*/
#include "audio_decoder.h"
-#include "stream.h"
+#include "audio_buffers.h"
+#include "exceptions.h"
+#include "log.h"
+#include "resampler.h"
+#include "util.h"
+#include "film.h"
+#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<Film> f, DecodeOptions o, Job* j)
- : Decoder (f, o, j)
+AudioDecoder::AudioDecoder (shared_ptr<const Film> film, shared_ptr<const AudioContent> content)
+ : Decoder (film)
+ , _audio_content (content)
+ , _audio_position (0)
+{
+ 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 ()));
+ }
+}
+
+/** 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.
+ */
+void
+AudioDecoder::audio (shared_ptr<const AudioBuffers> data)
+{
+ if (_resampler) {
+ data = _resampler->run (data);
+ }
+
+ _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (data, _audio_position)));
+ _audio_position += data->frames ();
+}
+
+void
+AudioDecoder::flush ()
{
+ if (!_resampler) {
+ return;
+ }
+ shared_ptr<const AudioBuffers> b = _resampler->flush ();
+ if (b) {
+ _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (b, _audio_position)));
+ _audio_position += b->frames ();
+ }
}
void
-AudioDecoder::set_audio_stream (shared_ptr<AudioStream> s)
+AudioDecoder::seek (ContentTime t, bool)
{
- _audio_stream = s;
+ shared_ptr<const Film> film = _film.lock ();
+ assert (film);
+
+ FrameRateChange frc = film->active_frame_rate_change (_audio_content->position ());
+ _audio_position = (t + first_audio()) / frc.speed_up;
}