hi..i'm thinking of using adau1452 board as a complete dsp in my audio system..it is possible? there is a way to impreve hardware performance of this board?
I'm interested to change sampling rate from 48khz to 192khz..it's possible?
The ADAU1452 will happily run at 192K, but the eval board's AD1938 codec is hardwired at 48 KHz -- forcing the whole thing to operate at this sample rate. To fix this requires cutting traces and hand wiring to get the codec off its hardwiring and onto the SPI bus, in order to render its features programmable. I wouldn't try this myself, but at least one brave soul has modified their ADAU1452MINIZ this way -- see https://ez.analog.com/message/76515#76515
Another limitation is that the board's op-amps, etc. operate on a 5V supply -- although with modern-day silicon this is not as bad as it once was.
Power is not a problem..I use a voltate regalato from 12v to 5v
Power is not a problem...I will use a voltage regulator from 12V to 5V
So i have to program ad1938 over its SPI interface? I need EVAL-ADUSB2EBZ ( the usb programmer included in the adau1463 evaluation kit) and SigmaStudio?
A EVAL-ADUSB2EBZ comes with all SigmaDSP eval boards, including the ADAU1452MINIZ kit. You'll need it to program the ADAU1452 as well as the AD1938. If you plan on running the board as stand-alone (self-booting) in your final application, the USBi is also involved in programming the ADAU1452MINIZ's E2PROM. In this case, the ADAU1452 can run as a bus master to set up the -1938, although someone else would have to help you as I lack experience with this arrangement. Perhaps when ADI gets back from their shutdown they can be of more assistance.
Generally, modifying the board is not a good idea. Like all surgery it's difficult and not guaranteed successful. I'm quite happy running my board at 48 K, although some applications may call for more. As for the op-amps, what I mean is that 5V doesn't give you the dynamic range available with good old + and - 15V rails as used on older analog equipment -- however, today you can do about as good with the latest chips at 5V.
So the solution is to wire the SPI port of ad1938?
Yes, the reference I have linked to above describes rewiring the AD1938 to allow programming its function (though I have not tried this myself).
In another post you had mentioned that FLAC has poor quality at 48 K. This makes me wonder if perhaps you're confusing the bit rate of compressed audio with the sample rate of linear PCM audio. SigmaDSPs handle a 24+ bit word at each sample. Even at 48K this is better than CD quality, which is 16-bit words @ 44.1 K/s. BTW, presently no SigmaDSP can decode compressed audio.
Hi..thanks for your answer..I think 24bit 96khz Is the best solution..So i'dd like to get this..
For compressed audio decoding there is the audio player..from audio Player its go into SigmaDSP
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