Suggestions needed for software, interface etc.

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I have a doubt about it,
Isn’t it possible than when doing that loop that the output circuitry of the soundcard (DAC and analog section) already limits the frequency response and that the input measures a lower kHz limite because the output can’t deliver that.

just to be sure shouldn’t the first tests of the input stage done with an external signal generator already known to be able to deliver the full sweep?

thanks
The idea in loopback is to evaluate the frequency response of the combined ADC and DAC.
Actually, the frequency response of both is usually determined by digital filters that are very abrupt (very high slope) and essentially similar.
Indeed you could measure them separately, but it requires test equipment that you may not have at hand.
 
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The idea in loopback is to evaluate the frequency response of teh combined ADC and DAC.
Actually, the frequency response of both is usually determined by digital filters that are very abrupt (very high slope) and essentially similar.
Indeed you could measure them separately, but it requires test equipment that you may not have at hand.

I have a signal generator that is able to do sweeps and goes quite high in frequency, I can try with that
 
Of course, second-hand lab gear is an excellent solution.
I still use my old (1970's) LEA generator and THD meter, with a Hameg. It takes about half the space of my dining table, though, which is not small.
Absolutely, I would be lost without my old Ferrograph test set which goes right up to 150KHz.

Cheers

ian
 
I use the venerable EMU0404USB, which given all the new devices that have come on the market in the last decade or more is still a standout. I'd have to go dig through Scarlett test reports to see if anyone has tested that in a similar way, as the Focusrite range would have to be the most common modern USB device that is relatively cheap and often available 2nd hand in its various revisions (and would be my goto if I came across a really cheap one).

The headphone open-circuit output voltage level droops 10% at 90kHz, and 30% at 94kHz in my 0404. The unbalanced input has a higher drop-off rate above 90kHz, so is the portion of the loopback that requires more REW compensation. And yes that type of measurement out to 100kHz requires lab grade metering.
 
I'd have to go dig through Scarlett test reports to see if anyone has tested that in a similar way, as the Focusrite range would have to be the most common modern USB device that is relatively cheap and often available 2nd hand in its various revisions (and would be my goto if I came across a really cheap one).
I believe Ian has one; he could chime on it...
 
TTLV320ADC6140 quad-channel 768-kHz Burr-Brown™ audio ADC evaluation module is also an option , it comes with an xmos usb controler card , every parameter is controlable in a very comprehensive software called pure path console...

...Im very tempted to try one of the E1DE Gen3 DAC's and use it as source for test signals, with the calibration available on the chip itself via the app it should be able to offer up an extremely accurate low distortion test signal to well beyond 100khz.
I wonder how the TI EV board would work in conjunction with a DAC such as this one
https://www.amazon.co.uk/s?k=topping+d50s&ref=nb_sb_noss_2or that one
https://fr.shenzhenaudio.com/produc...bit-768khz-dsd512-decoder-with-remote-controlHow would the software fare with two different drivers, one for the ADC, the other for the DAC?
Does anybody know?
I think I'll ask on the REW forum...
I believe I could try to run REW with two soundcards, one for send, the other for receive.
It would not go well in my set-up, since the Dell notebook has only one USB port; I would have to add a USB hub, or change computer...
EDIT: John Mulcahy (REW) suggested I use ASIO4ALL or FlexASIO to agregate the two "soundcards".
I'll start with that, with my two "ordinary" soundcards.
 
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I hadnt heard of Flexasio before .

I have tried Asio4all but I couldnt get it to work in 24/32 bit modes .
Flexasio does appear to allow 24/32 bit .
Let us know how it works out .
 
How would the software fare with two different drivers, one for the ADC, the other for the DAC?
Does anybody know?

I don’t know how it works in windows, but on a MAC that’s really easy to do. It recognizes both soundcards and you can make an “aggregate device” with the 2.
Then on REW you just choose the “aggregate device” as your driver, it takes care of the Clock Sync between both units also.

maybe there’s something similar for Windows
 
I don’t know how it works in windows, but on a MAC that’s really easy to do. It recognizes both soundcards and you can make an “aggregate device” with the 2.
Then on REW you just choose the “aggregate device” as your driver, it takes care of the Clock Sync between both units also.

maybe there’s something similar for Windows
That's one thing a Mac does native, but Windows must use a BandAid.
 
https://www.lifewire.com/best-usb-hubs-4142295
With a USB 3 hub it should be less of a bottleneck and should be easily capable of handling two audio interfaces . The ones with an external power supply might be better than using the laptop supply which is often a source of extra noise in audio interfaces .
That's interesting. Many people suggest using a USB powered interface in order to avoid potential issues like groundloop and extra noise.
 
I guess it varies alot depending on hardware and what filters ,regulation and dc-dc conversion is involved , but in general Id favour an audio interface on its own dedicated supply anyday , thats not to say RF mush cant get superimposed on the audio path via ground either . Be interesting to take a standard USB powered interface and check how the noise figures stack up from a really quiet 5 volts vs hashy 5v mobo supplies .
 
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