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/*
Copyright (C) 2013-2015 Carl Hetherington <cth@carlh.net>
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include "resampler.h"
#include "audio_buffers.h"
#include "exceptions.h"
#include "compose.hpp"
#include "dcpomatic_assert.h"
#include <samplerate.h>
#include "i18n.h"
using std::cout;
using std::pair;
using std::make_pair;
using boost::shared_ptr;
/** @param in Input sampling rate (Hz)
* @param out Output sampling rate (Hz)
* @param channels Number of channels.
* @param fast true to be fast rather than good.
*/
Resampler::Resampler (int in, int out, int channels, bool fast)
: _in_rate (in)
, _out_rate (out)
, _channels (channels)
{
int error;
_src = src_new (fast ? SRC_LINEAR : SRC_SINC_BEST_QUALITY, _channels, &error);
if (!_src) {
throw StringError (String::compose (N_("could not create sample-rate converter (%1)"), error));
}
}
Resampler::~Resampler ()
{
src_delete (_src);
}
shared_ptr<const AudioBuffers>
Resampler::run (shared_ptr<const AudioBuffers> in)
{
int in_frames = in->frames ();
int in_offset = 0;
int out_offset = 0;
shared_ptr<AudioBuffers> resampled (new AudioBuffers (_channels, 0));
while (in_frames > 0) {
/* Compute the resampled frames count and add 32 for luck */
int const max_resampled_frames = ceil ((double) in_frames * _out_rate / _in_rate) + 32;
SRC_DATA data;
data.data_in = new float[in_frames * _channels];
{
float** p = in->data ();
float* q = data.data_in;
for (int i = 0; i < in_frames; ++i) {
for (int j = 0; j < _channels; ++j) {
*q++ = p[j][in_offset + i];
}
}
}
data.input_frames = in_frames;
data.data_out = new float[max_resampled_frames * _channels];
data.output_frames = max_resampled_frames;
data.end_of_input = 0;
data.src_ratio = double (_out_rate) / _in_rate;
int const r = src_process (_src, &data);
if (r) {
delete[] data.data_in;
delete[] data.data_out;
throw EncodeError (String::compose (N_("could not run sample-rate converter (%1)"), src_strerror (r)));
}
if (data.output_frames_gen == 0) {
break;
}
resampled->ensure_size (out_offset + data.output_frames_gen);
resampled->set_frames (out_offset + data.output_frames_gen);
{
float* p = data.data_out;
float** q = resampled->data ();
for (int i = 0; i < data.output_frames_gen; ++i) {
for (int j = 0; j < _channels; ++j) {
q[j][out_offset + i] = *p++;
}
}
}
in_frames -= data.input_frames_used;
in_offset += data.input_frames_used;
out_offset += data.output_frames_gen;
delete[] data.data_in;
delete[] data.data_out;
}
return resampled;
}
shared_ptr<const AudioBuffers>
Resampler::flush ()
{
shared_ptr<AudioBuffers> out (new AudioBuffers (_channels, 0));
int out_offset = 0;
int64_t const output_size = 65536;
float dummy[1];
float buffer[output_size];
SRC_DATA data;
data.data_in = dummy;
data.input_frames = 0;
data.data_out = buffer;
data.output_frames = output_size;
data.end_of_input = 1;
data.src_ratio = double (_out_rate) / _in_rate;
int const r = src_process (_src, &data);
if (r) {
throw EncodeError (String::compose (N_("could not run sample-rate converter (%1)"), src_strerror (r)));
}
out->ensure_size (out_offset + data.output_frames_gen);
float* p = data.data_out;
float** q = out->data ();
for (int i = 0; i < data.output_frames_gen; ++i) {
for (int j = 0; j < _channels; ++j) {
q[j][out_offset + i] = *p++;
}
}
out_offset += data.output_frames_gen;
out->set_frames (out_offset);
return out;
}
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