#include <iostream>
#include <boost/thread.hpp>
#include <boost/filesystem.hpp>
+#include <boost/lexical_cast.hpp>
#include <sndfile.h>
#include <openjpeg.h>
#include "j2k_wav_encoder.h"
J2KWAVEncoder::J2KWAVEncoder (shared_ptr<const FilmState> s, shared_ptr<const Options> o, Log* l)
: Encoder (s, o, l)
+#ifdef HAVE_SWRESAMPLE
+ , _swr_context (0)
+#endif
, _deinterleave_buffer_size (8192)
, _deinterleave_buffer (0)
, _process_end (false)
/* 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,
int remote_backoff = 0;
while (1) {
+
+ TIMING ("encoder thread %1 sleeps", pthread_self ());
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", pthread_self(), _queue.size());
boost::shared_ptr<DCPVideoFrame> vf = _queue.front ();
_queue.pop_front ();
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", pthread_self(), vf->frame());
encoded = vf->encode_locally ();
+ TIMING ("encoder thread %1 finishes local encode of %2", pthread_self(), 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 ()));
}
}
}
void
-J2KWAVEncoder::process_begin ()
+J2KWAVEncoder::process_begin (int64_t audio_channel_layout, AVSampleFormat audio_sample_format)
{
+ 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,
+ _fs->target_sample_rate(),
+ audio_channel_layout,
+ audio_sample_format,
+ _fs->audio_sample_rate,
+ 0, 0
+ );
+
+ swr_init (_swr_context);
+#else
+ throw EncodeError ("Cannot resample audio as libswresample is not present");
+#endif
+ } else {
+#ifdef HAVE_SWRESAMPLE
+ _swr_context = 0;
+#endif
+ }
+
for (int i = 0; i < Config::instance()->num_local_encoding_threads (); ++i) {
_worker_threads.push_back (new boost::thread (boost::bind (&J2KWAVEncoder::encoder_thread, this, (ServerDescription *) 0)));
}
{
boost::mutex::scoped_lock lock (_worker_mutex);
+ _log->log ("Clearing queue of " + lexical_cast<string> (_queue.size ()));
+
/* Keep waking workers until the queue is empty */
while (!_queue.empty ()) {
+ _log->log ("Waking with " + lexical_cast<string> (_queue.size ()));
_worker_condition.notify_all ();
_worker_condition.wait (lock);
}
terminate_worker_threads ();
+ _log->log ("Mopping up " + lexical_cast<string> (_queue.size()));
+
/* The following sequence of events can occur in the above code:
1. a remote worker takes the last image off the queue
2. the loop above terminates
*/
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) {
+
+ while (1) {
+ uint8_t buffer[256 * _fs->bytes_per_sample() * _fs->audio_channels];
+ uint8_t* out[2] = {
+ buffer,
+ 0
+ };
+
+ int const frames = swr_convert (_swr_context, out, 256, 0, 0);
+
+ if (frames < 0) {
+ throw EncodeError ("could not run sample-rate converter");
+ }
+
+ if (frames == 0) {
+ break;
+ }
+
+ write_audio (buffer, frames * _fs->bytes_per_sample() * _fs->audio_channels);
}
+
+ swr_free (&_swr_context);
}
+#endif
close_sound_files ();
}
void
-J2KWAVEncoder::process_audio (uint8_t* data, int data_size)
+J2KWAVEncoder::process_audio (uint8_t* data, int size)
{
- /* Size of a sample in bytes */
- int const sample_size = 2;
+ /* This is a buffer we might use if we are sample-rate converting;
+ it will need freeing if so.
+ */
+ uint8_t* out_buffer = 0;
- /* 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.
+ /* Maybe sample-rate convert */
+#if HAVE_SWRESAMPLE
+ if (_swr_context) {
+
+ uint8_t const * in[2] = {
+ data,
+ 0
+ };
+
+ /* Here's samples per channel */
+ int const samples = size / _fs->bytes_per_sample();
+
+ /* And here's frames (where 1 frame is a collection of samples, 1 for each channel,
+ so for 5.1 a frame would be 6 samples)
+ */
+ 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 * _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];
+
+ uint8_t* out[2] = {
+ out_buffer,
+ 0
+ };
+
+ /* Resample audio */
+ 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");
+ }
+
+ /* And point our variables at the resampled audio */
+ data = out_buffer;
+ size = out_frames * _fs->audio_channels * _fs->bytes_per_sample();
+ }
+#endif
+
+ write_audio (data, size);
+
+ /* Delete the sample-rate conversion buffer, if it exists */
+ delete[] out_buffer;
+}
+
+void
+J2KWAVEncoder::write_audio (uint8_t* data, int 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 ... */
/* Number of bytes left to read this time */
- int remaining = data_size;
+ int remaining = size;
/* Our position in the output buffers, in bytes */
int position = 0;
while (remaining > 0) {
/* 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 DecodeError ("unknown audio sample format");
+ throw EncodeError ("unknown audio sample format");
}
}
remaining -= this_time * _fs->audio_channels;
}
}
+
projects / dcpomatic.git / blobdiff