I am currently using a Zedboard with the ADAU1761. I am a VHDL firmware engineer but I've been instructed to implement a bi-directional 8k audio interface with anti-aliasing.
Does the ADAU1761 do this out of the box?
Are there registers I can set to implement this or do I need to use SigmaStudio to program the device?
I apologise for my ignorance on the subject, I have very little (none) audio and DSP experience and having read the datasheet it's not clear.
Even a point in the right direction or where to start would help no end.
You will need SigmaStudio to program the DSP. If you use the part only as a codec then you can do it only with register writes.
The 1761 can operate at 8kHz fs. The anti-aliasing will have to be done as an analog filter coming into the part. The DAC is an oversampling DAC but you will still need some filtering on the output to filter out the out-of-band information.
There are some other parts that will sample at a higher sample rate and have built in sample rate converters that can go down to 8kHz. This would ease the difficulty of the analog filters and you can get away with a single pole filter.
Since it seems to me you are asked to design an interface, I am assuming you really do not need to do any DSP programming. Is that correct? Just convert analog to a digital signal and then a digital signal back to analog? The digital signal would be sent down a USB or Ethernet serial bus. Is that close to what you need?
Then I would take a long look at the ADAU1372. It will run the ADC and the DACs at 192kHz fs and the serial data ports can operate at 8kHz and it has sample rate converters to filter and convert the audio to the new sample rate. It is very low latency because of the internal speed of the converters but that advantage is somewhat lost running the serial port at 8kHz.