/*
Copyright (C) 2013-2015 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/>.
*/
-extern "C" {
-#include "libavutil/channel_layout.h"
-#include "libavutil/opt.h"
-}
#include "resampler.h"
#include "audio_buffers.h"
#include "exceptions.h"
#include "compose.hpp"
+#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::runtime_error;
using boost::shared_ptr;
+/** @param in Input sampling rate (Hz)
+ * @param out Output sampling rate (Hz)
+ * @param channels Number of channels.
+ */
Resampler::Resampler (int in, int out, int channels)
: _in_rate (in)
, _out_rate (out)
, _channels (channels)
{
- _swr_context = swr_alloc ();
-
- /* Sample formats */
- av_opt_set_int (_swr_context, "isf", AV_SAMPLE_FMT_FLTP, 0);
- av_opt_set_int (_swr_context, "osf", AV_SAMPLE_FMT_FLTP, 0);
-
- /* Channel counts */
- av_opt_set_int (_swr_context, "ich", _channels, 0);
- av_opt_set_int (_swr_context, "och", _channels, 0);
-
- /* Sample rates */
- av_opt_set_int (_swr_context, "isr", _in_rate, 0);
- av_opt_set_int (_swr_context, "osr", _out_rate, 0);
-
- av_opt_set (_swr_context, "resampler", "soxr", 0);
-
- swr_init (_swr_context);
+ int 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));
+ }
}
Resampler::~Resampler ()
{
- swr_free (&_swr_context);
+ if (_src) {
+ src_delete (_src);
+ }
+}
+
+void
+Resampler::set_fast ()
+{
+ src_delete (_src);
+ _src = 0;
+
+ 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)
{
- /* Compute the resampled frames count and add 32 for luck */
- int const max_resampled_frames = ceil ((double) in->frames() * _out_rate / _in_rate) + 32;
- shared_ptr<AudioBuffers> resampled (new AudioBuffers (_channels, max_resampled_frames));
-
- int const resampled_frames = swr_convert (
- _swr_context, (uint8_t **) resampled->data(), max_resampled_frames, (uint8_t const **) in->data(), in->frames()
- );
-
- if (resampled_frames < 0) {
- char buf[256];
- av_strerror (resampled_frames, buf, sizeof(buf));
- throw EncodeError (String::compose (_("could not run sample-rate converter for %1 samples (%2) (%3)"), in->frames(), resampled_frames, buf));
+ 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;
+ float* in_buffer = new float[in_frames * _channels];
+
+ {
+ float** p = in->data ();
+ float* q = in_buffer;
+ 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.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) [processing %2 to %3, %4 channels]"),
+ src_strerror (r),
+ in_frames,
+ max_resampled_frames,
+ _channels
+ )
+ );
+ }
+
+ if (data.output_frames_gen == 0) {
+ delete[] data.data_in;
+ delete[] data.data_out;
+ 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;
}
- resampled->set_frames (resampled_frames);
return resampled;
}
{
shared_ptr<AudioBuffers> out (new AudioBuffers (_channels, 0));
int out_offset = 0;
- int64_t const pass_size = 256;
- shared_ptr<AudioBuffers> pass (new AudioBuffers (_channels, 256));
+ int64_t const output_size = 65536;
+
+ float dummy[1];
+ float* buffer = new float[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) {
+ delete[] buffer;
+ throw EncodeError (String::compose (N_("could not run sample-rate converter (%1)"), src_strerror (r)));
+ }
- while (true) {
- int const frames = swr_convert (_swr_context, (uint8_t **) pass->data(), pass_size, 0, 0);
+ out->ensure_size (out_offset + data.output_frames_gen);
- if (frames < 0) {
- throw EncodeError (_("could not run sample-rate converter"));
+ 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++;
}
-
- if (frames == 0) {
- break;
- }
-
- out->ensure_size (out_offset + frames);
- out->copy_from (pass.get(), frames, 0, out_offset);
- out_offset += frames;
- out->set_frames (out_offset);
}
+ out_offset += data.output_frames_gen;
+ out->set_frames (out_offset);
+
+ delete[] buffer;
return out;
}
+
+void
+Resampler::reset ()
+{
+ src_reset (_src);
+}
projects / dcpomatic.git / blobdiff