#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"
#include "server.h"
#include "filter.h"
#include "log.h"
+#include "cross.h"
using namespace std;
using namespace boost;
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)
*/
for (int i = 0; i < _fs->audio_channels; ++i) {
SF_INFO sf_info;
- sf_info.samplerate = _fs->audio_sample_rate;
+ sf_info.samplerate = dcp_audio_sample_rate (_fs->audio_sample_rate);
/* We write mono files */
sf_info.channels = 1;
sf_info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_24;
));
_worker_condition.notify_all ();
+ } else {
+ frame_skipped ();
}
}
void
-J2KWAVEncoder::encoder_thread (Server* server)
+J2KWAVEncoder::encoder_thread (ServerDescription* server)
{
/* Number of seconds that we currently wait between attempts
to connect to the server; not relevant for localhost
remote_backoff += 10;
}
stringstream s;
- s << "Remote encode on " << server->host_name() << " failed (" << e.what() << "); thread sleeping for " << remote_backoff << "s.";
+ s << "Remote encode of " << vf->frame() << " on " << server->host_name() << " failed (" << e.what() << "); thread sleeping for " << remote_backoff << "s.";
_log->log (s.str ());
}
if (encoded) {
encoded->write (_opt, vf->frame ());
- frame_done ();
+ frame_done (vf->frame ());
} else {
lock.lock ();
_queue.push_front (vf);
}
if (remote_backoff > 0) {
- sleep (remote_backoff);
+ dvdomatic_sleep (remote_backoff);
}
lock.lock ();
}
void
-J2KWAVEncoder::process_begin ()
+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
+
+ stringstream s;
+ s << "Will resample audio from " << _fs->audio_sample_rate << " to " << target_sample_rate();
+ _log->log (s.str ());
+
+ _swr_context = swr_alloc_set_opts (
+ 0,
+ audio_channel_layout,
+ audio_sample_format,
+ 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, (Server *) 0)));
+ _worker_threads.push_back (new boost::thread (boost::bind (&J2KWAVEncoder::encoder_thread, this, (ServerDescription *) 0)));
}
- vector<Server*> servers = Config::instance()->servers ();
+ vector<ServerDescription*> servers = Config::instance()->servers ();
- for (vector<Server*>::iterator i = servers.begin(); i != servers.end(); ++i) {
+ for (vector<ServerDescription*>::iterator i = servers.begin(); i != servers.end(); ++i) {
for (int j = 0; j < (*i)->threads (); ++j) {
_worker_threads.push_back (new boost::thread (boost::bind (&J2KWAVEncoder::encoder_thread, this, *i)));
}
{
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);
}
lock.unlock ();
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
+ 3. the remote worker fails to encode the image and puts it back on the queue
+ 4. the remote worker is then terminated by terminate_worker_threads
+
+ So just mop up anything left in the queue here.
+ */
+
+ 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 ());
+ 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 ());
+ }
+ }
+
+#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
+ };
+
+ 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;
+ }
+
+ mop += frames;
+ write_audio (buffer, frames);
+ }
+
+ swr_free (&_swr_context);
+ }
+#endif
+
close_sound_files ();
/* Rename .wav.tmp files to .wav */
}
void
-J2KWAVEncoder::process_audio (uint8_t* data, int data_size)
+J2KWAVEncoder::process_audio (uint8_t* data, int size)
+{
+ /* This is a buffer we might use if we are sample-rate converting;
+ it will need freeing if so.
+ */
+ uint8_t* out_buffer = 0;
+
+ /* 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 * 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, size);
+ 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)
{
/* 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: 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) {
sf_write_short (_sound_files[i], (const short *) _deinterleave_buffer, this_time / sample_size);
break;
default:
- throw DecodeError ("unknown audio sample format");
+ throw EncodeError ("unknown audio sample format");
}
}
remaining -= this_time * _fs->audio_channels;
}
}
+
+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