mirror of
https://gitlab.freedesktop.org/pulseaudio/pulseaudio.git
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resampler: Split the resampler implementations into separate files
Rebased by Peter Meerwald. Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net> Signed-off-by: poljar (Damir Jelić) <poljarinho@gmail.com>
This commit is contained in:
poljar (Damir Jelić)
Tanu Kaskinen
parent
64d17a6b0a
commit
72103e1e33
8 changed files with 748 additions and 604 deletions
132
src/pulsecore/resampler/ffmpeg.c
Normal file
132
src/pulsecore/resampler/ffmpeg.c
Normal file
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@ -0,0 +1,132 @@
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/***
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This file is part of PulseAudio.
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Copyright 2004-2006 Lennart Poettering
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PulseAudio is free software; you can redistribute it and/or modify
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it under the terms of the GNU Lesser General Public License as published
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by the Free Software Foundation; either version 2.1 of the License,
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or (at your option) any later version.
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PulseAudio is distributed in the hope that it will be useful, but
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WITHOUT ANY WARRANTY; without even the implied warranty of
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||||
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||
General Public License for more details.
|
||||
|
||||
You should have received a copy of the GNU Lesser General Public License
|
||||
along with PulseAudio; if not, write to the Free Software
|
||||
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
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USA.
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***/
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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#include <pulse/xmalloc.h>
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#include "pulsecore/ffmpeg/avcodec.h"
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#include "pulsecore/resampler.h"
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struct ffmpeg_data { /* data specific to ffmpeg */
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struct AVResampleContext *state;
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};
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static unsigned ffmpeg_resample(pa_resampler *r, const pa_memchunk *input, unsigned in_n_frames, pa_memchunk *output, unsigned *out_n_frames) {
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unsigned used_frames = 0, c;
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int previous_consumed_frames = -1;
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struct ffmpeg_data *ffmpeg_data;
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pa_assert(r);
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pa_assert(input);
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pa_assert(output);
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pa_assert(out_n_frames);
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ffmpeg_data = r->impl.data;
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for (c = 0; c < r->work_channels; c++) {
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unsigned u;
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pa_memblock *b, *w;
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int16_t *p, *t, *k, *q, *s;
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int consumed_frames;
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/* Allocate a new block */
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b = pa_memblock_new(r->mempool, in_n_frames * sizeof(int16_t));
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p = pa_memblock_acquire(b);
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/* Now copy the input data, splitting up channels */
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t = (int16_t*) pa_memblock_acquire_chunk(input) + c;
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k = p;
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for (u = 0; u < in_n_frames; u++) {
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*k = *t;
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t += r->work_channels;
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k ++;
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}
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pa_memblock_release(input->memblock);
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/* Allocate buffer for the result */
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w = pa_memblock_new(r->mempool, *out_n_frames * sizeof(int16_t));
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q = pa_memblock_acquire(w);
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/* Now, resample */
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used_frames = (unsigned) av_resample(ffmpeg_data->state,
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q, p,
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&consumed_frames,
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(int) in_n_frames, (int) *out_n_frames,
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c >= (unsigned) (r->work_channels-1));
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pa_memblock_release(b);
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pa_memblock_unref(b);
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pa_assert(consumed_frames <= (int) in_n_frames);
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pa_assert(previous_consumed_frames == -1 || consumed_frames == previous_consumed_frames);
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previous_consumed_frames = consumed_frames;
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/* And place the results in the output buffer */
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s = (int16_t *) pa_memblock_acquire_chunk(output) + c;
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for (u = 0; u < used_frames; u++) {
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*s = *q;
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q++;
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s += r->work_channels;
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}
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pa_memblock_release(output->memblock);
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pa_memblock_release(w);
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pa_memblock_unref(w);
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}
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*out_n_frames = used_frames;
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return in_n_frames - previous_consumed_frames;
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}
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static void ffmpeg_free(pa_resampler *r) {
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struct ffmpeg_data *ffmpeg_data;
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pa_assert(r);
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ffmpeg_data = r->impl.data;
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if (ffmpeg_data->state)
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av_resample_close(ffmpeg_data->state);
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}
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int pa_resampler_ffmpeg_init(pa_resampler *r) {
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struct ffmpeg_data *ffmpeg_data;
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pa_assert(r);
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ffmpeg_data = pa_xnew(struct ffmpeg_data, 1);
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/* We could probably implement different quality levels by
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* adjusting the filter parameters here. However, ffmpeg
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* internally only uses these hardcoded values, so let's use them
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* here for now as well until ffmpeg makes this configurable. */
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if (!(ffmpeg_data->state = av_resample_init((int) r->o_ss.rate, (int) r->i_ss.rate, 16, 10, 0, 0.8)))
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return -1;
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r->impl.free = ffmpeg_free;
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r->impl.resample = ffmpeg_resample;
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r->impl.data = (void *) ffmpeg_data;
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return 0;
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}
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102
src/pulsecore/resampler/libsamplerate.c
Normal file
102
src/pulsecore/resampler/libsamplerate.c
Normal file
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@ -0,0 +1,102 @@
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/***
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This file is part of PulseAudio.
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Copyright 2004-2006 Lennart Poettering
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PulseAudio is free software; you can redistribute it and/or modify
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it under the terms of the GNU Lesser General Public License as published
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by the Free Software Foundation; either version 2.1 of the License,
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or (at your option) any later version.
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PulseAudio is distributed in the hope that it will be useful, but
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WITHOUT ANY WARRANTY; without even the implied warranty of
|
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
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General Public License for more details.
|
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|
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You should have received a copy of the GNU Lesser General Public License
|
||||
along with PulseAudio; if not, write to the Free Software
|
||||
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
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USA.
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***/
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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#include <samplerate.h>
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#include "pulsecore/resampler.h"
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static unsigned libsamplerate_resample(pa_resampler *r, const pa_memchunk *input, unsigned in_n_frames, pa_memchunk *output, unsigned *out_n_frames) {
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SRC_DATA data;
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SRC_STATE *state;
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pa_assert(r);
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pa_assert(input);
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pa_assert(output);
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pa_assert(out_n_frames);
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state = r->impl.data;
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memset(&data, 0, sizeof(data));
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data.data_in = pa_memblock_acquire_chunk(input);
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data.input_frames = (long int) in_n_frames;
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data.data_out = pa_memblock_acquire_chunk(output);
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data.output_frames = (long int) *out_n_frames;
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data.src_ratio = (double) r->o_ss.rate / r->i_ss.rate;
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data.end_of_input = 0;
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pa_assert_se(src_process(state, &data) == 0);
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pa_memblock_release(input->memblock);
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pa_memblock_release(output->memblock);
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*out_n_frames = (unsigned) data.output_frames_gen;
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return in_n_frames - data.input_frames_used;
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}
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static void libsamplerate_update_rates(pa_resampler *r) {
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SRC_STATE *state;
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pa_assert(r);
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state = r->impl.data;
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pa_assert_se(src_set_ratio(state, (double) r->o_ss.rate / r->i_ss.rate) == 0);
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}
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static void libsamplerate_reset(pa_resampler *r) {
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SRC_STATE *state;
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pa_assert(r);
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state = r->impl.data;
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pa_assert_se(src_reset(state) == 0);
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}
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static void libsamplerate_free(pa_resampler *r) {
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SRC_STATE *state;
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pa_assert(r);
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state = r->impl.data;
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if (state)
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src_delete(state);
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}
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int pa_resampler_libsamplerate_init(pa_resampler *r) {
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int err;
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SRC_STATE *state;
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pa_assert(r);
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if (!(state = src_new(r->method, r->work_channels, &err)))
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return -1;
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r->impl.free = libsamplerate_free;
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r->impl.update_rates = libsamplerate_update_rates;
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r->impl.resample = libsamplerate_resample;
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r->impl.reset = libsamplerate_reset;
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r->impl.data = state;
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return 0;
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}
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163
src/pulsecore/resampler/peaks.c
Normal file
163
src/pulsecore/resampler/peaks.c
Normal file
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@ -0,0 +1,163 @@
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/***
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This file is part of PulseAudio.
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Copyright 2004-2006 Lennart Poettering
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PulseAudio is free software; you can redistribute it and/or modify
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it under the terms of the GNU Lesser General Public License as published
|
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by the Free Software Foundation; either version 2.1 of the License,
|
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or (at your option) any later version.
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|
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PulseAudio is distributed in the hope that it will be useful, but
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WITHOUT ANY WARRANTY; without even the implied warranty of
|
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
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General Public License for more details.
|
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|
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You should have received a copy of the GNU Lesser General Public License
|
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along with PulseAudio; if not, write to the Free Software
|
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Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
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USA.
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***/
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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#include <pulse/xmalloc.h>
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#include <math.h>
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#include "pulsecore/resampler.h"
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struct peaks_data { /* data specific to the peak finder pseudo resampler */
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unsigned o_counter;
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unsigned i_counter;
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float max_f[PA_CHANNELS_MAX];
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int16_t max_i[PA_CHANNELS_MAX];
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};
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static unsigned peaks_resample(pa_resampler *r, const pa_memchunk *input, unsigned in_n_frames, pa_memchunk *output, unsigned *out_n_frames) {
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unsigned c, o_index = 0;
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unsigned i, i_end = 0;
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void *src, *dst;
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struct peaks_data *peaks_data;
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pa_assert(r);
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pa_assert(input);
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pa_assert(output);
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pa_assert(out_n_frames);
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peaks_data = r->impl.data;
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src = pa_memblock_acquire_chunk(input);
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dst = pa_memblock_acquire_chunk(output);
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i = ((uint64_t) peaks_data->o_counter * r->i_ss.rate) / r->o_ss.rate;
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i = i > peaks_data->i_counter ? i - peaks_data->i_counter : 0;
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while (i_end < in_n_frames) {
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i_end = ((uint64_t) (peaks_data->o_counter + 1) * r->i_ss.rate) / r->o_ss.rate;
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i_end = i_end > peaks_data->i_counter ? i_end - peaks_data->i_counter : 0;
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pa_assert_fp(o_index * r->w_fz < pa_memblock_get_length(output->memblock));
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/* 1ch float is treated separately, because that is the common case */
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if (r->work_channels == 1 && r->work_format == PA_SAMPLE_FLOAT32NE) {
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float *s = (float*) src + i;
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float *d = (float*) dst + o_index;
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for (; i < i_end && i < in_n_frames; i++) {
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float n = fabsf(*s++);
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if (n > peaks_data->max_f[0])
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peaks_data->max_f[0] = n;
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}
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if (i == i_end) {
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*d = peaks_data->max_f[0];
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peaks_data->max_f[0] = 0;
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o_index++, peaks_data->o_counter++;
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}
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} else if (r->work_format == PA_SAMPLE_S16NE) {
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int16_t *s = (int16_t*) src + r->work_channels * i;
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int16_t *d = (int16_t*) dst + r->work_channels * o_index;
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for (; i < i_end && i < in_n_frames; i++)
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for (c = 0; c < r->work_channels; c++) {
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int16_t n = abs(*s++);
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if (n > peaks_data->max_i[c])
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peaks_data->max_i[c] = n;
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}
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if (i == i_end) {
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for (c = 0; c < r->work_channels; c++, d++) {
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*d = peaks_data->max_i[c];
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peaks_data->max_i[c] = 0;
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}
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o_index++, peaks_data->o_counter++;
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}
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} else {
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float *s = (float*) src + r->work_channels * i;
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float *d = (float*) dst + r->work_channels * o_index;
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for (; i < i_end && i < in_n_frames; i++)
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for (c = 0; c < r->work_channels; c++) {
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float n = fabsf(*s++);
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if (n > peaks_data->max_f[c])
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peaks_data->max_f[c] = n;
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}
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if (i == i_end) {
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for (c = 0; c < r->work_channels; c++, d++) {
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*d = peaks_data->max_f[c];
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peaks_data->max_f[c] = 0;
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}
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o_index++, peaks_data->o_counter++;
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}
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}
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}
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pa_memblock_release(input->memblock);
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pa_memblock_release(output->memblock);
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*out_n_frames = o_index;
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peaks_data->i_counter += in_n_frames;
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/* Normalize counters */
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while (peaks_data->i_counter >= r->i_ss.rate) {
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pa_assert(peaks_data->o_counter >= r->o_ss.rate);
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peaks_data->i_counter -= r->i_ss.rate;
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peaks_data->o_counter -= r->o_ss.rate;
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}
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return 0;
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}
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static void peaks_update_rates_or_reset(pa_resampler *r) {
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struct peaks_data *peaks_data;
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pa_assert(r);
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peaks_data = r->impl.data;
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peaks_data->i_counter = 0;
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peaks_data->o_counter = 0;
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}
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int pa_resampler_peaks_init(pa_resampler*r) {
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struct peaks_data *peaks_data;
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pa_assert(r);
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pa_assert(r->i_ss.rate >= r->o_ss.rate);
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pa_assert(r->work_format == PA_SAMPLE_S16NE || r->work_format == PA_SAMPLE_FLOAT32NE);
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peaks_data = pa_xnew0(struct peaks_data, 1);
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r->impl.resample = peaks_resample;
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r->impl.update_rates = peaks_update_rates_or_reset;
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r->impl.reset = peaks_update_rates_or_reset;
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r->impl.data = peaks_data;
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return 0;
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}
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151
src/pulsecore/resampler/speex.c
Normal file
151
src/pulsecore/resampler/speex.c
Normal file
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@ -0,0 +1,151 @@
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/***
|
||||
This file is part of PulseAudio.
|
||||
|
||||
Copyright 2004-2006 Lennart Poettering
|
||||
|
||||
PulseAudio 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.
|
||||
|
||||
PulseAudio 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
|
||||
General Public License for more details.
|
||||
|
||||
You should have received a copy of the GNU Lesser General Public License
|
||||
along with PulseAudio; if not, write to the Free Software
|
||||
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
|
||||
USA.
|
||||
***/
|
||||
|
||||
#ifdef HAVE_CONFIG_H
|
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#include <config.h>
|
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#endif
|
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|
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#include <speex/speex_resampler.h>
|
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|
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#include "pulsecore/resampler.h"
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|
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static unsigned speex_resample_float(pa_resampler *r, const pa_memchunk *input, unsigned in_n_frames, pa_memchunk *output, unsigned *out_n_frames) {
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float *in, *out;
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uint32_t inf = in_n_frames, outf = *out_n_frames;
|
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SpeexResamplerState *state;
|
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|
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pa_assert(r);
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pa_assert(input);
|
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pa_assert(output);
|
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pa_assert(out_n_frames);
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|
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state = r->impl.data;
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in = pa_memblock_acquire_chunk(input);
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out = pa_memblock_acquire_chunk(output);
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/* Strictly speaking, speex resampler expects its input
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* to be normalized to the [-32768.0 .. 32767.0] range.
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* This matters if speex has been compiled with --enable-fixed-point,
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* because such speex will round the samples to the nearest
|
||||
* integer. speex with --enable-fixed-point is therefore incompatible
|
||||
* with PulseAudio's floating-point sample range [-1 .. 1]. speex
|
||||
* without --enable-fixed-point works fine with this range.
|
||||
* Care has been taken to call speex_resample_float() only
|
||||
* for speex compiled without --enable-fixed-point.
|
||||
*/
|
||||
pa_assert_se(speex_resampler_process_interleaved_float(state, in, &inf, out, &outf) == 0);
|
||||
|
||||
pa_memblock_release(input->memblock);
|
||||
pa_memblock_release(output->memblock);
|
||||
|
||||
pa_assert(inf == in_n_frames);
|
||||
*out_n_frames = outf;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static unsigned speex_resample_int(pa_resampler *r, const pa_memchunk *input, unsigned in_n_frames, pa_memchunk *output, unsigned *out_n_frames) {
|
||||
int16_t *in, *out;
|
||||
uint32_t inf = in_n_frames, outf = *out_n_frames;
|
||||
SpeexResamplerState *state;
|
||||
|
||||
pa_assert(r);
|
||||
pa_assert(input);
|
||||
pa_assert(output);
|
||||
pa_assert(out_n_frames);
|
||||
|
||||
state = r->impl.data;
|
||||
|
||||
in = pa_memblock_acquire_chunk(input);
|
||||
out = pa_memblock_acquire_chunk(output);
|
||||
|
||||
pa_assert_se(speex_resampler_process_interleaved_int(state, in, &inf, out, &outf) == 0);
|
||||
|
||||
pa_memblock_release(input->memblock);
|
||||
pa_memblock_release(output->memblock);
|
||||
|
||||
pa_assert(inf == in_n_frames);
|
||||
*out_n_frames = outf;
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static void speex_update_rates(pa_resampler *r) {
|
||||
SpeexResamplerState *state;
|
||||
pa_assert(r);
|
||||
|
||||
state = r->impl.data;
|
||||
|
||||
pa_assert_se(speex_resampler_set_rate(state, r->i_ss.rate, r->o_ss.rate) == 0);
|
||||
}
|
||||
|
||||
static void speex_reset(pa_resampler *r) {
|
||||
SpeexResamplerState *state;
|
||||
pa_assert(r);
|
||||
|
||||
state = r->impl.data;
|
||||
|
||||
pa_assert_se(speex_resampler_reset_mem(state) == 0);
|
||||
}
|
||||
|
||||
static void speex_free(pa_resampler *r) {
|
||||
SpeexResamplerState *state;
|
||||
pa_assert(r);
|
||||
|
||||
state = r->impl.data;
|
||||
if (!state)
|
||||
return;
|
||||
|
||||
speex_resampler_destroy(state);
|
||||
}
|
||||
|
||||
int pa_resampler_speex_init(pa_resampler *r) {
|
||||
int q, err;
|
||||
SpeexResamplerState *state;
|
||||
|
||||
pa_assert(r);
|
||||
|
||||
r->impl.free = speex_free;
|
||||
r->impl.update_rates = speex_update_rates;
|
||||
r->impl.reset = speex_reset;
|
||||
|
||||
if (r->method >= PA_RESAMPLER_SPEEX_FIXED_BASE && r->method <= PA_RESAMPLER_SPEEX_FIXED_MAX) {
|
||||
|
||||
q = r->method - PA_RESAMPLER_SPEEX_FIXED_BASE;
|
||||
r->impl.resample = speex_resample_int;
|
||||
|
||||
} else {
|
||||
pa_assert(r->method >= PA_RESAMPLER_SPEEX_FLOAT_BASE && r->method <= PA_RESAMPLER_SPEEX_FLOAT_MAX);
|
||||
|
||||
q = r->method - PA_RESAMPLER_SPEEX_FLOAT_BASE;
|
||||
r->impl.resample = speex_resample_float;
|
||||
}
|
||||
|
||||
pa_log_info("Choosing speex quality setting %i.", q);
|
||||
|
||||
if (!(state = speex_resampler_init(r->work_channels, r->i_ss.rate, r->o_ss.rate, q, &err)))
|
||||
return -1;
|
||||
|
||||
r->impl.data = state;
|
||||
|
||||
return 0;
|
||||
}
|
||||
102
src/pulsecore/resampler/trivial.c
Normal file
102
src/pulsecore/resampler/trivial.c
Normal file
|
|
@ -0,0 +1,102 @@
|
|||
/***
|
||||
This file is part of PulseAudio.
|
||||
|
||||
Copyright 2004-2006 Lennart Poettering
|
||||
|
||||
PulseAudio 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.
|
||||
|
||||
PulseAudio 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
|
||||
General Public License for more details.
|
||||
|
||||
You should have received a copy of the GNU Lesser General Public License
|
||||
along with PulseAudio; if not, write to the Free Software
|
||||
Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
|
||||
USA.
|
||||
***/
|
||||
|
||||
#ifdef HAVE_CONFIG_H
|
||||
#include <config.h>
|
||||
#endif
|
||||
|
||||
#include <pulse/xmalloc.h>
|
||||
|
||||
#include "pulsecore/resampler.h"
|
||||
|
||||
struct trivial_data { /* data specific to the trivial resampler */
|
||||
unsigned o_counter;
|
||||
unsigned i_counter;
|
||||
};
|
||||
|
||||
static unsigned trivial_resample(pa_resampler *r, const pa_memchunk *input, unsigned in_n_frames, pa_memchunk *output, unsigned *out_n_frames) {
|
||||
unsigned i_index, o_index;
|
||||
void *src, *dst;
|
||||
struct trivial_data *trivial_data;
|
||||
|
||||
pa_assert(r);
|
||||
pa_assert(input);
|
||||
pa_assert(output);
|
||||
pa_assert(out_n_frames);
|
||||
|
||||
trivial_data = r->impl.data;
|
||||
|
||||
src = pa_memblock_acquire_chunk(input);
|
||||
dst = pa_memblock_acquire_chunk(output);
|
||||
|
||||
for (o_index = 0;; o_index++, trivial_data->o_counter++) {
|
||||
i_index = ((uint64_t) trivial_data->o_counter * r->i_ss.rate) / r->o_ss.rate;
|
||||
i_index = i_index > trivial_data->i_counter ? i_index - trivial_data->i_counter : 0;
|
||||
|
||||
if (i_index >= in_n_frames)
|
||||
break;
|
||||
|
||||
pa_assert_fp(o_index * r->w_fz < pa_memblock_get_length(output->memblock));
|
||||
|
||||
memcpy((uint8_t*) dst + r->w_fz * o_index, (uint8_t*) src + r->w_fz * i_index, (int) r->w_fz);
|
||||
}
|
||||
|
||||
pa_memblock_release(input->memblock);
|
||||
pa_memblock_release(output->memblock);
|
||||
|
||||
*out_n_frames = o_index;
|
||||
|
||||
trivial_data->i_counter += in_n_frames;
|
||||
|
||||
/* Normalize counters */
|
||||
while (trivial_data->i_counter >= r->i_ss.rate) {
|
||||
pa_assert(trivial_data->o_counter >= r->o_ss.rate);
|
||||
|
||||
trivial_data->i_counter -= r->i_ss.rate;
|
||||
trivial_data->o_counter -= r->o_ss.rate;
|
||||
}
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static void trivial_update_rates_or_reset(pa_resampler *r) {
|
||||
struct trivial_data *trivial_data;
|
||||
pa_assert(r);
|
||||
|
||||
trivial_data = r->impl.data;
|
||||
|
||||
trivial_data->i_counter = 0;
|
||||
trivial_data->o_counter = 0;
|
||||
}
|
||||
|
||||
int pa_resampler_trivial_init(pa_resampler *r) {
|
||||
struct trivial_data *trivial_data;
|
||||
pa_assert(r);
|
||||
|
||||
trivial_data = pa_xnew0(struct trivial_data, 1);
|
||||
|
||||
r->impl.resample = trivial_resample;
|
||||
r->impl.update_rates = trivial_update_rates_or_reset;
|
||||
r->impl.reset = trivial_update_rates_or_reset;
|
||||
r->impl.data = trivial_data;
|
||||
|
||||
return 0;
|
||||
}
|
||||
Loading…
Add table
Add a link
Reference in a new issue