Hopefully much cleaner handling of PTS changes under resample.

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

diff --git a/src/lib/sndfile_decoder.cc b/src/lib/sndfile_decoder.cc
index dd9e654c77de7984a4672a7a577fb8d91f340f44..1fc1ecaf2cc66c1cbcb650f0af9cad0f0d2a82fc 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"
 
@@ -38,7 +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().string().c_str(), SFM_READ, &_info);
        if (!_sndfile) {
                throw DecodeError (_("could not open audio file for reading"));
        }
@@ -53,22 +55,22 @@ SndfileDecoder::~SndfileDecoder ()
        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 (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) {
@@ -77,7 +79,7 @@ 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) {
@@ -87,12 +89,10 @@ SndfileDecoder::pass ()
                }
        }
                
-       audio->set_frames (this_time);
-       Audio (audio, double(_done) / audio_frame_rate());
+       data->set_frames (this_time);
+       audio (data, _done);
        _done += this_time;
        _remaining -= this_time;
-
-       return (_remaining == 0);
 }
 
 int
@@ -101,7 +101,7 @@ SndfileDecoder::audio_channels () const
        return _info.channels;
 }
 
-ContentAudioFrame
+AudioContent::Frame
 SndfileDecoder::audio_length () const
 {
        return _info.frames;
@@ -112,3 +112,9 @@ SndfileDecoder::audio_frame_rate () const
 {
        return _info.samplerate;
 }
+
+bool
+SndfileDecoder::done () const
+{
+       return _audio_position >= _sndfile_content->audio_length ();
+}