High Resolution Multi-Channel Digital Interface

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DSD - PCM MuxIt Transmitter Gerbers
(.zip of 274x)

Note that I ended up isolating pin 74 of the SM5816AF (XMTPCM). I then connected this pin to the DSD mode pin of a Sony DVP-NS500V SACD player.
(The Sony had garbage on the DSD lines when it wasn't playing an SACD. This mod mutes the output when an SACD isn't being played.)
 

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Michele Spinolo said:
Great work Brian!
Thanks Michele!
Could you please post the Gerbers of the DSD to PCM conversion board too?
It took me a bit to dig up these old files, but here they are.
On which player did you succesfully installed the boards?
A Sony DVP-NS500V and a Pioneer DV-563A.
Did you then give a try to SM5819AF DSD->PCM conversioin chip??
Not yet.

The Pioneer DV-563A has much easier access to its DSD and PCM signals, because it brings them across an FPC cable. I'd like to try making a SM5819AF based board that would directly plug in.

No, I'm not making any time estimates... :xeye:

Regards,
Brian.:cubist:
 
Really impressive DIY work.

Regarding the use of ASRC, I wonder if you are interested in implementing flow control somehow with different physical link than LVDS. 1394 or USB would allow it, but I know the 1394 implementation is terribly complicated, and don't have a good idea on the complexity of USB for this purpose. However, if it can be done, you can get rid of the ASRC's and its possible negative effect on sound quality.

I also really wanted similar kind of systems, incorporating full digital processing for all CD/DVDA/DVDV/SACD signals, room correction, and most importantly digital active crossover for my future speakers...^^;
I have been thinking of different approach, as I was not convinced in the superiority of D-amp solution to analog high-end. I (un)fortunately have a big-*** analog stereo amp, and that's keeping me from full-digital amplification path, at least for now.
So my plan was to buy a processor with 1394, along with a universal player with 1394 audio output, and modify it to incorporate the wanted features. I might end up having to add another DSP with my custom features.

There are so many different ideas going on, but I am really glad to see people really pursuing this with real works. I hope we can share info. and ideas on this pure digital audio path. :)
 
" ... 1394 or USB would allow it, but I know the 1394 implementation is terribly complicated ..."

I don't know why this keeps coming up, but 1394 is actually less complicated than USB. (Peer to peer, slimmer address space, bi-directional double duplex (twice as many wires as USB), but 4 times the bandwidth.)

As long as your not planning to re-invent any wheels or writing a lot of mods to existing ACISs, you just use a decent TI 1394b hub chip and a choice audio ASIC.

These guys introduce 1394 DACs on a regular basis for pro musicians and studio work, just about every month: http://m-audio.com & http://rolandus.com

They simply pick from the litter of 1394 audio ASICs like this: http://www.oxsemi.com/oxford/documents/pages/audio/audio.html ... then glue on some knobs & LEDs and package it nicely. Their latest is: http://www.m-audio.com/products/en_us/ProFireLightbridge-main.html

This is something USB just can't do: multi-channel, bi-directional, 24 bit / 96k or better ...

" ... I also really wanted similar kind of systems, incorporating full digital processing for all CD/DVDA/DVDV/SACD signals ..."

Get a computer with a decent DVD player (of "universal" capabilities) and then build your 1394 DAC / A2D and plug 'er in.

(A mercenary announcement: http://industrialcomponent.com/firewirestuff/fws46603.html ... $24 and the 6-pin port card fits any PCI bus equipped machine, Windows or Mac = no drivers required.)

:smash:
 
Hi,

@Brian (or anyone who can help.) :)

I'm just trying to layout some PCB designs for muxit interfaces using SRC4192's, but I'm a bit stuck about how to configure them.....

I see in the datasheet that they don't need to be linked for matched phase, but how should I set the MODE pins? I'm guessing the first chip should be set as master and the rest of the chips set as slave seeing that the output BCLK and LRCLK pins are tied together?

The crystals on the Audiotrak Prodigy I will be using as a source are 24.576MHz and 22.5792 - does this mean I should be using 512fs or 256fs mode?

I've attached a screenshot of how I have the chips connected atm (no master clock or reset yet etc.).

Also, is your master clock actually on your receiver board, or is it taken from your amp's master clock? What speed clock are you using on your receiver board (I can't quite make it out on the pdf file.)

I'm also assuming that if I was to modify my sound card with a muxit transmitter, is it a good idea to have a low jitter clock on the sound card as well as a low jitter clock for the master clock at the receiver end?

Another area of confusion is - do you use the same low jitter clock for both the ASRC reference clock and equibit amps?


Thanks,
OzOnE.
 

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Ok, I think I'm getting my head around some of it.....

I've set the first SRC4192's output port as master / 256fs giving me a 96KHz output sample rate when using a 24.576MHz reference clock.

The rest of the chips are set with both ports as slave, so they will receive the BCLK and LRCLK signals from the first chip.

So, the master clock on the receiver board WILL have to clock the DAC (or digital amp) section as well as the ASRC chips?

But, does this also mean that the input ports can be slaved, and they don't need a master clock from the LVDS receiver chip (because the ASRC is reclocking everything anyway?)

Also, I've grounded the TDM inputs on all chips - is this correct?

Plus, I'm probably going to add a multiplexer to the receiver board so I can switch between an external input and the amp's own inputs (from it's ADC's / DSPs) - can anyone suggest which type of logic to use for minimal jitter (I don't think it will matter too much though.)


OzOnE.
 
A glaring error I think I'm making is that I assumed the BCLK and LRCLK signals on the output ports were being generated by the ASRC chips themselves.

I'm thinking now that all the chips are slaved from the DAC / Equibit's own master clock, so the BCLK and LRCLK are in fact the "inputs" to the output ports on the ASRCs.

The BCLK and LRCLK signals mainly serve to allow the ASRC to sync the output data to the DAC's clock signals. (I'm probably wrong again here.)

So, your super-duper low jitter Kwak / Tent / Platypus / Tipi clock (only kidding) can still be connected to your DAC, and the ASRC chips will simply sync their data outputs to it.

But, does that mean the reference clock is literally a reference, and is asyncronous to the input and output ports?

OzOnE.

EDIT: BTW, fantastic work Brian! Without this thread I would never have dreamt of trying something like this. I think this type of interface should be made more commonplace sources / amps and should have been standardized way before the advent of HDMI.
 
OzOnE_2k3 said:
A glaring error I think I'm making is that I assumed the BCLK and LRCLK signals on the output ports were being generated by the ASRC chips themselves.

I'm thinking now that all the chips are slaved from the DAC / Equibit's own master clock, so the BCLK and LRCLK are in fact the "inputs" to the output ports on the ASRCs.

Correct.
The way I am using them, and the way I would probably recommend for most situations, is for the ASRC to be run as slaves.

The BCLK and LRCLK signals mainly serve to allow the ASRC to sync the output data to the DAC's clock signals. (I'm probably wrong again here.)

Actually, you are correct here.

So, your super-duper low jitter Kwak / Tent / Platypus / Tipi clock (only kidding) can still be connected to your DAC, and the ASRC chips will simply sync their data outputs to it.

Yes, and an important point here is that the critical low-jitter Master MCLK signal that is feeding the Equibit or DAC IC doesn't even have to go anywhere near the ASRC. This is especially useful if the ASRC is being retrofitted on a separate board.

But, does that mean the reference clock is literally a reference, and is asyncronous to the input and output ports?

Correct. IF the ASRC is being used in slave mode, the reference clock is used to initialize the ASRC, and to drive the interface logic and rate ratio estimator circuitry. It isn't directly used to time any of the audio clocks.

(If the ASRC is in master mode, then the reference clock would also be used to generate the audio output clocks, which is why I usually wouldn't want to use it that way.)

Regards,
Brian.:cubist:
 
I'm still trying to find an HDMI Breakout Box / DAC ...

The new Apple TV device (US$229) only has HDMI, optical audio and RCA output for connection to stereo or multi channel equipment. Since that leaves something to be desired for us "golden ear" types, I would really like to find some way to extract that 24 bit / 192K, dolby 5.1 off of the cable.

Anyone have a suggestion ??
 
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