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DIY Audio Analyzer with AK5397/AK5394A and AK4490
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Old 25th November 2020, 10:37 AM   #3031
Edmond Stuart is offline Edmond Stuart  Netherlands
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Quote:
Originally Posted by phofman View Post
What's wrong with that USB driver? What would a constant latency every run be beneficial for? IMO the issue was just "why", not that it hurts anything. This is a measurement device, not an interactive musical instrument or a DAW interface. A latency of several hundred ms would not hurt any measurements, IMO.
Hi Pavel,

Indeed, it doesn't hurt, but in some cases you need to know the exact latency and/or the phase relationship between DAC input and ADC output.

BTW, there are several tools out there that measure the latency, but the results are not always consistent. The RTL utility form Oblique Audio for example, says 5.620ms, while DiAna measures 5.6328ms (@ buffer size =256 and SR=192kHz).

Cheers,
E.
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Old 25th November 2020, 11:18 AM   #3032
phofman is offline phofman  Czech Republic
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Edmond, I understand the need for knowing exact latency, but I would assume it must be measured for every run (start of the continuous stream) anyway. Every start will have a different latency, with any type of audio device. Therefore I do not see any problem with having a latency of 5.1ms in one run and 5.9ms in another run. But maybe I am missing something.

Actually, how do you measure the latency? IMO if a timer is used in the actual user-space application, then process scheduling changes the exact moment of running the code for reading the final timestamp. Differences in tens or even hundreds of microsecs would not be surprising.

In linux alsa the driver can be asked to accompany the data with DMA or link timestamp ALSA PCM Timestamping — The Linux Kernel documentation . The link timestamp is optional (if the driver can get it for its hardware), the DMA timestamp is obtained by the driver at IRQ callback. IMO this is as precise as it can get, yet still varies as DMA and servicing IRQs has delays too.
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Old 25th November 2020, 12:21 PM   #3033
Edmond Stuart is offline Edmond Stuart  Netherlands
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Hi Pavel

Quote:
Originally Posted by phofman View Post
Edmond, I understand the need for knowing exact latency, but I would assume it must be measured for every run (start of the continuous stream) anyway. Every start will have a different latency, with any type of audio device. Therefore I do not see any problem with having a latency of 5.1ms in one run and 5.9ms in another run. But maybe I am missing something.
No, you don't miss anything (I didn't say I've problems with variable latencies, only remarks).

Quote:
Actually, how do you measure the latency? IMO if a timer is used in the actual user-space application, then process scheduling changes the exact moment of running the code for reading the final timestamp. Differences in tens or even hundreds of microsecs would not be surprising.
[...]
I measure the latency by comparing the phase of the DAC input and ADC output at two distinct frequency, normally around 997Hz. In my applications it's the phase difference that counts, because in this way the phase lead or lag from filters (or whatever) in the whole DAC-ADC chain is also taken into account.

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E.
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Old 25th November 2020, 08:04 PM   #3034
chris719 is offline chris719  United States
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Quote:
Originally Posted by phofman View Post
IMO UAC2 is fully capable of serving the measurement device needs.
I agree, but I never see things like dropouts or high DPC latency in Windows with PCIe devices, and I see them occasionally across various systems with some USB Audio Class devices.

For measurement, I have a non-audio board I designed using multiple LTC2387-18 and I'm pretty sure it would be hard to find a combination of UAC stack and drivers that work at ~15 MHz. I use FTDI FT601 for now.
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Old 25th November 2020, 09:37 PM   #3035
BlacK_Chicken is offline BlacK_Chicken  Germany
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Dear Edmond, wouldn't it be the easiest to use a ADC reference channel (usually the second input of the DAQ/soundcard) in oder to obtain phase/amplitude response of a DUT independently from hardware latencies in the aquisition path?
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Old 26th November 2020, 10:04 AM   #3036
Edmond Stuart is offline Edmond Stuart  Netherlands
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Quote:
Originally Posted by BlacK_Chicken View Post
Dear Edmond, wouldn't it be the easiest to use a ADC reference channel (usually the second input of the DAQ/soundcard) in order to obtain phase/amplitude response of a DUT independently from hardware latencies in the acquisition path?
Indeed a good question. And yes, DiAna can operate this way. But if the fundamental is heavily attenuated and/or phase shifted when using notch filters on both channels, the only thing that is left for synchronization is the sine at the DAC input. Now the latency comes into play and a phase correction has to be applied.

There is one more reason to not use an ADC channel for synchronization: crosstalk. Believe it or not, even a tiny pure sine wave at the one channel introduces harmonics at the other channel. This appears to be a nasty habit of single chip stereo ADC's and a hindrance for ultra low THD measurements. Read here for more info on this topic.

Cheers,
E.
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Last edited by Edmond Stuart; 26th November 2020 at 10:06 AM.
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Old 26th November 2020, 05:53 PM   #3037
BlacK_Chicken is offline BlacK_Chicken  Germany
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Yes, got your point. In case you need a reference of the stimulus for THD measurements the second aquisition channel is useless. That is a very unique analysis implemented in DiAna.

My statement was targeted to generic transfer function measurements (either derived by stepped sine oder impulse response), where ultra low THD is not a prime requirement.
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Old 18th December 2020, 08:57 PM   #3038
zfe is offline zfe
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For measuring electrostatic headphone amps I would like to increase the input attenuation.
Would it be good practice to add a, say, 900k resistor in series to the single ended input? This would make a 10:1 divider with the 100k input impedance of the RTX6001. A hight input impedance is desired for electrostatic amps anyway.
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Old 19th December 2020, 12:24 PM   #3039
JensH is offline JensH  Denmark
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That will probably work fine. To fine tune it and avoid drop off at high frequencies, you could mount a small (high voltage NP0 or similar quality) capacitor in parallel with the 900k resistor, e.g. 3.9 pF or ideally around 4.1 pF.
If you mount the components inside the XLR connector you can avoid attenuation from the cable capacitance. And you get some shielding as well.
Without the capacitors you will get a drop of around 3 dB at 50 kHz.
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Old 19th December 2020, 08:19 PM   #3040
nyt is offline nyt
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Originally Posted by JensH View Post
It's been very little for several months. In fact I haven't really worked on any spare time (audio) projects for some time, mainly because I moved to a new house recently.

I will take a new look at it and try to figure out a way to make it available, perhaps as a bare board solution.

any joy?
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