/*
- Copyright (C) 2012 Carl Hetherington <cth@carlh.net>
+ Copyright (C) 2012-2015 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
#include "audio_decoder.h"
#include "audio_buffers.h"
-#include "exceptions.h"
-#include "log.h"
+#include "audio_decoder_stream.h"
+#include "audio_content.h"
+#include <boost/foreach.hpp>
+#include <iostream>
#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<const Film> f, shared_ptr<const AudioContent> c)
- : Decoder (f)
- , _next_audio (0)
- , _audio_content (c)
+AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> 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<const Film> 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<AudioDecoderStream> (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<const AudioBuffers> data, ContentTime time)
{
- if (_swr_context) {
+ if (_ignore_audio) {
+ return;
+ }
+
+ if (_streams.find (stream) == _streams.end ()) {
- shared_ptr<const Film> film = _film.lock ();
- assert (film);
-
- shared_ptr<AudioBuffers> 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<const AudioBuffers> 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<AudioBuffers> 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<AudioStreamPtr, shared_ptr<AudioDecoderStream> >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) {
+ i->second->flush ();
}
+}
- shared_ptr<const Film> film = _film.lock ();
- assert (film);
-
- /* Remap channels */
- shared_ptr<AudioBuffers> dcp_mapped (new AudioBuffers (film->dcp_audio_channels(), data->frames()));
- dcp_mapped->make_silent ();
- list<pair<int, libdcp::Channel> > map = _audio_content->audio_mapping().content_to_dcp ();
- for (list<pair<int, libdcp::Channel> >::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<AudioStreamPtr, shared_ptr<AudioDecoderStream> >::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<const Film> film = _film.lock ();
- assert (film);
-
- return (_audio_content->length() - _next_audio) < film->audio_frames_to_time (1);
+ _ignore_audio = true;
}
-
projects / dcpomatic.git / blobdiff