#include "audio_buffers.h"
#include "exceptions.h"
#include "log.h"
+#include "resampler.h"
+#include "util.h"
#include "i18n.h"
using boost::optional;
using boost::shared_ptr;
-AudioDecoder::AudioDecoder (shared_ptr<const Film> f)
- : Decoder (f)
- , _audio_position (0)
+AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> content)
+ : _audio_content (content)
{
+ if (content->output_audio_frame_rate() != content->content_audio_frame_rate() && content->audio_channels ()) {
+ _resampler.reset (new Resampler (content->content_audio_frame_rate(), content->output_audio_frame_rate(), content->audio_channels ()));
+ }
}
-#if 0
+/** Audio timestamping is made hard by many factors, but the final nail in the coffin is resampling.
+ * We have to assume that we are feeding continuous data into the resampler, and so we get continuous
+ * data out. Hence we do the timestamping here, post-resampler, just by counting samples.
+ *
+ * The time is passed in here so that after a seek we can set up our _audio_position. The
+ * time is ignored once this has been done.
+ */
void
-AudioDecoder::process_end ()
+AudioDecoder::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
{
- 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 (_resampler) {
+ data = _resampler->run (data);
+ }
- if (frames < 0) {
- throw EncodeError (_("could not run sample-rate converter"));
- }
+ if (!_audio_position) {
+ _audio_position = time;
+ }
- if (frames == 0) {
- break;
- }
+ _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (_audio_position.get (), data)));
+ _audio_position = _audio_position.get() + ContentTime (data->frames (), _audio_content->output_audio_frame_rate ());
+}
- out->set_frames (frames);
- _writer->write (out);
- }
+void
+AudioDecoder::flush ()
+{
+ if (!_resampler) {
+ return;
+ }
+ shared_ptr<const AudioBuffers> b = _resampler->flush ();
+ if (b) {
+ _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (_audio_position.get (), b)));
+ _audio_position = _audio_position.get() + ContentTime (b->frames (), _audio_content->output_audio_frame_rate ());
}
}
-#endif
void
-AudioDecoder::audio (shared_ptr<const AudioBuffers> data, AudioContent::Frame frame)
+AudioDecoder::seek (ContentTime, bool)
{
- Audio (data, frame);
- _audio_position = frame + data->frames ();
+ _audio_position.reset ();
}