Untested use of Frame for video/audio content lengths.

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

diff --git a/src/lib/sndfile_decoder.cc b/src/lib/sndfile_decoder.cc
index d6537843e8970ebde3ae6872bb99f04fd66e5de3..ecd1a7e16d46d74ba49e84c2a22482e24839b38c 100644 (file)
--- a/src/lib/sndfile_decoder.cc
+++ b/src/lib/sndfile_decoder.cc
@@ -1,5 +1,5 @@
 /*
-    Copyright (C) 2012 Carl Hetherington <cth@carlh.net>
+    Copyright (C) 2012-2015 Carl Hetherington <cth@carlh.net>
 
     This program is free software; you can redistribute it and/or modify
     it under the terms of the GNU General Public License as published by
@@ -18,10 +18,13 @@
 */
 
 #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"
-#include "film.h"
 #include "exceptions.h"
 #include "audio_buffers.h"
 
@@ -33,14 +36,20 @@ 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)
 {
        _info.format = 0;
+
+       /* Here be monsters.  See fopen_boost for similar shenanigans */
+#ifdef DCPOMATIC_WINDOWS
+       _sndfile = sf_wchar_open (_sndfile_content->path(0).c_str(), SFM_READ, &_info);
+#else  
        _sndfile = sf_open (_sndfile_content->path(0).string().c_str(), SFM_READ, &_info);
+#endif
+       
        if (!_sndfile) {
                throw DecodeError (_("could not open audio file for reading"));
        }
@@ -56,7 +65,7 @@ SndfileDecoder::~SndfileDecoder ()
 }
 
 bool
-SndfileDecoder::pass ()
+SndfileDecoder::pass (PassReason)
 {
        if (_remaining == 0) {
                return true;
@@ -65,7 +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 ();
@@ -94,7 +103,7 @@ SndfileDecoder::pass ()
        }
                
        data->set_frames (this_time);
-       audio (data, _done * TIME_HZ / audio_frame_rate ());
+       audio (data, ContentTime::from_frames (_done, audio_frame_rate ()));
        _done += this_time;
        _remaining -= this_time;
 
@@ -107,7 +116,7 @@ SndfileDecoder::audio_channels () const
        return _info.channels;
 }
 
-AudioFrame
+Frame
 SndfileDecoder::audio_length () const
 {
        return _info.frames;
@@ -122,9 +131,8 @@ SndfileDecoder::audio_frame_rate () const
 void
 SndfileDecoder::seek (ContentTime t, bool accurate)
 {
-       Decoder::seek (t, accurate);
        AudioDecoder::seek (t, accurate);
 
-       _done = t * audio_frame_rate() / TIME_HZ;
+       _done = t.frames (audio_frame_rate ());
        _remaining = _info.frames - _done;
 }