Handle multiple audio streams in a single piece of content

in a similar way to the V1 patch.

1 files changed, 236 insertions, 0 deletions

diff --git a/src/lib/audio_decoder_stream.cc b/src/lib/audio_decoder_stream.cc
new file mode 100644
index 000000000..b9a8f2657
--- /dev/null
+++ b/src/lib/audio_decoder_stream.cc
@@ -0,0 +1,236 @@
+/*
+    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
+    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,
+    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.
+
+*/
+
+#include "audio_decoder_stream.h"
+#include "audio_buffers.h"
+#include "audio_processor.h"
+#include "audio_decoder.h"
+#include "resampler.h"
+#include "util.h"
+#include <iostream>
+
+#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;
+
+AudioDecoderStream::AudioDecoderStream (shared_ptr<const AudioContent> content, AudioStreamPtr stream, AudioDecoder* decoder)
+	: _content (content)
+	, _stream (stream)
+	, _decoder (decoder)
+{
+	if (content->resampled_audio_frame_rate() != _stream->frame_rate()) {
+		_resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_audio_frame_rate(), _stream->channels ()));
+	}
+
+	reset_decoded ();
+}
+
+void
+AudioDecoderStream::reset_decoded ()
+{
+	_decoded = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_stream->channels(), 0)), 0);
+}
+
+ContentAudio
+AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
+{
+	shared_ptr<ContentAudio> dec;
+
+	Frame const end = frame + length - 1;
+		
+	if (frame < _decoded.frame || end > (_decoded.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, _content->resampled_audio_frame_rate()), accurate);
+	}
+
+	/* Offset of the data that we want from the start of _decoded.audio
+	   (to be set up shortly)
+	*/
+	Frame 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 until we have enough data, or the subclass does not want to give us any more */
+		while (
+			(_decoded.frame > frame || (_decoded.frame + _decoded.audio->frames()) < end) &&
+			!_decoder->pass (Decoder::PASS_REASON_AUDIO)
+			)
+		{}
+		
+		decoded_offset = frame - _decoded.frame;
+	} else {
+		while (
+			_decoded.audio->frames() < length &&
+			!_decoder->pass (Decoder::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 starting from `frame'.  This could be -ve
+	   if pass() returned true before we got enough data.
+	*/
+	Frame const available = _decoded.audio->frames() - decoded_offset;
+
+	/* We will return either that, or the requested amount, whichever is smaller */
+	Frame const to_return = max ((Frame) 0, min (available, length));
+
+	/* Copy our data to the output */
+	shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded.audio->channels(), to_return));
+	out->copy_from (_decoded.audio.get(), to_return, decoded_offset, 0);
+
+	Frame const remaining = max ((Frame) 0, available - to_return);
+
+	/* Clean up decoded; first, move the data after what we just returned to the start of the buffer */
+	_decoded.audio->move (decoded_offset + to_return, 0, remaining);
+	/* And set up the number of frames we have left */
+	_decoded.audio->set_frames (remaining);
+	/* Also bump where those frames are in terms of the content */
+	_decoded.frame += decoded_offset + to_return;
+
+	return ContentAudio (out, frame);
+}
+
+/** 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 _position.  The
+ *  time is ignored once this has been done.
+ */
+void
+AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
+{
+	if (_resampler) {
+		data = _resampler->run (data);
+	}
+
+	Frame const frame_rate = _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 ();
+		Frame 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 */
+			Frame const to_discard = min (-delta_frames, static_cast<Frame> (data->frames()));
+			Frame 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 (!_position) {
+		_position = time.frames (frame_rate);
+	}
+
+	DCPOMATIC_ASSERT (_position.get() >= (_decoded.frame + _decoded.audio->frames()));
+
+	add (data);
+}
+
+void
+AudioDecoderStream::add (shared_ptr<const AudioBuffers> data)
+{
+	if (!_position) {
+		/* This should only happen when there is a seek followed by a flush, but
+		   we need to cope with it.
+		*/
+		return;
+	}
+	
+	/* Resize _decoded to fit the new data */
+	int new_size = 0;
+	if (_decoded.audio->frames() == 0) {
+		/* There's nothing in there, so just store the new data */
+		new_size = data->frames ();
+		_decoded.frame = _position.get ();
+	} else {
+		/* Otherwise we need to extend _decoded to include the new stuff */
+		new_size = _position.get() + data->frames() - _decoded.frame;
+	}
+	
+	_decoded.audio->ensure_size (new_size);
+	_decoded.audio->set_frames (new_size);
+
+	/* Copy new data in */
+	_decoded.audio->copy_from (data.get(), data->frames(), 0, _position.get() - _decoded.frame);
+	_position = _position.get() + data->frames ();
+
+	/* Limit the amount of data we keep in case nobody is asking for it */
+	int const max_frames = _content->resampled_audio_frame_rate () * 10;
+	if (_decoded.audio->frames() > max_frames) {
+		int const to_remove = _decoded.audio->frames() - max_frames;
+		_decoded.frame += to_remove;
+		_decoded.audio->move (to_remove, 0, max_frames);
+		_decoded.audio->set_frames (max_frames);
+	}
+}
+
+void
+AudioDecoderStream::flush ()
+{
+	if (!_resampler) {
+		return;
+	}
+
+	shared_ptr<const AudioBuffers> b = _resampler->flush ();
+	if (b) {
+		add (b);
+	}
+}
+
+void
+AudioDecoderStream::seek (ContentTime t, bool accurate)
+{
+	_position.reset ();
+	reset_decoded ();
+	if (accurate) {
+		_seek_reference = t;
+	}
+}