Audio: General fixes and improvements

Signed-off-by: Christopher Snowhill <kode54@gmail.com>

Audio/Chain/AudioChunk.h

@@ -87,7 +87,7 @@ enum {
 - (id)init;
 - (id)initWithProperties:(NSDictionary *)properties;
 
-- (void)assignSamples:(const void *)data frameCount:(size_t)count;
+- (void)assignSamples:(const void *_Nonnull)data frameCount:(size_t)count;
 - (void)assignData:(NSData *)data;
 
 - (NSData *)removeSamples:(size_t)frameCount;

Audio/Chain/AudioChunk.m

@@ -159,7 +159,7 @@ static const uint32_t AudioChannelConfigTable[] = {
 	channelConfig = config;
 }
 
-- (void)assignSamples:(const void *)data frameCount:(size_t)count {
+- (void)assignSamples:(const void *_Nonnull)data frameCount:(size_t)count {
 	if(formatAssigned) {
 		const size_t bytesPerPacket = format.mBytesPerPacket;
 		[chunkData appendBytes:data length:bytesPerPacket * count];

Audio/Chain/ChunkList.h

@@ -25,6 +25,8 @@ NS_ASSUME_NONNULL_BEGIN
 	BOOL inAdder;
 	BOOL inRemover;
 	BOOL inPeeker;
+	BOOL inMerger;
+	BOOL inConverter;
 	BOOL stopping;
 	
 	// For format converter

Audio/Chain/ChunkList.m

@@ -384,6 +384,8 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
 		inAdder = NO;
 		inRemover = NO;
 		inPeeker = NO;
+		inMerger = NO;
+		inConverter = NO;
 		stopping = NO;
 		
 		formatRead = NO;
@@ -407,7 +409,7 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
 
 - (void)dealloc {
 	stopping = YES;
-	while(inAdder || inRemover || inPeeker) {
+	while(inAdder || inRemover || inPeeker || inMerger || inConverter) {
 		usleep(500);
 	}
 	if(hdcd_decoder) {
@@ -456,7 +458,9 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
 }
 
 - (BOOL)isFull {
-	return (maxDuration - listDuration) < 0.05;
+	@synchronized (chunkList) {
+		return (maxDuration - listDuration) < 0.05;
+	}
 }
 
 - (void)addChunk:(AudioChunk *)chunk {
@@ -553,13 +557,20 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
 }
 
 - (AudioChunk *)removeAndMergeSamples:(size_t)maxFrameCount {
+	if(stopping) {
+		return [[AudioChunk alloc] init];
+	}
+
+	inMerger = YES;
+
 	BOOL formatSet = NO;
 	AudioStreamBasicDescription currentFormat;
 	uint32_t currentChannelConfig = 0;
 
 	double streamTimestamp = 0.0;
 	double streamTimeRatio = 1.0;
-	if(![self peekTimestamp:&streamTimestamp timeRatio:&streamTimeRatio]) {
+	if (![self peekTimestamp:&streamTimestamp timeRatio:&streamTimeRatio]) {
+		inMerger = NO;
 		return [[AudioChunk alloc] init];
 	}
 
@@ -570,11 +581,12 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
 	[outputChunk setStreamTimestamp:streamTimestamp];
 	[outputChunk setStreamTimeRatio:streamTimeRatio];
 
-	while(totalFrameCount < maxFrameCount) {
+	while(!stopping && totalFrameCount < maxFrameCount) {
 		AudioStreamBasicDescription newFormat;
 		uint32_t newChannelConfig;
 		if(![self peekFormat:&newFormat channelConfig:&newChannelConfig]) {
-			break;
+			usleep(500);
+			continue;
 		}
 		if(formatSet &&
 		   (memcmp(&newFormat, &currentFormat, sizeof(newFormat)) != 0 ||
@@ -589,8 +601,9 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
 		}
 
 		chunk = [self removeSamples:maxFrameCount - totalFrameCount];
-		if(![chunk duration]) {
+		if(!chunk || ![chunk frameCount]) {
-			break;
+			usleep(500);
+			continue;
 		}
 
 		if([chunk isHDCD]) {
@@ -606,9 +619,11 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
 	}
 
 	if(!totalFrameCount) {
+		inMerger = NO;
 		return [[AudioChunk alloc] init];
 	}
 
+	inMerger = NO;
 	return outputChunk;
 }
 
@@ -618,10 +633,15 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
 }
 
 - (AudioChunk *)convertChunk:(AudioChunk *)inChunk {
+	if(stopping) return [[AudioChunk alloc] init];
+
+	inConverter = YES;
+
 	AudioStreamBasicDescription chunkFormat = [inChunk format];
 	if(![inChunk duration] ||
 	   (chunkFormat.mFormatFlags == kAudioFormatFlagsNativeFloatPacked &&
 		chunkFormat.mBitsPerChannel == 32)) {
+		inConverter = NO;
 		return inChunk;
 	}
 
@@ -635,8 +655,10 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
 		inputLossless = chunkLossless;
 
 		BOOL isFloat = !!(inputFormat.mFormatFlags & kAudioFormatFlagIsFloat);
-		if((!isFloat && !(inputFormat.mBitsPerChannel >= 1 && inputFormat.mBitsPerChannel <= 32)) || (isFloat && !(inputFormat.mBitsPerChannel == 32 || inputFormat.mBitsPerChannel == 64)))
+		if((!isFloat && !(inputFormat.mBitsPerChannel >= 1 && inputFormat.mBitsPerChannel <= 32)) || (isFloat && !(inputFormat.mBitsPerChannel == 32 || inputFormat.mBitsPerChannel == 64))) {
+			inConverter = NO;
 			return [[AudioChunk alloc] init];
+		}
 
 		// These are really placeholders, as we're doing everything internally now
 		if(inputLossless &&
@@ -684,6 +706,7 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
 	NSUInteger samplesRead = [inChunk frameCount];
 	
 	if(!samplesRead) {
+		inConverter = NO;
 		return [[AudioChunk alloc] init];
 	}
 	
@@ -865,34 +888,41 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
 	
 	[outChunk assignSamples:inputBuffer frameCount:bytesReadFromInput / floatFormat.mBytesPerPacket];
 
+	inConverter = NO;
 	return outChunk;
 }
 
 - (BOOL)peekFormat:(AudioStreamBasicDescription *)format channelConfig:(uint32_t *)config {
 	if(stopping) return NO;
+	inPeeker = YES;
 	@synchronized(chunkList) {
 		if([chunkList count]) {
 			AudioChunk *chunk = [chunkList objectAtIndex:0];
 			*format = [chunk format];
 			*config = [chunk channelConfig];
+			inPeeker = NO;
 			return YES;
 		}
 	}
+	inPeeker = NO;
 	return NO;
 }
 
 - (BOOL)peekTimestamp:(double *)timestamp timeRatio:(double *)timeRatio {
 	if(stopping) return NO;
+	inPeeker = YES;
 	@synchronized (chunkList) {
 		if([chunkList count]) {
 			AudioChunk *chunk = [chunkList objectAtIndex:0];
 			*timestamp = [chunk streamTimestamp];
 			*timeRatio = [chunk streamTimeRatio];
+			inPeeker = NO;
 			return YES;
 		}
 	}
 	*timestamp = 0.0;
 	*timeRatio = 1.0;
+	inPeeker = NO;
 	return NO;
 }
 

Audio/Chain/ConverterNode.m

@@ -101,7 +101,7 @@ void scale_by_volume(float *buffer, size_t count, float volume) {
 		@autoreleasepool {
 			AudioChunk *chunk = nil;
 			chunk = [self convert];
-			if(!chunk || ![chunk duration]) {
+			if(!chunk || ![chunk frameCount]) {
 				if([self endOfStream] == YES) {
 					break;
 				}

Audio/Chain/DSP/DSPEqualizerNode.m

@@ -299,7 +299,7 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
 - (void)cleanUp {
 	stopping = YES;
 	while(processEntered) {
-		usleep(1000);
+		usleep(500);
 	}
 	[self fullShutdown];
 }
@@ -323,7 +323,7 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
 		@autoreleasepool {
 			AudioChunk *chunk = nil;
 			chunk = [self convert];
-			if(!chunk || ![chunk duration]) {
+			if(!chunk || ![chunk frameCount]) {
 				if([self endOfStream] == YES) {
 					break;
 				}
@@ -387,7 +387,7 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
 
 	size_t totalFrameCount = 0;
 	AudioChunk *chunk = [self readAndMergeChunksAsFloat32:4096];
-	if(![chunk duration]) {
+	if(!chunk || ![chunk frameCount]) {
 		processEntered = NO;
 		return nil;
 	}

Audio/Chain/DSP/DSPFSurroundNode.m

@@ -22,7 +22,6 @@ static void * kDSPFSurroundNodeContext = &kDSPFSurroundNodeContext;
 @implementation DSPFSurroundNode {
 	BOOL enableFSurround;
 	BOOL FSurroundDelayRemoved;
-	BOOL resetStreamFormat;
 	FSurroundFilter *fsurround;
 
 	BOOL stopping, paused;
@@ -96,7 +95,6 @@ static void * kDSPFSurroundNodeContext = &kDSPFSurroundNodeContext;
 		outputChannelConfig = [fsurround channelConfig];
 
 		FSurroundDelayRemoved = NO;
-		resetStreamFormat = YES;
 	} else {
 		fsurround = nil;
 	}
@@ -118,7 +116,7 @@ static void * kDSPFSurroundNodeContext = &kDSPFSurroundNodeContext;
 - (void)cleanUp {
 	stopping = YES;
 	while(processEntered) {
-		usleep(1000);
+		usleep(500);
 	}
 	[self fullShutdown];
 }
@@ -142,7 +140,7 @@ static void * kDSPFSurroundNodeContext = &kDSPFSurroundNodeContext;
 		@autoreleasepool {
 			AudioChunk *chunk = nil;
 			chunk = [self convert];
-			if(!chunk || ![chunk duration]) {
+			if(!chunk || ![chunk frameCount]) {
 				if([self endOfStream] == YES) {
 					break;
 				}
@@ -204,18 +202,18 @@ static void * kDSPFSurroundNodeContext = &kDSPFSurroundNodeContext;
 		return [self readChunk:4096];
 	}
 
-	size_t totalRequestedSamples = resetStreamFormat ? 2048 : 4096;
+	size_t totalRequestedSamples = 4096;
 
 	size_t totalFrameCount = 0;
 	AudioChunk *chunk = [self readAndMergeChunksAsFloat32:totalRequestedSamples];
-	if(![chunk duration]) {
+	if(!chunk || ![chunk frameCount]) {
 		processEntered = NO;
 		return nil;
 	}
 
 	double streamTimestamp = [chunk streamTimestamp];
 
-	float *samplePtr = resetStreamFormat ? &inBuffer[2048 * 2] : &inBuffer[0];
+	float *samplePtr = &inBuffer[0];
 
 	size_t frameCount = [chunk frameCount];
 	NSData *sampleData = [chunk removeSamples:frameCount];
@@ -224,12 +222,6 @@ static void * kDSPFSurroundNodeContext = &kDSPFSurroundNodeContext;
 
 	totalFrameCount = frameCount;
 
-	if(resetStreamFormat) {
-		bzero(&inBuffer[0], 2048 * 2 * sizeof(float));
-		totalFrameCount += 2048;
-		resetStreamFormat = NO;
-	}
-	
 	size_t countToProcess = totalFrameCount;
 	size_t samplesRendered;
 	if(countToProcess < 4096) {
@@ -237,8 +229,8 @@ static void * kDSPFSurroundNodeContext = &kDSPFSurroundNodeContext;
 		countToProcess = 4096;
 	}
 
-	[fsurround process:&inBuffer[0] output:&outBuffer[4096 * 6] count:(int)countToProcess];
+	[fsurround process:&inBuffer[0] output:&outBuffer[0] count:(int)countToProcess];
-	samplePtr = &outBuffer[4096 * 6];
+	samplePtr = &outBuffer[0];
 	samplesRendered = totalFrameCount;
 
 	if(totalFrameCount < 4096) {

Audio/Chain/DSP/DSPHRTFNode.m

@@ -241,7 +241,7 @@ static void unregisterMotionListener(void) {
 - (void)cleanUp {
 	stopping = YES;
 	while(processEntered) {
-		usleep(1000);
+		usleep(500);
 	}
 	[self fullShutdown];
 }
@@ -265,7 +265,7 @@ static void unregisterMotionListener(void) {
 		@autoreleasepool {
 			AudioChunk *chunk = nil;
 			chunk = [self convert];
-			if(!chunk || ![chunk duration]) {
+			if(!chunk || ![chunk frameCount]) {
 				if([self endOfStream] == YES) {
 					break;
 				}
@@ -328,7 +328,7 @@ static void unregisterMotionListener(void) {
 	}
 
 	AudioChunk *chunk = [self readChunkAsFloat32:4096];
-	if(!chunk || ![chunk duration]) {
+	if(!chunk || ![chunk frameCount]) {
 		processEntered = NO;
 		return nil;
 	}

Audio/Chain/DSP/DSPRubberbandNode.m

@@ -322,7 +322,7 @@ static void * kDSPRubberbandNodeContext = &kDSPRubberbandNodeContext;
 - (void)cleanUp {
 	stopping = YES;
 	while(processEntered) {
-		usleep(1000);
+		usleep(500);
 	}
 	[self fullShutdown];
 }
@@ -346,7 +346,7 @@ static void * kDSPRubberbandNodeContext = &kDSPRubberbandNodeContext;
 		@autoreleasepool {
 			AudioChunk *chunk = nil;
 			chunk = [self convert];
-			if(!chunk || ![chunk duration]) {
+			if(!chunk || ![chunk frameCount]) {
 				if([self endOfStream] == YES) {
 					break;
 				}
@@ -417,7 +417,7 @@ static void * kDSPRubberbandNodeContext = &kDSPRubberbandNodeContext;
 		samplesToProcess = blockSize;
 
 	AudioChunk *chunk = [self readAndMergeChunksAsFloat32:samplesToProcess];
-	if(![chunk duration]) {
+	if(!chunk || ![chunk frameCount]) {
 		processEntered = NO;
 		return nil;
 	}

Audio/Chain/Node.m

@@ -259,14 +259,14 @@
 		[[previousNode semaphore] signal];
 	}
 
+	[accessLock unlock];
+
 	AudioChunk *ret;
 
 	@autoreleasepool {
 		ret = [[previousNode buffer] removeAndMergeSamples:maxFrames];
 	}
 
-	[accessLock unlock];
-
 	if([ret frameCount]) {
 		[[previousNode semaphore] signal];
 	}
@@ -294,14 +294,14 @@
 		[[previousNode semaphore] signal];
 	}
 
+	[accessLock unlock];
+
 	AudioChunk *ret;
 
 	@autoreleasepool {
 		ret = [[previousNode buffer] removeAndMergeSamplesAsFloat32:maxFrames];
 	}
 
-	[accessLock unlock];
-
 	if([ret frameCount]) {
 		[[previousNode semaphore] signal];
 	}

Audio/Chain/VisualizationNode.m

@@ -233,7 +233,7 @@ static VisualizationCollection *theCollection = nil;
 			}
 			AudioChunk *chunk = nil;
 			chunk = [self readAndMergeChunksAsFloat32:512];
-			if(!chunk || ![chunk duration]) {
+			if(!chunk || ![chunk frameCount]) {
 				if([self endOfStream] == YES) {
 					break;
 				}