No-op: remove all trailing whitespace.

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

diff --git a/src/lib/sndfile_decoder.cc b/src/lib/sndfile_decoder.cc
index 2c47fcd9e9ca9ddff3946590dc9008d3ff8d8815..2fa9ae2a3c48df0328f618016c65a5f1bc64e599 100644 (file)
--- a/src/lib/sndfile_decoder.cc
+++ b/src/lib/sndfile_decoder.cc
@@ -18,10 +18,6 @@
 */
 
 #include <iostream>
-#ifdef DCPOMATIC_WINDOWS
-#include <windows.h>
-#define ENABLE_SNDFILE_WINDOWS_PROTOTYPES 1
-#endif
 #include <sndfile.h>
 #include "sndfile_content.h"
 #include "sndfile_decoder.h"
@@ -43,7 +39,7 @@ SndfileDecoder::SndfileDecoder (shared_ptr<const SndfileContent> c)
        , _remaining (_info.frames)
        , _deinterleave_buffer (0)
 {
-       
+
 }
 
 SndfileDecoder::~SndfileDecoder ()
@@ -52,12 +48,12 @@ SndfileDecoder::~SndfileDecoder ()
 }
 
 bool
-SndfileDecoder::pass (PassReason)
+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.
        */
@@ -65,7 +61,7 @@ SndfileDecoder::pass (PassReason)
        sf_count_t const this_time = min (block, _remaining);
 
        int const channels = _sndfile_content->audio_stream()->channels ();
-       
+
        shared_ptr<AudioBuffers> data (new AudioBuffers (channels, this_time));
 
        if (_sndfile_content->audio_stream()->channels() == 1) {
@@ -88,7 +84,7 @@ SndfileDecoder::pass (PassReason)
                        }
                }
        }
-               
+
        data->set_frames (this_time);
        audio (_sndfile_content->audio_stream (), data, ContentTime::from_frames (_done, _info.samplerate));
        _done += this_time;