2 * audio resampling with soxr
3 * Copyright (c) 2012 Rob Sykes <robs@users.sourceforge.net>
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24 * audio resampling with soxr
27 #include "libavutil/log.h"
28 #include "swresample_internal.h"
32 static struct ResampleContext *create(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
33 double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, double kaiser_beta, double precision, int cheby, int exact_rational){
36 soxr_datatype_t type =
37 format == AV_SAMPLE_FMT_S16P? SOXR_INT16_S :
38 format == AV_SAMPLE_FMT_S16 ? SOXR_INT16_I :
39 format == AV_SAMPLE_FMT_S32P? SOXR_INT32_S :
40 format == AV_SAMPLE_FMT_S32 ? SOXR_INT32_I :
41 format == AV_SAMPLE_FMT_FLTP? SOXR_FLOAT32_S :
42 format == AV_SAMPLE_FMT_FLT ? SOXR_FLOAT32_I :
43 format == AV_SAMPLE_FMT_DBLP? SOXR_FLOAT64_S :
44 format == AV_SAMPLE_FMT_DBL ? SOXR_FLOAT64_I : (soxr_datatype_t)-1;
46 soxr_io_spec_t io_spec = soxr_io_spec(type, type);
48 soxr_quality_spec_t q_spec = soxr_quality_spec((int)((precision-2)/4), (SOXR_HI_PREC_CLOCK|SOXR_ROLLOFF_NONE)*!!cheby);
49 q_spec.precision = precision;
50 #if !defined SOXR_VERSION /* Deprecated @ March 2013: */
51 q_spec.bw_pc = cutoff? FFMAX(FFMIN(cutoff,.995),.8)*100 : q_spec.bw_pc;
53 q_spec.passband_end = cutoff? FFMAX(FFMIN(cutoff,.995),.8) : q_spec.passband_end;
56 soxr_delete((soxr_t)c);
57 c = (struct ResampleContext *)
58 soxr_create(in_rate, out_rate, 0, &error, &io_spec, &q_spec, 0);
60 av_log(NULL, AV_LOG_ERROR, "soxr_create: %s\n", error);
64 static void destroy(struct ResampleContext * *c){
65 soxr_delete((soxr_t)*c);
69 static int flush(struct SwrContext *s){
70 s->delayed_samples_fixup = soxr_delay((soxr_t)s->resample);
72 soxr_process((soxr_t)s->resample, NULL, 0, NULL, NULL, 0, NULL);
77 soxr_process((soxr_t)s->resample, &f, 0, &idone, &f, 0, &odone);
78 s->delayed_samples_fixup -= soxr_delay((soxr_t)s->resample);
85 struct ResampleContext * c, AudioData *dst, int dst_size,
86 AudioData *src, int src_size, int *consumed){
88 soxr_error_t error = soxr_set_error((soxr_t)c, soxr_set_num_channels((soxr_t)c, src->ch_count));
90 error = soxr_process((soxr_t)c, src->ch, (size_t)src_size,
91 &idone, dst->ch, (size_t)dst_size, &odone);
95 *consumed = (int)idone;
96 return error? -1 : odone;
99 static int64_t get_delay(struct SwrContext *s, int64_t base){
100 double delayed_samples = soxr_delay((soxr_t)s->resample);
104 delayed_samples += s->delayed_samples_fixup;
106 delay_s = delayed_samples / s->out_sample_rate;
108 return (int64_t)(delay_s * base + .5);
111 static int invert_initial_buffer(struct ResampleContext *c, AudioData *dst, const AudioData *src,
112 int in_count, int *out_idx, int *out_sz){
116 static int64_t get_out_samples(struct SwrContext *s, int in_samples){
117 double out_samples = (double)s->out_sample_rate / s->in_sample_rate * in_samples;
118 double delayed_samples = soxr_delay((soxr_t)s->resample);
121 delayed_samples += s->delayed_samples_fixup;
123 return (int64_t)(out_samples + delayed_samples + 1 + .5);
126 struct Resampler const swri_soxr_resampler={
127 create, destroy, process, flush, NULL /* set_compensation */, get_delay,
128 invert_initial_buffer, get_out_samples