On Sat, 29 Nov 2003, Mark Constable wrote:
> > On Fri, 28 Nov 2003, Joerg Anders wrote:
> >
> > Forget this mail! I've got it!
> >
> > > ... yes if anybody explains the jack usage to me:
>
> But how about letting the rest of us know the exact procedure
> that worked for you :-)
>
> I'd love to put the "recipe" for how you did this on the
> Alsa Wiki with links to your page and mp3 examples.
>
Ok! A presupposition is: FluidSynth is compiled with jack support.
In first Xterm:
$ su -
# jackd -d alsa
In 2nd Xterm:
$ su -
# fluidsynth -a jack <soundfontfile>
In 3rd Xterm:
$ pmidi -list
Port Client name Port name
64:0 Rawmidi 0 - Audigy MPU-401 (UA Audigy MPU-401 (UART)
64:32 Rawmidi 0 - Audigy MPU-401 (UA Audigy MPU-401 #2
65:0 Emu10k1 WaveTable Emu10k1 Port 0
65:1 Emu10k1 WaveTable Emu10k1 Port 1
65:2 Emu10k1 WaveTable Emu10k1 Port 2
65:3 Emu10k1 WaveTable Emu10k1 Port 3
128:0 FLUID Synth (4187) Synth input port (4187)
(Thinking: Aha! The FluidSynth ALSA-MIDI input port is 128:0)
$ pmidi -p 128:0 <file>.mid (NO ENTER!!!)
In 4rd Xterm:
$ su -
# jack_lsp
alsa_pcm:capture_1
alsa_pcm:capture_2
alsa_pcm:playback_1
alsa_pcm:playback_2
fluidsynth:left
fluidsynth:right
(Thinking: Aha the 2 fluidsynth WAV output ports are named: "fluidsynth:left" and "fluidsynth:right")
# jackrec -f <file>.wav -d 5 -b 16 fluidsynth:left fluidsynth:right (NO ENTER !!!)
Then: ENTER in 3rd Xterm and immediately in 4rd Xterm.
Why I must be root ? ---> Don't know!
I assume "jackrec -f <file>.wav -d 5 -b 16 fluidsynth:left fluidsynth:right"
means : add fluidsynth:left + fluidsynth:right. Perhaps it is stereo ???
--
J.Anders, Chemnitz, GERMANY (ja(a)informatik.tu-chemnitz.de)
Has anyone else seen dropping of notes with EMU10K1 ALSA drivers mentioned by Joerg during wavetable synthesis?
I have seen dropping of notes with both EMU10K1 and ICE1712 EXTERNAL MPU-401, but I don't believe this is because of the drivers. Neither Muse nor Rosegarden function well on my machine, but pmidi has absolutely no problems. I have been assuming that these dropped notes are due to the larger load of GUI MIDI sequencers versus command-line sequencers, not the drivers. Rosegarden is worst, Muse next. BTW, Rosegarden is much, much worse for me than any Windows sequencers I have. I run 2.4.20 with the usual kernel patches for low-latency.
Just curious about what others have seen. Thanks!
-------- Original Message --------
> Subject: Re: [linux-audio-user] Help with basic midi configuration
> Date: Mon, 01 Dec 2003 09:01:21 +0100
> From: Robert Jonsson <robert.jonsson(a)dataductus.se>
> Reply-To: A list for linux audio users
> <linux-audio-user(a)music.columbia.edu>
> To: A list for linux audio users <linux-audio-user(a)music.columbia.edu>
> References: <3FCADB77.30705(a)domovolcanico.com>
>
> Hi Eduardo,
>
> A question, did you load any soundfonts?
>
> The SB-Live soundcard has built in synth capabilities but no internal
> sounds. You have to load a soundfont with the utility sfxload (available
> in the aweutils package). This also implies that you need to setup
> OSS-compatibility support in ALSA since the awe utils use that.
>
> I would also advice you to install a newer version of muse, check
> rpm.nyvalls.se/sounds9.1.html.
>
> Regards,
> Robert
>
> Eduardo Guerrero wrote:
>
>> Hi, I'm on my first attempt at midi on Linux. I'm using MusE
>> 0.6.0pre5 in Mandrake 9.1. I think I can start jack in a root
>> console, but I guess for a simple test it isn't necessary, so I start
>> muse and I get the GUI. I managed to record some notes but if I press
>> play on the transport I don't hear anything. I'm using a sblive
>> platinum and, for setting levels I'm using Alsa-Mixer GUI and for
>> connections I'm using Alsa Patch Bay. I have been tweaking everything
>> and haven't got any sound. It surely works since I can use XMMS and
>> also I can load and hear midi tracks on KMidi.
>>
>> In my MusE Midi Port Table I have port 1, generic midi, going to the
>> device EMU10K1 MPU-401 (UART) and the state shows as OK. Trying with
>> different settings I have seen a message about DEVICE PLAYING or
>> similar but can't remember what I did.
>>
>> On MusE the black activity meter is moving and mute cell has a green
>> LED (?) on, I don't know if its on why I see such activity. But
>> anyway I don't get any sound with it turned on or off.
>>
>> Any help would be appreciated,
>> Eduardo
>>
>
>
>
>
>
Thanks, now it's working. I did try to load a soundfont, but I had each
track strip turned off. The button seems pushed when it's off, contrary
to my belief.
For the moment it's just playing what I've recorded, is there a way to
play live with the soundfont sounds?
Later today I will try a newer muse version, as Robert suggests.
Thank you,
Eduardo.
Patrick Shirkey <pshirkey(a)boosthardware.com> wrote:
>
>So you are talking about designing an app which allows multitrack
>recording from a digital source but doesn't have any requirement to
>connect to a sound cards inputs?
I'm talking about a fancy phone answering machine where the phone
does not support modem voice commands. The application will accept
an incoming phone call on one of up to 16 phones and start an
application that stores the incoming voice into a WAV file.
The phone will generate an analogue signal, because the D/A step
accounts for an important part of the quality of the final voice, so
it will have to connect to a sound cards inputs.
Ideally we would use an artificial ear, but then we would not be able
place it into a 19" rack because of all the background noise.
>It should be relatively easy to design your own app/script which takes
>the raw data and writes it to multiple files.
I hope so. The fancy part is the voice quality assessment.
>IIRC ecasound can read raw data and write it to a multitrack file.
>
>http://www.wakkanet.fi/~kaiv/ecasound/
>
>
>--
>Patrick Shirkey - Boost Hardware Ltd.
>Http://www.boosthardware.com
>Http://www.djcj.org - The Linux Audio Users guide
>========================================
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Gene Heskett wrote:
> On Monday 01 December 2003 08:06, Patrick Shirkey wrote:
>
>>As some of you already know I am the occasional webmaster/docs
>>maintainer for ALSA.
>>
>>I have been planning some additions to the site to make it more
>>accessible but before I jump into anything that could be considered
>>a facelift I would like to get some opinions from y'all.
>>
>>Can you tell me what things you would like to have on the website
>>that are not already there? What things have you noticed while
>>using it that are not user friendly?
>>
>>I would like to add a database for some kind of card rating system.
>>
>>What would you consider valuable things to rate a card against?
>>
>>All opinions will be considered so don't hold back ;-]
>
>
> One thing that I didn't see the last time I was there, is a set of
> instructions on howto replace the now somewhat ageing version of alsa
> being shipped with the 2.6.0-test series of kernels, with the latest
> versions, including a patchfile to fix the .config too.
>
> This would be very very nice for us "bleeding edge" types. The 2.6
> kernel is a genuine winner so far in terms of user interactivity. I
> haven't booted to a 2.4 kernel more than 10 minutes to test something
> in a month or more.
>
This would be useful. Hopefully someone will find the time soon to make
a contribution. Any takers?
--
Patrick Shirkey - Boost Hardware Ltd.
Http://www.boosthardware.comHttp://www.djcj.org - The Linux Audio Users guide
========================================
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Tethered to the now toppled cart, the donkey was unable to escape before
the arrival of U.S. troops.
United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM
Patrick,
thanks for your quick reply. I'm not quite sure whether your answer
is really the answer on what I wanted to ask, so I'll clarify below.
Patrick Shirkey <pshirkey(a)boosthardware.com> wrote:
>
>Your be either one rme hammerfall + digital mixing desk with adat
>connectors. Otherwise you can use multiple cards but they need to be
>wordclock synced so you tracks stay in sample sync with each other. In
>that case you could use two rme hdsp multifaces, 2 midiman delta1010s
>and there are a couple of other 8 i/o cards with ALSA support.
I think to make sure that this is really what we need I'll have to
provide a better description of the problem. We are developing a
testbed for wireless voice transmission. We will have up to 16 active
phones, that are independently used to receive voice samples. We have
to record each voice transmission to a .WAV file (because this is what
the quality measurement tool understands).
To do this we will have to connect the 16 phones to a Linux box (prefered)
and need the ability to save the voice sample to a file. Each sample is
completely independent of all the other samples. Basically we need the
ability to do "sox /dev/phonespeakerX phoneXsample.wav" where 0 <= X <= 15.
>> Will we have 16 /dev/dsp devices [e.g. /dev/dsp0 .. /dev/dsp16]?
>>
>
>If you use jack you will have an easy to understand interface for
>routing the i/o's.
Following your suggestion I glanced at jack. I'm not quite sure whether
this is really what we are looking for.
Thanks in advance
Joerg
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Robert, Patrick,
this was what I hoped to learn. I'll do some more research
and suggest the Hammerfall for our project.
Thanks a lot and kind regards
Joerg
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Hi there,
A colleague of mine is having major problems using multiple (three) RME
Hammerfall cards using Mac OS 9 and X and consulted me for help.
The problem is that he has to transfer a few hours of 72-channel music
(recorded with Mackie HDR's) to his harddisk for further mixing and
processing.
I only have experience with (single and dual) stereo cards using alsa,
jack, ardour and ladspa. I wonder if anybody have experience with
multiple RME cards using alsa, and particullary experience doing such a
task with mentioned tools?
I haven't had the chance to examine his (Mac-related) problem but it
sounds like he might suffering from problems related to his
power-supply?
Regards,
Ronald van Engelen
Hi List,
for a project where we want to measure voice quality in a special network
we need a way to connect 16 audio sources to a computer running Linux.
What is the best way to do this? One card with 16 input channels? Which one?
Or multiple cards with less input channels (say 4 cards with 4 channels)?
Once again, which one?
Will we have 16 /dev/dsp devices [e.g. /dev/dsp0 .. /dev/dsp16]?
Thanks in advance
Joerg
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Joerg wrote:
>It has nothing to do with the sequencer. Please have a look at:
Please see Chris's comments about much more robust midi sequencing in the
latest release for Rosegarden. Apparently my problems with dropped notes
with EXTERNAL synths DO have something to do with sequencers, just as I
thought.
Joerg also wrote:
>The bug is well know by the driver developers. Please have a look at
In this posting, assuming that I found the correct place --- had to go
there indirectly --- there is no mention of the cause of the problem. The
description "well know(n)" appears to be overstatement. The testing
conditions were not specified, etc. So there is no way for any reader
to conclude that this is due to drivers rather than sequencer.
Again, I have never had a problem with pmidi and external synths. No
pitch-bending drops, etc., even though they do occur with very intense MIDI
under Windows, Rosegarden, and Muse. pmidi with external synths and emu10k1
drivers with SB Live! Value have never had a problem --- not once.
So I find the emu10k1 drivers to be fine for me so far.