2 Copyright (C) 2013-2015 Carl Hetherington <cth@carlh.net>
4 This file is part of DCP-o-matic.
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.
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.
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/>.
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>
34 using std::runtime_error;
35 using boost::shared_ptr;
37 /** @param in Input sampling rate (Hz)
38 * @param out Output sampling rate (Hz)
39 * @param channels Number of channels.
41 Resampler::Resampler (int in, int out, int channels)
44 , _channels (channels)
47 _src = src_new (SRC_SINC_BEST_QUALITY, _channels, &error);
49 throw runtime_error (String::compose (N_("could not create sample-rate converter (%1)"), error));
53 Resampler::~Resampler ()
59 Resampler::set_fast ()
63 _src = src_new (SRC_LINEAR, _channels, &error);
65 throw runtime_error (String::compose (N_("could not create sample-rate converter (%1)"), error));
69 shared_ptr<const AudioBuffers>
70 Resampler::run (shared_ptr<const AudioBuffers> in)
72 int in_frames = in->frames ();
75 shared_ptr<AudioBuffers> resampled (new AudioBuffers (_channels, 0));
77 while (in_frames > 0) {
79 /* Compute the resampled frames count and add 32 for luck */
80 int const max_resampled_frames = ceil ((double) in_frames * _out_rate / _in_rate) + 32;
83 data.data_in = new float[in_frames * _channels];
86 float** p = in->data ();
87 float* q = data.data_in;
88 for (int i = 0; i < in_frames; ++i) {
89 for (int j = 0; j < _channels; ++j) {
90 *q++ = p[j][in_offset + i];
95 data.input_frames = in_frames;
97 data.data_out = new float[max_resampled_frames * _channels];
98 data.output_frames = max_resampled_frames;
100 data.end_of_input = 0;
101 data.src_ratio = double (_out_rate) / _in_rate;
103 int const r = src_process (_src, &data);
105 delete[] data.data_in;
106 delete[] data.data_out;
109 N_("could not run sample-rate converter (%1) [processing %2 to %3, %4 channels]"),
112 max_resampled_frames,
118 if (data.output_frames_gen == 0) {
122 resampled->ensure_size (out_offset + data.output_frames_gen);
123 resampled->set_frames (out_offset + data.output_frames_gen);
126 float* p = data.data_out;
127 float** q = resampled->data ();
128 for (int i = 0; i < data.output_frames_gen; ++i) {
129 for (int j = 0; j < _channels; ++j) {
130 q[j][out_offset + i] = *p++;
135 in_frames -= data.input_frames_used;
136 in_offset += data.input_frames_used;
137 out_offset += data.output_frames_gen;
139 delete[] data.data_in;
140 delete[] data.data_out;
146 shared_ptr<const AudioBuffers>
149 shared_ptr<AudioBuffers> out (new AudioBuffers (_channels, 0));
151 int64_t const output_size = 65536;
154 float* buffer = new float[output_size];
157 data.data_in = dummy;
158 data.input_frames = 0;
159 data.data_out = buffer;
160 data.output_frames = output_size;
161 data.end_of_input = 1;
162 data.src_ratio = double (_out_rate) / _in_rate;
164 int const r = src_process (_src, &data);
167 throw EncodeError (String::compose (N_("could not run sample-rate converter (%1)"), src_strerror (r)));
170 out->ensure_size (out_offset + data.output_frames_gen);
172 float* p = data.data_out;
173 float** q = out->data ();
174 for (int i = 0; i < data.output_frames_gen; ++i) {
175 for (int j = 0; j < _channels; ++j) {
176 q[j][out_offset + i] = *p++;
180 out_offset += data.output_frames_gen;
181 out->set_frames (out_offset);