/*
- Copyright (C) 2012-2014 Carl Hetherington <cth@carlh.net>
+ Copyright (C) 2012-2016 Carl Hetherington <cth@carlh.net>
- This program is free software; you can redistribute it and/or modify
+ This file is part of DCP-o-matic.
+
+ DCP-o-matic is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
- This program is distributed in the hope that it will be useful,
+ DCP-o-matic is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
- along with this program; if not, write to the Free Software
- Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ along with DCP-o-matic. If not, see <http://www.gnu.org/licenses/>.
*/
#include "audio_decoder.h"
#include "audio_buffers.h"
-#include "audio_processor.h"
-#include "resampler.h"
-#include "util.h"
+#include "audio_decoder_stream.h"
+#include "audio_content.h"
+#include <boost/foreach.hpp>
+#include <iostream>
#include "i18n.h"
-using std::list;
-using std::pair;
using std::cout;
-using std::min;
-using std::max;
-using boost::optional;
+using std::map;
using boost::shared_ptr;
-AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> content)
- : _audio_content (content)
+AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr<const AudioContent> content, shared_ptr<Log> log)
+ : _ignore (false)
{
- 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 ());
+ BOOST_FOREACH (AudioStreamPtr i, content->streams ()) {
+ _streams[i] = shared_ptr<AudioDecoderStream> (new AudioDecoderStream (content, i, parent, log));
}
-
- reset_decoded_audio ();
}
-void
-AudioDecoder::reset_decoded_audio ()
+ContentAudio
+AudioDecoder::get (AudioStreamPtr stream, Frame frame, Frame length, bool accurate)
{
- _decoded_audio = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_audio_content->processed_audio_channels(), 0)), 0);
+ return _streams[stream]->get (frame, length, accurate);
}
-shared_ptr<ContentAudio>
-AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate)
+void
+AudioDecoder::give (AudioStreamPtr stream, shared_ptr<const AudioBuffers> data, ContentTime time)
{
- 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);
+ if (_ignore) {
+ return;
}
- /* 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 (!pass() && (_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end)) {}
- decoded_offset = frame - _decoded_audio.frame;
- } else {
- while (!pass() && _decoded_audio.audio->frames() < length) {}
- /* Use decoded_offset of 0, as we don't really care what frames we return */
- }
+ if (_streams.find (stream) == _streams.end ()) {
- /* 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;
+ /* 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.
- /* We will return either that, or the requested amount, whichever is smaller */
- AudioFrame const to_return = max ((AudioFrame) 0, min (available, length));
+ At this point the AudioDecoder does not know about the new stream.
- /* 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);
+ Then
+ - Some other property change signal is delivered which marks the player's pieces invalid.
+ - Film viewer re-gets again.
+ - Everything is OK.
- AudioFrame const remaining = max ((AudioFrame) 0, available - to_return);
+ In this situation it is fine for us to silently drop the audio.
+ */
- /* 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;
+ }
- return shared_ptr<ContentAudio> (new ContentAudio (out, frame));
+ _streams[stream]->audio (data, time);
}
-/** 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.
- */
void
-AudioDecoder::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
+AudioDecoder::flush ()
{
- if (_resampler) {
- data = _resampler->run (data);
+ for (map<AudioStreamPtr, shared_ptr<AudioDecoderStream> >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) {
+ i->second->flush ();
}
-
- 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<AudioBuffers> 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<AudioFrame> (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<AudioBuffers> 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<ContentTime> ();
- }
-
- if (!_audio_position) {
- _audio_position = time.frames (frame_rate);
- }
-
- assert (_audio_position.get() >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
-
- add (data);
}
void
-AudioDecoder::add (shared_ptr<const AudioBuffers> data)
+AudioDecoder::seek (ContentTime t, bool accurate)
{
- /* 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);
+ for (map<AudioStreamPtr, shared_ptr<AudioDecoderStream> >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) {
+ i->second->seek (t, accurate);
}
}
+/** Set this decoder never to produce any data */
void
-AudioDecoder::flush ()
+AudioDecoder::set_ignore ()
{
- if (!_resampler) {
- return;
- }
-
- shared_ptr<const AudioBuffers> b = _resampler->flush ();
- if (b) {
- add (b);
- }
+ _ignore = true;
}
void
-AudioDecoder::seek (ContentTime t, bool accurate)
+AudioDecoder::set_fast ()
{
- _audio_position.reset ();
- reset_decoded_audio ();
- if (accurate) {
- _seek_reference = t;
- }
- if (_processor) {
- _processor->flush ();
+ for (map<AudioStreamPtr, shared_ptr<AudioDecoderStream> >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) {
+ i->second->set_fast ();
}
}
projects / dcpomatic.git / blobdiff