Revert "Core Audio output: Rewrote major portions"

This reverts commit 637ea4efe1.

Audio/AudioPlayer.m

@@ -93,7 +93,7 @@
 	bufferChain = [[BufferChain alloc] initWithController:self];
 	[self notifyStreamChanged:userInfo];
     
-	while (![bufferChain open:url withOutputFormatHint:[output format] withRGInfo:rgi])
+	while (![bufferChain open:url withOutputFormat:[output format] withRGInfo:rgi])
 	{
 		bufferChain = nil;
 		
@@ -401,7 +401,7 @@
 			&& [[nextStream path] isEqualToString:[[lastChain streamURL] path]]))
 		{
 			if ([lastChain setTrack:nextStream] 
-				&& [newChain openWithInput:[lastChain inputNode] withOutputFormatHint:[output format] withRGInfo:nextStreamRGInfo])
+				&& [newChain openWithInput:[lastChain inputNode] withOutputFormat:[output format] withRGInfo:nextStreamRGInfo])
 			{
 				[newChain setStreamURL:nextStream];
 				[newChain setUserInfo:nextStreamUserInfo];
@@ -418,7 +418,7 @@
         
         lastChain = nil;
 		
-		while (shouldContinue && ![newChain open:nextStream withOutputFormatHint:[output format] withRGInfo:nextStreamRGInfo])
+		while (shouldContinue && ![newChain open:nextStream withOutputFormat:[output format] withRGInfo:nextStreamRGInfo])
 		{
 			if (nextStream == nil)
 			{

Audio/Chain/BufferChain.h

@@ -30,14 +30,14 @@
 - (id)initWithController:(id)c;
 - (void)buildChain;
 
-- (BOOL)open:(NSURL *)url withOutputFormatHint:(AudioStreamBasicDescription)outputFormat withRGInfo:(NSDictionary*)rgi;
+- (BOOL)open:(NSURL *)url withOutputFormat:(AudioStreamBasicDescription)outputFormat withRGInfo:(NSDictionary*)rgi;
 
 //Used when changing tracks to reuse the same decoder
-- (BOOL)openWithInput:(InputNode *)i  withOutputFormatHint:(AudioStreamBasicDescription)outputFormat withRGInfo:(NSDictionary*)rgi;
+- (BOOL)openWithInput:(InputNode *)i  withOutputFormat:(AudioStreamBasicDescription)outputFormat withRGInfo:(NSDictionary*)rgi;
 
 //Used when resetting the decoder on seek
 - (BOOL)openWithDecoder:(id<CogDecoder>)decoder
-    withOutputFormatHint:(AudioStreamBasicDescription)outputFormat
+    withOutputFormat:(AudioStreamBasicDescription)outputFormat
     withRGInfo:(NSDictionary*)rgi;
 
 - (void)seek:(double)time;

Audio/Chain/BufferChain.m

@@ -43,7 +43,7 @@
 	finalNode = converterNode;
 }
 
-- (BOOL)open:(NSURL *)url withOutputFormatHint:(AudioStreamBasicDescription)outputFormat withRGInfo:(NSDictionary *)rgi
+- (BOOL)open:(NSURL *)url withOutputFormat:(AudioStreamBasicDescription)outputFormat withRGInfo:(NSDictionary *)rgi
 {	
 	[self setStreamURL:url];
 
@@ -65,13 +65,7 @@
     
     NSDictionary * properties = [inputNode properties];
 
-    inputFormat = [inputNode nodeFormat];
+    if (![converterNode setupWithInputFormat:(inputFormat = propertiesToASBD(properties)) outputFormat:outputFormat 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 outputFormat:outputFormat isLossless:[[properties valueForKey:@"encoding"] isEqualToString:@"lossless"]])
 		return NO;
 
     [self setRGInfo:rgi];
@@ -81,7 +75,7 @@
 	return YES;
 }
 
-- (BOOL)openWithInput:(InputNode *)i withOutputFormatHint:(AudioStreamBasicDescription)outputFormat withRGInfo:(NSDictionary *)rgi
+- (BOOL)openWithInput:(InputNode *)i withOutputFormat:(AudioStreamBasicDescription)outputFormat withRGInfo:(NSDictionary *)rgi
 {
 	DLog(@"New buffer chain!");
 	[self buildChain];
@@ -92,14 +86,7 @@
     NSDictionary * properties = [inputNode properties];
     
 	DLog(@"Input Properties: %@", properties);
-
+    if (![converterNode setupWithInputFormat:(inputFormat = propertiesToASBD(properties)) outputFormat:outputFormat isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
-    inputFormat = [inputNode nodeFormat];
-    
-    outputFormat.mChannelsPerFrame = inputFormat.mChannelsPerFrame;
-    outputFormat.mBytesPerFrame = ((outputFormat.mBitsPerChannel + 7) / 8) * outputFormat.mChannelsPerFrame;
-    outputFormat.mBytesPerPacket = outputFormat.mBytesPerFrame * outputFormat.mFramesPerPacket;
-
-    if (![converterNode setupWithInputFormat:inputFormat outputFormat:outputFormat isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
 		return NO;
 
     [self setRGInfo:rgi];
@@ -108,7 +95,7 @@
 }
 
 - (BOOL)openWithDecoder:(id<CogDecoder>)decoder
-    withOutputFormatHint:(AudioStreamBasicDescription)outputFormat
+    withOutputFormat:(AudioStreamBasicDescription)outputFormat
     withRGInfo:(NSDictionary*)rgi;
 {
     DLog(@"New buffer chain!");
@@ -120,14 +107,7 @@
     NSDictionary * properties = [inputNode properties];
     
     DLog(@"Input Properties: %@", properties);
-
+    if (![converterNode setupWithInputFormat:(inputFormat = propertiesToASBD(properties)) outputFormat:outputFormat isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
-    inputFormat = [inputNode nodeFormat];
-    
-    outputFormat.mChannelsPerFrame = inputFormat.mChannelsPerFrame;
-    outputFormat.mBytesPerFrame = ((outputFormat.mBitsPerChannel + 7) / 8) * outputFormat.mChannelsPerFrame;
-    outputFormat.mBytesPerPacket = outputFormat.mBytesPerFrame * outputFormat.mFramesPerPacket;
-
-    if (![converterNode setupWithInputFormat:inputFormat outputFormat:outputFormat isLossless:[[properties objectForKey:@"encoding"] isEqualToString:@"lossless"]])
         return NO;
     
     [self setRGInfo:rgi];

Audio/Chain/ConverterNode.h

@@ -74,6 +74,8 @@
     id __weak originalPreviousNode;
     
     void *hdcd_decoder;
+    
+    HeadphoneFilter *hFilter;
 }
 
 @property AudioStreamBasicDescription inputFormat;

Audio/Chain/ConverterNode.m

@@ -72,11 +72,189 @@ void PrintStreamDesc (AudioStreamBasicDescription *inDesc)
         hdcd_decoder = NULL;
 
         [[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.volumeScaling"		options:0 context:nil];
+        [[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.headphoneVirtualization" options:0 context:nil];
+        [[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.hrirPath" options:0 context:nil];
     }
     
     return self;
 }
 
+static const float STEREO_DOWNMIX[8-2][8][2]={
+    /*3.0*/
+    {
+        {0.5858F,0.0F},{0.0F,0.5858F},{0.4142F,0.4142F}
+    },
+    /*quadrophonic*/
+    {
+        {0.4226F,0.0F},{0.0F,0.4226F},{0.366F,0.2114F},{0.2114F,0.336F}
+    },
+    /*5.0*/
+    {
+        {0.651F,0.0F},{0.0F,0.651F},{0.46F,0.46F},{0.5636F,0.3254F},
+        {0.3254F,0.5636F}
+    },
+    /*5.1*/
+    {
+        {0.529F,0.0F},{0.0F,0.529F},{0.3741F,0.3741F},{0.3741F,0.3741F},{0.4582F,0.2645F},
+        {0.2645F,0.4582F}
+    },
+    /*6.1*/
+    {
+        {0.4553F,0.0F},{0.0F,0.4553F},{0.322F,0.322F},{0.322F,0.322F},{0.2788F,0.2788F},
+        {0.3943F,0.2277F},{0.2277F,0.3943F}
+    },
+    /*7.1*/
+    {
+        {0.3886F,0.0F},{0.0F,0.3886F},{0.2748F,0.2748F},{0.2748F,0.2748F},{0.3366F,0.1943F},
+        {0.1943F,0.3366F},{0.3366F,0.1943F},{0.1943F,0.3366F}
+    }
+};
+
+static void downmix_to_stereo(float * buffer, int channels, size_t count)
+{
+    if (channels >= 3 && channels <= 8)
+    for (size_t i = 0; i < count; ++i)
+    {
+        float left = 0, right = 0;
+        for (int j = 0; j < channels; ++j)
+        {
+            left += buffer[i * channels + j] * STEREO_DOWNMIX[channels - 3][j][0];
+            right += buffer[i * channels + j] * STEREO_DOWNMIX[channels - 3][j][1];
+        }
+        buffer[i * 2 + 0] = left;
+        buffer[i * 2 + 1] = right;
+    }
+}
+
+static void downmix_to_mono(float * buffer, int channels, size_t count)
+{
+    if (channels >= 3 && channels <= 8)
+    {
+        downmix_to_stereo(buffer, channels, count);
+        channels = 2;
+    }
+    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 += buffer[i * channels + j];
+        }
+        buffer[i] = sample * invchannels;
+    }
+}
+
+static void upmix(float * buffer, int inchannels, int outchannels, size_t count)
+{
+    for (ssize_t i = count - 1; i >= 0; --i)
+    {
+        if (inchannels == 1 && outchannels == 2)
+        {
+            // upmix mono to stereo
+            float sample = buffer[i];
+            buffer[i * 2 + 0] = sample;
+            buffer[i * 2 + 1] = sample;
+        }
+        else if (inchannels == 1 && outchannels == 4)
+        {
+            // upmix mono to quad
+            float sample = buffer[i];
+            buffer[i * 4 + 0] = sample;
+            buffer[i * 4 + 1] = sample;
+            buffer[i * 4 + 2] = 0;
+            buffer[i * 4 + 3] = 0;
+        }
+        else if (inchannels == 1 && (outchannels == 3 || outchannels >= 5))
+        {
+            // upmix mono to center channel
+            float sample = buffer[i];
+            buffer[i * outchannels + 2] = sample;
+            for (int j = 0; j < 2; ++j)
+            {
+                buffer[i * outchannels + j] = 0;
+            }
+            for (int j = 3; j < outchannels; ++j)
+            {
+                buffer[i * outchannels + j] = 0;
+            }
+        }
+        else if (inchannels == 4 && outchannels >= 5)
+        {
+            float fl = buffer[i * 4 + 0];
+            float fr = buffer[i * 4 + 1];
+            float bl = buffer[i * 4 + 2];
+            float br = buffer[i * 4 + 3];
+            const int skipclfe = (outchannels == 5) ? 1 : 2;
+            buffer[i * outchannels + 0] = fl;
+            buffer[i * outchannels + 1] = fr;
+            buffer[i * outchannels + skipclfe + 2] = bl;
+            buffer[i * outchannels + skipclfe + 3] = br;
+            for (int j = 0; j < skipclfe; ++j)
+            {
+                buffer[i * outchannels + 2 + j] = 0;
+            }
+            for (int j = 4 + skipclfe; j < outchannels; ++j)
+            {
+                buffer[i * outchannels + j] = 0;
+            }
+        }
+        else if (inchannels == 5 && outchannels >= 6)
+        {
+            float fl = buffer[i * 5 + 0];
+            float fr = buffer[i * 5 + 1];
+            float c = buffer[i * 5 + 2];
+            float bl = buffer[i * 5 + 3];
+            float br = buffer[i * 5 + 4];
+            buffer[i * outchannels + 0] = fl;
+            buffer[i * outchannels + 1] = fr;
+            buffer[i * outchannels + 2] = c;
+            buffer[i * outchannels + 3] = 0;
+            buffer[i * outchannels + 4] = bl;
+            buffer[i * outchannels + 5] = br;
+            for (int j = 6; j < outchannels; ++j)
+            {
+                buffer[i * outchannels + j] = 0;
+            }
+        }
+        else if (inchannels == 7 && outchannels == 8)
+        {
+            float fl = buffer[i * 7 + 0];
+            float fr = buffer[i * 7 + 1];
+            float c = buffer[i * 7 + 2];
+            float lfe = buffer[i * 7 + 3];
+            float sl = buffer[i * 7 + 4];
+            float sr = buffer[i * 7 + 5];
+            float bc = buffer[i * 7 + 6];
+            buffer[i * 8 + 0] = fl;
+            buffer[i * 8 + 1] = fr;
+            buffer[i * 8 + 2] = c;
+            buffer[i * 8 + 3] = lfe;
+            buffer[i * 8 + 4] = bc;
+            buffer[i * 8 + 5] = bc;
+            buffer[i * 8 + 6] = sl;
+            buffer[i * 8 + 7] = sr;
+        }
+        else
+        {
+            // upmix N channels to N channels plus silence the empty channels
+            float samples[inchannels];
+            for (int j = 0; j < inchannels; ++j)
+            {
+                samples[j] = buffer[i * inchannels + j];
+            }
+            for (int j = 0; j < inchannels; ++j)
+            {
+                buffer[i * outchannels + j] = samples[j];
+            }
+            for (int j = inchannels; j < outchannels; ++j)
+            {
+                buffer[i * outchannels + j] = 0;
+            }
+        }
+    }
+}
+
 void scale_by_volume(float * buffer, size_t count, float volume)
 {
     if ( volume != 1.0 )
@@ -842,6 +1020,31 @@ tryagain:
         
         scale_by_volume( (float*) floatBuffer, amountReadFromFC / sizeof(float), volumeScale);
         
+        if ( hFilter ) {
+            int samples = amountReadFromFC / floatFormat.mBytesPerFrame;
+            [hFilter process:floatBuffer sampleCount:samples toBuffer:floatBuffer + amountReadFromFC];
+            memmove(floatBuffer, floatBuffer + amountReadFromFC, samples * sizeof(float) * 2);
+            amountReadFromFC = samples * sizeof(float) * 2;
+        }
+        else if ( inputFormat.mChannelsPerFrame > 2 && outputFormat.mChannelsPerFrame == 2 )
+        {
+            int samples = amountReadFromFC / floatFormat.mBytesPerFrame;
+            downmix_to_stereo( (float*) floatBuffer, inputFormat.mChannelsPerFrame, samples );
+            amountReadFromFC = samples * sizeof(float) * 2;
+        }
+        else if ( inputFormat.mChannelsPerFrame > 1 && outputFormat.mChannelsPerFrame == 1 )
+        {
+            int samples = amountReadFromFC / floatFormat.mBytesPerFrame;
+            downmix_to_mono( (float*) floatBuffer, inputFormat.mChannelsPerFrame, samples );
+            amountReadFromFC = samples * sizeof(float);
+        }
+        else if ( inputFormat.mChannelsPerFrame < outputFormat.mChannelsPerFrame )
+        {
+            int samples = amountReadFromFC / floatFormat.mBytesPerFrame;
+            upmix( (float*) floatBuffer, inputFormat.mChannelsPerFrame, outputFormat.mChannelsPerFrame, samples );
+            amountReadFromFC = samples * sizeof(float) * outputFormat.mChannelsPerFrame;
+        }
+        
         floatSize = amountReadFromFC;
         floatOffset = 0;
     }
@@ -872,6 +1075,15 @@ tryagain:
         //User reset the volume scaling option
         [self refreshVolumeScaling];
     }
+    else if ([keyPath isEqualToString:@"values.headphoneVirtualization"] ||
+             [keyPath isEqualToString:@"values.hrirPath"]) {
+        // Reset the converter, without rebuffering
+        if (outputFormat.mChannelsPerFrame == 2 &&
+            inputFormat.mChannelsPerFrame >= 1 &&
+            inputFormat.mChannelsPerFrame <= 8) {
+            [self inputFormatDidChange:inputFormat];
+        }
+    }
 }
 
 static float db_to_scale(float db)
@@ -929,8 +1141,6 @@ static float db_to_scale(float db)
 	inputFormat = inf;
 	outputFormat = outf;
     
-    nodeFormat = outputFormat;
-    
     rememberedLossless = lossless;
     
     // These are the only sample formats we support translating
@@ -982,6 +1192,33 @@ static float db_to_scale(float db)
     dmFloatFormat.mBytesPerFrame = (32/8)*dmFloatFormat.mChannelsPerFrame;
     dmFloatFormat.mBytesPerPacket = dmFloatFormat.mBytesPerFrame * floatFormat.mFramesPerPacket;
     
+    BOOL hVirt = [[[NSUserDefaultsController sharedUserDefaultsController] defaults] boolForKey:@"headphoneVirtualization"];
+    
+    if (hVirt &&
+        outputFormat.mChannelsPerFrame == 2 &&
+        inputFormat.mChannelsPerFrame >= 1 &&
+         inputFormat.mChannelsPerFrame <= 8) {
+        NSString * userPreset = [[[NSUserDefaultsController sharedUserDefaultsController] defaults] stringForKey:@"hrirPath"];
+        
+        NSURL * presetUrl = nil;
+        
+        if (userPreset && ![userPreset isEqualToString:@""]) {
+            presetUrl = [NSURL fileURLWithPath:userPreset];
+            if (![HeadphoneFilter validateImpulseFile:presetUrl])
+                presetUrl = nil;
+        }
+        
+        if (!presetUrl) {
+            presetUrl = [[NSBundle mainBundle] URLForResource:@"gsx" withExtension:@"wv"];
+            if (![HeadphoneFilter validateImpulseFile:presetUrl])
+                presetUrl = nil;
+        }
+
+        if (presetUrl) {
+            hFilter = [[HeadphoneFilter alloc] initWithImpulseFile:presetUrl forSampleRate:outputFormat.mSampleRate withInputChannels:inputFormat.mChannelsPerFrame];
+        }
+    }
+
     skipResampler = outputFormat.mSampleRate == floatFormat.mSampleRate;
     
     sampleRatio = (double)outputFormat.mSampleRate / (double)floatFormat.mSampleRate;
@@ -1034,6 +1271,8 @@ static float db_to_scale(float db)
 	DLog(@"Decoder dealloc");
 
     [[NSUserDefaultsController sharedUserDefaultsController] removeObserver:self forKeyPath:@"values.volumeScaling"];
+    [[NSUserDefaultsController sharedUserDefaultsController] removeObserver:self forKeyPath:@"values.headphoneVirtualization"];
+    [[NSUserDefaultsController sharedUserDefaultsController] removeObserver:self forKeyPath:@"values.hrirPath"];
 
     paused = NO;
 	[self cleanUp];
@@ -1098,6 +1337,9 @@ static float db_to_scale(float db)
     {
         usleep(500);
     }
+    if (hFilter) {
+        hFilter = nil;
+    }
     if (hdcd_decoder)
     {
         free(hdcd_decoder);

Audio/Chain/Downmix.h

@@ -1,26 +0,0 @@
-//
-//  Downmix.h
-//  Cog
-//
-//  Created by Christopher Snowhill on 2/05/22.
-//  Copyright 2022 __LoSnoCo__. All rights reserved.
-//
-
-#import <Foundation/Foundation.h>
-#import <CoreAudio/CoreAudio.h>
-
-#import "HeadphoneFilter.h"
-
-@interface DownmixProcessor : NSObject {
-    HeadphoneFilter *hFilter;
-    
-    AudioStreamBasicDescription inputFormat;
-    AudioStreamBasicDescription outputFormat;
-}
-
-- (id) initWithInputFormat:(AudioStreamBasicDescription)inf andOutputFormat:(AudioStreamBasicDescription)outf;
-
-- (void) process:(const void*)inBuffer frameCount:(size_t)frames output:(void *)outBuffer;
-
-@end
-

Audio/Chain/Downmix.m

@@ -1,302 +0,0 @@
-//
-//  Downmix.m
-//  Cog
-//
-//  Created by Christopher Snowhill on 2/05/22.
-//  Copyright 2022 __LoSnoCo__. All rights reserved.
-//
-
-#import "Downmix.h"
-
-#import "Logging.h"
-
-static const float STEREO_DOWNMIX[8-2][8][2]={
-    /*3.0*/
-    {
-        {0.5858F,0.0F},{0.0F,0.5858F},{0.4142F,0.4142F}
-    },
-    /*quadrophonic*/
-    {
-        {0.4226F,0.0F},{0.0F,0.4226F},{0.366F,0.2114F},{0.2114F,0.336F}
-    },
-    /*5.0*/
-    {
-        {0.651F,0.0F},{0.0F,0.651F},{0.46F,0.46F},{0.5636F,0.3254F},
-        {0.3254F,0.5636F}
-    },
-    /*5.1*/
-    {
-        {0.529F,0.0F},{0.0F,0.529F},{0.3741F,0.3741F},{0.3741F,0.3741F},{0.4582F,0.2645F},
-        {0.2645F,0.4582F}
-    },
-    /*6.1*/
-    {
-        {0.4553F,0.0F},{0.0F,0.4553F},{0.322F,0.322F},{0.322F,0.322F},{0.2788F,0.2788F},
-        {0.3943F,0.2277F},{0.2277F,0.3943F}
-    },
-    /*7.1*/
-    {
-        {0.3886F,0.0F},{0.0F,0.3886F},{0.2748F,0.2748F},{0.2748F,0.2748F},{0.3366F,0.1943F},
-        {0.1943F,0.3366F},{0.3366F,0.1943F},{0.1943F,0.3366F}
-    }
-};
-
-static void downmix_to_stereo(const float * inBuffer, int channels, float * outBuffer, size_t count)
-{
-    if (channels >= 3 && channels <= 8)
-    for (size_t i = 0; i < count; ++i)
-    {
-        float left = 0, right = 0;
-        for (int j = 0; j < channels; ++j)
-        {
-            left += inBuffer[i * channels + j] * STEREO_DOWNMIX[channels - 3][j][0];
-            right += inBuffer[i * channels + j] * STEREO_DOWNMIX[channels - 3][j][1];
-        }
-        outBuffer[i * 2 + 0] = left;
-        outBuffer[i * 2 + 1] = right;
-    }
-}
-
-static void downmix_to_mono(const float * inBuffer, int channels, float * outBuffer, size_t count)
-{
-    float tempBuffer[count * 2];
-    if (channels >= 3 && channels <= 8)
-    {
-        downmix_to_stereo(inBuffer, channels, tempBuffer, count);
-        inBuffer = tempBuffer;
-        channels = 2;
-    }
-    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;
-    }
-}
-
-static void upmix(const float * inBuffer, int inchannels, float * outBuffer, int outchannels, size_t count)
-{
-    for (ssize_t i = 0; i < count; ++i)
-    {
-        if (inchannels == 1 && outchannels == 2)
-        {
-            // upmix mono to stereo
-            float sample = inBuffer[i];
-            outBuffer[i * 2 + 0] = sample;
-            outBuffer[i * 2 + 1] = sample;
-        }
-        else if (inchannels == 1 && outchannels == 4)
-        {
-            // upmix mono to quad
-            float sample = inBuffer[i];
-            outBuffer[i * 4 + 0] = sample;
-            outBuffer[i * 4 + 1] = sample;
-            outBuffer[i * 4 + 2] = 0;
-            outBuffer[i * 4 + 3] = 0;
-        }
-        else if (inchannels == 1 && (outchannels == 3 || outchannels >= 5))
-        {
-            // upmix mono to center channel
-            float sample = inBuffer[i];
-            outBuffer[i * outchannels + 2] = sample;
-            for (int j = 0; j < 2; ++j)
-            {
-                outBuffer[i * outchannels + j] = 0;
-            }
-            for (int j = 3; j < outchannels; ++j)
-            {
-                outBuffer[i * outchannels + j] = 0;
-            }
-        }
-        else if (inchannels == 4 && outchannels >= 5)
-        {
-            float fl = inBuffer[i * 4 + 0];
-            float fr = inBuffer[i * 4 + 1];
-            float bl = inBuffer[i * 4 + 2];
-            float br = inBuffer[i * 4 + 3];
-            const int skipclfe = (outchannels == 5) ? 1 : 2;
-            outBuffer[i * outchannels + 0] = fl;
-            outBuffer[i * outchannels + 1] = fr;
-            outBuffer[i * outchannels + skipclfe + 2] = bl;
-            outBuffer[i * outchannels + skipclfe + 3] = br;
-            for (int j = 0; j < skipclfe; ++j)
-            {
-                outBuffer[i * outchannels + 2 + j] = 0;
-            }
-            for (int j = 4 + skipclfe; j < outchannels; ++j)
-            {
-                outBuffer[i * outchannels + j] = 0;
-            }
-        }
-        else if (inchannels == 5 && outchannels >= 6)
-        {
-            float fl = inBuffer[i * 5 + 0];
-            float fr = inBuffer[i * 5 + 1];
-            float c = inBuffer[i * 5 + 2];
-            float bl = inBuffer[i * 5 + 3];
-            float br = inBuffer[i * 5 + 4];
-            outBuffer[i * outchannels + 0] = fl;
-            outBuffer[i * outchannels + 1] = fr;
-            outBuffer[i * outchannels + 2] = c;
-            outBuffer[i * outchannels + 3] = 0;
-            outBuffer[i * outchannels + 4] = bl;
-            outBuffer[i * outchannels + 5] = br;
-            for (int j = 6; j < outchannels; ++j)
-            {
-                outBuffer[i * outchannels + j] = 0;
-            }
-        }
-        else if (inchannels == 7 && outchannels == 8)
-        {
-            float fl = inBuffer[i * 7 + 0];
-            float fr = inBuffer[i * 7 + 1];
-            float c = inBuffer[i * 7 + 2];
-            float lfe = inBuffer[i * 7 + 3];
-            float sl = inBuffer[i * 7 + 4];
-            float sr = inBuffer[i * 7 + 5];
-            float bc = inBuffer[i * 7 + 6];
-            outBuffer[i * 8 + 0] = fl;
-            outBuffer[i * 8 + 1] = fr;
-            outBuffer[i * 8 + 2] = c;
-            outBuffer[i * 8 + 3] = lfe;
-            outBuffer[i * 8 + 4] = bc;
-            outBuffer[i * 8 + 5] = bc;
-            outBuffer[i * 8 + 6] = sl;
-            outBuffer[i * 8 + 7] = sr;
-        }
-        else
-        {
-            // upmix N channels to N channels plus silence the empty channels
-            float samples[inchannels];
-            for (int j = 0; j < inchannels; ++j)
-            {
-                samples[j] = inBuffer[i * inchannels + j];
-            }
-            for (int j = 0; j < inchannels; ++j)
-            {
-                outBuffer[i * outchannels + j] = samples[j];
-            }
-            for (int j = inchannels; j < outchannels; ++j)
-            {
-                outBuffer[i * outchannels + j] = 0;
-            }
-        }
-    }
-}
-
-@implementation DownmixProcessor
-
-- (id) initWithInputFormat:(AudioStreamBasicDescription)inf andOutputFormat:(AudioStreamBasicDescription)outf {
-    self = [super init];
-    
-    if (self) {
-        if (inf.mFormatID != kAudioFormatLinearPCM ||
-            (inf.mFormatFlags & kAudioFormatFlagsNativeFloatPacked) != kAudioFormatFlagsNativeFloatPacked ||
-            inf.mBitsPerChannel != 32 ||
-            inf.mBytesPerFrame != (4 * inf.mChannelsPerFrame) ||
-            inf.mBytesPerPacket != inf.mFramesPerPacket * inf.mBytesPerFrame)
-            return nil;
-        
-        if (outf.mFormatID != kAudioFormatLinearPCM ||
-            (outf.mFormatFlags & kAudioFormatFlagsNativeFloatPacked) != kAudioFormatFlagsNativeFloatPacked ||
-            outf.mBitsPerChannel != 32 ||
-            outf.mBytesPerFrame != (4 * outf.mChannelsPerFrame) ||
-            outf.mBytesPerPacket != outf.mFramesPerPacket * outf.mBytesPerFrame)
-            return nil;
-        
-        inputFormat = inf;
-        outputFormat = outf;
-
-        [self setupVirt];
-
-        [[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.headphoneVirtualization" options:0 context:nil];
-        [[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.hrirPath" options:0 context:nil];
-    }
-    
-    return self;
-}
-
-- (void) dealloc {
-    [[NSUserDefaultsController sharedUserDefaultsController] removeObserver:self forKeyPath:@"values.headphoneVirtualization"];
-    [[NSUserDefaultsController sharedUserDefaultsController] removeObserver:self forKeyPath:@"values.hrirPath"];
-}
-
-- (void) setupVirt {
-    @synchronized(hFilter) {
-        hFilter = nil;
-    }
-    
-    BOOL hVirt = [[[NSUserDefaultsController sharedUserDefaultsController] defaults] boolForKey:@"headphoneVirtualization"];
-    
-    if (hVirt &&
-        outputFormat.mChannelsPerFrame == 2 &&
-        inputFormat.mChannelsPerFrame >= 1 &&
-         inputFormat.mChannelsPerFrame <= 8) {
-        NSString * userPreset = [[[NSUserDefaultsController sharedUserDefaultsController] defaults] stringForKey:@"hrirPath"];
-        
-        NSURL * presetUrl = nil;
-        
-        if (userPreset && ![userPreset isEqualToString:@""]) {
-            presetUrl = [NSURL fileURLWithPath:userPreset];
-            if (![HeadphoneFilter validateImpulseFile:presetUrl])
-                presetUrl = nil;
-        }
-        
-        if (!presetUrl) {
-            presetUrl = [[NSBundle mainBundle] URLForResource:@"gsx" withExtension:@"wv"];
-            if (![HeadphoneFilter validateImpulseFile:presetUrl])
-                presetUrl = nil;
-        }
-
-        if (presetUrl) {
-            @synchronized(hFilter) {
-                hFilter = [[HeadphoneFilter alloc] initWithImpulseFile:presetUrl forSampleRate:outputFormat.mSampleRate withInputChannels:inputFormat.mChannelsPerFrame];
-            }
-        }
-    }
-}
-
-- (void)observeValueForKeyPath:(NSString *)keyPath
-                      ofObject:(id)object
-                        change:(NSDictionary *)change
-                       context:(void *)context
-{
-    DLog(@"SOMETHING CHANGED!");
-    if ([keyPath isEqualToString:@"values.headphoneVirtualization"] ||
-             [keyPath isEqualToString:@"values.hrirPath"]) {
-        // Reset the converter, without rebuffering
-        [self setupVirt];
-    }
-}
-
-- (void) process:(const void *)inBuffer frameCount:(size_t)frames output:(void *)outBuffer {
-    @synchronized (hFilter) {
-        if ( hFilter ) {
-            [hFilter process:(const float *) inBuffer sampleCount:frames toBuffer:(float *) outBuffer];
-            return;
-        }
-    }
-    
-    if ( inputFormat.mChannelsPerFrame > 2 && outputFormat.mChannelsPerFrame == 2 )
-    {
-        downmix_to_stereo( (const float*) inBuffer, inputFormat.mChannelsPerFrame, (float*) outBuffer, frames );
-    }
-    else if ( inputFormat.mChannelsPerFrame > 1 && outputFormat.mChannelsPerFrame == 1 )
-    {
-        downmix_to_mono( (const float*) inBuffer, inputFormat.mChannelsPerFrame, (float*) outBuffer, frames );
-    }
-    else if ( inputFormat.mChannelsPerFrame < outputFormat.mChannelsPerFrame )
-    {
-        upmix( (const float*) inBuffer, inputFormat.mChannelsPerFrame, (float*) outBuffer, outputFormat.mChannelsPerFrame, frames );
-    }
-    else if ( inputFormat.mChannelsPerFrame == outputFormat.mChannelsPerFrame )
-    {
-        memcpy(outBuffer, inBuffer, frames * outputFormat.mBytesPerPacket);
-    }
-}
-
-@end

Audio/Chain/InputNode.m

@@ -51,8 +51,6 @@
 	
 	bytesPerFrame = ((bitsPerSample + 7) / 8) * channels;
     
-    nodeFormat = propertiesToASBD(properties);
-    
 	shouldContinue = YES;
 	shouldSeek = NO;
 
@@ -70,8 +68,6 @@
 	
 	bytesPerFrame = ((bitsPerSample + 7) / 8) * channels;
     
-    nodeFormat = propertiesToASBD(properties);
-    
 	[self registerObservers];
 
 	shouldContinue = YES;

Audio/Chain/Node.h

@@ -7,7 +7,6 @@
 //
 
 #import <Cocoa/Cocoa.h>
-#import <CoreAudio/CoreAudio.h>
 #import "VirtualRingBuffer.h"
 #import "Semaphore.h"
 
@@ -26,12 +25,7 @@
 	BOOL shouldContinue;	
 	BOOL endOfStream; //All data is now in buffer
 	BOOL initialBufferFilled;
-    
-    AudioStreamBasicDescription nodeFormat;
 }
-
-@property (readonly) AudioStreamBasicDescription nodeFormat;
-
 - (id)initWithController:(id)c previous:(id)p;
 
 - (int)writeData:(void *)ptr amount:(int)a;

Audio/Chain/Node.m

@@ -13,8 +13,6 @@
 
 @implementation Node
 
-@synthesize nodeFormat;
-
 - (id)initWithController:(id)c previous:(id)p
 {
 	self = [super init];
@@ -43,8 +41,7 @@
 	
 	while (shouldContinue == YES && amountLeft > 0)
 	{
-        BOOL wrapped;
+		availOutput = [buffer lengthAvailableToWriteReturningPointer:&writePtr];
-		availOutput = [buffer lengthAvailableToWriteReturningPointer:&writePtr bufferWrapped:&wrapped];
 		if (availOutput == 0) {
 			if (initialBufferFilled == NO) {
 				initialBufferFilled = YES;

Audio/Chain/OutputNode.h

@@ -24,9 +24,6 @@
     
     BOOL paused;
     BOOL started;
-    
-    BOOL formatSetup;
-    BOOL formatChanged;
 }
 
 - (void)beginEqualizer:(AudioUnit)eq;
@@ -65,6 +62,4 @@
 
 - (void)sustainHDCD;
 
-- (BOOL)formatChanged;
-
 @end

Audio/Chain/OutputNode.m

@@ -23,9 +23,6 @@
     paused = YES;
     started = NO;
 
-    formatSetup = NO;
-    formatChanged = NO;
-
 	output = [[OutputCoreAudio alloc] initWithController:self];
 	
 	[output setup];
@@ -131,14 +128,6 @@
             if (oldSampleRatio)
                 amountPlayed += oldSampleRatio * [[converter buffer] bufferedLength];
 #endif
-            AudioStreamBasicDescription inf = [bufferChain inputFormat];
-            
-            format.mChannelsPerFrame = inf.mChannelsPerFrame;
-            format.mBytesPerFrame = ((format.mBitsPerChannel + 7) / 8) * format.mChannelsPerFrame;
-            format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket;
-            
-            sampleRatio = 1.0 / (format.mSampleRate * format.mBytesPerPacket);
-
             [converter setOutputFormat:format];
             [converter inputFormatDidChange:[bufferChain inputFormat]];
         }
@@ -189,21 +178,4 @@
     [output sustainHDCD];
 }
 
-- (BOOL)formatChanged
-{
-    [self setPreviousNode:[[controller bufferChain] finalNode]];
-    
-    AudioStreamBasicDescription inf = [[self previousNode] nodeFormat];
-    
-    if (!formatSetup || memcmp(&nodeFormat, &inf, sizeof(nodeFormat)) != 0) {
-        nodeFormat = inf;
-        formatSetup = YES;
-        formatChanged = YES;
-    }
-
-    BOOL copyFormatChanged = formatChanged;
-    formatChanged = NO;
-    return copyFormatChanged;
-}
-
 @end

Audio/CogAudio.xcodeproj/project.pbxproj

@@ -59,8 +59,6 @@
 		83725A8E27AA0DE60003F694 /* libsoxr.0.dylib in CopyFiles */ = {isa = PBXBuildFile; fileRef = 83725A8A27AA0DBF0003F694 /* libsoxr.0.dylib */; settings = {ATTRIBUTES = (CodeSignOnCopy, ); }; };
 		83725A9027AA16C90003F694 /* Accelerate.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 83725A7B27AA0D8A0003F694 /* Accelerate.framework */; };
 		83725A9127AA16D50003F694 /* AVFoundation.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 83725A7C27AA0D8E0003F694 /* AVFoundation.framework */; };
-		8380F2D927AE6053009183C1 /* Downmix.m in Sources */ = {isa = PBXBuildFile; fileRef = 8380F2D727AE6053009183C1 /* Downmix.m */; };
-		8380F2DA27AE6053009183C1 /* Downmix.h in Headers */ = {isa = PBXBuildFile; fileRef = 8380F2D827AE6053009183C1 /* Downmix.h */; };
 		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 */; };
@@ -153,8 +151,6 @@
 		83725A7C27AA0D8E0003F694 /* AVFoundation.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = AVFoundation.framework; path = System/Library/Frameworks/AVFoundation.framework; sourceTree = SDKROOT; };
 		83725A8827AA0DBF0003F694 /* soxr.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = soxr.h; sourceTree = "<group>"; };
 		83725A8A27AA0DBF0003F694 /* libsoxr.0.dylib */ = {isa = PBXFileReference; lastKnownFileType = "compiled.mach-o.dylib"; path = libsoxr.0.dylib; sourceTree = "<group>"; };
-		8380F2D727AE6053009183C1 /* Downmix.m */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.objc; path = Downmix.m; sourceTree = "<group>"; };
-		8380F2D827AE6053009183C1 /* Downmix.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = Downmix.h; sourceTree = "<group>"; };
 		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>"; };
@@ -288,8 +284,6 @@
 		17D21C750B8BE4BA00D1EBDE /* Chain */ = {
 			isa = PBXGroup;
 			children = (
-				8380F2D827AE6053009183C1 /* Downmix.h */,
-				8380F2D727AE6053009183C1 /* Downmix.m */,
 				83A44A00279119B50049B6E2 /* RefillNode.h */,
 				83A449FF279119B50049B6E2 /* RefillNode.m */,
 				17D21C760B8BE4BA00D1EBDE /* BufferChain.h */,
@@ -452,7 +446,6 @@
 				8347C7412796C58800FA8A7D /* NSFileHandle+CreateFile.h in Headers */,
 				17C940230B900909008627D6 /* AudioMetadataReader.h in Headers */,
 				83725A8B27AA0DBF0003F694 /* soxr.h in Headers */,
-				8380F2DA27AE6053009183C1 /* Downmix.h in Headers */,
 				17B619300B909BC300BC003F /* AudioPropertiesReader.h in Headers */,
 				835EDD7D279FE307001EDCCE /* HeadphoneFilter.h in Headers */,
 				839366671815923C006DD712 /* CogPluginMulti.h in Headers */,
@@ -553,7 +546,6 @@
 				8347C7422796C58800FA8A7D /* NSFileHandle+CreateFile.m in Sources */,
 				17D21DC80B8BE79700D1EBDE /* CoreAudioUtils.m in Sources */,
 				839366681815923C006DD712 /* CogPluginMulti.m in Sources */,
-				8380F2D927AE6053009183C1 /* Downmix.m in Sources */,
 				835C88AA2797D4D400E28EAE /* lpc.c in Sources */,
 				17D21EBE0B8BF44000D1EBDE /* AudioPlayer.m in Sources */,
 				17F94DD60B8D0F7000A34E87 /* PluginController.m in Sources */,

Audio/Output/OutputCoreAudio.h

@@ -19,8 +19,6 @@
 
 #import "Semaphore.h"
 
-#import "Downmix.h"
-
 //#define OUTPUT_LOG
 #ifdef OUTPUT_LOG
 #import <stdio.h>
@@ -44,9 +42,6 @@
     
     BOOL eqEnabled;
     
-    BOOL streamFormatSetup;
-    
-    atomic_long bytesBuffered;
     atomic_long bytesRendered;
     atomic_long bytesHdcdSustained;
     
@@ -59,19 +54,12 @@
 
     AudioDeviceID outputDeviceID;
     AudioStreamBasicDescription deviceFormat;    // info about the default device
-    AudioStreamBasicDescription outerStreamFormat; // this is set when the buffer changes
-    AudioStreamBasicDescription streamFormat;      // this is set when the output callback gets it
 
     AUAudioUnit *_au;
     size_t _bufferSize;
     
     AudioUnit _eq;
     
-    DownmixProcessor *downmixer;
-    
-    void * savedBuffer;
-    size_t savedSize;
-    size_t savedMaxSize;
 #ifdef OUTPUT_LOG
     FILE *_logFile;
 #endif

Audio/Output/OutputCoreAudio.m

@@ -83,132 +83,45 @@ static OSStatus renderCallback( void *inRefCon, AudioUnitRenderActionFlags *ioAc
         return 0;
     }
     
-    if (_self->savedSize) {
+    void * readPtr;
-        int readBytes = (int) _self->savedSize;
+    int toRead = [[_self->outputController buffer] lengthAvailableToReadReturningPointer:&readPtr];
     
-        const int streamchannels = _self->streamFormat.mChannelsPerFrame;
+    if (toRead > amountToRead)
-        const int streamBytesPerPacket = streamchannels * sizeof(float);
+        toRead = amountToRead;
     
-        int samplesToRead = readBytes / streamBytesPerPacket;
+    if (toRead) {
-        
+        fillBuffers(ioData, (float*)readPtr, toRead / bytesPerPacket, 0);
-        if (samplesToRead > (amountToRead / bytesPerPacket))
+        amountRead = toRead;
-            samplesToRead = amountToRead / bytesPerPacket;
+        [[_self->outputController buffer] didReadLength:toRead];
-        
+        [_self->outputController incrementAmountPlayed:amountRead];
-        readBytes = samplesToRead * streamBytesPerPacket;
+        atomic_fetch_add(&_self->bytesRendered, amountRead);
-        
-        atomic_fetch_sub(&_self->bytesBuffered, readBytes);
-        
-        float downmixBuffer[samplesToRead * channels];
-        [_self->downmixer process:_self->savedBuffer frameCount:samplesToRead output:downmixBuffer];
-        fillBuffers(ioData, downmixBuffer, samplesToRead, 0);
-        amountRead += samplesToRead * bytesPerPacket;
-        [_self->outputController incrementAmountPlayed:samplesToRead * bytesPerPacket];
-        atomic_fetch_add(&_self->bytesRendered, samplesToRead * bytesPerPacket);
         [_self->writeSemaphore signal];
-
-        if (_self->savedSize > readBytes) {
-            _self->savedSize -= readBytes;
-            memmove(_self->savedBuffer, _self->savedBuffer + readBytes, _self->savedSize);
-        }
-        else {
-            _self->savedSize = 0;
-        }
     }
+    else
+        [[_self->outputController buffer] didReadLength:0];
 
-    while (amountRead < amountToRead && [_self->outputController shouldContinue])
+    // Try repeatedly! Buffer wraps can cause a slight data shortage, as can
+    // unexpected track changes.
+    while ((amountRead < amountToRead) && [_self->outputController shouldContinue] == YES)
     {
-        void * readPtr;
+        int amountRead2; //Use this since return type of readdata isnt known...may want to fix then can do a simple += to readdata
-        int toRead = 0;
+        amountRead2 = [[_self->outputController buffer] lengthAvailableToReadReturningPointer:&readPtr];
-        do {
+        if (amountRead2 > (amountToRead - amountRead))
-            toRead = [[_self->outputController buffer] lengthAvailableToReadReturningPointer:&readPtr];
+            amountRead2 = amountToRead - amountRead;
-            if (toRead && *((uint8_t*)readPtr) == 0xFF) {
+        if (amountRead2) {
-                size_t toSkip = 0;
+            atomic_fetch_add(&_self->bytesRendered, amountRead2);
-                while (toRead && *((uint8_t*)readPtr) == 0xFF) {
+            fillBuffers(ioData, (float*)readPtr, amountRead2 / bytesPerPacket, amountRead / bytesPerPacket);
-                    toSkip++;
+            [[_self->outputController buffer] didReadLength:amountRead2];
-                    readPtr++;
-                    toRead--;
-                }
-                [[_self->outputController buffer] didReadLength:(int)toSkip];
-                toRead = 0;
-            }
-        }
-        while (!toRead);
 
-        int bytesRead = 0;
+            [_self->outputController incrementAmountPlayed:amountRead2];
             
-        int32_t chunkId = -1;
+            amountRead += amountRead2;
-
-        if (toRead >= 4) {
-            memcpy(&chunkId, readPtr, 4);
-            readPtr += 4;
-            toRead -= 4;
-            bytesRead += 4;
-        }
-
-        if (chunkId == 1 && toRead >= sizeof(AudioStreamBasicDescription)) {
-            AudioStreamBasicDescription inf;
-            memcpy(&inf, readPtr, sizeof(inf));
-            readPtr += sizeof(inf);
-            toRead -= sizeof(inf);
-            bytesRead += sizeof(inf);
-            
-            if (!_self->streamFormatSetup || memcmp(&inf, &_self->streamFormat, sizeof(inf)) != 0) {
-                _self->streamFormatSetup = YES;
-                _self->streamFormat = inf;
-                _self->downmixer = [[DownmixProcessor alloc] initWithInputFormat:inf andOutputFormat:_self->deviceFormat];
-            }
-            
-            if (toRead >= 4) {
-                memcpy(&chunkId, readPtr, 4);
-                readPtr += 4;
-                toRead -= 4;
-                bytesRead += 4;
-            }
-            else chunkId = -1;
-        }
-
-        const int streamchannels = _self->streamFormat.mChannelsPerFrame;
-        const int streamBytesPerPacket = streamchannels * sizeof(float);
-
-        if (chunkId == 0 && toRead >= 4) {
-            memcpy(&chunkId, readPtr, 4);
-            readPtr += 4;
-            bytesRead += 4;
-            toRead = chunkId;
-        }
-
-        if (toRead) {
-            size_t samplesToRead = toRead / streamBytesPerPacket;
-            size_t saveBytes = 0;
-            
-            if (samplesToRead * bytesPerPacket > (amountToRead - amountRead)) {
-                size_t shortToRead = (amountToRead - amountRead) / bytesPerPacket;
-                saveBytes = (samplesToRead - shortToRead) * streamBytesPerPacket;
-                samplesToRead = shortToRead;
-            }
-            float downmixBuffer[samplesToRead * channels];
-            [_self->downmixer process:readPtr frameCount:samplesToRead output:downmixBuffer];
-            fillBuffers(ioData, downmixBuffer, samplesToRead, amountRead / bytesPerPacket);
-            amountRead += samplesToRead * bytesPerPacket;
-            bytesRead += toRead;
-            
-            if (saveBytes) {
-                if (!_self->savedBuffer || _self->savedMaxSize < saveBytes) {
-                    _self->savedBuffer = realloc(_self->savedBuffer, _self->savedMaxSize = saveBytes * 3);
-                }
-                _self->savedSize = saveBytes;
-                memcpy(_self->savedBuffer, readPtr + toRead - saveBytes, saveBytes);
-            }
-            
-            atomic_fetch_sub(&_self->bytesBuffered, toRead - saveBytes);
-            
-            [[_self->outputController buffer] didReadLength:bytesRead];
-            [_self->outputController incrementAmountPlayed:samplesToRead * bytesPerPacket];
-            atomic_fetch_add(&_self->bytesRendered, samplesToRead * bytesPerPacket);
             [_self->writeSemaphore signal];
         }
-        else
+        else {
-            [[_self->outputController buffer] didReadLength:bytesRead];
+            [[_self->outputController buffer] didReadLength:0];
+            [_self->readSemaphore timedWait:500];
+        }
     }
     
     float volumeScale = 1.0;
@@ -225,6 +138,14 @@ static OSStatus renderCallback( void *inRefCon, AudioUnitRenderActionFlags *ioAc
 
     scaleBuffersByVolume(ioData, _self->volume * volumeScale);
     
+    if (amountRead < amountToRead)
+    {
+        // Either underrun, or no data at all. Caller output tends to just
+        // buffer loop if it doesn't get anything, so always produce a full
+        // buffer, and silence anything we couldn't supply.
+        clearBuffers(ioData, (amountToRead - amountRead) / bytesPerPacket, amountRead / bytesPerPacket);
+    }
+    
     return 0;
 };
 
@@ -244,15 +165,6 @@ static OSStatus renderCallback( void *inRefCon, AudioUnitRenderActionFlags *ioAc
         started = NO;
         stopNext = NO;
         
-        streamFormatSetup = NO;
-
-        downmixer = nil;
-        
-        savedBuffer = NULL;
-        savedSize = 0;
-        savedMaxSize = 0;
-        
-        atomic_init(&bytesBuffered, 0);
         atomic_init(&bytesRendered, 0);
         atomic_init(&bytesHdcdSustained, 0);
         
@@ -312,7 +224,6 @@ default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const
         
         if ([outputController shouldReset]) {
             [[outputController buffer] empty];
-            atomic_store(&bytesBuffered, 0);
             [outputController setShouldReset:NO];
             [delayedEvents removeAllObjects];
             delayedEventsPopped = YES;
@@ -333,73 +244,13 @@ default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const
             break;
         
         void *writePtr;
-        BOOL wrapped = NO;
+        int toWrite = [[outputController buffer] lengthAvailableToWriteReturningPointer:&writePtr];
-        int toWrite = [[outputController buffer] lengthAvailableToWriteReturningPointer:&writePtr bufferWrapped:&wrapped];
-        int bytesWritten = 0;
-        if (toWrite >= 4 + sizeof(AudioStreamBasicDescription)) {
-            if ([outputController formatChanged]) {
-                int32_t chunkId = 1; // ASBD
-                memcpy(writePtr, &chunkId, 4);
-
-                writePtr += 4;
-                toWrite -= 4;
-                bytesWritten += 4;
-                
-                AudioStreamBasicDescription inf = [outputController nodeFormat];
-                
-                outerStreamFormat = inf;
-
-                memcpy(writePtr, &inf, sizeof(inf));
-                
-                writePtr += sizeof(inf);
-                toWrite -= sizeof(inf);
-                bytesWritten += sizeof(inf);
-            }
-        }
-        [[outputController buffer] didWriteLength:bytesWritten];
-        
-        toWrite = [[outputController buffer] lengthAvailableToWriteReturningPointer:&writePtr bufferWrapped:&wrapped];
         int bytesRead = 0;
-        bytesWritten = 0;
+        if (toWrite > CHUNK_SIZE)
-        if (toWrite >= 4 + 4 + 512 * outerStreamFormat.mBytesPerPacket) {
+            toWrite = CHUNK_SIZE;
-            uint8_t buffer[512 * outerStreamFormat.mBytesPerPacket];
+        if (toWrite)
-
+            bytesRead = [outputController readData:writePtr amount:toWrite];
-            bytesRead = [outputController readData:buffer amount:(int)sizeof(buffer)];
+        [[outputController buffer] didWriteLength:bytesRead];
-            
-            while (bytesRead < sizeof(buffer) && ![outputController endOfStream]) {
-                int bytesRead2 = [outputController readData:buffer + bytesRead amount:(int)(sizeof(buffer) - bytesRead)];
-                bytesRead += bytesRead2;
-            }
-           
-            int32_t chunkId = 0; // audio data
-            memcpy(writePtr, &chunkId, 4);
-            writePtr += 4;
-            toWrite -= 4;
-            bytesWritten += 4;
-            
-            chunkId = bytesRead;
-            memcpy(writePtr, &chunkId, 4);
-            writePtr += 4;
-            toWrite -= 4;
-            bytesWritten += 4;
-            
-            memcpy(writePtr, buffer, bytesRead);
-            writePtr += bytesRead;
-            toWrite -= bytesRead;
-            bytesWritten += bytesRead;
-            
-            atomic_fetch_add(&bytesBuffered, bytesRead);
-            
-            [[outputController buffer] didWriteLength:bytesWritten];
-        }
-        else if (wrapped && toWrite > 0) {
-            memset(writePtr, 0xFF, toWrite);
-            [[outputController buffer] didWriteLength:toWrite];
-        }
-        else if (toWrite) {
-            [[outputController buffer] didWriteLength:0];
-            toWrite = 0;
-        }
         if (bytesRead) {
             [readSemaphore signal];
             continue;
@@ -419,20 +270,20 @@ default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const
             // End of input possibly reached
             if (delayedEventsPopped && [outputController endOfStream] == YES)
             {
-                long _bytesBuffered = atomic_load_explicit(&bytesBuffered, memory_order_relaxed) * deviceFormat.mBytesPerPacket / outerStreamFormat.mBytesPerPacket;
+                long bytesBuffered = [[outputController buffer] bufferedLength];
-                _bytesBuffered += atomic_load_explicit(&bytesRendered, memory_order_relaxed);
+                bytesBuffered += atomic_load_explicit(&bytesRendered, memory_order_relaxed);
                 if ([outputController chainQueueHasTracks])
                 {
-                    if (_bytesBuffered < CHUNK_SIZE / 2)
+                    if (bytesBuffered < CHUNK_SIZE / 2)
-                        _bytesBuffered = 0;
+                        bytesBuffered = 0;
                     else
-                        _bytesBuffered -= CHUNK_SIZE / 2;
+                        bytesBuffered -= CHUNK_SIZE / 2;
                 }
                 else {
                     stopNext = YES;
                     break;
                 }
-                [delayedEvents addObject:[NSNumber numberWithLong:_bytesBuffered]];
+                [delayedEvents addObject:[NSNumber numberWithLong:bytesBuffered]];
                 delayedEventsPopped = NO;
                 if (!started) {
                     started = YES;
@@ -626,9 +477,8 @@ default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const
         NSError *err;
         AVAudioFormat *renderFormat;
 
-        [outputController incrementAmountPlayed:atomic_load_explicit(&bytesBuffered, memory_order_relaxed)];
+        [outputController incrementAmountPlayed:[[outputController buffer] bufferedLength]];
         [[outputController buffer] empty];
-        atomic_store(&bytesBuffered, 0);
 
         _deviceFormat = format;
         deviceFormat = *(format.streamDescription);
@@ -713,12 +563,6 @@ default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const
     stopNext = NO;
     outputDeviceID = -1;
 	
-    streamFormatSetup = NO;
-    
-    savedBuffer = NULL;
-    savedSize = 0;
-    savedMaxSize = 0;
-	
     AudioComponentDescription desc;
     NSError *err;
 	
@@ -825,8 +669,6 @@ default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const
         return 0;
     };
 
-    [_au setMaximumFramesToRender:512];
-
     UInt32 value;
     UInt32 size = sizeof(value);
 
@@ -946,14 +788,6 @@ default_device_changed(AudioObjectID inObjectID, UInt32 inNumberAddresses, const
         _logFile = NULL;
     }
 #endif
-    if (savedBuffer)
-    {
-        free(savedBuffer);
-        savedBuffer = NULL;
-        savedSize = 0;
-        savedMaxSize = 0;
-    }
-    
     outputController = nil;
 }
 

Audio/ThirdParty/VirtualRingBuffer/VirtualRingBuffer.h

@@ -83,7 +83,7 @@
 // Write operations:
 
 // The writing thread must call this method first.
-- (UInt32)lengthAvailableToWriteReturningPointer:(void **)returnedWritePointer bufferWrapped:(BOOL*)wrapped;
+- (UInt32)lengthAvailableToWriteReturningPointer:(void **)returnedWritePointer;
 // Iff a value > 0 is returned, the writing thread may then write that much data into the returned pointer.
 // Afterwards, the writing thread must call didWriteLength:.
 - (void)didWriteLength:(UInt32)length;

Audio/ThirdParty/VirtualRingBuffer/VirtualRingBuffer.m

@@ -140,7 +140,7 @@ static void deallocateVirtualBuffer(void *buffer, UInt32 bufferLength);
 // Write operations
 //
 
-- (UInt32)lengthAvailableToWriteReturningPointer:(void **)returnedWritePointer bufferWrapped:(BOOL *)wrapped
+- (UInt32)lengthAvailableToWriteReturningPointer:(void **)returnedWritePointer
 {
     // Assumptions:
     // returnedWritePointer != NULL
@@ -152,10 +152,8 @@ static void deallocateVirtualBuffer(void *buffer, UInt32 bufferLength);
     int localBufferFilled = atomic_load_explicit(&bufferFilled, memory_order_relaxed);
 
     length = bufferLength - localBufferFilled;
-    if (length >= bufferLength - localWritePointer) {
+    if (length > bufferLength - localWritePointer)
-        *wrapped = YES;
         length = bufferLength - localWritePointer;
-    }
 
     *returnedWritePointer = buffer + localWritePointer;