#include "exceptions.h"
#include "log.h"
#include "resampler.h"
+#include "util.h"
+#include "film.h"
#include "i18n.h"
AudioDecoder::AudioDecoder (shared_ptr<const Film> film, shared_ptr<const AudioContent> content)
: Decoder (film)
, _audio_content (content)
- , _last_audio (0)
+ , _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, ContentTime time)
+AudioDecoder::audio (shared_ptr<const AudioBuffers> data)
{
if (_resampler) {
data = _resampler->run (data);
}
- _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (data, time)));
- _last_audio = time + (data->frames() * TIME_HZ / _audio_content->output_audio_frame_rate());
+ _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (data, _audio_position)));
+ _audio_position += data->frames ();
}
void
shared_ptr<const AudioBuffers> b = _resampler->flush ();
if (b) {
- audio (b, _last_audio);
+ _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (b, _audio_position)));
+ _audio_position += b->frames ();
}
}
+
+void
+AudioDecoder::seek (ContentTime t, bool)
+{
+ 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;
+}