Linux-audio-user January 2004

linux-audio-user@lists.linuxaudio.org

148 participants
168 discussions

[linux-audio-user] Live music with Linux.

by Patrick Shirkey

Not that many of you live in Korea but I am playing a live ambient set tomorrow night using only my computer. http://www.djcj.org/shevaka The software I will be using: spiralsynthmodular x1, alsaplayer x3, jackeq x1, and jack timemachine to hopefully record it all. I also have been working for the past month with ardour and audacity creating a lot of the samples which I will inject into the soundscape with alsaplayer. This will be my first real gig performing my music. Not just djing other peoples cuts (which I have been enjoying doing for the past 3 years). It wouldn't be possible for me to accomplish this without the awesome assistance from Steve Harris who designed the djeq plugin and dj flange. I'm also relying heavily on the latest cvs version of ssm as it has improvements to the jack plugin and echo plugin among other things which I will be thrashing heavily :). I feel this is important to let y'all know about because I have worked (and I mean work) for the past 4 years to get to this position and in the process I had a lot of fun with and help from many of you to get here. I chose not to invest in windows or mac apps 5 years ago and I am pleased that my gamble and investment in time and energy has paid off. Big ups to all the people who are following the dream through to concept through to reality. -- Patrick Shirkey - Boost Hardware Ltd. Http://www.boosthardware.com Http://www.djcj.org - The Linux Audio Users guide ======================================== Apparently upon the beginning of the barrage, the donkey broke discipline and panicked, toppling the cart. At that point, the rockets disconnected from the timer, leaving them strewn around the street. 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

20 years, 8 months

Re: [linux-audio-user] The trouble with disks

by Kjetil Svalastog Matheussen

John Anderson: > > I've had some more time to play around with things. Eventually I ended > up going back to 2.4.22 patched for low latency and pre-emption. 2.6.0 > and 2.6.0-mm2 would both xrun when I was recording. The 2.4 kernel will > record reliably, but I still get dropouts on disk reads. > > What's interesting is that if I'm running jack (0.93.6 now, but the last > few releases as well) with *no* connections (except for the automatic > capture and playback ports on the card, 12 and 10 respectively) and I do > anything that involves a fair amount of disk reading (starting ardour, > or cat some-file.wav >/dev/null), I'll get a bunch of xruns. > > jackd -R -d alsa -p 256 -n 3, running as root. > > It's been while since this thread was started, so here's a refresher on > the box: Uniprocessor Athlon XP 2200+, 1Gb RAM, 36Gb Ultra3 SCSI > (Adaptec 29160 with Fujitsu MAN3367 drive). MSI K7N2 Delta motherboard. > Terratec EWS88MT soundcard. Using reiserfs as the filesystem. > I'm very surprised that you are running stable(?) with this machine and a 2.4.22 kernel. I battled with two different kind of nforce2 boards earlier this year, no last year, without succeeding with the 2.4.22 kernel. There is something with the local apic on the nforce2 chipset that does not work with some/all 2.4 kernels. Running 2.4.23 and using the "nolapic" option on the kernel command line solves the problem completely. Perhaps that might be worth a try to solve the xruns problem? --

20 years, 8 months

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

by lau＠hippie-online.de

Jan Depner wrote: > >> Moreover I found out that xmms and audiacity play >> it a bit faster and higher than the original data, even though I always >> use the DAT's master clock! :-( > >The problem there is that your DAT is probably at 44100 and your card >defaults to 48000. Check in envy24control. You are right - sorry! When playing around I changed from a low level 48 kHz to a full level 44,1 kHz tape... Frank Barknecht wrote: > > Please upgrade... > > ... if alone because dmix'ing as I described in c't will not work with > this ALSA version and the Audiophile. ;) Hm, I am a bit averse from upgrading because I am afraid that this may result in conflicts with all that YaST stuff (and actually I do not need dmix for recording). Do I also have to build a new kernel to upgrade ALSA? >> Well, -f S16_LE -r 48000 -c 2 is identical to -f dat and actually also >> results in a mute file. > Your problem probably is, that you are *not* recording from your > digital input, but from the device called "default", which arecord > uses by default. "default" corresponds to "plughw:0W unless you > changed something in asoundrc (but you didn't do this). Yes, I think this is the reason for my problem. As I said there are no asoundrc files on my box. So I am going to learn about asoundrc and then create one. > Although I also have the Audiophile, I don't have any digital audio > gear, so I never tried to record from that and thus I don't know the > name of the digital ALSA device off-hand, but maybe someone else here > does? I am sure I can find this information somewhere on the ALSA webite. > BTW: OSS emulation on the Audiophile can be a bit tricky sometimes > because of the chipset, so you should try to use ALSA wherever > possible with this card. Sounds like a good idea to me. Anyhow I think that arecord is the perfect tool for recording from DAT - if it works... ;-) davidrclark(a)earthlink.net wrote: > Regarding low levels with some 24/96 cards: The inputs are lowered to 8.3% > to account for 12 (or so) channels so that clipping won't occur, I presume. > So if you have a 2-channel 24/96 card, your inputs are way too low > when using ICE1712, for example. (This is true for arecord, not qarecord.) > If you do arecord with verbose output (-v), you will see exactly what the > reduction is. I should mention that this is with analog --- I would expect > the same with SPDIF. Is this also true if you do not use the mixer? What a nonsense! :-( > Using qarecord, this problem doesn't exist. I looked at the code, but > again couldn't find where the input levels were maintained versus arecord > where they are lowered. At present I cannot access my Linux box to find out if qarecord is installed. I am going to check for this as soon as possible. Thank you all for your valuable help so far! I am confident of getting it running now. :-) Ciao, HippiE

20 years, 8 months

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

by davidrclark＠earthlink.net

Regarding low levels with some 24/96 cards: The inputs are lowered to 8.3% to account for 12 (or so) channels so that clipping won't occur, I presume. So if you have a 2-channel 24/96 card, your inputs are way too low when using ICE1712, for example. (This is true for arecord, not qarecord.) If you do arecord with verbose output (-v), you will see exactly what the reduction is. I should mention that this is with analog --- I would expect the same with SPDIF. Transformation table: 0 <- 0*0.0833333 + 1*0.0833333 + 2*0.0833333 + 3*0.0833333 + 4*0.0833333 + 5*0.0833333 + 6*0.0833333 + 7*0.0833333 + 8*0.0833333 + 9*0.0833333 + 10*0.0833333 + 11*0.0833333 I would be interested in how to alter the routing myself, if anyone has the information. I looked at some of the configuration files but could not immediately see how to do this. I would like NO reduction on inputs. It appears to have something to do with ttable routing and gain factors. Using qarecord, this problem doesn't exist. I looked at the code, but again couldn't find where the input levels were maintained versus arecord where they are lowered. Thanks to anyone for information on how to do this. I'm sure there are a number of folks out there who have cranked up their volumes, only to clip on the card, then lower them again, only to throw away perfectly good signal. But many of them may not realize that this is totally unnecessary.

20 years, 8 months

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

by lau＠hippie-online.de

Jan Depner wrote: > > What version of Audacity are you using? 1.1.1. Note that there is no problem in *playing* files with audacity. Harald Milz wrote: > >>In the meantime I tried to use arecord (which unfortunately does not >>provide man pages) > > It does - I have one on my machine! Ah yes, there is one on my one, too. Seems that I typed something wrong?! But man does not really provide more information than --help (at least in version 0.9.0rc7). :-( >>arecord -f S16_LE -r 48000 test.wav >> >>*but* this results in a mono file with too low signal level. > > You may have to set -c 2 as well to get stereo, this is not the default. Well, -f S16_LE -r 48000 -c 2 is identical to -f dat and actually also results in a mute file. > Alsio the hardware device maybe. In the meantime I checked modules.conf by the great c't article of Frank Barknecht. Everything seems to be okay there. But I could not find /etc/asoundrc or ~/.asoundrc. May be this is the reason for my problem - I am going to find out next. > As for low level - are you sure the DAT is correctly leveled? After all, > these tools do nothing with the level except if explicitly (and kindly) > asked. Yes, arecord seems to be exactly what I need for a 1:1 DAT copy. Nevertheless I still can only record mono files with definitely much lower level than the input signal (which takes the full range of the envy24control meters). Moreover I found out that xmms and audiacity play it a bit faster and higher than the original data, even though I always use the DAT's master clock! :-( By the way: Does anybody know, what HardwareSettings -> VolumeChange means in envy24control? Ciao, HippiE

20 years, 8 months

[linux-audio-user] Module options for usb-audio

by David Baron

So far, I have gotten to install on my Debian system, the MPU401 and USB-Audio drivers. Here is my cat /proc/asound/cards: 1 [UM1 ]: USB-Audio - UM-1 EDIROL UM-1 at usb-00:07.2-2 2 [UART ]: MPU-401 UART - MPU-401 UART MPU-401 UART at 0x300, polled Both are listed as RWxE (or similar) but I cannot remember how to get that at present. amidi -l cannot read the cards. (I still cannot get snd-cs46xx to install for my Dman2044 because the pci auto-config system thinks it has a Maestro AGOPO ESS chip. About to give up on this and install my old MediaVision just to have something on which to listen.) The MPU-401 is feeding a Yamaha sw60xg card. Yamaha did it this way for windows2000 so I tried it here. Windows programs running under WINE can play to the MPU401 beautifully. Trying to get MIDI in from the UM1 will hang the polling program. Nothing native to the Linux can record or play from either one of them. Are there any options that I need to feed the usb-audio to get that working? How do I use these devices with the Linux?

20 years, 8 months

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

by lau＠hippie-online.de

Mark Constable wrote: > > http://alsa.opensrc.org/index.php?action=find&find=envy24control > > Probably more exciting info somewhere but the above may help. Thank you. There is some information about installing and OSS emulation, but unfortunately nothing helpful about how to use envy24control. Anahata wrote: > > Does this have something to do with Audacity only working with OSS sound > drivers? > > I think it can be made to work, either by later versions of Audacity > having ALSA support or by using ALSA's OSS emulation, but as you can > probably tell, I haven't tried this. I am using Audacity with OSS, > which works fine for my purposes, including recording via S/PDIF input. Harald Milz wrote: > > man arecord maybe. In the meantime I tried to use arecord (which unfortunately does not provide man pages) to avoid OSS emulation problems: arecord -f dat test.wav produces mute files again. However it works with arecord -f S16_LE -r 48000 test.wav *but* this results in a mono file with too low signal level. I also did not succeed in playing around with the recording/muting/volume settings of alsamixer. :-( HippiE

20 years, 8 months

[linux-audio-user] Gating

by John Anderson

I'm trying to use the Gate plugins (from the swh plugin collection) with Ardour. Seems like that isn't quite the right place for them - there seems to be no place for the non-audio input / output? So what other app could I use them in? thanks John

20 years, 8 months

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Linux-audio-user January 2004