X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder_stream.cc;h=4fedf9291eea8e55515a187ae688dbfb437748e0;hb=a9936a35fbae63311b1132e6283bf28db1d8c233;hp=fdcebbc9f0648a24a22cc8770a56c1bf52e512b6;hpb=a1201ace8a1d5d3e0a9eca769cb9e0a12bf32693;p=dcpomatic.git

diff --git a/src/lib/audio_decoder_stream.cc b/src/lib/audio_decoder_stream.cc
index fdcebbc9f..4fedf9291 100644
--- 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"
@@ -42,8 +44,8 @@ AudioDecoderStream::AudioDecoderStream (shared_ptr<const AudioContent> content,
 	, _stream (stream)
 	, _decoder (decoder)
 {
-	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 +62,7 @@ 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);
+	_content->film()->log()->log (String::compose ("-> ADS has request for %1 %2", frame, length), LogEntry::TYPE_DEBUG_DECODE);
 
 	Frame const end = frame + length - 1;
 
@@ -85,7 +87,7 @@ 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)
 			)
 		{}
 
@@ -93,7 +95,7 @@ AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
 	} else {
 		while (
 			_decoded.audio->frames() < length &&
-			!_decoder->pass ()
+			!_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
 			)
 		{}
 
@@ -134,7 +136,7 @@ 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);
+	_content->film()->log()->log (String::compose ("ADS receives %1 %2", time, data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
 
 	if (_resampler) {
 		data = _resampler->run (data);
@@ -145,7 +147,7 @@ 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 +174,7 @@ 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()));