using std::cout;
-using std::map;
-using std::pair;
using std::shared_ptr;
using std::make_shared;
using boost::optional;
/** @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 time_already_delayed)
+AudioDecoder::emit(shared_ptr<const Film> film, AudioStreamPtr stream, shared_ptr<const AudioBuffers> data, ContentTime time, bool flushing)
{
if (ignore ()) {
return;
}
- /* Amount of error we will tolerate on audio timestamps; see comment below.
- * We'll use 1 24fps video frame at 48kHz as this seems to be roughly how
- * ffplay does it.
- */
- static Frame const slack_frames = 48000 / 24;
-
int const resampled_rate = _content->resampled_frame_rate(film);
- if (!time_already_delayed) {
+ if (!flushing) {
time += ContentTime::from_seconds (_content->delay() / 1000.0);
}
- auto reset = false;
- if (_positions[stream] == 0) {
- /* This is the first data we have received since initialisation or seek. 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.
- */
- if (_content->delay() > 0) {
- /* Insert silence to give the delay */
- silence (_content->delay ());
- }
- reset = true;
- } else if (std::abs(_positions[stream] - time.frames_round(resampled_rate)) > slack_frames) {
- reset = true;
+ /* 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.
+ */
+
+ 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);
+
+ if (need_reset) {
LOG_GENERAL (
"Reset audio position: was %1, new data at %2, slack: %3 frames",
_positions[stream],
);
}
- if (reset) {
+ if (first_since_seek || need_reset) {
_positions[stream] = time.frames_round (resampled_rate);
}
+ if (first_since_seek && _content->delay() > 0) {
+ silence (stream, _content->delay());
+ }
+
shared_ptr<Resampler> resampler;
auto i = _resamplers.find(stream);
if (i != _resamplers.end()) {
}
}
- if (resampler) {
- auto ro = resampler->run (data);
- if (ro->frames() == 0) {
+ 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;
}
- data = ro;
}
Data(stream, ContentAudio (data, _positions[stream]));
if (_content->delay() < 0) {
/* Finish off with the gap caused by the delay */
- silence (-_content->delay ());
+ for (auto stream: _content->streams()) {
+ silence (stream, -_content->delay());
+ }
}
}
void
-AudioDecoder::silence (int milliseconds)
+AudioDecoder::silence (AudioStreamPtr stream, int milliseconds)
{
- for (auto i: _content->streams()) {
- int const samples = ContentTime::from_seconds(milliseconds / 1000.0).frames_round(i->frame_rate());
- auto silence = make_shared<AudioBuffers>(i->channels(), samples);
- silence->make_silent ();
- Data (i, ContentAudio (silence, _positions[i]));
- }
+ 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