Introducing Ground Sound

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Myth?

4real - You are quiet right – ”a long while” is a bit of an understatement, but it isn’t a myth though. :D

Facts are that the hardware is working as proto type module, but haven’t been tested with audio in normal crossover operation mode – only in “direct through mode” and I haven’t evaluated this for sound quality. :rolleyes:

I really hope for at presentation on the homepage in January, but…no promise.
I could put a picture here, but it will properly lead to more questions and more excuses…. :violin:

Okay – one promise – I will reveal a picture here the 24.dec 3.00 PM C.E.T. – a little Christmas present from me to you. :santa2:

Robert
 
TNT said:
Maybe you should put a webserver in there and a web interface + the XIwizard - then anyone could run the box from any OS.


TNT – why can’t you just use a Slim Device like everybody else? I’m not especially interested in making the product more expensive – just to add another feature that many got already. You’ll have to settle with the Wintendo OS or simulate it – that’s how the world works today. ;)

- Too many are satisfied with the MS dominance and I'm not going to start the revolution! :D

Did I misunderstand your question – did you want the web server thing only for controlling the XOverWizard? This would be totally overkill to have such a control for something you very seldom will be using. :confused:
 
4real said:
Wow, looks kind of huge!

Am I correct in thinking that this one is fully balanced? I see an unconnected pin at every in- and output.


4real – no it’s not balanced al the way. The analogue input is balanced to the ADC and after the DAC it’s “only” balanced to the end of the DAC output filter – again why add extra circuitry when it’s not necessary? - Not really my stile. ;)

There is a pseudo balanced output after the relays – which works just as well as a “real” one when long signal cables are used. It’s a matter of taste I guess. :cool:
 
Robert GS said:
4real – no it’s not balanced al the way. The analogue input is balanced to the ADC and after the DAC it’s “only” balanced to the end of the DAC output filter – again why add extra circuitry when it’s not necessary? - Not really my stile. ;)

Well, since all my components are fully balanced I would not mind having a fully balanced filter.

There is a pseudo balanced output after the relays – which works just as well as a “real” one when long signal cables are used. It’s a matter of taste I guess. :cool: [/B]

I guess, it will be good enough :)

Can you comment on the used DAC chips? Same as the old board? How does the volume control work? It it the control integrated in the DAC chips?
 
I know that the converters are all balanced, but the volume control, buffers and the relays will be the double requiring more surface, power and extra passive components.

The DAC will be PCM1798, which should be quite adequate for this job and it haven’t got volume control on board. The volume control is a separate chip – it will be fun to evaluate it. :D Very soon :D
 
4real said:
Seperate volume control? There are not a lot of options I guess, it could be either a BB or a Wolfson chip? Or do you have something special?


You should be able to spot the volume control very easily just above the second DAC output row – but it isn’t a BB or Wolfson chip!!! :cool:

It’s a LQFP-48 chip -> 8-channels. :) Now you'll have to guess a little ;)
 
4real said:
So you're just throwing bits away for volume control ?

I guess, doing that for 24 bit is fine, but what about 16 bit sources via SPDIF? A 60 dB decrease in volume will only leave you with 6 bits of resolution (14 bits for 24 bit sources).


Sorry - I really don't understand you calculation. :confused:

A digital input will be full range 16bits or 24bits and handled as a 24bit signal in the DSP + full resolution output in the DACs, then the volume will attenuate the level as you see on the display. Always – something in dB. ;)
 
TNT said:
Doing a volume control rigth, having 24 bit available, it should be possible to "transpose" the 16 bis to represent a lower level and thus, not sacrifying any resolution !?



Sorry - I really don't understand your "transpose". :confused:

You simply set the DAI receiver to an output format of your choice - in our case I2S 24bits as the rest works in I2S! :)

If the DAI receive a 16bit resolution signal - the msb of the incoming will be the msb of the 24bit output, but it will not have better resolution than 16bits.
 
Robert GS said:
A digital input will be full range 16bits or 24bits and handled as a 24bit signal in the DSP + full resolution output in the DACs, then the volume will attenuate the level as you see on the display. Always – something in dB. ;)

I guessed that you would do the volume control in the DSP ;) Turn out you don't!

A bit of google brings me to CS3308. Am I right? You might also go for the PGA4311 (twice) better specs and it might even be cheaper :)

About the sound I don't know. -110 dB THD should be more than adequate. Anyway, right now, I'm spoiled with a Relay volume control ;) I'd have to sacrifice it if I'm going for the groundsound module. I guess, it will bring me much more benefits than drawbacks :)
 
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