src/lib/resampler.cc

   1 /*
   2     Copyright (C) 2013-2015 Carl Hetherington <cth@carlh.net>
   3
   4     This file is part of DCP-o-matic.
   5
   6     DCP-o-matic is free software; you can redistribute it and/or modify
   7     it under the terms of the GNU General Public License as published by
   8     the Free Software Foundation; either version 2 of the License, or
   9     (at your option) any later version.
  10
  11     DCP-o-matic is distributed in the hope that it will be useful,
  12     but WITHOUT ANY WARRANTY; without even the implied warranty of
  13     MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  14     GNU General Public License for more details.
  15
  16     You should have received a copy of the GNU General Public License
  17     along with DCP-o-matic.  If not, see <http://www.gnu.org/licenses/>.
  18
  19 */
  20
  21 #include "resampler.h"
  22 #include "audio_buffers.h"
  23 #include "exceptions.h"
  24 #include "compose.hpp"
  25 #include "dcpomatic_assert.h"
  26 #include <samplerate.h>
  27 #include <iostream>
  28 #include <cmath>
  29
  30 #include "i18n.h"
  31
  32 using std::cout;
  33 using std::pair;
  34 using std::make_pair;
  35 using std::runtime_error;
  36 using boost::shared_ptr;
  37
  38 /** @param in Input sampling rate (Hz)
  39  *  @param out Output sampling rate (Hz)
  40  *  @param channels Number of channels.
  41  */
  42 Resampler::Resampler (int in, int out, int channels)
  43         : _in_rate (in)
  44         , _out_rate (out)
  45         , _channels (channels)
  46 {
  47         int error;
  48         _src = src_new (SRC_SINC_BEST_QUALITY, _channels, &error);
  49         if (!_src) {
  50                 throw runtime_error (String::compose (N_("could not create sample-rate converter (%1)"), error));
  51         }
  52 }
  53
  54 Resampler::~Resampler ()
  55 {
  56         src_delete (_src);
  57 }
  58
  59 void
  60 Resampler::set_fast ()
  61 {
  62         src_delete (_src);
  63         int error;
  64         _src = src_new (SRC_LINEAR, _channels, &error);
  65         if (!_src) {
  66                 throw runtime_error (String::compose (N_("could not create sample-rate converter (%1)"), error));
  67         }
  68 }
  69
  70 shared_ptr<const AudioBuffers>
  71 Resampler::run (shared_ptr<const AudioBuffers> in)
  72 {
  73         int in_frames = in->frames ();
  74         int in_offset = 0;
  75         int out_offset = 0;
  76         shared_ptr<AudioBuffers> resampled (new AudioBuffers (_channels, 0));
  77
  78         while (in_frames > 0) {
  79
  80                 /* Compute the resampled frames count and add 32 for luck */
  81                 int const max_resampled_frames = ceil ((double) in_frames * _out_rate / _in_rate) + 32;
  82
  83                 SRC_DATA data;
  84                 float* in_buffer = new float[in_frames * _channels];
  85
  86                 {
  87                         float** p = in->data ();
  88                         float* q = in_buffer;
  89                         for (int i = 0; i < in_frames; ++i) {
  90                                 for (int j = 0; j < _channels; ++j) {
  91                                         *q++ = p[j][in_offset + i];
  92                                 }
  93                         }
  94                 }
  95
  96                 data.data_in = in_buffer;
  97                 data.input_frames = in_frames;
  98
  99                 data.data_out = new float[max_resampled_frames * _channels];
 100                 data.output_frames = max_resampled_frames;
 101
 102                 data.end_of_input = 0;
 103                 data.src_ratio = double (_out_rate) / _in_rate;
 104
 105                 int const r = src_process (_src, &data);
 106                 if (r) {
 107                         delete[] data.data_in;
 108                         delete[] data.data_out;
 109                         throw EncodeError (
 110                                 String::compose (
 111                                         N_("could not run sample-rate converter (%1) [processing %2 to %3, %4 channels]"),
 112                                         src_strerror (r),
 113                                         in_frames,
 114                                         max_resampled_frames,
 115                                         _channels
 116                                         )
 117                                 );
 118                 }
 119
 120                 if (data.output_frames_gen == 0) {
 121                         delete[] data.data_in;
 122                         delete[] data.data_out;
 123                         break;
 124                 }
 125
 126                 resampled->ensure_size (out_offset + data.output_frames_gen);
 127                 resampled->set_frames (out_offset + data.output_frames_gen);
 128
 129                 {
 130                         float* p = data.data_out;
 131                         float** q = resampled->data ();
 132                         for (int i = 0; i < data.output_frames_gen; ++i) {
 133                                 for (int j = 0; j < _channels; ++j) {
 134                                         q[j][out_offset + i] = *p++;
 135                                 }
 136                         }
 137                 }
 138
 139                 in_frames -= data.input_frames_used;
 140                 in_offset += data.input_frames_used;
 141                 out_offset += data.output_frames_gen;
 142
 143                 delete[] data.data_in;
 144                 delete[] data.data_out;
 145         }
 146
 147         return resampled;
 148 }
 149
 150 shared_ptr<const AudioBuffers>
 151 Resampler::flush ()
 152 {
 153         shared_ptr<AudioBuffers> out (new AudioBuffers (_channels, 0));
 154         int out_offset = 0;
 155         int64_t const output_size = 65536;
 156
 157         float dummy[1];
 158         float* buffer = new float[output_size];
 159
 160         SRC_DATA data;
 161         data.data_in = dummy;
 162         data.input_frames = 0;
 163         data.data_out = buffer;
 164         data.output_frames = output_size;
 165         data.end_of_input = 1;
 166         data.src_ratio = double (_out_rate) / _in_rate;
 167
 168         int const r = src_process (_src, &data);
 169         if (r) {
 170                 delete[] buffer;
 171                 throw EncodeError (String::compose (N_("could not run sample-rate converter (%1)"), src_strerror (r)));
 172         }
 173
 174         out->ensure_size (out_offset + data.output_frames_gen);
 175
 176         float* p = data.data_out;
 177         float** q = out->data ();
 178         for (int i = 0; i < data.output_frames_gen; ++i) {
 179                 for (int j = 0; j < _channels; ++j) {
 180                         q[j][out_offset + i] = *p++;
 181                 }
 182         }
 183
 184         out_offset += data.output_frames_gen;
 185         out->set_frames (out_offset);
 186
 187         delete[] buffer;
 188         return out;
 189 }
 190
 191 void
 192 Resampler::reset ()
 193 {
 194         src_reset (_src);
 195 }