Merge master.

[dcpomatic.git] / src / lib / sndfile_decoder.cc

diff --git a/src/lib/sndfile_decoder.cc b/src/lib/sndfile_decoder.cc
index f66a7c7dc9dd363f649f2c71632a9a5057d78949..f98d4d8be83ae0e8c6bb23ce90d1d2b2d0017344 100644 (file)
--- a/src/lib/sndfile_decoder.cc
+++ b/src/lib/sndfile_decoder.cc
@@ -66,9 +66,13 @@ 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.
        */
@@ -101,9 +105,11 @@ 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
@@ -112,7 +118,7 @@ SndfileDecoder::audio_channels () const
        return _info.channels;
 }
 
-AudioContent::Frame
+AudioFrame
 SndfileDecoder::audio_length () const
 {
        return _info.frames;
@@ -124,8 +130,12 @@ 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;
 }