/*
- Copyright (C) 2012 Carl Hetherington <cth@carlh.net>
+ 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
#include "exceptions.h"
#include "log.h"
#include "resampler.h"
+#include "util.h"
+#include "film.h"
#include "i18n.h"
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 Film> film, shared_ptr<const AudioContent> content)
- : Decoder (film)
- , _audio_content (content)
- , _last_audio (0)
+AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> content)
+ : _audio_content (content)
{
- if (content->output_audio_frame_rate() != content->content_audio_frame_rate() && content->audio_channels ()) {
- _resampler.reset (new Resampler (content->content_audio_frame_rate(), content->output_audio_frame_rate(), content->audio_channels ()));
+ 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 new_size = 0;
+ if (_decoded_audio.audio->frames() == 0) {
+ /* There's nothing in there, so just store the new data */
+ new_size = data->frames ();
+ _decoded_audio.frame = _audio_position.get ();
+ } else {
+ /* Otherwise we need to extend _decoded_audio to include the new stuff */
+ new_size = _audio_position.get() + data->frames() - _decoded_audio.frame;
+ }
- _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (data, time)));
- _last_audio = time + (data->frames() * TIME_HZ / _audio_content->output_audio_frame_rate());
+ _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 ()
{
return;
}
+ /*
shared_ptr<const AudioBuffers> b = _resampler->flush ();
if (b) {
- audio (b, _last_audio);
+ _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 ();
}
projects / dcpomatic.git / blobdiff