AudioUnit Input Samples

So I am having some trouble here with my AudioUnit taking in data from microphone/line-in in iOS. I am able to set everything up to what I think is okay and it is calling my recordingCallback, but the data that I am getting out of the buffer is not correct. It always returns exactly the same thing, which is mostly zeros and random large numbers. Does anyone know what could be causing this. My code is as follows.

Setting up Audio Unit

OSStatus status;

// Describe audio component
AudioComponentDescription desc;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_RemoteIO;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;

// Get component
AudioComponent inputComponent = AudioComponentFindNext(NULL, &desc);
status = AudioComponentInstanceNew(inputComponent, &audioUnit);

// Enable IO for recording
UInt32 flag = 1;
status = AudioUnitSetProperty(audioUnit, 
                              kAudioOutputUnitProperty_EnableIO, 
                              kAudioUnitScope_Input, 
                              kInputBusNumber,
                              &flag, 
                              sizeof(flag));
// Disable playback IO
flag = 0;
status = AudioUnitSetProperty(audioUnit, 
                              kAudioOutputUnitProperty_EnableIO, 
                              kAudioUnitScope_Output, 
                              kOutputBusNumber,
                              &flag, 
                              sizeof(flag));

// Describe format
AudioStreamBasicDescription audioFormat;
audioFormat.mSampleRate         = 44100.00;
audioFormat.mFormatID           = kAudioFormatLinearPCM;
audioFormat.mFormatFlags        = kAudioFormatFlagsNativeFloatPacked |kAudioFormatFlagIsNonInterleaved;
audioFormat.mFramesPerPacket    = 1;
audioFormat.mChannelsPerFrame   = 1;
audioFormat.mBitsPerChannel     = 32;
audioFormat.mBytesPerPacket     = 4;
audioFormat.mBytesPerFrame      = 4;

// Apply format
status = AudioUnitSetProperty(audioUnit, 
                              kAudioUnitProperty_StreamFormat, 
                              kAudioUnitScope_Output, 
                              kInputBusNumber, 
                              &audioFormat, 
                              sizeof(audioFormat));

// Set input callback
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = recordingCallback;
callbackStruct.inputProcRefCon = (__bridge void*)self;
status = AudioUnitSetProperty(audioUnit, 
                              kAudioOutputUnitProperty_SetInputCallback, 
                              kAudioUnitScope_Global, 
                              kInputBusNumber, 
                              &callbackStruct, 
                              sizeof(callbackStruct));
status = AudioUnitInitialize(audioUnit);

Input Callback

static OSStatus recordingCallback(void *inRefCon, 
                                  AudioUnitRenderActionFlags *ioActionFlags, 
                                  const AudioTimeStamp *inTimeStamp, 
                                  UInt32 inBusNumber, 
                                  UInt32 inNumberFrames, 
                                  AudioBufferList *ioData) {

    AudioBufferList bufferList;
    bufferList.mNumberBuffers = 1;
    bufferList.mBuffers[0].mDataByteSize = 4;
    bufferList.mBuffers[0].mNumberChannels = 1;
    bufferList.mBuffers[0].mData = malloc(sizeof(float)*inNumberFrames); //
    InputAudio *input = (__bridge InputAudio*)inRefCon;

    OSStatus status;

    status = AudioUnitRender([input audioUnit], 
                             ioActionFlags, 
                             inTimeStamp, 
                             inBusNumber, 
                             inNumberFrames, 
                             &bufferList);

    float* result = (float*)&bufferList.mBuffers[0].mData;

    if (input->counter == 5) {
        for (int i = 0;i<inNumberFrames;i++) {
            printf("%f ",result[i]);
        }
    }
    input->counter++;
    return noErr;
}

Anyone ever encounter a similar problem or see something clearly wrong in my code. Thanks in advance for any help!

I am basing all of it off of Michael Tysons Core Audio RemoteIO code

If I remember correctly, the samples you get from the audio buffer in the callback aren't floats, they're SInt16. Try casting the samples like this:

SInt16 *sn16AudioData= (SInt16 *)(bufferList.mBuffers[0].mData);

And these should be the max and min values:

#define sn16_MAX_SAMPLE_VALUE 32767
#define sn16_MIN_SAMPLE_VALUE -32768