I have a nice working mixer recorder prototype working well using ADAU1467 with 16 channels
analouge IO. ADC DAC etc.
I'd like to make an AES3 digital IO card to add to the design.
I see in the ADAU146x datasheet " The S/PDIF interface meets the S/PDIF consumer performance specification. It does not meet the AES3 professional specification. "
Does Analogue Devices make a multi-channel capable AES3 IC ? transceiver , transmitter or receiver that can interface to the DSP ?
Is there a common way to interface AES3 channels to ADAU146x using AD chips ?
Unfortunately, we do not have anything in the SigmaDSP family that could so AES3. You would have to go to a SHARC or a BLACKFIN to get that capability. I do not work on those parts so I am not sure if there are any off the shelf solutions available either as an example or from a 3rd party developer. You would have to ask this question on that part of the forum.
> Does Analogue Devices make a multi-channel capable AES3 IC ?
AES3 is a stereo format that is very similar to S/PDIF. ADI supports multi-channel digital audio standard on processors in the SHARC family.
The two major differences are the physical layer (generally 110 ohm balanced vs. 75 ohm coax), and the subcode. The transport layer is the same. The ADAU146x parts support most of AES3, but the decoder does not automatically interpret the subcode as it does for S/PDIF. However, all of the user and status bits are extracted from the stream and buffered.
The biggest reason that we don't claim to support AES3 is that the Rx/Tx blocks are not tested to AES3 standards for jitter generation and tolerance. They may well (likely) meet those specifications, but the ICs have not been characterized over temperature and process as they have for S/PDIF. Also, the SigmaDSP transceivers do not support time code. You may want to do some tests to determine of the transceivers meet your needs.
If you do use the S/PDIF transceiver, note that you must either (a) clock the entire core at the received sample rate, (b) force the sending device to be synchronous with the DSP by using a common word clock, or (c) use an ASRC in the signal path losing bit accuracy (still a high quality solution).
Hi Ken, thanks for your reply.
I had assumed I would need to use an ASRC in the signal path.
I found a solution for 8 receive channels. I can daisy chain CS8422 ( ASRC with receiver ) to TDM input on ADAU1467.
I can live with simple 2 channel output of AES3 mix for now.
I think SHARC with integrated ARM A5 die may use to much power for the design i'm working on?
Is there a low power SHARC DSP based on M7?
I am not familiar with SHARC IDE , I think its commercial IDE based on eclipse and approx 8000 usd +?
Is there demo code for LTC timecode IO, multichannel audio recording, SSD drive support in the IDE ?
is there some kind of real time OS support ?
I'm using arm M7 low power MC connected to ADAU1467 for mixing and recording which is working well but it will have limitations.
I will implement LTC timecode next but that has to all be done in the MC ( as you say , no timecode support in sigmaDSP )
maybe this is already all done in SHARC DSP and i am re-inventing the wheel ?
I'm interested in learning more about the SHARC dev environment, where is best to start ?
This is by far the best deal for SHARC development.
I do not know some of the other questions you are asking about. You should search the SHARC forums for some of these answers.
Thanks Dave ,
I'll take a look at the SHARC forum.