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Basic release notes support (#2282).
[dcpomatic.git]
/
src
/
lib
/
resampler.cc
diff --git
a/src/lib/resampler.cc
b/src/lib/resampler.cc
index 4010390b8e663ebf413efe09ceb10498aace41a2..4447ccf0dfc2728ad6cffe3b78fee435c2f3ec09 100644
(file)
--- a/
src/lib/resampler.cc
+++ b/
src/lib/resampler.cc
@@
-1,22
+1,24
@@
/*
/*
- Copyright (C) 2013-20
15
Carl Hetherington <cth@carlh.net>
+ Copyright (C) 2013-20
21
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.
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
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 "resampler.h"
#include "audio_buffers.h"
#include "exceptions.h"
@@
-24,56
+26,78
@@
#include "dcpomatic_assert.h"
#include <samplerate.h>
#include <iostream>
#include "dcpomatic_assert.h"
#include <samplerate.h>
#include <iostream>
+#include <cmath>
#include "i18n.h"
#include "i18n.h"
+
using std::cout;
using std::cout;
-using std::pair;
using std::make_pair;
using std::make_pair;
+using std::make_shared;
+using std::pair;
using std::runtime_error;
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 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;
: _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) {
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 ()
Resampler::~Resampler ()
+{
+ if (_src) {
+ src_delete (_src);
+ }
+}
+
+
+void
+Resampler::set_fast ()
{
src_delete (_src);
{
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)
{
int in_frames = in->frames ();
int in_offset = 0;
int out_offset = 0;
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)
);
+
auto resampled = make_shared<AudioBuffers>(_channels, 0
);
while (in_frames > 0) {
/* Compute the resampled frames count and add 32 for luck */
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;
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];
for (int i = 0; i < in_frames; ++i) {
for (int j = 0; j < _channels; ++j) {
*q++ = p[j][in_offset + i];
@@
-81,9
+105,10
@@
Resampler::run (shared_ptr<const AudioBuffers> in)
}
}
}
}
+ data.data_in = in_buffer.data();
data.input_frames = in_frames;
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;
data.output_frames = max_resampled_frames;
data.end_of_input = 0;
@@
-91,21
+116,26
@@
Resampler::run (shared_ptr<const AudioBuffers> in)
int const r = src_process (_src, &data);
if (r) {
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)));
+ throw EncodeError (
+ String::compose (
+ N_("could not run sample-rate converter (%1) [processing %2 to %3, %4 channels]"),
+ src_strerror (r),
+ in_frames,
+ max_resampled_frames,
+ _channels
+ )
+ );
}
if (data.output_frames_gen == 0) {
break;
}
}
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);
{
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++;
for (int i = 0; i < data.output_frames_gen; ++i) {
for (int j = 0; j < _channels; ++j) {
q[j][out_offset + i] = *p++;
@@
-116,41
+146,39
@@
Resampler::run (shared_ptr<const AudioBuffers> in)
in_frames -= data.input_frames_used;
in_offset += data.input_frames_used;
out_offset += data.output_frames_gen;
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;
}
}
return resampled;
}
+
shared_ptr<const AudioBuffers>
Resampler::flush ()
{
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;
float dummy[1];
int out_offset = 0;
int64_t const output_size = 65536;
float dummy[1];
-
float buffer[output_size]
;
+
std::vector<float> buffer(output_size)
;
SRC_DATA data;
data.data_in = dummy;
data.input_frames = 0;
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) {
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++;
for (int i = 0; i < data.output_frames_gen; ++i) {
for (int j = 0; j < _channels; ++j) {
q[j][out_offset + i] = *p++;
@@
-158,7
+186,14
@@
Resampler::flush ()
}
out_offset += data.output_frames_gen;
}
out_offset += data.output_frames_gen;
- out->set_frames (out_offset);
return out;
}
return out;
}
+
+
+void
+Resampler::reset ()
+{
+ src_reset (_src);
+}
+