Audio Visualizer from wav looks wrong

So I'm having trouble making an audio visualizer look accurate. The bins that have a significant amount of sound tend to draw correctly, but the problem I'm having is that all the frequencies with no significant sound seem to be coming back with a value that usually bounces between -60dB and -40dB. This forms a flat bouncing line (usually in the higher freqencies).

I want to display 512 bins or less at 30 frames per second. I've been reading up on FFT and audio non stop for a couple weeks now, and so far my process has been:

• Load pcm data from wav file. This comes in as 44100 samples per second that have a range of -/+ 32767. I'm assuming I treat these as real numbers when passing them to the FFT.


• Divide these samples up into 1470 per frame. (446 are ignored)


• Take 1024 samples and apply a hann window.


• Pass the samples to FFT as an array of real[1024] as well as another array of the same size filled with zeros for the imaginary part.


• Get the magnitude by looping through the (samples/2) bins and do a sqrt(real[i]*real[i] + img[i]*img[i]).


• Taking 20 * log(magnitude) to get the decibel level of each bin


• Draw a rectangle for each bin. Draw these bins for each frame.

I've tested it with a couple songs, and a wav file I generated that just plays a tone at 440. With the wav file, I do get a spike at the 440 bin, but all the other bins form a line that isn't much shorter than the 440 bin. Also every other frame, the bins apart from 440 look like a graphed log function with a dip on some other bin.

I'm writing this in c++. Using STK to only load left channel from the audio file:

//put every sample in the song into a temporary vector
for (int i = 0; i < stkObject->getSize(); i++)
{
    standardVector.push_back(stkObject->tick(LEFT));
}

I'm using FFTReal to preform the FFT:

    std::vector<std::vector <double> > leftChannelData;
    int numberOfFrames = stkObject->getSize()/samplesPerFrame;

    leftChannelData.resize(numberOfFrames);
    for(int i = 0; i < numberOfFrames; i++)
    {
        for(int j = 0; j < FFT_SAMPLE_LENGTH; j++)
        {
            real[j] = standardVector[j + (i*samplesPerFrame)];
        }

        applyHannWindow(real, FFT_SAMPLE_LENGTH);
        fft_object.do_fft(imaginary,real);

        //FFTReal instructions say to run this after an fft
        fft_object.rescale(real);

        leftChannelData[i].resize(FFT_SAMPLE_LENGTH/2);
        for (int j = 0; j < FFT_SAMPLE_LENGTH/2; j++)
        {
            double magnitude = sqrt(real[j]*real[j] + imaginary[j]*imaginary[j]);
            double dbValue = 20 * log(std::abs(magnitude)/maxMagnitude);

            leftChannelData[i].at(j) = dbValue;
        }
    }

I'm at a loss as to what's causing this. I've tried various ways to pull those 446 samples I'm ignoring, but the results don't seem to change. I think I may be doing something fundamentally wrong. I've tried normalizing the pcm data before handing it to the fft and I've tried normalizing the magnitude before finding the decibels, but it doesn't seem to be working. Any thoughts?