Linux-audio-dev May 2003

linux-audio-dev@lists.linuxaudio.org

71 participants
56 discussions

[admin] Re: [linux-audio-dev] AMERÝKA'DA YAÞAMAK ÝSTER MÝSÝNÝZ ?

by Joern Nettingsmeier

ABD/GÖÇMENLÝK wrote: [some turkish spam bs] shit. they sneaked it past the html filter. but i'm reluctant to plonk all multipart messages, many mailers produce them by default. guess we have to live with it for now. -- All Members shall refrain in their international relations from the threat or use of force against the territorial integrity or political independence of any state, or in any other manner inconsistent with the Purposes of the United Nations. -- Charter of the United Nations, Article 2.4 Jörn Nettingsmeier Kurfürstenstr 49, 45138 Essen, Germany http://spunk.dnsalias.org (my server) http://www.linuxdj.com/audio/lad/ (Linux Audio Developers)

20 years, 10 months

[linux-audio-dev] Hydrogen 0.8.0 released

by Comix

Hi all, the 0.8.0 version of Hydrogen Gnu/Linux drum machine is available at http://hydrogen.sf.net Features: * Graphical user interface based on QT 3, * Sample based audio engine, * Oss Audio driver, * Jack Audio driver, * Export to disk audio driver, * Alsa Midi input, * Ability to import/export xml-based song file, * 64 ticks per pattern, * 16 voices with volume, mute, solo, pan capabilities, * Import of samples in wave, au, aiff format. * Humanize and swing functions * Delay FX (new) * Assignable Jack ports in preferences file (new) * Assignable midi-in channel (1..16, ALL) (new) * Import/export of drumkits (new) Changes: * Delay FX * Bug fix in Alsa Midi Driver * Assignable Jack ports in preferences file * Assignable midi-in channel (1..16, ALL) * Drumkit support (load, save, import, export) * Acoustic drumkit included * various GUI improvements Happy drumming! ;) -- Alessandro <Comix> Cominu http://hydrogen.sf.net e-mail: comix(a)despammed.com Icq: 116354077 Linux User # 203765 [...Codito Ergo Sum...]

20 years, 11 months

[linux-audio-dev] Processing a chain of machines

by Raf Geens

Hi all, I'm currently working on creating Mess, a Buzz-like software studio written in C++ on Linux, and have run into a somewhat critical problem (note: this is my first "real" audio app): I'm using PortAudio as audio API and it's outputbuffer keeps underflowing because my callback function is too slow. One of my goals is to enable the user to make a network of synths, samples, effects,etc. ("machines" in Buzz-terminology). My initial idea for implementing this was to associate a buffer and a process-function with each machine. If the machine only generated sound, the process-function would simply fill the buffer, if it was an effect the buffer would contain the effect's input which would be consequently overwritten by the effect's output. The MachineManager class contained a processChain-function which was responsible for calling each machine's process-function in the right order and copying and mixing the contents of the buffers according to the connections in the network. PortAudio's callback called the processChain-function and next copied the contents of the master machine's buffer (the master represents the output in the network) to PortAudio's outputbuffer. As a test I made a simple sine tone generating machine and connected it to the master. This resulted in some weird noises when using small buffers and in clear sine tones with pauses inbetween when using very large buffers. In an attempt to let the callback do less work, I made an extra thread together with a set of two buffers with read/write flags, the idea being that the thread continuously tried to fill the buffers with the write flag set, using processChain (after this the buffer's flag would be set to 'read' ; if there was nothing left to write the thread would go to sleep) and the callback function given to PortAudio would check each time it ran for a readable buffer and copy that to it's own outputbuffer (and setting the just read buffer's flag to 'write', awakening the thread with processChain in it). This again resulted in some weird noises (mostly due to thread scheduling not quite going the way I expected). The problem with my first method seems to be the copying of all the buffers that makes the callback function too slow, but I don't immediately see any other way of doing it (is there one?). In retrospect the second way just seems wrong. So my question right now is: how can I process a chain of machines in a fast enough way? Raf

20 years, 11 months

[linux-audio-dev] libsndfile-python

by rm

hello again kids, if you like python and libsndfile and want to use python with libsndfile, you might try a preliminary set of bindings i put up at http://www.arcsin.org/archive/20030520025359.shtml . if you don't like python with/and libsndfile, then this has likely been a waste of your time. thanks, rob ---- Robert Melby #&)*&)!$_! !&$@*($(_)!#& !$&*(!@#$ uucp: ...!org!arcsin!rm Internet: yes

20 years, 11 months

[linux-audio-dev] steve's next plugin?

by Paul Davis

http://www.nytimes.com/2003/05/20/science/20OBSE.html?ex=1054008000&en=82a1…

20 years, 11 months

[linux-audio-dev] Re: Sblive on LAD (was: SBLive again)

by Lukas Degener

Linium wrote: >Hi, > >I saw your request about the crackling noise you get with the sblive card. > >I have such a beast too, and in the past i had problem with full-duplex at >44.1 khz. > >The Sblive work internally at 48khz and work very well at this samplerate >under Alsa, but at 44.1 khz some noise appeared. >May be it is the same problem ? try 48 khz and see if it solves it. > >Note: i don't know if this issue has been adressed now by the Alsa team, so >may be i am writting for nothing... but who know ? :) > >Linium > Just let me say thanks to Linium and others: The problem seems to have disapeared after i made jack use 48khz instead of 44.1khz I have to do some more testing, but it seems i can now drive my apps in full-duplex without major problems. Why the different behaviour of alsa and jack? I guess that was because the app i mainly used for testing (ams) would set the sample rate to 48khz when using alsa and accept the preset rate (44,1khz) when used with jackd. which lead me into the false assumption that there could be something wrong with jack. Glad that's solved! :-) Lukas

20 years, 11 months

[linux-audio-dev] Re: [Alsamodular-devel] subharmonics

by Alfons Adriaensen

On Wed, May 21, 2003 at 03:03:58PM +0200, Matthias Nagorni wrote: > On Wed, 21 May 2003, Lukas Degener wrote: > > > Matthias, is there a special (e.g. technical) reason for limiting the > > range of the sub-harmonics param to [1/1,1/16] ? > > Actualy i could use up (down?) to 1/32 or better 1/64. Will i experience > > any bad surprises if i change this in the code? > > No surprises. Go ahead ! I just thought 4 octaves down were sufficient, > but if you have a good subwoofer and patient neighbours... ;-) Be careful ! If like me you have vintage DC300A ready to deliver a kilowatt, and your system is fully DC coupled, watch out for the funny smell from you speakers..... FA

20 years, 11 months

[linux-audio-dev] new plugins

by Alfons Adriaensen

Hi list, Two new plugins are available at <alsamodular.sf.net>, together with the moogvcf plugin announced earlier. These are: cs_phaser: Similar the CSOund's phaser1 : a series of up to 30 first order allpass filters with modulation and feedback, cs_chorus: The three voice chorus from Sean Costello's "stringphaser" CSOund orchestra. Note: all three plugins (vcf, phaser anc chorus) may be used to generate singing birds :-) Enjoy ! Fons Adriaensen

20 years, 11 months

[linux-audio-dev] SBLive again

by Lukas Degener

Hello, please excuse me posting to both lad and jack-dev, i wasn't sure which one fits better. I am experiencing again what i may call "static crackling noises" when jackd with my sblive. They apear every now and then and go away after some time. The reason why i send this to the jack-dev list too, is that this does not happen when i use the same application (ams in this particular case, but also other jack clients like freqtweak or muse show this behaviour) without jack, i.e. directly interfacing the alsa layer. So could it be that jack does something which sort of provocates this crackling, something that does not happen when using alsa "directly" ? And yes, i am using a VIA chipset, first (on my old mobo) the KT133 and now (new mobo) the KT266. Reading through all kinds of web forums i have found this issues been discussed in length but without a clear result apart of statements like "Don't use SBLive!" or "Don't use VIA mobos!" (maybe both are right, but neither does much to make me feel better :-) ) Does anybody actualy _know_ what the reason for this noises is? Is there anything i can do? Would it help to tweak pci latency timers? I read somewhere that this solved the problem for someone using Win98, but i have to admit that even after reading two HOWTOs on this, i'm not realy understanding what this exactly does. Here are some additional observations i made, maybe this is of help: - After juggeling pci slots a bit, my sblive now uses irq 10 and does not share it with any other device. That did noticable improve the situation, i.e. crackling does apear less frequently. - As described above, the noise starts every now and then (i haven't yet found something that would proofable trigger it). First there are a few clicks and pops, then it becomes more "dense" and soon it sounds like some high frequent crackling. (dropped frames i suppose, lots of them). This lasts till jackd reports an x-run, and that's where it stops abruptly. (So there are situations when xruns become your friend :-) ) BTW, any idea how i could get more "run-time" info on what happens in my machine when this crackling starts? Is there a way to "see" irq activity? - The problem does not seem to be related to system load, it happens even in situations where jackd reports less than 5 percent cpu usage. - Sometimes when i create realy loud noises in ams, which result in hard-clipping somewhere in the signal chain, this sounds quiet similar. But i guess this hardly is related? Hope this wasn't too much of a redundant question. Kind Regards, Lukas

20 years, 11 months

[linux-audio-dev] Static class data and dlopen()

by Alfons Adriaensen

This is not strictly a Linux Audio question, but I encountered this matter in the context of writing LADSPA plugins, so... Imagine you have a C++ class S which has some static data. Derive A from S and use A to create a plugin a.so Derive B from S and use B to create a plugin b.so When a host links (at runtime) with both a.so and b.so, then the static data from S will be created twice (I verified this to be the case). This is what I expected, but it may shock C++ purists, so my question now is this: Is this the 'official' behaviour when loading plugins, or is this something that could change with future version of g++ and the DL libs ? Fons

20 years, 11 months

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Linux-audio-dev May 2003