-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 ((_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 */
- }
-
- /* 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.
- */