A couple of sndfile fixes for the new world order.

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

diff --git a/src/lib/sndfile_decoder.cc b/src/lib/sndfile_decoder.cc
index 0f2d54d153c971d73344b2ba246b182cea984ed2..432f73f0d66f5b30e6c8068a13266719cf9c197b 100644 (file)
--- a/src/lib/sndfile_decoder.cc
+++ b/src/lib/sndfile_decoder.cc
@@ -39,7 +39,8 @@ SndfileDecoder::SndfileDecoder (shared_ptr<const Film> f, shared_ptr<const Sndfi
        , _sndfile_content (c)
        , _deinterleave_buffer (0)
 {
-       _sndfile = sf_open (_sndfile_content->file().string().c_str(), SFM_READ, &_info);
+       _info.format = 0;
+       _sndfile = sf_open (_sndfile_content->path(0).string().c_str(), SFM_READ, &_info);
        if (!_sndfile) {
                throw DecodeError (_("could not open audio file for reading"));
        }
@@ -57,6 +58,10 @@ SndfileDecoder::~SndfileDecoder ()
 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.
        */
@@ -65,11 +70,11 @@ SndfileDecoder::pass ()
 
        int const channels = _sndfile_content->audio_channels ();
        
-       shared_ptr<AudioBuffers> audio (new AudioBuffers (channels, this_time));
+       shared_ptr<AudioBuffers> data (new AudioBuffers (channels, this_time));
 
        if (_sndfile_content->audio_channels() == 1) {
                /* No de-interleaving required */
-               sf_read_float (_sndfile, audio->data(0), this_time);
+               sf_read_float (_sndfile, data->data(0), this_time);
        } else {
                /* Deinterleave */
                if (!_deinterleave_buffer) {
@@ -78,7 +83,7 @@ SndfileDecoder::pass ()
                sf_readf_float (_sndfile, _deinterleave_buffer, this_time);
                vector<float*> out_ptr (channels);
                for (int i = 0; i < channels; ++i) {
-                       out_ptr[i] = audio->data(i);
+                       out_ptr[i] = data->data(i);
                }
                float* in_ptr = _deinterleave_buffer;
                for (int i = 0; i < this_time; ++i) {
@@ -88,12 +93,12 @@ SndfileDecoder::pass ()
                }
        }
                
-       audio->set_frames (this_time);
-       Audio (audio, double(_done) / audio_frame_rate());
+       data->set_frames (this_time);
+       audio (data, _done * TIME_HZ / audio_frame_rate ());
        _done += this_time;
        _remaining -= this_time;
 
-       return (_remaining == 0);
+       return true;
 }
 
 int
@@ -102,7 +107,7 @@ SndfileDecoder::audio_channels () const
        return _info.channels;
 }
 
-ContentAudioFrame
+AudioContent::Frame
 SndfileDecoder::audio_length () const
 {
        return _info.frames;
@@ -114,8 +119,11 @@ SndfileDecoder::audio_frame_rate () const
        return _info.samplerate;
 }
 
-Time
-SndfileDecoder::next () const
+void
+SndfileDecoder::seek (ContentTime t, bool accurate)
 {
-       return _next_audio;
+       Decoder::seek (t, accurate);
+
+       _done = t * audio_frame_rate() / TIME_HZ;
+       _remaining = _info.frames - _done;
 }