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From: Talbot Andrew <ACTALBOT@mail.dstl.gov.uk>
To: rsgb_lf_group@blacksheep.org
Date: Wed, 13 Dec 2006 11:21:02 -0000
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Subject: RE: LF: RE: SAQ Receiver for Window$
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 www.dspguide.com/

Its actually called "The Scientists and Engineers Guide to DSP"

Andy  G4JNT


-----Original Message-----
From: owner-rsgb_lf_group@blacksheep.org
[mailto:owner-rsgb_lf_group@blacksheep.org] On Behalf Of Hugh_m0wye
Sent: 13 December 2006 11:13
To: rsgb_lf_group@blacksheep.org
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" <dl4yhf@freenet.de>
To: <rsgb_lf_group@blacksheep.org>
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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