Linux-audio-user March 2003

linux-audio-user@lists.linuxaudio.org

134 participants
134 discussions

[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

21 years, 8 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.

21 years, 8 months

[linux-audio-user] jack connections

by Levi Burton

Greetings, Is there a command line interface to jack connections? Something similar to aconnect? qjackconnect is broken for me. Besides, I would like to know what is going on "under the hood" so to speak. -- Levi Burton http://www.puresimplicity.net/~ldb/

21 years, 8 months

Re: [linux-audio-user] Collaboration

by DevDSP Info

Hi all, I've got a scheme going that might be of interest to those people who are looking to collaborate on tracks: at http://src.devdsp.net I've set up a small web label type thing where every "release" consists of 20 MB worth of MP3s + the source files that were used to create these MP3s. So far, two people have opted to do this, one of whom is Dave/Nebogeo. It would be a possibility to set up a series of these things where each release takes what was done in a previous release in the series and build off that. If any of you are interested, drop me a line. There is one small but important restriction: I'm only going to put up five "releases" simultaneously at one time. When number six comes around, number one gets taken down. Sorry about that, but it's well beyond my financial means to pay for the bandwidth involved in hosting gigabytes worth of audio. Take care, Matthijs de Jonge http://devdsp.net - news and resources for computer musicians

21 years, 8 months

[linux-audio-user] RE:linux-audio-user digest, Vol 1 #327 - 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 #327 - 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

21 years, 8 months

[linux-audio-user] Software for cutting interviews

by Ralf Haller

Hi there, I am running a radio show on a local radio station where I am broadcasting interviews. For cutting these interview I am looking for some software to get this task done under LINUX. In the past I used a tool called ddclip under Windows, but I completely switched over to LINUX. What I need now is a software to cut the interview in pieces, remove unwanted stuff and put it all together again. Some DSP effects would be nice to. I tried to get ardour to work, but didn't succeed yet. Is there any other LINUX software around? Thanks for your Help, r(a)l.f

21 years, 8 months

[linux-audio-user] now streaming: linux audio developers meeting at zkm in karlsruhe

by J�rn Nettingsmeier

hi everyone ! i'd like to invite you all to tune in to our live streaming program from the linux audio developers meeting at the zkm karlsruhe. please visit http://www.linuxdj.com/audio/lad/eventszkm2003.php3 for a list of streaming mirrors and additional material (slides). dust off your mp3 players, it's only 32 kbit/sec ! currently speaking is alsa core developer takashi iwai, further speakers will include paul davis and dave philips. best, joern J�rn Nettingsmeier Kurf�rstenstr 49, 45138 Essen, Germany http://spunk.dnsalias.org (my server) http://www.linuxdj.com/audio/lad/ (Linux Audio Developers)

21 years, 8 months

[linux-audio-user] GWC quits

by Reid Vail

I'm running GWC and have run into a problem with the app just aborting. Quits in the midst of the Pop/clicks removal routine. It's consistant. The hardware I'm running is an Athlon 850 with 256k RAM. The sound card is an SB16 ISA. I know it's not much, but it was free, back a few years ago. Now I wonder if the routines that GWC is performing are just too much for that little engine to execute, so the program aborts. Any advice would be appreciated. RV

21 years, 8 months

[linux-audio-user] Dirty, Nasty Raunchy Guitars

by Levi Burton

Greetings, Searched the list archives but found nothing appropriate, so I thought I would just ask. I am a metal guitar player. I like a heavily saturated guitar tone. Could someone point me in the right direction for processing a clean guitar signal in real-time such that it can (even partially - i like weird sounding stuff) emulate tube circuitry. I am thinking a chain of LADSPA plug-ins would do the job. The sound im looking for doesn't have to emulate anything to a T, it just has to be raunchy (think Nine Inch Nails - Broken, or any Nine Inch Nails album for that matter). Thanks! -- Levi Burton http://www.puresimplicity.net/~ldb/

21 years, 8 months

[linux-audio-user] vsound

by jdennett＠science.uva.nl

>> cygnus:~/mars> vsound --timing -f output.wav realplay >> _1790918_blur_beagle_vi.ram >> About to start the application. The output will not be available until the >> application exits. >> /usr/bin/vsound: line 163: 1366 Aborted >> LD_PRELOAD="$pkglibdir/libvsound.so" "$@" >I find I have to use vsound -t hi, vsound -t and vsound --timing are meant to be synonyms, and indeed both give me the same problem. has anyone else had this problem? i don't even know what it means. why does it abort on this line? cheers jane

21 years, 8 months

← Newer
1
...
6
7
8
9
10
11
12
13
14
Older →

Jump to page:

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

Linux-audio-user March 2003