Do not reset output sample rate automatically

This was buggy as hell, and resulted in errors. Now the user should
restart playback if they change output device formats.

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

Audio/Chain/BufferChain.h

@@ -16,9 +16,6 @@
 	InputNode *inputNode;
 	ConverterNode *converterNode;
 
-	AudioStreamBasicDescription inputFormat;
-	uint32_t inputChannelConfig;
-
 	NSURL *streamURL;
 	id userInfo;
 	NSDictionary *rgInfo;

Audio/Chain/BufferChain.m

@@ -60,7 +60,8 @@
 
 	NSDictionary *properties = [inputNode properties];
 
-	inputFormat = [inputNode nodeFormat];
+	AudioStreamBasicDescription inputFormat = [inputNode nodeFormat];
+	uint32_t inputChannelConfig = 0;
 	if([properties valueForKey:@"channelConfig"])
 		inputChannelConfig = [[properties valueForKey:@"channelConfig"] unsignedIntValue];
 
@@ -89,7 +90,8 @@
 
 	NSDictionary *properties = [inputNode properties];
 
-	inputFormat = [inputNode nodeFormat];
+	AudioStreamBasicDescription inputFormat = [inputNode nodeFormat];
+	uint32_t inputChannelConfig = 0;
 	if([properties valueForKey:@"channelConfig"])
 		inputChannelConfig = [[properties valueForKey:@"channelConfig"] unsignedIntValue];
 
@@ -123,7 +125,8 @@
 
 	DLog(@"Input Properties: %@", properties);
 
-	inputFormat = [inputNode nodeFormat];
+	AudioStreamBasicDescription inputFormat = [inputNode nodeFormat];
+	uint32_t inputChannelConfig = 0;
 	if([properties valueForKey:@"channelConfig"])
 		inputChannelConfig = [[properties valueForKey:@"channelConfig"] unsignedIntValue];
 
@@ -231,11 +234,11 @@
 }
 
 - (AudioStreamBasicDescription)inputFormat {
-	return inputFormat;
+	return [inputNode nodeFormat];
 }
 
 - (uint32_t)inputConfig {
-	return inputChannelConfig;
+	return [inputNode nodeChannelConfig];
 }
 
 - (double)secondsBuffered {

Audio/Chain/ChunkList.m

@@ -50,7 +50,7 @@
 }
 
 - (BOOL)isFull {
-	return (maxDuration - listDuration) < 0.01;
+	return (maxDuration - listDuration) < 0.05;
 }
 
 - (void)addChunk:(AudioChunk *)chunk {

Audio/Chain/ConverterNode.h

@@ -15,7 +15,6 @@
 #import <soxr.h>
 
 #import "Node.h"
-#import "RefillNode.h"
 
 #import "HeadphoneFilter.h"
 
@@ -31,7 +30,6 @@
 	BOOL stopping;
 	BOOL convertEntered;
 	BOOL paused;
-	BOOL outputFormatChanged;
 
 	BOOL skipResampler;
 
@@ -77,13 +75,6 @@
 
 	AudioChunk *lastChunkIn;
 
-	AudioStreamBasicDescription previousOutputFormat;
-	uint32_t previousOutputConfig;
-	AudioStreamBasicDescription rememberedInputFormat;
-	uint32_t rememberedInputConfig;
-	RefillNode *refillNode;
-	id __weak originalPreviousNode;
-
 	void *hdcd_decoder;
 
 	HeadphoneFilter *hFilter;

Audio/Chain/ConverterNode.m

@@ -53,13 +53,9 @@ void PrintStreamDesc(AudioStreamBasicDescription *inDesc) {
 		stopping = NO;
 		convertEntered = NO;
 		paused = NO;
-		outputFormatChanged = NO;
 
 		skipResampler = YES;
 
-		refillNode = nil;
-		originalPreviousNode = nil;
-
 		extrapolateBuffer = NULL;
 		extrapolateBufferSize = 0;
 
@@ -68,6 +64,8 @@ void PrintStreamDesc(AudioStreamBasicDescription *inDesc) {
 
 		hdcd_decoder = NULL;
 
+		lastChunkIn = nil;
+
 		[[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.volumeScaling" options:0 context:nil];
 	}
 
@@ -423,22 +421,14 @@ static void convert_be_to_le(uint8_t *buffer, size_t bitsPerSample, size_t bytes
 	// and returns 0 samples when it has nothing more to process at the end of stream.
 	while([self shouldContinue] == YES) {
 		int amountConverted;
+		while(paused) {
+			usleep(500);
+		}
 		@autoreleasepool {
 			amountConverted = [self convert:writeBuf amount:CHUNK_SIZE];
 		}
 		if(!amountConverted) {
 			if(paused) {
-				while(paused)
-					usleep(500);
-				continue;
-			} else if(refillNode) {
-				// refill node just ended, file resumes
-				[self setPreviousNode:originalPreviousNode];
-				[self setEndOfStream:NO];
-				[self setShouldContinue:YES];
-				refillNode = nil;
-				[self cleanUp];
-				[self setupWithInputFormat:rememberedInputFormat withInputConfig:rememberedInputConfig outputFormat:outputFormat outputConfig:outputChannelConfig isLossless:rememberedLossless];
 				continue;
 			} else if(streamFormatChanged) {
 				[self cleanUp];
@@ -503,7 +493,7 @@ tryagain:
 
 			ssize_t bytesReadFromInput = 0;
 
-			while(bytesReadFromInput < amountToWrite && !stopping && !streamFormatChanged && [self shouldContinue] == YES && [self endOfStream] == NO) {
+			while(bytesReadFromInput < amountToWrite && !stopping && !paused && !streamFormatChanged && [self shouldContinue] == YES && [self endOfStream] == NO) {
 				AudioStreamBasicDescription inf;
 				uint32_t config;
 				if([self peekFormat:&inf channelConfig:&config]) {
@@ -536,16 +526,13 @@ tryagain:
 				}
 				bytesReadFromInput += bytesRead;
 				if(!frameCount) {
-					if(refillNode)
+					usleep(500);
-						[self setEndOfStream:YES];
-					else
-						usleep(500);
 				}
 			}
 
 			// Pad end of track with input format silence
 
-			if(stopping || streamFormatChanged || [self shouldContinue] == NO || [self endOfStream] == YES) {
+			if(stopping || paused || streamFormatChanged || [self shouldContinue] == NO || [self endOfStream] == YES) {
 				if(!skipResampler && !is_postextrapolated_) {
 					if(dsd2pcm) {
 						amountToSkip = dsd2pcmLatency * inputFormat.mBytesPerPacket;
@@ -989,9 +976,8 @@ static float db_to_scale(float db) {
 	// Move this here so process call isn't running the resampler until it's allocated
 	stopping = NO;
 	convertEntered = NO;
-	paused = NO;
-	outputFormatChanged = NO;
 	streamFormatChanged = NO;
+	paused = NO;
 
 	return YES;
 }
@@ -1007,46 +993,18 @@ static float db_to_scale(float db) {
 
 - (void)setOutputFormat:(AudioStreamBasicDescription)format outputConfig:(uint32_t)outputConfig {
 	DLog(@"SETTING OUTPUT FORMAT!");
-	previousOutputFormat = outputFormat;
-	previousOutputConfig = outputChannelConfig;
 	outputFormat = format;
 	outputChannelConfig = outputConfig;
-	outputFormatChanged = YES;
 }
 
 - (void)inputFormatDidChange:(AudioStreamBasicDescription)format inputConfig:(uint32_t)inputConfig {
 	DLog(@"FORMAT CHANGED");
 	paused = YES;
-	[self cleanUp];
+	while(convertEntered) {
-	if(outputFormatChanged && ![buffer isEmpty] &&
+		usleep(500);
-	   (outputChannelConfig != previousOutputConfig ||
-	    memcmp(&outputFormat, &previousOutputFormat, sizeof(outputFormat)) != 0)) {
-		// Transfer previously buffered data, remember input format
-		rememberedInputFormat = format;
-		rememberedInputConfig = inputChannelConfig;
-		originalPreviousNode = previousNode;
-		refillNode = [[RefillNode alloc] initWithController:controller previous:nil];
-		[self setPreviousNode:refillNode];
-
-		[refillNode setFormat:previousOutputFormat];
-		[refillNode setChannelConfig:previousOutputConfig];
-
-		for(;;) {
-			@autoreleasepool {
-				AudioChunk *chunk = [buffer removeSamples:16384];
-				size_t frameCount = [chunk frameCount];
-				if(frameCount) {
-					NSData *samples = [chunk removeSamples:frameCount];
-					[refillNode writeData:[samples bytes] amount:frameCount];
-				} else
-					break;
-			}
-		}
-
-		[self setupWithInputFormat:previousOutputFormat withInputConfig:[AudioChunk guessChannelConfig:previousOutputFormat.mChannelsPerFrame] outputFormat:outputFormat outputConfig:outputChannelConfig isLossless:rememberedLossless];
-	} else {
-		[self setupWithInputFormat:format withInputConfig:inputConfig outputFormat:outputFormat outputConfig:outputChannelConfig isLossless:rememberedLossless];
 	}
+	[self cleanUp];
+	[self setupWithInputFormat:format withInputConfig:inputConfig outputFormat:outputFormat outputConfig:outputChannelConfig isLossless:rememberedLossless];
 }
 
 - (void)setRGInfo:(NSDictionary *)rgi {

Audio/Chain/Downmix.m

@@ -147,13 +147,12 @@ static void downmix_to_mono(const float *inBuffer, int channels, uint32_t config
 	inBuffer = tempBuffer;
 	channels = 2;
 	config = AudioConfigStereo;
-	float invchannels = 1.0 / (float)channels;
 	for(size_t i = 0; i < count; ++i) {
 		float sample = 0;
 		for(int j = 0; j < channels; ++j) {
 			sample += inBuffer[i * channels + j];
 		}
-		outBuffer[i] = sample * invchannels;
+		outBuffer[i] = sample;
 	}
 }
 

Audio/Chain/OutputNode.h

@@ -20,7 +20,6 @@
 	uint32_t config;
 
 	double amountPlayed;
-	double sampleRatio;
 	OutputCoreAudio *output;
 
 	BOOL paused;

Audio/Chain/OutputNode.m

@@ -17,7 +17,6 @@
 
 - (void)setup {
 	amountPlayed = 0.0;
-	sampleRatio = 0.0;
 
 	paused = YES;
 	started = NO;
@@ -100,7 +99,6 @@
 	config = channelConfig;
 	// Calculate a ratio and add to double(seconds) instead, as format may change
 	// double oldSampleRatio = sampleRatio;
-	sampleRatio = 1.0 / (format.mSampleRate * format.mBytesPerPacket);
 	BufferChain *bufferChain = [controller bufferChain];
 	if(bufferChain) {
 		ConverterNode *converter = [bufferChain converter];
@@ -108,13 +106,16 @@
 			// This clears the resampler buffer, but not the input buffer
 			// We also have to jump the play position ahead accounting for
 			// the data we are flushing
-#if 0
+			amountPlayed += [[converter buffer] listDuration];
-            // We no longer need to do this, because outputchanged converter
+
-            // now uses the RefillNode to slap the previous samples into
+			AudioStreamBasicDescription inf = [bufferChain inputFormat];
-            // itself
+			uint32_t config = [bufferChain inputConfig];
-            if (oldSampleRatio)
+
-                amountPlayed += oldSampleRatio * [[converter buffer] bufferedLength];
+			format.mChannelsPerFrame = inf.mChannelsPerFrame;
-#endif
+			format.mBytesPerFrame = ((inf.mBitsPerChannel + 7) / 8) * format.mChannelsPerFrame;
+			format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket;
+			channelConfig = config;
+
 			[converter setOutputFormat:format
 			              outputConfig:channelConfig];
 			[converter inputFormatDidChange:[bufferChain inputFormat] inputConfig:[bufferChain inputConfig]];

Audio/Chain/RefillNode.h

@@ -1,27 +0,0 @@
-//
-//  RefillNode.h
-//  Cog
-//
-//  Created by Christopher SNowhill on 1/13/22.
-//  Copyright 2022 __LoSnoCo__. All rights reserved.
-//
-
-#import <Cocoa/Cocoa.h>
-
-#import <AudioToolbox/AudioToolbox.h>
-#import <AudioUnit/AudioUnit.h>
-#import <CoreAudio/AudioHardware.h>
-
-#import "Node.h"
-#import "Plugin.h"
-
-#define INPUT_NODE_SEEK
-
-@interface RefillNode : Node {
-	// This node just slaps pre-converted data into its buffer for re-buffering
-}
-
-- (void)setFormat:(AudioStreamBasicDescription)format;
-- (void)setChannelConfig:(uint32_t)config;
-
-@end

Audio/Chain/RefillNode.m

@@ -1,49 +0,0 @@
-//
-//  RefillNode.m
-//  Cog
-//
-//  Created by Christopher Snowhill on 1/13/22.
-//  Copyright 2022 __LoSnoCo__. All rights reserved.
-//
-
-#import "RefillNode.h"
-#import "Plugin.h"
-
-#import "Logging.h"
-
-@implementation RefillNode
-
-- (id)initWithController:(id)c previous:(id)p {
-	self = [super init];
-	if(self) {
-		// This special node should be able to handle up to four buffers
-		buffer = [[ChunkList alloc] initWithMaximumDuration:12.0];
-		semaphore = [[Semaphore alloc] init];
-
-		initialBufferFilled = NO;
-
-		controller = c;
-		endOfStream = NO;
-		shouldContinue = YES;
-
-		nodeLossless = NO;
-
-		[self setPreviousNode:p];
-	}
-
-	return self;
-}
-
-- (void)dealloc {
-	DLog(@"Refill Node dealloc");
-}
-
-- (void)setFormat:(AudioStreamBasicDescription)format {
-	nodeFormat = format;
-}
-
-- (void)setChannelConfig:(uint32_t)config {
-	nodeChannelConfig = config;
-}
-
-@end

Audio/CogAudio.xcodeproj/project.pbxproj

@@ -66,8 +66,6 @@
 		8384912718080FF100E7332D /* Logging.h in Headers */ = {isa = PBXBuildFile; fileRef = 8384912618080FF100E7332D /* Logging.h */; };
 		839366671815923C006DD712 /* CogPluginMulti.h in Headers */ = {isa = PBXBuildFile; fileRef = 839366651815923C006DD712 /* CogPluginMulti.h */; };
 		839366681815923C006DD712 /* CogPluginMulti.m in Sources */ = {isa = PBXBuildFile; fileRef = 839366661815923C006DD712 /* CogPluginMulti.m */; };
-		83A44A01279119B50049B6E2 /* RefillNode.m in Sources */ = {isa = PBXBuildFile; fileRef = 83A449FF279119B50049B6E2 /* RefillNode.m */; };
-		83A44A02279119B50049B6E2 /* RefillNode.h in Headers */ = {isa = PBXBuildFile; fileRef = 83A44A00279119B50049B6E2 /* RefillNode.h */; };
 		8DC2EF570486A6940098B216 /* Cocoa.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 1058C7B1FEA5585E11CA2CBB /* Cocoa.framework */; };
 		8E8D3D2F0CBAEE6E00135C1B /* AudioContainer.h in Headers */ = {isa = PBXBuildFile; fileRef = 8E8D3D2D0CBAEE6E00135C1B /* AudioContainer.h */; settings = {ATTRIBUTES = (Public, ); }; };
 		8E8D3D300CBAEE6E00135C1B /* AudioContainer.m in Sources */ = {isa = PBXBuildFile; fileRef = 8E8D3D2E0CBAEE6E00135C1B /* AudioContainer.m */; };
@@ -162,8 +160,6 @@
 		8384912618080FF100E7332D /* Logging.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = Logging.h; path = ../../Utils/Logging.h; sourceTree = "<group>"; };
 		839366651815923C006DD712 /* CogPluginMulti.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CogPluginMulti.h; sourceTree = "<group>"; };
 		839366661815923C006DD712 /* CogPluginMulti.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; path = CogPluginMulti.m; sourceTree = "<group>"; };
-		83A449FF279119B50049B6E2 /* RefillNode.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; path = RefillNode.m; sourceTree = "<group>"; };
-		83A44A00279119B50049B6E2 /* RefillNode.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RefillNode.h; sourceTree = "<group>"; };
 		8DC2EF5A0486A6940098B216 /* Info.plist */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text.plist; path = Info.plist; sourceTree = "<group>"; };
 		8DC2EF5B0486A6940098B216 /* CogAudio.framework */ = {isa = PBXFileReference; explicitFileType = wrapper.framework; includeInIndex = 0; path = CogAudio.framework; sourceTree = BUILT_PRODUCTS_DIR; };
 		8E8D3D2D0CBAEE6E00135C1B /* AudioContainer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = AudioContainer.h; sourceTree = "<group>"; };
@@ -298,8 +294,6 @@
 				834FD4EF27AF93680063BC83 /* ChunkList.m */,
 				834FD4F227AFA2150063BC83 /* Downmix.h */,
 				834FD4F327AFA2150063BC83 /* Downmix.m */,
-				83A44A00279119B50049B6E2 /* RefillNode.h */,
-				83A449FF279119B50049B6E2 /* RefillNode.m */,
 				17D21C760B8BE4BA00D1EBDE /* BufferChain.h */,
 				17D21C770B8BE4BA00D1EBDE /* BufferChain.m */,
 				8EC1225D0B993BD500C5B3AD /* ConverterNode.h */,
@@ -457,7 +451,6 @@
 				839366671815923C006DD712 /* CogPluginMulti.h in Headers */,
 				17ADB13C0B97926D00257CA2 /* AudioSource.h in Headers */,
 				835C88B1279811A500E28EAE /* hdcd_decode2.h in Headers */,
-				83A44A02279119B50049B6E2 /* RefillNode.h in Headers */,
 				8EC1225F0B993BD500C5B3AD /* ConverterNode.h in Headers */,
 				8384912718080FF100E7332D /* Logging.h in Headers */,
 				8E8D3D2F0CBAEE6E00135C1B /* AudioContainer.h in Headers */,
@@ -560,7 +553,6 @@
 				17C940240B900909008627D6 /* AudioMetadataReader.m in Sources */,
 				17B619310B909BC300BC003F /* AudioPropertiesReader.m in Sources */,
 				17ADB13D0B97926D00257CA2 /* AudioSource.m in Sources */,
-				83A44A01279119B50049B6E2 /* RefillNode.m in Sources */,
 				834FD4F127AF93680063BC83 /* ChunkList.m in Sources */,
 				8EC122600B993BD500C5B3AD /* ConverterNode.m in Sources */,
 				8E8D3D300CBAEE6E00135C1B /* AudioContainer.m in Sources */,

Audio/Output/OutputCoreAudio.m

@@ -211,16 +211,10 @@ default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const
 	running = YES;
 	started = NO;
 	stopNext = NO;
-	size_t eventCount = 0;
 	atomic_store(&bytesRendered, 0);
 	NSMutableArray *delayedEvents = [[NSMutableArray alloc] init];
 	BOOL delayedEventsPopped = YES;
 	while(!stopping) {
-		if(++eventCount == 128) {
-			[self updateDeviceFormat];
-			eventCount = 0;
-		}
-
 		if([outputController shouldReset]) {
 			@autoreleasepool {
 				[[outputController buffer] reset];
@@ -671,7 +665,7 @@ default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const
 #endif
 
 		inputData->mBuffers[0].mNumberChannels = channels;
-
+		
 		return 0;
 	};