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# Audio filtering for radio communications using ADAU1700/1

Im working on a project that would benefit from DSP on transmit and receive in a radio communications transceiver. The test board we have, has 4 pots connected to 4 analog inputs. We want to use 2 channels both audio, one for transmit, the other for receive. The transmit channel would have boost for a preset range of frequencies but the receive side would have a variable filter giving a boost, and a variable filter giving a cut or a notch effect. I thought of using the first two pots to set the frequency and the 'depth' of effect for the boost, the last two pots to set the frequency and the depth of cut.

Is there any ready made software specifically for radio communications, SSB in particular, im pretty open minded how to approach this.

John

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•     Hello Roman,

The demodulator loses 6 dB because only half its output is the recovered signal.  The other half is the sum frequency, which is then filtered out.  In mathematical terms,

sin(A) x sin(B)  =  0.5 x [cos(A-B) - cos(A+B)]

Unfortunately as you're finding out, phase-locked loops only work within a relatively narrow frequency range for which they were designed.  This is sufficient for this project's original purpose -- a manually adjustable notch filter for amateur radio, where the PLL tracking was a neat little bonus.  PLLs are not suitable for blind frequency measurement.

If your application is auto feedback nulling for sound reinforcement, likely you'll need  more horsepower than SigmaDSP can provide.  For example, I've helped install a few auditorium sound systems which included a Behringer Ultra-Curve DSP EQ.  Inside (I got curious, see my avatar!) is two SHARCs and a Blackfin.  Any one of these processors is way more powerful than a -1761.   Typically, feedback-killing boxes measure frequency via Fast Fourier Transform (FFT).  Presently the only FFT-capable SigmaDSP is the ADAU1452 series, and we're still waiting for enough block-processing algorithms in SigmaStudio to make good use of this power.

This being said, a fellow engineer on EZ did build a blind frequency measurement scheme using discrete filters -- see https://ez.analog.com/message/140809#140809

Best regards,

Bob

•     Hello Roman,

The demodulator loses 6 dB because only half its output is the recovered signal.  The other half is the sum frequency, which is then filtered out.  In mathematical terms,

sin(A) x sin(B)  =  0.5 x [cos(A-B) - cos(A+B)]

Unfortunately as you're finding out, phase-locked loops only work within a relatively narrow frequency range for which they were designed.  This is sufficient for this project's original purpose -- a manually adjustable notch filter for amateur radio, where the PLL tracking was a neat little bonus.  PLLs are not suitable for blind frequency measurement.

If your application is auto feedback nulling for sound reinforcement, likely you'll need  more horsepower than SigmaDSP can provide.  For example, I've helped install a few auditorium sound systems which included a Behringer Ultra-Curve DSP EQ.  Inside (I got curious, see my avatar!) is two SHARCs and a Blackfin.  Any one of these processors is way more powerful than a -1761.   Typically, feedback-killing boxes measure frequency via Fast Fourier Transform (FFT).  Presently the only FFT-capable SigmaDSP is the ADAU1452 series, and we're still waiting for enough block-processing algorithms in SigmaStudio to make good use of this power.

This being said, a fellow engineer on EZ did build a blind frequency measurement scheme using discrete filters -- see https://ez.analog.com/message/140809#140809

Best regards,

Bob

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