Merge master and multifarious hackery.

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

diff --git a/src/lib/sndfile_decoder.cc b/src/lib/sndfile_decoder.cc
index c7311112acdd28cf4303acb5fb581912bdaa932e..c4c6e5f4ee065f6dca6145defbb293bedcb5b7d1 100644 (file)
--- a/src/lib/sndfile_decoder.cc
+++ b/src/lib/sndfile_decoder.cc
@@ -23,6 +23,7 @@
 #include "sndfile_decoder.h"
 #include "film.h"
 #include "exceptions.h"
+#include "audio_buffers.h"
 
 #include "i18n.h"
 
@@ -34,40 +35,63 @@ using boost::shared_ptr;
 
 SndfileDecoder::SndfileDecoder (shared_ptr<const Film> f, shared_ptr<const SndfileContent> c)
        : Decoder (f)
-       , AudioDecoder (f)
+       , AudioDecoder (f, c)
        , _sndfile_content (c)
+       , _deinterleave_buffer (0)
 {
        _sndfile = sf_open (_sndfile_content->file().string().c_str(), SFM_READ, &_info);
        if (!_sndfile) {
                throw DecodeError (_("could not open audio file for reading"));
        }
 
+       _done = 0;
        _remaining = _info.frames;
 }
 
 SndfileDecoder::~SndfileDecoder ()
 {
-       if (_sndfile) {
-               sf_close (_sndfile);
-       }
+       sf_close (_sndfile);
+       delete[] _deinterleave_buffer;
 }
 
-bool
+void
 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->audio_frame_rate() / 2;
+       sf_count_t const block = _sndfile_content->content_audio_frame_rate() / 2;
        sf_count_t const this_time = min (block, _remaining);
+
+       int const channels = _sndfile_content->audio_channels ();
        
-       shared_ptr<AudioBuffers> audio (new AudioBuffers (_sndfile_content->audio_channels(), this_time));
-       sf_read_float (_sndfile, audio->data(0), this_time);
-       audio->set_frames (this_time);
-       Audio (audio);
-       _remaining -= this_time;
+       shared_ptr<AudioBuffers> data (new AudioBuffers (channels, this_time));
 
-       return (_remaining == 0);
+       if (_sndfile_content->audio_channels() == 1) {
+               /* No de-interleaving required */
+               sf_read_float (_sndfile, data->data(0), this_time);
+       } else {
+               /* Deinterleave */
+               if (!_deinterleave_buffer) {
+                       _deinterleave_buffer = new float[block * channels];
+               }
+               sf_readf_float (_sndfile, _deinterleave_buffer, this_time);
+               vector<float*> out_ptr (channels);
+               for (int i = 0; i < channels; ++i) {
+                       out_ptr[i] = data->data(i);
+               }
+               float* in_ptr = _deinterleave_buffer;
+               for (int i = 0; i < this_time; ++i) {
+                       for (int j = 0; j < channels; ++j) {
+                               *out_ptr[j]++ = *in_ptr++;
+                       }
+               }
+       }
+               
+       data->set_frames (this_time);
+       audio (data, double(_done) / audio_frame_rate());
+       _done += this_time;
+       _remaining -= this_time;
 }
 
 int
@@ -87,3 +111,9 @@ SndfileDecoder::audio_frame_rate () const
 {
        return _info.samplerate;
 }
+
+Time
+SndfileDecoder::next () const
+{
+       return _next_audio;
+}