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

Florian,

Just gave your dry and wet sample a quick listen. It
sounds promising. I'm looking forward to giving it a
more serious run.

ron

--- Florian Schmidt <mista.tapas@email-addr-hidden> wrote:

>
> 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;
> }
>
>
=== message truncated ===

                
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Received on Sat Jan 29 08:15:09 2005