Considerable rework of decoder timing; tests pass, at least.

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

diff --git a/src/lib/sndfile_decoder.cc b/src/lib/sndfile_decoder.cc
index b5a7f139bbf586722ced1bcb1892aa762b1b5fb1..3af683c570a89b21a59cf6fd68d8a9f6d37d3fb7 100644 (file)
--- 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.
        */
@@ -93,6 +97,8 @@ SndfileDecoder::pass ()
        audio (data, _done);
        _done += this_time;
        _remaining -= this_time;
+
+       return true;
 }
 
 int
@@ -113,14 +119,10 @@ SndfileDecoder::audio_frame_rate () const
        return _info.samplerate;
 }
 
-bool
-SndfileDecoder::done () const
-{
-       return _audio_position >= _sndfile_content->audio_length ();
-}
-
 void
-SndfileDecoder::seek (DCPTime t, bool accurate)
+SndfileDecoder::seek (ContentTime t, bool accurate)
 {
+       Decoder::seek (t, accurate);
+       
        /* XXX */
 }