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FFmpeg/libswresample/soxr_resample.c
Ganesh Ajjanagadde 1bed09a30e swresample: allow double precision beta value for the Kaiser window
Kaiser windows inherently don't require beta to be an integer. This was
an arbitrary restriction. Moreover, soxr does not require it, and in
fact often estimates beta to a non-integral value.

Thus, this patch allows greater flexibility for swresample clients.
Micro version is updated.

Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
2015-11-08 21:11:07 -05:00

131 lines
4.4 KiB
C

/*
* 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){
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 = linear? 0 : 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
};