iOS 18 audio bugfix

Author: gaborszanto

Superpowered/OpenSource/SuperpoweredIOSAudioIO.mm

@@ -18,6 +18,18 @@ static audioDeviceType NSStringToAudioDeviceType(NSString *str) {
     else return audioDeviceType_other;
 }
 
+static unsigned int nearestPowerOfTwo(unsigned int n) {
+    unsigned int v = n - 1;
+    v |= v >> 1;
+    v |= v >> 2;
+    v |= v >> 4;
+    v |= v >> 8;
+    v |= v >> 16;
+    v++;
+    unsigned int x = v >> 1;
+    return (v - n) > (n - x) ? x : v;
+}
+
 // Initialization
 @implementation SuperpoweredIOSAudioIO {
 #if __has_feature(objc_arc)
@@ -307,7 +319,10 @@ - (void)onRouteChange:(NSNotification *)notification {
 }
 
 - (void)applyBuffersize {
-    [[AVAudioSession sharedInstance] setPreferredIOBufferDuration:double(preferredBufferSizeMs) * 0.001 error:NULL];
+    double sr = double(self->samplerate);
+    if (sr < 1000) sr = 48000.0;
+    unsigned int powerOfTwoBufferSizeFrames = nearestPowerOfTwo((unsigned int)(sr * preferredBufferSizeMs * 0.001));
+    [[AVAudioSession sharedInstance] setPreferredIOBufferDuration:powerOfTwoBufferSizeFrames / sr error:NULL];
 }
 
 - (void)applySamplerate {
@@ -374,8 +389,8 @@ static void streamFormatChangedCallback(void *inRefCon, AudioUnit inUnit, AudioU
         if (self->samplerate != sr) {
             self->samplerate = sr;
             int minimum = int(self->samplerate * 0.001f), maximum = int(self->samplerate * 0.025f);
-            self->minimumNumberOfFrames = (minimum >> 3) << 3;
-            self->maximumNumberOfFrames = (maximum >> 3) << 3;
+            self->minimumNumberOfFrames = nearestPowerOfTwo(minimum / 8) * 8;
+            self->maximumNumberOfFrames = nearestPowerOfTwo(maximum / 8) * 8;
             if (self->maximumNumberOfFrames < 1024) self->maximumNumberOfFrames = 1024;
             if (self->minimumNumberOfFrames < 16) self->minimumNumberOfFrames = 16;
             [self performSelectorOnMainThread:@selector(applyBuffersize) withObject:nil waitUntilDone:NO];
@@ -504,7 +519,7 @@ - (void)setPreferredSamplerate:(int)hz {
 - (void)mapChannels {
     outputChannelMap.deviceChannels[0] = outputChannelMap.deviceChannels[1] = -1;
     for (int n = 0; n < 8; n++) outputChannelMap.HDMIChannels[n] = -1;
-    for (int n = 0; n < 32; n++) outputChannelMap.USBChannels[n] = inputChannelMap.USBChannels[n] = - 1;
+    for (int n = 0; n < 32; n++) outputChannelMap.USBChannels[n] = inputChannelMap.USBChannels[n] = -1;
 
     if ([(NSObject *)self->delegate respondsToSelector:@selector(mapChannels:inputMap:externalAudioDeviceName:outputsAndInputs:)]) [delegate mapChannels:&outputChannelMap inputMap:&inputChannelMap externalAudioDeviceName:externalAudioDeviceName outputsAndInputs:audioSystemInfo];
     if (!audioUnit || (numberOfChannels <= 2)) return;

Superpowered/OpenSource/SuperpoweredOSXAudioIO.mm

@@ -214,7 +214,7 @@ static void makeStreamFormat(AudioUnit au, AudioStreamBasicDescription *format,
 
 static void setBufferSize(int samplerate, int preferredBufferSizeMs, AudioDeviceID deviceID) {
     if (samplerate < 1) return;
-    UInt32 frames = powf(2.0f, floorf(log2f(float(samplerate) * 0.001f * float(preferredBufferSizeMs))));
+    UInt32 frames = (UInt32)powf(2.0f, floorf(log2f(float(samplerate) * 0.001f * float(preferredBufferSizeMs))));
     if (frames > 4096) frames = 4096;
     AudioObjectPropertyAddress address = { kAudioDevicePropertyBufferFrameSize, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMain };
     if (deviceID != UINT_MAX) AudioObjectSetPropertyData(deviceID, &address, 0, NULL, sizeof(UInt32), &frames);
@@ -277,7 +277,7 @@ - (void)createAudioUnits {
                 AudioUnitAddPropertyListener(outau, kAudioUnitProperty_StreamFormat, streamFormatChangedCallback, (__bridge void *)self);
                 AudioStreamBasicDescription format;
                 makeStreamFormat(outau, &format, false, numberOfChannels);
-                samplerate = format.mSampleRate;
+                samplerate = (int)format.mSampleRate;
                 if (!AudioUnitSetProperty(outau, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &format, sizeof(format))) {
                     AURenderCallbackStruct callbackStruct;
                     callbackStruct.inputProc = audioOutputCallback;
@@ -313,7 +313,7 @@ - (void)createAudioUnits {
                 AudioUnitAddPropertyListener(inau, kAudioUnitProperty_StreamFormat, streamFormatChangedCallback, (__bridge void *)self);
                 AudioStreamBasicDescription format;
                 makeStreamFormat(inau, &format, true, numberOfChannels);
-                if (outputEnabled) format.mSampleRate = samplerate; else samplerate = format.mSampleRate;
+                if (outputEnabled) format.mSampleRate = samplerate; else samplerate = (int)format.mSampleRate;
                 if (!AudioUnitSetProperty(inau, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &format, sizeof(format))) {
                     AURenderCallbackStruct callbackStruct;
                     callbackStruct.inputProc = audioInputCallback;

Superpowered/SuperpoweredAdvancedAudioPlayer.h

@@ -385,10 +385,14 @@ class AdvancedAudioPlayer {
 
 /// @return Returns with the current pitch bend percent. Will be 1 if there is no pitch bend happening.
     JSWASM float getCurrentPitchBendPercent();
-
+    
 /// @brief Reset the pitch bend offset to the beatgrid to zero.
     JSWASM void resetBendMsOffset();
 
+/// @brief Set the pitch bend offset to the beatgrid.
+/// @param ms The value.
+    JSWASM void setBendOffsetMs(double ms);
+
 /// @return Indicates if returning from scratching or reverse playback will maintain the playback position as if the player had never entered into scratching or reverse playback.
     JSWASM bool isPerformingSlip();