The discussion centers on troubleshooting Fast Fourier Transform (FFT) implementation in C# at a sampling rate of 44100 Hz with 16384 samples. The user compares their FFT results with MATLAB, noting discrepancies in frequency peaks, particularly with a sine wave input. Key issues identified include the lack of windowing before FFT, potential under-sampling, and incorrect handling of signed 16-bit audio data. The conversation emphasizes the importance of verifying microphone input and suggests testing with known sine wave signals to validate the FFT implementation.
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Are you looking at stereo information?btb4198 said:however, I am still getting half the Frequencies...
I can just do * by 2 at the very end and get the right Frequencies but I really want to know why it is doing this
I don't know how you are grabbing the information. At some point you must have specified 44100Hz, 16 bits per sample. When you did that, the same data structure should have included whether the signal would be stereo or mono.btb4198 said:I do not know... how can I find out ?
I am using the default microphone on my computer..
what is the different between stereo and mono for a microphone ?
mono is one.. so one mic
so it stereo more that one mic?
how would did affect the input samples ?
int tempint = 0;
for (int index = 0; index < 32768; index += 2)
{
buffer1[tempint] = ((buffer[index + 1] << 8) |
buffer[index + 0]);
if (buffer1[tempint] > 32767)
buffer1[tempint] = buffer1[tempint] - 65536;
tempint++;
} int tempint = 0;
for (int index = 0; index < 32768; index += 2)
{
buffer1[tempint] = ((buffer[index + 1] << 8) |
buffer[index + 0]);
if (buffer1[tempint] > 44100)
buffer1[tempint] = buffer1[tempint] - 88200;
tempint++;