diff --git a/Audio/Chain/ConverterNode.h b/Audio/Chain/ConverterNode.h
index 397741070..2a23f455b 100644
--- a/Audio/Chain/ConverterNode.h
+++ b/Audio/Chain/ConverterNode.h
@@ -62,8 +62,12 @@
     AudioStreamBasicDescription rememberedInputFormat;
     RefillNode *refillNode;
     id __weak originalPreviousNode;
+    
+    NSString *outputResampling;
 }
 
+@property AudioStreamBasicDescription inputFormat;
+
 - (id)initWithController:(id)c previous:(id)p;
 
 - (BOOL)setupWithInputFormat:(AudioStreamBasicDescription)inputFormat outputFormat:(AudioStreamBasicDescription)outputFormat;
diff --git a/Audio/Chain/ConverterNode.m b/Audio/Chain/ConverterNode.m
index d288f9906..e5a339826 100644
--- a/Audio/Chain/ConverterNode.m
+++ b/Audio/Chain/ConverterNode.m
@@ -35,6 +35,8 @@ void PrintStreamDesc (AudioStreamBasicDescription *inDesc)
 
 @implementation ConverterNode
 
+@synthesize inputFormat;
+
 - (id)initWithController:(id)c previous:(id)p
 {
     self = [super initWithController:c previous:p];
@@ -68,6 +70,8 @@ void PrintStreamDesc (AudioStreamBasicDescription *inDesc)
         
         dsd2pcm = NULL;
         dsd2pcmCount = 0;
+        
+        outputResampling = @"";
 
         [[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.volumeScaling"		options:0 context:nil];
         [[NSUserDefaultsController sharedUserDefaultsController] addObserver:self forKeyPath:@"values.outputResampling" options:0 context:nil];
@@ -1038,8 +1042,11 @@ tryagain:
     }
     else if ([keyPath isEqual:@"values.outputResampling"]) {
         // Reset resampler
-        if (resampler && resampler_data)
-            [self inputFormatDidChange:inputFormat];
+        if (resampler && resampler_data) {
+            NSString *value = [[NSUserDefaults standardUserDefaults] stringForKey:@"outputResampling"];
+            if (![value isEqualToString:outputResampling])
+                [self inputFormatDidChange:inputFormat];
+        }
     }
 }
 
@@ -1104,10 +1111,18 @@ static float db_to_scale(float db)
         || (isFloat && !(inputFormat.mBitsPerChannel == 32 || inputFormat.mBitsPerChannel == 64)))
         return NO;
     
+    // These are really placeholders, as we're doing everything internally now
+    
+    floatFormat = inputFormat;
+    floatFormat.mFormatFlags = kAudioFormatFlagsNativeFloatPacked;
+    floatFormat.mBitsPerChannel = 32;
+    floatFormat.mBytesPerFrame = (32/8)*floatFormat.mChannelsPerFrame;
+    floatFormat.mBytesPerPacket = floatFormat.mBytesPerFrame * floatFormat.mFramesPerPacket;
+    
     if (inputFormat.mBitsPerChannel == 1) {
         // Decimate this for speed
-        inputFormat.mSampleRate *= 1.0 / 8.0;
-        dsd2pcmCount = inputFormat.mChannelsPerFrame;
+        floatFormat.mSampleRate *= 1.0 / 8.0;
+        dsd2pcmCount = floatFormat.mChannelsPerFrame;
         dsd2pcm = calloc(dsd2pcmCount, sizeof(void*));
         dsd2pcm[0] = dsd2pcm_alloc();
         dsd2pcmLatency = dsd2pcm_latency(dsd2pcm[0]);
@@ -1117,14 +1132,6 @@ static float db_to_scale(float db)
         }
     }
     
-    // These are really placeholders, as we're doing everything internally now
-    
-    floatFormat = inputFormat;
-    floatFormat.mFormatFlags = kAudioFormatFlagsNativeFloatPacked;
-    floatFormat.mBitsPerChannel = 32;
-    floatFormat.mBytesPerFrame = (32/8)*floatFormat.mChannelsPerFrame;
-    floatFormat.mBytesPerPacket = floatFormat.mBytesPerFrame * floatFormat.mFramesPerPacket;
-    
     inpOffset = 0;
     inpSize = 0;
     
@@ -1142,9 +1149,9 @@ static float db_to_scale(float db)
     convert_s16_to_float_init_simd();
     convert_s32_to_float_init_simd();
     
-    skipResampler = outputFormat.mSampleRate == inputFormat.mSampleRate;
+    skipResampler = outputFormat.mSampleRate == floatFormat.mSampleRate;
     
-    sampleRatio = (double)outputFormat.mSampleRate / (double)inputFormat.mSampleRate;
+    sampleRatio = (double)outputFormat.mSampleRate / (double)floatFormat.mSampleRate;
     
     if (!skipResampler)
     {
@@ -1162,6 +1169,8 @@ static float db_to_scale(float db)
         else if ([resampling isEqualToString:@"highest"])
             quality = RESAMPLER_QUALITY_HIGHEST;
         
+        outputResampling = resampling;
+        
         if (!retro_resampler_realloc(&resampler_data, &resampler, "sinc", quality, inputFormat.mChannelsPerFrame, sampleRatio))
         {
             return NO;
@@ -1211,7 +1220,8 @@ static float db_to_scale(float db)
 	DLog(@"FORMAT CHANGED");
     paused = YES;
 	[self cleanUp];
-    if (outputFormatChanged && ![buffer isEmpty])
+    if (outputFormatChanged && ![buffer isEmpty] &&
+        memcmp(&outputFormat, &previousOutputFormat, sizeof(outputFormat)) != 0)
     {
         // Transfer previously buffered data, remember input format
         rememberedInputFormat = format;
diff --git a/Audio/Chain/InputNode.m b/Audio/Chain/InputNode.m
index 8568dc501..f63437bb4 100644
--- a/Audio/Chain/InputNode.m
+++ b/Audio/Chain/InputNode.m
@@ -102,7 +102,10 @@
         DLog(@"Input format changed");
         // Converter doesn't need resetting for this, as output format hasn't changed
         ConverterNode *converter = [[[controller controller] bufferChain] converter];
-        [converter inputFormatDidChange:[[[controller controller] bufferChain] inputFormat]];
+        AudioStreamBasicDescription newInputFormat = [[[controller controller] bufferChain] inputFormat];
+        AudioStreamBasicDescription oldInputFormat = [converter inputFormat];
+        if (memcmp(&oldInputFormat, &newInputFormat, sizeof(oldInputFormat)) != 0)
+            [converter inputFormatDidChange:newInputFormat];
 	}
 	else if ([keyPath isEqual:@"metadata"]) {
 		//Inform something of metadata change