+/*
+ 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;
+ }
+}
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