Linux-audio-user December 2003

linux-audio-user@lists.linuxaudio.org

149 participants
190 discussions

RE: [linux-audio-user] rme Digi96/8 adat

by hayesjaj＠notes.udayton.edu

>> From: hayesjaj(a)notes.udayton.edu [mailto:hayesjaj@notes.udayton.edu] >> >> Have you managed to find a solution to this problem? I am >> having the same >> issues. >> >> Thanks, >> James >Most of my problems were fixed with the addition of three new features: > >1) Jack knows to use controller 0 when accessing device 1 >2) Jack makes a 16 bit connection when the connected interface doesn't support 32 bit >3) The ability to tell Jack how many ports to make available. > >What kind of problems are you having? >regards >-Reuben Specifically, I am interested in the third item. I have successfully captured and sent 2 channel audio from both dev 0 and 1, but I (as I am VERY new to ALSA and the .asoundrc voodoo) have not been able to use all 8 channels of ADAT optical io. The goal of my setup is to capture 8 streams into Ardour via jackd from the ADAT in but I have yet to figure out how to access all of the interleaved data. Please excuse me if my difficulty is naive but as I said I am new to the finer workings of alsa and am eager to get some direction beyond the fragments found in news group archives. Sincerely, James

21 years, 1 month

[linux-audio-user] jack-configuration

by Julien Claassen

Hi everyone! Do you know, if there is a pendant to the .asoundrc in jack. Can you write a .jackrc to declare named inputs/outputs with a specified number of channels? If this is possible, how? I.e. let's say I wanna combine my third and fourth channel on the soundcard (delta 1010lt ice1712) to one jack-input. Thanks for any help! Kindest regards Julien Julien Patrick Claassen jclaassen(a)gmx.de julien(a)c-lab.de http://www.geocities.com/jjs_home SBS C-LAB Fuerstenallee 11 33102 Paderborn Phone: (+49) 5251 60 6060 Fax: (+49) 5251 60 6065 www.c-lab.de

21 years, 1 month

Re: [linux-audio-user] How to record PCM signal?

by Eduardo Guerrero

I'm sorry, seems like my reply button only sends to the author and not to the list. Here's my reply: > hardware playback port. You can use the monitor ports as input > to qarecord to capture everything you hear from your playback > ports. > > I checked the monitor box in the options and I got 2 more outputs in qjackctl connection panel, however I can't find how to connect qarecord, it doesn't show anywhere and the meters don't move when I play. Maybe a command line option has to be enabled. I'm starting it with just qarecord & Thanks, Eduardo.

21 years, 1 month

[linux-audio-user] How to record PCM signal?

by Eduardo Guerrero

Hi, I'm trying to record what I play with Zynaddsubfx and Hydrogen in a stereo wav file but haven't got results so far using Audacity and QARecord. My card is a sblive and for everything it works fine. I've been testing different settings on the connection panel in qjackctl but still nothing. It seems like I'm not routing the audio as it should go. Any pointers? Thanks, Eduardo.

21 years, 1 month

RE: [linux-audio-user] rme Digi96/8 adat

by Reuben Martin

> From: hayesjaj(a)notes.udayton.edu [mailto:hayesjaj@notes.udayton.edu] > > Have you managed to find a solution to this problem? I am > having the same > issues. > > Thanks, > James Most of my problems were fixed with the addition of three new features: 1) Jack knows to use controller 0 when accessing device 1 2) Jack makes a 16 bit connection when the connected interface doesn't support 32 bit 3) The ability to tell Jack how many ports to make available. What kind of problems are you having? regards -Reuben

21 years, 1 month

Re: [linux-audio-user] rme Digi96/8 adat

by hayesjaj＠notes.udayton.edu

Have you managed to find a solution to this problem? I am having the same issues. Thanks, James

21 years, 1 month

[linux-audio-user] ALSA -> OSS DSP question

by Akos Maroy

Hi, I posted on this list a while ago, but still couldn't resolve one of my problems. I have a Dela 1010LT card, which works nice with the envy24 ALSA driver. What I want to achieve is to record from this soundcard through the OSS interface. I've been experimenting with /etc/asound.conf and the aoss utility, but to no avail. Basically what I want to achieve is to have a two channel /dev/dsp device, which corresponds to adc 3&4 on the Delta card. in /etc/asound.conf I have: # adcdac 3&4 pcm.channel2 { type plug ttable.0.2 1 ttable.0.3 1 slave.pcm ice1712 } which works nice when I use arecord -D channel2 ... now, what I want to work is something following: aoss sox -t ossdsp -r 44100 -c 2 -w /dev/dsp0 -t wav tmp.wav but no matter how I try to generate a pcm.dsp0 section in /etc/asound.conf, what I record is total silence :( as far as I understand, I need a pcm.dsp0 section, which would describe the /dev/dsp0 features. what I tried, among others is: pcm.dsp0 { type plug slave.pcm "channel2" } but this doesn't work. (it may be a problem that there is actually an OSS-driven /dev/dsp0 in the system. but setting pcm.dsp1 didn't work, it gave me non-existent device errors) can someone give me hints on setting up a proper pcm.dsp0 section?

21 years, 1 month

Re: Re: [linux-audio-user] Recording from SPDIF with SB live

by lau＠hippie-online.de

> lau(a)hippie-online.de a écrit : > > Is this a particular weakness of the SB live? What card supports exact > > recording from SPDIF-in? Christophe Vescovi wrote: > Yes this is a well known problem of the SB Live. The EMU10K1 Chips work > only at 48kHz and the SPDIF input is resampling the data to 48kHz > whatever is the input sample rate (even if this is 48kHz). So it's not > possible to do sample accurate digital copy from SPDIF-in. Oh God, what a botch! :-( > Most semi-pro > audio cards support exact recording from SPDIF, the M-audio cards for > example are well supported under Linux (the Audiophile is a good > candidate if you don't need multiple IO). Thanks a lot for your help! Ciao, HippiE

21 years, 1 month

[linux-audio-user] [ANN] Linux Audio Conference #2 - Call for Music

by Frank Neumann

Hi all, the following announcement gets sent to all of linux-audio-dev, linux-audio-user and linux-audio-announce mailing lists in order to reach as many possible interested parties as possible; I'm sorry if you receive this twice or even more often. I would be glad if we get a lot of participation from your side! Frank ----------------------------------------------------------------------- >From April 29th to May 2nd, 2004, the Institute for Music and Acoustics of ZKM Karlsruhe, Germany, will host the 2nd conference of the Linux Audio Developers (LAD). As a new feature there will be presentations of music in addition to technical talks. For this, we are looking for music that has been produced completely or mostly under Linux. We are looking for: * Interesting demos of sound synthesis, sound processing, etc. * "Classical" computer music compositions, to be played in a concert setting * Pieces from areas such as Electronica, Chill-Out, Ambient etc. If you would like to participate, please send your composition(s) to this address: Linux Sound Night ZKM, Institut fuer Musik und Akustik Lorenzstr. 19 D-76135 Karlsruhe Germany Please make use of one of the following media formats: - Audio-CD, DVD or CD-ROM Possible audio file formats: aiff or wav; mono, stereo or multi-channel; 44.1 or 48 kHz; 16 or 24 bit resolution. Please include the following items with your submission (in English): * A short commentary on the compositions * A short Curriculum Vitae * A completed and signed printout of the form available here: http://www.zkm.de/lad Deadline for submissions is February 29th, 2004. A jury will select the compositions that will be performed/played. The jury will award 3 grants to participants to contribute to their travel expenses. Terms and conditions for participation can be found in the form above. Up-to-date information about the conference is available here: http://www.zkm.de/lad

21 years, 1 month

Re: [linux-audio-user] Recording from SPDIF with SB live

by lau＠hippie-online.de

lau(a)hippie-online.de wrote: >> I would like to record directly from my DAT recorder (48 kHz) via SPDIF. >> As a greenhorn in harddisk recording I expect that there should be a way >> to get an exact copy of the data on the tape with no input level / mixer >> and no DA-AD conversion in between?! >> >> Unfortunately so far I did not succeed in recording from the SPDIF-input >> at all. I am just able to route the DAT signal to the analog line out by >> setting its volume using alsamixer. But the SPDIF-in does not appear in >> other mixers like kmix and the signal is not available in recording >> programs like audacity or krecord. :-( >> >> I am using: >> >> - SuSE 8.2 / kernel 2.4.20 / i86 >> - alsa 0.9.0 >> - emu10k1 >> - Soundblaster live! rev. 4 Joern Nettingsmeier wrote: > i don't have an spdif source, so i can't test, but it used to work on my > sblive when i set the capture flag in alsamixer on "IEC958 Coaxial" > *and* the "capture" channel. Yes, you are right, this works (and is not possible using alsamixergui) - thank you! *But* it definitely results in a DA-AD conversion because - I can record in arbitrary sampling rates without problems - When I stop the tape the input level remains about -70 dB Is this a particular weakness of the SB live? What card supports exact recording from SPDIF-in? Ciao, HippiE

21 years, 1 month

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Linux-audio-user December 2003