Linux-audio-user January 2005

linux-audio-user@lists.linuxaudio.org

207 participants
246 discussions

[linux-audio-user] Re: [Jackit-devel] Re: [linux-audio-dev] [ANNOUNCE] jack_convolve-0.0.1

by Jack O'Quin

Fons Adriaensen <fons.adriaensen(a)skynet.be> writes: > On Sun, Jan 30, 2005 at 10:23:29AM -0600, Jack O'Quin wrote: > >> One simple improvement I'd recommend. JAMin runs its DSP thread at a >> RT priority one less than the JACK process thread. It would be better >> to add (log2(bufsize)-log2(chunksize)) to the JACK priority (only when >> negative). Then, multiple JACK client helper threads could share RT >> scheduling cooperatively, so the ones with lower scheduling deadlines >> get to run first. I think that works for any collection of sizes, >> assuming the total CPU bandwidth is sufficient. > > That would certainly help in the case you decribe, but ISTR (should > search for my notes from previous lifes), there is a limit of about > 70% CPU load where such a scheme will break down. It's probably > quite hard to improve on this with fixed priorities. True, but in many cases 70% isn't bad. > IIRC, the reason it breaks down well before 100% load is that the > priorities should really be a dynamic function of how far the > deadline actually is at any time, rather than being fixed. > For example, a calculation for a relative chunk size of 8 that has > one period left is more urgent than one for a size 4 chunk that > still has three periods until its deadline. Right. I haven't done the math, but my intuition is that in the special case of JACK clients and various larger power-of-two chunk sizes, this scheme should work very well. Maybe even optimally. > To avoid this limit, the work should be divided evenly over the > periods that are available to do it, and in that case you don't > even need a second thread. But that's probably very hard. Sure. In the case of JAMin it was dictated by our desire to use FFTW rather than writing our own FFT and trying to figure out how to divide the computation into relatively equal parts. -- joq

20 years, 4 months

[linux-audio-user] software for a live audio desk

by Anthony Baxter

Hi there, I'm looking for software for driving a sound system during a live performance (of a play). All the sound is pre-recorded (no live mikes) but it should be able to deal with multiple sound devices (for different channels), and live mixing (e.g. backing music, then sound effects over the top, cued manually). I can't find anything that seems to fit this - am I going to have to write it myself? Thanks, Anthony -- Anthony Baxter <anthony(a)interlink.com.au> It's never too late to have a happy childhood.

20 years, 4 months

[linux-audio-user] Speeding up Speech

by Phillip Blevins

Hi, I've got a lot of speeches on cassette tape that I have converted to .wav and cleaned up a bit. I remember reading somewhere that in radio broadcast they will record talk show hosts and then somehow shorten the space between words to that there is more room for commercials. Has anyone done this or heard of a tool that does this? I would also like to split the speeches up into about 1 minute sections breaking between words. Also, I've got about 4,000 cassette tapes I would like to digitize. I can do about 1.5 a day with it playing at regular speeds. If I do it about 200 days out of the year I figure it will take me about 13 years!! Do anyone know of a handy way to digitize at high speeds. I apologize in advance if someone has already asked these questions but I was unable to use the search utility on the list serve. Thanks, Phillip

20 years, 4 months

[linux-audio-user] Re: [PlanetCCRMA] Why we we have to use Red Hat?

by Barton Bosch

Dave Phillips wrote: > Matt Barber wrote: > >> Do you want to completely uninstall gnome (including gdm), or would just >> installing fluxbox work? You can grab fluxbox from many >> apt-repositories.. >> I think freshrpms has one (and I think Fernando's sources.list that >> you get >> when you install apt from ccrma has freshrpms as a commented option). >> > No, I don't want to completely uninstall GNOME, I assume there are apps > in the Planet that depend on it, and I may want to build other > GNOME-dependent apps in future. Main Menu->Gnome Configuration Editor->Apps->Nautilus->Preferences->Always_Use_Browser and you're back to the old school style nautilus. Barton

20 years, 4 months

[linux-audio-user] Way to suspend Jack clients

by JP Mercury

Hi everyone, I'm looking for a way to suspend Jack clients so that their DSP process() callbacks are temporarily bypassed and outputs replaced with silence. I'd like to do this from a server level. I see using various softsynths and apps together in a live situation. This is part of refining my toolkit of instruments that I use for looping and performance. What I'd like to do is suspend apps that I'm not currently using to keep CPU cycles down. For example, AMS alone eats 50% for those fabulous analog sounds in 4 voices. I'd like to be able to switch between AMS, Beatrix, Aeolus, Hexter, and other apps without all that loading, exiting, and reconnecting of ports. As far as I know there is currently no way to tell the Jack server to suspend clients. Short of patching each of my favorite apps to do the suspend, can anyone see a way?? Or perhaps Paul would consider adding this functionality to Jack. Thank You & Kind Regards, -Mercury

20 years, 4 months

[linux-audio-user] Mic recording issues

by Hans Fugal

I was trying to do some basic mic recording last night, and I was having strange problems. My sound card is a turtle beach santa cruz using the snd-cs46xx module. Audacity had no trouble recording - sounded fine. Timemachine on the other hand experienced horrible hissing. I tried every conceivable mixer level/muting/gain combination I could think of and always got the awful hiss. Why might the same mixer settings work great in Audacity which uses OSS (via the emulation), and not in Jack? -- De gustibus non disputandum est.

20 years, 4 months

[linux-audio-user] FW: iPods boot Linux

by JakeHarries＠aol.com

Subj: [microsound] iPods boot Linux Date: Sun, 30 1 2005 1:32:31 am GMT From: Phil Thomson <hellomynameisphil(a)gmail.com> To: microsound <microsound(a)hyperreal.org> Reply-To: microsound <microsound(a)hyperreal.org>,Phil Thomson <hellomynameisphil(a)gmail.com> Sent from the Internet (Details) http://apple.slashdot.org/article.pl?sid=05/01/29/2017244 -- Phil Thomson hellomynameisphil(a)gmail.com -- best, Jake

20 years, 4 months

[linux-audio-user] Traktion

by David Baron

This one may well be the first fully capable audio/midi DAW program to work in both Windows and Linux. This is because it is not based on Microsoft's framework. I got the freebie while this offer was around. Version 2 is supposed to come out soon. (I presently use Cakewalk Home Studio--Sonar's not so little sister and a lot of bang for the buck. I have gotten used to drawing automation envelopes rather than attempting to record them. Traktion offers recording only.) The program almost works using WINE but if it can be gotten to work this way, I would not expect acceptable audio performance. I would love to see it play, though. Two problems persist: 1. Child pop-up menus do not appear, rendering most functions unaccessible including finding a file to open. Others have reported and not reported this problem so may be fixable. 2. The program detects MIDI devices properly allowing WINE (using alsa or OSS interfaces) to feed it the correct devices. It detects audio devices by looking for drivers in the windows directory. How do I know? It finds a Delta44 asio driver even though I do not have the card, just the file. This is a very bad and unportable approach, obviously.

20 years, 4 months

[linux-audio-user] [ANNOUNCE] jack_convolve-0.0.1

by Florian Schmidt

Hi, i implemented a small convolution engine for jack.. grab it at http://affenbande.org/~tapas/jack_convolve-0.0.1.tgz untar make ./jack_convolve responsefile.wav It creates as many in/out ports as the response file has channels. Uses fftw3f and libsndfile. Will add libsamplerate support in near future. So for now, make sure the samplerate of jack and the response file match. Here's a ca 1 sec 48khz (resampled from 96khz) stereo response file of a room: http://affenbande.org/~tapas/FrontFacing%20TLM170.wav It's from this package: http://www.noisevault.com/index.php?page=3&action=file&file_id=130 which has 96khz responses.. Consumes ca. 25-30% cpu load on my 1.2ghz athlon at jack buffer size 2048 [;)] So there's plenty room for optimization (and some return value checking will be added too ;)).. If you know some tricks, let me know.. The sourcecode is pasted below for easier reference. Flo P.S.: thanks to mario lang for collaborating and giving some hints towards using fftw3f instead of fftw and some other optimizations.. P.P.S.: oh yeah, example sound, here you go [hydrogen dry then with output of jack_convolve mixed to it]: http://affenbande.org/~tapas/jack_conv_ex1.ogg And here the convoluted signal alone: http://affenbande.org/~tapas/jack_conv_ex2.ogg P.P.S.: known issues: - won't handle samplerate or buffersize changes gracefully - will bring your machine to a crawl ;) jack_convolve.cc: --------------- /* Copyright (C) 2004 Florian Schmidt This program is free software; you can redistribute it and/or modify it under the terms of the GNU Lesser General Public License as published by the Free Software Foundation; either version 2.1 of the License, or (at your option) any later version. This program is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser General Public License for more details. You should have received a copy of the GNU Lesser General Public License along with this program; if not, write to the Free Software Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. $Id: types_8h-source.html,v 1.1 2004/04/27 18:21:48 joq Exp $ */ #include <jack/jack.h> #include <iostream> #include <sstream> #include <unistd.h> #include <signal.h> #include <stdio.h> #include <sndfile.h> #include <vector> #include <cmath> #include <fftw3.h> jack_client_t *client; std::vector<jack_port_t *> iports; std::vector<jack_port_t *> oports; jack_nframes_t jack_buffer_size; int chunks_per_channel; // channel chunk data std::vector<std::vector <fftwf_complex*> > chunks; // the buffers for the fft float *fft_float; fftwf_complex *fft_complex; // the plan fftwf_plan fft_plan_forward; fftwf_plan fft_plan_backward; float normalize_factor; // per channel we need a ringbuffer holding the fft results of the // audio periods passed to us by jackd. // each needs to be sized jack_buffer_size * chunks_per_channel (see main()) std::vector<fftwf_complex *> ringbuffers; // this gets advanced by jack_buffer_size after each process() callback unsigned int ringbuffer_index = 0; // a vector to hold the jack buffer pointers.. these get resized // during init in main std::vector<jack_default_audio_sample_t *> ibuffers; std::vector<jack_default_audio_sample_t *> obuffers; // channel data std::vector<jack_default_audio_sample_t *>overlaps; int process(jack_nframes_t frames, void *arg) { // std::cout << " " << ringbuffer_index; // get pointer[s] to the buffer[s] int channels = chunks.size(); for (int channel = 0; channel < channels; ++channel) { ibuffers[channel] = ((jack_default_audio_sample_t*)jack_port_get_buffer(iports[channel], frames)); obuffers[channel] = ((jack_default_audio_sample_t*)jack_port_get_buffer(oports[channel], frames)); } for (int channel = 0; channel < channels; ++channel) { // copy input buffer to fft buffer for (int frame = 0; frame < jack_buffer_size; ++frame) { fft_float[frame] = (float)(ibuffers[channel][frame]); fft_float[frame+jack_buffer_size] = 0.0; } // fft the input[s] fftwf_execute(fft_plan_forward); // store the new result into the ringbuffer for this channel for (int frame = 0; frame < jack_buffer_size * 2; ++frame) { ringbuffers[channel][ringbuffer_index+frame][0] = fft_complex[frame][0] / normalize_factor; ringbuffers[channel][ringbuffer_index+frame][1] = fft_complex[frame][1] / normalize_factor; } // zero our buffer for the inverse FFT, so we can simply += the // values in the next step. for (int frame = 0; frame < jack_buffer_size * 2; ++frame) { fft_complex[frame][0] = 0; fft_complex[frame][1] = 0; } // multiply corresponding chunks of the fft'ed response[s] // we start with the chunk for the current part of the response and work our // way to the oldest data in the ringbuffer (we need to go backwards for that) for (int chunk = 0; chunk < chunks_per_channel; ++chunk) { // we go backwards and constraint to the whole buffersize ("%") long int chunk_rb_index = ((ringbuffer_index - (2 * chunk * jack_buffer_size)) + 2 * chunks_per_channel * jack_buffer_size) % (chunks_per_channel * jack_buffer_size * 2); for (int frame = 0; frame < jack_buffer_size * 2; ++frame) { // complex multiplication (a+bi)(c+di) = (ac - bd)+(ad + bc)i long int running_ringbuffer_index = chunk_rb_index + frame; float a,b,c,d; a = ringbuffers[channel][running_ringbuffer_index][0]; b = ringbuffers[channel][running_ringbuffer_index][1]; c = chunks[channel][chunk][frame][0]; d = chunks[channel][chunk][frame][1]; fft_complex[frame][0] += (a * c) - (b * d); fft_complex[frame][1] += (a * d) + (b * c); } } // inverse fft the input[s] fftwf_execute(fft_plan_backward); // copy fft result to output buffer for (int frame = 0; frame < jack_buffer_size; ++frame) { obuffers[channel][frame] = (jack_default_audio_sample_t)(fft_float[frame] / normalize_factor); } // add previous overlap to this output buffer for (int frame = 0; frame < jack_buffer_size; ++frame) { obuffers[channel][frame] += overlaps[channel][frame]; } // save overlap for (int frame = 0; frame < jack_buffer_size; ++frame) { overlaps[channel][frame] = fft_float[frame+jack_buffer_size] / normalize_factor; } } // advance ringbuffer index ringbuffer_index += jack_buffer_size * 2; ringbuffer_index %= jack_buffer_size * 2 * chunks_per_channel; return 0; } bool quit = false; void signalled(int sig) { std::cout << "exiting.." << std::endl; quit = true; } int main(int argc, char *argv[]) { // we need to become jack client first so we can ask for the buffer // size. std::cout << "jack_convolve (C) 2004 Florian Schmidt - protected by GPL2" << std::endl; if (argc < 2) { std::cout << "usage: jack_convolve responsefile.wav" << std::endl; exit(0); } // hook up signal handler for ctrl-c signal(SIGINT, signalled); client = jack_client_new("convolve"); jack_buffer_size = jack_get_buffer_size(client); normalize_factor = sqrt(2.0 * (float)jack_buffer_size); std::cout << "buffer size: " << jack_buffer_size << std::endl; // first we load the response file. we simply assume it has // the right samplerate ;) the channel count of the // response file governs how many Ins and Outs // we provide to jack.. // filename of the soundfile is the first commandline // parameter, argv[1] struct SF_INFO sf_info; SNDFILE *response_file = sf_open (argv[1], SFM_READ, &sf_info) ; // register ports for each channel in the response file std::cout << "channels in response file: " << sf_info.channels << std::endl; std::cout << "registering ports:"; for (int i = 0; i < sf_info.channels; ++i) { std::stringstream stream_in; std::stringstream stream_out; stream_in << "in" << i; stream_out << "out" << i; std::cout << " " << stream_in.str(); std::cout << " " << stream_out.str(); jack_port_t *tmp_in = jack_port_register(client, stream_in.str().c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0); jack_port_t *tmp_out = jack_port_register(client, stream_out.str().c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0); iports.push_back(tmp_in); oports.push_back(tmp_out); } std::cout << std::endl; std::cout << "length of response file in frames: " << sf_info.frames << std::endl; if (sf_info.samplerate != jack_get_sample_rate(client)) { std::cout << "warning: samplerate in responseFile: " << sf_info.samplerate << "; jack-samplerate: " << jack_get_sample_rate(client) << std::endl; std::cout << "will resample response file" << std::endl; } // find out how many chunks we need per channel: chunks_per_channel = (int)ceil((float)sf_info.frames/(float)jack_buffer_size); std::cout << "chunks per channel: " << chunks_per_channel << std::endl; // allocate chunk memory for (int i = 0; i < sf_info.channels; ++i) { std::vector<fftwf_complex*> channel; for (int j = 0; j < chunks_per_channel; ++j) { // zero padded to twice the length fftwf_complex *tmp = (fftwf_complex*)fftwf_malloc(sizeof(fftwf_complex) * jack_buffer_size * 2); // zero for (int frame = 0; frame < jack_buffer_size * 2; ++frame) tmp[frame][0] = tmp[frame][1] = 0; channel.push_back(tmp); } chunks.push_back(channel); } std::cout << "chopping response file..."; // fill the chunks with the appropriate data float *tmp = new float[sf_info.channels]; for (int chunk = 0; chunk < chunks_per_channel; ++chunk) { for (int frame = 0; frame < jack_buffer_size * 2; ++frame) { // pad with 0's if (chunk*jack_buffer_size + frame < sf_info.frames && frame < jack_buffer_size) { int result = sf_readf_float(response_file, tmp, 1); if (result != 1) std::cout << "problem reading the soundfile" << std::endl; } else { for (int channel = 0; channel < sf_info.channels; ++channel) { tmp[channel] = 0; } } for (int channel = 0; channel < sf_info.channels; ++channel) { // set real value to sound data chunks[channel][chunk][frame][0] = tmp[channel]; // std::cout << tmp[channel] << " "; // imaginary value to 0 chunks[channel][chunk][frame][1] = 0; } } } std::cout << "done." << std::endl; std::cout << "creating fftw3 plan..."; // ok, now we need to FFT each chunk.. For this we need an FFT plan. // buffers fft_float = new float[jack_buffer_size * 2]; fft_complex = (fftwf_complex*)fftwf_malloc(sizeof(fftwf_complex) * jack_buffer_size * 2); // create fftw plan fft_plan_forward = fftwf_plan_dft_r2c_1d(jack_buffer_size * 2, fft_float, fft_complex, FFTW_MEASURE); fft_plan_backward = fftwf_plan_dft_c2r_1d(jack_buffer_size * 2, fft_complex, fft_float, FFTW_MEASURE); std::cout << "done" << std::endl; // fft the chunks std::cout << "FFT'ing response file chunks..." << std::endl; for (int channel = 0; channel < sf_info.channels; ++channel) { std::cout << "channel: " << channel << ": "; for (int chunk = 0; chunk < chunks_per_channel; ++chunk) { std::cout << "."; // copy chunk to input buffer for (int frame = 0; frame < jack_buffer_size * 2; ++frame) { fft_float[frame] = chunks[channel][chunk][frame][0]; // fft_in[frame][1] = 0; } // fft fftwf_execute(fft_plan_forward); // copy output buffer to chunk for (int frame = 0; frame < jack_buffer_size * 2; ++frame) { chunks[channel][chunk][frame][0] = fft_complex[frame][0] / normalize_factor; chunks[channel][chunk][frame][1] = fft_complex[frame][1] / normalize_factor;; } } std::cout << std::endl; } std::cout << "done." << std::endl; // make room so we can store the buffer pointers for each channel ibuffers.resize(sf_info.channels); obuffers.resize(sf_info.channels); // allocate ram for ringbuffers and zero out for 0 noise :) for (int channel = 0; channel < sf_info.channels; ++channel) { fftwf_complex *tmp = (fftwf_complex*)fftwf_malloc(sizeof(fftwf_complex) * jack_buffer_size * 2 * chunks_per_channel); // zero out buffers for (int frame = 0; frame < jack_buffer_size * chunks_per_channel * 2; ++frame) { tmp[frame][0] = 0; tmp[frame][1] = 0; } ringbuffers.push_back(tmp); } // allocate buffers for overlap for (int channel = 0; channel < sf_info.channels; ++channel) { jack_default_audio_sample_t *tmp = new jack_default_audio_sample_t[jack_buffer_size]; overlaps.push_back(tmp); } // now we should be ready to go // std::cout << chunks.size() << std::endl; jack_set_process_callback(client, process, 0); jack_activate(client); std::cout << "running (press ctrl-c to quit)..." << std::endl; while(!quit) {sleep(1);}; // sleep(seconds_to_run); jack_deactivate(client); jack_client_close(client); } --------------- -- Palimm Palimm! http://affenbande.org/~tapas/

20 years, 4 months

[linux-audio-user] Re: Decent reverb, and DSP in hardware?

by Juhana Sadeharju

>From: Lee Revell <rlrevell(a)joe-job.com> > >The EAX 1.0, 2.0 and 3.0 reverbs have been extracted from the Creative >dlls and the reverse engineered DSP source code is available here (check >out the SDK in the downloads section): > >http://www.kxproject.com Do you mean this file? http://kxproject.lugosoft.com/mirrors.php?&file=kxdrv3537-sdk.exe&language=… It is a Windows exe. Could anyone convert the content to a zip or a tar file. Upload to "ftp://ftp.funet.fi/incoming/audio/" or place it temporarily available at your own webpage for me. Juhana -- http://music.columbia.edu/mailman/listinfo/linux-graphics-dev for developers of open source graphics software

20 years, 4 months

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Linux-audio-user January 2005