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/Received on Sat Jan 29 04:15:26 2005
This archive was generated by hypermail 2.1.8 : Sat Jan 29 2005 - 04:15:27 EET