X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;ds=inline;f=src%2Flib%2Fsndfile_decoder.cc;h=432f73f0d66f5b30e6c8068a13266719cf9c197b;hb=e194f0003b60b2607da0822485c56cd8267e78dc;hp=c7311112acdd28cf4303acb5fb581912bdaa932e;hpb=afc495f722f89fea0bcc579046d1a5d362e36f69;p=dcpomatic.git

diff --git a/src/lib/sndfile_decoder.cc b/src/lib/sndfile_decoder.cc
index c7311112a..432f73f0d 100644
--- 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 +35,70 @@ 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);
+	_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"));
 	}
 
+	_done = 0;
 	_remaining = _info.frames;
 }
 
 SndfileDecoder::~SndfileDecoder ()
 {
-	if (_sndfile) {
-		sf_close (_sndfile);
-	}
+	sf_close (_sndfile);
+	delete[] _deinterleave_buffer;
 }
 
 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.
 	*/
-	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);
+	shared_ptr<AudioBuffers> data (new AudioBuffers (channels, this_time));
+
+	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, _done * TIME_HZ / audio_frame_rate ());
+	_done += this_time;
 	_remaining -= this_time;
 
-	return (_remaining == 0);
+	return true;
 }
 
 int
@@ -76,7 +107,7 @@ SndfileDecoder::audio_channels () const
 	return _info.channels;
 }
 
-ContentAudioFrame
+AudioContent::Frame
 SndfileDecoder::audio_length () const
 {
 	return _info.frames;
@@ -87,3 +118,12 @@ SndfileDecoder::audio_frame_rate () const
 {
 	return _info.samplerate;
 }
+
+void
+SndfileDecoder::seek (ContentTime t, bool accurate)
+{
+	Decoder::seek (t, accurate);
+
+	_done = t * audio_frame_rate() / TIME_HZ;
+	_remaining = _info.frames - _done;
+}