Revert "Use make_shared<>."

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

diff --git a/src/lib/audio_decoder_stream.cc b/src/lib/audio_decoder_stream.cc
index 4d1a5e4c58837e87d1d5e9fcc1446fc614dbe43b..1bfc65871da4d2bdd962d0258a8d24e90b6f66f9 100644 (file)
--- a/src/lib/audio_decoder_stream.cc
+++ b/src/lib/audio_decoder_stream.cc
@@ -1,19 +1,20 @@
 /*
-    Copyright (C) 2012-2015 Carl Hetherington <cth@carlh.net>
+    Copyright (C) 2012-2016 Carl Hetherington <cth@carlh.net>
 
-    This program is free software; you can redistribute it and/or modify
+    This file is part of DCP-o-matic.
+
+    DCP-o-matic 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,
+    DCP-o-matic 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.
+    along with DCP-o-matic.  If not, see <http://www.gnu.org/licenses/>.
 
 */
 
@@ -39,13 +40,14 @@ using std::max;
 using boost::optional;
 using boost::shared_ptr;
 
-AudioDecoderStream::AudioDecoderStream (shared_ptr<const AudioContent> content, AudioStreamPtr stream, AudioDecoder* decoder)
+AudioDecoderStream::AudioDecoderStream (shared_ptr<const AudioContent> content, AudioStreamPtr stream, Decoder* decoder, bool fast, shared_ptr<Log> log)
        : _content (content)
        , _stream (stream)
        , _decoder (decoder)
+       , _log (log)
 {
-       if (content->resampled_audio_frame_rate() != _stream->frame_rate() && _stream->channels() > 0) {
-               _resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_audio_frame_rate(), _stream->channels (), decoder->fast ()));
+       if (content->resampled_frame_rate() != _stream->frame_rate() && _stream->channels() > 0) {
+               _resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_frame_rate(), _stream->channels (), fast));
        }
 
        reset_decoded ();
@@ -62,13 +64,13 @@ AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
 {
        shared_ptr<ContentAudio> dec;
 
-       _content->film()->log()->log (String::compose ("-> ADS has request for %1 %2", frame, length), LogEntry::TYPE_DEBUG_DECODE);
+       _log->log (String::compose ("-> ADS has request for %1 %2", frame, length), LogEntry::TYPE_DEBUG_DECODE);
 
        Frame const end = frame + length - 1;
 
        if (frame < _decoded.frame || end > (_decoded.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, _content->resampled_audio_frame_rate()), accurate);
+               _decoder->seek (ContentTime::from_frames (frame, _content->resampled_frame_rate()), accurate);
        }
 
        /* Offset of the data that we want from the start of _decoded.audio
@@ -87,15 +89,20 @@ AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
                /* Keep stuffing data into _decoded until we have enough data, or the subclass does not want to give us any more */
                while (
                        (_decoded.frame > frame || (_decoded.frame + _decoded.audio->frames()) < end) &&
-                       !_decoder->pass (Decoder::PASS_REASON_AUDIO)
+                       !_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
                        )
                {}
 
                decoded_offset = frame - _decoded.frame;
+
+               _log->log (
+                       String::compose ("Accurate ADS::get has offset %1 from request %2 and available %3", decoded_offset, frame, _decoded.frame),
+                       LogEntry::TYPE_DEBUG_DECODE
+                       );
        } else {
                while (
                        _decoded.audio->frames() < length &&
-                       !_decoder->pass (Decoder::PASS_REASON_AUDIO)
+                       !_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
                        )
                {}
 
@@ -136,13 +143,13 @@ AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
 void
 AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
 {
-       _content->film()->log()->log (String::compose ("ADS receives %1 %2", time, data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
+       _log->log (String::compose ("ADS receives %1 %2", time, data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
 
        if (_resampler) {
                data = _resampler->run (data);
        }
 
-       Frame const frame_rate = _content->resampled_audio_frame_rate ();
+       Frame const frame_rate = _content->resampled_frame_rate ();
 
        if (_seek_reference) {
                /* We've had an accurate seek and now we're seeing some data */
@@ -211,7 +218,7 @@ AudioDecoderStream::add (shared_ptr<const AudioBuffers> data)
        _position = _position.get() + data->frames ();
 
        /* Limit the amount of data we keep in case nobody is asking for it */
-       int const max_frames = _content->resampled_audio_frame_rate () * 10;
+       int const max_frames = _content->resampled_frame_rate () * 10;
        if (_decoded.audio->frames() > max_frames) {
                int const to_remove = _decoded.audio->frames() - max_frames;
                _decoded.frame += to_remove;