For a feedback suppression algorithm I am looking to implement variable gain / rejection / frequency notch or band-stop filters.
On other words, I need to be able to control the center frequency and amount of rejection dynamically.
What would be the best way to implement this on an ADAU 1761?
Hello Roman,
Your final application will influence which of several approaches might work for you. I'll try to describe these methods with their advantages and limitations.
Second-Order IIR
…The spreadsheet is attached below
Hi KJBob,
thank you! for your comprehensive answer!
I got some important pointers from it - like dynamic calculation of coefficients not being possible on the DSP itself (what a shame btw! Can't I do…
Hello Roman,
Your final application will influence which of several approaches might work for you. I'll try to describe these methods with their advantages and limitations.
Second-Order IIR
These are by far the most versatile and capable, but also the most difficult to control in real-time:
Problem is, to adjust these filters at runtime you'll need an external means of control -- either SigmaStudio via the USBi, or a microcontroller in your final product. This is because the five IIR coefficients needed to drive these filters have calculations too complex for a SigmaDSP to do on its own. So instead you have your microcontroller calculate the needed coefficients for a desired response, then have it safeload them into the DSP's parameter memory via I2C or SPI bus. I attached a spreadsheet with formulas to calculate coefficients for various filter types -- it's fairly straightforward to translate them into C code for your microcontroller. To troubleshoot your own calculations, enter them into the IIR Coefficient version of the filter, as with the notch filter shown below:
Once you've calculated your coefficients, safeloading them has been covered in various forum topics -- for example, https://ez.analog.com/message/61175#61175
State-Variable Filter
This is the only SigmaStudio filter for which the DSP figures its own coefficients, thus allowing filter control to originate within a signal flow. It has simultaneous lowpass, highpass, and bandpass outputs. It's possible to create a notch by subtracting its bandpass output from the original signal, but tricky to obtain a deep notch this way.
Communications-Grade Notch
If your application is in communications with a narrow voice bandwidth, you can borrow an idea used in radio since the 1930's -- it's far easier to move the incoming signal's frequency than to move the filter's frequency. See an example of this technique at https://ez.analog.com/message/173545#173545
Best regards,
Bob
Hello Roman,
Your final application will influence which of several approaches might work for you. I'll try to describe these methods with their advantages and limitations.
Second-Order IIR
These are by far the most versatile and capable, but also the most difficult to control in real-time:
Problem is, to adjust these filters at runtime you'll need an external means of control -- either SigmaStudio via the USBi, or a microcontroller in your final product. This is because the five IIR coefficients needed to drive these filters have calculations too complex for a SigmaDSP to do on its own. So instead you have your microcontroller calculate the needed coefficients for a desired response, then have it safeload them into the DSP's parameter memory via I2C or SPI bus. I attached a spreadsheet with formulas to calculate coefficients for various filter types -- it's fairly straightforward to translate them into C code for your microcontroller. To troubleshoot your own calculations, enter them into the IIR Coefficient version of the filter, as with the notch filter shown below:
Once you've calculated your coefficients, safeloading them has been covered in various forum topics -- for example, https://ez.analog.com/message/61175#61175
State-Variable Filter
This is the only SigmaStudio filter for which the DSP figures its own coefficients, thus allowing filter control to originate within a signal flow. It has simultaneous lowpass, highpass, and bandpass outputs. It's possible to create a notch by subtracting its bandpass output from the original signal, but tricky to obtain a deep notch this way.
Communications-Grade Notch
If your application is in communications with a narrow voice bandwidth, you can borrow an idea used in radio since the 1930's -- it's far easier to move the incoming signal's frequency than to move the filter's frequency. See an example of this technique at https://ez.analog.com/message/173545#173545
Best regards,
Bob
