Getting started with DSP for audio - sanity check please

[BCZ]

So I would like to get started with a DSP for audio processing for a speaker. Please let me know if what I'm doing makes sense and not crazy. And feel free to offer any suggestions and improvements.

I want to build a DSP to implement some active filters and compressors and FIR for 8 channels of audio. Here is what my DSP workflow will look like, except with some differences such as I'll add another FIR filtering right after the input and a multi-band compressor at the end of the chain, and of course a different input.



I'm thinking of getting a EVAL-21489-EZLITE board and use SigmaStudio and all their built in libraries to build what I want.

Will this be as easy as just buying the evaluation board, hooking it up to my computer, drawing the DSP workflow on SigmaStudio, hit compile and boom it'll work?

Also, I noticed in SigmaStudio, the bass synth algorithm is available for the ADAU1701, but not available for ADAU1452 or ADSP214xx. Anyone know what's up with that? In fact, the ADSP214xx is missing a lot of algorithms that are available to the ADAU1701 and ADAU1452.

Thanks