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26th September 2021, 01:59 PM  #10471  
Banned
Join Date: Sep 2021

Quote:
That paper is about the resolution of ADC and not DAC. Measure the accuracy of an ADC is not as simple as DAC, it needs to find "decision levels", i.e. input voltages at all code boundaries. In a DAC, there is one unique output voltage for each digital input code, regardless of DNL or INL errors. An ADC, on the other hand, does not have a unique voltage input corresponding to each output code—there is a small input voltage range equal to 1 LSB in width (for an ideal noiseless ADC) that will produce the same digital output code. This is called the quantization uncertainty, and it can be the source of confusion when specifying and measuring ADC static transfer characteristics. Directly measured code transiction method only works well if the ADC (and the input voltage source) has an effective peaktopeak inputreferred noise which is less than 1 LSB. Larger amounts of inputreferred noise tend to mask the transitions and make the measurement increasingly difficult. A computerbased ADC servoloop tester is needed. This test method allows for data averaging to remove the effects of excess noise. 

26th September 2021, 03:00 PM  #10472  
diyAudio Member
Join Date: Aug 2005
Location: Toronto

Quote:
The effect is the same in the ADCs and DACs, uncertainty in the ADC digital output matches the analog output level uncertainty. That is, when your partner is testing the DAC with his 6 1/2 voltmeter, the LSBs are flickering, no way to figure out the real value, other than statistically, with a certain probability, something your DIY partner clearly did not do. The peak in such a measurements distribution histogram would give the peak resolution, which is what apparently your DIY partner is obsessed with. If I recall correctly, the average (not the peak) of these measurements will give the effective resolution. Of course, the voltmeter has to have a resolution better than the peak resolution of the DAC. Data averaging, as much as analog averaging DOES NOT remove noise or any of its effects. The only way to remove noise is filtering in a limited bandwidth, or synchronous averaging (or averaging in the time domain, or autocorrelation if you prefer) which do not apply to spot values like the digital ADC or analog DAC outputs. To make it crystal clear, when you get a signal spectrum in your favorite software controlling a sound card, averaging the result does lead to a sharper image, but the noise is the same! To lower the noise effect you would have to lower the resolution bandwidth (or increase the process gain if you prefer). Last edited by syn08; 26th September 2021 at 03:16 PM. 

26th September 2021, 04:07 PM  #10473 
Banned
Join Date: Sep 2021

This is again wrong for a binary weighted DAC like the one presented in this thread.
Firstly I have no partner. Secondly I have read 7 1/2 digit DMM (Keithley 2001) and "Median" in the reports, which means the value separating the higher half from the lower half of a data sample, a population, or a probability distribution. Moreover it would be useful to consider the property of superposition, and be able to recognize its signature. Mathematically, superposition, a property of linear systems, implies that, if the influences of a number of phenomena at a particular point are measured individually, with all other influences at zero as each is asserted, the resulting total, with any number of these influences operating, will always be equal to the arithmetic sum of the individual measurements. For example, let us assume that a simple binaryweighted DAC is ideal, except that each bit has a small linearity error associated with it. If each bit error is independent of the state of the other bits, then the linearity error at any code is simply the algebraic sum of the errors of each bit in that code (i.e., superposition holds). In addition, by using endpoint linearity, we have defined the linearity error at zero and fullscale to be zero. Thus, the sum of all the bit errors must be zero, since all bits are summed to give the all "1"s value. The bit errors can be either positive or negative; therefore, if their sum is zero, the sum of the positive errors (positive summation) must be equal to the sum of the negative errors (negative summation). These two summations constitute the worstcase integral nonlinearities of the DAC. Intelligent use of superposition, coupled with a complete understanding of the architecture of the DAC under test, generally allows for a reduction in the number of measurements required to adequately determine DNL and INL. Superposition generally holds true for binary weighted DACs. Because of superposition, it is not necessary to measure the linearity error at all codes. Therefore, absolute accuracy of the voltage measured at the DAC output is not required. 
26th September 2021, 04:48 PM  #10474 
diyAudio Member
Join Date: Aug 2005
Location: Toronto

Sorry, any further dialogue is impossible. You, like your non existing DIY partner, choose to ignore reality and replace it with your very own construct. I'll let Andreas know you are doing a fine job while he's MIA.

26th September 2021, 06:16 PM  #10475  
diyAudio Member
Join Date: Aug 2005
Location: Toronto

For those wondering, ain't search engines a biatch...
Quote:
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26th September 2021, 10:15 PM  #10476  
Banned
Join Date: Sep 2021

Quote:
In a DAC, we are concerned with two measures of the linearity of its transfer function: integral nonlinearity, INL (or relative accuracy), and differential nonlinearity, DNL. Integral nonlinearity is the maximum deviation, at any point in the transfer function, of the output voltage level from its ideal value which is a straight line drawn through the actual zero and fullscale of the DAC. Differential nonlinearity is the maximum deviation of an actual analog output step, between adjacent input codes, from the ideal step value of +1 LSB, calibrated based on the gain of the particular DAC. If the differential nonlinearity is more negative than –1 LSB, the DACs transfer function is nonmonotonic. 

26th September 2021, 11:08 PM  #10477 
diyAudio Member
Join Date: Aug 2005
Location: Toronto

And how is this relevant to the Andrea/Roberto measurements? Hint: it is not, the definitions above are text book correct, what is incorrect is the use of the numbers provided by the Keithley 2001 71/2 DVM.
You cannot determine 1 LSB at DC without advanced statistics as mentioned. Again, the difference is as between the peak resolution and the effective resolution. You are hoping to measure the former, but you only can measure the latter. 2.7bit difference, that's a lot. 
27th September 2021, 12:28 AM  #10478 
diyAudio Member
Join Date: Oct 2008

Get a room, you two!

27th September 2021, 07:23 PM  #10479 
diyAudio Member
Join Date: Apr 2008

Wow, even after being placed on my ignore list, andrea_mori still somehow manages to deliver the goods.
*** Anyway, moving right along. Nobody has posted their build here for months! I finished up my dam1021 rev7 build a couple of weeks back, so here it is. Deliberately minimalistic, I almost called it "Singular Purpose". HiFi2000 case, FPE panels, JLSounds I2SoverUSB v.III input, DIYinHK LT3045based power supplies, direct R2R output intended for my headphone amp, and buffered output for my powered desk speakers. Sounds bloody marvellous, I feel no need to modify the board in any way. My only regret is that I didn't add a second USB input and leave the USB/serial adapter connected internally for filter testing. I could add a small USB hub internally, but it would defeat the isolation provided by the I2SoverUSB module, which is essential given how damn noisy my computer's USB is. So Søren.... thanks for your design, your patience with the community, and for the ongoing and continual upgrades over the years! 
27th September 2021, 08:03 PM  #10480  
diyAudio Member

Quote:
You could use the isolated serial port with an interface and a hub, keeping it on the noisy side....
__________________
Søren  www.soekris.dk 

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