/*
- Copyright (C) 2012-2014 Carl Hetherington <cth@carlh.net>
+ Copyright (C) 2012-2021 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 "audio_content.h"
+#include "dcpomatic_log.h"
+#include "log.h"
#include "resampler.h"
-#include "util.h"
+#include "compose.hpp"
+#include <iostream>
#include "i18n.h"
-using std::list;
-using std::pair;
+
using std::cout;
-using std::min;
-using std::max;
+using std::shared_ptr;
+using std::make_shared;
using boost::optional;
-using boost::shared_ptr;
+using namespace dcpomatic;
-AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> content)
- : _audio_content (content)
-{
- 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 ());
+AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr<const AudioContent> content, bool fast)
+ : DecoderPart (parent)
+ , _content (content)
+ , _fast (fast)
+{
+ /* Set up _positions so that we have one for each stream */
+ for (auto i: content->streams ()) {
+ _positions[i] = 0;
}
-
- reset_decoded_audio ();
}
-void
-AudioDecoder::reset_decoded_audio ()
-{
- _decoded_audio = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_audio_content->processed_audio_channels(), 0)), 0);
-}
-shared_ptr<ContentAudio>
-AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate)
+/** @param time_already_delayed true if the delay should not be added to time */
+void
+AudioDecoder::emit(shared_ptr<const Film> film, AudioStreamPtr stream, shared_ptr<const AudioBuffers> data, ContentTime time, bool flushing)
{
- 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 */
+ int const resampled_rate = _content->resampled_frame_rate(film);
+ if (!flushing) {
+ time += ContentTime::from_seconds (_content->delay() / 1000.0);
}
- /* 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.
+ /* Amount of error we will tolerate on audio timestamps; see comment below.
+ * We'll use 1 24fps video frame as this seems to be roughly how ffplay does it.
+ */
+ Frame const slack_frames = resampled_rate / 24;
+
+ /* first_since_seek is set to true if this is the first data we have
+ received since initialisation or seek. We'll set the position based
+ on the ContentTime that was given. After this first time we just
+ count samples unless the timestamp is more than slack_frames away
+ from where we think it should be. This is because ContentTimes seem
+ to be slightly unreliable from FFmpegDecoder (i.e. not sample
+ accurate), but we still need to obey them sometimes otherwise we get
+ sync problems such as #1833.
*/
- AudioFrame const available = _decoded_audio.audio->frames() - decoded_offset;
-
- /* We will return either that, or the requested amount, whichever is smaller */
- AudioFrame const to_return = max ((AudioFrame) 0, min (available, length));
- /* 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);
+ auto const first_since_seek = _positions[stream] == 0;
+ auto const need_reset = !first_since_seek && (std::abs(_positions[stream] - time.frames_round(resampled_rate)) > slack_frames);
- AudioFrame const remaining = max ((AudioFrame) 0, available - to_return);
-
- /* 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 shared_ptr<ContentAudio> (new ContentAudio (out, frame));
-}
+ if (need_reset) {
+ LOG_GENERAL (
+ "Reset audio position: was %1, new data at %2, slack: %3 frames",
+ _positions[stream],
+ time.frames_round(resampled_rate),
+ std::abs(_positions[stream] - time.frames_round(resampled_rate))
+ );
+ }
-/** 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)
-{
- if (_resampler) {
- data = _resampler->run (data);
+ if (first_since_seek || need_reset) {
+ _positions[stream] = time.frames_round (resampled_rate);
}
- if (_processor) {
- data = _processor->run (data);
+ if (first_since_seek && _content->delay() > 0) {
+ silence (stream, _content->delay());
}
- 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<Resampler> resampler;
+ auto i = _resamplers.find(stream);
+ if (i != _resamplers.end()) {
+ resampler = i->second;
+ } else {
+ if (stream->frame_rate() != resampled_rate) {
+ LOG_GENERAL (
+ "Creating new resampler from %1 to %2 with %3 channels",
+ stream->frame_rate(),
+ resampled_rate,
+ stream->channels()
+ );
+
+ resampler = make_shared<Resampler>(stream->frame_rate(), resampled_rate, stream->channels());
+ if (_fast) {
+ resampler->set_fast ();
}
- 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);
+ _resamplers[stream] = resampler;
}
- _seek_reference = optional<ContentTime> ();
}
- if (!_audio_position) {
- _audio_position = time.frames (frame_rate);
+ if (resampler && !flushing) {
+ /* It can be the the data here has a different number of channels than the stream
+ * it comes from (e.g. the files decoded by FFmpegDecoder sometimes have a random
+ * frame, often at the end, with more channels). Insert silence or discard channels
+ * here.
+ */
+ if (resampler->channels() != data->channels()) {
+ LOG_WARNING("Received audio data with an unexpected channel count of %1 instead of %2", data->channels(), resampler->channels());
+ auto data_copy = data->clone();
+ data_copy->set_channels(resampler->channels());
+ data = resampler->run(data_copy);
+ } else {
+ data = resampler->run(data);
+ }
+
+ if (data->frames() == 0) {
+ return;
+ }
}
- assert (_audio_position.get() >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
+ Data(stream, ContentAudio (data, _positions[stream]));
+ _positions[stream] += data->frames();
+}
+
+
+/** @return Time just after the last thing that was emitted from a given stream */
+ContentTime
+AudioDecoder::stream_position (shared_ptr<const Film> film, AudioStreamPtr stream) const
+{
+ auto i = _positions.find (stream);
+ DCPOMATIC_ASSERT (i != _positions.end ());
+ return ContentTime::from_frames (i->second, _content->resampled_frame_rate(film));
+}
+
+
+boost::optional<ContentTime>
+AudioDecoder::position (shared_ptr<const Film> film) const
+{
+ optional<ContentTime> p;
+ for (auto i: _positions) {
+ auto const ct = stream_position (film, i.first);
+ if (!p || ct < *p) {
+ p = ct;
+ }
+ }
- add (data);
+ return p;
}
+
void
-AudioDecoder::add (shared_ptr<const AudioBuffers> data)
+AudioDecoder::seek ()
{
- /* 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;
+ for (auto i: _resamplers) {
+ i.second->flush ();
+ i.second->reset ();
}
-
- _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 (auto& i: _positions) {
+ i.second = 0;
}
}
+
void
AudioDecoder::flush ()
{
- if (!_resampler) {
- return;
+ for (auto const& i: _resamplers) {
+ auto ro = i.second->flush ();
+ if (ro->frames() > 0) {
+ Data (i.first, ContentAudio (ro, _positions[i.first]));
+ _positions[i.first] += ro->frames();
+ }
}
- shared_ptr<const AudioBuffers> b = _resampler->flush ();
- if (b) {
- add (b);
+ if (_content->delay() < 0) {
+ /* Finish off with the gap caused by the delay */
+ for (auto stream: _content->streams()) {
+ silence (stream, -_content->delay());
+ }
}
}
+
void
-AudioDecoder::seek (ContentTime t, bool accurate)
+AudioDecoder::silence (AudioStreamPtr stream, int milliseconds)
{
- _audio_position.reset ();
- reset_decoded_audio ();
- if (accurate) {
- _seek_reference = t;
- }
- if (_processor) {
- _processor->flush ();
- }
+ int const samples = ContentTime::from_seconds(milliseconds / 1000.0).frames_round(stream->frame_rate());
+ auto silence = make_shared<AudioBuffers>(stream->channels(), samples);
+ silence->make_silent ();
+ Data (stream, ContentAudio(silence, _positions[stream]));
}
projects / dcpomatic.git / blobdiff