Revert to previous low latency output system

This reverts usage of the AVFoundation output to use
the previous lower latency CoreAudio output, and
paves the way for a change I am cooking up soon.

Fixes several issues with playback and seeking latency.

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

Audio/AudioPlayer.m

@@ -91,7 +91,7 @@
 	bufferChain = [[BufferChain alloc] initWithController:self];
 	[self notifyStreamChanged:userInfo];
 
-	while(![bufferChain open:url withUserInfo:userInfo withRGInfo:rgi]) {
+	while(![bufferChain open:url withOutputFormat:[output format] withUserInfo:userInfo withRGInfo:rgi]) {
 		bufferChain = nil;
 
 		[self requestNextStream:userInfo];
@@ -401,7 +401,7 @@
 	}
 
 	if(pathsEqual || ([[nextStream scheme] isEqualToString:[[lastChain streamURL] scheme]] && (([nextStream host] == nil && [[lastChain streamURL] host] == nil) || [[nextStream host] isEqualToString:[[lastChain streamURL] host]]) && [[nextStream path] isEqualToString:[[lastChain streamURL] path]])) {
-		if([lastChain setTrack:nextStream] && [newChain openWithInput:[lastChain inputNode] withUserInfo:nextStreamUserInfo withRGInfo:nextStreamRGInfo]) {
+		if([lastChain setTrack:nextStream] && [newChain openWithInput:[lastChain inputNode] withOutputFormat:[output format] withUserInfo:nextStreamUserInfo withRGInfo:nextStreamRGInfo]) {
 			[newChain setStreamURL:nextStream];
 
 			@synchronized(chainQueue) {
@@ -420,7 +420,7 @@
 
 	NSURL *url = nextStream;
 
-	while(shouldContinue && ![newChain open:url withUserInfo:nextStreamUserInfo withRGInfo:nextStreamRGInfo]) {
+	while(shouldContinue && ![newChain open:url withOutputFormat:[output format] withUserInfo:nextStreamUserInfo withRGInfo:nextStreamRGInfo]) {
 		if(nextStream == nil) {
 			newChain = nil;
 			if(output)

Audio/Chain/BufferChain.h

@@ -28,13 +28,14 @@
 - (id)initWithController:(id)c;
 - (void)buildChain;
 
-- (BOOL)open:(NSURL *)url withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi;
+- (BOOL)open:(NSURL *)url withOutputFormat:(AudioStreamBasicDescription)outputFormat withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi;
 
 // Used when changing tracks to reuse the same decoder
-- (BOOL)openWithInput:(InputNode *)i withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi;
+- (BOOL)openWithInput:(InputNode *)i withOutputFormat:(AudioStreamBasicDescription)outputFormat withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi;
 
 // Used when resetting the decoder on seek
 - (BOOL)openWithDecoder:(id<CogDecoder>)decoder
+       withOutputFormat:(AudioStreamBasicDescription)outputFormat
            withUserInfo:(id)userInfo
              withRGInfo:(NSDictionary *)rgi;
 

Audio/Chain/BufferChain.m

@@ -40,7 +40,7 @@
 	finalNode = converterNode;
 }
 
-- (BOOL)open:(NSURL *)url withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi {
+- (BOOL)open:(NSURL *)url withOutputFormat:(AudioStreamBasicDescription)outputFormat withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi {
 	[self setStreamURL:url];
 	[self setUserInfo:userInfo];
 
@@ -66,7 +66,11 @@
 	if([properties valueForKey:@"channelConfig"])
 		inputChannelConfig = [[properties valueForKey:@"channelConfig"] unsignedIntValue];
 
-	if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig isLossless:[[properties valueForKey:@"encoding"] isEqualToString:@"lossless"]])
+	outputFormat.mChannelsPerFrame = inputFormat.mChannelsPerFrame;
+	outputFormat.mBytesPerFrame = ((outputFormat.mBitsPerChannel + 7) / 8) * outputFormat.mChannelsPerFrame;
+	outputFormat.mBytesPerPacket = outputFormat.mBytesPerFrame * outputFormat.mFramesPerPacket;
+
+	if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig outputFormat:outputFormat isLossless:[[properties valueForKey:@"encoding"] isEqualToString:@"lossless"]])
 		return NO;
 
 	[self setRGInfo:rgi];
@@ -76,7 +80,7 @@
 	return YES;
 }
 
-- (BOOL)openWithInput:(InputNode *)i withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi {
+- (BOOL)openWithInput:(InputNode *)i withOutputFormat:(AudioStreamBasicDescription)outputFormat withUserInfo:(id)userInfo withRGInfo:(NSDictionary *)rgi {
 	DLog(@"New buffer chain!");
 	[self setUserInfo:userInfo];
 	[self buildChain];
@@ -91,8 +95,12 @@
 	if([properties valueForKey:@"channelConfig"])
 		inputChannelConfig = [[properties valueForKey:@"channelConfig"] unsignedIntValue];
 
+	outputFormat.mChannelsPerFrame = inputFormat.mChannelsPerFrame;
+	outputFormat.mBytesPerFrame = ((outputFormat.mBitsPerChannel + 7) / 8) * outputFormat.mChannelsPerFrame;
+	outputFormat.mBytesPerPacket = outputFormat.mBytesPerFrame * outputFormat.mFramesPerPacket;
+
 	DLog(@"Input Properties: %@", properties);
-	if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
+	if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig outputFormat:outputFormat isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
 		return NO;
 
 	[self setRGInfo:rgi];
@@ -101,6 +109,7 @@
 }
 
 - (BOOL)openWithDecoder:(id<CogDecoder>)decoder
+	   withOutputFormat:(AudioStreamBasicDescription)outputFormat
            withUserInfo:(id)userInfo
              withRGInfo:(NSDictionary *)rgi;
 {
@@ -120,7 +129,11 @@
 	if([properties valueForKey:@"channelConfig"])
 		inputChannelConfig = [[properties valueForKey:@"channelConfig"] unsignedIntValue];
 
-	if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
+	outputFormat.mChannelsPerFrame = inputFormat.mChannelsPerFrame;
+	outputFormat.mBytesPerFrame = ((outputFormat.mBitsPerChannel + 7) / 8) * outputFormat.mChannelsPerFrame;
+	outputFormat.mBytesPerPacket = outputFormat.mBytesPerFrame * outputFormat.mFramesPerPacket;
+
+	if(![converterNode setupWithInputFormat:inputFormat withInputConfig:inputChannelConfig outputFormat:outputFormat isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
 		return NO;
 
 	[self setRGInfo:rgi];

Audio/Chain/ConverterNode.h

@@ -12,11 +12,15 @@
 #import <AudioUnit/AudioUnit.h>
 #import <CoreAudio/AudioHardware.h>
 
+#import <soxr.h>
+
 #import "Node.h"
 
 @interface ConverterNode : Node {
 	NSDictionary *rgInfo;
 
+	soxr_t soxr;
+
 	void *inputBuffer;
 	size_t inputBufferSize;
 	size_t inpSize, inpOffset;
@@ -25,12 +29,33 @@
 	BOOL convertEntered;
 	BOOL paused;
 
+	BOOL skipResampler;
+
+	unsigned int PRIME_LEN_;
+	unsigned int N_samples_to_add_;
+	unsigned int N_samples_to_drop_;
+
+	BOOL is_preextrapolated_;
+	int is_postextrapolated_;
+
+	int latencyEaten;
+	int latencyEatenPost;
+
+	double sampleRatio;
+
 	float volumeScale;
 
+	void *floatBuffer;
+	size_t floatBufferSize;
+
+	void *extrapolateBuffer;
+	size_t extrapolateBufferSize;
+
 	BOOL rememberedLossless;
 
 	AudioStreamBasicDescription inputFormat;
 	AudioStreamBasicDescription floatFormat;
+	AudioStreamBasicDescription outputFormat;
 
 	uint32_t inputChannelConfig;
 
@@ -43,7 +68,7 @@
 
 - (id)initWithController:(id)c previous:(id)p;
 
-- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inputFormat withInputConfig:(uint32_t)inputConfig isLossless:(BOOL)lossless;
+- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inputFormat withInputConfig:(uint32_t)inputConfig outputFormat:(AudioStreamBasicDescription)outputFormat isLossless:(BOOL)lossless;
 - (void)cleanUp;
 
 - (void)process;
@@ -51,6 +76,8 @@
 
 - (void)setRGInfo:(NSDictionary *)rgi;
 
+- (void)setOutputFormat:(AudioStreamBasicDescription)outputFormat;
+
 - (void)inputFormatDidChange:(AudioStreamBasicDescription)format inputConfig:(uint32_t)inputConfig;
 
 - (void)refreshVolumeScaling;

Audio/Chain/ConverterNode.m

@@ -16,6 +16,9 @@
 
 #import "Logging.h"
 
+#import "lpc.h"
+#import "util.h"
+
 #ifdef _DEBUG
 #import "BadSampleCleaner.h"
 #endif
@@ -48,13 +51,21 @@ static void *kConverterNodeContext = &kConverterNodeContext;
 	if(self) {
 		rgInfo = nil;
 
+		soxr = 0;
 		inputBuffer = NULL;
 		inputBufferSize = 0;
+		floatBuffer = NULL;
+		floatBufferSize = 0;
 
 		stopping = NO;
 		convertEntered = NO;
 		paused = NO;
 
+		skipResampler = YES;
+
+		extrapolateBuffer = NULL;
+		extrapolateBufferSize = 0;
+
 		[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.volumeScaling" options:0 context:kConverterNodeContext];
 	}
 
@@ -102,8 +113,10 @@ void scale_by_volume(float *buffer, size_t count, float volume) {
 			}
 		}
 		if(streamFormatChanged) {
-			[self cleanUp];
+			@autoreleasepool {
-			[self setupWithInputFormat:newInputFormat withInputConfig:newInputChannelConfig isLossless:rememberedLossless];
+				[self cleanUp];
+				[self setupWithInputFormat:newInputFormat withInputConfig:newInputChannelConfig outputFormat:self->outputFormat isLossless:rememberedLossless];
+			}
 		}
 	}
 }
@@ -173,6 +186,57 @@ void scale_by_volume(float *buffer, size_t count, float volume) {
 			return nil;
 		}
 
+		if(stopping || paused || streamFormatChanged || [self shouldContinue] == NO || [self endOfStream] == YES) {
+			if(!skipResampler) {
+				if(!is_postextrapolated_) {
+					is_postextrapolated_ = 1;
+				}
+			} else {
+				is_postextrapolated_ = 3;
+			}
+		}
+
+		// Extrapolate start
+		if(!skipResampler && !is_preextrapolated_) {
+			size_t inputSamples = bytesReadFromInput / floatFormat.mBytesPerPacket;
+			size_t prime = MIN(inputSamples, PRIME_LEN_);
+			size_t _N_samples_to_add_ = N_samples_to_add_;
+			size_t newSize = _N_samples_to_add_ * floatFormat.mBytesPerPacket;
+			newSize += bytesReadFromInput;
+
+			if(newSize > inputBufferSize) {
+				inputBuffer = realloc(inputBuffer, inputBufferSize = newSize * 3);
+			}
+
+			memmove(inputBuffer + _N_samples_to_add_ * floatFormat.mBytesPerPacket, inputBuffer, bytesReadFromInput);
+
+			lpc_extrapolate_bkwd(inputBuffer + _N_samples_to_add_ * floatFormat.mBytesPerPacket, inputSamples, prime, floatFormat.mChannelsPerFrame, LPC_ORDER, _N_samples_to_add_, &extrapolateBuffer, &extrapolateBufferSize);
+
+			bytesReadFromInput += _N_samples_to_add_ * floatFormat.mBytesPerPacket;
+			latencyEaten = N_samples_to_drop_;
+			is_preextrapolated_ = YES;
+		}
+
+		if(is_postextrapolated_ == 1) {
+			size_t inputSamples = bytesReadFromInput / floatFormat.mBytesPerPacket;
+			size_t prime = MIN(inputSamples, PRIME_LEN_);
+			size_t _N_samples_to_add_ = N_samples_to_add_;
+
+			size_t newSize = bytesReadFromInput;
+			newSize += _N_samples_to_add_ * floatFormat.mBytesPerPacket;
+			if(newSize > inputBufferSize) {
+				inputBuffer = realloc(inputBuffer, inputBufferSize = newSize * 3);
+			}
+
+			lpc_extrapolate_fwd(inputBuffer, inputSamples, prime, floatFormat.mChannelsPerFrame, LPC_ORDER, _N_samples_to_add_, &extrapolateBuffer, &extrapolateBufferSize);
+
+			bytesReadFromInput += _N_samples_to_add_ * floatFormat.mBytesPerPacket;
+			latencyEatenPost = N_samples_to_drop_;
+			is_postextrapolated_ = 2;
+		} else if(is_postextrapolated_ == 3) {
+			latencyEatenPost = 0;
+		}
+
 		// Input now contains bytesReadFromInput worth of floats, in the input sample rate
 		inpSize = bytesReadFromInput;
 		inpOffset = 0;
@@ -182,16 +246,77 @@ void scale_by_volume(float *buffer, size_t count, float volume) {
 
 	ioNumberPackets -= ioNumberPackets % floatFormat.mBytesPerPacket;
 
+	if(ioNumberPackets) {
+		size_t inputSamples = ioNumberPackets / floatFormat.mBytesPerPacket;
+		ioNumberPackets = (UInt32)inputSamples;
+		ioNumberPackets = (UInt32)ceil((float)ioNumberPackets * sampleRatio);
+		ioNumberPackets = (ioNumberPackets + 255) & ~255;
+
+		size_t newSize = ioNumberPackets * floatFormat.mBytesPerPacket;
+		if(!floatBuffer || floatBufferSize < newSize) {
+			floatBuffer = realloc(floatBuffer, floatBufferSize = newSize * 3);
+		}
+
+		if(stopping) {
+			convertEntered = NO;
+			return nil;
+		}
+
+		size_t inputDone = 0;
+		size_t outputDone = 0;
+
+		if(!skipResampler) {
+			ioNumberPackets += soxr_delay(soxr);
+			soxr_process(soxr, (float *)(((uint8_t *)inputBuffer) + inpOffset), inputSamples, &inputDone, floatBuffer, ioNumberPackets, &outputDone);
+
+			if(latencyEatenPost) {
+				// Post file or format change flush
+				size_t idone = 0, odone = 0;
+
+				do {
+					soxr_process(soxr, NULL, 0, &idone, floatBuffer + outputDone * floatFormat.mBytesPerPacket, ioNumberPackets - outputDone, &odone);
+					outputDone += odone;
+				} while(odone > 0);
+			}
+		} else {
+			memcpy(floatBuffer, (((uint8_t *)inputBuffer) + inpOffset), inputSamples * floatFormat.mBytesPerPacket);
+			inputDone = inputSamples;
+			outputDone = inputSamples;
+		}
+
+		inpOffset += inputDone * floatFormat.mBytesPerPacket;
+
+		if(latencyEaten) {
+			if(outputDone > latencyEaten) {
+				outputDone -= latencyEaten;
+				memmove(floatBuffer, floatBuffer + latencyEaten * floatFormat.mBytesPerPacket, outputDone * floatFormat.mBytesPerPacket);
+				latencyEaten = 0;
+			} else {
+				latencyEaten -= outputDone;
+				outputDone = 0;
+			}
+		}
+
+		if(latencyEatenPost) {
+			if(outputDone > latencyEatenPost) {
+				outputDone -= latencyEatenPost;
+			} else {
+				outputDone = 0;
+			}
+			latencyEatenPost = 0;
+		}
+
+		ioNumberPackets = (UInt32)outputDone * floatFormat.mBytesPerPacket;
+	}
+
 	if(ioNumberPackets) {
 		AudioChunk *chunk = [[AudioChunk alloc] init];
 		[chunk setFormat:nodeFormat];
 		if(nodeChannelConfig) {
 			[chunk setChannelConfig:nodeChannelConfig];
 		}
-		scale_by_volume((float *)(((uint8_t *)inputBuffer) + inpOffset), ioNumberPackets / sizeof(float), volumeScale);
+		scale_by_volume(floatBuffer, ioNumberPackets / sizeof(float), volumeScale);
-		[chunk assignSamples:(((uint8_t *)inputBuffer) + inpOffset) frameCount:ioNumberPackets / floatFormat.mBytesPerPacket];
+		[chunk assignSamples:floatBuffer frameCount:ioNumberPackets / floatFormat.mBytesPerPacket];
-
-		inpOffset += ioNumberPackets;
 		convertEntered = NO;
 		return chunk;
 	}
@@ -260,9 +385,10 @@ static float db_to_scale(float db) {
 	volumeScale = scale;
 }
 
-- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inf withInputConfig:(uint32_t)inputConfig isLossless:(BOOL)lossless {
+- (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inf withInputConfig:(uint32_t)inputConfig outputFormat:(AudioStreamBasicDescription)outf isLossless:(BOOL)lossless {
 	// Make the converter
 	inputFormat = inf;
+	outputFormat = outf;
 
 	inputChannelConfig = inputConfig;
 
@@ -289,11 +415,40 @@ static float db_to_scale(float db) {
 	inpOffset = 0;
 	inpSize = 0;
 
-	// This is a post resampler, post-down/upmix format
+	// This is a post resampler format
 
 	nodeFormat = floatFormat;
+	nodeFormat.mSampleRate = outputFormat.mSampleRate;
 	nodeChannelConfig = inputChannelConfig;
 
+	sampleRatio = (double)outputFormat.mSampleRate / (double)floatFormat.mSampleRate;
+	skipResampler = fabs(sampleRatio - 1.0) < 1e-7;
+	if(!skipResampler) {
+		soxr_quality_spec_t q_spec = soxr_quality_spec(SOXR_HQ, 0);
+		soxr_io_spec_t io_spec = soxr_io_spec(SOXR_FLOAT32_I, SOXR_FLOAT32_I);
+		soxr_runtime_spec_t runtime_spec = soxr_runtime_spec(0);
+		soxr_error_t error;
+
+		soxr = soxr_create(floatFormat.mSampleRate, outputFormat.mSampleRate, floatFormat.mChannelsPerFrame, &error, &io_spec, &q_spec, &runtime_spec);
+		if(error)
+			return NO;
+
+		PRIME_LEN_ = MAX(floatFormat.mSampleRate / 20, 1024u);
+		PRIME_LEN_ = MIN(PRIME_LEN_, 16384u);
+		PRIME_LEN_ = MAX(PRIME_LEN_, (unsigned int)(2 * LPC_ORDER + 1));
+
+		N_samples_to_add_ = floatFormat.mSampleRate;
+		N_samples_to_drop_ = outputFormat.mSampleRate;
+
+		samples_len(&N_samples_to_add_, &N_samples_to_drop_, 20, 8192u);
+
+		is_preextrapolated_ = NO;
+		is_postextrapolated_ = 0;
+	}
+
+	latencyEaten = 0;
+	latencyEatenPost = 0;
+
 	PrintStreamDesc(&inf);
 	PrintStreamDesc(&nodeFormat);
 
@@ -324,7 +479,7 @@ static float db_to_scale(float db) {
 		usleep(500);
 	}
 	[self cleanUp];
-	[self setupWithInputFormat:format withInputConfig:inputConfig isLossless:rememberedLossless];
+	[self setupWithInputFormat:format withInputConfig:inputConfig outputFormat:self->outputFormat isLossless:rememberedLossless];
 }
 
 - (void)setRGInfo:(NSDictionary *)rgi {
@@ -338,6 +493,20 @@ static float db_to_scale(float db) {
 	while(convertEntered) {
 		usleep(500);
 	}
+	if(soxr) {
+		soxr_delete(soxr);
+		soxr = NULL;
+	}
+	if(extrapolateBuffer) {
+		free(extrapolateBuffer);
+		extrapolateBuffer = NULL;
+		extrapolateBufferSize = 0;
+	}
+	if(floatBuffer) {
+		free(floatBuffer);
+		floatBuffer = NULL;
+		floatBufferSize = 0;
+	}
 	if(inputBuffer) {
 		free(inputBuffer);
 		inputBuffer = NULL;

Audio/Chain/Node.m

@@ -11,7 +11,7 @@
 #import "BufferChain.h"
 #import "Logging.h"
 
-#import "OutputAVFoundation.h"
+#import "OutputCoreAudio.h"
 
 #import <pthread.h>
 

Audio/Chain/OutputNode.h

@@ -13,7 +13,7 @@
 #import <CoreAudio/AudioHardware.h>
 
 #import "Node.h"
-#import "OutputAVFoundation.h"
+#import "OutputCoreAudio.h"
 
 @interface OutputNode : Node {
 	AudioStreamBasicDescription format;
@@ -21,7 +21,7 @@
 
 	double amountPlayed;
 	double amountPlayedInterval;
-	OutputAVFoundation *output;
+	OutputCoreAudio *output;
 
 	BOOL paused;
 	BOOL started;

Audio/Chain/OutputNode.m

@@ -9,7 +9,7 @@
 #import "OutputNode.h"
 #import "AudioPlayer.h"
 #import "BufferChain.h"
-#import "OutputAVFoundation.h"
+#import "OutputCoreAudio.h"
 
 #import "Logging.h"
 
@@ -23,7 +23,7 @@
 	started = NO;
 	intervalReported = NO;
 
-	output = [[OutputAVFoundation alloc] initWithController:self];
+	output = [[OutputCoreAudio alloc] initWithController:self];
 
 	[output setup];
 }
@@ -149,6 +149,7 @@
 			format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket;
 			channelConfig = config;
 
+			[converter setOutputFormat:format];
 			[converter inputFormatDidChange:[bufferChain inputFormat] inputConfig:[bufferChain inputConfig]];
 		}
 	}

Audio/CogAudio.xcodeproj/project.pbxproj

@@ -25,8 +25,6 @@
 		17D21CA80B8BE4BA00D1EBDE /* Node.m in Sources */ = {isa = PBXBuildFile; fileRef = 17D21C7D0B8BE4BA00D1EBDE /* Node.m */; };
 		17D21CA90B8BE4BA00D1EBDE /* OutputNode.h in Headers */ = {isa = PBXBuildFile; fileRef = 17D21C7E0B8BE4BA00D1EBDE /* OutputNode.h */; settings = {ATTRIBUTES = (Public, ); }; };
 		17D21CAA0B8BE4BA00D1EBDE /* OutputNode.m in Sources */ = {isa = PBXBuildFile; fileRef = 17D21C7F0B8BE4BA00D1EBDE /* OutputNode.m */; };
-		17D21CC50B8BE4BA00D1EBDE /* OutputAVFoundation.h in Headers */ = {isa = PBXBuildFile; fileRef = 17D21C9C0B8BE4BA00D1EBDE /* OutputAVFoundation.h */; settings = {ATTRIBUTES = (Public, ); }; };
-		17D21CC60B8BE4BA00D1EBDE /* OutputAVFoundation.m in Sources */ = {isa = PBXBuildFile; fileRef = 17D21C9D0B8BE4BA00D1EBDE /* OutputAVFoundation.m */; };
 		17D21CC70B8BE4BA00D1EBDE /* Status.h in Headers */ = {isa = PBXBuildFile; fileRef = 17D21C9E0B8BE4BA00D1EBDE /* Status.h */; settings = {ATTRIBUTES = (Public, ); }; };
 		17D21CF30B8BE5EF00D1EBDE /* CogSemaphore.h in Headers */ = {isa = PBXBuildFile; fileRef = 17D21CF10B8BE5EF00D1EBDE /* CogSemaphore.h */; settings = {ATTRIBUTES = (Public, ); }; };
 		17D21CF40B8BE5EF00D1EBDE /* CogSemaphore.m in Sources */ = {isa = PBXBuildFile; fileRef = 17D21CF20B8BE5EF00D1EBDE /* CogSemaphore.m */; };
@@ -66,6 +64,13 @@
 		8350416D28646149006B32CC /* CoreMedia.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 8350416C28646149006B32CC /* CoreMedia.framework */; };
 		835C88B1279811A500E28EAE /* hdcd_decode2.h in Headers */ = {isa = PBXBuildFile; fileRef = 835C88AF279811A500E28EAE /* hdcd_decode2.h */; };
 		835C88B2279811A500E28EAE /* hdcd_decode2.c in Sources */ = {isa = PBXBuildFile; fileRef = 835C88B0279811A500E28EAE /* hdcd_decode2.c */; };
+		835DD2672ACAF1D90057E319 /* OutputCoreAudio.m in Sources */ = {isa = PBXBuildFile; fileRef = 835DD2652ACAF1D90057E319 /* OutputCoreAudio.m */; };
+		835DD2682ACAF1D90057E319 /* OutputCoreAudio.h in Headers */ = {isa = PBXBuildFile; fileRef = 835DD2662ACAF1D90057E319 /* OutputCoreAudio.h */; };
+		835DD2702ACAF5AD0057E319 /* LICENSE.LGPL in Resources */ = {isa = PBXBuildFile; fileRef = 835DD26B2ACAF5AD0057E319 /* LICENSE.LGPL */; };
+		835DD2712ACAF5AD0057E319 /* License.txt in Resources */ = {isa = PBXBuildFile; fileRef = 835DD26C2ACAF5AD0057E319 /* License.txt */; };
+		835DD2722ACAF5AD0057E319 /* lpc.h in Headers */ = {isa = PBXBuildFile; fileRef = 835DD26D2ACAF5AD0057E319 /* lpc.h */; };
+		835DD2732ACAF5AD0057E319 /* util.h in Headers */ = {isa = PBXBuildFile; fileRef = 835DD26E2ACAF5AD0057E319 /* util.h */; };
+		835DD2742ACAF5AD0057E319 /* lpc.c in Sources */ = {isa = PBXBuildFile; fileRef = 835DD26F2ACAF5AD0057E319 /* lpc.c */; };
 		835FAC5E27BCA14D00BA8562 /* BadSampleCleaner.h in Headers */ = {isa = PBXBuildFile; fileRef = 835FAC5C27BCA14D00BA8562 /* BadSampleCleaner.h */; };
 		835FAC5F27BCA14D00BA8562 /* BadSampleCleaner.m in Sources */ = {isa = PBXBuildFile; fileRef = 835FAC5D27BCA14D00BA8562 /* BadSampleCleaner.m */; };
 		836DF618298F6F5F00CD0580 /* libsoxr.0.dylib in Frameworks */ = {isa = PBXBuildFile; fileRef = 836DF615298F6E8900CD0580 /* libsoxr.0.dylib */; };
@@ -132,8 +137,6 @@
 		17D21C7D0B8BE4BA00D1EBDE /* Node.m */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.objc; path = Node.m; sourceTree = "<group>"; };
 		17D21C7E0B8BE4BA00D1EBDE /* OutputNode.h */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.h; path = OutputNode.h; sourceTree = "<group>"; };
 		17D21C7F0B8BE4BA00D1EBDE /* OutputNode.m */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.objc; path = OutputNode.m; sourceTree = "<group>"; };
-		17D21C9C0B8BE4BA00D1EBDE /* OutputAVFoundation.h */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.h; path = OutputAVFoundation.h; sourceTree = "<group>"; };
-		17D21C9D0B8BE4BA00D1EBDE /* OutputAVFoundation.m */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.objc; path = OutputAVFoundation.m; sourceTree = "<group>"; };
 		17D21C9E0B8BE4BA00D1EBDE /* Status.h */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.h; path = Status.h; sourceTree = "<group>"; };
 		17D21CF10B8BE5EF00D1EBDE /* CogSemaphore.h */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.h; path = CogSemaphore.h; sourceTree = "<group>"; };
 		17D21CF20B8BE5EF00D1EBDE /* CogSemaphore.m */ = {isa = PBXFileReference; fileEncoding = 30; lastKnownFileType = sourcecode.c.objc; path = CogSemaphore.m; sourceTree = "<group>"; };
@@ -174,6 +177,13 @@
 		8350416C28646149006B32CC /* CoreMedia.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = CoreMedia.framework; path = System/Library/Frameworks/CoreMedia.framework; sourceTree = SDKROOT; };
 		835C88AF279811A500E28EAE /* hdcd_decode2.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = hdcd_decode2.h; sourceTree = "<group>"; };
 		835C88B0279811A500E28EAE /* hdcd_decode2.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = hdcd_decode2.c; sourceTree = "<group>"; };
+		835DD2652ACAF1D90057E319 /* OutputCoreAudio.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; path = OutputCoreAudio.m; sourceTree = "<group>"; };
+		835DD2662ACAF1D90057E319 /* OutputCoreAudio.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = OutputCoreAudio.h; sourceTree = "<group>"; };
+		835DD26B2ACAF5AD0057E319 /* LICENSE.LGPL */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = LICENSE.LGPL; sourceTree = "<group>"; };
+		835DD26C2ACAF5AD0057E319 /* License.txt */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = License.txt; sourceTree = "<group>"; };
+		835DD26D2ACAF5AD0057E319 /* lpc.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = lpc.h; sourceTree = "<group>"; };
+		835DD26E2ACAF5AD0057E319 /* util.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = util.h; sourceTree = "<group>"; };
+		835DD26F2ACAF5AD0057E319 /* lpc.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = lpc.c; sourceTree = "<group>"; };
 		835FAC5C27BCA14D00BA8562 /* BadSampleCleaner.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = BadSampleCleaner.h; path = Utils/BadSampleCleaner.h; sourceTree = SOURCE_ROOT; };
 		835FAC5D27BCA14D00BA8562 /* BadSampleCleaner.m */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.objc; name = BadSampleCleaner.m; path = Utils/BadSampleCleaner.m; sourceTree = SOURCE_ROOT; };
 		836DF615298F6E8900CD0580 /* libsoxr.0.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; name = libsoxr.0.dylib; path = ../ThirdParty/soxr/lib/libsoxr.0.dylib; sourceTree = "<group>"; };
@@ -354,12 +364,12 @@
 		17D21C9B0B8BE4BA00D1EBDE /* Output */ = {
 			isa = PBXGroup;
 			children = (
+				835DD2662ACAF1D90057E319 /* OutputCoreAudio.h */,
+				835DD2652ACAF1D90057E319 /* OutputCoreAudio.m */,
 				834A41AD287ABD6F00EB9D9B /* FSurroundFilter.h */,
 				834A41AE287ABD6F00EB9D9B /* FSurroundFilter.mm */,
 				839E56EB2879515D00DFB5F4 /* HeadphoneFilter.h */,
 				839E56EC2879515D00DFB5F4 /* HeadphoneFilter.mm */,
-				17D21C9C0B8BE4BA00D1EBDE /* OutputAVFoundation.h */,
-				17D21C9D0B8BE4BA00D1EBDE /* OutputAVFoundation.m */,
 			);
 			path = Output;
 			sourceTree = "<group>";
@@ -367,6 +377,7 @@
 		17D21CD80B8BE5B400D1EBDE /* ThirdParty */ = {
 			isa = PBXGroup;
 			children = (
+				835DD2692ACAF5AD0057E319 /* lvqcl */,
 				834A41A4287A90AB00EB9D9B /* fsurround */,
 				839E56E02879450300DFB5F4 /* hrtf */,
 				831A50152865A8800049CFE4 /* rsstate.cpp */,
@@ -437,6 +448,26 @@
 			path = hdcd;
 			sourceTree = "<group>";
 		};
+		835DD2692ACAF5AD0057E319 /* lvqcl */ = {
+			isa = PBXGroup;
+			children = (
+				835DD26A2ACAF5AD0057E319 /* License */,
+				835DD26D2ACAF5AD0057E319 /* lpc.h */,
+				835DD26E2ACAF5AD0057E319 /* util.h */,
+				835DD26F2ACAF5AD0057E319 /* lpc.c */,
+			);
+			path = lvqcl;
+			sourceTree = "<group>";
+		};
+		835DD26A2ACAF5AD0057E319 /* License */ = {
+			isa = PBXGroup;
+			children = (
+				835DD26B2ACAF5AD0057E319 /* LICENSE.LGPL */,
+				835DD26C2ACAF5AD0057E319 /* License.txt */,
+			);
+			path = License;
+			sourceTree = "<group>";
+		};
 		83725A8F27AA16C90003F694 /* Frameworks */ = {
 			isa = PBXGroup;
 			children = (
@@ -488,18 +519,20 @@
 				834A41AC287A90AB00EB9D9B /* channelmaps.h in Headers */,
 				17D21CA50B8BE4BA00D1EBDE /* InputNode.h in Headers */,
 				834A41A9287A90AB00EB9D9B /* freesurround_decoder.h in Headers */,
+				835DD2732ACAF5AD0057E319 /* util.h in Headers */,
 				17D21CA70B8BE4BA00D1EBDE /* Node.h in Headers */,
 				8399CF2C27B5D1D5008751F1 /* NSDictionary+Merge.h in Headers */,
 				17D21CA90B8BE4BA00D1EBDE /* OutputNode.h in Headers */,
 				8328995427CB511000D7F028 /* RedundantPlaylistDataStore.h in Headers */,
-				17D21CC50B8BE4BA00D1EBDE /* OutputAVFoundation.h in Headers */,
 				83504165286447DA006B32CC /* Downmix.h in Headers */,
 				839E56E52879450300DFB5F4 /* HrtfData.h in Headers */,
 				83B74281289E027F005AAC28 /* CogAudio-Bridging-Header.h in Headers */,
 				17D21CC70B8BE4BA00D1EBDE /* Status.h in Headers */,
 				17D21CF30B8BE5EF00D1EBDE /* CogSemaphore.h in Headers */,
+				835DD2682ACAF1D90057E319 /* OutputCoreAudio.h in Headers */,
 				839E56E62879450300DFB5F4 /* Endianness.h in Headers */,
 				17D21DC70B8BE79700D1EBDE /* CoreAudioUtils.h in Headers */,
+				835DD2722ACAF5AD0057E319 /* lpc.h in Headers */,
 				839E56ED2879515D00DFB5F4 /* HeadphoneFilter.h in Headers */,
 				17D21EBD0B8BF44000D1EBDE /* AudioPlayer.h in Headers */,
 				831A50182865A8B30049CFE4 /* rsstate.h in Headers */,
@@ -589,6 +622,8 @@
 			isa = PBXResourcesBuildPhase;
 			buildActionMask = 2147483647;
 			files = (
+				835DD2712ACAF5AD0057E319 /* License.txt in Resources */,
+				835DD2702ACAF5AD0057E319 /* LICENSE.LGPL in Resources */,
 			);
 			runOnlyForDeploymentPostprocessing = 0;
 		};
@@ -608,7 +643,6 @@
 				831A50162865A8800049CFE4 /* rsstate.cpp in Sources */,
 				17D21CA80B8BE4BA00D1EBDE /* Node.m in Sources */,
 				17D21CAA0B8BE4BA00D1EBDE /* OutputNode.m in Sources */,
-				17D21CC60B8BE4BA00D1EBDE /* OutputAVFoundation.m in Sources */,
 				835C88B2279811A500E28EAE /* hdcd_decode2.c in Sources */,
 				835FAC5F27BCA14D00BA8562 /* BadSampleCleaner.m in Sources */,
 				834FD4ED27AF91220063BC83 /* AudioChunk.m in Sources */,
@@ -630,8 +664,10 @@
 				834FD4F127AF93680063BC83 /* ChunkList.m in Sources */,
 				834A41B0287ABD6F00EB9D9B /* FSurroundFilter.mm in Sources */,
 				8EC122600B993BD500C5B3AD /* ConverterNode.m in Sources */,
+				835DD2672ACAF1D90057E319 /* OutputCoreAudio.m in Sources */,
 				8E8D3D300CBAEE6E00135C1B /* AudioContainer.m in Sources */,
 				B0575F300D687A4000411D77 /* Helper.m in Sources */,
+				835DD2742ACAF5AD0057E319 /* lpc.c in Sources */,
 				834A41AA287A90AB00EB9D9B /* freesurround_decoder.cpp in Sources */,
 				07DB5F3F0ED353A900C2E3EF /* AudioMetadataWriter.m in Sources */,
 			);

Audio/Output/OutputCoreAudio.h

@@ -47,8 +47,6 @@ using std::atomic_long;
 
 	NSLock *outputLock;
 
-	BOOL rsDone;
-	void *rsstate, *rsold;
 	double secondsLatency;
 	double visPushed;
 
@@ -126,7 +124,7 @@ using std::atomic_long;
 	
 	AudioChunk *chunkRemain;
 	
-	int resamplerRemain, visResamplerRemain;
+	int visResamplerRemain;
 
 	BOOL resetStreamFormat;
 	

Audio/Output/OutputCoreAudio.m

@@ -72,27 +72,12 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
 	AudioStreamBasicDescription format;
 	uint32_t config;
 	if([outputController peekFormat:&format channelConfig:&config]) {
-		// XXX ERROR with AirPods - Can't go higher than CD*8 surround - 192k stereo
-		// Emits to console: [AUScotty] Initialize: invalid FFT size 16384
-		// DSD256 stereo emits: [AUScotty] Initialize: invalid FFT size 65536
 		AudioStreamBasicDescription origFormat;
 		uint32_t origConfig = config;
 		memcpy(&origFormat, &format, sizeof(origFormat));
 
-		BOOL formatClipped = NO;
-		const double targetSampleRate = deviceFormat.mSampleRate;
-		double srcRate = format.mSampleRate;
-		double dstRate = targetSampleRate;
 		UInt32 srcChannels = format.mChannelsPerFrame;
 		UInt32 dstChannels = deviceFormat.mChannelsPerFrame;
-		if(fabs(format.mSampleRate - targetSampleRate) > 1e-5) {
-			format.mSampleRate = targetSampleRate;
-			formatClipped = YES;
-			if(fabs(srcRate - lastClippedSampleRate) > 1e-5) {
-				lastClippedSampleRate = srcRate;
-				streamFormatStarted = NO;
-			}
-		}
 		if(srcChannels != dstChannels) {
 			format.mChannelsPerFrame = dstChannels;
 			format.mBytesPerFrame = ((format.mBitsPerChannel + 7) / 8) * dstChannels;
@@ -103,19 +88,6 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
 			downmixer = nil;
 		}
 		if(!streamFormatStarted || config != realStreamChannelConfig || memcmp(&newFormat, &format, sizeof(format)) != 0) {
-			[outputLock lock];
-			if(formatClipped) {
-				ALog(@"Format clipped to %f Hz", targetSampleRate);
-				if(rsstate) {
-					rsold = rsstate;
-					rsstate = NULL;
-				}
-				rsstate = rsstate_new(srcChannels, srcRate, dstRate);
-			} else if(rsstate) {
-				rsold = rsstate;
-				rsstate = NULL;
-			}
-			[outputLock unlock];
 			newFormat = format;
 			newChannelConfig = config;
 			streamFormatStarted = YES;
@@ -126,13 +98,6 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
 			visFormat.mBytesPerPacket = visFormat.mBytesPerFrame * visFormat.mFramesPerPacket;
 
 			downmixerForVis = [[DownmixProcessor alloc] initWithInputFormat:origFormat inputConfig:origConfig andOutputFormat:visFormat outputConfig:AudioConfigMono];
-			[outputLock lock];
-			if(!rsold) {
-				realStreamFormat = format;
-				realStreamChannelConfig = config;
-				streamFormatChanged = YES;
-			}
-			[outputLock unlock];
 		}
 	}
 
@@ -164,35 +129,6 @@ static OSStatus eqRenderCallback(void *inRefCon, AudioUnitRenderActionFlags *ioA
 							  location:@"pre downmix"];
 #endif
 		const float *outputPtr = (const float *)[samples bytes];
-		[outputLock lock];
-		if(rsstate) {
-			if(frameCount + resamplerRemain > 8192) {
-				int leftoverFrameCount = frameCount + resamplerRemain - 8192;
-				int frameOffset = frameCount - leftoverFrameCount;
-				chunkRemain = chunk;
-				[chunk assignSamples:&outputPtr[frameOffset * format.mChannelsPerFrame] frameCount:leftoverFrameCount];
-				frameCount = frameOffset;
-			}
-			cblas_scopy(frameCount * format.mChannelsPerFrame, outputPtr, 1, &rsInBuffer[resamplerRemain * format.mChannelsPerFrame], 1);
-			outputPtr = &rsInBuffer[0];
-			frameCount += resamplerRemain;
-			resamplerRemain = 0;
-			size_t inDone = 0;
-			size_t framesDone = rsstate_resample(rsstate, outputPtr, frameCount, &inDone, &rsTempBuffer[0], amountToRead);
-			resamplerRemain = (int)(frameCount - inDone);
-			if(resamplerRemain && inDone) {
-				memmove(&rsInBuffer[0], &rsInBuffer[inDone * format.mChannelsPerFrame], resamplerRemain * format.mBytesPerPacket);
-			}
-			if(!framesDone) {
-				[outputLock unlock];
-				return 0;
-			}
-			frameCount = (int)framesDone;
-			outputPtr = &rsTempBuffer[0];
-			chunkDuration = frameCount / newFormat.mSampleRate;
-		}
-		[outputLock unlock];
-
 		[downmixerForVis process:outputPtr
 					  frameCount:frameCount
 						  output:&visAudio[0]];
@@ -403,14 +339,6 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
 			visPushed = 0.0;
 			started = NO;
 			restarted = NO;
-			if(rsstate) {
-				rsstate_delete(rsstate);
-				rsstate = NULL;
-			}
-			if(rsold) {
-				rsstate_delete(rsold);
-				rsold = NULL;
-			}
 			if(rsvis) {
 				rsstate_delete(rsvis);
 				rsvis = NULL;
@@ -665,6 +593,9 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
 		_deviceFormat = format;
 		deviceFormat = *(format.streamDescription);
 
+		// Force format for HRTF
+		deviceFormat.mSampleRate = 96000.0;
+
 		/// Seems some 3rd party devices return incorrect stuff...or I just don't like noninterleaved data.
 		deviceFormat.mFormatFlags &= ~kLinearPCMFormatFlagIsNonInterleaved;
 		//    deviceFormat.mFormatFlags &= ~kLinearPCMFormatFlagIsFloat;
@@ -892,127 +823,89 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
 
 	int samplesRenderedTotal = 0;
 
-	for(size_t i = 0; i < 2;) {
+	int samplesRendered = inputRendered;
-		int samplesRendered;
 
-		if(i == 0) {
+	samplePtr = &inputBuffer[0];
-			[outputLock lock];
+
-			if(!rsold) {
+	if(samplesRendered || fsurround) {
-				[outputLock unlock];
+		[outputLock lock];
-				++i;
+		if(fsurround) {
-				continue;
+			int countToProcess = samplesRendered;
+			if(countToProcess < 4096) {
+				bzero(samplePtr + countToProcess * 2, (4096 - countToProcess) * 2 * sizeof(float));
+				countToProcess = 4096;
 			}
-			samplesRendered = rsstate_flush(rsold, &rsTempBuffer[0], 4096);
+			[fsurround process:samplePtr output:&fsurroundBuffer[0] count:countToProcess];
-			if(samplesRendered < 4096) {
+			samplePtr = &fsurroundBuffer[0];
-				rsstate_delete(rsold);
+			if(resetStreamFormat || samplesRendered < 4096) {
-				rsold = NULL;
+				bzero(&fsurroundBuffer[4096 * 6], 4096 * 2 * sizeof(float));
-				[outputLock unlock];
+				[fsurround process:&fsurroundBuffer[4096 * 6] output:&fsurroundBuffer[4096 * 6] count:4096];
-				rsDone = YES;
+				samplesRendered += 2048;
-				++i;
-				continue;
 			}
-			[outputLock unlock];
+			if(!FSurroundDelayRemoved) {
-			samplePtr = &rsTempBuffer[0];
+				FSurroundDelayRemoved = YES;
-		} else {
+				if(samplesRendered > 2048) {
-			samplesRendered = inputRendered;
+					samplePtr += 2048 * 6;
-			samplePtr = &inputBuffer[0];
+					samplesRendered -= 2048;
-			if(rsDone) {
+				}
-				rsDone = NO;
-				realStreamFormat = newFormat;
-				realStreamChannelConfig = newChannelConfig;
-				[self updateStreamFormat];
 			}
 		}
+		[outputLock unlock];
 
-		if(samplesRendered || fsurround) {
+		if(!samplesRendered) {
-			[outputLock lock];
+			return 0;
-			if(fsurround) {
+		}
-				int countToProcess = samplesRendered;
+
-				if(countToProcess < 4096) {
+		[outputLock lock];
-					bzero(samplePtr + countToProcess * 2, (4096 - countToProcess) * 2 * sizeof(float));
+		if(hrtf) {
-					countToProcess = 4096;
+			[hrtf process:samplePtr sampleCount:samplesRendered toBuffer:&hrtfBuffer[0]];
+			samplePtr = &hrtfBuffer[0];
+		}
+		[outputLock unlock];
+
+		if(eqEnabled && eqInitialized) {
+			const int channels = streamFormat.mChannelsPerFrame;
+			if(channels > 0) {
+				const size_t channelsminusone = channels - 1;
+				uint8_t tempBuffer[sizeof(AudioBufferList) + sizeof(AudioBuffer) * channelsminusone];
+				AudioBufferList *ioData = (AudioBufferList *)&tempBuffer[0];
+
+				ioData->mNumberBuffers = channels;
+				for(size_t i = 0; i < channels; ++i) {
+					ioData->mBuffers[i].mData = &eqBuffer[4096 * i];
+					ioData->mBuffers[i].mDataByteSize = samplesRendered * sizeof(float);
+					ioData->mBuffers[i].mNumberChannels = 1;
 				}
-				[fsurround process:samplePtr output:&fsurroundBuffer[0] count:countToProcess];
+
-				samplePtr = &fsurroundBuffer[0];
+				status = AudioUnitRender(_eq, NULL, &timeStamp, 0, samplesRendered, ioData);
-				if(resetStreamFormat || samplesRendered < 4096) {
+
-					bzero(&fsurroundBuffer[4096 * 6], 4096 * 2 * sizeof(float));
+				if(status != noErr) {
-					[fsurround process:&fsurroundBuffer[4096 * 6] output:&fsurroundBuffer[4096 * 6] count:4096];
+					return 0;
-					samplesRendered += 2048;
 				}
-				if(!FSurroundDelayRemoved) {
+
-					FSurroundDelayRemoved = YES;
+				timeStamp.mSampleTime += ((double)samplesRendered) / streamFormat.mSampleRate;
-					if(samplesRendered > 2048) {
+
-						samplePtr += 2048 * 6;
+				for(int i = 0; i < channels; ++i) {
-						samplesRendered -= 2048;
+					cblas_scopy(samplesRendered, &eqBuffer[4096 * i], 1, samplePtr + i, channels);
-					}
-				}
-			}
-			[outputLock unlock];
-
-			if(!samplesRendered) {
-				break;
-			}
-
-			[outputLock lock];
-			if(hrtf) {
-				[hrtf process:samplePtr sampleCount:samplesRendered toBuffer:&hrtfBuffer[0]];
-				samplePtr = &hrtfBuffer[0];
-			}
-			[outputLock unlock];
-
-			if(eqEnabled && eqInitialized) {
-				const int channels = streamFormat.mChannelsPerFrame;
-				if(channels > 0) {
-					const size_t channelsminusone = channels - 1;
-					uint8_t tempBuffer[sizeof(AudioBufferList) + sizeof(AudioBuffer) * channelsminusone];
-					AudioBufferList *ioData = (AudioBufferList *)&tempBuffer[0];
-
-					ioData->mNumberBuffers = channels;
-					for(size_t i = 0; i < channels; ++i) {
-						ioData->mBuffers[i].mData = &eqBuffer[4096 * i];
-						ioData->mBuffers[i].mDataByteSize = samplesRendered * sizeof(float);
-						ioData->mBuffers[i].mNumberChannels = 1;
-					}
-
-					status = AudioUnitRender(_eq, NULL, &timeStamp, 0, samplesRendered, ioData);
-
-					if(status != noErr) {
-						return 0;
-					}
-
-					timeStamp.mSampleTime += ((double)samplesRendered) / streamFormat.mSampleRate;
-
-					for(int i = 0; i < channels; ++i) {
-						cblas_scopy(samplesRendered, &eqBuffer[4096 * i], 1, samplePtr + i, channels);
-					}
 				}
 			}
+		}
 			
-			if(downmixer) {
+		if(downmixer) {
-				[downmixer process:samplePtr frameCount:samplesRendered output:&downmixBuffer[0]];
+			[downmixer process:samplePtr frameCount:samplesRendered output:&downmixBuffer[0]];
-				samplePtr = &downmixBuffer[0];
+			samplePtr = &downmixBuffer[0];
-			}
+		}
 
 #ifdef OUTPUT_LOG
-			size_t dataByteSize = samplesRendered * sizeof(float) * deviceFormat.mChannelsPerFrame;
+		size_t dataByteSize = samplesRendered * sizeof(float) * deviceFormat.mChannelsPerFrame;
 
-			fwrite(samplePtr, 1, dataByteSize, _logFile);
+		fwrite(samplePtr, 1, dataByteSize, _logFile);
 #endif
-		}
-
-		if(i == 0) {
-			samplesRenderedTotal += samplesRendered;
-			if(samplesRendered) {
-				break;
-			}
-			++i;
-		} else {
-			inputBufferLastTime = 0;
-			samplesRenderedTotal += samplesRendered;
-			++i;
-		}
 	}
 
+	inputBufferLastTime = 0;
+	samplesRenderedTotal += samplesRendered;
+
 	return samplesRenderedTotal;
 }
 
@@ -1093,10 +986,6 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
 		downmixer = nil;
 		downmixerForVis = nil;
 
-		rsDone = NO;
-		rsstate = NULL;
-		rsold = NULL;
-		
 		lastClippedSampleRate = 0.0;
 
 		rsvis = NULL;
@@ -1106,7 +995,6 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
 
 		chunkRemain = nil;
 
-		resamplerRemain = 0;
 		visResamplerRemain = 0;
 		
 		secondsLatency = 0;
@@ -1187,7 +1075,9 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
 		[self audioOutputBlock];
 
 		[_au allocateRenderResourcesAndReturnError:&err];
-		
+
+		[self updateDeviceFormat];
+
 		visController = [VisualizationController sharedController];
 
 		[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.outputDevice" options:0 context:kOutputCoreAudioContext];
@@ -1326,14 +1216,6 @@ current_device_listener(AudioObjectID inObjectID, UInt32 inNumberAddresses, cons
 #endif
 		outputController = nil;
 		visController = nil;
-		if(rsstate) {
-			rsstate_delete(rsstate);
-			rsstate = NULL;
-		}
-		if(rsold) {
-			rsstate_delete(rsold);
-			rsold = NULL;
-		}
 		if(rsvis) {
 			rsstate_delete(rsvis);
 			rsvis = NULL;

Audio/ThirdParty/lvqcl/License/LICENSE.LGPL

@@ -0,0 +1,504 @@
+		  GNU LESSER GENERAL PUBLIC LICENSE
+		       Version 2.1, February 1999
+
+ Copyright (C) 1991, 1999 Free Software Foundation, Inc.
+ 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
+
+[This is the first released version of the Lesser GPL.  It also counts
+ as the successor of the GNU Library Public License, version 2, hence
+ the version number 2.1.]
+
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+
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+
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+
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+

Audio/ThirdParty/lvqcl/License/License.txt

@@ -0,0 +1,39 @@
+SoX resampler plugin for foobar2000 audio player
+Copyright (C) lvqcl
+
+This library is free software; you can redistribute it and/or
+modify it under the terms of the GNU Lesser General Public
+License as published by the Free Software Foundation; either
+version 2.1 of the License, or (at your option) any later version.
+
+This library is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+Lesser General Public License for more details.
+
+You should have received a copy of the GNU Lesser General Public
+License along with this library; if not, write to the Free Software
+Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301 USA
+
+
+This software uses code of SoX licensed under the terms of LGPLv2.1
+Copyright (C) robs@users.sourceforge.net
+Copyright (C) Chris Bagwell and SoX contributors
+Copyright (C) Reuben Thomas
+
+
+This software uses code of FFmpeg licensed under the terms of LGPLv2.1
+Copyright (C) Fabrice Bellard
+Copyright (C) Michael Niedermayer <michaelni@gmx.at>
+Copyright (C) Alex Converse <alex dot converse at gmail dot com>
+Copyright (C) Loren Merritt
+Copyright (C) Vitor Sessak
+Copyright (C) x264 project
+
+
+This software uses code of General Purpose FFT Package 
+Copyright (C) Takuya OOURA
+
+
+This software uses code of foobar2000 1.4 SDK
+Copyright (C) 2001-2018, Peter Pawlowski

Audio/ThirdParty/lvqcl/lpc.c

@@ -0,0 +1,193 @@
+/*
+ * Copyright (c) 2013, 2018 lvqcl
+ *
+ * Permission to use, copy, modify, and distribute this software for any
+ * purpose with or without fee is hereby granted, provided that the above
+ * copyright notice and this permission notice appear in all copies.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
+ * WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
+ * ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
+ * WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
+ * ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
+ * OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
+ */
+
+#include <memory.h>
+#include <stdlib.h>
+#include <stdbool.h>
+#include "lpc.h"
+
+static void apply_window(float *const data, const size_t data_len) {
+#if 0
+	if (0) // subtract the mean
+	{
+		double mean = 0;
+		for(int i = 0; i < (int)data_len; i++)
+			mean += data[i];
+		mean /= data_len;
+
+		for(int i = 0; i < (int)data_len; i++)
+			data[i] -= (float)mean;
+	}
+#endif
+
+	if(1) // Welch window
+	{
+		const float n2 = (data_len + 1) / 2.0f;
+		for(int i = 0; i < (int)data_len; i++) {
+			float k = (i + 1 - n2) / n2;
+			data[data_len - 1 - i] *= 1.0f - k * k;
+		}
+	}
+}
+
+static float vorbis_lpc_from_data(float *data, float *lpci, int n, int m, double *aut, double *lpc) {
+	double error;
+	double epsilon;
+	int i, j;
+
+	/* autocorrelation, p+1 lag coefficients */
+	j = m + 1;
+	while(j--) {
+		double d = 0; /* double needed for accumulator depth */
+		for(i = j; i < n; i++) d += (double)data[i] * data[i - j];
+		aut[j] = d;
+	}
+
+	/* Generate lpc coefficients from autocorr values */
+
+	/* set our noise floor to about -100dB */
+	error = aut[0] * (1. + 1e-10);
+	epsilon = 1e-9 * aut[0] + 1e-10;
+
+	for(i = 0; i < m; i++) {
+		double r = -aut[i + 1];
+
+		if(error < epsilon) {
+			memset(lpc + i, 0, (m - i) * sizeof(*lpc));
+			goto done;
+		}
+
+		/* Sum up this iteration's reflection coefficient; note that in
+		   Vorbis we don't save it.  If anyone wants to recycle this code
+		   and needs reflection coefficients, save the results of 'r' from
+		   each iteration. */
+
+		for(j = 0; j < i; j++) r -= lpc[j] * aut[i - j];
+		r /= error;
+
+		/* Update LPC coefficients and total error */
+
+		lpc[i] = r;
+		for(j = 0; j < i / 2; j++) {
+			double tmp = lpc[j];
+
+			lpc[j] += r * lpc[i - 1 - j];
+			lpc[i - 1 - j] += r * tmp;
+		}
+		if(i & 1) lpc[j] += lpc[j] * r;
+
+		error *= 1. - r * r;
+	}
+
+done:
+
+	/* slightly damp the filter */
+	{
+		double g = .99;
+		double damp = g;
+		for(j = 0; j < m; j++) {
+			lpc[j] *= damp;
+			damp *= g;
+		}
+	}
+
+	for(j = 0; j < m; j++) lpci[j] = (float)lpc[j];
+
+	/* we need the error value to know how big an impulse to hit the
+	   filter with later */
+
+	return error;
+}
+
+static void vorbis_lpc_predict(float *coeff, float *prime, int m, float *data, long n, float *work) {
+	/* in: coeff[0...m-1] LPC coefficients
+	       prime[0...m-1] initial values (allocated size of n+m-1)
+	  out: data[0...n-1] data samples */
+
+	long i, j, o, p;
+	float y;
+
+	if(!prime)
+		for(i = 0; i < m; i++)
+			work[i] = 0.f;
+	else
+		for(i = 0; i < m; i++)
+			work[i] = prime[i];
+
+	for(i = 0; i < n; i++) {
+		y = 0;
+		o = i;
+		p = m;
+		for(j = 0; j < m; j++)
+			y -= work[o++] * coeff[--p];
+
+		data[i] = work[o] = y;
+	}
+}
+
+void lpc_extrapolate2(float *const data, const size_t data_len, const int nch, const int lpc_order, const size_t extra_bkwd, const size_t extra_fwd, void **extrapolate_buffer, size_t *extrapolate_buffer_size) {
+	const size_t tdata_size = sizeof(float) * (extra_bkwd + data_len + extra_fwd);
+	const size_t aut_size = sizeof(double) * (lpc_order + 1);
+	const size_t lpc_size = sizeof(double) * lpc_order;
+	const size_t lpci_size = sizeof(float) * lpc_order;
+	const size_t work_size = sizeof(float) * (extra_bkwd + lpc_order + extra_fwd);
+
+	const size_t new_size = tdata_size + aut_size + lpc_size + lpci_size + work_size;
+
+	if(new_size > *extrapolate_buffer_size) {
+		*extrapolate_buffer = realloc(*extrapolate_buffer, new_size);
+		*extrapolate_buffer_size = new_size;
+	}
+
+	float *tdata = (float *)(*extrapolate_buffer); // for 1 channel only
+
+	double *aut = (double *)(*extrapolate_buffer + tdata_size);
+	double *lpc = (double *)(*extrapolate_buffer + tdata_size + aut_size);
+	float *lpci = (float *)(*extrapolate_buffer + tdata_size + aut_size + lpc_size);
+	float *work = (float *)(*extrapolate_buffer + tdata_size + aut_size + lpc_size + lpci_size);
+
+	for(int c = 0; c < nch; c++) {
+		if(extra_bkwd) {
+			for(int i = 0; i < (int)data_len; i++)
+				tdata[data_len - 1 - i] = data[i * nch + c];
+		} else {
+			for(int i = 0; i < (int)data_len; i++)
+				tdata[i] = data[i * nch + c];
+		}
+
+		apply_window(tdata, data_len);
+		vorbis_lpc_from_data(tdata, lpci, (int)data_len, lpc_order, aut, lpc);
+
+		// restore after apply_window
+		if(extra_bkwd) {
+			for(int i = 0; i < (int)data_len; i++)
+				tdata[data_len - 1 - i] = data[i * nch + c];
+		} else {
+			for(int i = 0; i < (int)data_len; i++)
+				tdata[i] = data[i * nch + c];
+		}
+
+		vorbis_lpc_predict(lpci, tdata + data_len - lpc_order, lpc_order, tdata + data_len, extra_fwd + extra_bkwd, work);
+
+		if(extra_bkwd) {
+			for(int i = 0; i < extra_bkwd; i++)
+				data[(-i - 1) * nch + c] = tdata[data_len + i];
+		} else {
+			for(int i = 0; i < extra_fwd; i++)
+				data[(i + data_len) * nch + c] = tdata[data_len + i];
+		}
+	}
+}

Audio/ThirdParty/lvqcl/lpc.h

@@ -0,0 +1,40 @@
+#ifndef MY_LPC_H
+#define MY_LPC_H
+
+/*    data        - beginning of the data
+ *    data_len    - length of data (in samples) that are base for extrapolation
+ *    nch            - number of (interleaved) channels
+ *    lpc_order    - LPC order
+ *    extra_bkwd    - number of samples to pre-extrapolate
+ *    extra_fwd    - number of samples to post-extrapolate
+ *
+ *    D = data; N = num_channels; LEN = data_len*N; EX = extra*N
+ *
+ *    memory layout when invdir == false:
+ *
+ *    [||||||||||||||||||||||||||||||||][||||||||||||||||||||][
+ *    ^ D[0]                            ^ D[LEN]              ^ D[LEN+EX]
+ *
+ *    memory layout when invdir == true:
+ *    ][||||||||||||||||||||][||||||||||||||||||||||||||||||||][
+ *                           ^ D[0]                            ^ D[LEN]
+ *    ^ D[-1*N-EX]          ^ D[-1*N]
+ *
+ */
+
+static const size_t LPC_ORDER = 32;
+
+void lpc_extrapolate2(float * const data, const size_t data_len, const int nch, const int lpc_order, const size_t extra_bkwd, const size_t extra_fwd, void ** extrapolate_buffer, size_t * extrapolate_buffer_size);
+
+static inline void lpc_extrapolate_bkwd(float * const data, const size_t data_len, const size_t prime_len, const int nch, const int lpc_order, const size_t extra_bkwd, void ** extrapolate_buffer, size_t * extrapolate_buffer_size)
+{
+    (void)data_len;
+    lpc_extrapolate2(data, prime_len, nch, lpc_order, extra_bkwd, 0, extrapolate_buffer, extrapolate_buffer_size);
+}
+
+static inline void lpc_extrapolate_fwd(float * const data, const size_t data_len, const size_t prime_len, const int nch, const int lpc_order, const size_t extra_fwd, void ** extrapolate_buffer, size_t * extrapolate_buffer_size)
+{
+    lpc_extrapolate2(data + (data_len - prime_len)*nch, prime_len, nch, lpc_order, 0, extra_fwd, extrapolate_buffer, extrapolate_buffer_size);
+}
+
+#endif

Audio/ThirdParty/lvqcl/util.h

@@ -0,0 +1,55 @@
+/* Copyright (c) 2018 lvqcl. All rights reserved.
+ *
+ * This library is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation; either version 2.1 of the License, or (at
+ * your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU Lesser
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with this library; if not, write to the Free Software Foundation,
+ * Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef UTIL_H_
+#define UTIL_H_
+
+#ifndef min
+#define min(a,b) (((a)<(b))?(a):(b))
+#endif
+#ifndef max
+#define max(a,b) (((a)>(b))?(a):(b))
+#endif
+
+static inline unsigned local_gcd(unsigned a, unsigned b)
+{
+    if (a == 0 || b == 0) return 0;
+    unsigned c = a % b;
+    while (c != 0) { a = b; b = c; c = a % b; }
+    return b;
+}
+
+/*
+    In:  *r1 and *r2: samplerates;
+    Out: *r1 and *r2: numbers of samples;
+
+    multiply r1 and r2 by n so that durations is 1/N th of second;
+    limit n so that r1 and r2 aren't bigger than M
+*/
+static void samples_len(unsigned* r1, unsigned* r2, unsigned N, unsigned M)        // example: r1 = 44100, r2 = 48000, N = 20, M = 8192
+{
+    if (r1 == 0 || r2 == 0) return;
+    unsigned v = local_gcd(*r1, *r2);        // v = 300
+    *r1 /= v; *r2 /= v;                        // r1 = 147; r2 = 160  == 1/300th of second
+    unsigned n = (v + N-1) / N;                // n = 300/20 = 15 times
+    unsigned z = max(*r1, *r2);        // z = 160
+    if (z*n > M) n = M / z;                    // 160*15 = 2400 < 8192;; if M == 1024: n = 1024/160 = 6; 160*6 = 960
+    if (n < 1) n = 1;
+    *r1 *= n; *r2 *= n;                        // r1 = 147*15 = 2205 samples, r2 = 160*15 = 2400 samples
+}
+
+#endif