libswresample/soxr_resample.c

   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, double kaiser_beta, double precision, int cheby, int exact_rational){
  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 = 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