- 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(), _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
- 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,
- av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
- AV_SAMPLE_FMT_FLTP,
- _audio_content->output_audio_frame_rate(),
- av_get_default_channel_layout (MAX_AUDIO_CHANNELS),
- AV_SAMPLE_FMT_FLTP,
- _audio_content->content_audio_frame_rate(),
- 0, 0
- );
-
- swr_init (_swr_context);
- } else {
- _swr_context = 0;
- }
-}
-
-AudioDecoder::~AudioDecoder ()
-{
- if (_swr_context) {
- swr_free (&_swr_context);
- }
-}
-
-
-#if 0
-void
-AudioDecoder::process_end ()
-{
- if (_swr_context) {
-
- 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);
-
- if (frames < 0) {
- throw EncodeError (_("could not run sample-rate converter"));
- }
-
- if (frames == 0) {
- break;
- }
-
- out->set_frames (frames);
- _writer->write (out);
- }
-
- }