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

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/