Merge master.

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

diff --git a/src/lib/sndfile_decoder.cc b/src/lib/sndfile_decoder.cc
index 3a71fab528508ece0dfdf713356308b2e8af8cfa..602014d5883d2dba1381588bb44a7036debc9c96 100644 (file)
--- a/src/lib/sndfile_decoder.cc
+++ b/src/lib/sndfile_decoder.cc
@@ -25,7 +25,6 @@
 #include <sndfile.h>
 #include "sndfile_content.h"
 #include "sndfile_decoder.h"
-#include "film.h"
 #include "exceptions.h"
 #include "audio_buffers.h"
 
@@ -37,9 +36,8 @@ using std::min;
 using std::cout;
 using boost::shared_ptr;
 
-SndfileDecoder::SndfileDecoder (shared_ptr<const Film> f, shared_ptr<const SndfileContent> c)
-       : Decoder (f)
-       , AudioDecoder (f, c)
+SndfileDecoder::SndfileDecoder (shared_ptr<const SndfileContent> c)
+       : AudioDecoder (c)
        , _sndfile_content (c)
        , _deinterleave_buffer (0)
 {
@@ -76,7 +74,7 @@ SndfileDecoder::pass ()
        /* Do things in half second blocks as I think there may be limits
           to what FFmpeg (and in particular the resampler) can cope with.
        */
-       sf_count_t const block = _sndfile_content->content_audio_frame_rate() / 2;
+       sf_count_t const block = _sndfile_content->audio_frame_rate() / 2;
        sf_count_t const this_time = min (block, _remaining);
 
        int const channels = _sndfile_content->audio_channels ();
@@ -133,7 +131,6 @@ SndfileDecoder::audio_frame_rate () const
 void
 SndfileDecoder::seek (ContentTime t, bool accurate)
 {
-       Decoder::seek (t, accurate);
        AudioDecoder::seek (t, accurate);
 
        _done = t.frames (audio_frame_rate ());