New DCPTime/ContentTime types.

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

diff --git a/src/lib/audio_merger.cc b/src/lib/audio_merger.cc
new file mode 100644 (file)
index 0000000..d50420c
--- /dev/null
+++ b/src/lib/audio_merger.cc
@@ -0,0 +1,92 @@
+/*
+    Copyright (C) 2013-2014 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
+    the Free Software Foundation; either version 2 of the License, or
+    (at your option) any later version.
+
+    This program is distributed in the hope that it will be useful,
+    but WITHOUT ANY WARRANTY; without even the implied warranty of
+    MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+    GNU General Public License for more details.
+
+    You should have received a copy of the GNU General Public License
+    along with this program; if not, write to the Free Software
+    Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+
+*/
+
+#include "audio_buffers.h"
+#include "audio_merger.h"
+
+using std::min;
+using std::max;
+using boost::shared_ptr;
+
+AudioMerger::AudioMerger (int channels, int frame_rate)
+       : _buffers (new AudioBuffers (channels, 0))
+       , _frame_rate (frame_rate)
+       , _last_pull (0)
+{
+
+}
+
+
+TimedAudioBuffers<DCPTime>
+AudioMerger::pull (DCPTime time)
+{
+       assert (time >= _last_pull);
+       
+       TimedAudioBuffers<DCPTime> out;
+       
+       int64_t const to_return = DCPTime (time - _last_pull).frames (_frame_rate);
+       out.audio.reset (new AudioBuffers (_buffers->channels(), to_return));
+       /* And this is how many we will get from our buffer */
+       int64_t const to_return_from_buffers = min (to_return, int64_t (_buffers->frames ()));
+       
+       /* Copy the data that we have to the back end of the return buffer */
+       out.audio->copy_from (_buffers.get(), to_return_from_buffers, 0, to_return - to_return_from_buffers);
+       /* Silence any gap at the start */
+       out.audio->make_silent (0, to_return - to_return_from_buffers);
+       
+       out.time = _last_pull;
+       _last_pull = time;
+       
+       /* And remove the data we're returning from our buffers */
+       if (_buffers->frames() > to_return_from_buffers) {
+               _buffers->move (to_return_from_buffers, 0, _buffers->frames() - to_return_from_buffers);
+       }
+       _buffers->set_frames (_buffers->frames() - to_return_from_buffers);
+       
+       return out;
+}
+
+void
+AudioMerger::push (shared_ptr<const AudioBuffers> audio, DCPTime time)
+{
+       assert (time >= _last_pull);
+       
+       int64_t frame = time.frames (_frame_rate);
+       int64_t after = max (int64_t (_buffers->frames()), frame + audio->frames() - _last_pull.frames (_frame_rate));
+       _buffers->ensure_size (after);
+       _buffers->accumulate_frames (audio.get(), 0, frame - _last_pull.frames (_frame_rate), audio->frames ());
+       _buffers->set_frames (after);
+}
+
+TimedAudioBuffers<DCPTime>
+AudioMerger::flush ()
+{
+       if (_buffers->frames() == 0) {
+               return TimedAudioBuffers<DCPTime> ();
+       }
+       
+       return TimedAudioBuffers<DCPTime> (_buffers, _last_pull);
+}
+
+void
+AudioMerger::clear (DCPTime t)
+{
+       _last_pull = t;
+       _buffers.reset (new AudioBuffers (_buffers->channels(), 0));
+}