Hello Fred, Alberto and all, I'm investing an interesting time in the SdR theory. One particular message of this group makes me problem. The problematic sentence is the following: "1° Mix the time domain Q & I signal with a NCO to make a near to zero signal"(this before the FFT) It seems that the "NCO" must mean something as "numerical oscillator". I don't suppose that it is an analytic oscillator (cos wot / sin wot) but a simple oscillator (cos wot). This simple oscillator applies to I and Q. I suppose that "I" is a sum of cos (wt) from 0 to the sampling frequency/2 I suppose that "Q" is a sum of sin (wt) from 0 to the sampling frequency/2
Now you want to slightly shift the spectrum by w0 (1 KHz for example), you will have at the output: - terms in (w-w0) which are the basebands components - terms in (w+w0) which are non-desired components. As w0 is weak, you can't discriminate (by filtering) w-w0 from w+w0 as the bands will be in the band from 0 to 3 KHz (now, with a big w0 it would be possible to discriminate w-w0 from w+w0). Other thing: the very high components (the last 1 KHz in the example, close to the sampling frequency/2) are going to fold in the base band with the w+w0 operation. It can't be avoided either. Note 1: with an analytical mixer, it would possible to discriminate w-w0 from w+w0 without filter but you will lose the quadrature configuration. Note 2: the following FFT will not help to discriminate the two components. TKS for ideas. 73 Patrick ----- Original Message ----- From: Fred Krom To: [email protected] Sent: Tuesday, September 26, 2006 3:32 PM Subject: [soft_radio] Change sample rate FFT 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]
