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/*
Copyright (C) 2012-2014 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 "resampler.h"
#include "util.h"
#include "film.h"
#include "i18n.h"
using std::stringstream;
using std::list;
using std::pair;
using std::cout;
using std::min;
using std::max;
using boost::optional;
using boost::shared_ptr;
AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> content)
: _audio_content (content)
{
if (content->resampled_audio_frame_rate() != content->audio_frame_rate() && content->audio_channels ()) {
_resampler.reset (new Resampler (content->audio_frame_rate(), content->resampled_audio_frame_rate(), content->audio_channels ()));
}
reset_decoded_audio ();
}
void
AudioDecoder::reset_decoded_audio ()
{
_decoded_audio = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_audio_content->audio_channels(), 0)), 0);
}
shared_ptr<ContentAudio>
AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate)
{
shared_ptr<ContentAudio> dec;
AudioFrame const end = frame + length - 1;
if (frame < _decoded_audio.frame || end > (_decoded_audio.frame + length * 4)) {
/* Either we have no decoded data, or what we do have is a long way from what we want: seek */
seek (ContentTime::from_frames (frame, _audio_content->audio_frame_rate()), accurate);
}
/* Offset of the data that we want from the start of _decoded_audio.audio
(to be set up shortly)
*/
AudioFrame decoded_offset = 0;
/* Now enough pass() calls will either:
* (a) give us what we want, or
* (b) hit the end of the decoder.
*
* If we are being accurate, we want the right frames,
* otherwise any frames will do.
*/
if (accurate) {
/* Keep stuffing data into _decoded_audio until we have enough data, or the subclass does not want to give us any more */
while (!pass() && (_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end)) {}
decoded_offset = frame - _decoded_audio.frame;
} else {
while (!pass() && _decoded_audio.audio->frames() < length) {}
/* Use decoded_offset of 0, as we don't really care what frames we return */
}
/* The amount of data available in _decoded_audio.audio starting from `frame'. This could be -ve
if pass() returned true before we got enough data.
*/
AudioFrame const available = _decoded_audio.audio->frames() - decoded_offset;
/* We will return either that, or the requested amount, whichever is smaller */
AudioFrame const to_return = max ((AudioFrame) 0, min (available, length));
/* Copy our data to the output */
shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded_audio.audio->channels(), to_return));
out->copy_from (_decoded_audio.audio.get(), to_return, decoded_offset, 0);
AudioFrame const remaining = max ((AudioFrame) 0, available - to_return);
/* Clean up decoded; first, move the data after what we just returned to the start of the buffer */
_decoded_audio.audio->move (decoded_offset + to_return, 0, remaining);
/* And set up the number of frames we have left */
_decoded_audio.audio->set_frames (remaining);
/* Also bump where those frames are in terms of the content */
_decoded_audio.frame += decoded_offset + to_return;
return shared_ptr<ContentAudio> (new ContentAudio (out, frame));
}
/** Called by subclasses when audio data is ready.
*
* Audio timestamping is made hard by many factors, but perhaps the most entertaining is resampling.
* We have to assume that we are feeding continuous data into the resampler, and so we get continuous
* data out. Hence we do the timestamping here, post-resampler, just by counting samples.
*
* The time is passed in here so that after a seek we can set up our _audio_position. The
* time is ignored once this has been done.
*/
void
AudioDecoder::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
{
if (_resampler) {
data = _resampler->run (data);
}
if (!_audio_position) {
_audio_position = time.frames (_audio_content->resampled_audio_frame_rate ());
}
assert (_audio_position.get() >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
/* Resize _decoded_audio to fit the new data */
int const new_size = _audio_position.get() + data->frames() - _decoded_audio.frame;
_decoded_audio.audio->ensure_size (new_size);
_decoded_audio.audio->set_frames (new_size);
/* Copy new data in */
_decoded_audio.audio->copy_from (data.get(), data->frames(), 0, _audio_position.get() - _decoded_audio.frame);
_audio_position = _audio_position.get() + data->frames ();
}
/* XXX: called? */
void
AudioDecoder::flush ()
{
if (!_resampler) {
return;
}
/*
shared_ptr<const AudioBuffers> b = _resampler->flush ();
if (b) {
_pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (b, _audio_position.get ())));
_audio_position = _audio_position.get() + b->frames ();
}
*/
}
void
AudioDecoder::seek (ContentTime, bool)
{
_audio_position.reset ();
reset_decoded_audio ();
}
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