Basics of splitting up Decoder tree like Content.

[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 8c521f94dfbdee956dbc5085b2764a73e651ee01..bbe26b6eff86cc75057f7e5595eb775e495d61bc 100644 (file)
--- a/src/lib/audio_decoder_stream.cc
+++ b/src/lib/audio_decoder_stream.cc
@@ -1,5 +1,5 @@
 /*
-    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
     it under the terms of the GNU General Public License as published by
@@ -23,6 +23,10 @@
 #include "audio_decoder.h"
 #include "resampler.h"
 #include "util.h"
+#include "film.h"
+#include "log.h"
+#include "audio_content.h"
+#include "compose.hpp"
 #include <iostream>
 
 #include "i18n.h"
@@ -35,13 +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, Decoder* 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_frame_rate() != _stream->frame_rate() && _stream->channels() > 0) {
+               _resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_frame_rate(), _stream->channels (), decoder->audio->fast ()));
        }
 
        reset_decoded ();
@@ -58,11 +63,13 @@ AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
 {
        shared_ptr<ContentAudio> dec;
 
+       _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);
+               seek (ContentTime::from_frames (frame, _content->resampled_frame_rate()), accurate);
        }
 
        /* Offset of the data that we want from the start of _decoded.audio
@@ -81,15 +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)
                        )
                {}
 
@@ -130,16 +142,18 @@ 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);
+
        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 */
                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));
@@ -166,7 +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()));
@@ -203,7 +217,7 @@ 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;