Some more tidying up.

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

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index fbc66c335bf21ec149335e28e029c94e09f03de1..c33f68c01c0c0bd3e910c8e46b08774772c8688a 100644 (file)
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -36,9 +36,12 @@ AudioDecoder::AudioDecoder (shared_ptr<const Film> f, shared_ptr<const AudioCont
 {
        if (_audio_content->content_audio_frame_rate() != _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 ());
+               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
@@ -79,7 +82,10 @@ 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);
@@ -130,7 +136,9 @@ AudioDecoder::emit_audio (shared_ptr<const AudioBuffers> data, Time time)
 
        Audio (data, time);
 
-       _next_audio = time + _film->audio_frames_to_time (data->frames());
+       shared_ptr<const Film> film = _film.lock ();
+       assert (film);
+       _next_audio = time + film->audio_frames_to_time (data->frames());
 }