KXStudio to 
> your desired audio outputs?

You can still use qjackctl, but its jackdbus support is not that great.
Plus it tends to stop jack when closed...

Cadence includes the Catia and Claudia tools to manage connections (see 
Cadence tools tab).
Catia is the simple version that only does the basic stuff,
Claudia is a frontend to LADISH (a session manager) which is obviously a 
bit more complex.

These tools are described into a bit more detail here:
http://kxstudio.sourceforge.net/Applications

>> The 14.04 ISO is a bit old now.
>> There has been some serious Cadence improvements and a new Zyn release
>> since then.
>> A new ISO will be released in a few days.
> OK. The one I downloaded was 14.04b.

The new ISO should be released in 2 weeks tops I hope.
Most things are ready, I just need to update a few packages and do some 
testing.

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From nettings at stackingdwarves.net  Thu Aug 28 15:27:50 2014
From: nettings at stackingdwarves.net (=?ISO-8859-1?Q?J=F6rn_Nettingsmeier?=)
Date: Thu, 28 Aug 2014 17:27:50 +0200
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <20140827192248.GC7320@aol.com>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>
 <20140827192248.GC7320@aol.com>
Message-ID: <53FF4A76.8000100@stackingdwarves.net>

On 08/27/2014 09:22 PM, Peter P. wrote:
> * Moshe Werner <moshwe at gmail.com> [2014-08-27 12:27]:
>> I think there is a pcmcia mad I interface now. Should work with Linux.
> The Madiface with its ExpressCard work fine under Linux and can be
> considered the best mobile card with 64 channels I/O each.
>> Also there are the fireface 800 and 400 devices that are supported by
>> ffado. They should have what you're looking for.
> It would be interesting what the current status of ffado and the
> FF400/800 is. Last time I spoke to someone about it, it was more like
> "8 channels, and no mixer yet".


no personal experience, but i guess the ff400/800 support is fairly 
complete by now. check with jonathan woithe on the ffado list to make sure.


-- 
J?rn Nettingsmeier
Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487

Meister f?r Veranstaltungstechnik (B?hne/Studio)
Tonmeister VDT

http://stackingdwarves.net


From paul at linuxaudiosystems.com  Thu Aug 28 15:48:13 2014
From: paul at linuxaudiosystems.com (Paul Davis)
Date: Thu, 28 Aug 2014 11:48:13 -0400
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <53FF4A76.8000100@stackingdwarves.net>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>
 <20140827192248.GC7320@aol.com>
 <53FF4A76.8000100@stackingdwarves.net>
Message-ID: <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>

On Thu, Aug 28, 2014 at 11:27 AM, J?rn Nettingsmeier <
nettings at stackingdwarves.net> wrote:

>
>
> no personal experience, but i guess the ff400/800 support is fairly
> complete by now. check with jonathan woithe on the ffado list to make sure.
>

just in time for firewire to ride off into the sunset ...
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From moshwe at gmail.com  Thu Aug 28 15:52:12 2014
From: moshwe at gmail.com (Moshe Werner)
Date: Thu, 28 Aug 2014 18:52:12 +0300
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>
 <20140827192248.GC7320@aol.com>
 <53FF4A76.8000100@stackingdwarves.net>
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
Message-ID: <CAGFf1ogKFJp5veLfz1xXN3cAoX_YynKLkiO6AdfS1bnLSOE=pw@mail.gmail.com>

?????? 28 ???? 2014 18:48, "Paul Davis" <paul at linuxaudiosystems.com> ???:
>
>
>
>
> On Thu, Aug 28, 2014 at 11:27 AM, J?rn Nettingsmeier <
nettings at stackingdwarves.net> wrote:
>>
>>
>>
>> no personal experience, but i guess the ff400/800 support is fairly
complete by now. check with jonathan woithe on the ffado list to make sure.
>
>
> just in time for firewire to ride off into the sunset ..

This made me laugh. Interesting what the next technology is going to be.
Thunderbolt? USB 3?
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From paul at linuxaudiosystems.com  Thu Aug 28 15:58:15 2014
From: paul at linuxaudiosystems.com (Paul Davis)
Date: Thu, 28 Aug 2014 11:58:15 -0400
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <CAGFf1ogKFJp5veLfz1xXN3cAoX_YynKLkiO6AdfS1bnLSOE=pw@mail.gmail.com>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>
 <20140827192248.GC7320@aol.com>
 <53FF4A76.8000100@stackingdwarves.net>
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
 <CAGFf1ogKFJp5veLfz1xXN3cAoX_YynKLkiO6AdfS1bnLSOE=pw@mail.gmail.com>
Message-ID: <CAFa_cKnOdzJXNWnHaVq0EwwoOF2FQ34PB0N3+zn=O=KEYUE0KA@mail.gmail.com>

On Thu, Aug 28, 2014 at 11:52 AM, Moshe Werner <moshwe at gmail.com> wrote:

>
> ?????? 28 ???? 2014 18:48, "Paul Davis" <paul at linuxaudiosystems.com> ???:
>
> > On Thu, Aug 28, 2014 at 11:27 AM, J?rn Nettingsmeier <
> nettings at stackingdwarves.net> wrote:
> >>
> >>
> >>
> >> no personal experience, but i guess the ff400/800 support is fairly
> complete by now. check with jonathan woithe on the ffado list to make sure.
> >
> >
> > just in time for firewire to ride off into the sunset ..
>
> This made me laugh. Interesting what the next technology is going to be.
> Thunderbolt? USB 3?
>
USB3 without a doubt.
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From nettings at stackingdwarves.net  Thu Aug 28 16:02:56 2014
From: nettings at stackingdwarves.net (=?UTF-8?B?SsO2cm4gTmV0dGluZ3NtZWllcg==?=)
Date: Thu, 28 Aug 2014 18:02:56 +0200
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
References: <20140827143830.GA7320@aol.com>	<d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>	<daa3b263e38f9bd369249cab0792f342@peterlutek.com>	<20140827154520.GB7320@aol.com>	<alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>	<CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>	<20140827192248.GC7320@aol.com>	<53FF4A76.8000100@stackingdwarves.net>
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
Message-ID: <53FF52B0.4030702@stackingdwarves.net>

On 08/28/2014 05:48 PM, Paul Davis wrote:
>
>
>
> On Thu, Aug 28, 2014 at 11:27 AM, J?rn Nettingsmeier
> <nettings at stackingdwarves.net <mailto:nettings at stackingdwarves.net>> wrote:
>
>
>
>     no personal experience, but i guess the ff400/800 support is fairly
>     complete by now. check with jonathan woithe on the ffado list to
>     make sure.
>
>
> just in time for firewire to ride off into the sunset ...
>

fair enough, but a surprisingly large number of those boxes are lying 
around in institutions and studios (among them, I believe, Ardour 
headquarters :) so it's still kind of useful - the ffs are still more or 
less state-of-the-art in terms of audio quality, so it's well worth the 
extra effort to dig up a firewire card and jump through a few hoops if 
the prize is access to those preamps and converters.


-- 
J?rn Nettingsmeier
Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487

Meister f?r Veranstaltungstechnik (B?hne/Studio)
Tonmeister VDT

http://stackingdwarves.net


From len at ovenwerks.net  Thu Aug 28 16:14:18 2014
From: len at ovenwerks.net (Len Ovens)
Date: Thu, 28 Aug 2014 09:14:18 -0700 (PDT)
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <CAFa_cKnOdzJXNWnHaVq0EwwoOF2FQ34PB0N3+zn=O=KEYUE0KA@mail.gmail.com>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>
 <20140827192248.GC7320@aol.com> <53FF4A76.8000100@stackingdwarves.net>
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
 <CAGFf1ogKFJp5veLfz1xXN3cAoX_YynKLkiO6AdfS1bnLSOE=pw@mail.gmail.com>
 <CAFa_cKnOdzJXNWnHaVq0EwwoOF2FQ34PB0N3+zn=O=KEYUE0KA@mail.gmail.com>
Message-ID: <alpine.DEB.2.10.1408280904290.2476@scott.cbbs.org>

On Thu, 28 Aug 2014, Paul Davis wrote:

> On Thu, Aug 28, 2014 at 11:52 AM, Moshe Werner <moshwe at gmail.com> wrote:
>
>       ?????? 28 ???? 2014 18:48, "Paul Davis" <paul at linuxaudiosystems.com>
>       ???:
>
>       > On Thu, Aug 28, 2014 at 11:27 AM, J?rn Nettingsmeier
>       <nettings at stackingdwarves.net> wrote:
>       >>
>       >>
>       >>
>       >> no personal experience, but i guess the ff400/800 support is
>       fairly complete by now. check with jonathan woithe on the ffado list
>       to make sure.
>       >
>       >
>       > just in time for firewire to ride off into the sunset ..
> 
> This made me laugh. Interesting what the next technology is going to be.
> Thunderbolt? USB 3?
> 
> USB3 without a doubt.

I notice many of the FireWire audio interfaces call themselves Thunderbolt 
compatible, with an adaptor. I would be interested to know if that adaptor 
is just for pinout or needs logic/firmware to work. I think that the FW 
chipsets, like the ICE1712, are all old. I have seen the odd audio 
interface that says USB3 though (as part of a digital mixing console, if I 
recall correctly). However, USB3 will be the next set of new interfaces.

I am sure that about the time most people have bought a USB3 device (audio 
or otherwise) a new interface will come out so everyone can buy evrything 
all over again... It seems about the time I buy a new system, I have to 
buy all new peripherals too.

--
Len Ovens
www.ovenwerks.net

From clemens at ladisch.de  Thu Aug 28 16:37:52 2014
From: clemens at ladisch.de (Clemens Ladisch)
Date: Thu, 28 Aug 2014 18:37:52 +0200
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <alpine.DEB.2.10.1408280904290.2476@scott.cbbs.org>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>
 <20140827192248.GC7320@aol.com>
 <53FF4A76.8000100@stackingdwarves.net>
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
 <CAGFf1ogKFJp5veLfz1xXN3cAoX_YynKLkiO6AdfS1bnLSOE=pw@mail.gmail.com>
 <CAFa_cKnOdzJXNWnHaVq0EwwoOF2FQ34PB0N3+zn=O=KEYUE0KA@mail.gmail.com>
 <alpine.DEB.2.10.1408280904290.2476@scott.cbbs.org>
Message-ID: <53FF5AE0.9040002@ladisch.de>

Len Ovens wrote:
> I notice many of the FireWire audio interfaces call themselves
> Thunderbolt compatible, with an adaptor. I would be interested to know
> if that adaptor is just for pinout or needs logic/firmware to work.

Thunderbolt is just a PCI Express connection (plus the ability to
transport video signals, which does not matter here).

A Thunderbolt-to-FireWire adapter is just a plain PCIe FireWire
controller chip.

> I think that the FW chipsets, like the ICE1712, are all old.

The ICE1712 is a PCI chip.

The FireWire chipsets I know of are:
* OXFW970: died years ago.
* BeBoB: now belongs to Archwave.  The website talks only about USB 2.0
  and "Audiolan".
* iceLynx: used by EchoAudio.  This is a generic FireWire chip that
  is also designed for video applications, which might be the only
  reason it is still sold by TI.  EchoAudio is pivoting to AVB and will
  only grudgingly sell you an AudioFire12.
* DICE: the current chips have been discontinued, but they are building
  the "DICE III" chip, which can also do USB and AVB.


Regards,
Clemens

From jh at brainiac.com  Thu Aug 28 16:44:49 2014
From: jh at brainiac.com (Joe Hartley)
Date: Thu, 28 Aug 2014 12:44:49 -0400
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <alpine.DEB.2.10.1408280904290.2476@scott.cbbs.org>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>
 <20140827192248.GC7320@aol.com>
 <53FF4A76.8000100@stackingdwarves.net>
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
 <CAGFf1ogKFJp5veLfz1xXN3cAoX_YynKLkiO6AdfS1bnLSOE=pw@mail.gmail.com>
 <CAFa_cKnOdzJXNWnHaVq0EwwoOF2FQ34PB0N3+zn=O=KEYUE0KA@mail.gmail.com>
 <alpine.DEB.2.10.1408280904290.2476@scott.cbbs.org>
Message-ID: <20140828124449.23dd0c6a2a386bde288d3096@brainiac.com>

On Thu, 28 Aug 2014 09:14:18 -0700 (PDT)
Len Ovens <len at ovenwerks.net> wrote:
> I think that the FW chipsets, like the ICE1712, are all old. 

The ICE1712 doesn't have anything to do with FW - it's an audio chip.
FW needs a different controlling chip, like a VT6306.

You can still get FW cards and even motherboards with it onboard, but it's
getting a lot more scarce.  I have an Alesis 16 channel FW mixer I use for
live work and it's nice getting discrete channels out for recording as a 
bonus.  

I think USB3 will become much more of a standard than TB for the
exact same reasons that USB2 became much more prevalent than FW.

-- 
======================================================================
       Joe Hartley - UNIX/network Consultant - jh at brainiac.com
 Without deviation from the norm, "progress" is not possible. - FZappa

From len at ovenwerks.net  Thu Aug 28 16:52:10 2014
From: len at ovenwerks.net (Len Ovens)
Date: Thu, 28 Aug 2014 09:52:10 -0700 (PDT)
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <53FF5AE0.9040002@ladisch.de>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>
 <20140827192248.GC7320@aol.com> <53FF4A76.8000100@stackingdwarves.net>
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
 <CAGFf1ogKFJp5veLfz1xXN3cAoX_YynKLkiO6AdfS1bnLSOE=pw@mail.gmail.com>
 <CAFa_cKnOdzJXNWnHaVq0EwwoOF2FQ34PB0N3+zn=O=KEYUE0KA@mail.gmail.com>
 <alpine.DEB.2.10.1408280904290.2476@scott.cbbs.org>
 <53FF5AE0.9040002@ladisch.de>
Message-ID: <alpine.DEB.2.10.1408280949080.2476@scott.cbbs.org>

On Thu, 28 Aug 2014, Clemens Ladisch wrote:

> Len Ovens wrote:
>> I think that the FW chipsets, like the ICE1712, are all old.
>
> The ICE1712 is a PCI chip.

Yes, my meaning was that most FW chipsets seem to be of the same 
generation as the ICE1712, that is obsolete.

--
Len Ovens
www.ovenwerks.net


From moshwe at gmail.com  Thu Aug 28 18:53:17 2014
From: moshwe at gmail.com (Moshe Werner)
Date: Thu, 28 Aug 2014 21:53:17 +0300
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <alpine.DEB.2.10.1408280949080.2476@scott.cbbs.org>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>
 <20140827192248.GC7320@aol.com>
 <53FF4A76.8000100@stackingdwarves.net>
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
 <CAGFf1ogKFJp5veLfz1xXN3cAoX_YynKLkiO6AdfS1bnLSOE=pw@mail.gmail.com>
 <CAFa_cKnOdzJXNWnHaVq0EwwoOF2FQ34PB0N3+zn=O=KEYUE0KA@mail.gmail.com>
 <alpine.DEB.2.10.1408280904290.2476@scott.cbbs.org>
 <53FF5AE0.9040002@ladisch.de>
 <alpine.DEB.2.10.1408280949080.2476@scott.cbbs.org>
Message-ID: <CAGFf1oiWmZtv83RJEXQh2xLHuDwU8UvxMM4uQOQiXAbFBZMQbQ@mail.gmail.com>

Just saw this http://www.rme-audio.de/en_products_madiface_xt.php

They claim it's the first USB3 AI.
Interesting if it's going to be supported under Linux sometime.


On Thu, Aug 28, 2014 at 7:52 PM, Len Ovens <len at ovenwerks.net> wrote:

> On Thu, 28 Aug 2014, Clemens Ladisch wrote:
>
>
>  Len Ovens wrote:
>>
>>> I think that the FW chipsets, like the ICE1712, are all old.
>>>
>>
>> The ICE1712 is a PCI chip.
>>
>
> Yes, my meaning was that most FW chipsets seem to be of the same
> generation as the ICE1712, that is obsolete.
>
> --
> Len Ovens
> www.ovenwerks.net
>
>
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user
>
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From p8rpp at aol.com  Thu Aug 28 20:07:27 2014
From: p8rpp at aol.com (Peter P.)
Date: Thu, 28 Aug 2014 16:07:27 -0400
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <53FF52B0.4030702@stackingdwarves.net>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>
 <20140827192248.GC7320@aol.com>
 <53FF4A76.8000100@stackingdwarves.net>
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
 <53FF52B0.4030702@stackingdwarves.net>
Message-ID: <20140828200726.GA16743@aol.com>

* J?rn Nettingsmeier <nettings at stackingdwarves.net> [2014-08-28 12:03]:
> On 08/28/2014 05:48 PM, Paul Davis wrote:
> >
> >
> >
> >On Thu, Aug 28, 2014 at 11:27 AM, J?rn Nettingsmeier
> ><nettings at stackingdwarves.net <mailto:nettings at stackingdwarves.net>> wrote:
> >
> >
> >
> >    no personal experience, but i guess the ff400/800 support is fairly
> >    complete by now. check with jonathan woithe on the ffado list to
> >    make sure.
> >
> >
> >just in time for firewire to ride off into the sunset ...
Yes, nice comment Paul. I feel I have to keep chasing laptops that
have either an express card slot or a firewire connector.
> >
> 
> fair enough, but a surprisingly large number of those boxes are
> lying around in institutions and studios (among them, I believe,
> Ardour headquarters :) so it's still kind of useful - the ffs are
> still more or less state-of-the-art in terms of audio quality, so
> it's well worth the extra effort to dig up a firewire card and jump
> through a few hoops if the prize is access to those preamps and
> converters.
Which nicely circles back to my question if there are any settings on
the FF400/800 are settable from within Linux, such as, lets say, +48V
and switchable input jacks etc.

Thanks for all the answers which help to get a good picture of the
current not too bright future vision re pro audio interfaces.

P

From murks at tuxfamily.org  Thu Aug 28 20:42:47 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Thu, 28 Aug 2014 22:42:47 +0200
Subject: [LAU] Session management with NSM
Message-ID: <20140828224247.236111e1@eeyore>

Hi there,

I'm just getting my toes wet with NSM and have a couple of questions.

1. NSM configuration. Is there a configuration file? I haven't seen
anything in the manual. The default session root ($HOME/NSM Sessions) is
quite horrible IMHO and I would have liked to configure that. Instead I
created an alias to set the root (alias nsm='non-session-manager --
--session-root ~/.nsm-sessions').

2. Adding programs to sessions through the GUI ("Add Client to
Session") is the only way? Is there no way to attach running clients
or at least have some comfort like tab completion to add clients? 

3. Jack and NSM. How do you handle that? It is possible to start jack
through NSM proxy and I guess it is OK to do that as long as jack
reliably starts before jackpatch (something I'm not sure of). First I
had just jackpatch in there and it started jack for me with a whole lot
of options that are unfamiliar to me and probably not needed.

4. CLI clients. Are they generally not supported? I added the lv2 host
that was recommended to me (jalv) and had to do that through the NSM
proxy, so the settings won't be saved even though the plugin (fabla in
this case) can save its settings. This sort of defeats session
management. With all the CLI tools we have it would be a pitty if that
was generally not supported. On a sidenote, can someone recommend a
plugin host that is supported?

Well, that's it for now. Last time I heard about NSM I got the
impression that it takes care of session management once and for all,
but the first half our gave me a different impression.

Regards,
Philipp

From ralf.mardorf at rocketmail.com  Thu Aug 28 21:07:05 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Thu, 28 Aug 2014 23:07:05 +0200
Subject: [LAU] Update Phasex:  Sequencer suggestions?
In-Reply-To: <20140826162353.GD16859@linuxaudio.org>
References: <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
 <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
 <20140826160630.GC16859@linuxaudio.org>
 <1409069791.7093.33.camel@rocketmail.com>
 <20140826162353.GD16859@linuxaudio.org>
Message-ID: <1409260025.4790.8.camel@rocketmail.com>

1.
        On Tue, 2014-08-26 at 16:23 +0000, Fons Adriaensen wrote:
>         I'm not the maintainer of the AUR package :-) Whoever that is
>         may be lurking here and pick up the fix.
        
        https://aur.archlinux.org/packages/ph/phasex-git/PKGBUILD
        https://aur.archlinux.org/packages/ph/phasex/PKGBUILD

2.
        https://github.com/williamweston/phasex/issues/10



From ralf.mardorf at rocketmail.com  Thu Aug 28 21:29:00 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Thu, 28 Aug 2014 23:29:00 +0200
Subject: [LAU] Update Phasex:  Sequencer suggestions?
In-Reply-To: <1409260025.4790.8.camel@rocketmail.com>
References: <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
 <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
 <20140826160630.GC16859@linuxaudio.org>
 <1409069791.7093.33.camel@rocketmail.com>
 <20140826162353.GD16859@linuxaudio.org>
 <1409260025.4790.8.camel@rocketmail.com>
Message-ID: <1409261340.4790.11.camel@rocketmail.com>

On Thu, 2014-08-28 at 23:07 +0200, Ralf Mardorf wrote:
> 1.
>         On Tue, 2014-08-26 at 16:23 +0000, Fons Adriaensen wrote:
> >         I'm not the maintainer of the AUR package :-) Whoever that is
> >         may be lurking here and pick up the fix.
>         
>         https://aur.archlinux.org/packages/ph/phasex-git/PKGBUILD
>         https://aur.archlinux.org/packages/ph/phasex/PKGBUILD
> 
> 2.
>         https://github.com/williamweston/phasex/issues/10

3.
        Reported: https://aur.archlinux.org/packages.php?K=phasex




From dj_kaza at hotmail.com  Thu Aug 28 23:40:01 2014
From: dj_kaza at hotmail.com (Kaza Kore)
Date: Thu, 28 Aug 2014 23:40:01 +0000
Subject: [LAU] i5 Hyper-Threading, BIOS settings and Arch n00b pointers
In-Reply-To: <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>,
 <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
Message-ID: <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>



Hello all.

I hope this isn't too rambling a post to get some replies...

I have seen on this list multiple times lately (possibly by the same person, but never corrected) a statement similar to thus:
"I chose an i5 over an i7 as the i5 doesn't have Hyperthreading whereas the i7 does."

I always thought I remembered this as wrong, especially as I believed I was running a 2 core i5 in my laptop and it shows up as 4 cores. I have just confirmed this to be the case. But it is partially correct, so if anybody has been taking this information for recommendation towards a new CPU purchase this appears to be a clearer picture of the situation.

"The quick explanation is that all Core i7 CPUs use Hyper-Threading, so a six-core CPU can handle 12 streams, a four-core can handle eight streams, and a dual-core can handle four streams. Core i5 uses Hyper-Threading to make a dual-core CPU act like a four-core one, but if you have a Core i5 processor with four true cores, it won't have Hyper-Threading. For the time being, Core i5 tops out at handling four streams, using four real cores or two cores with Hyper-Threading."[1]


As I say, Hyperthreading definitely "works" on my i5 and did with 12.04 as well as 14.04. Whether it helps or hinders I have no idea! I do know that the current Hyperthreading technology has nothing, or very little, to do with the Hyperthrading of the old P4 CPUs and most of the comments I read on it when the processors first came out seem to assume they were the same beast. I also know on Doze systems having it enabled on an i7 gave massive performance boosts with audio software, whereas on the old P4 performance was better with HT disabled (at least initially.)

So can anybody point to any conclusive evidence that i-series processors benefit from having HT disabled on a Linux based DAW? Preferably benchmarks on a system installed with HT Enabled and Disabled using a recent kernel and system.


There are also other BIOS settings I would like some recommendation on how to set. Also how much difference does it make it functions are turned on in BIOS and then Disabled later? I would imaging the other way around would cause more difficulties (as maybe the relevant parts of the kernel wouldn't be installed?) but have definitely read recommendations to make sure to set up BIOS first.

The items in question are:
Intel SpeedStep
CPU Management
Intel Hyper-Threading (mentioned above.)


Also: Are there any good resources on setting up an Arch DAW system? I have been reading as much as possible on the Arch Wiki as I can while we have power and internet here (really not much just lately! About 2 hours the whole of today!!)

Some offline documentation would be very useful, so I can read when the internet is down!
http://archaudio.org seems to be dead in the water! Was it superseded?
Plus these couple of articles I found but admit yet to read as been concentrating on the general Wiki.
http://www.linux.com/learn/tutorials/607117-build-a-serious-multimedia-production-workstation-with-arch-linux
https://wiki.archlinux.org/index.php/Pro_Audio
Anything else you can point me to I would be very thankful. I believe Arch is the next step I wan to take. :)
Also starting to feel a little disenfranchised with XFCE. What are you guys running your Arch DAW with?


Thanks for any pointers. :)


[1] http://www.pcmag.com/article2/0,2817,2404675,00.asp


 		 	   		   		 	   		   		 	   		  
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From gheskett at wdtv.com  Fri Aug 29 00:37:53 2014
From: gheskett at wdtv.com (Gene Heskett)
Date: Thu, 28 Aug 2014 20:37:53 -0400
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
References: <20140827143830.GA7320@aol.com>
 <53FF4A76.8000100@stackingdwarves.net>
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
Message-ID: <201408282037.53956.gheskett@wdtv.com>

On Thursday 28 August 2014 11:48:13 Paul Davis did opine
And Gene did reply:
> On Thu, Aug 28, 2014 at 11:27 AM, J??rn Nettingsmeier <
> 
> nettings at stackingdwarves.net> wrote:
> > no personal experience, but i guess the ff400/800 support is fairly
> > complete by now. check with jonathan woithe on the ffado list to make
> > sure.
> 
> just in time for firewire to ride off into the sunset ...

Sorry Paul, but while Apple has took aim and shot it down in flames over a 
decade ago because they wanted a royalty on every socket made, its not 
dead yet, and we will not have services over it until such time as we have 
the same performance from USB.  And we are one heck of a long row of apple 
trees from getting that kind of performance from USB.

When I can plug in a usb movie camera, download it in real time, edit out 
my camera shakies, and re-record it back to the digital Hi-8 tape it came 
from, again in real time, I might change my mind.  USB is nice, you can 
hang stuff on the buss and keep adding hubs until someone brings back a 
sample of the moon and it really is green cheese.  But unlike firewire, I 
have yet to see it Just Work(TM), for realtime video WITH 100% 
synchronized audio, like firewire does.

Cheers, Gene Heskett
-- 
"There are four boxes to be used in defense of liberty:
 soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
Genes Web page <http://geneslinuxbox.net:6309/gene>
US V Castleman, SCOTUS, Mar 2014 is grounds for Impeaching SCOTUS

From dj_kaza at hotmail.com  Fri Aug 29 01:14:48 2014
From: dj_kaza at hotmail.com (Kaza Kore)
Date: Fri, 29 Aug 2014 01:14:48 +0000
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <201408282037.53956.gheskett@wdtv.com>
References: <20140827143830.GA7320@aol.com>,
 <53FF4A76.8000100@stackingdwarves.net>,
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>,
 <201408282037.53956.gheskett@wdtv.com>
Message-ID: <DUB113-W1107FDBFBE7CDE95335DD6F8FDB0@phx.gbl>



> From: gheskett at wdtv.com
> To: linux-audio-user at lists.linuxaudio.org
> Date: Thu, 28 Aug 2014 20:37:53 -0400
> Subject: Re: [LAU] Successor/replacement for RME HDSP+Multiface?
> 
>...Hi-8 tape...
>


I thought we were talking about the future here! The 80s wants its property back!!

Also Hi8 is an analogue format so everything in the post is plain bollocks! Maybe you meant Digital8?? Still 15 years old and any tape format is pretty much dead and definitely not the future!

USB2 really doesn't perform that less well than most Firewire, especially with the low quality chipsets this protocol often seems to go through (whether the actual Firewire chipset, or the PCI(e) chipset that an adaptor hangs off. Biggest problem is finding a port which isn't shared with other devices, and many internal devices often share USB Hubs/Ports so it doesn't only depend on what you connect yourself.

If USB is not good enough for you (higher requirement for number of duplex channels for example) then go the route of PCI(e)/ExpressCard, or wait and see how USB3 develops over the coming months/years, although so far it seems pretty slow.
 		 	   		  
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From vogel at ct.metrocast.net  Fri Aug 29 02:05:39 2014
From: vogel at ct.metrocast.net (Robert Vogel)
Date: Thu, 28 Aug 2014 22:05:39 -0400
Subject: [LAU] GMorgan0.70 tar file uploaded to Sourceforge.
Message-ID: <53FFDFF3.1030408@ct.metrocast.net>

Download the Gmorgan 0.70 tar file from Sourceforge.com. Don't use svn, 
it is not updated.

Gmorgan is a midi processor. It can be voiced using Linux synths, midi 
connected equipment, or using one or more of the many soundcards 
available. I like to use a velocity sensing keyboard so that voices can 
be mixed or layered. It recognizes chords being played on the keyboard, 
and on the background terminal prints the input notes in both numeric 
and alpha form. Other features include a rhythm arranger, a style based 
sequencer, midi recorder. See the Youtube video links from the 
sourceforge home page.

GMorgan .70, the latest version for GNU/Linux, incorporates a new drum 
pattern panel based on the FLTK spreadsheet example, and
includes some miscellaneous cleanup.

Runs well on Ubuntu 12.4 with proprietary NVidia drivers.

If you have tried it before, be sure to go to Settings->global, renew 
the file paths to the current .70 gmorgan directory, and save them. The 
sound file has been renamed sounds.gmox and slightly reformated, so 
results could be unpredictable if you don't at least rest this file.

Let me know how it works for you.

Bob


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From gheskett at wdtv.com  Fri Aug 29 02:45:41 2014
From: gheskett at wdtv.com (Gene Heskett)
Date: Thu, 28 Aug 2014 22:45:41 -0400
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <DUB113-W1107FDBFBE7CDE95335DD6F8FDB0@phx.gbl>
References: <20140827143830.GA7320@aol.com>
 <201408282037.53956.gheskett@wdtv.com>
 <DUB113-W1107FDBFBE7CDE95335DD6F8FDB0@phx.gbl>
Message-ID: <201408282245.42030.gheskett@wdtv.com>

On Thursday 28 August 2014 21:14:48 Kaza Kore did opine
And Gene did reply:
> > From: gheskett at wdtv.com
> > To: linux-audio-user at lists.linuxaudio.org
> > Date: Thu, 28 Aug 2014 20:37:53 -0400
> > Subject: Re: [LAU] Successor/replacement for RME HDSP+Multiface?
> >
> >...Hi-8 tape...
> 
> I thought we were talking about the future here! The 80s wants its
> property back!!
> 
> Also Hi8 is an analogue format so everything in the post is plain
> bollocks! Maybe you meant Digital8?? Still 15 years old and any tape
> format is pretty much dead and definitely not the future!

Not this one, it uses metal tape in the same casette as a Hi-8 would use, 
but about a tenner more expensive. and is "digital Hi-8" format.

Reasonably sharp too at 720p.  Go look it up, its a Sony HandyCam DCR-
TRV460 NTSC. and about 11 years old IIRC.  And one of the first with 
lithium batteries.  I can't quickly find the charger, but after laying for 
at least 2 years, it still fires right up.

I have shot several weddings with it, processed it down to fit on a dvd 
using kino and sold the disks several times now.  Many many times sharper 
than a vhs deck.


> USB2 really doesn't perform that less well than most Firewire,
> especially with the low quality chipsets this protocol often seems to
> go through (whether the actual Firewire chipset, or the PCI(e) chipset
> that an adaptor hangs off. Biggest problem is finding a port which
> isn't shared with other devices, and many internal devices often share
> USB Hubs/Ports so it doesn't only depend on what you connect yourself.
> 
> If USB is not good enough for you (higher requirement for number of
> duplex channels for example) then go the route of PCI(e)/ExpressCard,
> or wait and see how USB3 develops over the coming months/years,
> although so far it seems pretty slow.


Cheers, Gene Heskett
-- 
"There are four boxes to be used in defense of liberty:
 soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
Genes Web page <http://geneslinuxbox.net:6309/gene>
US V Castleman, SCOTUS, Mar 2014 is grounds for Impeaching SCOTUS

From sam at vis.nu  Fri Aug 29 03:01:10 2014
From: sam at vis.nu (Sam Mulvey)
Date: Thu, 28 Aug 2014 20:01:10 -0700
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <201408282245.42030.gheskett@wdtv.com>
References: <20140827143830.GA7320@aol.com>
 <201408282037.53956.gheskett@wdtv.com>
 <DUB113-W1107FDBFBE7CDE95335DD6F8FDB0@phx.gbl>
 <201408282245.42030.gheskett@wdtv.com>
Message-ID: <53FFECF6.8060004@vis.nu>

On 08/28/2014 07:45 PM, Gene Heskett wrote:
>> > I thought we were talking about the future here! The 80s wants its
>> > property back!!
>> > 
>> > Also Hi8 is an analogue format so everything in the post is plain
>> > bollocks! Maybe you meant Digital8?? Still 15 years old and any tape
>> > format is pretty much dead and definitely not the future!
> Not this one, it uses metal tape in the same casette as a Hi-8 would use, 
> but about a tenner more expensive. and is "digital Hi-8" format.
>
> Reasonably sharp too at 720p.  Go look it up, its a Sony HandyCam DCR-
> TRV460 NTSC. and about 11 years old IIRC.  And one of the first with 
> lithium batteries.  I can't quickly find the charger, but after laying for 
> at least 2 years, it still fires right up.
>
> I have shot several weddings with it, processed it down to fit on a dvd 
> using kino and sold the disks several times now.  Many many times sharper 
> than a vhs deck.

..and I think there's a notable point here in the ability to read and
write to a device on the chain in realtime is a question of bandwidth
regardless of what the media is.   

-Sam

From brouits at free.fr  Fri Aug 29 03:21:52 2014
From: brouits at free.fr (=?windows-1252?Q?Beno=EEt_Rouits?=)
Date: Fri, 29 Aug 2014 05:21:52 +0200
Subject: [LAU] band organs
In-Reply-To: <53FEA590.60306@holgerdanske.com>
References: <53FEA590.60306@holgerdanske.com>
Message-ID: <53FFF1D0.9010606@free.fr>

Le 28/08/2014 05:44, David Christensen a ?crit :
> linux-audio-user:
>
> I'm a Linux user would like to play (sequence) band organ MIDI files
> using soundfont synthesis.  I have set up a Core Duo laptop with Debian
> Wheezy, recompiled the kernel with the realtime patch, and installed
> various music software packages (RoseGarden, FluidSynth, etc.).
>
>
> STFW I am able to find free/ open source software (FOSS) MIDI files of
> band organ music rolls, but I am unable to find any FOSS band organ
> soundfont files.
>
>
> I am also unable to find technical information on band organs, notably
> music roll encoding/ decoding -- e.g. which organ functions and
> instruments/ pitches are on which music roll channels -- and how this is
> dealt with when the rolls are converted to MIDI and played using
> something other than the make and model band organ intended by the
> arranger.
>
>
> Any suggestions?
>
>
> TIA,
>
> David

With keyword 'barrel organ' or 'french street organ' in google,
i found a link on a soundfont (sf2), source:
http://forum.renoise.com/index.php/topic/21134-barrel-organcalliope-instrument/
HTH,
- Beno?t


From jeb at ponderworthy.com  Fri Aug 29 04:57:00 2014
From: jeb at ponderworthy.com (Jonathan E Brickman)
Date: Fri, 29 Aug 2014 04:57:00 +0000
Subject: [LAU] Status of lisalo/lisaloQt and calfbox?
Message-ID: <5e8f2428ae0d4fc08d83cdebd6fcb69e@Ex13DAG10-N2.dataoncloud.net>

Have just read a bit of Lisalo/LisaloQt, front end for calfbox; what is the status of this project?  Also, are there usage examples for calfbox I can learn from?

--
Jonathan E. Brickman
Ponderworthy Music | jeb at ponderworthy.com<mailto:jeb at ponderworthy.com> | (785)233-9977 | http://ponderworthy.com
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From len at ovenwerks.net  Fri Aug 29 05:40:49 2014
From: len at ovenwerks.net (Len Ovens)
Date: Thu, 28 Aug 2014 22:40:49 -0700 (PDT)
Subject: [LAU] i5 Hyper-Threading, BIOS settings and Arch n00b pointers
In-Reply-To: <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>,
 <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
Message-ID: <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>

On Thu, 28 Aug 2014, Kaza Kore wrote:

> I have seen on this list multiple times lately (possibly by the same person, but
> never corrected) a statement similar to thus:
> "I chose an i5 over an i7 as the i5 doesn't have Hyperthreading whereas the i7
> does."
> 
> I always thought I remembered this as wrong, especially as I believed I was
> running a 2 core i5 in my laptop and it shows up as 4 cores. I have just

That is correct. I had forgotten that there was a model of i5 with two 
cores and that one has hyperthreading to make up for it. As I was buying 
MB and processor separate, I looked through Intel's spec list to make sure 
I got one without hyperthreading. I was hopping not to pay for "Boost 
Technology" as well, but the only higher end cpu that can be had without 
boost is the xeon (some of them anyway).

> "The quick explanation is that all Core i7 CPUs use Hyper-Threading, so a
> six-core CPU can handle 12 streams, a four-core can handle eight streams, and a
> dual-core can handle four streams. Core i5 uses Hyper-Threading to make a
> dual-core CPU act like a four-core one, but if you have a Core i5 processor with
> four true cores, it won't have Hyper-Threading. For the time being, Core i5 tops
> out at handling four streams, using four real cores or two cores with
> Hyper-Threading."[1]

Yes that is correct. Thank you for pointing that out. As always, this list 
is a good start, but one's own research is important too.

> As I say, Hyperthreading definitely "works" on my i5 and did with 12.04 as well
> as 14.04. Whether it helps or hinders I have no idea! I do know that the current
> Hyperthreading technology has nothing, or very little, to do with the
> Hyperthrading of the old P4 CPUs and most of the comments I read on it when the
> processors first came out seem to assume they were the same beast. I also know on
> Doze systems having it enabled on an i7 gave massive performance boosts with
> audio software, whereas on the old P4 performance was better with HT disabled (at
> least initially.)

Performance is an interesting word, hyperthreading has always helped 
performance, even audio performance if you are talking about CPU 
throughput. In jackd terms, even on the single core P4, audio was pretty 
solid down to about -p64, then xruns. With hyperthreading off -p16 with no 
xruns was possible. However, overall performance was less too.

> So can anybody point to any conclusive evidence that i-series processors benefit
> from having HT disabled on a Linux based DAW? Preferably benchmarks on a system
> installed with HT Enabled and Disabled using a recent kernel and system.

That does sound interesting. There is a site that tests lowlatency 
performance of a number of machines with RT kernels. Here is one for an i3 
with hyperthreading:
https://www.osadl.org/Profile-of-system-in-rack-0-slot-0.qa-profile-r0s0.0.html

It looks very good, so your comment about the difference between old 
hyperthreading and new may be right. (the highest latency in the plot is 
34us with cycle test) How these test relate to real world audio use I 
don't know. At the top of the above page there is access to the rest of 
the rack slots with different machines that might be there.

> There are also other BIOS settings I would like some recommendation on how to
> set. Also how much difference does it make it functions are turned on in BIOS and
> then Disabled later? I would imaging the other way around would cause more
> difficulties (as maybe the relevant parts of the kernel wouldn't be installed?)
> but have definitely read recommendations to make sure to set up BIOS first.
> 
> The items in question are:
> Intel SpeedStep

Even with the i5, a few months old, setting the CPU to performance over 
ondemand seems to make a difference. Ondemand pretty much means 10ms for 
no xrun latency. Ondemand changes cpu speed in such a way that the OS is 
aware of the cpu speed and can control that speed. Letting the CPU decide 
on it's own can't be better. This is why I turn boost technology off. -p16 
is about .6ms at 48k sample rate and seems to be very stable. In fact 
setting CPU governor to userspace and fixing cpu frequency to 800 MHz 
seems to be more stable than with on demand. (tested on both the i5 and an atom 
netbook)

> CPU Management

Not sure what you mean by that. SMIs? Any time the OS doesn't know what 
the CPU is doing, RT latency suffers. Intel agrees turning them off 
improves low latency, but they also say that in some cases this voids 
warranty...

> Intel Hyper-Threading (mentioned above.)

I don't know that the physical technology matters so much as the OS being 
hyperthreading aware and treating each pair of cores like one. That is 
making sure that core 0 does not do anything that takes too long for core 
1 to meet it's dead line. I do not know if new Linux kernels do this, 
older ones did not. They logged that the chip had hyperthreading, but 
still seemed to treat two threads as two different cores.

Certainly, common wisdom has not kept up with tech changes. I would be 
nice to know more.

--
Len Ovens
www.ovenwerks.net


From gnome at hawaii.rr.com  Fri Aug 29 06:56:55 2014
From: gnome at hawaii.rr.com (david)
Date: Thu, 28 Aug 2014 20:56:55 -1000
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <20140828200726.GA16743@aol.com>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>
 <20140827192248.GC7320@aol.com> <53FF4A76.8000100@stackingdwarves.net>
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
 <53FF52B0.4030702@stackingdwarves.net> <20140828200726.GA16743@aol.com>
Message-ID: <54002437.6080607@hawaii.rr.com>

On 08/28/2014 10:07 AM, Peter P. wrote:
> * J?rn Nettingsmeier <nettings at stackingdwarves.net> [2014-08-28 12:03]:
>> On 08/28/2014 05:48 PM, Paul Davis wrote:
>>>
>>> On Thu, Aug 28, 2014 at 11:27 AM, J?rn Nettingsmeier
>>> <nettings at stackingdwarves.net <mailto:nettings at stackingdwarves.net>> wrote:
>>>
>>>     no personal experience, but i guess the ff400/800 support is fairly
>>>     complete by now. check with jonathan woithe on the ffado list to
>>>     make sure.
>>>
>>> just in time for firewire to ride off into the sunset ...
> Yes, nice comment Paul. I feel I have to keep chasing laptops that
> have either an express card slot or a firewire connector.

Does anyone make a laptop with either of those anymore? Didn't find one 
on Amazon.

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From dj_kaza at hotmail.com  Fri Aug 29 07:21:27 2014
From: dj_kaza at hotmail.com (Kaza Kore)
Date: Fri, 29 Aug 2014 07:21:27 +0000
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <201408282245.42030.gheskett@wdtv.com>
References: <20140827143830.GA7320@aol.com>,
 <201408282037.53956.gheskett@wdtv.com>,
 <DUB113-W1107FDBFBE7CDE95335DD6F8FDB0@phx.gbl>,
 <201408282245.42030.gheskett@wdtv.com>
Message-ID: <DUB113-W3024DDF80474D625B42FD48FDB0@phx.gbl>



> From: gheskett at wdtv.com
> To: linux-audio-user at lists.linuxaudio.org
> Date: Thu, 28 Aug 2014 22:45:41 -0400
> Subject: Re: [LAU] Successor/replacement for RME HDSP+Multiface?
> 
> On Thursday 28 August 2014 21:14:48 Kaza Kore did opine
> And Gene did reply:
> > > From: gheskett at wdtv.com
> > > To: linux-audio-user at lists.linuxaudio.org
> > > Date: Thu, 28 Aug 2014 20:37:53 -0400
> > > Subject: Re: [LAU] Successor/replacement for RME HDSP+Multiface?
> > >
> > >...Hi-8 tape...
> > 
> > I thought we were talking about the future here! The 80s wants its
> > property back!!
> > 
> > Also Hi8 is an analogue format so everything in the post is plain
> > bollocks! Maybe you meant Digital8?? Still 15 years old and any tape
> > format is pretty much dead and definitely not the future!
> 
> Not this one, it uses metal tape in the same casette as a Hi-8 would use, 
> but about a tenner more expensive. and is "digital Hi-8" format.
> 
> Reasonably sharp too at 720p.  Go look it up, its a Sony HandyCam DCR-
> TRV460 NTSC. and about 11 years old IIRC.

So not Hi8 then! :p (If you look I did mention Digital8 too.) Not sure where you get the idea it's 720P capable! Specs on website state 640x480 and you even state in the name you provided it's NTSC, which is never 720P, same as PAL and SECAM aren't. They are old, SD standards. 720/1080 P/I are very different beasts really.

Anyway it's probably more important to talk about the standardised DV25 and DV50 protocol all these commercial/prosumer products use for communication that tape/card formats. There are some Sony and Panasonic camera that do this fine over USB so it's not impossible or a problem with USB itself. I see yours (and apparently many others) claim to have some kind of USB Streaming but for some reason it's not usually full quality, as you would get from Firewire. Wonder why...


 		 	   		  
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From harryhaaren at gmail.com  Fri Aug 29 07:59:28 2014
From: harryhaaren at gmail.com (Harry van Haaren)
Date: Fri, 29 Aug 2014 08:59:28 +0100
Subject: [LAU] Session management with NSM
In-Reply-To: <20140828224247.236111e1@eeyore>
References: <20140828224247.236111e1@eeyore>
Message-ID: <CAKudYbP4LiQDs4kWgU3a_YKCM9HVjmS3qO+hngNOky91r0XOUQ@mail.gmail.com>

On Thu, Aug 28, 2014 at 9:42 PM, Philipp ?berbacher <murks at tuxfamily.org> wrote:
> I'm just getting my toes wet with NSM and have a couple of questions.
Cool, lets see :)

> 1. NSM configuration. Is there a configuration file? I haven't seen
> anything in the manual. The default session root ($HOME/NSM Sessions) is
> quite horrible IMHO and I would have liked to configure that. Instead I
> created an alias to set the root (alias nsm='non-session-manager --
> --session-root ~/.nsm-sessions').

That's the correct way to handle this, as far as I know. Its useful to
have different directories on one system: it allows subdiving your
available sessions into groups like "albums" or
"projects-with-certain-people". Although I agree it feels a little
clunky, its quite powerful and useful.

> 2. Adding programs to sessions through the GUI ("Add Client to
> Session") is the only way? Is there no way to attach running clients
> or at least have some comfort like tab completion to add clients?
NSM does not support this "attach" workflow, but tab completion or a
list of available (fully supported) NSM clients would be a good
improvement on workflow. This should be discussed as to how best
implement it: i'm not sure.

> 3. Jack and NSM. How do you handle that? It is possible to start jack
> through NSM proxy and I guess it is OK to do that as long as jack
> reliably starts before jackpatch (something I'm not sure of). First I
> had just jackpatch in there and it started jack for me with a whole lot
> of options that are unfamiliar to me and probably not needed.

I imagine that NSM will launch said JACK apps, and if one is set to
"start JACK" on jack_client_open() in its code, then it will start
JACK with the settings in  ~/.jackdrc   Perhaps the inclusion of a
"Start JACK" type client with particular settings can be implemented
in order to handle this? I'm open for suggestions too.

> 4. CLI clients. Are they generally not supported? I added the lv2 host
> that was recommended to me (jalv) and had to do that through the NSM
> proxy, so the settings won't be saved even though the plugin (fabla in
> this case) can save its settings. This sort of defeats session
> management. With all the CLI tools we have it would be a pitty if that
> was generally not supported. On a sidenote, can someone recommend a
> plugin host that is supported?

CLI clients are supported just like clients with a GUI, there is no
difference to NSM. The issue you're encountering here is that JALV
currently doesn't support NSM, which is something that I agree needs
fixing. I'll put JALV NSM support on the TODO, its something I've
lacked myself too.

> Well, that's it for now. Last time I heard about NSM I got the
> impression that it takes care of session management once and for all,
> but the first half our gave me a different impression.
OpenAV stands behind NSM: I'm willing to do my best to cooperate with
project developers to implement NSM in various programs, and improve
the workflow of session management.

If there's any furthur questions, please ask, in the mean time, I'll
try code up some NSM :) -Harry

From simonzwise at gmail.com  Fri Aug 29 08:33:43 2014
From: simonzwise at gmail.com (Simon Wise)
Date: Fri, 29 Aug 2014 18:33:43 +1000
Subject: [LAU] i5 Hyper-Threading, BIOS settings and Arch n00b pointers
In-Reply-To: <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>,
 <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
Message-ID: <54003AE7.1010304@gmail.com>

On 29/08/14 15:40, Len Ovens wrote:

>
> I don't know that the physical technology matters so much as the OS being
> hyperthreading aware and treating each pair of cores like one. That is making
> sure that core 0 does not do anything that takes too long for core 1 to meet
> it's dead line. I do not know if new Linux kernels do this, older ones did not.
> They logged that the chip had hyperthreading, but still seemed to treat two
> threads as two different cores.
>
> Certainly, common wisdom has not kept up with tech changes. I would be nice to
> know more.
>

Not quite on topic, since this isn't to do with Hyper-threading, but certainly 
the Linux scheduler has been getting much more sophisticated in dealing with 
different kinds of cores ... in ARM it now schedules tasks for chips with some 
smaller cores and some faster ones, keeping them busy with suitable sized tasks.

The ARM kernels running the most recent Samsung tablets (with 4 big plus 4 
little cores) have this GTS in the 3.14 kernels ... it runs all 8 cores together 
assigning tasks appropriate to each, rather than just switching between big or 
little of each pair to save power. Selling hardware on that scale certainly 
brings a budget, and since the kernel is GPL it can't be kept in-house.

Seems that 3.14 has also added a deadline-based scheduler that is closer to what 
audio needs from realtime than the extremely low latency preemption based on 
priorities that the two older realtime schedulers offer.

http://www.linuxfoundation.org/news-media/blogs/browse/2014/01/deadline-scheduling-314

Simon

From dj_kaza at hotmail.com  Fri Aug 29 09:33:40 2014
From: dj_kaza at hotmail.com (Kaza Kore)
Date: Fri, 29 Aug 2014 09:33:40 +0000
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <54002437.6080607@hawaii.rr.com>
References: <20140827143830.GA7320@aol.com>,
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>,
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>,
 <20140827154520.GB7320@aol.com>,
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>,
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>,
 <20140827192248.GC7320@aol.com>
 <53FF4A76.8000100@stackingdwarves.net>,
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>,
 <53FF52B0.4030702@stackingdwarves.net>
 <20140828200726.GA16743@aol.com>,<54002437.6080607@hawaii.rr.com>
Message-ID: <DUB113-W18E942790CECC11578A7BE8FDB0@phx.gbl>



> Date: Thu, 28 Aug 2014 20:56:55 -1000
> From: gnome at hawaii.rr.com
> To: linux-audio-user at lists.linuxaudio.org
> Subject: Re: [LAU] Successor/replacement for RME HDSP+Multiface?
> 
> On 08/28/2014 10:07 AM, Peter P. wrote:
> > * J?rn Nettingsmeier <nettings at stackingdwarves.net> [2014-08-28 12:03]:
> >> On 08/28/2014 05:48 PM, Paul Davis wrote:
> >>>
> >>> On Thu, Aug 28, 2014 at 11:27 AM, J?rn Nettingsmeier
> >>> <nettings at stackingdwarves.net <mailto:nettings at stackingdwarves.net>> wrote:
> >>>
> >>>     no personal experience, but i guess the ff400/800 support is fairly
> >>>     complete by now. check with jonathan woithe on the ffado list to
> >>>     make sure.
> >>>
> >>> just in time for firewire to ride off into the sunset ...
> > Yes, nice comment Paul. I feel I have to keep chasing laptops that
> > have either an express card slot or a firewire connector.
> 
> Does anyone make a laptop with either of those anymore? Didn't find one 
> on Amazon.
> 

LOADS of laptops have ExpressCard slots! PCMCIA on the other hand you'd be lucky to find these days. Don't think they were ever much more popular for audio interfaces than the newer ExpressCard though anyway, so unless it's about using hardware you already own there is little point looking for it.
 		 	   		  
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From fons at linuxaudio.org  Fri Aug 29 09:47:25 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Fri, 29 Aug 2014 09:47:25 +0000
Subject: [LAU] i5 Hyper-Threading, BIOS settings and Arch n00b pointers
In-Reply-To: <54003AE7.1010304@gmail.com>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>
 <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
 <54003AE7.1010304@gmail.com>
Message-ID: <20140829094725.GA32083@linuxaudio.org>

On Fri, Aug 29, 2014 at 06:33:43PM +1000, Simon Wise wrote:

> Seems that 3.14 has also added a deadline-based scheduler that is
> closer to what audio needs from realtime than the extremely low
> latency preemption based on priorities that the two older realtime
> schedulers offer.
> 
> http://www.linuxfoundation.org/news-media/blogs/browse/2014/01/deadline-scheduling-314

Not really.

I went to a workshop on this some weeks ago, with presentations
by the author(s) of this deadline scheduler. 

I presented a short intro to Jack, what it does and how and why,
with the aim to enable a discussion on if the deadline scheduler
would be a good solution for running a dynamic set of audio apps.
The conclusion was that (apart from some specific uses in e.g.
drivers) it was not designed to solve this type of problem, and
it would quite difficult to use it in such a context. 

The new scheduler is designed to run a set of periodic tasks,
with arbitrary periods and having few or no dependencies between
them, at up to 100% CPU usage, while meeting their deadlines.

Rate-proportional scheduling (shorter period -> higher priority)
can only do this up to 69% CPU use, except in some special cases
such as all periods being equal or having simple integer ratios
between them.

The situation with a set of audio apps running under Jack (or
something similar) is very different: the order in which they
have to run depends on their connections. 

Another problem with the deadline scheduler as implemented is
that you can not overcommit CPU use. If P_i is the period of
task 'i' and T_i its running time, then Sum_i { T_i / P_i }
must be smaller than the number of CPUs available to the
scheduler. This is enforced by the API, where T_i,  P_i and
D_i (the relative deadline) are parameters. If a task ever
tries to take more than its T_i, it will be interrupted and
have to wait until its CPU budget is updated, which will be
after a time P_i. 

In fact, having tasks pre-empted like this is a very normal
way to use the scheduler. The idea is that each task should
get a defined share T_i of time in each period P_i, and meet
its deadline which is relative to the instant it becomes
runnable. 

If on the other hand a task has to finish a defined amount
of work in each period (e.g process a period of audio because
others depend on the output), then its configured T_i must be
the absolute worst case time it could ever take to do that. 
So to make this work with audio processes you need to *know*
their running time, and if these times are variable you must
use the worst case one. Which means that CPU use will be well
below 100%.

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From murks at tuxfamily.org  Fri Aug 29 11:02:33 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Fri, 29 Aug 2014 13:02:33 +0200
Subject: [LAU] Update Phasex:  Sequencer suggestions?
In-Reply-To: <1409261340.4790.11.camel@rocketmail.com>
References: <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
 <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
 <20140826160630.GC16859@linuxaudio.org>
 <1409069791.7093.33.camel@rocketmail.com>
 <20140826162353.GD16859@linuxaudio.org>
 <1409260025.4790.8.camel@rocketmail.com>
 <1409261340.4790.11.camel@rocketmail.com>
Message-ID: <20140829130233.38095a15@eeyore.mozart.uni-klu.ac.at>

On Thu, 28 Aug 2014 23:29:00 +0200
Ralf Mardorf <ralf.mardorf at rocketmail.com> wrote:

> On Thu, 2014-08-28 at 23:07 +0200, Ralf Mardorf wrote:
> > 1.
> >         On Tue, 2014-08-26 at 16:23 +0000, Fons Adriaensen wrote:
> > >         I'm not the maintainer of the AUR package :-) Whoever
> > > that is may be lurking here and pick up the fix.
> >         
> >         https://aur.archlinux.org/packages/ph/phasex-git/PKGBUILD
> >         https://aur.archlinux.org/packages/ph/phasex/PKGBUILD
> > 
> > 2.
> >         https://github.com/williamweston/phasex/issues/10
> 
> 3.
>         Reported: https://aur.archlinux.org/packages.php?K=phasex

The one is orphaned, the other 'maintained' by speps, so nothing will
happen unless one of us adopts and patches and create and patch a
duplicate for the other one. I'm too stupid to patch apparently,
otherwise I would have done it already.

Regards,
Philipp

From dj_kaza at hotmail.com  Fri Aug 29 12:54:22 2014
From: dj_kaza at hotmail.com (dale)
Date: Fri, 29 Aug 2014 18:39:22 +0545
Subject: [LAU] i5 Hyper-Threading, BIOS settings and Arch n00b pointers
In-Reply-To: <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>
 , <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
Message-ID: <BLU436-SMTP1261D057A799C5A4F5BB0D48FDB0@phx.gbl>

On Thu, 2014-08-28 at 22:40 -0700, Len Ovens wrote:

> Performance is an interesting word, hyperthreading has always helped 
> performance, even audio performance if you are talking about CPU 
> throughput. In jackd terms, even on the single core P4, audio was pretty 
> solid down to about -p64, then xruns. With hyperthreading off -p16 with no 
> xruns was possible. However, overall performance was less too.
> 
> > So can anybody point to any conclusive evidence that i-series processors benefit
> > from having HT disabled on a Linux based DAW? Preferably benchmarks on a system
> > installed with HT Enabled and Disabled using a recent kernel and system.
> 
> That does sound interesting. There is a site that tests lowlatency 
> performance of a number of machines with RT kernels. Here is one for an i3 
> with hyperthreading:
> https://www.osadl.org/Profile-of-system-in-rack-0-slot-0.qa-profile-r0s0.0.html
> 
> It looks very good, so your comment about the difference between old 
> hyperthreading and new may be right. (the highest latency in the plot is 
> 34us with cycle test) How these test relate to real world audio use I 
> don't know. At the top of the above page there is access to the rest of 
> the rack slots with different machines that might be there.

Have done some further reading, and although I can't say for definite it
sounds like support for modern hyperthreading mode is quite well
supported in Linux now and for general/desktop use. Seen one report that
it specifically caused problems with (some?) rt-kernels, but not with
generic. No mention of pre-empt. Another reporting that it increases
power but generally at the cost of latency.

I would be interested to know if it would allow you to play something in
your DAW that uses greater CPU power but at the cost of low loads
requiring a higher buffer setting within Jack. It does seem these days
it may be down to the question of whether minimum latency or maximum
available power is of most importance to you. (Although without some
real checks I'm only just postulating. If somebody wants to provide me
with a project file that will play on a default install of KXStudio
installed with it turned on and off.

Although that still begs the question of whether it being turned off in
BIOS/UFEI before or after installation actually makes any difference...

> 
> > CPU Management
> 
> Not sure what you mean by that. SMIs? Any time the OS doesn't know what 
> the CPU is doing, RT latency suffers. Intel agrees turning them off 
> improves low latency, but they also say that in some cases this voids 
> warranty...

I now think this is CPU Power Management and turns off some levels such
as Sleep or Hibernate but I'm not positive...

Thanks, Dale.


From dj_kaza at hotmail.com  Fri Aug 29 13:20:08 2014
From: dj_kaza at hotmail.com (dale)
Date: Fri, 29 Aug 2014 19:05:08 +0545
Subject: [LAU] NOW - UEFI
In-Reply-To: <BLU436-SMTP1261D057A799C5A4F5BB0D48FDB0@phx.gbl>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>
 , <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
 <BLU436-SMTP1261D057A799C5A4F5BB0D48FDB0@phx.gbl>
Message-ID: <BLU437-SMTP19EF8B72E796EA0DC27E558FDB0@phx.gbl>

On Fri, 2014-08-29 at 18:39 +0545, dale wrote:

> Although that still begs the question of whether it being turned off in
> BIOS/UFEI before or after installation actually makes any difference...
> 

Now this is something that has been harder for me to get my head around
than benefit/disadvantages to HT!

When I got this laptop it had Window7 pre-installed with a bunch of
Lenovo software on it. At the time I know nothing of changes from BIOS
to UEFI! One of the first things I did was install Ubuntu Studio 12.04
on it, using Legacy mode as I wasn't sure if it was UEFI compatible and
my laptop seems to allow both.

This has caused various headaches I have kind learned to live with. Main
one being relating to screen brightness. If I set a low default screen
brightness in Windows then it seems to change the range of brightness
Linux can display. This was even more noticable when running LiveUSB
distros! I actually couldn't get the screen bright enough to see what I
was doing without setting the level within Windows bright again!

Then I read (or skimmed) this yesterday and feel even more confused.
https://wiki.archlinux.org/index.php/UEFI

But I am starting to think I should try enabling it with a Linux
install. Especially as I plan to completely banish Windows from the
computer now... But I thought I would ask what more experienced Linux
users have to say about UEFI mode vs BIOS/Legacy mode.

Again thanks for all the help. Dale.


From len at ovenwerks.net  Fri Aug 29 14:24:00 2014
From: len at ovenwerks.net (Len Ovens)
Date: Fri, 29 Aug 2014 07:24:00 -0700 (PDT)
Subject: [LAU] Status of lisalo/lisaloQt and calfbox?
In-Reply-To: <5e8f2428ae0d4fc08d83cdebd6fcb69e@Ex13DAG10-N2.dataoncloud.net>
References: <5e8f2428ae0d4fc08d83cdebd6fcb69e@Ex13DAG10-N2.dataoncloud.net>
Message-ID: <alpine.DEB.2.10.1408290708270.2499@scott.cbbs.org>

On Fri, 29 Aug 2014, Jonathan E Brickman wrote:

> Have just read a bit of Lisalo/LisaloQt, front end for calfbox; what is the
> status of this project?? Also, are there usage examples for calfbox I can learn
> from?

Please post some urls  :)  I am thinking Dr Lisa Lo and lisalot and "calf 
box" (milk means more) are not what this is about even if google does.

In truth, I did find calfbox on github but the README file gave me no clue 
what this does. And the comment by a distro packager that as it was cli he 
didn't know either even though he includes it was not enlightening either. 
I am thinking something to do with playing samples. I found a comment in 
Debian multimedia that says: "calfbox (free alternative for linuxsampler 
SFZ sampler, http://repo.or.cz/w/calfbox.git)" Which sounds very 
interesting.


--
Len Ovens
www.ovenwerks.net

From simonzwise at gmail.com  Fri Aug 29 14:28:24 2014
From: simonzwise at gmail.com (Simon Wise)
Date: Sat, 30 Aug 2014 00:28:24 +1000
Subject: [LAU] i5 Hyper-Threading, BIOS settings and Arch n00b pointers
In-Reply-To: <20140829094725.GA32083@linuxaudio.org>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>
 <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
 <54003AE7.1010304@gmail.com> <20140829094725.GA32083@linuxaudio.org>
Message-ID: <54008E08.4070003@gmail.com>

On 29/08/14 19:47, Fons Adriaensen wrote:
> On Fri, Aug 29, 2014 at 06:33:43PM +1000, Simon Wise wrote:
>
>> Seems that 3.14 has also added a deadline-based scheduler that is
>> closer to what audio needs from realtime than the extremely low
>> latency preemption based on priorities that the two older realtime
>> schedulers offer.
>>
>> http://www.linuxfoundation.org/news-media/blogs/browse/2014/01/deadline-scheduling-314
>
> Not really.
>
> I went to a workshop on this some weeks ago, with presentations
> by the author(s) of this deadline scheduler.
>
> I presented a short intro to Jack, what it does and how and why,
> with the aim to enable a discussion on if the deadline scheduler
> would be a good solution for running a dynamic set of audio apps.
> The conclusion was that (apart from some specific uses in e.g.
> drivers) it was not designed to solve this type of problem, and
> it would quite difficult to use it in such a context.
>
> The new scheduler is designed to run a set of periodic tasks,
> with arbitrary periods and having few or no dependencies between
> them, at up to 100% CPU usage, while meeting their deadlines.

that's interesting, it seemed promising at first ... so this is for being 
certain that particular and predictable sized tasks will get scheduled in time 
for their own individual deadlines, and I guess finding out immediately that 
something can't be guaranteed ... while most audio tasks need to be done as soon 
as possible since they are of less predictable duration, are inter-dependant and 
all work together towards essentially the same deadline ... so the challenge in 
multi core machines with a set of audio tasks taking most of the available 
processing is scheduling the tasks with the most depending on them earliest, 
hence putting tasks on queues with various priorities makes a lot more sense.

>
> Rate-proportional scheduling (shorter period ->  higher priority)
> can only do this up to 69% CPU use, except in some special cases
> such as all periods being equal or having simple integer ratios
> between them.
>
> The situation with a set of audio apps running under Jack (or
> something similar) is very different: the order in which they
> have to run depends on their connections.
>
> Another problem with the deadline scheduler as implemented is
> that you can not overcommit CPU use. If P_i is the period of
> task 'i' and T_i its running time, then Sum_i { T_i / P_i }
> must be smaller than the number of CPUs available to the
> scheduler. This is enforced by the API, where T_i,  P_i and
> D_i (the relative deadline) are parameters. If a task ever
> tries to take more than its T_i, it will be interrupted and
> have to wait until its CPU budget is updated, which will be
> after a time P_i.

so it is probably very useful if you need to be certain of tasks completing and 
perhaps change strategies early to avoid missing deadlines, but that isn't what 
audio needs.

>
> In fact, having tasks pre-empted like this is a very normal
> way to use the scheduler. The idea is that each task should
> get a defined share T_i of time in each period P_i, and meet
> its deadline which is relative to the instant it becomes
> runnable.
>
> If on the other hand a task has to finish a defined amount
> of work in each period (e.g process a period of audio because
> others depend on the output), then its configured T_i must be
> the absolute worst case time it could ever take to do that.
> So to make this work with audio processes you need to *know*
> their running time, and if these times are variable you must
> use the worst case one. Which means that CPU use will be well
> below 100%.

thanks for that explanation


Simon

From grib at billgribble.com  Fri Aug 29 14:29:42 2014
From: grib at billgribble.com (Bill Gribble)
Date: Fri, 29 Aug 2014 10:29:42 -0400
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <54002437.6080607@hawaii.rr.com>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>
 <20140827192248.GC7320@aol.com> <53FF4A76.8000100@stackingdwarves.net>
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
 <53FF52B0.4030702@stackingdwarves.net> <20140828200726.GA16743@aol.com>
 <54002437.6080607@hawaii.rr.com>
Message-ID: <9B60CEE4-F83D-4531-A9AD-2A229AD5309B@billgribble.com>

I bought a Dell about a year ago with an ExpressCard slot (E6420).  ISTR many manufacturers make them, tho it's tough sometimes to meet a specific combination of requirements including EC.  

Thanks,
Bill Gribble

> On Aug 29, 2014, at 2:56, david <gnome at hawaii.rr.com> wrote:
> 
>> On 08/28/2014 10:07 AM, Peter P. wrote:
>> * J?rn Nettingsmeier <nettings at stackingdwarves.net> [2014-08-28 12:03]:
>>>> On 08/28/2014 05:48 PM, Paul Davis wrote:
>>>> 
>>>> On Thu, Aug 28, 2014 at 11:27 AM, J?rn Nettingsmeier
>>>> <nettings at stackingdwarves.net <mailto:nettings at stackingdwarves.net>> wrote:
>>>> 
>>>>    no personal experience, but i guess the ff400/800 support is fairly
>>>>    complete by now. check with jonathan woithe on the ffado list to
>>>>    make sure.
>>>> 
>>>> just in time for firewire to ride off into the sunset ...
>> Yes, nice comment Paul. I feel I have to keep chasing laptops that
>> have either an express card slot or a firewire connector.
> 
> Does anyone make a laptop with either of those anymore? Didn't find one on Amazon.
> 
> -- 
> David W. Jones
> gnome at hawaii.rr.com
> authenticity, honesty, community
> http://dancingtreefrog.com
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

From jeb at ponderworthy.com  Fri Aug 29 14:34:31 2014
From: jeb at ponderworthy.com (Jonathan E Brickman)
Date: Fri, 29 Aug 2014 14:34:31 +0000
Subject: [LAU] Status of lisalo/lisaloQt and calfbox?
In-Reply-To: <alpine.DEB.2.10.1408290708270.2499@scott.cbbs.org>
Message-ID: <em12278a3f-93d5-4e5e-bdf2-0bb1afabc317@dcs-jb-8a>

Greetings, Len.  The creator of calfbox suggested I look at this file:

sampler_api_example2.py

which can be seen in the git repo here:

http://repo.or.cz/w/calfbox.git/blob/HEAD:/sampler_api_example2.py

My impression has become, that calfbox is probably a very nice Python 
toolkit by which one can build applets to do a number of musical things, 
including build a live synth using SFZ instruments.  This thought is 
analogous to mididings, which is indeed a very nice Python toolkit by 
which one can build applets to do a number of MIDI conversions, 
combinations, and other renobulations of many sorts.  I have not tried 
calfbox yet, but do use mididings regularly, and am looking forward to 
some careful experimentation; I am trying to build a very dense Strings 
instrument for my synth box ( http://lsn.ponderworthy.com ), and 
although the 14-item LinuxSynth setup I have right now sounds great for 
a second or two of note-hold, one hears increasing spits and pops and 
static after more hold even on one note.

Jonathan E. Brickman
Ponderworthy Music | jeb at ponderworthy.com | (785)233-9977 | 
http://ponderworthy.com <http://ponderworthy.com/>





------ Original Message ------
From: "Len Ovens" <len at ovenwerks.net>
To: "Jonathan E Brickman" <jeb at ponderworthy.com>
Cc: "linux-audio-user at lists.linuxaudio.org" 
<linux-audio-user at lists.linuxaudio.org>
Sent: 8/29/2014 9:24:00 AM
Subject: Re: [LAU] Status of lisalo/lisaloQt and calfbox?

>On Fri, 29 Aug 2014, Jonathan E Brickman wrote:
>
>>  Have just read a bit of Lisalo/LisaloQt, front end for calfbox; what 
>>is the
>>  status of this project?  Also, are there usage examples for calfbox I 
>>can learn
>>  from?
>
>Please post some urls :) I am thinking Dr Lisa Lo and lisalot and "calf
>box" (milk means more) are not what this is about even if google does.
>
>In truth, I did find calfbox on github but the README file gave me no 
>clue
>what this does. And the comment by a distro packager that as it was cli 
>he
>didn't know either even though he includes it was not enlightening 
>either.
>I am thinking something to do with playing samples. I found a comment 
>in
>Debian multimedia that says: "calfbox (free alternative for 
>linuxsampler
>SFZ sampler, http://repo.or.cz/w/calfbox.git)" Which sounds very
>interesting.
>
>
>--
>Len Ovens
>www.ovenwerks.net

From len at ovenwerks.net  Fri Aug 29 15:12:29 2014
From: len at ovenwerks.net (Len Ovens)
Date: Fri, 29 Aug 2014 08:12:29 -0700 (PDT)
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <DUB113-W3024DDF80474D625B42FD48FDB0@phx.gbl>
References: <20140827143830.GA7320@aol.com>,
 <201408282037.53956.gheskett@wdtv.com>,
 <DUB113-W1107FDBFBE7CDE95335DD6F8FDB0@phx.gbl>,
 <201408282245.42030.gheskett@wdtv.com>
 <DUB113-W3024DDF80474D625B42FD48FDB0@phx.gbl>
Message-ID: <alpine.DEB.2.10.1408290728170.2499@scott.cbbs.org>

On Fri, 29 Aug 2014, Kaza Kore wrote:

> Anyway it's probably more important to talk about the standardised DV25 and DV50
> protocol all these commercial/prosumer products use for communication that
> tape/card formats. There are some Sony and Panasonic camera that do this fine
> over USB so it's not impossible or a problem with USB itself. I see yours (and
> apparently many others) claim to have some kind of USB Streaming but for some
> reason it's not usually full quality, as you would get from Firewire. Wonder
> why...

Intel says that nobody really needs that low of a latency anyway in 
response some sound cards problems with one of their implementations of 
USB3. (the problem was USB2 cards with USB3 ports which are supposed to be 
compatable)

In recording latency is not the issue it is with live work. For recording 
latency needs to known and constant and reasonably low for monitoring. I 
have heard/read people who say what does it matter if you move your head a 
foot or two closer or father from the speaker? But in live work if the 
same audio comes from different places it is called a filter.

Looking at some of the computers around and their latency spec, there is 
no reason they should not be able to be used as a live mix engine for FOH 
except the audio IF is too slow. Often this is because the IF includes an 
internal router/mixer/effects section which I find useless anyway.

Looked at from a different POV, the cost of a digital mixer makes it an 
atractive audio interface. For ~$2k you can have:
 	- 22 digital i/o (USB as happens)
 	- 16 mic/line preamps
 	- a 16 motor fader DAW control (can be MCP)
 	- 16 digital channel strips with complete eq+
 	- low enough internal latency for live work
 	- can be used as an on site live digital recorder (18 tracks)
 	- 32 track/channel available (for more money)
( http://www.allen-heath.com/ahproducts/qu-16/ )

That looks very attractive for a small working studio. For specialty mics 
I would still use a dedicated pre if I could. The presence of eq and other 
effects in this box should not preclude the use of other plugins in the 
DAW as these are live related (make an sm58 sound good) and there may be 
better plugins available in some cases.

I am sure there are other digital live mixers around with similar 
features.

--
Len Ovens
www.ovenwerks.net


From jhernberg at alchemy.lu  Fri Aug 29 15:37:09 2014
From: jhernberg at alchemy.lu (Joakim Hernberg)
Date: Fri, 29 Aug 2014 17:37:09 +0200
Subject: [LAU] i5 Hyper-Threading, BIOS settings and Arch n00b pointers
In-Reply-To: <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>
 <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
Message-ID: <20140829173709.44b02ce8@alchemy.lu>

On Thu, 28 Aug 2014 23:40:01 +0000
Kaza Kore <dj_kaza at hotmail.com> wrote:

> So can anybody point to any conclusive evidence that i-series
> processors benefit from having HT disabled on a Linux based DAW?
> Preferably benchmarks on a system installed with HT Enabled and
> Disabled using a recent kernel and system.

Afaik there are 2 points to be made.

1. SMT (at least as implemented on iX cpus) break the entire concept
of realtime.  You do get an estimated increase in CPU throughput of
about 25%, at the cost of stealing CPU time from your SCHED_FIFO/RR
thread.  What it in effect means is that a SCHED_NORMAL thread can run
on the sibling CPU stealing CPU time from your SCHED_FIFO/RR thread.

2. I suspect that the CPU instruction cache is too small. and that
SMT can also cause cache depletion, which would explain some xruns I
have seen while torture testing the system doing real audio and
running hackbench at the same time.

Speedstep was already covered by someone else.

CPU management is a thorny question.  That setting is probably
related to System Management Interrupts (SMIs), which can be very bad
indeed.  Normally used for fan control and other functions by the
BIOS.  The really bad thing is that they block execution and there is
nothing the kernel can do about it.  if it's only for a few usecs,
it's probably insignificant and not a problem, but if it's
milliseconds, then you will get xruns...  

There is a program called hwlatdetect in the rt-tests package that can
be used to determine how big the problem is.  It consists of a kernel
module and a python script to start it and to report the results back
to the user.  What is it does is basically to stop all kernel execution
and then to loop reading TSC timestamps.  If it finds breaks in the
data stream they will have been caused by SMIs.

-- 

   Joakim

From len at ovenwerks.net  Fri Aug 29 15:38:28 2014
From: len at ovenwerks.net (Len Ovens)
Date: Fri, 29 Aug 2014 08:38:28 -0700 (PDT)
Subject: [LAU] i5 Hyper-Threading, BIOS settings and Arch n00b pointers
In-Reply-To: <BLU436-SMTP1261D057A799C5A4F5BB0D48FDB0@phx.gbl>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl> ,
 <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
 <BLU436-SMTP1261D057A799C5A4F5BB0D48FDB0@phx.gbl>
Message-ID: <alpine.DEB.2.10.1408290828340.2499@scott.cbbs.org>

On Fri, 29 Aug 2014, dale wrote:

> Although that still begs the question of whether it being turned off in
> BIOS/UFEI before or after installation actually makes any difference...

(hyperthreading) can be turned off after boot by telling the kernel not to 
use either odd or even numbered Cores. I do not think doing it in the bios 
is better or more effective as the idea is to make sure one thread does 
not interfere with another. The only difference I could see is that it may 
be that the bios switch may remove power from the secondary thread parts 
of the chip for cooler running, but I do not know.

It should be possible to turn off the use of cores on the fly so that one 
could have a low latency record session with less performance and a higher 
latency mixdown with more performance (or video processing/editing).

I am just thinking that the P4 I gave to my son still has HT turned off, I 
should turn it back on.

--
Len Ovens
www.ovenwerks.net


From fons at linuxaudio.org  Fri Aug 29 16:01:11 2014
From: fons at linuxaudio.org (Fons Adriaensen)
Date: Fri, 29 Aug 2014 16:01:11 +0000
Subject: [LAU] i5 Hyper-Threading, BIOS settings and Arch n00b pointers
In-Reply-To: <54008E08.4070003@gmail.com>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>
 <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
 <54003AE7.1010304@gmail.com>
 <20140829094725.GA32083@linuxaudio.org>
 <54008E08.4070003@gmail.com>
Message-ID: <20140829160111.GA22009@linuxaudio.org>

On Sat, Aug 30, 2014 at 12:28:24AM +1000, Simon Wise wrote:

> that's interesting, it seemed promising at first ... so this is for
> being certain that particular and predictable sized tasks will get
> scheduled in time for their own individual deadlines, and I guess
> finding out immediately that something can't be guaranteed 

Yes. On a single CPU machine, if the API accepts your parameters
(the condition involving the P_i and T_i is satisfied), then in
theory things should work up to 100% CPU. In practice a bit less,
due to implementation losses. The strange thing is that the same
algorithm can not provide the same guarantee on multi-CPU systems.
All it can do in that case is provide an upper bound on by how 
much deadlines will be missed. There seems to be no solution for
this somewhat counter-intuitive result. 

> most audio tasks need to be done as soon as possible since they are
> of less predictable duration, are inter-dependant and all work
> together towards essentially the same deadline ...

Yes, it's those dependencies and the fact that the whole set of tasks
has a common deadline instead of each an individual one which doesn't 
fit into the scheme. It's really meant for e.g. servers which should
use a well defined percentage of the time for each of a number of tasks.
Meeting the deadlines is in fact not the end, but the means by which
this is achieved by the new scheduler.

> in multi core machines with a set of audio tasks taking most of the
> available processing is scheduling the tasks with the most depending
> on them earliest, hence putting tasks on queues with various
> priorities makes a lot more sense.

The most common case in our world is all tasks having the same period.
Then you schedule them on their dependencies, and nothing else. The
exception would be algorithms such as used in zita-convolver in which
part (or even most of) the work is done in larger periods, which are
all an exact multiple (even power of 2) of Jack's one. Rate-monotonic
scheduling (giving the tasks with larger periods lower priority) works
very well in that case, up to full CPU usage, provided all app use the
same mapping from period size to priority.

Ciao,

-- 
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)


From len at ovenwerks.net  Fri Aug 29 16:45:34 2014
From: len at ovenwerks.net (Len Ovens)
Date: Fri, 29 Aug 2014 09:45:34 -0700 (PDT)
Subject: [LAU] NOW - UEFI
In-Reply-To: <BLU437-SMTP19EF8B72E796EA0DC27E558FDB0@phx.gbl>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>  ,
 <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
 <BLU436-SMTP1261D057A799C5A4F5BB0D48FDB0@phx.gbl>
 <BLU437-SMTP19EF8B72E796EA0DC27E558FDB0@phx.gbl>
Message-ID: <alpine.DEB.2.10.1408290924290.2499@scott.cbbs.org>

On Fri, 29 Aug 2014, dale wrote:

> This has caused various headaches I have kind learned to live with. Main
> one being relating to screen brightness. If I set a low default screen
> brightness in Windows then it seems to change the range of brightness
> Linux can display. This was even more noticable when running LiveUSB
> distros! I actually couldn't get the screen bright enough to see what I
> was doing without setting the level within Windows bright again!

Is there a setting in regular bios for screen brightness? EFI my leave 
some variables around that the OS can continue to manipulat after boot... 
wonderful :P

> Then I read (or skimmed) this yesterday and feel even more confused.
> https://wiki.archlinux.org/index.php/UEFI

I see what you mean, every time I thought I was getting somewhere, I got 
sent to yet another page. The most I got out of it was that EFI is intel's 
answer to grub but with more control of the firware settings at the same 
time. The old bios included calls to access some of the HW, but no one 
used them as they were not muti-task/user friendly. It appears EFI does 
the same thing and windows uses it and Linux does not know how to access 
at least part of it. (or windows sets up its own calls within it)


> But I am starting to think I should try enabling it with a Linux
> install. Especially as I plan to completely banish Windows from the
> computer now... But I thought I would ask what more experienced Linux
> users have to say about UEFI mode vs BIOS/Legacy mode.

You will get some I don't doubt... it may be hard to tell the truth from 
FUD  ;)   EFI and Linux are still relatively new (3.10ish) and complaints 
found in one kernel version may no longer be true.

--
Len Ovens
www.ovenwerks.net


From len at ovenwerks.net  Fri Aug 29 17:24:34 2014
From: len at ovenwerks.net (Len Ovens)
Date: Fri, 29 Aug 2014 10:24:34 -0700 (PDT)
Subject: [LAU] NOW - UEFI
In-Reply-To: <alpine.DEB.2.10.1408290924290.2499@scott.cbbs.org>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>  ,
 <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
 <BLU436-SMTP1261D057A799C5A4F5BB0D48FDB0@phx.gbl>
 <BLU437-SMTP19EF8B72E796EA0DC27E558FDB0@phx.gbl>
 <alpine.DEB.2.10.1408290924290.2499@scott.cbbs.org>
Message-ID: <alpine.DEB.2.10.1408290958360.2499@scott.cbbs.org>

On Fri, 29 Aug 2014, Len Ovens wrote:

> I see what you mean, every time I thought I was getting somewhere, I got sent 
> to yet another page. The most I got out of it was that EFI is intel's answer 
> to grub but with more control of the firware settings at the same time. The 
> old bios included calls to access some of the HW, but no one used them as 
> they were not muti-task/user friendly. It appears EFI does the same thing and 
> windows uses it and Linux does not know how to access at least part of it. 
> (or windows sets up its own calls within it)

What I forgot to add, is that it is obvious that a larger part of this 
code sticks around and is still running after boot. At least with the bios 
that didn't seem to be true. Some of this code may be called by SMIs and 
of course the OS has less control of what is there. This code is not open 
source and aside from security considerations (inteligent ethernet IFs 
have that problem) RT performance may suffer.

I would suggest the xeon set of processors and MB may be more controlable 
just because the server market demands it be so. Turning SMIs off on these 
boards voids the warranty due to heat considerations, but my monitoring of 
temperature while in performance mode with all cores at 100% use on my 4 
core i5 has shown no problems. The temperature has remained well below the 
point speed reduction would be indicated. In fact ondemand can make things 
hotter, I found manually running one core at a lower speed actually 
increased the temperature. There are a lot of different xeon models and it 
is possible to get them with no HT or boost. I was thinking of getting 
such a MB, but could find none with PCI slots. I do not know if they would 
make better audio boards though. At the other end of the server market is 
the atom MB which are good headless audio machines (the graphics part is 
not linux friendly) using less power, cooling and being small besides. 
Many of these MB have one PCI slot and there are "L" adaptors that allow 
the card to use less space so an older (well supported) audio card could 
be used.

--
Len Ovens
www.ovenwerks.net


From czhenry at gmail.com  Fri Aug 29 17:42:29 2014
From: czhenry at gmail.com (Charles Z Henry)
Date: Fri, 29 Aug 2014 12:42:29 -0500
Subject: [LAU] i5 Hyper-Threading, BIOS settings and Arch n00b pointers
In-Reply-To: <20140829160111.GA22009@linuxaudio.org>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>
 <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
 <54003AE7.1010304@gmail.com>
 <20140829094725.GA32083@linuxaudio.org>
 <54008E08.4070003@gmail.com>
 <20140829160111.GA22009@linuxaudio.org>
Message-ID: <CAPfmNOHgaL8OWzDK8Euov_JTu8AwbM3yKbRhczBk9uELKL2tew@mail.gmail.com>

On Fri, Aug 29, 2014 at 11:01 AM, Fons Adriaensen <fons at linuxaudio.org> wrote:
> On Sat, Aug 30, 2014 at 12:28:24AM +1000, Simon Wise wrote:
...
>> in multi core machines with a set of audio tasks taking most of the
>> available processing is scheduling the tasks with the most depending
>> on them earliest, hence putting tasks on queues with various
>> priorities makes a lot more sense.
>
> The most common case in our world is all tasks having the same period.
> Then you schedule them on their dependencies, and nothing else. The
> exception would be algorithms such as used in zita-convolver in which
> part (or even most of) the work is done in larger periods, which are
> all an exact multiple (even power of 2) of Jack's one. Rate-monotonic
> scheduling (giving the tasks with larger periods lower priority) works
> very well in that case, up to full CPU usage, provided all app use the
> same mapping from period size to priority.

I recently sat in on Karthik Poduval's thesis defense on the topic of
Jack scheduling with the hierarchical group scheduler (I hope you
don't mind the mention, Karthik).  He showed great scheduling
efficiency for a real-time thread and reduced jitter.

Hardware locality and group scheduling looks like a great approach for
real-time audio and video--this approach could handle many of the
issues mentioned in this thread.  For example, 1-2 ms is significant
for a real-time audio thread, but 16-17 ms is the important duration
for 60Hz video.  One may need low latency, the other may need high
throughput.  The solution is: 2 different schedulers in a hierarchy,
each handling different groups of processes.  There can be any number
of schedulers in group hierarchical scheduling, so the system
processes could be handled by another (the first/original) scheduler
as well.  HWLoc can add to this scheme, ensuring there's no contention
among different RT processes that operate on different time-scales (on
multi-core systems).

Aside from knowing this general scheme and principle, I'm ignorant of
the fine details of kernel schedulers.  I may not be able to add much
more to this discussion (but I'll keep reading).

From len at ovenwerks.net  Fri Aug 29 18:01:11 2014
From: len at ovenwerks.net (Len Ovens)
Date: Fri, 29 Aug 2014 11:01:11 -0700 (PDT)
Subject: [LAU] Status of lisalo/lisaloQt and calfbox?
In-Reply-To: <em12278a3f-93d5-4e5e-bdf2-0bb1afabc317@dcs-jb-8a>
References: <em12278a3f-93d5-4e5e-bdf2-0bb1afabc317@dcs-jb-8a>
Message-ID: <alpine.DEB.2.10.1408291026070.2499@scott.cbbs.org>

On Fri, 29 Aug 2014, Jonathan E Brickman wrote:

> some careful experimentation; I am trying to build a very dense Strings
> instrument for my synth box ( http://lsn.ponderworthy.com ), and
> although the 14-item LinuxSynth setup I have right now sounds great for
> a second or two of note-hold, one hears increasing spits and pops and
> static after more hold even on one note.

I was surprised that you include X and a wm at all. I would suggest 
running screen from dbus-launch could do the same thing so long as all 
your applications are CLI. I have set a box up this way and been able to 
run pulseaudio, jackdbus and other things (before I ran out of memory on 
the old P300). I would not suggest pulse in your case though. There are 
some very good CLI tools for keeping track of jack connections.

It is relatively easy using sys v init, upstart or systemd to run a shell 
script as a user that would start dbus and screen (or another cli session 
manager) with a number of screens open and running some predefined 
program. Ssh login to the box and type:
screen -df
will connect you to the already running screen instance running as your 
user. I have done this from an android phone, though the text was really 
too small :) but a tablet would be better. If you run out of buttons on 
your keyboard, USB keyboards (qwerty) are cheap and the controller inside 
can be directly connected to stomp switches. Actkbd can assign keypress to 
CLI command or with the extension I am working on (it is far enough along 
to be useful to you I think) send a jack midi command to your midi 
filter/combiner or direct to an app.

There are also a number of MIDI control apps that send MIDI via 
ethernet/wireless some of them for android. qmidinet should now be able to 
run headless too.

Just a note and you may already know this, you can stack commands with 
jack_control. For example I start jackdbus like this:
jack_control ds $DRIVER dps device $DEV dps rate $RATE dps period $FRAME \
         dps nperiods $PERIOD start
>From a script.

I understand using 96k for this project, but in the case of pops and such 
have you tried setting 48k just to see if it goes away? Then setting 
period 32 would give the same latency, but may still give cleaner sound. 
You can use jack_bufsize to change period on the fly, but note that there 
will be audiable artifacts and some applications (rakarrack for example) 
never recover and need to be restarted.

--
Len Ovens
www.ovenwerks.net


From jeb at ponderworthy.com  Fri Aug 29 18:28:00 2014
From: jeb at ponderworthy.com (Jonathan E Brickman)
Date: Fri, 29 Aug 2014 18:28:00 +0000
Subject: [LAU] Status of lisalo/lisaloQt and calfbox?
In-Reply-To: <alpine.DEB.2.10.1408291026070.2499@scott.cbbs.org>
Message-ID: <emf80d1baa-29d8-4b85-930c-69791a121a43@dcs-jb-8a>

Greetings, Len, and thanks for writing.  Reponses below.

>
>I was surprised that you include X and a wm at all. I would suggest
>running screen from dbus-launch could do the same thing so long as all
>your applications are CLI. I have set a box up this way and been able 
>to
>run pulseaudio, jackdbus and other things (before I ran out of memory 
>on
>the old P300). I would not suggest pulse in your case though. There are
>some very good CLI tools for keeping track of jack connections.
It is a perennial question for me, whether or not to start going CLI.  
So far every time I have begun to look there, I have found yet another 
tweak I want to do NOW which is very easy in GUI and rather arcane in 
CLI.  One of the better examples is the use of Calfjackhost, most of my 
patches use a single CJH with three plugins (usually EQ, reverb, and 
compression), and I haven't found a way to handle that anywhere near as 
easily in CLI, and the visualizations help.  Similarly I use Yoshimi a 
lot, three simultaneous instances in two different patches, and although 
I could run it CLI, I could not easily configure it CLI.  I have also 
thought about setting up two different major modes, CLI and GUI, using 
the same patchset items, but it seemed to me like a lot of 
time-expenditure and complexity for minimal gain; the current hardware 
is not very CPU- or RAM-limited, and it does seem best to use exactly 
the same setup in production as on the bench.  So I mix, CLI for some 
things, GUI when it's just easier.  And figuring out Jack connections is 
a lot easier in GUI :-)  I don't keep qjackctl or one of the newer ones 
running all the time, but it sure is nice to have it one click away.

>
>It is relatively easy using sys v init, upstart or systemd to run a 
>shell
>script as a user that would start dbus and screen (or another cli 
>session
>manager) with a number of screens open and running some predefined
>program. Ssh login to the box and type:
>screen -df
>will connect you to the already running screen instance running as your
>user. I have done this from an android phone, though the text was 
>really
>too small :) but a tablet would be better. If you run out of buttons on
>your keyboard, USB keyboards (qwerty) are cheap and the controller 
>inside
>can be directly connected to stomp switches. Actkbd can assign keypress 
>to
>CLI command or with the extension I am working on (it is far enough 
>along
>to be useful to you I think) send a jack midi command to your midi
>filter/combiner or direct to an app.
I have been wondering how much capacity I'm wasting keeping the desktop 
manager up.  I have a good graphics card so that isn't a limiter, but 
obviously there is some motherboard bandwidth being taken.  It may be 
that I should do something like the Archbang default, which is just 
enough.

>
>There are also a number of MIDI control apps that send MIDI via
>ethernet/wireless some of them for android. qmidinet should now be able 
>to
>run headless too.
>
>Just a note and you may already know this, you can stack commands with
>jack_control. For example I start jackdbus like this:
>jack_control ds $DRIVER dps device $DEV dps rate $RATE dps period 
>$FRAME \
>          dps nperiods $PERIOD start
>From a script.
Wow, I had forgotten about that, although I think I saw it in a doc once 
or twice.

>
>I understand using 96k for this project, but in the case of pops and 
>such
>have you tried setting 48k just to see if it goes away? Then setting
>period 32 would give the same latency, but may still give cleaner 
>sound.
>You can use jack_bufsize to change period on the fly, but note that 
>there
>will be audiable artifacts and some applications (rakarrack for 
>example)
>never recover and need to be restarted.
>
>--
>Len Ovens
>www.ovenwerks.net

Aha, did not think of reducing to 48k for just the one patch, that 
should work just fine, because I have found that to keep Jack completely 
stable I have to kill the process and restart between patches anyhow.  
I'll have to keep that one in the bag of tricks!

Jonathan E. Brickman
Ponderworthy Music | jeb at ponderworthy.com | (785)233-9977 | 
http://ponderworthy.com <http://ponderworthy.com/>
>

From gnome at hawaii.rr.com  Fri Aug 29 20:15:02 2014
From: gnome at hawaii.rr.com (david)
Date: Fri, 29 Aug 2014 10:15:02 -1000
Subject: [LAU] NOW - UEFI
In-Reply-To: <BLU437-SMTP19EF8B72E796EA0DC27E558FDB0@phx.gbl>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl> ,
 <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
 <BLU436-SMTP1261D057A799C5A4F5BB0D48FDB0@phx.gbl>
 <BLU437-SMTP19EF8B72E796EA0DC27E558FDB0@phx.gbl>
Message-ID: <5400DF46.1000207@hawaii.rr.com>

On 08/29/2014 03:20 AM, dale wrote:
> On Fri, 2014-08-29 at 18:39 +0545, dale wrote:
>
>> Although that still begs the question of whether it being turned off in
>> BIOS/UFEI before or after installation actually makes any difference...
>>
>
> Now this is something that has been harder for me to get my head around
> than benefit/disadvantages to HT!
>
> When I got this laptop it had Window7 pre-installed with a bunch of
> Lenovo software on it. At the time I know nothing of changes from BIOS
> to UEFI! One of the first things I did was install Ubuntu Studio 12.04
> on it, using Legacy mode as I wasn't sure if it was UEFI compatible and
> my laptop seems to allow both.
>
> This has caused various headaches I have kind learned to live with. Main
> one being relating to screen brightness. If I set a low default screen
> brightness in Windows then it seems to change the range of brightness
> Linux can display. This was even more noticable when running LiveUSB
> distros! I actually couldn't get the screen bright enough to see what I
> was doing without setting the level within Windows bright again!
>
> Then I read (or skimmed) this yesterday and feel even more confused.
> https://wiki.archlinux.org/index.php/UEFI
>
> But I am starting to think I should try enabling it with a Linux
> install. Especially as I plan to completely banish Windows from the
> computer now... But I thought I would ask what more experienced Linux
> users have to say about UEFI mode vs BIOS/Legacy mode.
>
> Again thanks for all the help. Dale.

Hmm, found this page at the Ubuntu Community about Ubuntu and UEFI. 
Apparently it can be installed in SecureBoot mode. I don't know what 
other distros can. I have Debian Sid on 2 machines here with EUFI, but 
both are in BIOS/Legacy mode.

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From federicogalland at gmail.com  Fri Aug 29 21:19:04 2014
From: federicogalland at gmail.com (Federico Galland)
Date: Fri, 29 Aug 2014 18:19:04 -0300
Subject: [LAU] Noo 'pooter :)
In-Reply-To: <20140826212439.352361f6@debian>
References: <20140826212439.352361f6@debian>
Message-ID: <20140829181904.21ec77372ba470ff8b24c1f3@gmail.com>

> As this will be a clean install, I'm wondering what people might suggest as
> for best distro to make full use of it - all my other machines have had a
> progression of debian upgrades so are probably full of crud.

I've used arch linux on my laptop for 5 years, and while I loved it as a general purpose OS, the AUR was just more problematic with audio software than building on my own.
In the end, I dumped it in favor of debian (a 6 years old install, still on its feet) because of the convenience of the kxstudio repos. I just feared having to update my system, and didn't really like to fill my bandwidth with large updates.

If I were you I'd stick with debian.

Of course there is the downside of packages being compiled without optimizations, and you do miss the PKGBUILD system, which is the best I've seen on linux land.

Cheers.

-- 
educados por el rock,
as? est?n las cosas.

<federicogalland at gmail.com>

From guido.aulisi at gmail.com  Fri Aug 29 21:39:09 2014
From: guido.aulisi at gmail.com (Guido Aulisi)
Date: Fri, 29 Aug 2014 23:39:09 +0200
Subject: [LAU] A MIDI tone editor for the Roland MKS-70 Super JX analog
	synthesizer
Message-ID: <1409348349.22362.4.camel@yoda.heavyware>

Hello,
this is my attempt to build a GTK+ (really GTKmm) C++ application to
edit patches on the Roland MKS 70.

Code is on github: https://github.com/tartina/pg800.git

Tarballs and (maybe) RPM packages will come... some day!

Ciao
Guido


From harryhaaren at gmail.com  Fri Aug 29 23:01:05 2014
From: harryhaaren at gmail.com (Harry van Haaren)
Date: Sat, 30 Aug 2014 00:01:05 +0100
Subject: [LAU] Session management with NSM
In-Reply-To: <20140829213201.25bca02a@eeyore.mozart.uni-klu.ac.at>
References: <20140828224247.236111e1@eeyore>
 <CAKudYbP4LiQDs4kWgU3a_YKCM9HVjmS3qO+hngNOky91r0XOUQ@mail.gmail.com>
 <20140829213201.25bca02a@eeyore.mozart.uni-klu.ac.at>
Message-ID: <CAKudYbNfJjNSJfHRgRSz7NX7O2Bd9=gyhAsR+qAwZf7Ofw5cwA@mail.gmail.com>

On Fri, Aug 29, 2014 at 8:32 PM, Philipp ?berbacher <murks at tuxfamily.org> wrote:
> Thanks a lot for your reply Harry.
Cheers, be careful to not remove the list from replies: its good to
keep everything in the archives for future reference :)

>> That's the correct way to handle this, as far as I know. Its useful to
>> have different directories on one system: it allows subdiving your
>> available sessions into groups like "albums" or
>> "projects-with-certain-people". Although I agree it feels a little
>> clunky, its quite powerful and useful.
>
> There could also be a subdivision in the NSM GUI. Well, the current way
> is certainly the simpler implementation, not sure it's simpler for the
> users :)

Sure, and my original suggestion was a "stepping-stone" type idea,
with hopes to improve the workflow furthur, once this has become the
"biggest" issue NSM has :D

>> > 2. Adding programs to sessions through the GUI ("Add Client to
>> > Session") is the only way? Is there no way to attach running clients
>> > or at least have some comfort like tab completion to add clients?
>> NSM does not support this "attach" workflow, but tab completion or a
>> list of available (fully supported) NSM clients would be a good
>> improvement on workflow. This should be discussed as to how best
>> implement it: i'm not sure.
>
> Right, a list of supported Clients would also be nice, however, I see
> two problems:
> 1. The list would need to be updated somehow, and even then it would be
> a bit problematic because different distributions ship different
> versions of the software. NSM might already list a program as supported
> while the installed version of the program does not yet support NSM.
> 2. The other programs, audio or just related, should ideally also be
> listed, and that task is impossible.

Actually this might be possible to solve with a "packaging" trick as
such: have programs install a file into a specific location (that is
currently *not* used by any program) to denote its NSM support. I'll
suggest installing a file in /usr/share/nsm/ , and if there's a file
there, then the filename without extension represents that a program
is capable of NSM. This would require *all* NSM clients to explicitly
add an NSM file.

Perhaps other developers more involved in packaging /
"feature-announcing" will have a better idea here, I'm all ears, my
suggestion above is just that: a suggestion.

>> > 3. Jack and NSM. How do you handle that? It is possible to start
>> > jack through NSM proxy and I guess it is OK to do that as long as
>> > jack reliably starts before jackpatch (something I'm not sure of).
>> > First I had just jackpatch in there and it started jack for me with
>> > a whole lot of options that are unfamiliar to me and probably not
>> > needed.
>>
>> I imagine that NSM will launch said JACK apps, and if one is set to
>> "start JACK" on jack_client_open() in its code, then it will start
>> JACK with the settings in  ~/.jackdrc   Perhaps the inclusion of a
>> "Start JACK" type client with particular settings can be implemented
>> in order to handle this? I'm open for suggestions too.
>
> That seems to be what happens, and its a race. In my experience
> jackpatch wins the race against jackd, so I have to start jack before
> the session.
> A start_jack client could be useful, but from what I have seen all we
> really need is the possibility to start a client before the others.
> The simple way would be a timeout, but you'd still have the
> race. Ideally there would be some way to tell NSM that jack has
> started and is ready. I have doubts that this is possible with plain
> jack1 and NSM proxy, maybe a special start_jack client could help here.

NSM doesn't *explicitly* require JACK to be running actually: its
probably its most common use right now, but setting an explicit
dependency on JACK should be avoided. Perhaps a flag could be
introduce on a per-client basis, that represents
"start-before-others". This way, a "jackd" or "start-jack" client can
be loaded before the rest. Or even two or more "before-others" clients
could set up whatever needs setting up, before "normal-time" NSM
clients are loaded.

Again, welcome input from users / devs.

>> > 4. CLI clients. Are they generally not supported? I added the lv2
>> > host that was recommended to me (jalv) and had to do that through
>> > the NSM proxy, so the settings won't be saved even though the
>> > plugin (fabla in this case) can save its settings. This sort of
>> > defeats session management. With all the CLI tools we have it would
>> > be a pitty if that was generally not supported. On a sidenote, can
>> > someone recommend a plugin host that is supported?
>>
>> CLI clients are supported just like clients with a GUI, there is no
>> difference to NSM. The issue you're encountering here is that JALV
>> currently doesn't support NSM, which is something that I agree needs
>> fixing. I'll put JALV NSM support on the TODO, its something I've
>> lacked myself too.
>
> Ok, great. Does a CLI NSM client exist that I can try?

None that I know of right now: Indeed JALV needs NSM, and jalv (the
command line version) will then be such a client.

> I also noticed that JALV keeps hanging around
> after I close the session it is part of, is that expected behavior?

This can be fixed by sending the "SIGTERM" in the lower part of the
"nsm-proxy" configuration dialog (where you fill in "jalv.gtk", and
the arguments to load a certain plugin).

>> > Well, that's it for now. Last time I heard about NSM I got the
>> > impression that it takes care of session management once and for
>> > all, but the first half our gave me a different impression.
>> OpenAV stands behind NSM: I'm willing to do my best to cooperate with
>> project developers to implement NSM in various programs, and improve
>> the workflow of session management.
>>
>> If there's any furthur questions, please ask, in the mean time, I'll
>> try code up some NSM :) -Harry
>
> Thanks a lot for your help Harry, we have used crutches for session
> management long enough.

Agreed, lets try fix this together with the communit in the next
weeks, and never look back ;)
Cheers, -Harry

From simonzwise at gmail.com  Sat Aug 30 02:39:46 2014
From: simonzwise at gmail.com (Simon Wise)
Date: Sat, 30 Aug 2014 12:39:46 +1000
Subject: [LAU] i5 Hyper-Threading, BIOS settings and Arch n00b pointers
In-Reply-To: <20140829160111.GA22009@linuxaudio.org>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>
 <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
 <54003AE7.1010304@gmail.com> <20140829094725.GA32083@linuxaudio.org>
 <54008E08.4070003@gmail.com> <20140829160111.GA22009@linuxaudio.org>
Message-ID: <54013972.90606@gmail.com>

On 30/08/14 02:01, Fons Adriaensen wrote:
> On Sat, Aug 30, 2014 at 12:28:24AM +1000, Simon Wise wrote:
>
>> that's interesting, it seemed promising at first ... so this is for
>> being certain that particular and predictable sized tasks will get
>> scheduled in time for their own individual deadlines, and I guess
>> finding out immediately that something can't be guaranteed
>
> Yes. On a single CPU machine, if the API accepts your parameters
> (the condition involving the P_i and T_i is satisfied), then in
> theory things should work up to 100% CPU. In practice a bit less,
> due to implementation losses. The strange thing is that the same
> algorithm can not provide the same guarantee on multi-CPU systems.
> All it can do in that case is provide an upper bound on by how
> much deadlines will be missed. There seems to be no solution for
> this somewhat counter-intuitive result.

while on a single core machine there isn't a problem with dependencies in DSP 
either ... audio is easier ... there is enough time to finish it all, or there 
isn't.



>
> Yes, it's those dependencies and the fact that the whole set of tasks
> has a common deadline instead of each an individual one which doesn't
> fit into the scheme. It's really meant for e.g. servers which should
> use a well defined percentage of the time for each of a number of tasks.
> Meeting the deadlines is in fact not the end, but the means by which
> this is achieved by the new scheduler.




so multicore DSP scheduling is something like dependency based boot scheduling 
... that has taken a lot of effort.

>
> The most common case in our world is all tasks having the same period.
> Then you schedule them on their dependencies, and nothing else. The
> exception would be algorithms such as used in zita-convolver in which
> part (or even most of) the work is done in larger periods, which are
> all an exact multiple (even power of 2) of Jack's one. Rate-monotonic
> scheduling (giving the tasks with larger periods lower priority) works
> very well in that case, up to full CPU usage, provided all app use the
> same mapping from period size to priority.

so a potential use for this scheduler in audio would be if we have a non-DSP 
task which does not take a big part of the processing overall but that must keep 
up to date, and should not be left to the last moment leaving it all in one 
sample period potentially causing a DSP glitch.

Something like dealing with the control inputs or the feedback required to play 
an instrument where responsiveness and lack of jitter are really important.

Then while the DSP threads are running with real-time priority we can still 
guarantee an allocation of a steady but strictly limited amount of processing 
time for the critical parts of perhaps midi or automation which is evenly spread 
over the DSP periods. Maybe also making sure some particular aspects of the GUI 
are always updated even when most of it slows down if the DSP load gets too high.

Simon

From dj_kaza at hotmail.com  Sat Aug 30 05:23:33 2014
From: dj_kaza at hotmail.com (dale)
Date: Sat, 30 Aug 2014 11:08:33 +0545
Subject: [LAU] NOW - UEFI
In-Reply-To: <alpine.DEB.2.10.1408290924290.2499@scott.cbbs.org>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>
 , <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
 <BLU436-SMTP1261D057A799C5A4F5BB0D48FDB0@phx.gbl>
 <BLU437-SMTP19EF8B72E796EA0DC27E558FDB0@phx.gbl>
 <alpine.DEB.2.10.1408290924290.2499@scott.cbbs.org>
Message-ID: <BLU436-SMTP2432AA3D14768BBFE11A83A8FD80@phx.gbl>

On Fri, 2014-08-29 at 09:45 -0700, Len Ovens wrote:
> On Fri, 29 Aug 2014, dale wrote:
> 
> > This has caused various headaches I have kind learned to live with. Main
> > one being relating to screen brightness. If I set a low default screen
> > brightness in Windows then it seems to change the range of brightness
> > Linux can display. This was even more noticable when running LiveUSB
> > distros! I actually couldn't get the screen bright enough to see what I
> > was doing without setting the level within Windows bright again!
> 
> Is there a setting in regular bios for screen brightness? EFI my leave 
> some variables around that the OS can continue to manipulat after boot... 
> wonderful :P
> 
No and I've looked again and again thinking it must be settable
elsewhere if it can be set from the Lenovo software/Windows and affect
all booted OSes.

> > Then I read (or skimmed) this yesterday and feel even more confused.
> > https://wiki.archlinux.org/index.php/UEFI
> 
> I see what you mean, every time I thought I was getting somewhere, I got 
> sent to yet another page. The most I got out of it was that EFI is intel's 
> answer to grub but with more control of the firware settings at the same 
> time. The old bios included calls to access some of the HW, but no one 
> used them as they were not muti-task/user friendly. It appears EFI does 
> the same thing and windows uses it and Linux does not know how to access 
> at least part of it. (or windows sets up its own calls within it)
> 
> 
> > But I am starting to think I should try enabling it with a Linux
> > install. Especially as I plan to completely banish Windows from the
> > computer now... But I thought I would ask what more experienced Linux
> > users have to say about UEFI mode vs BIOS/Legacy mode.
> 
> You will get some I don't doubt... it may be hard to tell the truth from 
> FUD  ;)   EFI and Linux are still relatively new (3.10ish) and complaints 
> found in one kernel version may no longer be true.
> 
> --
> Len Ovens
> www.ovenwerks.net
> 

I guess it must tie into ACPI somehow. Or would that be a silly
assumption? I believe ACPI controls hardware functions, including screen
brightness and fan controllers, while the system is running. Or am I
confused? I know on the rare occasions I've had a crash there have been
lots of ACPI Unknown (or similar) messages in my log! Did mean to look
into it further at some point...


From dj_kaza at hotmail.com  Sat Aug 30 05:29:33 2014
From: dj_kaza at hotmail.com (dale)
Date: Sat, 30 Aug 2014 11:14:33 +0545
Subject: [LAU] NOW - UEFI
In-Reply-To: <5400DF46.1000207@hawaii.rr.com>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>
 , <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
 <BLU436-SMTP1261D057A799C5A4F5BB0D48FDB0@phx.gbl>
 <BLU437-SMTP19EF8B72E796EA0DC27E558FDB0@phx.gbl>
 <5400DF46.1000207@hawaii.rr.com>
Message-ID: <BLU436-SMTP1004286747614A7A09B7F998FD80@phx.gbl>

On Fri, 2014-08-29 at 10:15 -1000, david wrote:

> 
> Hmm, found this page at the Ubuntu Community about Ubuntu and UEFI. 
> Apparently it can be installed in SecureBoot mode. I don't know what 
> other distros can. I have Debian Sid on 2 machines here with EUFI, but 
> both are in BIOS/Legacy mode.
> 

I haven't looked in probably over a year but from what I remember at the
time it seemed only Canonical (Ubuntu) and Red Hat Foundation were going
to get the keys required for Secure Boot.

On my laptop I have Secure Boot but it is disabled and separate to the
UEFI option.

Should probably have mentioned at the beginning it's a Lenovo X230,
although most of my questions are of the more general kind.

Dale.


From dj_kaza at hotmail.com  Sat Aug 30 05:42:24 2014
From: dj_kaza at hotmail.com (dale)
Date: Sat, 30 Aug 2014 11:27:24 +0545
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <alpine.DEB.2.10.1408290728170.2499@scott.cbbs.org>
References: <20140827143830.GA7320@aol.com>
 , <201408282037.53956.gheskett@wdtv.com>
 , <DUB113-W1107FDBFBE7CDE95335DD6F8FDB0@phx.gbl>
 , <201408282245.42030.gheskett@wdtv.com>
 <DUB113-W3024DDF80474D625B42FD48FDB0@phx.gbl>
 <alpine.DEB.2.10.1408290728170.2499@scott.cbbs.org>
Message-ID: <BLU436-SMTP115B35F8F0881A47DC9C1558FD80@phx.gbl>

On Fri, 2014-08-29 at 08:12 -0700, Len Ovens wrote:
> On Fri, 29 Aug 2014, Kaza Kore wrote:
> 
> > Anyway it's probably more important to talk about the standardised DV25 and DV50
> > protocol all these commercial/prosumer products use for communication that
> > tape/card formats. There are some Sony and Panasonic camera that do this fine
> > over USB so it's not impossible or a problem with USB itself. I see yours (and
> > apparently many others) claim to have some kind of USB Streaming but for some
> > reason it's not usually full quality, as you would get from Firewire. Wonder
> > why...
> 
> Intel says that nobody really needs that low of a latency anyway in 
> response some sound cards problems with one of their implementations of 
> USB3. (the problem was USB2 cards with USB3 ports which are supposed to be 
> compatable)
> 
> In recording latency is not the issue it is with live work. For recording 
> latency needs to known and constant and reasonably low for monitoring. I 
> have heard/read people who say what does it matter if you move your head a 
> foot or two closer or father from the speaker? But in live work if the 
> same audio comes from different places it is called a filter.
> 

Yes if it is coming from two sources of different distances you get a
comb filter but that's not what we're talking about with distance/delay
here. It's more absolute latency.

Examples I was presented at while at college/university were:
* A piano player striking a key and hearing a sound, from the downwards
motion to hearing the sound is about 6ms. Taking into account both
mechanical transference from key to string and the sound to the ear.
* For comparison this would be the same as a guitarist standing six feet
away from his guitar amplifier if we lived in a ideal world (where
electricity travelled the speed of light) but there is obviously
propagation delay plus any added by stomp boxes (s)he may have. This is
not a distance at which a guitarist finds it difficult to play!

I sometimes think the hunt for super-low latency is a bit absurd! 3ms,
to give you a 6ms round trip, should be a workable amount for pretty
much anybody and most I expect could cope with quite a lot higher (not
many working methods require the full round trip!)

But full round trip is probably also the time when delay effects (com
filter and echo) become important, such as using a PC as an LMS or FOH
mixing desk. I would consider these specialist cases though.

Just my 2c ;-)



From guido.aulisi at gmail.com  Sat Aug 30 07:07:28 2014
From: guido.aulisi at gmail.com (Guido Aulisi)
Date: Sat, 30 Aug 2014 09:07:28 +0200
Subject: [LAU] A MIDI tone editor for the Roland MKS-70 Super JX analog
	synthesizer
Message-ID: <1409382448.2395.1.camel@yoda.heavyware>

Hello,
this is my attempt to build a GTK+ (really GTKmm) C++ application to
edit patches on the Roland MKS 70.

Code is on github: https://github.com/tartina/pg800.git

Tarballs and (maybe) RPM packages will come... some day!

Ciao
Guido


From gnome at hawaii.rr.com  Sat Aug 30 08:10:06 2014
From: gnome at hawaii.rr.com (david)
Date: Fri, 29 Aug 2014 22:10:06 -1000
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <9B60CEE4-F83D-4531-A9AD-2A229AD5309B@billgribble.com>
References: <20140827143830.GA7320@aol.com>
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>
 <20140827154520.GB7320@aol.com>
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>
 <20140827192248.GC7320@aol.com> <53FF4A76.8000100@stackingdwarves.net>
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>
 <53FF52B0.4030702@stackingdwarves.net> <20140828200726.GA16743@aol.com>
 <54002437.6080607@hawaii.rr.com>
 <9B60CEE4-F83D-4531-A9AD-2A229AD5309B@billgribble.com>
Message-ID: <540186DE.7010700@hawaii.rr.com>

Well, the Dell E6530 seems to have an EC slot. Currently listed as 
"temporarily out of stock". Viewing their list of suggested alternatives 
offers me a bunch of refurbished Dell tablets and models running Celerons.

Googling around hasn't found any manufacturers selling laptops with EC 
slots. HP (junk laptops), Acer, Lenovo, Toshiba - what models do they 
sell anymore that have EC slots?

On 08/29/2014 04:29 AM, Bill Gribble wrote:
> I bought a Dell about a year ago with an ExpressCard slot (E6420).  ISTR many manufacturers make them, tho it's tough sometimes to meet a specific combination of requirements including EC.
>
> Thanks,
> Bill Gribble
>
>> On Aug 29, 2014, at 2:56, david <gnome at hawaii.rr.com> wrote:
>>
>>> On 08/28/2014 10:07 AM, Peter P. wrote:
>>> * J?rn Nettingsmeier <nettings at stackingdwarves.net> [2014-08-28 12:03]:
>>>>> On 08/28/2014 05:48 PM, Paul Davis wrote:
>>>>>
>>>>> On Thu, Aug 28, 2014 at 11:27 AM, J?rn Nettingsmeier
>>>>> <nettings at stackingdwarves.net <mailto:nettings at stackingdwarves.net>> wrote:
>>>>>
>>>>>     no personal experience, but i guess the ff400/800 support is fairly
>>>>>     complete by now. check with jonathan woithe on the ffado list to
>>>>>     make sure.
>>>>>
>>>>> just in time for firewire to ride off into the sunset ...
>>> Yes, nice comment Paul. I feel I have to keep chasing laptops that
>>> have either an express card slot or a firewire connector.
>>
>> Does anyone make a laptop with either of those anymore? Didn't find one on Amazon.


-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From gnome at hawaii.rr.com  Sat Aug 30 08:13:39 2014
From: gnome at hawaii.rr.com (david)
Date: Fri, 29 Aug 2014 22:13:39 -1000
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <DUB113-W18E942790CECC11578A7BE8FDB0@phx.gbl>
References: <20140827143830.GA7320@aol.com>,
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>,
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>,
 <20140827154520.GB7320@aol.com>,
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>,
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>,
 <20140827192248.GC7320@aol.com> <53FF4A76.8000100@stackingdwarves.net>,
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>,
 <53FF52B0.4030702@stackingdwarves.net> <20140828200726.GA16743@aol.com>,
 <54002437.6080607@hawaii.rr.com> <DUB113-W18E942790CECC11578A7BE8FDB0@phx.gbl>
Message-ID: <540187B3.1090504@hawaii.rr.com>

On 08/28/2014 11:33 PM, Kaza Kore wrote:
>
>
>  > Date: Thu, 28 Aug 2014 20:56:55 -1000
>  > From: gnome at hawaii.rr.com
>  > To: linux-audio-user at lists.linuxaudio.org
>  > Subject: Re: [LAU] Successor/replacement for RME HDSP+Multiface?
>  >
>  > On 08/28/2014 10:07 AM, Peter P. wrote:
>  > > * J?rn Nettingsmeier <nettings at stackingdwarves.net> [2014-08-28 12:03]:
>  > >> On 08/28/2014 05:48 PM, Paul Davis wrote:
>  > >>>
>  > >>> On Thu, Aug 28, 2014 at 11:27 AM, J?rn Nettingsmeier
>  > >>> <nettings at stackingdwarves.net
> <mailto:nettings at stackingdwarves.net>> wrote:
>  > >>>
>  > >>> no personal experience, but i guess the ff400/800 support is fairly
>  > >>> complete by now. check with jonathan woithe on the ffado list to
>  > >>> make sure.
>  > >>>
>  > >>> just in time for firewire to ride off into the sunset ...
>  > > Yes, nice comment Paul. I feel I have to keep chasing laptops that
>  > > have either an express card slot or a firewire connector.
>  >
>  > Does anyone make a laptop with either of those anymore? Didn't find one
>  > on Amazon.
>
> LOADS of laptops have ExpressCard slots!

Really? Which ones?

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From gnome at hawaii.rr.com  Sat Aug 30 08:17:49 2014
From: gnome at hawaii.rr.com (david)
Date: Fri, 29 Aug 2014 22:17:49 -1000
Subject: [LAU] NOW - UEFI
In-Reply-To: <BLU436-SMTP1004286747614A7A09B7F998FD80@phx.gbl>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>	 ,
 <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>	
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>	
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>	
 <BLU436-SMTP1261D057A799C5A4F5BB0D48FDB0@phx.gbl>	
 <BLU437-SMTP19EF8B72E796EA0DC27E558FDB0@phx.gbl>	
 <5400DF46.1000207@hawaii.rr.com>
 <BLU436-SMTP1004286747614A7A09B7F998FD80@phx.gbl>
Message-ID: <540188AD.5030507@hawaii.rr.com>

On 08/29/2014 07:29 PM, dale wrote:
> On Fri, 2014-08-29 at 10:15 -1000, david wrote:
>
>>
>> Hmm, found this page at the Ubuntu Community about Ubuntu and UEFI.
>> Apparently it can be installed in SecureBoot mode. I don't know what
>> other distros can. I have Debian Sid on 2 machines here with EUFI, but
>> both are in BIOS/Legacy mode.
>
> I haven't looked in probably over a year but from what I remember at the
> time it seemed only Canonical (Ubuntu) and Red Hat Foundation were going
> to get the keys required for Secure Boot.

Thanks for clarification, both my System76 laptop and ASUS-based desktop 
have SecureBoot disabled. The System76 came with Ubuntu 14 installed.

> On my laptop I have Secure Boot but it is disabled and separate to the
> UEFI option.
>
> Should probably have mentioned at the beginning it's a Lenovo X230,
> although most of my questions are of the more general kind.

Personally, I sometimes think the NSA is behind EUFI. What better place 
to hide your spyware than built into the firmware? Then, of course, 
someone realized that China, maker of most laptops and computers in 
world, could be doing the same thing ... ;)

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From gnome at hawaii.rr.com  Sat Aug 30 08:21:50 2014
From: gnome at hawaii.rr.com (david)
Date: Fri, 29 Aug 2014 22:21:50 -1000
Subject: [LAU] NOW - UEFI
In-Reply-To: <BLU436-SMTP2432AA3D14768BBFE11A83A8FD80@phx.gbl>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl> ,
 <DUB113-W10492F509E35C517AAE5F3C8FDA0@phx.gbl>
 <DUB113-W400A86D80FCEEB2B27DCD88FDA0@phx.gbl>
 <alpine.DEB.2.10.1408282139450.2499@scott.cbbs.org>
 <BLU436-SMTP1261D057A799C5A4F5BB0D48FDB0@phx.gbl>
 <BLU437-SMTP19EF8B72E796EA0DC27E558FDB0@phx.gbl>
 <alpine.DEB.2.10.1408290924290.2499@scott.cbbs.org>
 <BLU436-SMTP2432AA3D14768BBFE11A83A8FD80@phx.gbl>
Message-ID: <5401899E.2020101@hawaii.rr.com>

On 08/29/2014 07:23 PM, dale wrote:
> On Fri, 2014-08-29 at 09:45 -0700, Len Ovens wrote:
>> On Fri, 29 Aug 2014, dale wrote:
>>
>>> This has caused various headaches I have kind learned to live with. Main
>>> one being relating to screen brightness. If I set a low default screen
>>> brightness in Windows then it seems to change the range of brightness
>>> Linux can display. This was even more noticable when running LiveUSB
>>> distros! I actually couldn't get the screen bright enough to see what I
>>> was doing without setting the level within Windows bright again!
>>
>> Is there a setting in regular bios for screen brightness? EFI my leave
>> some variables around that the OS can continue to manipulat after boot...
>> wonderful :P
>>
> No and I've looked again and again thinking it must be settable
> elsewhere if it can be set from the Lenovo software/Windows and affect
> all booted OSes.
>
>>> Then I read (or skimmed) this yesterday and feel even more confused.
>>> https://wiki.archlinux.org/index.php/UEFI
>>
>> I see what you mean, every time I thought I was getting somewhere, I got
>> sent to yet another page. The most I got out of it was that EFI is intel's
>> answer to grub but with more control of the firware settings at the same
>> time. The old bios included calls to access some of the HW, but no one
>> used them as they were not muti-task/user friendly. It appears EFI does
>> the same thing and windows uses it and Linux does not know how to access
>> at least part of it. (or windows sets up its own calls within it)
>>
>>
>>> But I am starting to think I should try enabling it with a Linux
>>> install. Especially as I plan to completely banish Windows from the
>>> computer now... But I thought I would ask what more experienced Linux
>>> users have to say about UEFI mode vs BIOS/Legacy mode.
>>
>> You will get some I don't doubt... it may be hard to tell the truth from
>> FUD  ;)   EFI and Linux are still relatively new (3.10ish) and complaints
>> found in one kernel version may no longer be true.
>>
>> --
>> Len Ovens
>> www.ovenwerks.net
>>
>
> I guess it must tie into ACPI somehow. Or would that be a silly
> assumption? I believe ACPI controls hardware functions, including screen
> brightness and fan controllers, while the system is running. Or am I
> confused? I know on the rare occasions I've had a crash there have been
> lots of ACPI Unknown (or similar) messages in my log! Did mean to look
> into it further at some point...

ACPI handles power-related things. It (or something proprietary) may 
also handle things like the FN key functions on laptops, which include 
brightening, dimming, switching display outputs, on/off/volume of built 
in audio hardware, enabling touchpad, wireless, stuffs like that. Well, 
actually, keyboard driver has to first recognize those keystrokes, then 
the system has to carry out the correct command - which might require 
triggering an ACPI event?

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From gnome at hawaii.rr.com  Sat Aug 30 08:25:38 2014
From: gnome at hawaii.rr.com (david)
Date: Fri, 29 Aug 2014 22:25:38 -1000
Subject: [LAU] KXStudio reminded me of why I don't like ZynAddSubFX
In-Reply-To: <53FEE119.9090909@gmail.com>
References: <53FD9F6A.30805@hawaii.rr.com> <53FDAD8F.8080608@gmail.com>
 <53FEDD2E.9030309@hawaii.rr.com> <53FEE119.9090909@gmail.com>
Message-ID: <54018A82.4010008@hawaii.rr.com>

On 08/27/2014 09:58 PM, Filipe Coelho wrote:
> On 08/28/2014 08:41 AM, david wrote:
>> On 08/27/2014 12:06 AM, Filipe Coelho wrote:
>>> In any case, the Zyn in the kxstudio repos does not auto-connect to
>>> system outputs.
>> Zynn in Debian autoconnects to JACK, but I have to connect it manually
>> (that's the way I like it) within JACK.
>>> This is so it can be safely used in a session manager environment.
>> OK, makes sense. So how DO you connect Zynn using JACK in KXStudio to
>> your desired audio outputs?
>
> You can still use qjackctl, but its jackdbus support is not that great.
> Plus it tends to stop jack when closed...

Hmm, guess that's never bothered me. When I run JACK, I start it with 
QJackCtrl. When I'm done with QJackCtl, I'm also done with JACK. But, 
oddly enough, the times I've forgotten to STOP Jack before closing 
QJackCtl, it left JACK running just fine.

> Cadence includes the Catia and Claudia tools to manage connections (see
> Cadence tools tab).
> Catia is the simple version that only does the basic stuff,
> Claudia is a frontend to LADISH (a session manager) which is obviously a
> bit more complex.
>
> These tools are described into a bit more detail here:
> http://kxstudio.sourceforge.net/Applications

Sounds silly to me. Have to run yet another application to do something 
that QJackCtl does in a subwindow? Although Claudia sounds useful.

>>> The 14.04 ISO is a bit old now.
>>> There has been some serious Cadence improvements and a new Zyn release
>>> since then.
>>> A new ISO will be released in a few days.
>> OK. The one I downloaded was 14.04b.
>
> The new ISO should be released in 2 weeks tops I hope.
> Most things are ready, I just need to update a few packages and do some
> testing.

Will have to try that out. Thanks for updates!

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From kevinc at cosgroves.us  Sat Aug 30 09:26:59 2014
From: kevinc at cosgroves.us (Kevin Cosgrove)
Date: Sat, 30 Aug 2014 02:26:59 -0700
Subject: [LAU] Audio File to Graphic Thumbnail in Command Line?
Message-ID: <20140830092659.4BA84BE05B@joseph.cosgroves.us>


I have a huge number (about 2400) of 0.5GB WAV files from live 24
track recordings.  10-40% of the tracks on some recording sets
are not needed, an unused vocal mic stashed off stage.  I have
been using the RMS function in SOX by command line to help me
determine which files are not useful.  A low RMS level tends to
indicate less usable signal.

But, this doesn't always work.  For instance, it can miss that
great minute long harmonica solo during the hour long show.
Peak levels don't help, because frequently a drummer will slam
something loudly enough to falsely mark the file as useful, when
it's actually not.

Looking at the audio waveform with Ardour or Audacity seems
foolproof.  But, it's also way too much work to use a GUI
application on that many files.

I would like to run a script or program on the set of files and
produce something like an Audacity or Ardour waveform graphic to
be stored in a PNG (or similar) associated file.  Does anyone
know of something close to out-of-the-box ready to do this?

I did find these possibly helpful links.

http://www.network-theory.co.uk/docs/octave3/octave_263.html

https://wiki.ubuntu.com/Waveform_Viewers-Plotting_Large_Analog_Data

Thanks...


--
Kevin



From gheskett at wdtv.com  Sat Aug 30 10:31:36 2014
From: gheskett at wdtv.com (Gene Heskett)
Date: Sat, 30 Aug 2014 06:31:36 -0400
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
Message-ID: <201408300631.36097.gheskett@wdtv.com>

On Thursday 28 August 2014 23:01:10 Sam Mulvey did opine
And Gene did reply:
> On 08/28/2014 07:45 PM, Gene Heskett wrote:
> >> > I thought we were talking about the future here! The 80s wants its
> >> > property back!!
> >> > 
> >> > Also Hi8 is an analogue format so everything in the post is plain
> >> > bollocks! Maybe you meant Digital8?? Still 15 years old and any
> >> > tape format is pretty much dead and definitely not the future!
> > 
> > Not this one, it uses metal tape in the same casette as a Hi-8 would
> > use, but about a tenner more expensive. and is "digital Hi-8"
> > format.
> > 
> > Reasonably sharp too at 720p.  Go look it up, its a Sony HandyCam
> > DCR- TRV460 NTSC. and about 11 years old IIRC.  And one of the first
> > with lithium batteries.  I can't quickly find the charger, but after
> > laying for at least 2 years, it still fires right up.
> > 
> > I have shot several weddings with it, processed it down to fit on a
> > dvd using kino and sold the disks several times now.  Many many
> > times sharper than a vhs deck.
> 
> ..and I think there's a notable point here in the ability to read and
> write to a device on the chain in realtime is a question of bandwidth
> regardless of what the media is.
> 
> -Sam

I will not argue that point Sam.  But I will note that firewire Just Works 
in real time.  I have usb2.0 stuff both here, a veritable weeping willow 
ith everything plugged in and out in the shop with cnc machinery, using a 
usb2.0 camera for machine vision.  And while lsusb -vv says its recognized 
and running as a usb2.0 camera, and it is the only device on that port, 
its nearly worthless because the frame rate for a sustained image on the 
computer is about 2 to 3 frames a second.  Great picture, but you move the 
machine 3 thousandths of an inch, then move your eyes from the keyboard 
back to the monitor so you can check the crosshair registration on the 
work target and still have to wait for the video to catch up.  USB2.0 has 
more than enough bandwidth to handle a 60 fps interlaced pix, according to 
the specs.  I have never ever seen evidence of it.

Nother furinstance, my printer, a Brother HL-3170-CDW color laser, is 
hooked up both ways, a std usb cable in one port, and a 100mbit ethernet 
cable from my cheap switch to the RJ45 on it.  The pages per minute nearly 
doubles if I feed it thru the ethernet port.  Unless its running in duplex 
mode.  Thats a huge time equalizer...

And I found the charger cable, it was laying in plain sight, on the floor 
2 feet from my feet.

Cheers, Gene Heskett
-- 
"There are four boxes to be used in defense of liberty:
 soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
Genes Web page <http://geneslinuxbox.net:6309/gene>
US V Castleman, SCOTUS, Mar 2014 is grounds for Impeaching SCOTUS


Cheers, Gene Heskett
-- 
"There are four boxes to be used in defense of liberty:
 soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
Genes Web page <http://geneslinuxbox.net:6309/gene>
US V Castleman, SCOTUS, Mar 2014 is grounds for Impeaching SCOTUS

From gheskett at wdtv.com  Sat Aug 30 10:32:12 2014
From: gheskett at wdtv.com (Gene Heskett)
Date: Sat, 30 Aug 2014 06:32:12 -0400
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
Message-ID: <201408300632.12520.gheskett@wdtv.com>

On Friday 29 August 2014 03:21:27 Kaza Kore did opine
And Gene did reply:
> > From: gheskett at wdtv.com
> > To: linux-audio-user at lists.linuxaudio.org
> > Date: Thu, 28 Aug 2014 22:45:41 -0400
> > Subject: Re: [LAU] Successor/replacement for RME HDSP+Multiface?
> > 
> > On Thursday 28 August 2014 21:14:48 Kaza Kore did opine
> > 
> > And Gene did reply:
> > > > From: gheskett at wdtv.com
> > > > To: linux-audio-user at lists.linuxaudio.org
> > > > Date: Thu, 28 Aug 2014 20:37:53 -0400
> > > > Subject: Re: [LAU] Successor/replacement for RME HDSP+Multiface?
> > > >
> > > >...Hi-8 tape...
> > > 
> > > I thought we were talking about the future here! The 80s wants its
> > > property back!!
> > > 
> > > Also Hi8 is an analogue format so everything in the post is plain
> > > bollocks! Maybe you meant Digital8?? Still 15 years old and any
> > > tape format is pretty much dead and definitely not the future!
> > 
> > Not this one, it uses metal tape in the same casette as a Hi-8 would
> > use, but about a tenner more expensive. and is "digital Hi-8"
> > format.
> > 
> > Reasonably sharp too at 720p.  Go look it up, its a Sony HandyCam
> > DCR- TRV460 NTSC. and about 11 years old IIRC.
> 
> So not Hi8 then! :p (If you look I did mention Digital8 too.) Not sure
> where you get the idea it's 720P capable! Specs on website state
> 640x480 and you even state in the name you provided it's NTSC, which
> is never 720P, same as PAL and SECAM aren't. They are old, SD
> standards. 720/1080 P/I are very different beasts really.
> 
> Anyway it's probably more important to talk about the standardised DV25
> and DV50 protocol all these commercial/prosumer products use for
> communication that tape/card formats. There are some Sony and
> Panasonic camera that do this fine over USB so it's not impossible or
> a problem with USB itself. I see yours (and apparently many others)
> claim to have some kind of USB Streaming but for some reason it's not
> usually full quality, as you would get from Firewire. Wonder why...

The std says the speed is there.  But on this Asus M2N-SLI Deluxe 
motherboard that cost $287 USD when I bought it, all USB ports claim to be 
USB2.0.  The throughput to/from a hard drive in a self powered usb box 
that I have 2 of, one 40Gb, one 300Gb drive, has a hard time out running a 
floppy disk.  No mistakes ever, but the usable bandwidth simply is not 
there.  My next door neighbor bought one of the 40's the same day I bought 
mine, runs it as a backup on her windows machines.  On her windows boxes, 
it has no problem moving data in either direction at about 50 megabytes a 
second.

A 640x480 USB2.0 camera, plugged into the rear port of a D525MW Atom 
powered board, only make 3 frames a second.

The linux version of USB is a 1 legged dog in comparison.  Why we put up 
with that poor usb performance is beyond me. We had the original USB in 
full usage on linux a good year ahead of the Redmond version, but IMSNHO, 
linux has been sitting on its butt for at least a decade.

What the bloody hell, a copy of the std reference is well within the 
financial reach of both Red Hat and Ubuntu & even SuSe.  But I don't see 
any improvements in the speeds here, and I am currently running a 3.16.0 
kernel on a quad core phenom.

Cheers, Gene Heskett
-- 
"There are four boxes to be used in defense of liberty:
 soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
Genes Web page <http://geneslinuxbox.net:6309/gene>
US V Castleman, SCOTUS, Mar 2014 is grounds for Impeaching SCOTUS


From gheskett at wdtv.com  Sat Aug 30 10:32:26 2014
From: gheskett at wdtv.com (Gene Heskett)
Date: Sat, 30 Aug 2014 06:32:26 -0400
Subject: [LAU] i5 Hyper-Threading, BIOS settings and Arch n00b pointers
Message-ID: <201408300632.26622.gheskett@wdtv.com>

On Friday 29 August 2014 04:33:43 Simon Wise did opine
And Gene did reply:
> On 29/08/14 15:40, Len Ovens wrote:
> > I don't know that the physical technology matters so much as the OS
> > being hyperthreading aware and treating each pair of cores like one.
> > That is making sure that core 0 does not do anything that takes too
> > long for core 1 to meet it's dead line. I do not know if new Linux
> > kernels do this, older ones did not. They logged that the chip had
> > hyperthreading, but still seemed to treat two threads as two
> > different cores.
> > 
> > Certainly, common wisdom has not kept up with tech changes. I would
> > be nice to know more.
> 
> Not quite on topic, since this isn't to do with Hyper-threading, but
> certainly the Linux scheduler has been getting much more sophisticated
> in dealing with different kinds of cores ... in ARM it now schedules
> tasks for chips with some smaller cores and some faster ones, keeping
> them busy with suitable sized tasks.
> 
> The ARM kernels running the most recent Samsung tablets (with 4 big
> plus 4 little cores) have this GTS in the 3.14 kernels ... it runs all
> 8 cores together assigning tasks appropriate to each, rather than just
> switching between big or little of each pair to save power. Selling
> hardware on that scale certainly brings a budget, and since the kernel
> is GPL it can't be kept in-house.
> 
> Seems that 3.14 has also added a deadline-based scheduler that is
> closer to what audio needs from realtime than the extremely low
> latency preemption based on priorities that the two older realtime
> schedulers offer.
> 
> http://www.linuxfoundation.org/news-media/blogs/browse/2014/01/deadline
> -scheduling-314
> 
> Simon

This message is timely as I haven't tried the deadline scheduler in 
several years, so I just switched the config to make it the default, and 
its building now.  Perhaps it will actually improve both the USB lags, and 
the network video playback I get here.

Thank you for an informative post.

Cheers, Gene Heskett
-- 
"There are four boxes to be used in defense of liberty:
 soap, ballot, jury, and ammo. Please use in that order."
-Ed Howdershelt (Author)
Genes Web page <http://geneslinuxbox.net:6309/gene>
US V Castleman, SCOTUS, Mar 2014 is grounds for Impeaching SCOTUS


From mott at reverberant.com  Sat Aug 30 10:38:33 2014
From: mott at reverberant.com (Iain Mott)
Date: Sat, 30 Aug 2014 07:38:33 -0300
Subject: [LAU] html5 audio through jack
Message-ID: <1409395113.3154.5.camel@espelho>


Hi list,

I'm thinking of updating some of my web pages to use multi-platform
flash/html5 audio players, at present they use flash only and won't play
on iPads for example.

Due to some problems I was having with pulse audio in relation to my HDSP
interface I have recently disabled it and all my audio is running via jack/alsa and
the HDSP interface. With flash in firefox, there are no problems and the
audio plays. My .asoundrc is configured with the following:

pcm.rawjack {
    type jack
    playback_ports {
        0 system:playback_1
        1 system:playback_2
    }
    capture_ports {
        0 system:capture_1
        1 system:capture_2
    }
}

pcm.jack {
    type plug
    slave { pcm "rawjack" }
    hint {
 	description "JACK Audio Connection Kit"
    }
}


pcm.!default {
    type plug
    slave { pcm "rawjack" }
}



HTML5 players in firefox don't play however via jack. When pulse was enabled, HTML5
content would play through the computer's built-in sound card. Now that
it's disabled I can't get it to play through jack.

An example page with a HTML5 player is here:

http://www.html5tutorial.info/html5-audio.php

Any suggestions please? A modification of the .asoundrc?

I'm running Ubuntu 14.04

Thanks,


From dj_kaza at hotmail.com  Sat Aug 30 11:17:27 2014
From: dj_kaza at hotmail.com (Kazakore)
Date: Sat, 30 Aug 2014 17:02:27 +0545
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <540187B3.1090504@hawaii.rr.com>
References: <20140827143830.GA7320@aol.com>,
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>,
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>,
 <20140827154520.GB7320@aol.com>,
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>,
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>,
 <20140827192248.GC7320@aol.com> <53FF4A76.8000100@stackingdwarves.net>,
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>,
 <53FF52B0.4030702@stackingdwarves.net> <20140828200726.GA16743@aol.com>,
 <54002437.6080607@hawaii.rr.com> <DUB113-W18E942790CECC11578A7BE8FDB0@phx.gbl>
 <540187B3.1090504@hawaii.rr.com>
Message-ID: <BLU436-SMTP76898D38FEFACB3A4C0CD48FD80@phx.gbl>


>>
>> LOADS of laptops have ExpressCard slots!
>
> Really? Which ones?
>

My Lenovo X230 for starters, plus most the others I looked at in the T 
range at the time (just over a year ago.) My internet is far too slow to 
do any research on today's models. If things truly have changed in just 
the last year them I'm sorry. Didn't seem hard to find when I was 
looking last summer/autumn and that doesn't feel that long ago.

Well the basic T440 doesn't but it's taken me about 10 minutes just to 
check that so really am going to stop now. Hope you find something. 
(Maybe the T540 or T440P??) I remember being surprised when I heard 
Apple had dropped it from the MBP! Seems maybe that was the start of it 
being dropped everywhere... :-(

From dj_kaza at hotmail.com  Sat Aug 30 11:48:46 2014
From: dj_kaza at hotmail.com (Kazakore)
Date: Sat, 30 Aug 2014 17:33:46 +0545
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <201408300632.12520.gheskett@wdtv.com>
References: <201408300632.12520.gheskett@wdtv.com>
Message-ID: <BLU436-SMTP6803C4061141CFBC2ED8FD8FD80@phx.gbl>


On 30/08/14 16:17, Gene Heskett wrote:
> On Friday 29 August 2014 03:21:27 Kaza Kore did opine
> And Gene did reply:
>>> From: gheskett at wdtv.com
>>> To: linux-audio-user at lists.linuxaudio.org
>>> Date: Thu, 28 Aug 2014 22:45:41 -0400
>>> Subject: Re: [LAU] Successor/replacement for RME HDSP+Multiface?
>>>
>>> On Thursday 28 August 2014 21:14:48 Kaza Kore did opine
>>>
>>> And Gene did reply:
>>>>> From: gheskett at wdtv.com
>>>>> To: linux-audio-user at lists.linuxaudio.org
>>>>> Date: Thu, 28 Aug 2014 20:37:53 -0400
>>>>> Subject: Re: [LAU] Successor/replacement for RME HDSP+Multiface?
>>>>>
>>>>> ...Hi-8 tape...
>>>> I thought we were talking about the future here! The 80s wants its
>>>> property back!!
>>>>
>>>> Also Hi8 is an analogue format so everything in the post is plain
>>>> bollocks! Maybe you meant Digital8?? Still 15 years old and any
>>>> tape format is pretty much dead and definitely not the future!
>>> Not this one, it uses metal tape in the same casette as a Hi-8 would
>>> use, but about a tenner more expensive. and is "digital Hi-8"
>>> format.
>>>
>>> Reasonably sharp too at 720p.  Go look it up, its a Sony HandyCam
>>> DCR- TRV460 NTSC. and about 11 years old IIRC.
>> So not Hi8 then! :p (If you look I did mention Digital8 too.) Not sure
>> where you get the idea it's 720P capable! Specs on website state
>> 640x480 and you even state in the name you provided it's NTSC, which
>> is never 720P, same as PAL and SECAM aren't. They are old, SD
>> standards. 720/1080 P/I are very different beasts really.
>>
>> Anyway it's probably more important to talk about the standardised DV25
>> and DV50 protocol all these commercial/prosumer products use for
>> communication that tape/card formats. There are some Sony and
>> Panasonic camera that do this fine over USB so it's not impossible or
>> a problem with USB itself. I see yours (and apparently many others)
>> claim to have some kind of USB Streaming but for some reason it's not
>> usually full quality, as you would get from Firewire. Wonder why...
> The std says the speed is there.  But on this Asus M2N-SLI Deluxe
> motherboard that cost $287 USD when I bought it, all USB ports claim to be
> USB2.0.  The throughput to/from a hard drive in a self powered usb box
> that I have 2 of, one 40Gb, one 300Gb drive, has a hard time out running a
> floppy disk.  No mistakes ever, but the usable bandwidth simply is not
> there.  My next door neighbor bought one of the 40's the same day I bought
> mine, runs it as a backup on her windows machines.  On her windows boxes,
> it has no problem moving data in either direction at about 50 megabytes a
> second.
>
> A 640x480 USB2.0 camera, plugged into the rear port of a D525MW Atom
> powered board, only make 3 frames a second.
>
> The linux version of USB is a 1 legged dog in comparison.  Why we put up
> with that poor usb performance is beyond me. We had the original USB in
> full usage on linux a good year ahead of the Redmond version, but IMSNHO,
> linux has been sitting on its butt for at least a decade.
>
> What the bloody hell, a copy of the std reference is well within the
> financial reach of both Red Hat and Ubuntu & even SuSe.  But I don't see
> any improvements in the speeds here, and I am currently running a 3.16.0
> kernel on a quad core phenom.
>
> Cheers, Gene Heskett

Which is well more than enough for full bandwidth transfer of many video 
channels from a DV type device (which are very similar to your 
Digital8.) They use DV25, which is 25Mbit/s, or there was the DVCPRO50 
at 50Mbit/s and DVCHD which uses 100Mbit/s. The tape storage media 
stores it at these compression rates so there is no way to get better 
quality from them!

The specs of your camera specifically say that the USB is for WebCam 
output though! So I'm not surprised to head it's lower quality and 
framerate. I just don't understand why!

Sony and Panasonic have both implemented it on a very low number of 
their camcorders and as far as I know it works fine. But they require 
special, propriety software so not really any use for the FLOSS world! I 
honestly don't understand why there isn't a more widespread solution, 
especially as firewire seems so hard to get these days, but answers of 
the type "USB isn't up to it" don't bear up to the facts in my opinion.

Dale.


From dj_kaza at hotmail.com  Sat Aug 30 12:03:27 2014
From: dj_kaza at hotmail.com (Kazakore)
Date: Sat, 30 Aug 2014 17:48:27 +0545
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <201408300632.12520.gheskett@wdtv.com>
References: <201408300632.12520.gheskett@wdtv.com>
Message-ID: <BLU437-SMTP10136D77B132F3F480DBB1C8FD80@phx.gbl>


On 30/08/14 16:17, Gene Heskett wrote:
> The linux version of USB is a 1 legged dog in comparison. Why we put 
> up with that poor usb performance is beyond me. We had the original 
> USB in full usage on linux a good year ahead of the Redmond version, 
> but IMSNHO, linux has been sitting on its butt for at least a decade. 
> What the bloody hell, a copy of the std reference is well within the 
> financial reach of both Red Hat and Ubuntu & even SuSe. But I don't 
> see any improvements in the speeds here, and I am currently running a 
> 3.16.0 kernel on a quad core phenom. Cheers, Gene Heskett 

I plugged in a friend's USB3 hard drive to move some files around for 
her couple of days ago and was WOW'd by the speeds to be honest! And I'm 
running spinning disks internally, no SSD. We accidental grabbed a near 
6GB folder and it had done almost half of it by the time we had noticed 
and decided to cancel it rather than copy it all. Never seen data move 
so fast outside of RAID! :)

(Don't know what happened to the whitespace in my quote. Not sure about 
thunderbird... :( )

From dj_kaza at hotmail.com  Sat Aug 30 12:09:18 2014
From: dj_kaza at hotmail.com (Kazakore)
Date: Sat, 30 Aug 2014 17:54:18 +0545
Subject: [LAU] html5 audio through jack
In-Reply-To: <1409395113.3154.5.camel@espelho>
References: <1409395113.3154.5.camel@espelho>
Message-ID: <BLU437-SMTP21B335DEAC0E6903E9270E8FD80@phx.gbl>


On 30/08/14 16:23, Iain Mott wrote:
> Hi list,
>
> I'm thinking of updating some of my web pages to use multi-platform
> flash/html5 audio players, at present they use flash only and won't play
> on iPads for example.
>
> Due to some problems I was having with pulse audio in relation to my HDSP
> interface I have recently disabled it and all my audio is running via jack/alsa and
> the HDSP interface. With flash in firefox, there are no problems and the
> audio plays. My .asoundrc is configured with the following:
>
> pcm.rawjack {
>      type jack
>      playback_ports {
>          0 system:playback_1
>          1 system:playback_2
>      }
>      capture_ports {
>          0 system:capture_1
>          1 system:capture_2
>      }
> }
>
> pcm.jack {
>      type plug
>      slave { pcm "rawjack" }
>      hint {
>   	description "JACK Audio Connection Kit"
>      }
> }
>
>
> pcm.!default {
>      type plug
>      slave { pcm "rawjack" }
> }
>
>
>
> HTML5 players in firefox don't play however via jack. When pulse was enabled, HTML5
> content would play through the computer's built-in sound card. Now that
> it's disabled I can't get it to play through jack.
>
> An example page with a HTML5 player is here:
>
> http://www.html5tutorial.info/html5-audio.php
>
> Any suggestions please? A modification of the .asoundrc?
>
> I'm running Ubuntu 14.04
>
> Thanks,
>
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

Can't help but can tell you I have the exact same problem! I also don't 
get audio from streaming media directly from the internet (eg a web 
hosted .ogg) Have you managed to get that working? I use VLC's 
Multimedia Plugin for firefox, VLC is set to Jack, also have all 
GStreamer routed through Jack, but "normal" firefox audio never reaches it.

Good luck with this one. It's part of the reason Arch is feeling so 
appealing to me right now. I can truly get a system with no P(IT)A in 
there! ;-)

Dale.


From dj_kaza at hotmail.com  Sat Aug 30 12:15:19 2014
From: dj_kaza at hotmail.com (Kazakore)
Date: Sat, 30 Aug 2014 18:00:19 +0545
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <BLU436-SMTP76898D38FEFACB3A4C0CD48FD80@phx.gbl>
References: <20140827143830.GA7320@aol.com>,
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>,
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>,
 <20140827154520.GB7320@aol.com>,
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>,
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>,
 <20140827192248.GC7320@aol.com> <53FF4A76.8000100@stackingdwarves.net>,
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>,
 <53FF52B0.4030702@stackingdwarves.net> <20140828200726.GA16743@aol.com>,
 <54002437.6080607@hawaii.rr.com> <DUB113-W18E942790CECC11578A7BE8FDB0@phx.gbl>
 <540187B3.1090504@hawaii.rr.com>
 <BLU436-SMTP76898D38FEFACB3A4C0CD48FD80@phx.gbl>
Message-ID: <BLU436-SMTP158C7907D3B0163A6B68E6A8FD80@phx.gbl>


On 30/08/14 17:02, Kazakore wrote:
>
>>>
>>> LOADS of laptops have ExpressCard slots!
>>
>> Really? Which ones?
>>
>
> My Lenovo X230 for starters, plus most the others I looked at in the T 
> range at the time (just over a year ago.) My internet is far too slow 
> to do any research on today's models. If things truly have changed in 
> just the last year them I'm sorry. Didn't seem hard to find when I was 
> looking last summer/autumn and that doesn't feel that long ago.
>
> Well the basic T440 doesn't but it's taken me about 10 minutes just to 
> check that so really am going to stop now. Hope you find something. 
> (Maybe the T540 or T440P??) I remember being surprised when I heard 
> Apple had dropped it from the MBP! Seems maybe that was the start of 
> it being dropped everywhere... :-(


Seems the latest range have got rid of them but I can make you a 
suggested. Get a Refurbished one from Lenovo Outlet Store. I went on the 
assumption that there is a greater chance of you being in the States 
than the UK so checked the store for there and both the X230 and T530 
were listed as available and both have ExpressCard.

My X230 was from the UK Outlet store. Was quite lucky I think. About 
40-50% reduction when ever other laptop they had listed was only 10% or 
a bit more. I have my suspicions (hopes) a bulk order was cancelled and 
thus they had a lot of them and they price went cheap. Good thing about 
the X230 is that it's about the only ThinkPad model I found that goes 
through the same tests as the T-series ("MilSpec Tested") :-)

Hope this is of some help. Dale.

From dlphillips at woh.rr.com  Sat Aug 30 12:43:15 2014
From: dlphillips at woh.rr.com (Dave Phillips)
Date: Sat, 30 Aug 2014 08:43:15 -0400
Subject: [LAU] A MIDI tone editor for the Roland MKS-70 Super JX analog
 synthesizer
In-Reply-To: <1409382448.2395.1.camel@yoda.heavyware>
References: <1409382448.2395.1.camel@yoda.heavyware>
Message-ID: <5401C6E3.5070605@woh.rr.com>


On 08/30/2014 03:07 AM, Guido Aulisi wrote:
> Hello,
> this is my attempt to build a GTK+ (really GTKmm) C++ application to
> edit patches on the Roland MKS 70.
>
> Code is on github: https://github.com/tartina/pg800.git
>
> Tarballs and (maybe) RPM packages will come... some day!
>
> Ciao
> Guido
>
>

Greetings,

I'm glad you wrote an editor for it, but alas, I sold my MKS-70 years 
ago. :(

It is a great hardware synth, one of Roland's best.

Best regards,

dp


From rob at rektau.ukfsn.org  Sat Aug 30 17:03:57 2014
From: rob at rektau.ukfsn.org (rob)
Date: Sat, 30 Aug 2014 18:03:57 +0100
Subject: [LAU] html5 audio through jack
In-Reply-To: <BLU437-SMTP21B335DEAC0E6903E9270E8FD80@phx.gbl>
References: <1409395113.3154.5.camel@espelho>
 <BLU437-SMTP21B335DEAC0E6903E9270E8FD80@phx.gbl>
Message-ID: <540203FD.5070803@rektau.ukfsn.org>

On 30/08/14 13:09, Kazakore wrote:
>
> On 30/08/14 16:23, Iain Mott wrote:
>> Hi list,
>>
>> I'm thinking of updating some of my web pages to use multi-platform
>> flash/html5 audio players, at present they use flash only and won't play
>> on iPads for example.
>>
>> Due to some problems I was having with pulse audio in relation to my HDSP
>> interface I have recently disabled it and all my audio is running via
>> jack/alsa and
>> the HDSP interface. With flash in firefox, there are no problems and the
>> audio plays. My .asoundrc is configured with the following:
>>
>> pcm.rawjack {
>>      type jack
>>      playback_ports {
>>          0 system:playback_1
>>          1 system:playback_2
>>      }
>>      capture_ports {
>>          0 system:capture_1
>>          1 system:capture_2
>>      }
>> }
>>
>> pcm.jack {
>>      type plug
>>      slave { pcm "rawjack" }
>>      hint {
>>       description "JACK Audio Connection Kit"
>>      }
>> }
>>
>>
>> pcm.!default {
>>      type plug
>>      slave { pcm "rawjack" }
>> }
>>
>>
>>
>> HTML5 players in firefox don't play however via jack. When pulse was
>> enabled, HTML5
>> content would play through the computer's built-in sound card. Now that
>> it's disabled I can't get it to play through jack.
>>
>> An example page with a HTML5 player is here:
>>
>> http://www.html5tutorial.info/html5-audio.php
>>
>> Any suggestions please? A modification of the .asoundrc?
>>
>> I'm running Ubuntu 14.04
>>
>> Thanks,
>>
>> _______________________________________________
>> Linux-audio-user mailing list
>> Linux-audio-user at lists.linuxaudio.org
>> http://lists.linuxaudio.org/listinfo/linux-audio-user
>
> Can't help but can tell you I have the exact same problem! I also don't
> get audio from streaming media directly from the internet (eg a web
> hosted .ogg) Have you managed to get that working? I use VLC's
> Multimedia Plugin for firefox, VLC is set to Jack, also have all
> GStreamer routed through Jack, but "normal" firefox audio never reaches it.
>
> Good luck with this one. It's part of the reason Arch is feeling so
> appealing to me right now. I can truly get a system with no P(IT)A in
> there! ;-)
>
> Dale.
>
> _______________________________________________

http://alsa.opensrc.org/Jack_and_Loopback_device_as_Alsa-to-Jack_bridge 
might be worth looking at.

rob



From reuben.m at gmail.com  Sat Aug 30 18:08:09 2014
From: reuben.m at gmail.com (Reuben Martin)
Date: Sat, 30 Aug 2014 13:08:09 -0500
Subject: [LAU] i5 Hyper-Threading, BIOS settings and Arch n00b pointers
In-Reply-To: <20140829094725.GA32083@linuxaudio.org>
References: <DUB113-W56B49B57136F35F658C12F8FDA0@phx.gbl>
 <54003AE7.1010304@gmail.com> <20140829094725.GA32083@linuxaudio.org>
Message-ID: <3192050.vkUy8XFDQq@subterfuge>

On Friday, August 29, 2014 09:47:25 AM Fons Adriaensen wrote:
> On Fri, Aug 29, 2014 at 06:33:43PM +1000, Simon Wise wrote:
> > Seems that 3.14 has also added a deadline-based scheduler that is
> > closer to what audio needs from realtime than the extremely low
> > latency preemption based on priorities that the two older realtime
> > schedulers offer.
> > 
> > http://www.linuxfoundation.org/news-media/blogs/browse/2014/01/deadline-sc
> > heduling-314
> Not really.
> 
> The new scheduler is designed to run a set of periodic tasks,
> with arbitrary periods and having few or no dependencies between
> them, at up to 100% CPU usage, while meeting their deadlines.
> 
> Rate-proportional scheduling (shorter period -> higher priority)
> can only do this up to 69% CPU use, except in some special cases
> such as all periods being equal or having simple integer ratios
> between them.
> 

What about pinning a core (or cores) to audio processing? If the system is 
truly being stressed to the point that it has become challenging for the 
scheduler to keep up, wouldn't pinning make more sense?

-Reuben

From list at nilsgey.de  Sat Aug 30 19:50:16 2014
From: list at nilsgey.de (Nils)
Date: Sat, 30 Aug 2014 21:50:16 +0200
Subject: [LAU] Status of lisalo/lisaloQt and calfbox?
In-Reply-To: <5e8f2428ae0d4fc08d83cdebd6fcb69e@Ex13DAG10-N2.dataoncloud.net>
References: <5e8f2428ae0d4fc08d83cdebd6fcb69e@Ex13DAG10-N2.dataoncloud.net>
Message-ID: <54022AF8.8000309@nilsgey.de>

Hello Jonathan,

1) there will be never again a program of mine based on Linuxsampler. So 
if you hope that I reupload the old CLI lisalo which was for LS: no. 
Reason: Linuxsampler has a bad license. don't support it.

2) I took the prototypes based on calfbox down not because I don't want 
to work on them anymore but I want to release  proper software, not 
low-quality, half-done stuff. I still need a sampler myself, but this is 
not a trivial piece of software.

3) My current life planning is that I'm done with my thesis in October 
and then I choose what to do with my programming time. There are three 
options: laborejo 2, the sampler or a commercial screencasting helper/tool.
Everything that involves getting money is quite convincing to me right 
now, I have to say.

Nils
http://www.nilsgey.de

On 29.08.2014 06:57, Jonathan E Brickman wrote:
> Have just read a bit of Lisalo/LisaloQt, front end for calfbox; what 
> is the status of this project?  Also, are there usage examples for 
> calfbox I can learn from?
>
> -- 
> Jonathan E. Brickman
> Ponderworthy Music | jeb at ponderworthy.com | (785)233-9977 | 
> http://ponderworthy.com
>
>
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user

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From dj_kaza at hotmail.com  Sat Aug 30 20:12:04 2014
From: dj_kaza at hotmail.com (Kazakore)
Date: Sun, 31 Aug 2014 01:57:04 +0545
Subject: [LAU] html5 audio through jack
In-Reply-To: <540203FD.5070803@rektau.ukfsn.org>
References: <1409395113.3154.5.camel@espelho>
 <BLU437-SMTP21B335DEAC0E6903E9270E8FD80@phx.gbl>
 <540203FD.5070803@rektau.ukfsn.org>
Message-ID: <BLU436-SMTP227E4E9F322A3846F31D45F8FD80@phx.gbl>


On 30/08/14 22:48, rob wrote:
> On 30/08/14 13:09, Kazakore wrote:
>>
>> On 30/08/14 16:23, Iain Mott wrote:
>>> Hi list,
>>>
>>> I'm thinking of updating some of my web pages to use multi-platform
>>> flash/html5 audio players, at present they use flash only and won't 
>>> play
>>> on iPads for example.
>>>
>>> Due to some problems I was having with pulse audio in relation to my 
>>> HDSP
>>> interface I have recently disabled it and all my audio is running via
>>> jack/alsa and
>>> the HDSP interface. With flash in firefox, there are no problems and 
>>> the
>>> audio plays. My .asoundrc is configured with the following:
>>>
>>> pcm.rawjack {
>>>      type jack
>>>      playback_ports {
>>>          0 system:playback_1
>>>          1 system:playback_2
>>>      }
>>>      capture_ports {
>>>          0 system:capture_1
>>>          1 system:capture_2
>>>      }
>>> }
>>>
>>> pcm.jack {
>>>      type plug
>>>      slave { pcm "rawjack" }
>>>      hint {
>>>       description "JACK Audio Connection Kit"
>>>      }
>>> }
>>>
>>>
>>> pcm.!default {
>>>      type plug
>>>      slave { pcm "rawjack" }
>>> }
>>>
>>>
>>>
>>> HTML5 players in firefox don't play however via jack. When pulse was
>>> enabled, HTML5
>>> content would play through the computer's built-in sound card. Now that
>>> it's disabled I can't get it to play through jack.
>>>
>>> An example page with a HTML5 player is here:
>>>
>>> http://www.html5tutorial.info/html5-audio.php
>>>
>>> Any suggestions please? A modification of the .asoundrc?
>>>
>>> I'm running Ubuntu 14.04
>>>
>>> Thanks,
>>>
>>> _______________________________________________
>>> Linux-audio-user mailing list
>>> Linux-audio-user at lists.linuxaudio.org
>>> http://lists.linuxaudio.org/listinfo/linux-audio-user
>>
>> Can't help but can tell you I have the exact same problem! I also don't
>> get audio from streaming media directly from the internet (eg a web
>> hosted .ogg) Have you managed to get that working? I use VLC's
>> Multimedia Plugin for firefox, VLC is set to Jack, also have all
>> GStreamer routed through Jack, but "normal" firefox audio never 
>> reaches it.
>>
>> Good luck with this one. It's part of the reason Arch is feeling so
>> appealing to me right now. I can truly get a system with no P(IT)A in
>> there! ;-)
>>
>> Dale.
>>
>> _______________________________________________
>
> http://alsa.opensrc.org/Jack_and_Loopback_device_as_Alsa-to-Jack_bridge might 
> be worth looking at.
>
> rob
>
>
> _______________________________________________
>

I'm sure it used to be at the bottom of this page it had a note about 
how going the ALSA Loopback route would stop you being able to change 
settings from within QJackCtl, and just at the same point I read it 
there was a thread on here by somebody not being able to set up Jack 
through QJackCtl which at the time I assumed must be related. Can't see 
any comment on it anywhere at all now though...

http://jackaudio.org/faq/routing_alsa.html

But obviously that would be unacceptable and why I never tried 
re-routing ALSA.

Dale.

From bruviaro at scu.edu  Sun Aug 31 00:16:26 2014
From: bruviaro at scu.edu (Bruno Ruviaro)
Date: Sat, 30 Aug 2014 17:16:26 -0700
Subject: [LAU] KXStudio reminded me of why I don't like ZynAddSubFX
In-Reply-To: <54018A82.4010008@hawaii.rr.com>
References: <53FD9F6A.30805@hawaii.rr.com> <53FDAD8F.8080608@gmail.com>
 <53FEDD2E.9030309@hawaii.rr.com> <53FEE119.9090909@gmail.com>
 <54018A82.4010008@hawaii.rr.com>
Message-ID: <CALR3z0hk65obzwukUUxmZv_Egm=FX99_+7-j1mpgcY=TgqbdSw@mail.gmail.com>

On Sat, Aug 30, 2014 at 1:25 AM, david <gnome at hawaii.rr.com> wrote:

> ?[snip]
>
>  Cadence includes the Catia and Claudia tools to manage connections (see
>> Cadence tools tab).
>> Catia is the simple version that only does the basic stuff,
>> Claudia is a frontend to LADISH (a session manager) which is obviously a
>> bit more complex.
>>
>> These tools are described into a bit more detail here:
>> http://kxstudio.sourceforge.net/Applications
>>
>
> Sounds silly to me. Have to run yet another application to do something
> that QJackCtl does in a subwindow? Although Claudia sounds useful.
>
>
?It's not silly, it's just proposing a different workflow. One of the cool
things about Cadence (for me at least) you can easily set it up to start
jack by default when you login -- you don't have to open anything in the
next login, not even Cadence itself. Thanks to the available bridges (which
also can auto-start), you can also have a2jmidi and pulseaudio jack sink
starting and running automatically.

In short, it is one *less* application to open -- I simply no longer need
QJackCtl in most cases. And whenever I need to connect something manually,
*then* I open Catia, the simple interface that is there just to for that
purpose.

This set-up is particularly useful when I'm dealing with a group of ten or
more students in a class or workshop, with 99% of people trying Linux for
the first time. It is so much easier to be able to go straight to
SuperCollider or Ardour or Hydrogen, bypassing completely any initial set
up or talk about Jack -- especially when time is so short. Later, when more
complex routing needs arise, that's often the best time to introduce the
talk about Jack.

My 2 cents,

Bruno
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From bruviaro at scu.edu  Sun Aug 31 00:28:04 2014
From: bruviaro at scu.edu (Bruno Ruviaro)
Date: Sat, 30 Aug 2014 17:28:04 -0700
Subject: [LAU] Session management with NSM
In-Reply-To: <CAKudYbNfJjNSJfHRgRSz7NX7O2Bd9=gyhAsR+qAwZf7Ofw5cwA@mail.gmail.com>
References: <20140828224247.236111e1@eeyore>
 <CAKudYbP4LiQDs4kWgU3a_YKCM9HVjmS3qO+hngNOky91r0XOUQ@mail.gmail.com>
 <20140829213201.25bca02a@eeyore.mozart.uni-klu.ac.at>
 <CAKudYbNfJjNSJfHRgRSz7NX7O2Bd9=gyhAsR+qAwZf7Ofw5cwA@mail.gmail.com>
Message-ID: <CALR3z0h9D_o+wrvkF1GW13ZdV1bbdsQDxRe1F3iSpztkcz=_0g@mail.gmail.com>

Just cheering for NSM development here... ;-)

I'd love to see it grow and become fully supported by even more
applications. I'm not a developer so I can't help on that end, but I'm
thinking about creating some online tutorials and guides. Anything else I
could help with, let me know.

Bruno


On Fri, Aug 29, 2014 at 4:01 PM, Harry van Haaren <harryhaaren at gmail.com>
wrote:

> On Fri, Aug 29, 2014 at 8:32 PM, Philipp ?berbacher <murks at tuxfamily.org>
> wrote:
> > Thanks a lot for your reply Harry.
> Cheers, be careful to not remove the list from replies: its good to
> keep everything in the archives for future reference :)
>
> >> That's the correct way to handle this, as far as I know. Its useful to
> >> have different directories on one system: it allows subdiving your
> >> available sessions into groups like "albums" or
> >> "projects-with-certain-people". Although I agree it feels a little
> >> clunky, its quite powerful and useful.
> >
> > There could also be a subdivision in the NSM GUI. Well, the current way
> > is certainly the simpler implementation, not sure it's simpler for the
> > users :)
>
> Sure, and my original suggestion was a "stepping-stone" type idea,
> with hopes to improve the workflow furthur, once this has become the
> "biggest" issue NSM has :D
>
> >> > 2. Adding programs to sessions through the GUI ("Add Client to
> >> > Session") is the only way? Is there no way to attach running clients
> >> > or at least have some comfort like tab completion to add clients?
> >> NSM does not support this "attach" workflow, but tab completion or a
> >> list of available (fully supported) NSM clients would be a good
> >> improvement on workflow. This should be discussed as to how best
> >> implement it: i'm not sure.
> >
> > Right, a list of supported Clients would also be nice, however, I see
> > two problems:
> > 1. The list would need to be updated somehow, and even then it would be
> > a bit problematic because different distributions ship different
> > versions of the software. NSM might already list a program as supported
> > while the installed version of the program does not yet support NSM.
> > 2. The other programs, audio or just related, should ideally also be
> > listed, and that task is impossible.
>
> Actually this might be possible to solve with a "packaging" trick as
> such: have programs install a file into a specific location (that is
> currently *not* used by any program) to denote its NSM support. I'll
> suggest installing a file in /usr/share/nsm/ , and if there's a file
> there, then the filename without extension represents that a program
> is capable of NSM. This would require *all* NSM clients to explicitly
> add an NSM file.
>
> Perhaps other developers more involved in packaging /
> "feature-announcing" will have a better idea here, I'm all ears, my
> suggestion above is just that: a suggestion.
>
> >> > 3. Jack and NSM. How do you handle that? It is possible to start
> >> > jack through NSM proxy and I guess it is OK to do that as long as
> >> > jack reliably starts before jackpatch (something I'm not sure of).
> >> > First I had just jackpatch in there and it started jack for me with
> >> > a whole lot of options that are unfamiliar to me and probably not
> >> > needed.
> >>
> >> I imagine that NSM will launch said JACK apps, and if one is set to
> >> "start JACK" on jack_client_open() in its code, then it will start
> >> JACK with the settings in  ~/.jackdrc   Perhaps the inclusion of a
> >> "Start JACK" type client with particular settings can be implemented
> >> in order to handle this? I'm open for suggestions too.
> >
> > That seems to be what happens, and its a race. In my experience
> > jackpatch wins the race against jackd, so I have to start jack before
> > the session.
> > A start_jack client could be useful, but from what I have seen all we
> > really need is the possibility to start a client before the others.
> > The simple way would be a timeout, but you'd still have the
> > race. Ideally there would be some way to tell NSM that jack has
> > started and is ready. I have doubts that this is possible with plain
> > jack1 and NSM proxy, maybe a special start_jack client could help here.
>
> NSM doesn't *explicitly* require JACK to be running actually: its
> probably its most common use right now, but setting an explicit
> dependency on JACK should be avoided. Perhaps a flag could be
> introduce on a per-client basis, that represents
> "start-before-others". This way, a "jackd" or "start-jack" client can
> be loaded before the rest. Or even two or more "before-others" clients
> could set up whatever needs setting up, before "normal-time" NSM
> clients are loaded.
>
> Again, welcome input from users / devs.
>
> >> > 4. CLI clients. Are they generally not supported? I added the lv2
> >> > host that was recommended to me (jalv) and had to do that through
> >> > the NSM proxy, so the settings won't be saved even though the
> >> > plugin (fabla in this case) can save its settings. This sort of
> >> > defeats session management. With all the CLI tools we have it would
> >> > be a pitty if that was generally not supported. On a sidenote, can
> >> > someone recommend a plugin host that is supported?
> >>
> >> CLI clients are supported just like clients with a GUI, there is no
> >> difference to NSM. The issue you're encountering here is that JALV
> >> currently doesn't support NSM, which is something that I agree needs
> >> fixing. I'll put JALV NSM support on the TODO, its something I've
> >> lacked myself too.
> >
> > Ok, great. Does a CLI NSM client exist that I can try?
>
> None that I know of right now: Indeed JALV needs NSM, and jalv (the
> command line version) will then be such a client.
>
> > I also noticed that JALV keeps hanging around
> > after I close the session it is part of, is that expected behavior?
>
> This can be fixed by sending the "SIGTERM" in the lower part of the
> "nsm-proxy" configuration dialog (where you fill in "jalv.gtk", and
> the arguments to load a certain plugin).
>
> >> > Well, that's it for now. Last time I heard about NSM I got the
> >> > impression that it takes care of session management once and for
> >> > all, but the first half our gave me a different impression.
> >> OpenAV stands behind NSM: I'm willing to do my best to cooperate with
> >> project developers to implement NSM in various programs, and improve
> >> the workflow of session management.
> >>
> >> If there's any furthur questions, please ask, in the mean time, I'll
> >> try code up some NSM :) -Harry
> >
> > Thanks a lot for your help Harry, we have used crutches for session
> > management long enough.
>
> Agreed, lets try fix this together with the communit in the next
> weeks, and never look back ;)
> Cheers, -Harry
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user
>
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From dj_kaza at hotmail.com  Sun Aug 31 01:24:46 2014
From: dj_kaza at hotmail.com (Kazakore)
Date: Sun, 31 Aug 2014 07:09:46 +0545
Subject: [LAU] KXStudio reminded me of why I don't like ZynAddSubFX
In-Reply-To: <CALR3z0hk65obzwukUUxmZv_Egm=FX99_+7-j1mpgcY=TgqbdSw@mail.gmail.com>
References: <53FD9F6A.30805@hawaii.rr.com> <53FDAD8F.8080608@gmail.com>
 <53FEDD2E.9030309@hawaii.rr.com> <53FEE119.9090909@gmail.com>
 <54018A82.4010008@hawaii.rr.com>
 <CALR3z0hk65obzwukUUxmZv_Egm=FX99_+7-j1mpgcY=TgqbdSw@mail.gmail.com>
Message-ID: <BLU436-SMTP161BA69886F0DE42AFA9E748FD90@phx.gbl>


On 31/08/14 06:01, Bruno Ruviaro wrote:
> On Sat, Aug 30, 2014 at 1:25 AM, david <gnome at hawaii.rr.com 
> <mailto:gnome at hawaii.rr.com>> wrote:
>
>     ?[snip]
>
>         Cadence includes the Catia and Claudia tools to manage
>         connections (see
>         Cadence tools tab).
>         Catia is the simple version that only does the basic stuff,
>         Claudia is a frontend to LADISH (a session manager) which is
>         obviously a
>         bit more complex.
>
>         These tools are described into a bit more detail here:
>         http://kxstudio.sourceforge.net/Applications
>
>
>     Sounds silly to me. Have to run yet another application to do
>     something that QJackCtl does in a subwindow? Although Claudia
>     sounds useful.
>
>
> ?It's not silly, it's just proposing a different workflow. One of the 
> cool things about Cadence (for me at least) you can easily set it up 
> to start jack by default when you login -- you don't have to open 
> anything in the next login, not even Cadence itself. Thanks to the 
> available bridges (which also can auto-start), you can also have 
> a2jmidi and pulseaudio jack sink starting and running automatically.
>
>

And you can have QJackCtl automatically start when you login as well 
(which is what I do.) I still can't see any reason why Cadence is an 
improvement.... (Not arguing, trying to understand.)

Dale.
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From gnome at hawaii.rr.com  Sun Aug 31 01:45:54 2014
From: gnome at hawaii.rr.com (david)
Date: Sat, 30 Aug 2014 15:45:54 -1000
Subject: [LAU] KXStudio reminded me of why I don't like ZynAddSubFX
In-Reply-To: <BLU436-SMTP161BA69886F0DE42AFA9E748FD90@phx.gbl>
References: <53FD9F6A.30805@hawaii.rr.com> <53FDAD8F.8080608@gmail.com>
 <53FEDD2E.9030309@hawaii.rr.com> <53FEE119.9090909@gmail.com>
 <54018A82.4010008@hawaii.rr.com>
 <CALR3z0hk65obzwukUUxmZv_Egm=FX99_+7-j1mpgcY=TgqbdSw@mail.gmail.com>
 <BLU436-SMTP161BA69886F0DE42AFA9E748FD90@phx.gbl>
Message-ID: <54027E52.6020709@hawaii.rr.com>

On 08/30/2014 03:24 PM, Kazakore wrote:
>
> On 31/08/14 06:01, Bruno Ruviaro wrote:
>> On Sat, Aug 30, 2014 at 1:25 AM, david <gnome at hawaii.rr.com
>> <mailto:gnome at hawaii.rr.com>> wrote:
>>
>>     ?[snip]
>>
>>         Cadence includes the Catia and Claudia tools to manage
>>         connections (see
>>         Cadence tools tab).
>>         Catia is the simple version that only does the basic stuff,
>>         Claudia is a frontend to LADISH (a session manager) which is
>>         obviously a
>>         bit more complex.
>>
>>         These tools are described into a bit more detail here:
>>         http://kxstudio.sourceforge.net/Applications
>>
>>
>>     Sounds silly to me. Have to run yet another application to do
>>     something that QJackCtl does in a subwindow? Although Claudia
>>     sounds useful.
>>
>>
>> ?It's not silly, it's just proposing a different workflow. One of the
>> cool things about Cadence (for me at least) you can easily set it up
>> to start jack by default when you login -- you don't have to open
>> anything in the next login, not even Cadence itself. Thanks to the
>> available bridges (which also can auto-start), you can also have
>> a2jmidi and pulseaudio jack sink starting and running automatically.
>
> And you can have QJackCtl automatically start when you login as well
> (which is what I do.) I still can't see any reason why Cadence is an
> improvement.... (Not arguing, trying to understand.)

I found Cadence confusing in that area. I think it should default to 
maybe the simple connection tool running inside the Cadence window, with 
an option setting to make the advanced one the default. Or something 
like that.

My computers are all general-purpose ones, not audio-dedicated, so I 
don't want JACK starting automatically. But session management would 
speed things up.

One thing I liked about Musix 2 were the demos it had setup. You clicked 
an icon, it started up a bunch of programs (for example, JACK, Hydrogen, 
Qsynth or Zyn, and Rosegarden with a composition open), and everything 
would start playing when you hit play in RG's transport ... I think they 
do it through scripts?

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From gnome at hawaii.rr.com  Sun Aug 31 02:17:46 2014
From: gnome at hawaii.rr.com (david)
Date: Sat, 30 Aug 2014 16:17:46 -1000
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <BLU436-SMTP158C7907D3B0163A6B68E6A8FD80@phx.gbl>
References: <20140827143830.GA7320@aol.com>,
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>,
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>,
 <20140827154520.GB7320@aol.com>,
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>,
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>,
 <20140827192248.GC7320@aol.com> <53FF4A76.8000100@stackingdwarves.net>,
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>,
 <53FF52B0.4030702@stackingdwarves.net> <20140828200726.GA16743@aol.com>,
 <54002437.6080607@hawaii.rr.com> <DUB113-W18E942790CECC11578A7BE8FDB0@phx.gbl>
 <540187B3.1090504@hawaii.rr.com>
 <BLU436-SMTP76898D38FEFACB3A4C0CD48FD80@phx.gbl>
 <BLU436-SMTP158C7907D3B0163A6B68E6A8FD80@phx.gbl>
Message-ID: <540285CA.3070507@hawaii.rr.com>

On 08/30/2014 02:15 AM, Kazakore wrote:
>
> On 30/08/14 17:02, Kazakore wrote:
>>
>>>>
>>>> LOADS of laptops have ExpressCard slots!
>>>
>>> Really? Which ones?
>>>
>>
>> My Lenovo X230 for starters, plus most the others I looked at in the T
>> range at the time (just over a year ago.) My internet is far too slow
>> to do any research on today's models. If things truly have changed in
>> just the last year them I'm sorry. Didn't seem hard to find when I was
>> looking last summer/autumn and that doesn't feel that long ago.
>>
>> Well the basic T440 doesn't but it's taken me about 10 minutes just to
>> check that so really am going to stop now. Hope you find something.
>> (Maybe the T540 or T440P??) I remember being surprised when I heard
>> Apple had dropped it from the MBP! Seems maybe that was the start of
>> it being dropped everywhere... :-(
>
>
> Seems the latest range have got rid of them but I can make you a
> suggested. Get a Refurbished one from Lenovo Outlet Store. I went on the
> assumption that there is a greater chance of you being in the States
> than the UK so checked the store for there and both the X230 and T530
> were listed as available and both have ExpressCard.
>
> My X230 was from the UK Outlet store. Was quite lucky I think. About
> 40-50% reduction when ever other laptop they had listed was only 10% or
> a bit more. I have my suspicions (hopes) a bulk order was cancelled and
> thus they had a lot of them and they price went cheap. Good thing about
> the X230 is that it's about the only ThinkPad model I found that goes
> through the same tests as the T-series ("MilSpec Tested") :-)
>
> Hope this is of some help. Dale.

Thanks, sounds useful. Looks like their Outlet Stores are Europe only, 
don't see USA as an option.

I glanced at their current T540p model; no ExpressCard. T440, no EC.

Toshiba - can't tell.

Looks to me like ExpressCard is dead.

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From jeb at ponderworthy.com  Sun Aug 31 02:35:44 2014
From: jeb at ponderworthy.com (Jonathan E Brickman)
Date: Sun, 31 Aug 2014 02:35:44 +0000
Subject: [LAU] Status of lisalo/lisaloQt and calfbox?
References: <5e8f2428ae0d4fc08d83cdebd6fcb69e@Ex13DAG10-N2.dataoncloud.net>
 <54022AF8.8000309@nilsgey.de>
Message-ID: <8bdb7c8f6f8f4df2b119596289a827c9@Ex13DAG10-N1.dataoncloud.net>

Thank you, Nils.  Will be keeping my eyes open for your work :-)  Am not sure I am up to implementing even a simple SFZ test with calfbox, it is clearly a very powerful and flexible component, but am going to try again using its samples.

J.E.B.

On 08/30/2014 02:50 PM, Nils wrote:
Hello Jonathan,

1) there will be never again a program of mine based on Linuxsampler. So if you hope that I reupload the old CLI lisalo which was for LS: no. Reason: Linuxsampler has a bad license. don't support it.

2) I took the prototypes based on calfbox down not because I don't want to work on them anymore but I want to release  proper software, not low-quality, half-done stuff. I still need a sampler myself, but this is not a trivial piece of software.

3) My current life planning is that I'm done with my thesis in October and then I choose what to do with my programming time. There are three options: laborejo 2, the sampler or a commercial screencasting helper/tool.
Everything that involves getting money is quite convincing to me right now, I have to say.

Nils
http://www.nilsgey.de

On 29.08.2014 06:57, Jonathan E Brickman wrote:
Have just read a bit of Lisalo/LisaloQt, front end for calfbox; what is the status of this project?  Also, are there usage examples for calfbox I can learn from?

--
Jonathan E. Brickman
Ponderworthy Music | jeb at ponderworthy.com<mailto:jeb at ponderworthy.com> | (785)233-9977 | http://ponderworthy.com



_______________________________________________
Linux-audio-user mailing list
Linux-audio-user at lists.linuxaudio.org<mailto:Linux-audio-user at lists.linuxaudio.org>
http://lists.linuxaudio.org/listinfo/linux-audio-user




--
Jonathan E. Brickman
Ponderworthy Music | jeb at ponderworthy.com<mailto:jeb at ponderworthy.com> | (785)233-9977 | http://ponderworthy.com
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From dj_kaza at hotmail.com  Sun Aug 31 02:36:49 2014
From: dj_kaza at hotmail.com (Kazakore)
Date: Sun, 31 Aug 2014 08:21:49 +0545
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <540285CA.3070507@hawaii.rr.com>
References: <20140827143830.GA7320@aol.com>,
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>,
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>,
 <20140827154520.GB7320@aol.com>,
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>,
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>,
 <20140827192248.GC7320@aol.com> <53FF4A76.8000100@stackingdwarves.net>,
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>,
 <53FF52B0.4030702@stackingdwarves.net> <20140828200726.GA16743@aol.com>,
 <54002437.6080607@hawaii.rr.com> <DUB113-W18E942790CECC11578A7BE8FDB0@phx.gbl>
 <540187B3.1090504@hawaii.rr.com>
 <BLU436-SMTP76898D38FEFACB3A4C0CD48FD80@phx.gbl>
 <BLU436-SMTP158C7907D3B0163A6B68E6A8FD80@phx.gbl>
 <540285CA.3070507@hawaii.rr.com>
Message-ID: <BLU436-SMTP1601F5E0E33EA6C0615E4788FD90@phx.gbl>


On 31/08/14 08:02, david wrote:
>
> Thanks, sounds useful. Looks like their Outlet Stores are Europe only, 
> don't see USA as an option.
>

As I stated it was the USA one I checked yesterday on the assumption you 
might be that side of the pond.

http://outlet.lenovo.com/outlet_us/

From gnome at hawaii.rr.com  Sun Aug 31 03:51:00 2014
From: gnome at hawaii.rr.com (david)
Date: Sat, 30 Aug 2014 17:51:00 -1000
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <BLU436-SMTP1601F5E0E33EA6C0615E4788FD90@phx.gbl>
References: <20140827143830.GA7320@aol.com>,
 <d224c4f72d7668ccaaa409a3da40ecf3@peterlutek.com>,
 <daa3b263e38f9bd369249cab0792f342@peterlutek.com>,
 <20140827154520.GB7320@aol.com>,
 <alpine.LNX.2.00.1408271105280.3171@village.keycorner.org>,
 <CAGFf1ogyNkybjd=vOv71YOTS1gWUVS_rHhOU1s_T4q2VWBDJqA@mail.gmail.com>,
 <20140827192248.GC7320@aol.com> <53FF4A76.8000100@stackingdwarves.net>,
 <CAFa_cKm6iGC00R4Vt91_S_vbvjLhMCenc0Sy36d_8uL9RbxibA@mail.gmail.com>,
 <53FF52B0.4030702@stackingdwarves.net> <20140828200726.GA16743@aol.com>,
 <54002437.6080607@hawaii.rr.com> <DUB113-W18E942790CECC11578A7BE8FDB0@phx.gbl>
 <540187B3.1090504@hawaii.rr.com>
 <BLU436-SMTP76898D38FEFACB3A4C0CD48FD80@phx.gbl>
 <BLU436-SMTP158C7907D3B0163A6B68E6A8FD80@phx.gbl>
 <540285CA.3070507@hawaii.rr.com>
 <BLU436-SMTP1601F5E0E33EA6C0615E4788FD90@phx.gbl>
Message-ID: <54029BA4.50108@hawaii.rr.com>

On 08/30/2014 04:36 PM, Kazakore wrote:
>
> On 31/08/14 08:02, david wrote:
>>
>> Thanks, sounds useful. Looks like their Outlet Stores are Europe only,
>> don't see USA as an option.
>>
>
> As I stated it was the USA one I checked yesterday on the assumption you
> might be that side of the pond.
>
> http://outlet.lenovo.com/outlet_us/

Yah, I saw you mentioned the USA one, but the only one I could find via 
Google and the Lenovo site was the IE/EN one ...

Thanks for the link, browsing around on it right now. My church has a 
Presonus 12-channel Firewire interface we haven't been able to use for 
several years now because nobody has a laptop with working Firewire. Our 
main sound guy's ThinkPad finally died completely and we've never been 
able to make any ExpressCard Firewire port work on his HP laptop 
(saddled with Windows 7), something about the port/Firewire requiring a 
proprietary driver that's apparently XP-only.

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From len at ovenwerks.net  Sun Aug 31 04:56:01 2014
From: len at ovenwerks.net (Len Ovens)
Date: Sat, 30 Aug 2014 21:56:01 -0700 (PDT)
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <BLU436-SMTP115B35F8F0881A47DC9C1558FD80@phx.gbl>
References: <20140827143830.GA7320@aol.com> ,
 <201408282037.53956.gheskett@wdtv.com> ,
 <DUB113-W1107FDBFBE7CDE95335DD6F8FDB0@phx.gbl> ,
 <201408282245.42030.gheskett@wdtv.com>
 <DUB113-W3024DDF80474D625B42FD48FDB0@phx.gbl>
 <alpine.DEB.2.10.1408290728170.2499@scott.cbbs.org>
 <BLU436-SMTP115B35F8F0881A47DC9C1558FD80@phx.gbl>
Message-ID: <alpine.DEB.2.10.1408302119260.2499@scott.cbbs.org>

On Sat, 30 Aug 2014, dale wrote:

> On Fri, 2014-08-29 at 08:12 -0700, Len Ovens wrote:
>> Intel says that nobody really needs that low of a latency anyway in
>> response some sound cards problems with one of their implementations of
>> USB3. (the problem was USB2 cards with USB3 ports which are supposed to be
>> compatable)
>>
>> In recording latency is not the issue it is with live work. For recording
>> latency needs to known and constant and reasonably low for monitoring. I
>> have heard/read people who say what does it matter if you move your head a
>> foot or two closer or father from the speaker? But in live work if the
>> same audio comes from different places it is called a filter.
>>
>
> Yes if it is coming from two sources of different distances you get a
> comb filter but that's not what we're talking about with distance/delay
> here. It's more absolute latency.
>
> Examples I was presented at while at college/university were:
> * A piano player striking a key and hearing a sound, from the downwards
> motion to hearing the sound is about 6ms. Taking into account both
> mechanical transference from key to string and the sound to the ear.
> * For comparison this would be the same as a guitarist standing six feet
> away from his guitar amplifier if we lived in a ideal world (where
> electricity travelled the speed of light) but there is obviously
> propagation delay plus any added by stomp boxes (s)he may have. This is
> not a distance at which a guitarist finds it difficult to play!
>
> I sometimes think the hunt for super-low latency is a bit absurd! 3ms,
> to give you a 6ms round trip, should be a workable amount for pretty
> much anybody and most I expect could cope with quite a lot higher (not
> many working methods require the full round trip!)

Lets take my ice1712 based device. With jack set -p16 (as low as it can 
go), there is 3ms round trip:
1ms each way for the ice1712 internal monitor mixer and whatever else 
routing, plus .6ms each way for 3.2ms. So set -p32 for 4.4ms or -p64 for 
6.8ms.
Most USB2 IFs can do that fine in a USB2 only world. Some of 
intel's USB3 inplementations make -p64 for a USB2 unit full of xruns. So 
the "low" latency I was talking about is somewhat higher than 6ms.

> But full round trip is probably also the time when delay effects (com
> filter and echo) become important, such as using a PC as an LMS or FOH
> mixing desk. I would consider these specialist cases though.

Not that many years ago, the norm in computer based audio was sequencing 
and digital to disk (or memory) recording of audio was a specialist 
case... because the HW couldn't handle it. Now the HW can handle it and we 
do much more than just record audio to disk. So maybe the same thing is 
true here, The HW can't handle it so it becomes special case. The 
difference (from what I can tell) is that in this case, we seem to be 
going backwards. Latency is increasing. The reason it is increasing seems 
to be that someone has decided it is not important. So what might make 
sense in a very low latency environment, may never happen based on that 
decision.

What I am saying, is that to become standard application, the HW has to be 
cheap enough for most people to own it.



--
Len Ovens
www.ovenwerks.net


From moshwe at gmail.com  Sun Aug 31 07:13:40 2014
From: moshwe at gmail.com (Moshe Werner)
Date: Sun, 31 Aug 2014 10:13:40 +0300
Subject: [LAU] Open Source Audio Interface (was Successor/replacement for
	RME HDSP+Multiface?)
Message-ID: <CAGFf1oixjChzHw9kJBpiwDkAaoZD1-yQcK6QVR+R8r5JFnm6dQ@mail.gmail.com>

Hi all,

all the recent talk about less audio interfaces being supported for Linux
got me thinking.
First I want to mention that I'm neither a programmer nor an electrical
engineer, but what if we could develop our own AI?

We could start a Kickstarter project for the first open source AI.
I am sure many people would invest in the possibility of having a real
Linux natively supported high quality AI.

I imagine a modular approach with options to customize and add different
features like building blocks.
One user needs a lot of preamps, the other works mostly over ADAT or Madi...
We could even implement a DSP chip with LV2 effects to lessen the CPU load.
The possibilities are endless and I believe that it would broaden the
community instanly.

Maybe this is just wishful thinking, but it would solve one of the biggest
problems for Linux audio musicians.

Cheers

Moshe
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From moshwe at gmail.com  Sun Aug 31 09:26:45 2014
From: moshwe at gmail.com (Moshe Werner)
Date: Sun, 31 Aug 2014 12:26:45 +0300
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <alpine.DEB.2.10.1408302119260.2499@scott.cbbs.org>
References: <20140827143830.GA7320@aol.com>
 <201408282037.53956.gheskett@wdtv.com>
 <DUB113-W1107FDBFBE7CDE95335DD6F8FDB0@phx.gbl>
 <201408282245.42030.gheskett@wdtv.com>
 <DUB113-W3024DDF80474D625B42FD48FDB0@phx.gbl>
 <alpine.DEB.2.10.1408290728170.2499@scott.cbbs.org>
 <BLU436-SMTP115B35F8F0881A47DC9C1558FD80@phx.gbl>
 <alpine.DEB.2.10.1408302119260.2499@scott.cbbs.org>
Message-ID: <CAGFf1oh3unbsQbRysRsAbD2Moc=zK73hyRhZvNcfT0yW16mNYg@mail.gmail.com>

For those looking for a laptop with Express Card, Firewire, USB3 and
Thunderbolt:

http://shop.lenovo.com/us/en/laptops/thinkpad/w-series/w540/?sb=:00000025:00003883:

It is supported under Ubuntu 12.04 LTS.
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From mott at reverberant.com  Sun Aug 31 11:03:55 2014
From: mott at reverberant.com (Iain Mott)
Date: Sun, 31 Aug 2014 08:03:55 -0300
Subject: [LAU] html5 audio through jack
In-Reply-To: <BLU436-SMTP227E4E9F322A3846F31D45F8FD80@phx.gbl>
References: <1409395113.3154.5.camel@espelho>
 <BLU437-SMTP21B335DEAC0E6903E9270E8FD80@phx.gbl>
 <540203FD.5070803@rektau.ukfsn.org>
 <BLU436-SMTP227E4E9F322A3846F31D45F8FD80@phx.gbl>
Message-ID: <1409483035.3145.5.camel@espelho>

Thanks Dale and Rob - got more than half way into the configuration when
my hdsp card started fail again after reboot (it has not been well after
some recent ubuntu updates). Not sure what's going on, but I'll likely
only get back on to this at the end of the week. Until then...
Iain



Em Dom, 2014-08-31 ?s 01:57 +0545, Kazakore escreveu:
> On 30/08/14 22:48, rob wrote:
> > On 30/08/14 13:09, Kazakore wrote:
> >>
> >> On 30/08/14 16:23, Iain Mott wrote:
> >>> Hi list,
> >>>
> >>> I'm thinking of updating some of my web pages to use multi-platform
> >>> flash/html5 audio players, at present they use flash only and won't 
> >>> play
> >>> on iPads for example.
> >>>
> >>> Due to some problems I was having with pulse audio in relation to my 
> >>> HDSP
> >>> interface I have recently disabled it and all my audio is running via
> >>> jack/alsa and
> >>> the HDSP interface. With flash in firefox, there are no problems and 
> >>> the
> >>> audio plays. My .asoundrc is configured with the following:
> >>>
> >>> pcm.rawjack {
> >>>      type jack
> >>>      playback_ports {
> >>>          0 system:playback_1
> >>>          1 system:playback_2
> >>>      }
> >>>      capture_ports {
> >>>          0 system:capture_1
> >>>          1 system:capture_2
> >>>      }
> >>> }
> >>>
> >>> pcm.jack {
> >>>      type plug
> >>>      slave { pcm "rawjack" }
> >>>      hint {
> >>>       description "JACK Audio Connection Kit"
> >>>      }
> >>> }
> >>>
> >>>
> >>> pcm.!default {
> >>>      type plug
> >>>      slave { pcm "rawjack" }
> >>> }
> >>>
> >>>
> >>>
> >>> HTML5 players in firefox don't play however via jack. When pulse was
> >>> enabled, HTML5
> >>> content would play through the computer's built-in sound card. Now that
> >>> it's disabled I can't get it to play through jack.
> >>>
> >>> An example page with a HTML5 player is here:
> >>>
> >>> http://www.html5tutorial.info/html5-audio.php
> >>>
> >>> Any suggestions please? A modification of the .asoundrc?
> >>>
> >>> I'm running Ubuntu 14.04
> >>>
> >>> Thanks,
> >>>
> >>> _______________________________________________
> >>> Linux-audio-user mailing list
> >>> Linux-audio-user at lists.linuxaudio.org
> >>> http://lists.linuxaudio.org/listinfo/linux-audio-user
> >>
> >> Can't help but can tell you I have the exact same problem! I also don't
> >> get audio from streaming media directly from the internet (eg a web
> >> hosted .ogg) Have you managed to get that working? I use VLC's
> >> Multimedia Plugin for firefox, VLC is set to Jack, also have all
> >> GStreamer routed through Jack, but "normal" firefox audio never 
> >> reaches it.
> >>
> >> Good luck with this one. It's part of the reason Arch is feeling so
> >> appealing to me right now. I can truly get a system with no P(IT)A in
> >> there! ;-)
> >>
> >> Dale.
> >>
> >> _______________________________________________
> >
> > http://alsa.opensrc.org/Jack_and_Loopback_device_as_Alsa-to-Jack_bridge might 
> > be worth looking at.
> >
> > rob
> >
> >
> > _______________________________________________
> >
> 
> I'm sure it used to be at the bottom of this page it had a note about 
> how going the ALSA Loopback route would stop you being able to change 
> settings from within QJackCtl, and just at the same point I read it 
> there was a thread on here by somebody not being able to set up Jack 
> through QJackCtl which at the time I assumed must be related. Can't see 
> any comment on it anywhere at all now though...
> 
> http://jackaudio.org/faq/routing_alsa.html
> 
> But obviously that would be unacceptable and why I never tried 
> re-routing ALSA.
> 
> Dale.
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user



From ralf.mardorf at rocketmail.com  Sun Aug 31 13:15:50 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 31 Aug 2014 15:15:50 +0200
Subject: [LAU] Update Phasex:  Sequencer suggestions?
In-Reply-To: <20140829130233.38095a15@eeyore.mozart.uni-klu.ac.at>
References: <1409056818.7093.3.camel@rocketmail.com>
 <20140826152022.785793f5@eeyore.mozart.uni-klu.ac.at>
 <1409060454.7093.10.camel@rocketmail.com>
 <1409060819.7093.12.camel@rocketmail.com>
 <20140826140820.GA16859@linuxaudio.org>
 <1409064038.7093.22.camel@rocketmail.com>
 <20140826151337.GB16859@linuxaudio.org>
 <20140826172024.734dd13f@eeyore.mozart.uni-klu.ac.at>
 <20140826160630.GC16859@linuxaudio.org>
 <1409069791.7093.33.camel@rocketmail.com>
 <20140826162353.GD16859@linuxaudio.org>
 <1409260025.4790.8.camel@rocketmail.com>
 <1409261340.4790.11.camel@rocketmail.com>
 <20140829130233.38095a15@eeyore.mozart.uni-klu.ac.at>
Message-ID: <1409490950.14388.8.camel@rocketmail.com>

On Fri, 2014-08-29 at 13:02 +0200, Philipp ?berbacher wrote:
> I'm too stupid to patch apparently, otherwise I would have done it
> already.

Reported and patches were added by the maintainer. Nobody needs to do
everything ;).

Last Updated: 	2014-08-30 04:19

[rocketmouse at archlinux ~]$ yaourt -S phasex-git

==> Downloading phasex-git PKGBUILD from AUR...
x .AURINFO
x PKGBUILD
x fix_DC_offset.patch
x phasex-git.install
x snprintf-overflow.patch
x wave_sample.patch
Comment by Ralf_Mardorf  (2014-08-28 21:23)
Hi,

please consider to add the patches, to avoid DC offsets.

http://lists.linuxaudio.org/pipermail/linux-audio-user/2014-August/098653.html
https://github.com/williamweston/phasex/issues/10

Regards,
Ralf

Comment by rtfreedman  (2014-08-30 04:25)
This is an experimental release with some patches - please test it and report back.
fix_DC_offset.patch fixes this bug: https://github.com/williamweston/phasex/issues/10

Discussions: http://lists.linuxaudio.org/pipermail/linux-audio-user/2014-August/098653.html

[...]


From len at ovenwerks.net  Sun Aug 31 16:29:34 2014
From: len at ovenwerks.net (Len Ovens)
Date: Sun, 31 Aug 2014 09:29:34 -0700 (PDT)
Subject: [LAU] Open Source Audio Interface (was Successor/replacement
 for RME HDSP+Multiface?)
In-Reply-To: <CAGFf1oixjChzHw9kJBpiwDkAaoZD1-yQcK6QVR+R8r5JFnm6dQ@mail.gmail.com>
References: <CAGFf1oixjChzHw9kJBpiwDkAaoZD1-yQcK6QVR+R8r5JFnm6dQ@mail.gmail.com>
Message-ID: <alpine.DEB.2.10.1408310737500.2499@scott.cbbs.org>

On Sun, 31 Aug 2014, Moshe Werner wrote:

> all the recent talk about less audio interfaces being supported for Linux got me
> thinking.
> First I want to mention that I'm neither a programmer nor an electrical engineer, but
> what if we could develop our own AI?

This is not the first time for this idea. There are one or two people 
working on it. The idea that seems to be the best is an ethernet connected 
AI because this seems to be the digital interface that stays around and is 
best supported. The idea is to use an arm based board with a netjack 
master and built in audio IF. The only project I know of is to at first 
provide stereo i/o as a proof of concept.

> I imagine a modular approach with options to customize and add different features like
> building blocks.
> One user needs a lot of preamps, the other works mostly over ADAT or Madi...

A few comments about ADAT, while the format of the messages is well known, 
it is not open. Licencing may be a problem. True MADI uses obsolete HW. 
There are a number of audio over ethernet protocols with various openness 
or not.

Just a quick note about DIY HW. It is not generally cheaper than buying 
the same capability made by someone else. There are premade solutions out 
there, but they are not in the range of a lot of linux musicians.

> We could even implement a DSP chip with LV2 effects to lessen the CPU load.

Almost any arm based board would be able to add some DSP capabilities. It 
may also be possible to use a DSP board instead of an arm board, but the 
learning curve for programing it may be higher... like having a running up 
to date linux kernel and jackd running on it for example.

> Maybe this is just wishful thinking, but it would solve one of the biggest problems for
> Linux audio musicians.

The biggest limit to audio interfaces in basement studio is getting over 
the fact that a good audio interface is going to cost more than the 
computer it works with. This is first of all a psycological problem not 
technical.

Most of us are looking for those ~$500 solutions. That is, we expect to 
get 4 to 8 i/os with mic pres for $500 or so. That is why ADAT looks good 
to us. The Audio Science 8 i/o PCIe card is just over $1k but just has 
line in (my D66 is line level too BTW) so then you need mic pres. Most of 
us have supplied these with a mixer of some sort.

Really, an audio interface is two parts, The adc/dac, and the preamps. It 
is easy enough to do the line level input, even balanced.

Going one step up, there are a lot of mic preamps around that provide a 
lot more control than a mixer or 8 input audio IF. Most of them have 
s/pdif or aes3 out. So maybe an audio interface that provides that makes 
more sense. However, the reason the generic mic pres work for most people 
is that they are using a set of dynamic mics with a few condeser mics 
(probably low end like mine) and the pre doesn't matter as much as it 
might with a ribbon mic for example.

So the problem is less technical than it might seem. (on the technical 
end, using netjack as an audio interface does use more CPU than the same 
alsa card used locally) To make such a project worth while it needs to 
have a wide appeal, this means cheap... I can't think of any other way to 
put it. The goal ends up being 8 i/os with 8 mic pre for around $500. 
Having something more stable and lower latency than USB2 might give you 
another $200 (maybe more) to play with, but writing OSx/win drivers would 
give a lot more room (netjack does this OOTB).

Another idea might be to start looking at asking for linux drivers for 
stage end digital snakes. 24/12 i/o for $2200 is already within the range 
cost wise (per channel) we are looking at. and there may be cheaper ones 
like this one:
http://www.sweetwater.com/store/detail/S16
for $900.
I think it may make more sense to write drivers for these kinds of things 
even if it means adding a second ethernet card to the computer (or letting 
the driver take over the one on the laptop)

Just some quick thoughts, perhaps not as complete as they could be.

--
Len Ovens
www.ovenwerks.net


From ralf.mardorf at rocketmail.com  Sun Aug 31 17:03:38 2014
From: ralf.mardorf at rocketmail.com (Ralf Mardorf)
Date: Sun, 31 Aug 2014 19:03:38 +0200
Subject: [LAU] Open Source Audio Interface (was Successor/replacement
 for RME HDSP+Multiface?)
In-Reply-To: <alpine.DEB.2.10.1408310737500.2499@scott.cbbs.org>
References: <CAGFf1oixjChzHw9kJBpiwDkAaoZD1-yQcK6QVR+R8r5JFnm6dQ@mail.gmail.com>
 <alpine.DEB.2.10.1408310737500.2499@scott.cbbs.org>
Message-ID: <1409504618.25521.35.camel@rocketmail.com>

On Sun, 2014-08-31 at 09:29 -0700, Len Ovens wrote:
> Just a quick note about DIY HW. It is not generally cheaper than buying 
> the same capability made by someone else. There are premade solutions out 
> there, but they are not in the range of a lot of linux musicians.

In May 2011 I bought a RME HDSPe AIO for 478 ? and a ADA8000 for 169 ?.
I doubt that DIY is nearly that inexpensive when providing the same
quality. It likely would be much more expensive. At least for this PCIe
card RME is willing to help ALSA driver coders, the only problem is,
that when reporting the issues with this card, nobody seriously is
interested in the bug reports and feature requests. Assumed it would be
possible to build an audio interface for less costs, then there still
would be the same issue for the ALSA driver as we already have now.
Somebody with the knowledge, time and willing to write the driver is
needed. Assumed you should have the needed abilities to write drivers,
consider to that for existing sound cards instead of thinking to build
the hardware too.

> Most of us are looking for those ~$500 solutions. That is, we expect to 
> get 4 to 8 i/os with mic pres for $500 or so. That is why ADAT looks good 
> to us. The Audio Science 8 i/o PCIe card is just over $1k but just has 
> line in (my D66 is line level too BTW) so then you need mic pres. Most of 
> us have supplied these with a mixer of some sort.

Laptops are another issue, but for PC mobos the PCIe solution that
already does exist is around 478 ? + 169 ? = 647 ?.

$ hdspmixer 
[snip]
Looking for RME cards:
Card 0: RME AIO S/N 0x579bcc at 0xfddf0000, irq 18
RME AIO found!
[snip]


$ hdspconf 
[snip]
Looking for HDSP cards :
Card 0 : RME AIO S/N 0x579bcc at 0xfddf0000, irq 18
[snip]
No Hammerfall DSP card found.


So what ever I do wrong, only 2 of the 8 ADAT channels are available by
the jackd ports and I don't know how to get all the features that are
provided by the Windows driver. Not that I'm a Windows user, but to test
the sound card during warranty period, I tested it with a Windows
install. Already TotalMix is completely different for Windows, than for
Linux. There are also not the latency/xrun issues on the same machine as
for Linux. Perhaps somebody would car about such issues, if somebody
else but me would report it, since I received a mail off-list with the
warning to ban me from Linux audio mailing lists.

Whitewashing and ignoring issues seems to be more pleasant, than talking
about hard facts.


From moshwe at gmail.com  Sun Aug 31 19:20:38 2014
From: moshwe at gmail.com (Moshe Werner)
Date: Sun, 31 Aug 2014 22:20:38 +0300
Subject: [LAU] Open Source Audio Interface (was Successor/replacement
 for RME HDSP+Multiface?)
In-Reply-To: <alpine.DEB.2.10.1408310737500.2499@scott.cbbs.org>
References: <CAGFf1oixjChzHw9kJBpiwDkAaoZD1-yQcK6QVR+R8r5JFnm6dQ@mail.gmail.com>
 <alpine.DEB.2.10.1408310737500.2499@scott.cbbs.org>
Message-ID: <CAGFf1ogUL7+qM6VNVdw-jDsjGPn-L6xrXVGMQjdNYqOz83bFMw@mail.gmail.com>

On Sun, Aug 31, 2014 at 7:29 PM, Len Ovens <len at ovenwerks.net> wrote:

> On Sun, 31 Aug 2014, Moshe Werner wrote:
>
>  all the recent talk about less audio interfaces being supported for Linux
>> got me
>> thinking.
>> First I want to mention that I'm neither a programmer nor an electrical
>> engineer, but
>> what if we could develop our own AI?
>>
>
> This is not the first time for this idea. There are one or two people
> working on it. The idea that seems to be the best is an ethernet connected
> AI because this seems to be the digital interface that stays around and is
> best supported. The idea is to use an arm based board with a netjack master
> and built in audio IF. The only project I know of is to at first provide
> stereo i/o as a proof of concept.


Interesting, I didn't know this. Can you send a link to it?

>
>
>  I imagine a modular approach with options to customize and add different
>> features like
>> building blocks.
>> One user needs a lot of preamps, the other works mostly over ADAT or
>> Madi...
>>
>
> A few comments about ADAT, while the format of the messages is well known,
> it is not open. Licencing may be a problem. True MADI uses obsolete HW.
> There are a number of audio over ethernet protocols with various openness
> or not.
>

But ADAT is one of the most important and widespread interfaces, so we
can't just ignore it

>
> Just a quick note about DIY HW. It is not generally cheaper than buying
> the same capability made by someone else. There are premade solutions out
> there, but they are not in the range of a lot of linux musicians.


I didn't say cheaper. I am willing to pay for decent hardware, and also
software. I think I've donated more money to the Ardour project than I paid
for Protools (I admit, I've also got a Mac running Protools, there are
certain paying customers that still insist on that).
DIY gives you a certain amount of freedom, and quite some satisfaction. I'm
just about to finish a Gyraf G7 tube mic, which was a lot of fun building.
But really my point is not so much to build the AI by myself, but to
support a project that reflects the ideology that we stand behind.
I did pay a lot for my hardware, I would be much happier to pay it to
someone that invests in our community and also gives real support for
paying customers.
For now RME are the closest to this description but who knows for how long.


>
>  We could even implement a DSP chip with LV2 effects to lessen the CPU
>> load.
>>
>
> Almost any arm based board would be able to add some DSP capabilities. It
> may also be possible to use a DSP board instead of an arm board, but the
> learning curve for programing it may be higher... like having a running up
> to date linux kernel and jackd running on it for example.
>
>
>  Maybe this is just wishful thinking, but it would solve one of the
>> biggest problems for
>> Linux audio musicians.
>>
>
> The biggest limit to audio interfaces in basement studio is getting over
> the fact that a good audio interface is going to cost more than the
> computer it works with. This is first of all a psycological problem not
> technical.
>

I can see your point, but I don't totally agree. Cost is one matter but
IMHO the main factor of frustration is the lack of support, and generally
cold shoulder, we as a community get.
What will happen if RME decides do drop the support for Linux (whats left
of it)? How long can we chase after laptops with express cards just to run
the RME box.
Another thing we're missing is the diversity that exists in the "pro" world.
If RME would support their whole range of products under Linux that would
be slightly different, but today we're pretty much bound to a handful of
interfaces.
Now someone that has interest in Linux audio will be on the run at least
when he finds out that his dear AI is not supported under Linux. (Happened
to all my audio engineer buddies that saw Ardour, got excited about the
idea, then asked if it will work with "name your favorite AI", I said not
yet, that's it...)

>
> Most of us are looking for those ~$500 solutions. That is, we expect to
> get 4 to 8 i/os with mic pres for $500 or so. That is why ADAT looks good
> to us. The Audio Science 8 i/o PCIe card is just over $1k but just has line
> in (my D66 is line level too BTW) so then you need mic pres. Most of us
> have supplied these with a mixer of some sort.
>
> Really, an audio interface is two parts, The adc/dac, and the preamps. It
> is easy enough to do the line level input, even balanced.
>

My system consists of a 32 channel inline console->Alesis HDR24x AD->RME
hdsp9652 so tell me about it :)

>
> Going one step up, there are a lot of mic preamps around that provide a
> lot more control than a mixer or 8 input audio IF. Most of them have s/pdif
> or aes3 out. So maybe an audio interface that provides that makes more
> sense. However, the reason the generic mic pres work for most people is
> that they are using a set of dynamic mics with a few condeser mics
> (probably low end like mine) and the pre doesn't matter as much as it might
> with a ribbon mic for example.
>
> So the problem is less technical than it might seem. (on the technical
> end, using netjack as an audio interface does use more CPU than the same
> alsa card used locally) To make such a project worth while it needs to have
> a wide appeal, this means cheap... I can't think of any other way to put
> it. The goal ends up being 8 i/os with 8 mic pre for around $500. Having
> something more stable and lower latency than USB2 might give you another
> $200 (maybe more) to play with, but writing OSx/win drivers would give a
> lot more room (netjack does this OOTB).
>

Totally agree, we're witnessing genius on a daily basis in this list, so I
don't see a technical problem at all.
But I think the goal would be to bring something real new to the world,
like a "stackable" or modular interface that every user could more or less
configure using standard building blocks. That would cater the whole range
of users in our community.
The problem is that I'm neither a programmer nor a hardware designer, I'm a
user, so I'm just thinking out loud.

Best

Moshe
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From murks at tuxfamily.org  Sun Aug 31 19:42:50 2014
From: murks at tuxfamily.org (Philipp =?UTF-8?B?w5xiZXJiYWNoZXI=?=)
Date: Sun, 31 Aug 2014 21:42:50 +0200
Subject: [LAU] html5 audio through jack
In-Reply-To: <1409395113.3154.5.camel@espelho>
References: <1409395113.3154.5.camel@espelho>
Message-ID: <20140831214250.271947d4@eeyore.mozart.uni-klu.ac.at>

On Sat, 30 Aug 2014 07:38:33 -0300
Iain Mott <mott at reverberant.com> wrote:

> 
> Hi list,
> 
> I'm thinking of updating some of my web pages to use multi-platform
> flash/html5 audio players, at present they use flash only and won't
> play on iPads for example.
> 
> Due to some problems I was having with pulse audio in relation to my
> HDSP interface I have recently disabled it and all my audio is
> running via jack/alsa and the HDSP interface. With flash in firefox,
> there are no problems and the audio plays. My .asoundrc is configured
> with the following:
> 
> pcm.rawjack {
>     type jack
>     playback_ports {
>         0 system:playback_1
>         1 system:playback_2
>     }
>     capture_ports {
>         0 system:capture_1
>         1 system:capture_2
>     }
> }
> 
> pcm.jack {
>     type plug
>     slave { pcm "rawjack" }
>     hint {
>  	description "JACK Audio Connection Kit"
>     }
> }
> 
> 
> pcm.!default {
>     type plug
>     slave { pcm "rawjack" }
> }
> 
> 
> 
> HTML5 players in firefox don't play however via jack. When pulse was
> enabled, HTML5 content would play through the computer's built-in
> sound card. Now that it's disabled I can't get it to play through
> jack.
> 
> An example page with a HTML5 player is here:
> 
> http://www.html5tutorial.info/html5-audio.php
> 
> Any suggestions please? A modification of the .asoundrc?
> 
> I'm running Ubuntu 14.04
> 
> Thanks,

I'm not surprised. I am very much against anything audio (or
multimedia) in browsers. Nowadays browsers do pretty much everything,
but badly. One constant grieve for me since years has been that there
are virtually no audio settings for the browser (for example search
for 'audio' in firefox about:config). It just takes whatever it can
find, whatever is default on the system, and plays back through the
first two channels. It may work for 95% of the users, but if you're
part of the remaining 5% you can't do anything about it.
For that reason alone doing any specialised multimedia thing for the
browser is just crazy, there is basically no user control.
In your particular case I think it is the flash plugin itself that
handled audio output, and now with html5 it is the browser that does
it,and probably does something stupid. Maybe the way it rubs the ALSA
API the wrong way. It is really hard to say what's going on in a
browser.

Sorry to be of little help.

Philipp

From len at ovenwerks.net  Sun Aug 31 21:56:28 2014
From: len at ovenwerks.net (Len Ovens)
Date: Sun, 31 Aug 2014 14:56:28 -0700 (PDT)
Subject: [LAU] Open Source Audio Interface (was Successor/replacement
 for RME HDSP+Multiface?)
In-Reply-To: <CAGFf1ogUL7+qM6VNVdw-jDsjGPn-L6xrXVGMQjdNYqOz83bFMw@mail.gmail.com>
References: <CAGFf1oixjChzHw9kJBpiwDkAaoZD1-yQcK6QVR+R8r5JFnm6dQ@mail.gmail.com>
 <alpine.DEB.2.10.1408310737500.2499@scott.cbbs.org>
 <CAGFf1ogUL7+qM6VNVdw-jDsjGPn-L6xrXVGMQjdNYqOz83bFMw@mail.gmail.com>
Message-ID: <alpine.DEB.2.10.1408311407550.2499@scott.cbbs.org>

On Sun, 31 Aug 2014, Moshe Werner wrote:

> On Sun, Aug 31, 2014 at 7:29 PM, Len Ovens <len at ovenwerks.net> wrote:
>       On Sun, 31 Aug 2014, Moshe Werner wrote:
>
>             all the recent talk about less audio interfaces being supported
>             for Linux got me
>             thinking.
>             First I want to mention that I'm neither a programmer nor an
>             electrical engineer, but
>             what if we could develop our own AI?
> 
> 
> This is not the first time for this idea. There are one or two people working on
> it. The idea that seems to be the best is an ethernet connected AI because this
> seems to be the digital interface that stays around and is best supported. The
> idea is to use an arm based board with a netjack master and built in audio IF. The
> only project I know of is to at first provide stereo i/o as a proof of concept.
> 
> 
> Interesting, I didn't know this. Can you send a link to it?

http://lists.linuxaudio.org/pipermail/linux-audio-user/2011-October/081520.html
is the start I think... Though it may have surfaced since then too.

It seems to me that at some point the same project came back in either LAU 
or LAD but the topic got changed.

> A few comments about ADAT, while the format of the messages is well known, it is
> not open. Licencing may be a problem. True MADI uses obsolete HW. There are a
> number of audio over ethernet protocols with various openness or not.
> 
> ?
> But ADAT is one of the most important and widespread interfaces, so we can't just ignore
> it

Reading through the above thread, it seems to agree.

>
>       Just a quick note about DIY HW. It is not generally cheaper than buying the
>       same capability made by someone else. There are premade solutions out there,
>       but they are not in the range of a lot of linux musicians.
> 
> 
> I didn't say cheaper. I am willing to pay for decent hardware, and also software. I

Ok, nothing to argue about there. DIY is also a way of getting what can 
not be bought. I have done some small HW projects before (MIDI stuff) but 
am finding with older eyes even with glasses I tire sooner. SW is easier 
for me (and I am not that great at that either :)  )

>       Maybe this is just wishful thinking, but it would solve one of the
>       biggest problems for
>       Linux audio musicians.
> 
> 
> The biggest limit to audio interfaces in basement studio is getting over the fact
> that a good audio interface is going to cost more than the computer it works with.
> This is first of all a psycological problem not technical.
> 
> 
> I can see your point, but I don't totally agree. Cost is one matter but IMHO the main
> factor of frustration is the lack of support, and generally cold shoulder, we as a
> community get.

I was not thinking from a custom POV, but from a developing to manufacture 
POV. In which case it has to be profitable to fly.

>       So the problem is less technical than it might seem. (on the technical end,
>       using netjack as an audio interface does use more CPU than the same alsa
>       card used locally) To make such a project worth while it needs to have a
>       wide appeal, this means cheap... I can't think of any other way to put it.
>       The goal ends up being 8 i/os with 8 mic pre for around $500. Having
>       something more stable and lower latency than USB2 might give you another
>       $200 (maybe more) to play with, but writing OSx/win drivers would give a lot
>       more room (netjack does this OOTB).
> 
> 
> Totally agree, we're witnessing genius on a daily basis in this list, so I don't see a
> technical problem at all.
> But I think the goal would be to bring something real new to the world, like a
> "stackable" or modular interface that every user could more or less configure using
> standard building blocks. That would cater the whole range of users in our community.
> The problem is that I'm neither a programmer nor a hardware designer, I'm a user, so I'm
> just thinking out loud.

I'm not much of a designer ether, more of a hacker (not cracker thanks). I 
am not much at refining things once they work. I only have so much time to 
spare and limited energy due to physical problems.


--
Len Ovens
www.ovenwerks.net

From moshwe at gmail.com  Sun Aug 31 22:14:46 2014
From: moshwe at gmail.com (Moshe Werner)
Date: Mon, 1 Sep 2014 01:14:46 +0300
Subject: [LAU] Open Source Audio Interface (was Successor/replacement
 for RME HDSP+Multiface?)
In-Reply-To: <alpine.DEB.2.10.1408311407550.2499@scott.cbbs.org>
References: <CAGFf1oixjChzHw9kJBpiwDkAaoZD1-yQcK6QVR+R8r5JFnm6dQ@mail.gmail.com>
 <alpine.DEB.2.10.1408310737500.2499@scott.cbbs.org>
 <CAGFf1ogUL7+qM6VNVdw-jDsjGPn-L6xrXVGMQjdNYqOz83bFMw@mail.gmail.com>
 <alpine.DEB.2.10.1408311407550.2499@scott.cbbs.org>
Message-ID: <CAGFf1oh7iPOW2u56Yr2x1chzFDMFiyzCK_=So0ufApUbc932tA@mail.gmail.com>

>> This is not the first time for this idea. There are one or two people
>> working on
>> it. The idea that seems to be the best is an ethernet connected AI
>> because this
>> seems to be the digital interface that stays around and is best
>> supported. The
>> idea is to use an arm based board with a netjack master and built in
>> audio IF. The
>> only project I know of is to at first provide stereo i/o as a proof of
>> concept.
>>
>>
>> Interesting, I didn't know this. Can you send a link to it?
>>
>
> http://lists.linuxaudio.org/pipermail/linux-audio-user/
> 2011-October/081520.html
> is the start I think... Though it may have surfaced since then too.
>
> It seems to me that at some point the same project came back in either LAU
> or LAD but the topic got changed.
>
>
> Wow thanks Len!
Thats a very interesting read indeed.
How did I not come across this earlier.
But I can't find any recent info about it. Too bad.
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From gnome at hawaii.rr.com  Sun Aug 31 22:27:53 2014
From: gnome at hawaii.rr.com (david)
Date: Sun, 31 Aug 2014 12:27:53 -1000
Subject: [LAU] Successor/replacement for RME HDSP+Multiface?
In-Reply-To: <CAGFf1oh3unbsQbRysRsAbD2Moc=zK73hyRhZvNcfT0yW16mNYg@mail.gmail.com>
References: <20140827143830.GA7320@aol.com>
 <201408282037.53956.gheskett@wdtv.com>
 <DUB113-W1107FDBFBE7CDE95335DD6F8FDB0@phx.gbl>
 <201408282245.42030.gheskett@wdtv.com>
 <DUB113-W3024DDF80474D625B42FD48FDB0@phx.gbl>
 <alpine.DEB.2.10.1408290728170.2499@scott.cbbs.org>
 <BLU436-SMTP115B35F8F0881A47DC9C1558FD80@phx.gbl>
 <alpine.DEB.2.10.1408302119260.2499@scott.cbbs.org>
 <CAGFf1oh3unbsQbRysRsAbD2Moc=zK73hyRhZvNcfT0yW16mNYg@mail.gmail.com>
Message-ID: <5403A169.1050805@hawaii.rr.com>

On 08/30/2014 11:26 PM, Moshe Werner wrote:
> For those looking for a laptop with Express Card, Firewire, USB3 and
> Thunderbolt:
>
> http://shop.lenovo.com/us/en/laptops/thinkpad/w-series/w540/?sb=:00000025:00003883:
>
> It is supported under Ubuntu 12.04 LTS.

Whoa. That's just about perfect. And it's only $343 more than my present 
System76 Gazelle Pro 9 ended up costing, which has only USB connections, 
no ExpressCard port, and a 2.4GHz (vs 3.4GHz) 4th gen i7.

Thanks!

-- 
David W. Jones
gnome at hawaii.rr.com
authenticity, honesty, community
http://dancingtreefrog.com

From len at ovenwerks.net  Sun Aug 31 22:41:44 2014
From: len at ovenwerks.net (Len Ovens)
Date: Sun, 31 Aug 2014 15:41:44 -0700 (PDT)
Subject: [LAU] Open Source Audio Interface (was Successor/replacement
 for RME HDSP+Multiface?)
In-Reply-To: <CAGFf1ogUL7+qM6VNVdw-jDsjGPn-L6xrXVGMQjdNYqOz83bFMw@mail.gmail.com>
References: <CAGFf1oixjChzHw9kJBpiwDkAaoZD1-yQcK6QVR+R8r5JFnm6dQ@mail.gmail.com>
 <alpine.DEB.2.10.1408310737500.2499@scott.cbbs.org>
 <CAGFf1ogUL7+qM6VNVdw-jDsjGPn-L6xrXVGMQjdNYqOz83bFMw@mail.gmail.com>
Message-ID: <alpine.DEB.2.10.1408311517590.2499@scott.cbbs.org>

On Sun, 31 Aug 2014, Moshe Werner wrote:

> Interesting, I didn't know this. Can you send a link to it?

Here is another thread:
http://lists.linuxaudio.org/pipermail/linux-audio-user/2009-November/064520.html
It moves here:
http://lists.linuxaudio.org/pipermail/linux-audio-dev/2009-November/024713.html
And just in case you ever thoght of doing this:
http://lists.linuxaudio.org/pipermail/linux-audio-dev/2009-December/025080.html

I do not know where this has gotten to. I think it surfaces again as 
another thread, maybe related to the development board or cpu chip.

When I was looking at it, it seemed that a dac/adc had about the same 
interface as s/pdif, aes3. ADAT chips are harder to find. aes10 (I think 
thats the right one) or MADI, is data over an ethernet IF and as such is 
sent as aes3 channels in series (over simplified a lot) That is to say 
mostly software once the ethernet IF is chosen. However, it would mean 
rewriting the ethernet driver for the interface. There seems to a newer 
MADI that no longer uses optical, but cat5. It may be that a standard 
ethernet card would work for that with the right driver.

--
Len Ovens
www.ovenwerks.net


From marc at hacklava.net  Sun Aug 31 22:45:41 2014
From: marc at hacklava.net (Marc =?UTF-8?B?TGF2YWxsw6ll?=)
Date: Sun, 31 Aug 2014 18:45:41 -0400
Subject: [LAU] html5 audio through jack
In-Reply-To: <20140831214250.271947d4@eeyore.mozart.uni-klu.ac.at>
References: <1409395113.3154.5.camel@espelho>
 <20140831214250.271947d4@eeyore.mozart.uni-klu.ac.at>
Message-ID: <20140831184541.7d37587a@hacklava.net>


I play the audio output of my browsers (Firefox and Chrome/Chromium)
through jack. It works with HTML5 and Flash.

https://docs.fedoraproject.org/en-US/Fedora/18/html/Musicians_Guide/sect-Musicians_Guide-Integrating_PulseAudio_with_JACK.html

https://wiki.archlinux.org/index.php/PulseAudio/Examples#PulseAudio_through_JACK

The pulseaudio "jack" sink register as a 8 channels playback source in
jack, so it works with 5.1 and 7.1 streams, not only for stereo streams.

--
Marc

Le Sun, 31 Aug 2014 21:42:50 +0200,
Philipp ?berbacher <murks at tuxfamily.org> a ?crit :

> On Sat, 30 Aug 2014 07:38:33 -0300
> Iain Mott <mott at reverberant.com> wrote:
> 
> > 
> > Hi list,
> > 
> > I'm thinking of updating some of my web pages to use multi-platform
> > flash/html5 audio players, at present they use flash only and won't
> > play on iPads for example.
> > 
> > Due to some problems I was having with pulse audio in relation to my
> > HDSP interface I have recently disabled it and all my audio is
> > running via jack/alsa and the HDSP interface. With flash in firefox,
> > there are no problems and the audio plays. My .asoundrc is
> > configured with the following:
> > 
> > pcm.rawjack {
> >     type jack
> >     playback_ports {
> >         0 system:playback_1
> >         1 system:playback_2
> >     }
> >     capture_ports {
> >         0 system:capture_1
> >         1 system:capture_2
> >     }
> > }
> > 
> > pcm.jack {
> >     type plug
> >     slave { pcm "rawjack" }
> >     hint {
> >  	description "JACK Audio Connection Kit"
> >     }
> > }
> > 
> > 
> > pcm.!default {
> >     type plug
> >     slave { pcm "rawjack" }
> > }
> > 
> > 
> > 
> > HTML5 players in firefox don't play however via jack. When pulse was
> > enabled, HTML5 content would play through the computer's built-in
> > sound card. Now that it's disabled I can't get it to play through
> > jack.
> > 
> > An example page with a HTML5 player is here:
> > 
> > http://www.html5tutorial.info/html5-audio.php
> > 
> > Any suggestions please? A modification of the .asoundrc?
> > 
> > I'm running Ubuntu 14.04
> > 
> > Thanks,
> 
> I'm not surprised. I am very much against anything audio (or
> multimedia) in browsers. Nowadays browsers do pretty much everything,
> but badly. One constant grieve for me since years has been that there
> are virtually no audio settings for the browser (for example search
> for 'audio' in firefox about:config). It just takes whatever it can
> find, whatever is default on the system, and plays back through the
> first two channels. It may work for 95% of the users, but if you're
> part of the remaining 5% you can't do anything about it.
> For that reason alone doing any specialised multimedia thing for the
> browser is just crazy, there is basically no user control.
> In your particular case I think it is the flash plugin itself that
> handled audio output, and now with html5 it is the browser that does
> it,and probably does something stupid. Maybe the way it rubs the ALSA
> API the wrong way. It is really hard to say what's going on in a
> browser.
> 
> Sorry to be of little help.
> 
> Philipp
> _______________________________________________
> Linux-audio-user mailing list
> Linux-audio-user at lists.linuxaudio.org
> http://lists.linuxaudio.org/listinfo/linux-audio-user
> 


From jeb at ponderworthy.com  Sun Aug 31 23:08:24 2014
From: jeb at ponderworthy.com (Jonathan E Brickman)
Date: Sun, 31 Aug 2014 23:08:24 +0000
Subject: [LAU] Open Source Audio Interface (was Successor/replacement
 for RME HDSP+Multiface?)
References: <CAGFf1oixjChzHw9kJBpiwDkAaoZD1-yQcK6QVR+R8r5JFnm6dQ@mail.gmail.com>
 <alpine.DEB.2.10.1408310737500.2499@scott.cbbs.org>
Message-ID: <1ec8c3ec03704858b6d31a7571ca5843@Ex13DAG10-N1.dataoncloud.net>

 Just some quick thoughts, perhaps not as complete as they could be. -- Len Ovens
Doing pretty good I'd say, Len, I have been periodically studying this for quite a while.  And now that Firewire is going pass? I have to do it again :-)

There is one hardware approach which I have not seen at all yet: this is combination of (say) eight simple stereo USB interfaces, into one box.  Shouldn't it be fairly doable, to take eight satisfactory-quality USB interfaces, wire their timing chips together, write a custom driver, and go?  Or don't bother with the timing chips and the driver and use zita tools or multiple Jackd processes, and build this with a Raspberry Pi (or one of the more powerful act-alikes) as a jackd-over-tcp/ip audio appliance?

There is the fact, though, that USB2 and before, are monodirectional -- half duplex.  They transmit data only one direction at a time.  Yes, they flip back and forth very very fast, but no matter what, that's not good for us.  But USB3, happily, now gives us a full duplex capability.  Perhaps this will mean that once small-studio-priced USB3 multitrack interfaces come out, they may be simpler, and therefore potentially less expensive?

--
Jonathan E. Brickman
Ponderworthy Music | jeb at ponderworthy.com<mailto:jeb at ponderworthy.com> | (785)233-9977 | http://ponderworthy.com
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From jeb at ponderworthy.com  Sun Aug 31 23:11:02 2014
From: jeb at ponderworthy.com (Jonathan E Brickman)
Date: Sun, 31 Aug 2014 23:11:02 +0000
Subject: [LAU] SFZ on Carla -- how is it done?
Message-ID: <070530c6e093457eac328c20db64482a@Ex13DAG10-N1.dataoncloud.net>

Anyone know what Carla does when running SFZ's?  Does it encapsulate LinuxSynth or something else?  At 96 KHz I get a bit of background static all the time with an SFZ, but an SF2 is clear and beautiful.  Could it be a problem with the SFZ, or a problem adapting the SFZ to the high sampling rate?

--
Jonathan E. Brickman
Ponderworthy Music | jeb at ponderworthy.com<mailto:jeb at ponderworthy.com> | (785)233-9977 | http://ponderworthy.com
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From jeb at ponderworthy.com  Sun Aug 31 23:21:53 2014
From: jeb at ponderworthy.com (Jonathan E Brickman)
Date: Sun, 31 Aug 2014 23:21:53 +0000
Subject: [LAU] Yoshimi, in Carla?
Message-ID: <2d97bbdbc1814b4f91973e687768365b@Ex13DAG10-N1.dataoncloud.net>

I am fascinated by Carla's ability to call ZASFX.  How could this be implemented for Yoshimi?  Are there config files of some sort?

--
Jonathan E. Brickman
Ponderworthy Music | jeb at ponderworthy.com<mailto:jeb at ponderworthy.com> | (785)233-9977 | http://ponderworthy.com
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