Fred,
The basic idea is sound and it does work. The procedure is
2 take 1024 input samples and fft to frequency domain
3 truncate frequency vector throwing away upper 3/4 of frequency
channels giving 256 samples
4 LPF by attenuating high frequency components to zero
5 inverse fft back to time domain (sample rate is no 1/4 of input rate)
6 add resulting block of output samples to output sample train
7 take next 1024 samples but displaced only 512 samples in time to get
overlap
8 repeat steps 3 - 5
9 add resulting block of output samples to output sample vector but
displaced by 128 samples
etc, etc.
The initial (and final) 127 samples in the output sample train will be
slightly corrupted since due to not having been constructed from a full
set of input samples (similar effects occur in an FIR). Also at step 4
you need to have a smoothish transition from passband to stop band to
avoid aliasing problems.
Nick,
G4JNX
Fred Krom wrote:
>
> I got a question that maybe can be answered in this group.
> Most of the SDR radio's I know are using the following steps:
> 1e Mix the time domain Q & I signal with a NCO to make a near to zero
> signal
> 2e Take the FFT to convert it to frequence domain
> 3e Filter the signal by setting most of the bins to zero (multiply by
> filter)
> 4e Take the reverse FTT to convert to time domain
> 5e Down sample the signal to something like 8Khz sample rate.
> 6e Somekind of demod.
>
> What I like to do is take step 4 and 5 together, but can not get the
> code working.
> If for example the sample rate S=48KHz, FFT size N=1024 then
> BIN=S/N=47Hz.
> Then taking the reverse FFT with N=256 will result in a signal with
> sample rate S=12KHz (using the same BIN size).
> That will result in a smaller FFT and no down sample, so less cpu is used.
>
> What is wrong, the code or the idea?
> Fred
> PE0FKO
>
> [Non-text portions of this message have been removed]
>
>
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