}
void
-J2KWAVEncoder::process_video (shared_ptr<Image> yuv, int frame)
+J2KWAVEncoder::process_video (shared_ptr<Image> yuv, int frame, shared_ptr<Subtitle> sub)
{
boost::mutex::scoped_lock lock (_worker_mutex);
/* Wait until the queue has gone down a bit */
while (_queue.size() >= _worker_threads.size() * 2 && !_process_end) {
+ TIMING ("decoder sleeps with queue of %1", _queue.size());
_worker_condition.wait (lock);
+ TIMING ("decoder wakes with queue of %1", _queue.size());
}
if (_process_end) {
/* Only do the processing if we don't already have a file for this frame */
if (!boost::filesystem::exists (_opt->frame_out_path (frame, false))) {
pair<string, string> const s = Filter::ffmpeg_strings (_fs->filters);
+ TIMING ("adding to queue of %1", _queue.size ());
_queue.push_back (boost::shared_ptr<DCPVideoFrame> (
new DCPVideoFrame (
- yuv, _opt->out_size, _opt->padding, _fs->scaler, frame, _fs->frames_per_second, s.second,
+ yuv, sub, _opt->out_size, _opt->padding, _fs->scaler, frame, _fs->frames_per_second, s.second,
Config::instance()->colour_lut_index (), Config::instance()->j2k_bandwidth (),
_log
)
int remote_backoff = 0;
while (1) {
+
+ TIMING ("encoder thread %1 sleeps", boost::this_thread::get_id());
boost::mutex::scoped_lock lock (_worker_mutex);
while (_queue.empty () && !_process_end) {
_worker_condition.wait (lock);
return;
}
+ TIMING ("encoder thread %1 wakes with queue of %2", boost::this_thread::get_id(), _queue.size());
boost::shared_ptr<DCPVideoFrame> vf = _queue.front ();
+ _log->log (String::compose ("Encoder thread %1 pops frame %2 from queue", boost::this_thread::get_id(), vf->frame()));
_queue.pop_front ();
lock.unlock ();
encoded = vf->encode_remotely (server);
if (remote_backoff > 0) {
- stringstream s;
- s << server->host_name() << " was lost, but now she is found; removing backoff";
- _log->log (s.str ());
+ _log->log (String::compose ("%1 was lost, but now she is found; removing backoff", server->host_name ()));
}
/* This job succeeded, so remove any backoff */
/* back off more */
remote_backoff += 10;
}
- stringstream s;
- s << "Remote encode of " << vf->frame() << " on " << server->host_name() << " failed (" << e.what() << "); thread sleeping for " << remote_backoff << "s.";
- _log->log (s.str ());
+ _log->log (
+ String::compose (
+ "Remote encode of %1 on %2 failed (%3); thread sleeping for %4s",
+ vf->frame(), server->host_name(), e.what(), remote_backoff)
+ );
}
} else {
try {
+ TIMING ("encoder thread %1 begins local encode of %2", boost::this_thread::get_id(), vf->frame());
encoded = vf->encode_locally ();
+ TIMING ("encoder thread %1 finishes local encode of %2", boost::this_thread::get_id(), vf->frame());
} catch (std::exception& e) {
- stringstream s;
- s << "Local encode failed " << e.what() << ".";
- _log->log (s.str ());
+ _log->log (String::compose ("Local encode failed (%1)", e.what ()));
}
}
frame_done (vf->frame ());
} else {
lock.lock ();
+ _log->log (String::compose ("Encoder thread %1 pushes frame %2 back onto queue after failure", boost::this_thread::get_id(), vf->frame()));
_queue.push_front (vf);
lock.unlock ();
}
void
J2KWAVEncoder::process_begin (int64_t audio_channel_layout, AVSampleFormat audio_sample_format)
{
- if ((_fs->audio_sample_rate != dcp_audio_sample_rate (_fs->audio_sample_rate)) || (rint (_fs->frames_per_second) != _fs->frames_per_second)) {
-#ifdef HAVE_SWRESAMPLE
+ if (_fs->audio_sample_rate != _fs->target_sample_rate ()) {
+#ifdef HAVE_SWRESAMPLE
+
+ stringstream s;
+ s << "Will resample audio from " << _fs->audio_sample_rate << " to " << _fs->target_sample_rate();
+ _log->log (s.str ());
+
_swr_context = swr_alloc_set_opts (
0,
audio_channel_layout,
audio_sample_format,
- target_sample_rate(),
+ _fs->target_sample_rate(),
audio_channel_layout,
audio_sample_format,
_fs->audio_sample_rate,
*/
for (list<shared_ptr<DCPVideoFrame> >::iterator i = _queue.begin(); i != _queue.end(); ++i) {
- stringstream s;
- s << "Encode left-over frame " << (*i)->frame();
- _log->log (s.str ());
+ _log->log (String::compose ("Encode left-over frame %1", (*i)->frame ()));
try {
shared_ptr<EncodedData> e = (*i)->encode_locally ();
e->write (_opt, (*i)->frame ());
frame_done ((*i)->frame ());
} catch (std::exception& e) {
- stringstream s;
- s << "Local encode failed " << e.what() << ".";
- _log->log (s.str ());
+ _log->log (String::compose ("Local encode failed (%1)", e.what ()));
}
}
#if HAVE_SWRESAMPLE
if (_swr_context) {
- int mop = 0;
while (1) {
uint8_t buffer[256 * _fs->bytes_per_sample() * _fs->audio_channels];
- uint8_t* out[1] = {
- buffer
+ uint8_t* out[2] = {
+ buffer,
+ 0
};
int const frames = swr_convert (_swr_context, out, 256, 0, 0);
break;
}
- mop += frames;
- write_audio (buffer, frames);
+ write_audio (buffer, frames * _fs->bytes_per_sample() * _fs->audio_channels);
}
swr_free (&_swr_context);
int const frames = samples / _fs->audio_channels;
/* Compute the resampled frame count and add 32 for luck */
- int const out_buffer_size_frames = ceil (frames * target_sample_rate() / _fs->audio_sample_rate) + 32;
+ int const out_buffer_size_frames = ceil (frames * _fs->target_sample_rate() / _fs->audio_sample_rate) + 32;
int const out_buffer_size_bytes = out_buffer_size_frames * _fs->audio_channels * _fs->bytes_per_sample();
out_buffer = new uint8_t[out_buffer_size_bytes];
};
/* Resample audio */
- int out_frames = swr_convert (_swr_context, out, out_buffer_size_frames, in, size);
+ int out_frames = swr_convert (_swr_context, out, out_buffer_size_frames, in, frames);
if (out_frames < 0) {
throw EncodeError ("could not run sample-rate converter");
}
void
J2KWAVEncoder::write_audio (uint8_t* data, int size)
{
- /* Size of a sample in bytes */
- int const sample_size = 2;
-
- /* XXX: we are assuming that sample_size is right, the _deinterleave_buffer_size is a multiple
- of the sample size and that data_size is a multiple of _fs->audio_channels * sample_size.
+ /* XXX: we are assuming that the _deinterleave_buffer_size is a multiple
+ of the sample size and that size is a multiple of _fs->audio_channels * sample_size.
*/
+
+ assert ((size % (_fs->audio_channels * _fs->bytes_per_sample())) == 0);
+ assert ((_deinterleave_buffer_size % _fs->bytes_per_sample()) == 0);
/* XXX: this code is very tricksy and it must be possible to make it simpler ... */
/* How many bytes of the deinterleaved data to do this time */
int this_time = min (remaining / _fs->audio_channels, _deinterleave_buffer_size);
for (int i = 0; i < _fs->audio_channels; ++i) {
- for (int j = 0; j < this_time; j += sample_size) {
- for (int k = 0; k < sample_size; ++k) {
+ for (int j = 0; j < this_time; j += _fs->bytes_per_sample()) {
+ for (int k = 0; k < _fs->bytes_per_sample(); ++k) {
int const to = j + k;
- int const from = position + (i * sample_size) + (j * _fs->audio_channels) + k;
+ int const from = position + (i * _fs->bytes_per_sample()) + (j * _fs->audio_channels) + k;
_deinterleave_buffer[to] = data[from];
}
}
switch (_fs->audio_sample_format) {
case AV_SAMPLE_FMT_S16:
- sf_write_short (_sound_files[i], (const short *) _deinterleave_buffer, this_time / sample_size);
+ sf_write_short (_sound_files[i], (const short *) _deinterleave_buffer, this_time / _fs->bytes_per_sample());
break;
default:
throw EncodeError ("unknown audio sample format");
}
}
-int
-J2KWAVEncoder::target_sample_rate () const
-{
- double t = dcp_audio_sample_rate (_fs->audio_sample_rate);
- if (rint (_fs->frames_per_second) != _fs->frames_per_second) {
- if (_fs->frames_per_second == 23.976) {
- /* 24fps drop-frame ie 24 * 1000 / 1001 frames per second;
- hence we need to resample the audio to dcp_audio_sample_rate * 1000 / 1001
- so that when we play it back at dcp_audio_sample_rate it is sped up
- by the same amount that the video is
- */
- t *= double(1000) / 1001;
- } else {
- throw EncodeError ("unknown fractional frame rate");
- }
- }
-
- return rint (t);
-}
projects / dcpomatic.git / blobdiff