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20639 discussions

[linux-audio-user] [ANN] simsam-0.1.7 released

by Christian Henz

simsam-0.1.7 was released. Changes include: - multiple instruments - multiple outputs (JACK only) - config loading (at last) - some fixes & cleanups - some new bugs ;-) Source tarball and i386 .deb for Debian/unstable are available at simsam's SourceForge download page. http://sourceforge.net/project/showfiles.php?group_id=65022 http://simsam.sourceforge.net/ cheers, Christian Henz

21 years, 5 months

RE: [linux-audio-user] Some music made with Linux

by Matthew Allen

I would also vote for multi sample support **BUT**, and I may be in the minority here, I dig the current simplicity of specimen. I don't need filters or LFO (I can roll my own in any number of programs). Fast amplitude envelopes would be nice (so I can actually create my own quick percussion sounds from bits of found noise) but outside of that I like how it is now. I love the easy of use and simplicity. If I need something more complex (like say CC control over loop points, or filters or lfo) I would just build it in PD. Its nice to just be able to quickly start up an instance or 2 of specimen and create fun little or synth patterns. m. > -----Original Message----- > From: linux-audio-user-bounces(a)music.columbia.edu [mailto:linux-audio- > user-bounces(a)music.columbia.edu] On Behalf Of Pete Bessman > > > (Just another todo item for your undoubtedly long list!) > > I think it's in the tarball if you care to see what I'm working > on. I'm pretty sure I can knock out most of the high-priority items > this weekend and make another release. Filters, envelopes, LFOs, > portamento, and a whole-9-yards approach to JACK are must-haves, and > will really chang things for the better IMHO. > _________________________________________________ Scanned on 19 Feb 2004 23:50:38 Scanning by http://erado.com

21 years, 5 months

[linux-audio-user] More Mandrake 9.2 fine tuning...Dell latitude live fx box

by Russell Hanaghan

Followed the tips everyone offered on my "Latency, Mandrake and Dell post...missed the emails because I was on "digest" delivery and had only signed up that day... Anyway, here's what I have so far for using my Latitude CPXj 650 as a live fx unit... I have found that the "Freeverb" LADSPA plugin is about the best. Does not seem to be a resource hog and has very clean and flexible digital reverbs. I was using Ecamegapedal to host the plugins but it seems a bit lacking. For example, I can only run one plug at a time. If I want to eq the reverb I dont think I can.(?) So here's what I settled on (The Ardour guys will probably not be impressed) I'm using Ardour as the host because the layout is killer and the flexibility is there (in Mixer window) to add plugins as if it were on a real console. I can add the Reverb in pre fader and a parametric post fade, etc and have complete control. The latency is probably running about 10ms in real world terms. It's not real noticeable in this application anyway. This is an extreme underuseage of Ardour as it was intended, but for now it's what I want. Good news is the realtime latency and sound of the reverbs, etc, is running better than Sonar XL on Win XP so thats a good thing! I am still unable to run Jack at anything less than 1024 without having Xrun hell. I set the HD to DMA "on" I had apm as opposed to acpi running...turned it off and found no difference. I set /tmp up as tmpfs in fstab per Austin's suggestion, I'm running in IceWM instead of KDE because it seems to use less resources. But I have not been able to improve the overall performance that much with these tweaks... I notice that running Ardour (Beta .9beta 9 RPM from Thacs RPMs), after I have fired up Jackstart or Jackd-realtime from Qjackctl with -d alsa hw:0 1024 2 as root and then fire up Ardour and add a track, the pcm chain is connected (also visible in Qjackctl "Connect")and I have sound but the pc starts running slow and the Hard drive light is constantly on. I'm not recording anything...is this Jack writing to disc? I only have 128 megs in this machine...is the virtual memory trying to substitute for Jack writing to ram? It is still useable but if there was a solution for this one item, I'm in digital fx heaven ready for my next live gig running this as my PA fx. TIA

21 years, 5 months

[linux-audio-user] high-quality resampling

by Florin Andrei

I'm searching for a way to resample audio while preserving the highest quality possible. One possible target would be the 96 --> 44.1 conversion, for obvious reasons, but other ratios are not excluded. I'm speaking mostly from an "audiophile" perspective; raw numbers are good ("while using this tool, you preserve such-and-such signal to noise ratio and such-and-such bandwidth...") but i'd also like to hear about musical tests, if anyone performed anything like that. What would be the Linux tools that are supposed to provide the highest-quality resampling (according to the above definition of "quality")? libsamplerate? Something else? -- Florin Andrei http://florin.myip.org/

21 years, 5 months

[linux-audio-user] I need long latency

by Andrew Gaydenko

Hi, The aim is: to capture some source using JACK. So, latency is not significant here. The question is: which case to prefer for jack server settings - 2048 frames/period and 4 periods/buffer, or 4096 frames/period and 2 periods/buffer? Or - are these cases equivalent (buffer size has the same size)? Andrew

21 years, 5 months

[linux-audio-user] Sounds Expo video

by Daniel James

Here's a short interview that took place last week on the linuxaudio.org stand at Sounds Expo in London: http://www.sonicstate.com/news/shownews.cfm?newsid=1373 It works as a stream with Real Player 8 - I didn't try the other formats. Cheers Daniel

21 years, 5 months

[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

21 years, 5 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.

21 years, 5 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.

21 years, 5 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

21 years, 5 months

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Linux-audio-user