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  1. /*
  2.  * audio resampling with soxr
  3.  * Copyright (c) 2012 Rob Sykes <robs@users.sourceforge.net>
  4.  *
  5.  * This file is part of FFmpeg.
  6.  *
  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.
  11.  *
  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.
  16.  *
  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
  20.  */
  21.  
  22. /**
  23.  * @file
  24.  * audio resampling with soxr
  25.  */
  26.  
  27. #include "libavutil/log.h"
  28. #include "swresample_internal.h"
  29.  
  30. #include <soxr.h>
  31.  
  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, int kaiser_beta, double precision, int cheby){
  34.     soxr_error_t error;
  35.  
  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;
  45.  
  46.     soxr_io_spec_t io_spec = soxr_io_spec(type, type);
  47.  
  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 = linear? 0 : 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;
  52. #else
  53.     q_spec.passband_end = cutoff? FFMAX(FFMIN(cutoff,.995),.8) : q_spec.passband_end;
  54. #endif
  55.  
  56.     soxr_delete((soxr_t)c);
  57.     c = (struct ResampleContext *)
  58.         soxr_create(in_rate, out_rate, 0, &error, &io_spec, &q_spec, 0);
  59.     if (!c)
  60.         av_log(NULL, AV_LOG_ERROR, "soxr_create: %s\n", error);
  61.     return c;
  62. }
  63.  
  64. static void destroy(struct ResampleContext * *c){
  65.     soxr_delete((soxr_t)*c);
  66.     *c = NULL;
  67. }
  68.  
  69. static int flush(struct SwrContext *s){
  70.     s->delayed_samples_fixup = soxr_delay((soxr_t)s->resample);
  71.  
  72.     soxr_process((soxr_t)s->resample, NULL, 0, NULL, NULL, 0, NULL);
  73.  
  74.     {
  75.         float f;
  76.         size_t idone, odone;
  77.         soxr_process((soxr_t)s->resample, &f, 0, &idone, &f, 0, &odone);
  78.         s->delayed_samples_fixup -= soxr_delay((soxr_t)s->resample);
  79.     }
  80.  
  81.     return 0;
  82. }
  83.  
  84. static int process(
  85.         struct ResampleContext * c, AudioData *dst, int dst_size,
  86.         AudioData *src, int src_size, int *consumed){
  87.     size_t idone, odone;
  88.     soxr_error_t error = soxr_set_error((soxr_t)c, soxr_set_num_channels((soxr_t)c, src->ch_count));
  89.     if (!error)
  90.         error = soxr_process((soxr_t)c, src->ch, (size_t)src_size,
  91.                              &idone, dst->ch, (size_t)dst_size, &odone);
  92.     else
  93.         idone = 0;
  94.  
  95.     *consumed = (int)idone;
  96.     return error? -1 : odone;
  97. }
  98.  
  99. static int64_t get_delay(struct SwrContext *s, int64_t base){
  100.     double delayed_samples = soxr_delay((soxr_t)s->resample);
  101.     double delay_s;
  102.  
  103.     if (s->flushed)
  104.         delayed_samples += s->delayed_samples_fixup;
  105.  
  106.     delay_s = delayed_samples / s->out_sample_rate;
  107.  
  108.     return (int64_t)(delay_s * base + .5);
  109. }
  110.  
  111. static int invert_initial_buffer(struct ResampleContext *c, AudioData *dst, const AudioData *src,
  112.                                  int in_count, int *out_idx, int *out_sz){
  113.     return 0;
  114. }
  115.  
  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);
  119.  
  120.     if (s->flushed)
  121.         delayed_samples += s->delayed_samples_fixup;
  122.  
  123.     return (int64_t)(out_samples + delayed_samples + 1 + .5);
  124. }
  125.  
  126. struct Resampler const swri_soxr_resampler={
  127.     create, destroy, process, flush, NULL /* set_compensation */, get_delay,
  128.     invert_initial_buffer, get_out_samples
  129. };
  130.  
  131.