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/*
Copyright (C) 2012 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 "audio_decoder.h"
#include "audio_buffers.h"
#include "exceptions.h"
#include "log.h"
#include "i18n.h"
using std::stringstream;
using std::list;
using std::pair;
using boost::optional;
using boost::shared_ptr;
AudioDecoder::AudioDecoder (shared_ptr<const Film> f, shared_ptr<const AudioContent> c)
: Decoder (f)
, _next_audio (0)
, _audio_content (c)
{
if (_audio_content->content_audio_frame_rate() != _audio_content->output_audio_frame_rate()) {
shared_ptr<const Film> film = _film.lock ();
assert (film);
stringstream s;
s << String::compose (
"Will resample audio from %1 to %2",
_audio_content->content_audio_frame_rate(), _audio_content->output_audio_frame_rate()
);
film->log()->log (s.str ());
/* We will be using planar float data when we call the
resampler. As far as I can see, the audio channel
layout is not necessary for our purposes; it seems
only to be used get the number of channels and
decide if rematrixing is needed. It won't be, since
input and output layouts are the same.
*/
_swr_context = swr_alloc_set_opts (
0,
av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
AV_SAMPLE_FMT_FLTP,
_audio_content->output_audio_frame_rate(),
av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
AV_SAMPLE_FMT_FLTP,
_audio_content->content_audio_frame_rate(),
0, 0
);
swr_init (_swr_context);
} else {
_swr_context = 0;
}
}
AudioDecoder::~AudioDecoder ()
{
if (_swr_context) {
swr_free (&_swr_context);
}
}
#if 0
void
AudioDecoder::process_end ()
{
if (_swr_context) {
shared_ptr<const Film> film = _film.lock ();
assert (film);
shared_ptr<AudioBuffers> out (new AudioBuffers (film->audio_mapping().dcp_channels(), 256));
while (1) {
int const frames = swr_convert (_swr_context, (uint8_t **) out->data(), 256, 0, 0);
if (frames < 0) {
throw EncodeError (_("could not run sample-rate converter"));
}
if (frames == 0) {
break;
}
out->set_frames (frames);
_writer->write (out);
}
}
}
#endif
void
AudioDecoder::audio (shared_ptr<const AudioBuffers> data, Time time)
{
/* Maybe resample */
if (_swr_context) {
/* Compute the resampled frames count and add 32 for luck */
int const max_resampled_frames = ceil (
(int64_t) data->frames() * _audio_content->output_audio_frame_rate() / _audio_content->content_audio_frame_rate()
) + 32;
shared_ptr<AudioBuffers> resampled (new AudioBuffers (data->channels(), max_resampled_frames));
/* Resample audio */
int const resampled_frames = swr_convert (
_swr_context, (uint8_t **) resampled->data(), max_resampled_frames, (uint8_t const **) data->data(), data->frames()
);
if (resampled_frames < 0) {
throw EncodeError (_("could not run sample-rate converter"));
}
resampled->set_frames (resampled_frames);
/* And point our variables at the resampled audio */
data = resampled;
}
shared_ptr<const Film> film = _film.lock ();
assert (film);
/* Remap channels */
shared_ptr<AudioBuffers> dcp_mapped (new AudioBuffers (film->dcp_audio_channels(), data->frames()));
dcp_mapped->make_silent ();
list<pair<int, libdcp::Channel> > map = _audio_content->audio_mapping().content_to_dcp ();
for (list<pair<int, libdcp::Channel> >::iterator i = map.begin(); i != map.end(); ++i) {
dcp_mapped->accumulate_channel (data.get(), i->first, i->second);
}
Audio (dcp_mapped, time);
_next_audio = time + film->audio_frames_to_time (data->frames());
}
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