X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=2944357bac7fe0f279bb8da163915c1e730f65e9;hb=aeb835a18c8df347e0ed68fb24631b320abeb611;hp=bbd4ced6c1a2c114f337729f8fe609f71e1eb353;hpb=f45cbb73068321d857249aaa21f770b87493c7c2;p=dcpomatic.git diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc index bbd4ced6c..2944357ba 100644 --- a/src/lib/audio_decoder.cc +++ b/src/lib/audio_decoder.cc @@ -1,5 +1,5 @@ /* - Copyright (C) 2012 Carl Hetherington + Copyright (C) 2012-2015 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 @@ -19,149 +19,85 @@ #include "audio_decoder.h" #include "audio_buffers.h" -#include "exceptions.h" -#include "log.h" +#include "audio_decoder_stream.h" +#include "audio_content.h" +#include +#include #include "i18n.h" -using std::stringstream; -using std::list; -using std::pair; using std::cout; -using boost::optional; +using std::map; using boost::shared_ptr; -AudioDecoder::AudioDecoder (shared_ptr f, shared_ptr c) - : Decoder (f) - , _next_audio (0) - , _audio_content (c) +AudioDecoder::AudioDecoder (shared_ptr content, bool fast) + : _audio_content (content) + , _ignore_audio (false) + , _fast (fast) { - if (_audio_content->content_audio_frame_rate() != _audio_content->output_audio_frame_rate()) { - - shared_ptr film = _film.lock (); - assert (film); - - stringstream s; - s << String::compose ( - "Will resample audio from %1 to %2", - _audio_content->content_audio_frame_rate(), _audio_content->output_audio_frame_rate() - ); - - film->log()->log (s.str ()); - - /* We will be using planar float data when we call the - resampler. As far as I can see, the audio channel - layout is not necessary for our purposes; it seems - only to be used get the number of channels and - decide if rematrixing is needed. It won't be, since - input and output layouts are the same. - */ - - _swr_context = swr_alloc_set_opts ( - 0, - av_get_default_channel_layout (MAX_AUDIO_CHANNELS), - AV_SAMPLE_FMT_FLTP, - _audio_content->output_audio_frame_rate(), - av_get_default_channel_layout (MAX_AUDIO_CHANNELS), - AV_SAMPLE_FMT_FLTP, - _audio_content->content_audio_frame_rate(), - 0, 0 - ); - - swr_init (_swr_context); - } else { - _swr_context = 0; + BOOST_FOREACH (AudioStreamPtr i, content->audio_streams ()) { + _streams[i] = shared_ptr (new AudioDecoderStream (_audio_content, i, this)); } } -AudioDecoder::~AudioDecoder () +ContentAudio +AudioDecoder::get_audio (AudioStreamPtr stream, Frame frame, Frame length, bool accurate) { - if (_swr_context) { - swr_free (&_swr_context); - } + return _streams[stream]->get (frame, length, accurate); } - -#if 0 void -AudioDecoder::process_end () +AudioDecoder::audio (AudioStreamPtr stream, shared_ptr data, ContentTime time) { - if (_swr_context) { + if (_ignore_audio) { + return; + } + + if (_streams.find (stream) == _streams.end ()) { - shared_ptr film = _film.lock (); - assert (film); - - shared_ptr out (new AudioBuffers (film->audio_mapping().dcp_channels(), 256)); - - while (1) { - int const frames = swr_convert (_swr_context, (uint8_t **) out->data(), 256, 0, 0); + /* This method can be called with an unknown stream during the following sequence: + - Add KDM to some DCP content. + - Content gets re-examined. + - SingleStreamAudioContent::take_from_audio_examiner creates a new stream. + - Some content property change signal is delivered so Player::Changed is emitted. + - Film viewer to re-gets the frame. + - Player calls DCPDecoder pass which calls this method on the new stream. - if (frames < 0) { - throw EncodeError (_("could not run sample-rate converter")); - } + At this point the AudioDecoder does not know about the new stream. - if (frames == 0) { - break; - } + Then + - Some other property change signal is delivered which marks the player's pieces invalid. + - Film viewer re-gets again. + - Everything is OK. - out->set_frames (frames); - _writer->write (out); - } + In this situation it is fine for us to silently drop the audio. + */ + return; } + + _streams[stream]->audio (data, time); } -#endif void -AudioDecoder::audio (shared_ptr data, Time time) +AudioDecoder::flush () { - /* Maybe resample */ - if (_swr_context) { - - /* Compute the resampled frames count and add 32 for luck */ - int const max_resampled_frames = ceil ( - (int64_t) data->frames() * _audio_content->output_audio_frame_rate() / _audio_content->content_audio_frame_rate() - ) + 32; - - shared_ptr resampled (new AudioBuffers (data->channels(), max_resampled_frames)); - - /* Resample audio */ - int const resampled_frames = swr_convert ( - _swr_context, (uint8_t **) resampled->data(), max_resampled_frames, (uint8_t const **) data->data(), data->frames() - ); - - if (resampled_frames < 0) { - throw EncodeError (_("could not run sample-rate converter")); - } - - resampled->set_frames (resampled_frames); - - /* And point our variables at the resampled audio */ - data = resampled; + for (map >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) { + i->second->flush (); } +} - shared_ptr film = _film.lock (); - assert (film); - - /* Remap channels */ - shared_ptr dcp_mapped (new AudioBuffers (film->dcp_audio_channels(), data->frames())); - dcp_mapped->make_silent (); - list > map = _audio_content->audio_mapping().content_to_dcp (); - for (list >::iterator i = map.begin(); i != map.end(); ++i) { - dcp_mapped->accumulate_channel (data.get(), i->first, i->second); +void +AudioDecoder::seek (ContentTime t, bool accurate) +{ + for (map >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) { + i->second->seek (t, accurate); } - - Audio (dcp_mapped, time); - cout << "bumping n.a. by " << data->frames() << " ie " << film->audio_frames_to_time(data->frames()) << "\n"; - _next_audio = time + film->audio_frames_to_time (data->frames()); } -bool -AudioDecoder::audio_done () const +/** Set this player never to produce any audio data */ +void +AudioDecoder::set_ignore_audio () { - shared_ptr film = _film.lock (); - assert (film); - - return (_audio_content->length() - _next_audio) < film->audio_frames_to_time (1); + _ignore_audio = true; } -