X-Git-Url: https://main.carlh.net/gitweb/?p=dcpomatic.git;a=blobdiff_plain;f=src%2Flib%2Fencoder.cc;h=c1d1041ae539f9cdab1f087291eb2d8d7104abbb;hp=cff9899acb6151cc19990f372c0a091c7d63968f;hb=f861018389acd9d277fe34d7621182b9b54f977f;hpb=7b2ec1dd69951649f2c912fcf90b22913b1f6c3a diff --git a/src/lib/encoder.cc b/src/lib/encoder.cc index cff9899ac..c1d1041ae 100644 --- a/src/lib/encoder.cc +++ b/src/lib/encoder.cc @@ -27,7 +27,6 @@ #include #include "encoder.h" #include "util.h" -#include "options.h" #include "film.h" #include "log.h" #include "exceptions.h" @@ -38,6 +37,8 @@ #include "format.h" #include "cross.h" #include "writer.h" +#include "player.h" +#include "audio_mapping.h" #include "i18n.h" @@ -48,7 +49,8 @@ using std::vector; 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; @@ -77,22 +79,29 @@ Encoder::~Encoder () 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 ); @@ -126,9 +135,9 @@ void Encoder::process_end () { #if HAVE_SWRESAMPLE - if (_film->audio_stream() && _film->audio_stream()->channels() && _swr_context) { + if (_film->has_audio() && _swr_context) { - shared_ptr out (new AudioBuffers (_film->audio_stream()->channels(), 256)); + shared_ptr 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); @@ -142,7 +151,7 @@ Encoder::process_end () } out->set_frames (frames); - write_audio (out); + _writer->write (out); } swr_free (&_swr_context); @@ -193,7 +202,7 @@ Encoder::process_end () * 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) { @@ -231,9 +240,9 @@ Encoder::frame_done () } void -Encoder::process_video (shared_ptr image, bool same, boost::shared_ptr sub) +Encoder::process_video (shared_ptr image, bool same, shared_ptr 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; @@ -269,7 +278,7 @@ Encoder::process_video (shared_ptr image, bool same, boost::shared_ /* Queue this new frame for encoding */ pair const s = Filter::ffmpeg_strings (_film->filters()); TIMING ("adding to queue of %1", _queue.size ()); - _queue.push_back (boost::shared_ptr ( + _queue.push_back (shared_ptr ( new DCPVideoFrame ( image, sub, _film->format()->dcp_size(), _film->format()->dcp_padding (_film), _film->subtitle_offset(), _film->subtitle_scale(), @@ -301,9 +310,9 @@ Encoder::process_audio (shared_ptr data) 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 resampled (new AudioBuffers (_film->audio_stream()->channels(), max_resampled_frames)); + shared_ptr resampled (new AudioBuffers (_film->audio_mapping().dcp_channels(), max_resampled_frames)); /* Resample audio */ int const resampled_frames = swr_convert ( @@ -321,7 +330,7 @@ Encoder::process_audio (shared_ptr data) } #endif - write_audio (data); + _writer->write (data); } void @@ -362,7 +371,7 @@ Encoder::encoder_thread (ServerDescription* server) } TIMING ("encoder thread %1 wakes with queue of %2", boost::this_thread::get_id(), _queue.size()); - boost::shared_ptr vf = _queue.front (); + shared_ptr 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 (); @@ -416,34 +425,10 @@ Encoder::encoder_thread (ServerDescription* server) } if (remote_backoff > 0) { - dvdomatic_sleep (remote_backoff); + dcpomatic_sleep (remote_backoff); } lock.lock (); _condition.notify_all (); } } - -void -Encoder::write_audio (shared_ptr data) -{ - AudioMapping m (_film->audio_channels ()); - if (m.dcp_channels() != _film->audio_channels()) { - - /* Remap (currently just for mono -> 5.1) */ - - shared_ptr b (new AudioBuffers (m.dcp_channels(), data->frames ())); - for (int i = 0; i < m.dcp_channels(); ++i) { - optional s = m.dcp_to_source (static_cast (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); -}