Another attempt to do external audio moderately nicely.

[dcpomatic.git] / src / lib / j2k_wav_encoder.cc

diff --git a/src/lib/j2k_wav_encoder.cc b/src/lib/j2k_wav_encoder.cc
index 847ee948617e979df7596d95c9fe854042a7a322..be1f96fd4d6e9a8d64a96daaf2e6f14608eff421 100644 (file)
--- a/src/lib/j2k_wav_encoder.cc
+++ b/src/lib/j2k_wav_encoder.cc
@@ -46,6 +46,7 @@ using std::stringstream;
 using std::list;
 using std::vector;
 using std::pair;
+using std::cout;
 using boost::shared_ptr;
 using boost::thread;
 using boost::lexical_cast;
@@ -64,7 +65,7 @@ J2KWAVEncoder::J2KWAVEncoder (shared_ptr<const Film> f, shared_ptr<const Options
                */
                for (int i = 0; i < _film->audio_channels(); ++i) {
                        SF_INFO sf_info;
-                       sf_info.samplerate = dcp_audio_sample_rate (_film->audio_stream().get().sample_rate());
+                       sf_info.samplerate = dcp_audio_sample_rate (_film->audio_stream()->sample_rate());
                        /* We write mono files */
                        sf_info.channels = 1;
                        sf_info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_24;
@@ -229,22 +230,22 @@ J2KWAVEncoder::encoder_thread (ServerDescription* server)
 void
 J2KWAVEncoder::process_begin ()
 {
-       if (_film->audio_stream() && _film->audio_stream().get().sample_rate() != _film->target_audio_sample_rate()) {
+       if (_film->audio_stream() && _film->audio_stream()->sample_rate() != _film->target_audio_sample_rate()) {
 #ifdef HAVE_SWRESAMPLE
 
                stringstream s;
-               s << "Will resample audio from " << _film->audio_stream().get().sample_rate() << " to " << _film->target_audio_sample_rate();
+               s << "Will resample audio from " << _film->audio_stream()->sample_rate() << " to " << _film->target_audio_sample_rate();
                _film->log()->log (s.str ());
 
                /* We will be using planar float data when we call the resampler */
                _swr_context = swr_alloc_set_opts (
                        0,
-                       _film->audio_stream().get().channel_layout(),
+                       _film->audio_stream()->channel_layout(),
                        AV_SAMPLE_FMT_FLTP,
                        _film->target_audio_sample_rate(),
-                       _film->audio_stream().get().channel_layout(),
+                       _film->audio_stream()->channel_layout(),
                        AV_SAMPLE_FMT_FLTP,
-                       _film->audio_stream().get().sample_rate(),
+                       _film->audio_stream()->sample_rate(),
                        0, 0
                        );
                
@@ -314,7 +315,7 @@ J2KWAVEncoder::process_end ()
 #if HAVE_SWRESAMPLE    
        if (_film->audio_stream() && _swr_context) {
 
-               shared_ptr<AudioBuffers> out (new AudioBuffers (_film->audio_stream().get().channels(), 256));
+               shared_ptr<AudioBuffers> out (new AudioBuffers (_film->audio_stream()->channels(), 256));
                        
                while (1) {
                        int const frames = swr_convert (_swr_context, (uint8_t **) out->data(), 256, 0, 0);
@@ -336,9 +337,9 @@ J2KWAVEncoder::process_end ()
 #endif
 
        if (_film->audio_stream()) {
-               int const dcp_sr = dcp_audio_sample_rate (_film->audio_stream().get().sample_rate ());
+               int const dcp_sr = dcp_audio_sample_rate (_film->audio_stream()->sample_rate ());
                int64_t const extra_audio_frames = dcp_sr - (_audio_frames_written % dcp_sr);
-               shared_ptr<AudioBuffers> silence (new AudioBuffers (_film->audio_stream().get().channels(), extra_audio_frames));
+               shared_ptr<AudioBuffers> silence (new AudioBuffers (_film->audio_stream()->channels(), extra_audio_frames));
                silence->make_silent ();
                write_audio (silence);
                
@@ -364,9 +365,9 @@ J2KWAVEncoder::do_process_audio (shared_ptr<AudioBuffers> audio)
        if (_swr_context) {
 
                /* Compute the resampled frames count and add 32 for luck */
-               int const max_resampled_frames = ceil (audio->frames() * _film->target_audio_sample_rate() / _film->audio_stream().get().sample_rate()) + 32;
+               int const max_resampled_frames = ceil ((int64_t) audio->frames() * _film->target_audio_sample_rate() / _film->audio_stream()->sample_rate()) + 32;
 
-               resampled.reset (new AudioBuffers (_film->audio_stream().get().channels(), max_resampled_frames));
+               resampled.reset (new AudioBuffers (_film->audio_stream()->channels(), max_resampled_frames));
 
                /* Resample audio */
                int const resampled_frames = swr_convert (