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[linux-audio-user] notebook with midi

by Marco Milanesi

hi, anyone knows a notebook with midi capabilities? sorry for the ot. TIA, Marco

22 years, 6 months

[linux-audio-user] Capturing audio

by Levi Burton

Greetings, I am currently doing some stuff with rosegarden and digital audio. I have a composition which is a mixture of MIDI and audio. What I would like to do is "capture" incoming and outgoing sound. For example, the audio in rosegarden is sent to jack, which I assume is then sent to my pcm playback device. My MIDI stuff is an external synth going into my pcm input device (using my "line in"). I can hear my synth through my line in mixer setting, and I can hear the digital audio in rosegarden through my pcm mixer setting. What I want to do is route all of these signals to a another program that can record them, if that is possible. Is there some way I can tell jack to "listen to my line in and my pcm out, then route that to something else". Thanks -- Levi Burton http://www.puresimplicity.net/~ldb/

22 years, 6 months

[linux-audio-user] RE:linux-audio-user digest, Vol 1 #330 - 12 msgs

by RichardC＠europlex.ie

Hello linux-audio-user-request(a)music.columbia.edu I have received your e-mail regarding 'linux-audio-user digest, Vol 1 #330 - 12 msgs' I will be out of the office until the 24th of March. Please refer any queries that require immediate attention to Phil Carroll @ philc(a)europlex.ie Regards Richard Caldwell

22 years, 6 months

[linux-audio-user] Unwanted "tremolo" on mic-in with SB Audigy

by scott thomason

Greetings. I recently purchased a SoundBlaster Audigy, thinking it would be a good input device for use with my guitar (having had problems with on-board C-Media and my former Diamond MX-400). I am running it under alsa-0.9.0-rc6, with linux kernel 2.4.21-pre5 vanilla. While using various synth tools like ams or galan (and with the mic-in channel muted so I only hear the synth effect), I noted that I wasn't receiving any input at all from the audio input source. Upon further investigation, I noted that it was necessary for me to use KDE's mixer to bump up the "IGain" to a non-zero setting. Unfortunately, while this causes audio input to occur, it has an unwanted tremolo-like effect. Strangely, when I'm not using a synth tool and I'm just playing my guitar normally (with the mic-in unmuted and the IGain off), it sounds great. The card can accept input just fine. What's the deal with the IGain setting, and how can I set things up properly to avoid the "tremolo overload"? ---scott

22 years, 6 months

Re: [linux-audio-user] envy24control

by alex dray

hi, i don't mean to change the thread but, i was wondering if anyone knows (or has used) OSC with Pure Data before. I Am trying to set up a server to stream the output of a PD file, but am not quite sure of the logistics: i.e. icecast or shoutcast or what? >From: Luke Yelavich <lukethemuso(a)ozemail.com.au> >Reply-To: linux-audio-user(a)music.columbia.edu >To: linux-audio-user(a)music.columbia.edu >Subject: Re: [linux-audio-user] envy24control >Date: Mon, 17 Mar 2003 18:17:14 +1100 >MIME-Version: 1.0 >Received: from roar.music.columbia.edu ([128.59.195.116]) by >mc7-f14.law1.hotmail.com with Microsoft SMTPSVC(5.0.2195.5600); Sun, 16 Mar >2003 23:46:56 -0800 >Received: from roar.music.columbia.edu (IDENT:mailman@localhost >[127.0.0.1])by roar.music.columbia.edu (8.11.6/8.11.0) with ESMTP id >h2H7r0Q22751;Mon, 17 Mar 2003 02:53:00 -0500 >Received: from mta04.mail.mel.aone.net.au (mta04.mail.au.uu.net >[203.2.192.84])by roar.music.columbia.edu (8.11.6/8.11.0) with ESMTP id >h2H7n0Q22683for <linux-audio-user(a)music.columbia.edu>; Mon, 17 Mar 2003 >02:49:00 -0500 >Received: from luke-laptop.ozemail.com.au ([63.34.226.157]) by >mta04.mail.mel.aone.net.au with ESMTP id ><20030317074008.YSGH1381.mta04.mail.mel.aone.net.au(a)luke-laptop.ozemail.com.au> > for <linux-audio-user(a)music.columbia.edu>; Mon, 17 Mar >2003 18:40:08 +1100 >X-Message-Info: EoYTbT2lH2MsQxQLKd6QGpQxvU17UYmU >Message-Id: <5.2.0.9.1.20030317181622.00aa5a48(a)mail.ozemail.com.au> >X-Sender: lukethemuso(a)mail.ozemail.com.au >X-Mailer: QUALCOMM Windows Eudora Version 5.2.0.9 >In-Reply-To: <3E7499C0.8010209(a)kritek.net> >Sender: linux-audio-user-admin(a)music.columbia.edu >Errors-To: linux-audio-user-admin(a)music.columbia.edu >X-BeenThere: linux-audio-user(a)music.columbia.edu >X-Mailman-Version: 2.0.13 >Precedence: bulk >List-Help: ><mailto:linux-audio-user-request@music.columbia.edu?subject=help> >List-Post: <mailto:linux-audio-user@music.columbia.edu> >List-Subscribe: ><http://music.columbia.edu/mailman/listinfo/linux-audio-user>,<mailto:linux-audio-user-request@music.columbia.edu?subject=subscribe> >List-Id: A list for linux audio users <linux-audio-user.music.columbia.edu> >List-Unsubscribe: ><http://music.columbia.edu/mailman/listinfo/linux-audio-user>,<mailto:linux-audio-user-request@music.columbia.edu?subject=unsubscribe> >List-Archive: <http://music.columbia.edu/pipermail/linux-audio-user/> >X-Original-Date: Mon, 17 Mar 2003 18:17:14 +1100 >Return-Path: linux-audio-user-admin(a)music.columbia.edu >X-OriginalArrivalTime: 17 Mar 2003 07:46:56.0741 (UTC) >FILETIME=[626DE950:01C2EC59] > >Hi >My TerraTec DMX 6Fire 24/96 does the same thing. I haven't had any problems >with it, so I wouldn't worry. > >Regards >Luke > >At 02:35 AM 3/17/2003, you wrote: >>I have a Delta 44 card, when using envy24control it shows 10 PCM out >>channels, when I play an audio file that uses pcm out, all ten channels >>meters show activity, and any of these channels can set the volume for pcm >>out. My question, is there a way to disable any of these channels, a >>driver option maybe? My understanding of the card itself is that its >>4in/4out, thanks. _________________________________________________________________ Tired of spam? Get advanced junk mail protection with MSN 8. http://join.msn.com/?page=features/junkmail

22 years, 6 months

[linux-audio-user] envy24control

by Richard Dicaire

I have a Delta 44 card, when using envy24control it shows 10 PCM out channels, when I play an audio file that uses pcm out, all ten channels meters show activity, and any of these channels can set the volume for pcm out. My question, is there a way to disable any of these channels, a driver option maybe? My understanding of the card itself is that its 4in/4out, thanks.

22 years, 6 months

[linux-audio-user] RE:linux-audio-user digest, Vol 1 #329 - 16 msgs

by RichardC＠europlex.ie

Hello linux-audio-user-request(a)music.columbia.edu I have received your e-mail regarding 'linux-audio-user digest, Vol 1 #329 - 16 msgs' I will be out of the office until the 24th of March. Please refer any queries that require immediate attention to Phil Carroll @ philc(a)europlex.ie Regards Richard Caldwell

22 years, 6 months

[linux-audio-user] M-Audio Delta 44 and ALSA

by Richard Dicaire

Hi, new to the list, btw, there should be a search function built into the archives.... I have a working Delta 44 card using the ALSA drivers. However, using alsamixer and gamix, I have all these tracks and channels, no idea how/what they're for. The mixer interface is considerably different from the Windows control panel for the card. Do these drivers support the internal mixer on the card?

22 years, 6 months

[linux-audio-user] RE:linux-audio-user digest, Vol 1 #328 - 9 msgs

by RichardC＠europlex.ie

Hello linux-audio-user-request(a)music.columbia.edu I have received your e-mail regarding 'linux-audio-user digest, Vol 1 #328 - 9 msgs' I will be out of the office until the 24th of March. Please refer any queries that require immediate attention to Phil Carroll @ philc(a)europlex.ie Regards Richard Caldwell

22 years, 6 months

[linux-audio-user] ALSA, ens1371 and/or intel8x0, full-duplex, pcm back to input.

by Tomi Ollila

Hi I've been wondering/working a bit trying to use microphone and headphones at the same time in different Linux systems; I have exactly the same problem in both systems, and the "partial" fix works similarly in both systems. I've tried to find out documentation of the problem, with low success ratio; I'd be suprised is no-one else has same problems... maybe the search keywords I've been using has been poor choises... As a last resort I turn to this list ;/ The problem is when opening the audio device in full duplex mode the pcm output is feed back to the input; this continues even I mute the microphone input -- then I just can not feed more of my voice back to the "loop". So the problem can not be that the output coming out of headphones loops back to the microphone over the air -- and eventually I can verify this with a partial solution explained below. In the first place, to get full duplex audio working I installed alsa software to my machines. Both of the machines have Red Hat 8.0, and the alsa stuff was downloaded from www.freshrpms.net -- the latest version, alsa-kernel-0.9.1-fr1_2.4.18_26.8.0 was uploaded there in just a day or two ago... The machines were one desktop machine with soundblaster 64. More detailed information of it can be found at http://www.iki.fi/too/snd/info-64.txt The other one is Dell laptop, with Intel 8x0... More detailed information of that is at: http://www.iki.fi/too/snd/info-8x0.txt Note: Below I have not described which machine of the above I were using, for the simple reason, that the functionality was identical... To do my tests I began with $ arecord | aplay. Pretty soon I had horrible noise in my headphones, and started to adjust mixed settings with aumix (Since then I've used gnome-volume-control and alsamixer to setup, but with same results). The final values, with best results, of aumix settings are (graphical version at http://www.iki.fi/too/snd/aumix.png) vol 100, 100, R pcm 48, 48 speaker 0, 0 line 0, 0, P mic 100, 100, P cd 0, 0, P igain 100, 100 line1 0, 0, P phin 0, 0, P video 0, 0, P To get some delay between record and play I wrote simple utility program called delaypipe (http://www.iki.fi/too/snd/delaypipe.c). Delaypipe buffers the number of bytes given on the command line from stdin and feeds that to stdout after buffer is full. For example: $ arecord | ./delaypipe 16000 | aplay stored (about) 2 secs of data from arecord (default 8bits, 8000 hz, mono) before aplay gets the data. If I now say `foo' to the microphone, it is echoed after 2 secs -- and then again after 2 secs (with some additional noise, and then again and again... $ arecord -f S16_LE -r 44100 | ./delaypipe 176000 | aplay -f S16_LE -r 44100 does the same, with somewhat better sound quality. while reading the .asoundrc (which I could not understand much) there was an example $ arecord -f S16_LE -r 44100 -c 4 -D multi \ | aplay -f S16_LE -r 44100 -c 4 -D multi While this did not work: ALSA lib pcm.c:1906:(snd_pcm_open_noupdate) Unknown PCM multi ... and with removing -D multi from the above command line I could get only silence, after few trial&error tests I come up with: $ arecord -f S16_LE -r 44100 -c 3 | ./delaypipe 80000 | | aplay -f S16_LE -r 44100 -c 3 Now, if I spoke to the microphone, after a very short delay (1/3 of a second) I could hear myself speaking, in one of the headphone, *ONLY ONCE*. I had short fun with it, then gave headphones (with mic) to my 4 year old son -- he played with it about 10 minutes :D. At last I tested with $ arecord -c 3 | ./delaypipe 10000 | aplay -c 3 The difference was that now the sound came from the other headphone than with the previous command line. The interesting thing is that with the sb64 compared to intel8x0 the it was different headphone where I heard my voice (or at least I think it was so). This shows it is possible to record and listen sound through the soundcards I have at the same time. why simple arecord | aplay doesn't work properly I don't understand (due to lack of knowledge in this issue). Does anyone know (other) solutions how to get it working as I'd like. Preferably so that also oss -applications work. I need this stuff for VOIP purposes (is there clients that work with alsa out-of-the box and probably knows to tweak settings so that this works.) The only way I could do any solutions with my current knowledge would be as: $ arecord -c 3 | onechannel | gsmcompress | udpsend <host>/<port> & $ udpreceive <port> | gsmdecompress | threechannels | aplay -c 3 & where: onechannel strips the 2 other channels (how are the channels packed?) gsmcompress compresses the data with gsm... (or ADPCM or ...) udpsend ... udpreceive ... gsmdecompress ... threechannels adds 2 (silent) channels to the input data ... Anybody care to integrate all of this stuff to single `alsavoip' application. If not, and there is no no more suitable working voip applications I might take the task ACN. Anyway, any info to my problems is greatly appreciated. Tomi PS: If you get this mail twice, it is probably my fault.

22 years, 6 months

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Linux-audio-user