Split audio; builds.

[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 fdcebbc9f0648a24a22cc8770a56c1bf52e512b6..7037d9f47c21eb6a99bd8eb0fbbcc52df9bdaf24 100644 (file)
--- a/src/lib/audio_decoder_stream.cc
+++ b/src/lib/audio_decoder_stream.cc
@@ -25,6 +25,8 @@
 #include "util.h"
 #include "film.h"
 #include "log.h"
+#include "audio_content.h"
+#include "compose.hpp"
 #include <iostream>
 
 #include "i18n.h"
@@ -37,13 +39,14 @@ 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, AudioDecoder* decoder, shared_ptr<Log> log)
        : _content (content)
        , _stream (stream)
        , _decoder (decoder)
+       , _log (log)
 {
-       if (content->resampled_audio_frame_rate() != _stream->frame_rate()) {
-               _resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_audio_frame_rate(), _stream->channels ()));
+       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 ()));
        }
 
        reset_decoded ();
@@ -60,7 +63,7 @@ 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), Log::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;
 
@@ -85,15 +88,20 @@ 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 (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 (Decoder::PASS_REASON_AUDIO, accurate)
                        )
                {}
 
@@ -134,7 +142,7 @@ 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 ()), Log::TYPE_DEBUG_DECODE);
+       _log->log (String::compose ("ADS receives %1 %2", time, data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
 
        if (_resampler) {
                data = _resampler->run (data);
@@ -145,7 +153,7 @@ AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time
        if (_seek_reference) {
                /* We've had an accurate seek and now we're seeing some data */
                ContentTime const delta = time - _seek_reference.get ();
-               Frame const delta_frames = delta.frames (frame_rate);
+               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 (new AudioBuffers (data->channels(), data->frames() + delta_frames));
@@ -172,7 +180,7 @@ AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time
        }
 
        if (!_position) {
-               _position = time.frames (frame_rate);
+               _position = time.frames_round (frame_rate);
        }
 
        DCPOMATIC_ASSERT (_position.get() >= (_decoded.frame + _decoded.audio->frames()));