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/*
* audio resampling with soxr
* Copyright (c) 2012 Rob Sykes <robs@users.sourceforge.net>
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* audio resampling with soxr
*/
#include "libavutil/log.h"
#include "swresample_internal.h"
#include <soxr.h>
static struct ResampleContext *create(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta, double precision, int cheby, int exact_rational){
soxr_error_t error;
soxr_datatype_t type =
format == AV_SAMPLE_FMT_S16P? SOXR_INT16_S :
format == AV_SAMPLE_FMT_S16 ? SOXR_INT16_I :
format == AV_SAMPLE_FMT_S32P? SOXR_INT32_S :
format == AV_SAMPLE_FMT_S32 ? SOXR_INT32_I :
format == AV_SAMPLE_FMT_FLTP? SOXR_FLOAT32_S :
format == AV_SAMPLE_FMT_FLT ? SOXR_FLOAT32_I :
format == AV_SAMPLE_FMT_DBLP? SOXR_FLOAT64_S :
format == AV_SAMPLE_FMT_DBL ? SOXR_FLOAT64_I : (soxr_datatype_t)-1;
soxr_io_spec_t io_spec = soxr_io_spec(type, type);
soxr_quality_spec_t q_spec = soxr_quality_spec((int)((precision-2)/4), (SOXR_HI_PREC_CLOCK|SOXR_ROLLOFF_NONE)*!!cheby);
q_spec.precision = precision;
#if !defined SOXR_VERSION /* Deprecated @ March 2013: */
q_spec.bw_pc = cutoff? FFMAX(FFMIN(cutoff,.995),.8)*100 : q_spec.bw_pc;
#else
q_spec.passband_end = cutoff? FFMAX(FFMIN(cutoff,.995),.8) : q_spec.passband_end;
#endif
soxr_delete((soxr_t)c);
c = (struct ResampleContext *)
soxr_create(in_rate, out_rate, 0, &error, &io_spec, &q_spec, 0);
if (!c)
av_log(NULL, AV_LOG_ERROR, "soxr_create: %s\n", error);
return c;
}
static void destroy(struct ResampleContext * *c){
soxr_delete((soxr_t)*c);
*c = NULL;
}
static int flush(struct SwrContext *s){
s->delayed_samples_fixup = soxr_delay((soxr_t)s->resample);
soxr_process((soxr_t)s->resample, NULL, 0, NULL, NULL, 0, NULL);
{
float f;
size_t idone, odone;
soxr_process((soxr_t)s->resample, &f, 0, &idone, &f, 0, &odone);
s->delayed_samples_fixup -= soxr_delay((soxr_t)s->resample);
}
return 0;
}
static int process(
struct ResampleContext * c, AudioData *dst, int dst_size,
AudioData *src, int src_size, int *consumed){
size_t idone, odone;
soxr_error_t error = soxr_set_error((soxr_t)c, soxr_set_num_channels((soxr_t)c, src->ch_count));
if (!error)
error = soxr_process((soxr_t)c, src->ch, (size_t)src_size,
&idone, dst->ch, (size_t)dst_size, &odone);
else
idone = 0;
*consumed = (int)idone;
return error? -1 : odone;
}
static int64_t get_delay(struct SwrContext *s, int64_t base){
double delayed_samples = soxr_delay((soxr_t)s->resample);
double delay_s;
if (s->flushed)
delayed_samples += s->delayed_samples_fixup;
delay_s = delayed_samples / s->out_sample_rate;
return (int64_t)(delay_s * base + .5);
}
static int invert_initial_buffer(struct ResampleContext *c, AudioData *dst, const AudioData *src,
int in_count, int *out_idx, int *out_sz){
return 0;
}
static int64_t get_out_samples(struct SwrContext *s, int in_samples){
double out_samples = (double)s->out_sample_rate / s->in_sample_rate * in_samples;
double delayed_samples = soxr_delay((soxr_t)s->resample);
if (s->flushed)
delayed_samples += s->delayed_samples_fixup;
return (int64_t)(out_samples + delayed_samples + 1 + .5);
}
struct Resampler const swri_soxr_resampler={
create, destroy, process, flush, NULL /* set_compensation */, get_delay,
invert_initial_buffer, get_out_samples
};
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