X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=f6133947a0e4ad7e6ef57d38cd063102eb682411;hb=9922c1f2eaea674ba2ff6cce5f5853655fd8ad7a;hp=18f4b890d6b8878cc58eabc6a2aba5930eec6838;hpb=4388fff5376a6e5a6dc8d33e244a1245a728335c;p=dcpomatic.git diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc index 18f4b890d..f6133947a 100644 --- a/src/lib/audio_decoder.cc +++ b/src/lib/audio_decoder.cc @@ -1,5 +1,5 @@ /* - Copyright (C) 2012 Carl Hetherington + Copyright (C) 2012-2014 Carl Hetherington 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 @@ -19,39 +19,218 @@ #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 #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 film, shared_ptr content) - : Decoder (film) - , _audio_content (content) - , _audio_position (0) +AudioDecoder::AudioDecoder (shared_ptr 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 ())); + } + if (content->audio_processor ()) { + _processor = content->audio_processor()->clone (content->resampled_audio_frame_rate ()); + } + + reset_decoded_audio (); } void -AudioDecoder::audio (shared_ptr data, AudioContent::Frame frame) +AudioDecoder::reset_decoded_audio () { - Audio (data, frame); - _audio_position = frame + data->frames (); + _decoded_audio = ContentAudio (shared_ptr (new AudioBuffers (_audio_content->processed_audio_channels(), 0)), 0); } -/** This is a bit odd, but necessary when we have (e.g.) FFmpegDecoders with no audio. - * The player needs to know that there is no audio otherwise it will keep trying to - * pass() the decoder to get it to emit audio. +shared_ptr +AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate) +{ + shared_ptr 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->resampled_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 (PASS_REASON_AUDIO)) {} + decoded_offset = frame - _decoded_audio.frame; + } else { + while (_decoded_audio.audio->frames() < length && !pass (PASS_REASON_AUDIO)) {} + /* 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 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 (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. */ -bool -AudioDecoder::has_audio () const +void +AudioDecoder::audio (shared_ptr data, ContentTime time) +{ + if (_resampler) { + 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 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 (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 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 (); + } + + if (!_audio_position) { + _audio_position = time.frames (frame_rate); + } + + DCPOMATIC_ASSERT (_audio_position.get() >= (_decoded_audio.frame + _decoded_audio.audio->frames())); + + add (data); +} + +void +AudioDecoder::add (shared_ptr data) +{ + if (!_audio_position) { + /* This should only happen when there is a seek followed by a flush, but + we need to cope with it. + */ + return; + } + + /* 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; + } + + _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); + } +} + +void +AudioDecoder::flush () +{ + if (!_resampler) { + return; + } + + shared_ptr b = _resampler->flush (); + if (b) { + add (b); + } +} + +void +AudioDecoder::seek (ContentTime t, bool accurate) { - return _audio_content->audio_channels () > 0; + _audio_position.reset (); + reset_decoded_audio (); + if (accurate) { + _seek_reference = t; + } + if (_processor) { + _processor->flush (); + } }