www.dspguide.com/
Its actually called "The Scientists and Engineers Guide to DSP"
Andy G4JNT
-----Original Message-----
From: [email protected]
[mailto:[email protected]] On Behalf Of Hugh_m0wye
Sent: 13 December 2006 11:13
To: [email protected]
Subject: Re: LF: RE: SAQ Receiver for Window$
Hello Wolf,
You wrote ...
> A similar method is described in 'The Scientist's Guide to Digital
> Signal Processing" (look for the overlap/add method in chapter 18;
> it's available online for free ).
Please can you give us the URL for this, as Google can't find it - sounds
like a very useful document.
73
Hugh
----- Original Message -----
From: "Wolf DL4YHF" <[email protected]>
To: <[email protected]>
Sent: Wednesday, December 13, 2006 10:55 AM
Subject: Re: LF: RE: SAQ Receiver for Window$
> Hi Johan and group,
>
> You wrote..
>
>>
>> Yes, but SAQrx uses a complex FIR filter for the main
>> selectivity so it will take some CPU power if you want:
>>
>> 1. narrow passband width AND...
>> 2. small ripple in the passband AND...
>> 3. steep skirts AND...
>> 4. large stopband attenuation
>>
>> i.e. a "brickwall" response.
>>
>> A complex FIR with, for example:
>>
>> Passband width = 100 Hz
>> Passband ripple <= 0.1 dB
>> Stopband width = 150 Hz
>> Stopband attenuation = 80 dB (either side of +/-75 Hz)
>>
>> designed with Parks-McClellan requires 2 * 1526 = 3052 taps.
>> The main filter in SAQrx is run at Fs = 11025 Hz.
>> I think ~3000 taps would work OK on a fast machine although
>> I haven't checked it. The existing 1000Hz filter has 2*206 taps.
>>
> As an alternative (since you already have the forward and inverse FFT in
> your C-library) you could use FFT convolution for the filter, which
> *greatly* speeds up large filter kernels. The basic principle is to split
> up the incoming signal into overlapping FFTs (with say 4096 to 16384
> points), preferably using 50 % overlap and a raised cosine window, then do
> the filtering by multiplying the spectra with the filter's amplitude, and
> finally transform back to the time domain using the 'inverse' FFT.
> Because (sin(x)^2) + (cos(x)^2) = 1, there is no unwanted amplitude
> modulation caused by the FFT windowing if a 50 % overlap is used. I use
> this method in my own application, and it allows to run filters in 'real
> time' which are impossible with the classic FIR implementation (with
> convolution in the time domain). You don't even need to design the filter
> coefficients; all the algorithm needs is the filter's frequency response.
> A similar method is described in 'The Scientist's Guide to Digital Signal
> Processing" (look for the overlap/add method in chapter 18; it's available
> online for free ).
>
> Cheers,
> Wolf DL4YHF
>
> (happy to be back from hospital)
>
>
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