Various fixes to make audio analysis sort-of work.

[dcpomatic.git] / src / lib / audio_decoder.cc

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index e1c93ac77bcb23de9a5888cdc2babd36ffdef442..a9e01908c4782a6095821d080800b0568ce71214 100644 (file)
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -25,19 +25,29 @@
 #include "i18n.h"
 
 using std::stringstream;
+using std::list;
+using std::pair;
+using std::cout;
 using boost::optional;
 using boost::shared_ptr;
 
 AudioDecoder::AudioDecoder (shared_ptr<const Film> f, shared_ptr<const AudioContent> c)
        : Decoder (f)
+       , _next_audio (0)
        , _audio_content (c)
-       , _output_audio_frame_rate (_audio_content->output_audio_frame_rate (f))
 {
-       if (_audio_content->content_audio_frame_rate() != _output_audio_frame_rate) {
+       if (_audio_content->content_audio_frame_rate() != _audio_content->output_audio_frame_rate()) {
+
+               shared_ptr<const Film> film = _film.lock ();
+               assert (film);
 
                stringstream s;
-               s << String::compose ("Will resample audio from %1 to %2", _audio_content->content_audio_frame_rate(), _output_audio_frame_rate);
-               _film->log()->log (s.str ());
+               s << String::compose (
+                       "Will resample audio from %1 to %2",
+                       _audio_content->content_audio_frame_rate(), _audio_content->output_audio_frame_rate()
+                       );
+               
+               film->log()->log (s.str ());
 
                /* We will be using planar float data when we call the
                   resampler.  As far as I can see, the audio channel
@@ -49,10 +59,10 @@ AudioDecoder::AudioDecoder (shared_ptr<const Film> f, shared_ptr<const AudioCont
 
                _swr_context = swr_alloc_set_opts (
                        0,
-                       av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
+                       av_get_default_channel_layout (_audio_content->audio_channels ()),
                        AV_SAMPLE_FMT_FLTP,
-                       _output_audio_frame_rate,
-                       av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
+                       _audio_content->output_audio_frame_rate(),
+                       av_get_default_channel_layout (_audio_content->audio_channels ()),
                        AV_SAMPLE_FMT_FLTP,
                        _audio_content->content_audio_frame_rate(),
                        0, 0
@@ -78,7 +88,10 @@ AudioDecoder::process_end ()
 {
        if (_swr_context) {
 
-               shared_ptr<AudioBuffers> out (new AudioBuffers (_film->audio_mapping().dcp_channels(), 256));
+               shared_ptr<const Film> film = _film.lock ();
+               assert (film);
+               
+               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);
@@ -100,17 +113,17 @@ AudioDecoder::process_end ()
 #endif
 
 void
-AudioDecoder::emit_audio (shared_ptr<const AudioBuffers> data, Time time)
+AudioDecoder::audio (shared_ptr<const AudioBuffers> data, Time time)
 {
-       /* XXX: map audio to 5.1 */
-       
-       /* Maybe sample-rate convert */
+       /* Maybe resample */
        if (_swr_context) {
 
                /* Compute the resampled frames count and add 32 for luck */
-               int const max_resampled_frames = ceil ((int64_t) data->frames() * _output_audio_frame_rate / _audio_content->content_audio_frame_rate()) + 32;
+               int const max_resampled_frames = ceil (
+                       (int64_t) data->frames() * _audio_content->output_audio_frame_rate() / _audio_content->content_audio_frame_rate()
+                       ) + 32;
 
-               shared_ptr<AudioBuffers> resampled (new AudioBuffers (MAX_AUDIO_CHANNELS, max_resampled_frames));
+               shared_ptr<AudioBuffers> resampled (new AudioBuffers (data->channels(), max_resampled_frames));
 
                /* Resample audio */
                int const resampled_frames = swr_convert (
@@ -127,7 +140,27 @@ AudioDecoder::emit_audio (shared_ptr<const AudioBuffers> data, Time time)
                data = resampled;
        }
 
-       Audio (data, time);
+       shared_ptr<const Film> film = _film.lock ();
+       assert (film);
+       
+       /* Remap channels */
+       shared_ptr<AudioBuffers> dcp_mapped (new AudioBuffers (film->dcp_audio_channels(), data->frames()));
+       dcp_mapped->make_silent ();
+       list<pair<int, libdcp::Channel> > map = _audio_content->audio_mapping().content_to_dcp ();
+       for (list<pair<int, libdcp::Channel> >::iterator i = map.begin(); i != map.end(); ++i) {
+               dcp_mapped->accumulate_channel (data.get(), i->first, i->second);
+       }
+
+       Audio (dcp_mapped, time);
+       _next_audio = time + film->audio_frames_to_time (data->frames());
 }
 
-               
+bool
+AudioDecoder::audio_done () const
+{
+       shared_ptr<const Film> film = _film.lock ();
+       assert (film);
+       
+       return (_audio_content->length() - _next_audio) < film->audio_frames_to_time (1);
+}
+