/*
- Copyright (C) 2013-2015 Carl Hetherington <cth@carlh.net>
+ Copyright (C) 2013-2021 Carl Hetherington <cth@carlh.net>
- This program is free software; you can redistribute it and/or modify
+ This file is part of DCP-o-matic.
+
+ DCP-o-matic 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,
+ DCP-o-matic 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.
+ along with DCP-o-matic. If not, see <http://www.gnu.org/licenses/>.
*/
+
#include "resampler.h"
#include "audio_buffers.h"
#include "exceptions.h"
#include "dcpomatic_assert.h"
#include <samplerate.h>
#include <iostream>
+#include <cmath>
#include "i18n.h"
+
using std::cout;
-using std::pair;
using std::make_pair;
+using std::make_shared;
+using std::pair;
using std::runtime_error;
-using boost::shared_ptr;
+using std::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)
+Resampler::Resampler (int in, int out, int channels)
: _in_rate (in)
, _out_rate (out)
, _channels (channels)
{
int error;
- _src = src_new (fast ? SRC_LINEAR : SRC_SINC_BEST_QUALITY, _channels, &error);
+ _src = src_new (SRC_SINC_BEST_QUALITY, _channels, &error);
if (!_src) {
- throw runtime_error (String::compose (N_("could not create sample-rate converter (%1)"), error));
+ throw runtime_error (String::compose(N_("could not create sample-rate converter (%1)"), error));
}
}
+
Resampler::~Resampler ()
+{
+ if (_src) {
+ src_delete (_src);
+ }
+}
+
+
+void
+Resampler::set_fast ()
{
src_delete (_src);
+ _src = nullptr;
+
+ int error;
+ _src = src_new (SRC_LINEAR, _channels, &error);
+ if (!_src) {
+ throw runtime_error (String::compose(N_("could not create sample-rate converter (%1)"), error));
+ }
}
+
shared_ptr<const AudioBuffers>
Resampler::run (shared_ptr<const AudioBuffers> in)
{
+ DCPOMATIC_ASSERT(in->channels() == _channels);
+
int in_frames = in->frames ();
int in_offset = 0;
int out_offset = 0;
- shared_ptr<AudioBuffers> resampled (new AudioBuffers (_channels, 0));
+ auto resampled = make_shared<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;
+ int const max_resampled_frames = ceil (static_cast<double>(in_frames) * _out_rate / _in_rate) + 32;
SRC_DATA data;
- data.data_in = new float[in_frames * _channels];
+ std::vector<float> in_buffer(in_frames * _channels);
+ std::vector<float> out_buffer(max_resampled_frames * _channels);
{
- float** p = in->data ();
- float* q = data.data_in;
+ auto p = in->data ();
+ auto q = in_buffer.data();
for (int i = 0; i < in_frames; ++i) {
for (int j = 0; j < _channels; ++j) {
*q++ = p[j][in_offset + i];
}
}
+ data.data_in = in_buffer.data();
data.input_frames = in_frames;
- data.data_out = new float[max_resampled_frames * _channels];
+ data.data_out = out_buffer.data();
data.output_frames = max_resampled_frames;
data.end_of_input = 0;
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) [processing %2 to %3, %4 channels]"),
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 ();
+ auto p = data.data_out;
+ auto 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));
+ auto out = make_shared<AudioBuffers>(_channels, 0);
int out_offset = 0;
int64_t const output_size = 65536;
- /* I think this should only need to be 1 long, but I have seen
- src_process error with "input and output data arrays overlap"
- with dummy[1] (on OS X). I've added a few more for luck.
- */
- float dummy[16];
- float buffer[output_size];
+ float dummy[1];
+ std::vector<float> buffer(output_size);
SRC_DATA data;
data.data_in = dummy;
data.input_frames = 0;
- data.data_out = buffer;
+ data.data_out = buffer.data();
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)));
+ throw EncodeError (String::compose(N_("could not run sample-rate converter (%1)"), src_strerror(r)));
}
- out->ensure_size (out_offset + data.output_frames_gen);
+ out->set_frames (out_offset + data.output_frames_gen);
- float* p = data.data_out;
- float** q = out->data ();
+ auto p = data.data_out;
+ auto 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;
}
+
+
+void
+Resampler::reset ()
+{
+ src_reset (_src);
+}
+
projects / dcpomatic.git / blobdiff