Message: 2
From: Ralf Mardorf<ralf.mardorf(a)alice-dsl.net>
>Wouldn't do it a simple instrument tuner for this purpose?
>http://gillesdegottex.github.io/fmit/ss.html
That would be great if this one was working like it show up on the screenshots. I have tried all the instrument tuners in Ubuntu Studio - Synaptic. Only the smallest GxTuner works but not easy to get in another say exact 435Hz calibration. That is also what my project need.
> But something as Voxengo Span would be fine to.
> http://www.voxengo.com/product/span/
> This seems to be no frequency counter.
Your are right, But I also need a to see harmonics and that is why I mention this one - you can easily zoom in on one particular frequency an it holds the frequency figure stable in view. Compair this with the Calf Equalizers witch can not zoom in on a specific frequency and move to much even by a singel sinus.
* Please help me on my way to learn how to response on this list?
Regards Crojav
------------------------------
Message: 3
From: "Chris Caudle"<chris(a)chriscaudle.org>
On Tue, April 25, 2017 12:03 pm, Crojav wrote:
> I am looking for a "Frequency Counter"
>From your example it looks like the software you want is called a spectrum
analyzer.
You are right on that - I am looking for both a good Frequency counter and a good - functioning spectrum analyzer.
>Try Jack and ALSA Audio Analyzer (JAAA).
Also this one I tried, I think I tried all there is - what i could find in and out side Linux - Ubuntu Studio - Synaptic - Internet.
The JAAA is not workable - it is more ore funny to see what it show, but not any good to work with. I think this software is old fashion, and not up to date - regarding what now a day is posible with software. That is also why I came up with the Foxengo Span, this is really workable all function do what they have to do. The only thing is I can't use it under linux.
>But of course frequency and spectrum only apply to audio, MIDI by definition just sends a note number which can be mapped to an expected frequency if you assume standard tuning based on A3=440 Hz, so I am not sure what the reference to MIDI in the original message meant.
Your right about that to. What i forgot to say was - that I also will count the frequency of the audio out of the ZynaddsubFx Synth. But there are way to measure midi data frequency. That why I asked for suggestions?
* Please help me on my way to learn how to response on this list?
Regards Crojav
Hi,
Bigups to @NovaDeviator
Some of you might be interested in a new distribution system for your
music based on the #stream2own mechanism.
http://resonate.is
#stream2own aims to give listeners and artists a fair deal. Each paid
stream cost exponentially more upto the 9th stream and then the track is
owned by the listener and they can download the track or stream for free.
A fully paid track costs E1.25.
Artists who upload tracks get access to the beta program to start earning
credits immediately.*
Resonate is a co-op where all the shareholders get one vote independent of
the number of shares owned AND artists, fans are also eligible for a share
of the profits. It's a new paradigm for music distribution where the goal
is to fairly reimburse artists/producers for their hard work instead of
enabling a few board members to get 6 figure salaries and private
yachts/jets/etc...
* Artists earn 70% of every paid stream of their tracks. have voting
rights and also get access to a portion of the total profits.
--
Patrick Shirkey
Boost Hardware Ltd
I am looking for a "Frequency Counter"
For a project I need to count frequencies of midi note's
Something simulair as this would be fine
http://www.qsl.net/pa2ohh/11freqcnt1.htm - ( this one is not working for
me - it works only with Pulsaudio and that is what i removed - only Jack
here. )
But something as Voxengo Span would be fine to.
http://www.voxengo.com/product/span/
I tried I think all I could find under Linux, Vst,Lv2,Ladspa and a lot
from the Internet. But with out succes. Hope that someone from here will
bring me in the right direction.
Greeting Crojav
Op 25-04-17 om 14:00 schreef linux-audio-user-request(a)lists.linuxaudio.org:
> Send Linux-audio-user mailing list submissions to
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> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Linux-audio-user digest..."
>
>
> Today's Topics:
>
> 1. spectmorph-0.3.2 (Stefan Westerfeld)
> 2. default Jack connection between pulse-audio and ecasound
> (john gibby)
> 3. Re: default Jack connection between pulse-audio and ecasound
> (Len Ovens)
> 4. Re: default Jack connection between pulse-audio and ecasound
> (john gibby)
> 5. OT: Video Card Recommendation (Ivan K)
> 6. Re: OT: Video Card Recommendation (Ralf Mardorf)
> 7. Re: OT: Video Card Recommendation (Dave Phillips)
> 8. Re: OT: Video Card Recommendation (Ivan K)
> 9. Re: OT: Video Card Recommendation (Ralf Mardorf)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 24 Apr 2017 14:30:55 +0200
> From: Stefan Westerfeld <stefan(a)space.twc.de>
> To: linux-audio-announce(a)lists.linuxaudio.org
> Cc: linux-audio-user(a)lists.linuxaudio.org, beast(a)gnome.org,
> linux-audio-dev(a)lists.linuxaudio.org, spectmorph(a)googlegroups.com
> Subject: [LAU] spectmorph-0.3.2
> Message-ID: <20170424123055.GA2247(a)space.twc.de>
> Content-Type: text/plain; charset=us-ascii
>
> spectmorph-0.3.2 has been released.
>
> Overview of Changes in spectmorph-0.3.2:
> ----------------------------------------
> * Added new unison effect.
> * New instruments: pan-flute, synth-saw.
> * UI improvements:
> - support operator folding (to preserve screen space)
> - provide scrollbar if morph plan window height is large
> - repair operator move
> * VST plugin crash fixed.
> * No longer depend on BEAST/Rapicorn
> - use libsndfile for sound file I/O, added WavData API
> - refactoring, move libnobse code into SpectMorph
> * Add icon/.desktop file for smjack
> * Added debian package support.
> * LPC/LSF morphing code updates - but now disabled by default
>
> What is SpectMorph?
> -------------------
> SpectMorph is a free software project which allows to analyze samples of
> musical instruments, and to combine them (morphing). It can be used to
> construct hybrid sounds, for instance a sound between a trumpet and a flute; or
> smooth transitions, for instance a sound that starts as a trumpet and then
> gradually changes to a flute.
>
> SpectMorph ships with many ready-to-use instruments which can be combined using
> morphing.
>
> SpectMorph is implemented in C++ and licensed under the GNU LGPL version 3
>
> Integrating SpectMorph into your Work
> -------------------------------------
> In order to make music that contains SpectMorph, you currently need to use
> Linux. There are four ways of integrating SpectMorph sounds into music you
> create:
>
> - LV2 Plugin, for any sequencer that supports it.
> - VST Plugin, especially for proprietary solutions that don't support LV2.
> - JACK Client.
> - BEAST Module, integrating into BEASTs modular environment.
>
> Note that at this point, we may still change the way sound synthesis works, so
> newer versions of SpectMorph may sound (slightly) different than the current
> version.
>
> Links:
> ------
> Website: http://www.spectmorph.org
> Download: http://www.spectmorph.org/downloads/spectmorph-0.3.2.tar.bz2
>
> There are many audio demos on the website, which demonstrate morphing between
> instruments.
My 10 year old ASUS EN7600GS SILENT/HTD Series with 256MB
of video ram is acting flakey (I have narrowed the issues
down to a hardware problem) and I am looking for a replacement.
What do people recommend?
I prefer fan-less cards. Thanks.
Hi,
I tried really hard to figure this out on my own... I have a patchbay
profile that is supposed to connect PulseAudio JACK Sink-01 to ecasound.
But, it doesn't do it; when the computer boots, the connection I see in
Jack is PA Jack Sink to system. I always have to manually remove that
connection, and manually connect it to ecasound (so my ecasound crossover
network works with mpv media player or whatever uses PA).
The patchbay profile does work to connect Pianoteq to ecasound; just not
Pulse Audio Sink to ecasound. Any ideas? I have gotten a little familiar
with the scripts used, like pacmd and pajackconnect. I tried to customize
pajackconnect, adding a new start_ecasound function that made this
connection, but so far have not gotten that to work...
Thanks,
John
spectmorph-0.3.2 has been released.
Overview of Changes in spectmorph-0.3.2:
----------------------------------------
* Added new unison effect.
* New instruments: pan-flute, synth-saw.
* UI improvements:
- support operator folding (to preserve screen space)
- provide scrollbar if morph plan window height is large
- repair operator move
* VST plugin crash fixed.
* No longer depend on BEAST/Rapicorn
- use libsndfile for sound file I/O, added WavData API
- refactoring, move libnobse code into SpectMorph
* Add icon/.desktop file for smjack
* Added debian package support.
* LPC/LSF morphing code updates - but now disabled by default
What is SpectMorph?
-------------------
SpectMorph is a free software project which allows to analyze samples of
musical instruments, and to combine them (morphing). It can be used to
construct hybrid sounds, for instance a sound between a trumpet and a flute; or
smooth transitions, for instance a sound that starts as a trumpet and then
gradually changes to a flute.
SpectMorph ships with many ready-to-use instruments which can be combined using
morphing.
SpectMorph is implemented in C++ and licensed under the GNU LGPL version 3
Integrating SpectMorph into your Work
-------------------------------------
In order to make music that contains SpectMorph, you currently need to use
Linux. There are four ways of integrating SpectMorph sounds into music you
create:
- LV2 Plugin, for any sequencer that supports it.
- VST Plugin, especially for proprietary solutions that don't support LV2.
- JACK Client.
- BEAST Module, integrating into BEASTs modular environment.
Note that at this point, we may still change the way sound synthesis works, so
newer versions of SpectMorph may sound (slightly) different than the current
version.
Links:
------
Website: http://www.spectmorph.org
Download: http://www.spectmorph.org/downloads/spectmorph-0.3.2.tar.bz2
There are many audio demos on the website, which demonstrate morphing between
instruments.
--
Stefan Westerfeld, Hamburg/Germany, http://space.twc.de/~stefan
Dear Linux Audio Community,
Linux Audio plays a major role in hearing aid research in the north-west
of Germany. The HoerTech gGmbH - a small non-profit research institute
for hearing aid system technology is offering again a developer
position. More details about the job are here:
http://www.hoertech.de/en/h%C3%B6rtech/career.html
Best regards,
Giso
DrumGizmo version 0.9.13 now available!
Get it at http://www.drumgizmo.org/
DrumGizmo is an open source, multichannel, multilayered, cross-platform
drum plugin and stand-alone application. It enables you to compose drums
in midi and mix them with a multichannel approach. It is comparable to
that of mixing a real drumkit that has been recorded with a multimic setup.
Included in this release is:
* Diskstreaming support, so you no longer need huge amounts of memory
to run it, even with the big drumkits.
* A completely new and redesigned plugin UI.
* And for the hardcore FreeBSD users: OSS support!
We know a lot of you have been waiting for diskstreaming to be
implemented, so please help us out by giving it a thorough test. We're
interested in hearing about your experiences with it, especially on low
memory setups with slower HDD's. Any comments and / or bug reports can
be directed to us on IRC, Freenode, #drumgizmo. Or feel free to ask at
the official DrumGizmo forum[1].
For the full list of changes, check the roadmap for 0.9.13 [2].
And now, without further ado, go grab 0.9.13!!! [3].
Important note to package maintainers:
Since version 0.9.11 we copy vst source files into the build tree while
building the vst plugin. This mean that should you wish to make a
tar-ball available with the build directory after the build has finished
this must either be stripped of said files or not be made public.
[1] https://linuxmusicians.com/viewforum.php?f=55
[2]
http://www.drumgizmo.org/wiki/doku.php?id=roadmap:features_roadmap#version_…
[3] http://www.drumgizmo.org/wiki/doku.php?id=getting_drumgizmo
On Tue, April 18, 2017 12:36 pm, Claus Lensbøl wrote:
> Hi Chris
I hope you do not mind, but I have taken the liberty of replying also to
the linux-audio mailing list. The question was originally asked there,
and many others are more experienced than I with netjack, so keeping
replies there is better to get the most accurate information, as well as
to benefit others who may have similar questions and search the mailing
list archives for possible solutions.
> I have on my master node (Ubuntu Studio 16.10), let the system start
> jack, and the loaded the netmanager as:
> jack_load netmanager
I see here a confusion possibly on my part. Your original email stated
that you wished to " set up a Raspberry pi as an output for jack"
Which is why I asked explicitly:
>>> So I'm trying to set up a Raspberry pi as an output for jack.
>>
>> Meaning that the R-Pi has the audio hardware, i.e. R-Pi is master?
By that do you mean the Rasberry pi has the only audio output? In that
case the Rasberry pi should be the master, the Rasberry pi must start jack
first with the ALSA driver (assuming R-Pi uses ALSA for the audio
hardware, I am not directly familiar with the details of that hardware).
After the master (the device with the audio hardware) has started jack
successfully then you use jack_load to load the netmanager.
> At this point I can on the Pi start jack as:
> jackd -R -d net
That would allow using software on the R-Pi to play audio on your Ubuntu
computer.
> , and have it connect to the master (where I, on the master, can see
> the Pi using qjackctl).
> This is where I get into the problem where running (on the Pi):
> jack_load audioadapter
You have configured the R-Pi as a slave of the Ubuntu computer, you do not
usually load audioadapter on the slave.
Well, I guess that is not entirely accurate. You would not load
audioadapter on the slave if you wanted one machine to have audio output,
and the other machines to be connected by network to that single audio
output.
What are you trying to accomplish? Which machine will generate digital
audio data, and which machine should output audio?
Are you perhaps attempting to have both the local audio device on your
Ubuntu computer play audio, and also the R-Pi available to play audio? In
that case what you are attempting may be the correct way. If you do not
need to use an audio interface on your Ubuntu computer then make it slave
only (use net as the driver instead of ALSA, use ALSA as the driver for
the R-Pi).
> , results in:
> could not load audioadapter, intclient = 0 status = 0x 1
Look also at the jackd status messages. I attempted to load audioadapter
on a machine where the only audio interface was already in use by jackd
and got this on the console where I executed jack_load:
$ jack_load audioadapter
could not load audioadapter, intclient = 0 status = 0x 1
But also saw these additional messages from jackd:
../linux/alsa/JackAlsaAdapter.h:225, alsa error -16 : Device or resource busy
That makes sense on my machine, the audio interface is already exclusively
used so audioadapter could not use the device.
Perhaps on your R-Pi there are informative messages as well.
> I think I was assuming that I could not play music from the Pi using
> jack_play when the Pi had another master, but I should be able to test
> that out tonight.
That previous suggestion was based on the assumption that the Pi was the
master since you stated you wanted the Pi to be the audio output.
I was assuming that the Pi would be the jack master, since typically in an
audio production setup where jack is often used you do not want the audio
to be resampled, so you have just one master, a single device with the
audio output.
>>> What I've done is to follow this guide:
>>> https://github.com/jackaudio/jackaudio.github.com/wiki/WalkThrough_User_Net…
Definitely check the output of jackd when using jack_load.
Section 6. has this note:
By default this client will open the same number of input/output ports the
net driver has opened and will use the sampling rate the net backend is
currently using.
Perhaps the netbackend is using a sampling rate not compatible with the Pi
audio hardware. Perhaps there is a mismatch with the number of ports, I
do not know from that description if the number of ports on the netjack
slave must match the number of physical audio channels available or not.
Did you try the suggested:
jack_load audioadapter -i "-h"
That may give information on the correct number of ports to use.
If you really do want to use the audio adapters of both devices (Ubuntu
computer and Pi) perhaps zita-net2jack (zita-n2j) will work better for
you:
http://manpages.ubuntu.com/manpages/xenial/man1/zita-njbridge.1.html
--
Chris Caudle
