diff options
| author | Carl Hetherington <cth@carlh.net> | 2014-03-25 09:41:36 +0000 |
|---|---|---|
| committer | Carl Hetherington <cth@carlh.net> | 2014-03-25 09:41:36 +0000 |
| commit | ee77b3cf5f59f775e75e628aa28e8f2f9f941530 (patch) | |
| tree | bbf9ab4ef1f0f633591889cbbd6b7b65de8f5a57 /src/lib/audio_decoder.cc | |
| parent | e6f28e7cda23c1ba3c49cc1bf2dc1491c2f87160 (diff) | |
It builds.
Diffstat (limited to 'src/lib/audio_decoder.cc')
| -rw-r--r-- | src/lib/audio_decoder.cc | 64 |
1 files changed, 57 insertions, 7 deletions
diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc index 32453cc13..616abf846 100644 --- a/src/lib/audio_decoder.cc +++ b/src/lib/audio_decoder.cc @@ -1,5 +1,5 @@ /* - 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 @@ -23,6 +23,7 @@ #include "log.h" #include "resampler.h" #include "util.h" +#include "film.h" #include "i18n.h" @@ -41,7 +42,47 @@ AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> content) } } -/** Audio timestamping is made hard by many factors, but the final nail in the coffin is resampling. +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->content_audio_frame_rate()), accurate); + } + + /* 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) { + while (!pass() && _decoded_audio.audio->frames() < length) {} + } else { + while (!pass() && (_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end)) {} + } + + /* Clean up decoded */ + + AudioFrame const decoded_offset = frame - _decoded_audio.frame; + AudioFrame const amount_left = _decoded_audio.audio->frames() - decoded_offset; + _decoded_audio.audio->move (decoded_offset, 0, amount_left); + _decoded_audio.audio->set_frames (amount_left); + + shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded_audio.audio->channels(), length)); + out->copy_from (_decoded_audio.audio.get(), length, frame - _decoded_audio.frame, 0); + + 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. * @@ -56,13 +97,20 @@ AudioDecoder::audio (shared_ptr<const AudioBuffers> data, ContentTime time) } if (!_audio_position) { - _audio_position = time; + _audio_position = time.frames (_audio_content->output_audio_frame_rate ()); } - _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (_audio_position.get (), data))); - _audio_position = _audio_position.get() + ContentTime (data->frames (), _audio_content->output_audio_frame_rate ()); + assert (_audio_position >= (_decoded_audio.frame + _decoded_audio.audio->frames())); + + /* Resize _decoded_audio to fit the new data */ + _decoded_audio.audio->ensure_size (_audio_position.get() + data->frames() - _decoded_audio.frame); + + /* 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 () { @@ -70,11 +118,13 @@ AudioDecoder::flush () return; } + /* shared_ptr<const AudioBuffers> b = _resampler->flush (); if (b) { - _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (_audio_position.get (), b))); - _audio_position = _audio_position.get() + ContentTime (b->frames (), _audio_content->output_audio_frame_rate ()); + _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (b, _audio_position.get ()))); + _audio_position = _audio_position.get() + b->frames (); } + */ } void |
