iPhone AudioUnitRender error -50 in device

Question

I am working on one project in which i have used AudioUnitRender it runs fine in simulator but gives -50 error in the device.

If anyone have faced similar problem please give me some solution.

RIOInterface* THIS = (RIOInterface *)inRefCon;
COMPLEX_SPLIT A = THIS->A;
void *dataBuffer = THIS->dataBuffer;
float *outputBuffer = THIS->outputBuffer;
FFTSetup fftSetup = THIS->fftSetup;

uint32_t log2n = THIS->log2n;
uint32_t n = THIS->n;
uint32_t nOver2 = THIS->nOver2;
uint32_t stride = 1;
int bufferCapacity = THIS->bufferCapacity;
SInt16 index = THIS->index;

AudioUnit rioUnit = THIS->ioUnit;
OSStatus renderErr;
UInt32 bus1 = 1;

renderErr = AudioUnitRender(rioUnit, ioActionFlags, 
    inTimeStamp, bus1, inNumberFrames, THIS->bufferList);
NSLog(@"%d",renderErr);
if (renderErr < 0) {
    return renderErr;
}

data regarding sample size and frame...

bytesPerSample = sizeof(SInt16);
asbd.mFormatID = kAudioFormatLinearPCM;
asbd.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
asbd.mBitsPerChannel = 8 * bytesPerSample;
asbd.mFramesPerPacket = 1;
asbd.mChannelsPerFrame = 1; 

//asbd.mBytesPerPacket = asbd.mBytesPerFrame * asbd.mFramesPerPacket;
asbd.mBytesPerPacket = bytesPerSample * asbd.mFramesPerPacket;


//asbd.mBytesPerFrame = bytesPerSample * asbd.mChannelsPerFrame;    
asbd.mBytesPerFrame = bytesPerSample * asbd.mChannelsPerFrame;          

asbd.mSampleRate = sampleRate;      

thanks in advance..

Answer 1

The length of the buffer (inNumberFrames) can be different on the device and the simulator. From my experience it is often larger on the device. When you use your own AudioBufferList this is something you have to take into account. I would suggest allocating more memory for the buffer in the AudioBufferList.

Answer 2

I know this thread is old, but I just found the solution to this problem.

The buffer duration for the device is different from that on the simulator. So you have to change the buffer duration:

Float32 bufferDuration = ((Float32) <INSERT YOUR BUFFER DURATION HERE>) / sampleRate; // buffer duration in seconds

AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration, sizeof(bufferDuration), &bufferDuration);

Answer 3

Try adding kAudioFormatFlagsNativeEndian to your list of stream description format flags. Not sure if that will make a difference, but it can't hurt.

Also, I'm suspicious about the use of THIS for the userData member, which definitely does not fill that member with any meaningful data by default. Try running the code in a debugger and see if that instance is correctly extracted and casted. Assuming it is, just for fun try putting the AudioUnit object into a global variable (yeah, I know..) just to see if it works.

Finally, why use THIS->bufferList instead of the one passed into your render callback? That's probably not good.