Linux-audio-user February 2004

linux-audio-user@lists.linuxaudio.org

147 participants
192 discussions

[linux-audio-user] is (88200->44100) better than (96000->44100)?

by Andrew Gaydenko

Hi, The question is in subject. The situation is: I'm going to capture some of my old LP disks and (after some processing) write some mixes to CD. I'm able to record with any format from 24/96 and lower. Which recording samples rate and size are the best in accordance with next converting to CD format? Andrew

20 years, 9 months

Re: [linux-audio-user] Advice needed re: latency tuning/optimization, esp. video

by davidrclark＠earthlink.net

At the risk of confusing the issue even more or providing more misinformation: There is a lot of misunderstanding out there about the Live! cards, partly due to the lack of information from Creative Labs on the chipset. The internal clock runs at a fixed rate: 48 Ksamples/sec. All digital data are at this rate for all processes. This means that, for a recording session: The analog signal is sampled at 48 Ksamples/sec. If you store it at 44.1 Ksamples/sec, it is downsampled. When you play it back, it is read from the storage medium, upsampled to 48 Ksamples/sec, the converted to analog. This is why, IMO, it sounds so bad. It has been resampled twice. On the other hand, the card is capable of performing much better than this by working at 48 Ksamples/sec instead of 44.1 Ksample/sec. (48,000 is popular due to the large number of prime factors of two; but 44,100 is 2^2 * 3^2 * 5^2 * 7^2 so also has its advantages.) The Terratec EWX-2496, for example, is not fixed. The internal clock runs at whatever rate you specify. If you specify 44.1 Ksamples/sec, then this is what it runs at. Analog signals are sampled at 44.1 Ksamples/sec, then stored at this rate (assuming you've set it up that way). When they are played back, they are read at 44.1 Ksamples/sec, then converted back to analog. This results in more accurate sound --- what you hear on playback is more like what you heard when you did your recording. (I myself work at 24/96, mixing and all that, then downsample as the last step. I do this because I do a LOT of processing.) Now it is an interesting side effect that if you burn a CD, the Live! mixed result will sound better on another stereo system (one that runs at 44.1 Ksamples/sec) than it did when you were doing your mixing. Has anyone else noticed that? Suddenly things have improved. Well, that's to be expected because the signal was not upsampled. Although some people may be happy with this miraculous result, I myself am dissatisfied with this way of working because it never sounds like it what I heard previously. Playback for mixing sounds worse than what I heard during recording, yet the CD sounds better than the final mix. In summary, the Live! series of cards are providing you with very misleading information about the sounds you are working with if you insist on working at 44.1 Ksamples/sec. You'd probably be better off working at 48 Ksamples/sec, then downsampling. A major problem, though, is that many sample libraries are at 44.1 Ksamples/sec. In this case, you are forced into upsampling for audition and mixing! So... It all depends on what it is exactly that you are doing. This is definitely not a "one size fits all" situation. I wouldn't begin to tell anyone what they should do --- just provide some information about what is happening so they can make their own decisions. Do be careful!! As a footnote, Steve Harris mentioned that the Live! cards resample, even at 48 Ksamples/sec. Well, the signal path may include resampling, but technically, the interpolation algorithms used should reproduce the input exactly. So in fact, there should be no difference. In other words, don't be misled into thinking that resampling is occurring anyway, so you might as well work at 44.1 Ksamples/sec. No, you're worse off, assuming that there are no side issues such as heavy use of 44.1 Ksample/sec samples. Once again, it depends on exactly what it is that you are doing. I do have a Live! Value card myself, but avoid these issues by refusing to use it for serious audio work.

20 years, 9 months

[linux-audio-user] Re: cannot compile alsa for 2.6.2

by David Baron

Tried symlinks in the alsa source directory. No change. On Thursday 19 February 2004 13:55, linux-audio-user-request(a)music.columbia.edu wrote: > David Baron wrote: > > Contrary to popular belief, this kernel does not "come with" these > > built in! > > Last time I looked, it still did come with ALSA. Which doesn't mean > that it wasn't some obsolete version, so you might still want to > compile ALSA separately. The config file show nothing about ALSA at all. I would assume there would be entries such as: SND_******* y or m Actually, modules are being loaded but they may be coming from the older kernel. Alsa starts but the mixer fails to save or restore. /proc/asounc/cards has nothing. > > > So following the instructions in: > > http://www.linuxorbit.com/ > > modules.php?op=modload&name=Sections&file=index&req=viewarticle&artid=541 > >&page=1 (compiling on kernel-headers), I attempted to compile. cpp fails > > sanity check. I upgraded all the gcc stuff. Same. > > Does this still happen if you simply run "./configure" in the > alsa-driver package? If yes, what does configure.log say about this? That rules thingie IS running ./configure. I get the exact same message. The offender is this: | /* confdefs.h. */ | | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | /* end confdefs.h. */ | #ifdef __STDC__ | # include <limits.h> | #else | # include <assert.h> | #endif | linux/limit.h does not exist. As I am compiling off HEADERS, I would expect that the rules would define this directory as something. Usually, it is a symlink to source. I suppose I could set this up myself--easy enough. BTW, I tried to do this off source as well but got the same error. I did not have the symlink.

20 years, 9 months

[linux-audio-user] SoS DRM article now online

by Daniel James

Hello all, The piece I wrote for Sound on Sound about DRM for audio is now available online without subscription: http://www.soundonsound.com/sos/aug03/articles/drm.htm The arguments will be familiar to list members, but the piece was written for people who haven't been following DRM development closely. Cheers Daniel

20 years, 9 months

[linux-audio-user] Re: cannot compile alsa for 2.6.2

by David Baron

Straightened it all out -- get error "they are already there!". OK. (I do not expect the audio to work because Dman2044 is not yet supported but...) how do I get my snd-usbmidi and snd-mpu401 to work? The mixer fails on bootup (and on shutdown). proc/asound/cards has nothing.

20 years, 9 months

[linux-audio-user] Re: linux-audio-user Digest, Vol 5, Issue 60

by David Baron

On Thursday 19 February 2004 13:55, linux-audio-user-request(a)music.columbia.edu wrote: > David Baron wrote: > > Contrary to popular belief, this kernel does not "come with" these > > built in! > > Last time I looked, it still did come with ALSA. Which doesn't mean > that it wasn't some obsolete version, so you might still want to > compile ALSA separately. The config file show nothing about ALSA at all. I would assume there would be entries such as: SND_******* y or m Actually, modules are being loaded but they may be coming from the older kernel. Alsa starts but the mixer fails to save or restore. /proc/asounc/cards has nothing. > > > So following the instructions in: > > http://www.linuxorbit.com/ > > modules.php?op=modload&name=Sections&file=index&req=viewarticle&artid=541 > >&page=1 (compiling on kernel-headers), I attempted to compile. cpp fails > > sanity check. I upgraded all the gcc stuff. Same. > > Does this still happen if you simply run "./configure" in the > alsa-driver package? If yes, what does configure.log say about this? That rules thingie IS running ./configure. I get the exact same message. The offender is this: | /* confdefs.h. */ | | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | /* end confdefs.h. */ | #ifdef __STDC__ | # include <limits.h> | #else | # include <assert.h> | #endif | linux/limit.h does not exist. As I am compiling off HEADERS, I would expect that the rules would define this directory as something. Usually, it is a symlink to source. I suppose I could set this up myself--easy enough. BTW, I tried to do this off source as well but got the same error. I did not have the symlink.

20 years, 9 months

[linux-audio-user] idle JAMin eats CPU - is it for me only?

by Andrew Gaydenko

Hi, Idle JAMin (from current CVS) eats CPU: about 35%, if jack is started at 32/96, and about 20%, if jack is started at 32/44.1. Such CPU consumption takes place when there are no any jack client connected wih JAMin at all. Is it normal? Or - must I find something wrong in my configuration? I have P4 2.4GHz, 512MB DDRAM 333MHz. Andrew

20 years, 9 months

[linux-audio-user] Compiling alsa modules for 2.6.2

by David Baron

Contrary to popular belief, this kernel does not "come with" these built in! So following the instructions in: http://www.linuxorbit.com/ modules.php?op=modload&name=Sections&file=index&req=viewarticle&artid=541&page=1 (compiling on kernel-headers), I attempted to compile. cpp fails sanity check. I upgraded all the gcc stuff. Same. So ...

20 years, 9 months

RE: [linux-audio-user] suggested Window Manager good for audio?

by Matthew Allen

Ahh, funny you should ask this. I have been experimenting with both Fluxbox (http://www.fluxbox.org) and FVWM (http://www.fvwm.org) under CCRMA (fedora). I originally went with fluxbox but, at least for me, it actually had to much extra stuff (slit, tabs, icon bar). I really only needed a root menu, hotkeys and a pager, after a bunch of looking around I ended up with FVWM. Its learning curve is a bit daunting (this is linux isn't it?) but that is sort of mitigated by 2 things: The sheer amount of stuff you could do with it if you wanted, and the FVWM-themes project (http://fvwm-themes.sourceforge.net) Don't let the initial desktop scare you away from FVWM, it's so customizable you can make it look and work like pretty much any other WM out there. As far as resources go, neither of these are CPU hogs (FVWM does have some modules that potentially can eat up CPU). Installed FVWM takes up a bit more diskspace (even more so when using the themes package), but once again not enough space to warrant immediate dismissal. Fluxbox is much more easy to manage and it only took me about an hour 2 to get a theme that I liked and a full set of hotkeys and a CCRMA menu (the auto-menu scripts in both WM packages don't seem to find the CCRMA stuff, so you will have to edit by hand, relatively painless in fluxbox, a bit more time intensive in FVWM.) m. > -----Original Message----- > From: linux-audio-user-bounces(a)music.columbia.edu [mailto:linux-audio- > user-bounces(a)music.columbia.edu] On Behalf Of Kent, Gary > Sent: Wednesday, February 18, 2004 2:05 PM > To: linux-audio-user(a)music.columbia.edu > Subject: [linux-audio-user] suggested Window Manager good for audio? > > Hi: > I am trying out fvwm and it is pretty sparse, but was wondering > what others might suggest as best for audio? _________________________________________________ Scanned on 18 Feb 2004 22:30:28 Scanning by http://erado.com

20 years, 9 months

[linux-audio-user] Cannot compile alsa for 2.6.2

by David Baron

Get a C++ preprocessor fails sanity check.

20 years, 9 months

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Linux-audio-user February 2004