DSP Board Completed (DSP56371)
 

Took me a while but I've just finished the first revision of my DSP board with SE and USB inputs. This is a pretty nice project and I was thrilled to find that Freescale has a good evaluation board called the SoundBite and has Eagle files of that board to download.

I don't think this board is perfect -- and I still have to figure out how to solder all those 0805 resistors without going mad -- but I'd like if a you could take a look and make sure I didn't miss anything. I'm making the eagle files available for you to play and give better feedback; please be aware that I make no guaranties this will work. If you like this you are free to buy me something from my Amazon wish list :)

The DSP is set to boot of from SPI in slave mode as I will have another board for the UI (we're looking at a QVGA LCD with Atmel's qTouch like a QT60168).

Download DSP with RCA and USB & RevA (zip)

http://screenshot.ca/wp-content/uplo...AIN-Bottom.png

http://screenshot.ca/wp-content/uplo...-RCAIN-Top.png

Thank you for your comments :D I'm off to bed before I think of revision B... ahh too late - it's too large. 'll expose the ESAI with headers and put the ADC/DAC on another board that we will mount on top of the DSP board.

Nice work! Yeah I think it's a good idea to separate the boards and make things modular. I did so with my DAC and i'm very glad I did because it enables me to try out new ideas without throwing all the old stuff away.

I too think Its a good idea to use an ASRC like CS8422 to precede the digital signal processing, then, with fix clock/sample rate, you dont have to alter coefficients ever because of that. This DSP seems capable of some decent FIR stuff @ around 60khz. What you have in mind?

Not really much point in using the DSP if you plan to butcher the music with an ASRC first. Why on earth would you want to force everything from 44K1 to 192K into one arbitrarily chosen output rate arrived at with sub-optimal coefficients quite possibly chosen on a sample by sample basis.

I figured I can do the sample rate conversion in the DSP; insert N zero samples and low-pass filter the result. That is IF I want to resample the signal.

I dont think you want to go there, these IC-s are far more sophisticated than what you can come up within a ~year?? ASRC is , lets see, DPLL, ratio estimation, curve fitting, much much more complited than your average oversampling filter ! Basically you also spare the first stage of oversampling because of the ASRC chip, that is , not bad to start with.

Dont forget also FIR filters are grossly inefficient at 96khz .

ok then why dont you say he can just redesign his PCB around some different clock generation scheme, that can take up a year in itself. I doubt ASRC matters anything once he has crossover in mind, if thats IIR, he sets 200khz output from asrc and spare a lot, or , if he wants FIR, he sets 60khz. Not everyone is dCS or Wadia to mess around for years with an oversampling filter in itself! :fight:

lets see what Ed Meitner came up with :

MFAST™ vs. conventional PLLs

Most converters utilize PLL (Phase Locked Loop) circuits. MFAST™ has two distinct advantages. It's a high-speed asynchronous system that locks almost instantaneously to any data stream. Moreover, unlike PLLs which merely attenuate jitter, MFAST™ strips jitter out of the audio stream completely. Enabling you to enjoy pristine sonic clarity whether the incoming data stream is pure or anything but. The DAC2 also features:

:worship:

The DSP56371 has all the clock generation needed onboard.

I tought you just sent him to develop DDS clock generation, now I see the destination goes straight into jitter-hell.