Make KDM output options checkboxes rather than radios (part of #848).

[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 527610cdf9116d4cc0ebd241ad2ac9da6c6b716b..b7b96ddd4ad88e17b2b03057559f283a04e617da 100644 (file)
--- a/src/lib/audio_decoder_stream.cc
+++ b/src/lib/audio_decoder_stream.cc
@@ -28,7 +28,6 @@
 #include "log.h"
 #include "audio_content.h"
 #include "compose.hpp"
-#include <boost/make_shared.hpp>
 #include <iostream>
 
 #include "i18n.h"
@@ -40,16 +39,19 @@ using std::min;
 using std::max;
 using boost::optional;
 using boost::shared_ptr;
-using boost::make_shared;
 
-AudioDecoderStream::AudioDecoderStream (shared_ptr<const AudioContent> content, AudioStreamPtr stream, Decoder* decoder, bool fast, shared_ptr<Log> log)
+AudioDecoderStream::AudioDecoderStream (shared_ptr<const AudioContent> content, AudioStreamPtr stream, Decoder* decoder, shared_ptr<Log> log)
        : _content (content)
        , _stream (stream)
        , _decoder (decoder)
        , _log (log)
+         /* We effectively start having done a seek to zero; this allows silence-padding of the first
+            data that comes out of our decoder.
+         */
+       , _seek_reference (ContentTime ())
 {
        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));
+               _resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_frame_rate(), _stream->channels ()));
        }
 
        reset_decoded ();
@@ -58,7 +60,7 @@ AudioDecoderStream::AudioDecoderStream (shared_ptr<const AudioContent> content,
 void
 AudioDecoderStream::reset_decoded ()
 {
-       _decoded = ContentAudio (make_shared<AudioBuffers> (_stream->channels(), 0), 0);
+       _decoded = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_stream->channels(), 0)), 0);
 }
 
 ContentAudio
@@ -70,8 +72,14 @@ AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
 
        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 */
+       /* If we are less than (about) 5 seconds behind the data that we want we'll
+          run through it rather than seeking.
+       */
+       Frame const slack = 5 * 48000;
+
+       if (frame < _decoded.frame || end > (_decoded.frame + _decoded.audio->frames() + slack)) {
+               /* Either we have no decoded data, all our data is after the time that we
+                  want, or what we do have is a long way from what we want: seek */
                _decoder->seek (ContentTime::from_frames (frame, _content->resampled_frame_rate()), accurate);
        }
 
@@ -120,7 +128,7 @@ AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
        Frame const to_return = max ((Frame) 0, min (available, length));
 
        /* Copy our data to the output */
-       shared_ptr<AudioBuffers> out = make_shared<AudioBuffers> (_decoded.audio->channels(), to_return);
+       shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded.audio->channels(), to_return));
        out->copy_from (_decoded.audio.get(), to_return, decoded_offset, 0);
 
        Frame const remaining = max ((Frame) 0, available - to_return);
@@ -145,7 +153,7 @@ AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
 void
 AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
 {
-       _log->log (String::compose ("ADS receives %1 %2", time, data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
+       _log->log (String::compose ("ADS receives %1 %2", to_string(time), data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
 
        if (_resampler) {
                data = _resampler->run (data);
@@ -159,25 +167,11 @@ AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time
                Frame const delta_frames = delta.frames_round (frame_rate);
                if (delta_frames > 0) {
                        /* This data comes after the seek time.  Pad the data with some silence. */
-                       shared_ptr<AudioBuffers> padded = make_shared<AudioBuffers> (data->channels(), data->frames() + delta_frames);
+                       shared_ptr<AudioBuffers> padded (new AudioBuffers (data->channels(), data->frames() + delta_frames));
                        padded->make_silent ();
                        padded->copy_from (data.get(), data->frames(), 0, delta_frames);
                        data = padded;
                        time -= delta;
-               } else if (delta_frames < 0) {
-                       /* This data comes before the seek time.  Throw some data away */
-                       Frame const to_discard = min (-delta_frames, static_cast<Frame> (data->frames()));
-                       Frame const to_keep = data->frames() - to_discard;
-                       if (to_keep == 0) {
-                               /* We have to throw all this data away, so keep _seek_reference and
-                                  try again next time some data arrives.
-                               */
-                               return;
-                       }
-                       shared_ptr<AudioBuffers> trimmed = make_shared<AudioBuffers> (data->channels(), to_keep);
-                       trimmed->copy_from (data.get(), to_keep, to_discard, 0);
-                       data = trimmed;
-                       time += ContentTime::from_frames (to_discard, frame_rate);
                }
                _seek_reference = optional<ContentTime> ();
        }
@@ -251,3 +245,11 @@ AudioDecoderStream::seek (ContentTime t, bool accurate)
                _seek_reference = t;
        }
 }
+
+void
+AudioDecoderStream::set_fast ()
+{
+       if (_resampler) {
+               _resampler->set_fast ();
+       }
+}