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https://github.com/FFmpeg/FFmpeg.git
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101 lines
3.6 KiB
C
101 lines
3.6 KiB
C
/*
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* audio resampling with soxr
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* Copyright (c) 2012 Rob Sykes <robs@users.sourceforge.net>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but 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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* Lesser 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
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* audio resampling with soxr
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*/
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#include "libavutil/log.h"
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#include "swresample_internal.h"
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#include <soxr.h>
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static struct ResampleContext *create(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
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double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby){
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soxr_error_t error;
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soxr_datatype_t type =
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format == AV_SAMPLE_FMT_S16P? SOXR_INT16_S :
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format == AV_SAMPLE_FMT_S16 ? SOXR_INT16_I :
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format == AV_SAMPLE_FMT_S32P? SOXR_INT32_S :
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format == AV_SAMPLE_FMT_S32 ? SOXR_INT32_I :
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format == AV_SAMPLE_FMT_FLTP? SOXR_FLOAT32_S :
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format == AV_SAMPLE_FMT_FLT ? SOXR_FLOAT32_I :
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format == AV_SAMPLE_FMT_DBLP? SOXR_FLOAT64_S :
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format == AV_SAMPLE_FMT_DBL ? SOXR_FLOAT64_I : (soxr_datatype_t)-1;
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soxr_io_spec_t io_spec = soxr_io_spec(type, type);
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soxr_quality_spec_t q_spec = soxr_quality_spec((int)((precision-2)/4), (SOXR_HI_PREC_CLOCK|SOXR_ROLLOFF_NONE)*!!cheby);
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q_spec.precision = linear? 0 : precision;
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#if !defined SOXR_VERSION /* Deprecated @ March 2013: */
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q_spec.bw_pc = cutoff? FFMAX(FFMIN(cutoff,.995),.8)*100 : q_spec.bw_pc;
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#else
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q_spec.passband_end = cutoff? FFMAX(FFMIN(cutoff,.995),.8) : q_spec.passband_end;
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#endif
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soxr_delete((soxr_t)c);
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c = (struct ResampleContext *)
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soxr_create(in_rate, out_rate, 0, &error, &io_spec, &q_spec, 0);
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if (!c)
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av_log(NULL, AV_LOG_ERROR, "soxr_create: %s\n", error);
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return c;
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}
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static void destroy(struct ResampleContext * *c){
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soxr_delete((soxr_t)*c);
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*c = NULL;
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}
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static int flush(struct SwrContext *s){
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soxr_process((soxr_t)s->resample, NULL, 0, NULL, NULL, 0, NULL);
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return 0;
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}
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static int process(
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struct ResampleContext * c, AudioData *dst, int dst_size,
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AudioData *src, int src_size, int *consumed){
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size_t idone, odone;
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soxr_error_t error = soxr_set_error((soxr_t)c, soxr_set_num_channels((soxr_t)c, src->ch_count));
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error = soxr_process((soxr_t)c, src->ch, (size_t)src_size,
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&idone, dst->ch, (size_t)dst_size, &odone);
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*consumed = (int)idone;
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return error? -1 : odone;
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}
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static int64_t get_delay(struct SwrContext *s, int64_t base){
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double delay_s = soxr_delay((soxr_t)s->resample) / s->out_sample_rate;
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return (int64_t)(delay_s * base + .5);
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}
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static int invert_initial_buffer(struct ResampleContext *c, AudioData *dst, const AudioData *src,
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int in_count, int *out_idx, int *out_sz)
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{
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return 0;
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}
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struct Resampler const soxr_resampler={
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create, destroy, process, flush, NULL /* set_compensation */, get_delay,
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invert_initial_buffer,
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};
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