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>
35 using std::runtime_error;
36 using boost::shared_ptr;
38 /** @param in Input sampling rate (Hz)
39 * @param out Output sampling rate (Hz)
40 * @param channels Number of channels.
42 Resampler::Resampler (int in, int out, int channels)
45 , _channels (channels)
48 _src = src_new (SRC_SINC_BEST_QUALITY, _channels, &error);
50 throw runtime_error (String::compose (N_("could not create sample-rate converter (%1)"), error));
54 Resampler::~Resampler ()
60 Resampler::set_fast ()
64 _src = src_new (SRC_LINEAR, _channels, &error);
66 throw runtime_error (String::compose (N_("could not create sample-rate converter (%1)"), error));
70 pair<shared_ptr<const AudioBuffers>, Frame>
71 Resampler::run (shared_ptr<const AudioBuffers> in, Frame frame)
73 if (!_next_in || !_next_out || _next_in.get() != frame) {
74 /* Either there was a discontinuity in the input or this is the first input;
77 _next_out = lrintf (frame * _out_rate / _in_rate);
80 /* Expected next input frame */
81 _next_in = frame + in->frames ();
83 int in_frames = in->frames ();
86 shared_ptr<AudioBuffers> resampled (new AudioBuffers (_channels, 0));
88 while (in_frames > 0) {
90 /* Compute the resampled frames count and add 32 for luck */
91 int const max_resampled_frames = ceil ((double) in_frames * _out_rate / _in_rate) + 32;
94 float* in_buffer = new float[in_frames * _channels];
97 float** p = in->data ();
99 for (int i = 0; i < in_frames; ++i) {
100 for (int j = 0; j < _channels; ++j) {
101 *q++ = p[j][in_offset + i];
106 data.data_in = in_buffer;
107 data.input_frames = in_frames;
109 data.data_out = new float[max_resampled_frames * _channels];
110 data.output_frames = max_resampled_frames;
112 data.end_of_input = 0;
113 data.src_ratio = double (_out_rate) / _in_rate;
115 int const r = src_process (_src, &data);
117 delete[] data.data_in;
118 delete[] data.data_out;
121 N_("could not run sample-rate converter (%1) [processing %2 to %3, %4 channels]"),
124 max_resampled_frames,
130 if (data.output_frames_gen == 0) {
131 delete[] data.data_in;
132 delete[] data.data_out;
136 resampled->ensure_size (out_offset + data.output_frames_gen);
137 resampled->set_frames (out_offset + data.output_frames_gen);
140 float* p = data.data_out;
141 float** q = resampled->data ();
142 for (int i = 0; i < data.output_frames_gen; ++i) {
143 for (int j = 0; j < _channels; ++j) {
144 q[j][out_offset + i] = *p++;
149 in_frames -= data.input_frames_used;
150 in_offset += data.input_frames_used;
151 out_offset += data.output_frames_gen;
153 delete[] data.data_in;
154 delete[] data.data_out;
157 Frame out_frame = _next_out.get ();
159 /* Expected next output frame */
160 _next_out = _next_out.get() + resampled->frames();
162 return make_pair (resampled, out_frame);
165 pair<shared_ptr<const AudioBuffers>, Frame>
168 shared_ptr<AudioBuffers> out (new AudioBuffers (_channels, 0));
170 int64_t const output_size = 65536;
173 float* buffer = new float[output_size];
176 data.data_in = dummy;
177 data.input_frames = 0;
178 data.data_out = buffer;
179 data.output_frames = output_size;
180 data.end_of_input = 1;
181 data.src_ratio = double (_out_rate) / _in_rate;
183 int const r = src_process (_src, &data);
186 throw EncodeError (String::compose (N_("could not run sample-rate converter (%1)"), src_strerror (r)));
189 out->ensure_size (out_offset + data.output_frames_gen);
191 float* p = data.data_out;
192 float** q = out->data ();
193 for (int i = 0; i < data.output_frames_gen; ++i) {
194 for (int j = 0; j < _channels; ++j) {
195 q[j][out_offset + i] = *p++;
199 out_offset += data.output_frames_gen;
200 out->set_frames (out_offset);
203 return make_pair (out, _next_out.get ());