#include "audio_decoder.h"
#include "audio_buffers.h"
-#include "exceptions.h"
-#include "log.h"
+#include "audio_processor.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)
- , _decoded_audio (shared_ptr<AudioBuffers> (new AudioBuffers (content->audio_channels(), 0)), 0)
{
- 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 ()));
}
+
+ if (content->audio_processor ()) {
+ _processor = content->audio_processor()->clone (content->resampled_audio_frame_rate ());
+ }
+
+ reset_decoded_audio ();
+}
+
+void
+AudioDecoder::reset_decoded_audio ()
+{
+ _decoded_audio = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_audio_content->processed_audio_channels(), 0)), 0);
}
shared_ptr<ContentAudio>
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->content_audio_frame_rate()), accurate);
+ 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:
* otherwise any frames will do.
*/
if (accurate) {
- while (!pass() && _decoded_audio.audio->frames() < length) {}
- /* Use decoded_offset of 0, as we don't really care what frames we return */
- } else {
- while (!pass() && (_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end)) {}
+ /* Keep stuffing data into _decoded_audio until we have enough data, or the subclass does not want to give us any more */
+ while (_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end && !pass ()) {}
decoded_offset = frame - _decoded_audio.frame;
+ } else {
+ while (_decoded_audio.audio->frames() < length && !pass ()) {}
+ /* Use decoded_offset of 0, as we don't really care what frames we return */
}
- AudioFrame const amount_left = _decoded_audio.audio->frames() - decoded_offset;
-
- AudioFrame const to_return = min (amount_left, length);
+ /* 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);
-
- /* Clean up decoded */
- _decoded_audio.audio->move (decoded_offset + to_return, 0, amount_left - to_return);
- _decoded_audio.audio->set_frames (amount_left - to_return);
+
+ 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));
}
data = _resampler->run (data);
}
+ if (_processor) {
+ data = _processor->run (data);
+ }
+
+ AudioFrame const frame_rate = _audio_content->resampled_audio_frame_rate ();
+
+ if (_seek_reference) {
+ /* We've had an accurate seek and now we're seeing some data */
+ ContentTime const delta = time - _seek_reference.get ();
+ AudioFrame const delta_frames = delta.frames (frame_rate);
+ if (delta_frames > 0) {
+ /* This data comes after the seek time. Pad the data with some silence. */
+ shared_ptr<AudioBuffers> padded (new AudioBuffers (data->channels(), data->frames() + delta_frames));
+ padded->make_silent ();
+ padded->copy_from (data.get(), data->frames(), 0, delta_frames);
+ data = padded;
+ time -= delta;
+ } else if (delta_frames < 0) {
+ /* This data comes before the seek time. Throw some data away */
+ AudioFrame const to_discard = min (-delta_frames, static_cast<AudioFrame> (data->frames()));
+ AudioFrame const to_keep = data->frames() - to_discard;
+ if (to_keep == 0) {
+ /* We have to throw all this data away, so keep _seek_reference and
+ try again next time some data arrives.
+ */
+ return;
+ }
+ shared_ptr<AudioBuffers> trimmed (new AudioBuffers (data->channels(), to_keep));
+ trimmed->copy_from (data.get(), to_keep, to_discard, 0);
+ data = trimmed;
+ time += ContentTime::from_frames (to_discard, frame_rate);
+ }
+ _seek_reference = optional<ContentTime> ();
+ }
+
if (!_audio_position) {
- _audio_position = time.frames (_audio_content->output_audio_frame_rate ());
+ _audio_position = time.frames (frame_rate);
}
- assert (_audio_position >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
+ assert (_audio_position.get() >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
+ add (data);
+}
+
+void
+AudioDecoder::add (shared_ptr<const AudioBuffers> data)
+{
/* Resize _decoded_audio to fit the new data */
- int const new_size = _audio_position.get() + data->frames() - _decoded_audio.frame;
+ 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;
+ }
+
_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 ();
+
+ /* Limit the amount of data we keep in case nobody is asking for it */
+ int const max_frames = _audio_content->resampled_audio_frame_rate () * 10;
+ if (_decoded_audio.audio->frames() > max_frames) {
+ int const to_remove = _decoded_audio.audio->frames() - max_frames;
+ _decoded_audio.frame += to_remove;
+ _decoded_audio.audio->move (to_remove, 0, max_frames);
+ _decoded_audio.audio->set_frames (max_frames);
+ }
}
-/* XXX: called? */
void
AudioDecoder::flush ()
{
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 ();
+ add (b);
}
- */
}
void
-AudioDecoder::seek (ContentTime, bool)
+AudioDecoder::seek (ContentTime t, bool accurate)
{
_audio_position.reset ();
+ reset_decoded_audio ();
+ if (accurate) {
+ _seek_reference = t;
+ }
+ if (_processor) {
+ _processor->flush ();
+ }
}
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