#include <libdcp/picture_asset.h>
#include "encoder.h"
#include "util.h"
-#include "options.h"
#include "film.h"
#include "log.h"
#include "exceptions.h"
#include "format.h"
#include "cross.h"
#include "writer.h"
+#include "player.h"
+#include "audio_mapping.h"
#include "i18n.h"
using std::list;
using std::cout;
using std::make_pair;
-using namespace boost;
+using boost::shared_ptr;
+using boost::optional;
int const Encoder::_history_size = 25;
void
Encoder::process_begin ()
{
- if (_film->audio_stream() && _film->audio_stream()->sample_rate() != _film->target_audio_sample_rate()) {
+ if (_film->has_audio() && _film->audio_frame_rate() != _film->target_audio_sample_rate()) {
#ifdef HAVE_SWRESAMPLE
stringstream s;
- s << String::compose (N_("Will resample audio from %1 to %2"), _film->audio_stream()->sample_rate(), _film->target_audio_sample_rate());
+ s << String::compose (N_("Will resample audio from %1 to %2"), _film->audio_frame_rate(), _film->target_audio_sample_rate());
_film->log()->log (s.str ());
- /* We will be using planar float data when we call the resampler */
+ /* We will be using planar float data when we call the
+ resampler. As far as I can see, the audio channel
+ layout is not necessary for our purposes; it seems
+ only to be used get the number of channels and
+ decide if rematrixing is needed. It won't be, since
+ input and output layouts are the same.
+ */
+
_swr_context = swr_alloc_set_opts (
0,
- _film->audio_stream()->channel_layout(),
+ av_get_default_channel_layout (_film->audio_mapping().dcp_channels ()),
AV_SAMPLE_FMT_FLTP,
_film->target_audio_sample_rate(),
- _film->audio_stream()->channel_layout(),
+ av_get_default_channel_layout (_film->audio_mapping().dcp_channels ()),
AV_SAMPLE_FMT_FLTP,
- _film->audio_stream()->sample_rate(),
+ _film->audio_frame_rate(),
0, 0
);
Encoder::process_end ()
{
#if HAVE_SWRESAMPLE
- if (_film->audio_stream() && _film->audio_stream()->channels() && _swr_context) {
+ if (_film->has_audio() && _swr_context) {
- shared_ptr<AudioBuffers> out (new AudioBuffers (_film->audio_stream()->channels(), 256));
+ shared_ptr<AudioBuffers> out (new AudioBuffers (_film->audio_mapping().dcp_channels(), 256));
while (1) {
int const frames = swr_convert (_swr_context, (uint8_t **) out->data(), 256, 0, 0);
}
out->set_frames (frames);
- write_audio (out);
+ _writer->write (out);
}
swr_free (&_swr_context);
* or 0 if not known.
*/
float
-Encoder::current_frames_per_second () const
+Encoder::current_encoding_rate () const
{
boost::mutex::scoped_lock lock (_history_mutex);
if (int (_time_history.size()) < _history_size) {
}
void
-Encoder::process_video (shared_ptr<const Image> image, bool same, boost::shared_ptr<Subtitle> sub)
+Encoder::process_video (shared_ptr<const Image> image, bool same, shared_ptr<Subtitle> sub)
{
- FrameRateConversion frc (_film->source_frame_rate(), _film->dcp_frame_rate());
+ FrameRateConversion frc (_film->video_frame_rate(), _film->dcp_frame_rate());
if (frc.skip && (_video_frames_in % 2)) {
++_video_frames_in;
/* Queue this new frame for encoding */
pair<string, string> const s = Filter::ffmpeg_strings (_film->filters());
TIMING ("adding to queue of %1", _queue.size ());
- _queue.push_back (boost::shared_ptr<DCPVideoFrame> (
+ _queue.push_back (shared_ptr<DCPVideoFrame> (
new DCPVideoFrame (
image, sub, _film->format()->dcp_size(), _film->format()->dcp_padding (_film),
_film->subtitle_offset(), _film->subtitle_scale(),
if (_swr_context) {
/* Compute the resampled frames count and add 32 for luck */
- int const max_resampled_frames = ceil ((int64_t) data->frames() * _film->target_audio_sample_rate() / _film->audio_stream()->sample_rate()) + 32;
+ int const max_resampled_frames = ceil ((int64_t) data->frames() * _film->target_audio_sample_rate() / _film->audio_frame_rate()) + 32;
- shared_ptr<AudioBuffers> resampled (new AudioBuffers (_film->audio_stream()->channels(), max_resampled_frames));
+ shared_ptr<AudioBuffers> resampled (new AudioBuffers (_film->audio_mapping().dcp_channels(), max_resampled_frames));
/* Resample audio */
int const resampled_frames = swr_convert (
}
#endif
- write_audio (data);
+ _writer->write (data);
}
void
}
TIMING ("encoder thread %1 wakes with queue of %2", boost::this_thread::get_id(), _queue.size());
- boost::shared_ptr<DCPVideoFrame> vf = _queue.front ();
+ shared_ptr<DCPVideoFrame> vf = _queue.front ();
_film->log()->log (String::compose (N_("Encoder thread %1 pops frame %2 from queue"), boost::this_thread::get_id(), vf->frame()), Log::VERBOSE);
_queue.pop_front ();
}
if (remote_backoff > 0) {
- dvdomatic_sleep (remote_backoff);
+ dcpomatic_sleep (remote_backoff);
}
lock.lock ();
_condition.notify_all ();
}
}
-
-void
-Encoder::write_audio (shared_ptr<const AudioBuffers> data)
-{
- AudioMapping m (_film->audio_channels ());
- if (m.dcp_channels() != _film->audio_channels()) {
-
- /* Remap (currently just for mono -> 5.1) */
-
- shared_ptr<AudioBuffers> b (new AudioBuffers (m.dcp_channels(), data->frames ()));
- for (int i = 0; i < m.dcp_channels(); ++i) {
- optional<int> s = m.dcp_to_source (static_cast<libdcp::Channel> (i));
- if (!s) {
- b->make_silent (i);
- } else {
- memcpy (b->data()[i], data->data()[s.get()], data->frames() * sizeof(float));
- }
- }
-
- data = b;
- }
-
- _writer->write (data);
-}
projects / dcpomatic.git / blobdiff