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From: cpptutor2000 on 20 Apr 2008 22:03
Could some DSP guru please help me a bit ? I am creating a FFT based
application that uses an audio input. The sampling details are as
follows:
encoding : PCM
sampling frequency : 16000 Hz
resolution : 16 bits
channel : mono
signed : true
endianness : little (as this runs on an Intel processor)

The raw bytes get collected in a byte array, and I take two bytes at a
time (resolution is 16 bits) and correct for wrap-around in the first
byte and get the corresponding floating point number which is the
sampled value. The buffer is 2048 bytes long.

When I examine the raw numbers, I find a long sub-sequence of 0s at
the start, before I start seeing the sinusoidal pattern in the sampled
numbers.

I use a time shifted Gaussian window to filter the data before I
compute the FFT. When I examine the array containing the FFT
magnitudes, I find some extra peaks at the end of the array, i.e., the
spectrum does not look symmetric as it should.

I am not sure where the extra peaks are coming from. I am fairly
confident that my FFT routine is working, as I have tested it out with
hand-generated sample data, and the output spectrum has a very
symmetric structure in the peaks.
Any hints, suggestions would be immensely helpful. Thanks in advance
for your help.
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