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USB to I2S 384Khz - DSD Converter

The reason why the author of the blog page tried a various bypass capacitors was that the behavior of residual noises was variable depending on USB cabling methods. I think his finding may mean your worries.

correct, thats what is the source of my worries. its only logical too, since there is a direct connection to the USB ground and therefor PC ground, not only that, in this case its used directly for analogue line output ground reference.

I had hoped a transformer would help, but on second thoughts it wouldnt, since its a single ended signal, so any perturbations on ground will still be reflected. its not a common mode error, so even if you can create the correct analogue LPF to get rid of the digital hash in the very high frequency, you will still be stuck with this high ripple on signal ground.

I'm sure there is some way around this, but its not as simple as I had initially hoped
 

opc

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I would hazard a guess that you could use fast TTL logic IC's to generate a differential output and run the transformer across that, taking ground out of the equation entirely, and increasing your output voltage swing at the same time.

All you need is a buffer on one side and an inverter on the other.

Deriving the actual analogue audio signal directly from the output of a CPLD terrifies me though. Sounds to me like the worst idea anyone has ever had. Just because something is simple, doesn't mean it's going to be good.

I kinda get the feeling the DIY community is overly susceptible to gimmicks like this because it's so incredibly easy to implement and you get to feel an immediate sense of satisfaction and the "wow it's making music" feeling after having only soldered a single resistor. It's like using the ESS DAC in voltage mode.

Either way, I'll reserve judgement until after I get my boards and measure / listen to it. I have to admit the idea sounds interesting, but I think it will require a little more effort than a 1k resistor to properly implement.

Cheers,
Owen
 
I would hazard a guess that you could use fast TTL logic IC's to generate a differential output and run the transformer across that, taking ground out of the equation entirely, and increasing your output voltage swing at the same time.

All you need is a buffer on one side and an inverter on the other.

Deriving the actual analogue audio signal directly from the output of a CPLD terrifies me though. Sounds to me like the worst idea anyone has ever had. Just because something is simple, doesn't mean it's going to be good.

ya funny you mention that, when I saw Ians new design for the daughterboard to isolate ground between the fifo buffer and clock/buffer board, I had the same idea.

^^post 980 just in case direct link doesnt work

I had the plan of trying this direct out thing with this board as one experiment to do with a cheap device from the beginning, but the idea of just using a resistor doesnt appeal. the idea of a high performance active analogue filter to do it is curious though, given that in native dsd mode thats pretty much all the dac is doing. its a novelty more than anything.

I kinda get the feeling the DIY community is overly susceptible to gimmicks like this because it's so incredibly easy to implement and you get to feel an immediate sense of satisfaction and the "wow it's making music" feeling after having only soldered a single resistor. It's like using the ESS DAC in voltage mode.
lol bad day at the office mate? :bomb: agree completely but..... haha :rofl: I even have a suitable naked zfoil resistor… that should make ground less..erm..bouncy? ;) urushi painted perhaps? (not specifically aimed at Feastrex, i've just noticed that Urushi paint seems to be linked highly with a number of overpriced (but well made) Japanese audio/artisan products lately)

Either way, I'll reserve judgement until after I get my boards and measure / listen to it. I have to admit the idea sounds interesting, but I think it will require a little more effort than a 1k resistor to properly implement.

Cheers,
Owen
my thoughts exactly, well...i'll leave the measuring to someone like yourself with the gear/knowhow to make a proper job of it. dsd itself doesnt ever measure that flash afaik, thus why I havent really pursued it till now, so this could be an interesting trainwreck, OR maybe not...
 
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... the idea of a high performance active analogue filter to do it is curious though, given that in native dsd mode thats pretty much all the dac is doing. ...

A Japanese skillful audiophile, "sotaro", shows his design idea for the combination of isolated analog FIR and analog Filter. (In Japanese)
Please remember this is designed for DSD256 only. When any of you applies this architecture to DSD128 or DSD64 output of Combo384, you need to recalculate some values.

I'd like to say that DSD256 is constantly establishing its good reputation in Japanese DIY audio communities. I wish Amanero Combo can support this in the near future.
 
There is no doubt that a LP filtered DSD signal is the most minimalistic way to do DSD Analog conversion.
And I do use it myself in my old XB940 SACD player.

But don't forget that using the DSD direct out for LPF means you have zero PSRR.
And even if you can make a perfect power supply for your digital source with no noise in the audio band and capable of supply 24 or 98MHz,
there is no guarantee the output of your MCU will give you back 100% that voltage output level digitally.

There is no perfect solution. Else it is too simple. And no fun.

:)


Patrick
 
There is no doubt that a LP filtered DSD signal is the most minimalistic way to do DSD Analog conversion.
And I do use it myself in my old XB940 SACD player.

But don't forget that using the DSD direct out for LPF means you have zero PSRR.
And even if you can make a perfect power supply for your digital source with no noise in the audio band and capable of supply 24 or 98MHz,
there is no guarantee the output of your MCU will give you back 100% that voltage output level digitally.

There is no perfect solution. Else it is too simple. And no fun.

:)


Patrick

see the last couple of pages of posts from myself and opc discussing exactly that PSRR issue and possible solutions. good to know you liked the board, I expect mine here any day now
 
While I give you full credit of pointing out the PSRR issue, I think you missed the key points of my message.
The solution is NOT a perfect power supply for the MCU, which of course does not exist in the first place. :)

And yes, you can use a Class D amp. Perhaps with lots of feedback and some LPF at the output somewhere.


Patrick
 
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Joined 2009
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BTW we had the first batch of the Amanero before the GB.
We tested every functionality successfully without any problems.
We can only say we are very pleased with it.
So our most sincere congratulations to Domenico for a great product.

;)


Nice weekend to Italy,
Patrick
Good to hear. When I first saw the Amanero board I actually thought that it was kind of XEN style. Simple, solid, versatile, but without all the bells and whistles typically seen on these USB converters. It seems that it does one thing and does it well.
Not least, the sound is absolutely impeccable:)
Cheers,
Nic
 
While I give you full credit of pointing out the PSRR issue, I think you missed the key points of my message.
The solution is NOT a perfect power supply for the MCU, which of course does not exist in the first place. :)

And yes, you can use a Class D amp. Perhaps with lots of feedback and some LPF at the output somewhere.


Patrick
i'm not just talking about a perfect power supply for the ARM either, it'll still no matter what you do, have pretty serious ripple on ground simply because of the transient current demands and perhaps simultaneous switching noise of the chips used. i'm talking about decoupling the output from ground and using a fast analogue active filters and differential clock drivers/buffers of the kind used for LVDS, CMOS etc

but i'm not pushing it as being superior to anything, I just find it an interesting idea worth playing with and a cool little thought experiment.
 
current draw

if it is of any interest to anyone, the draw from my external 3.3V supply (supplies everything on the module) is around 120 mA for 44.1 khz.

compared to my hiface with 3 seperate salas supplies, (clock, modules and 1.8V), my single battery Combo dont have the dynamics and refinements of the hiface, but it does sound nice, given the size. But again, it is not fair as good supplies really make a difference.
 
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Joined 2007
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If you allow a multi-bit delta-sigma modulation for PCM, TI BB DSD1794A(and its family models) may meet your conditions because only analog filter is applied to DSD input signals under their "Advanced Segments" architecture. DSD1794A supports DSD64, DSD128 & DSD256.

Hi Bunpei - thanks for the suggestion but it doesn't go to 384 kHz as far as I can see from the datasheet ...? My main interest for this DAC is to have a DAC that can output PCM in very good quality (but not DSD conversion) and then have a very good DSD capability as well. Regarding the PCM1794A are you then sure that it can handle DSD256? It looks as if it accommodates various clock frequencies but only DSD64 ... (Electrical characteristics page 2).

Might there be a DAC that has all of it :) DSD256 & 384 kHz/24 bits.

For you who are most kind to reply please note that I need two DACs. One for monitoring purposes - which needs be very good - and later, one that is as good as I imagine they can get (IMHO)...

Evening greetings,

Jesper
 
Just on short listen, I can't hear any difference between this and the exaU2I.


Lucky you :)

Installed mine tonight. Didn't like wasapi under Win 7 64 using J River 17 much. KS played a bit better.

Fair comparison to EXA is not really possible as my EXA board already uses one outboard regulator for the clocks and will soon get a second one. As is, the comparison is not тоо flattering for the Amanero board: fine details and especially sound stage cues are missing. Overall the sound is of high quality and very listenable. Perhaps on par with Lorien's board with driver set to standard latency.

So far i've only tested PCM source material. The idea that DSD will have to play under WASAPI is not very appealing but so far i have not tested it.

Before further tests i will remove the onboard regulators and supply external power, either from a Jung super-reg or a Salas shunt.

As value for money and flexibility though, it is indeed fantastic.
 
... but it doesn't go to 384 kHz as far as I can see from the datasheet ...?
I assume you have a certain level of technical skills. An external digital interface of the DAC chip accepts 384 kHz/24 bit. M2Tech Young DAC employs this and achieves 384 kHz/24 bit play.

Regarding the PCM1794A are you then sure that it can handle DSD256?
Yes, sure. I have tested it. As for DSD1794A, two Japanese users have done that.