AudioKit update plot using AKAudioFile buffer and show full waveform

Question

I have a plot

self.plot = EZAudioPlot()
self.plot?.frame = CGRect(x:0, y:0, width:Int(self.plotWidth!), height:Int(self.plotOutlet!.bounds.height))
self.plot?.plotType = EZPlotType.buffer
self.plot?.shouldFill = true
self.plot?.shouldMirror = true
self.plot?.shouldOptimizeForRealtimePlot = true
self.plot?.color = UIColor.white
self.plot?.backgroundColor = UIColor(red:0,green:0,blue:0,alpha:0)
self.plot?.gain = 1
self.plotOutlet?.addSubview(self.plot!)

and I want to update it via the buffer copied from an AKAudioFile:

self.plot?.updateBuffer(file.pcmBuffer.floatChannelData![0], withBufferSize: file.pcmBuffer.frameLength)

This one says: pcmBuffer:265:error cannot readIntBuffer

I have tried another method, converting the buffer to EZAudioFloatData first but I can't get the type-casting right:

let data = EZAudioFloatData(numberOfChannels: 2, buffers: file.pcmBuffer.floatChannelData, bufferSize: file.pcmBuffer.frameLength)

The inspector says: Cannot convert value of type 'UnsafePointer < UnsafeMutablePointer < Float > > ?' to expected argument type 'UnsafeMutablePointer < UnsafeMutablePointer < Float > ? > !'

What am I doing wrong? By the way: I know I can load a wav file with EZAudioFile() and then get the buffer by using .getWaveformData() but for several reasons I want to know if there's a way to do the same using an AKAudioFile instead.

---UPDATE:

I was able to add the waveform feature right into the AKWaveTableDSPKernel.hpp: File:

EZAudioFloatData* getWaveformData(UInt32 numberOfPoints){

if (numCh == 0 || current_size < numberOfPoints){
    // prevent division by zero
    return nil;
}

EZAudioFloatData *waveformData;

float **data = (float **)malloc( sizeof(float*) * numCh );
for (int i = 0; i < numCh; i++)
{
    data[i] = (float *)malloc( sizeof(float) * numberOfPoints );
}

// calculate the required number of frames per buffer
SInt64 framesPerBuffer = ((SInt64) current_size / numberOfPoints);

// read through file and calculate rms at each point
for (SInt64 i = 0; i < numberOfPoints; i++){

    for (int channel = 0; channel < numCh; channel++){

        float channelData[framesPerBuffer];
        for (int frame = 0; frame < framesPerBuffer; frame++)
        {
            if(channel == 0){
                channelData[frame] = ftbl1->tbl[i * framesPerBuffer + frame];
            }else{
                channelData[frame] = ftbl2->tbl[i * framesPerBuffer + frame];
            }
        }
        float rms = [EZAudioUtilities RMS:channelData length:(UInt32)framesPerBuffer];
        data[channel][i] = rms;

    }

}

waveformData = [EZAudioFloatData dataWithNumberOfChannels:numCh buffers:(float **)data bufferSize:(UInt32)numberOfPoints];

// cleanup
for (int i = 0; i < numCh; i++){
    free(data[i]);
}

free(data);

return waveformData;

}

It just needs the common header binding. I will try to do a pull request with this feature soon.