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-rw-r--r--src/lib/sndfile_decoder.cc22
1 files changed, 16 insertions, 6 deletions
diff --git a/src/lib/sndfile_decoder.cc b/src/lib/sndfile_decoder.cc
index e10f4f568..d6537843e 100644
--- a/src/lib/sndfile_decoder.cc
+++ b/src/lib/sndfile_decoder.cc
@@ -55,9 +55,13 @@ SndfileDecoder::~SndfileDecoder ()
delete[] _deinterleave_buffer;
}
-void
+bool
SndfileDecoder::pass ()
{
+ if (_remaining == 0) {
+ return true;
+ }
+
/* Do things in half second blocks as I think there may be limits
to what FFmpeg (and in particular the resampler) can cope with.
*/
@@ -90,9 +94,11 @@ SndfileDecoder::pass ()
}
data->set_frames (this_time);
- audio (data, _done);
+ audio (data, _done * TIME_HZ / audio_frame_rate ());
_done += this_time;
_remaining -= this_time;
+
+ return _remaining == 0;
}
int
@@ -101,7 +107,7 @@ SndfileDecoder::audio_channels () const
return _info.channels;
}
-AudioContent::Frame
+AudioFrame
SndfileDecoder::audio_length () const
{
return _info.frames;
@@ -113,8 +119,12 @@ SndfileDecoder::audio_frame_rate () const
return _info.samplerate;
}
-bool
-SndfileDecoder::done () const
+void
+SndfileDecoder::seek (ContentTime t, bool accurate)
{
- return _audio_position >= _sndfile_content->audio_length ();
+ Decoder::seek (t, accurate);
+ AudioDecoder::seek (t, accurate);
+
+ _done = t * audio_frame_rate() / TIME_HZ;
+ _remaining = _info.frames - _done;
}