It builds.

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

diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc
index 32453cc13c974d58218cf5dae9eb4eb6da5047d2..616abf8464c85d538ae0f0ee16d8db7adb0b0e92 100644 (file)
--- a/src/lib/audio_decoder.cc
+++ b/src/lib/audio_decoder.cc
@@ -1,5 +1,5 @@
 /*
-    Copyright (C) 2012 Carl Hetherington <cth@carlh.net>
+    Copyright (C) 2012-2014 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
@@ -23,6 +23,7 @@
 #include "log.h"
 #include "resampler.h"
 #include "util.h"
+#include "film.h"
 
 #include "i18n.h"
 
@@ -41,7 +42,47 @@ AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> content)
        }
 }
 
-/** Audio timestamping is made hard by many factors, but the final nail in the coffin is resampling.
+shared_ptr<ContentAudio>
+AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate)
+{
+       shared_ptr<ContentAudio> dec;
+
+       AudioFrame const end = frame + length - 1;
+               
+       if (frame < _decoded_audio.frame || end > (_decoded_audio.frame + length * 4)) {
+               /* Either we have no decoded data, or what we do have is a long way from what we want: seek */
+               seek (ContentTime::from_frames (frame, _audio_content->content_audio_frame_rate()), accurate);
+       }
+
+       /* Now enough pass() calls will either:
+        *  (a) give us what we want, or
+        *  (b) hit the end of the decoder.
+        *
+        * If we are being accurate, we want the right frames,
+        * otherwise any frames will do.
+        */
+       if (accurate) {
+               while (!pass() && _decoded_audio.audio->frames() < length) {}
+       } else {
+               while (!pass() && (_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end)) {}
+       }
+       
+       /* Clean up decoded */
+
+       AudioFrame const decoded_offset = frame - _decoded_audio.frame;
+       AudioFrame const amount_left = _decoded_audio.audio->frames() - decoded_offset;
+       _decoded_audio.audio->move (decoded_offset, 0, amount_left);
+       _decoded_audio.audio->set_frames (amount_left);
+
+       shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded_audio.audio->channels(), length));
+       out->copy_from (_decoded_audio.audio.get(), length, frame - _decoded_audio.frame, 0);
+
+       return shared_ptr<ContentAudio> (new ContentAudio (out, frame));
+}
+
+/** Called by subclasses when audio data is ready.
+ *
+ *  Audio timestamping is made hard by many factors, but perhaps the most entertaining is resampling.
  *  We have to assume that we are feeding continuous data into the resampler, and so we get continuous
  *  data out.  Hence we do the timestamping here, post-resampler, just by counting samples.
  *
@@ -56,13 +97,20 @@ AudioDecoder::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
        }
 
        if (!_audio_position) {
-               _audio_position = time;
+               _audio_position = time.frames (_audio_content->output_audio_frame_rate ());
        }
 
-       _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (_audio_position.get (), data)));
-       _audio_position = _audio_position.get() + ContentTime (data->frames (), _audio_content->output_audio_frame_rate ());
+       assert (_audio_position >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
+
+       /* Resize _decoded_audio to fit the new data */
+       _decoded_audio.audio->ensure_size (_audio_position.get() + data->frames() - _decoded_audio.frame);
+
+       /* Copy new data in */
+       _decoded_audio.audio->copy_from (data.get(), data->frames(), 0, _audio_position.get() - _decoded_audio.frame);
+       _audio_position = _audio_position.get() + data->frames ();
 }
 
+/* XXX: called? */
 void
 AudioDecoder::flush ()
 {
@@ -70,11 +118,13 @@ AudioDecoder::flush ()
                return;
        }
 
+       /*
        shared_ptr<const AudioBuffers> b = _resampler->flush ();
        if (b) {
-               _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (_audio_position.get (), b)));
-               _audio_position = _audio_position.get() + ContentTime (b->frames (), _audio_content->output_audio_frame_rate ());
+               _pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (b, _audio_position.get ())));
+               _audio_position = _audio_position.get() + b->frames ();
        }
+       */
 }
 
 void