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Old 8th June 2009, 01:04 PM   #71
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Default "Keep In/out stream alive" ?

Thanks again

Quote:
Originally posted by KSTR
BTW, do you actually calculate the "measurement signal" response with the same process or do you just draw flat lines on the screen, assuming it is flat per se? I think this might be the core of the problem, I mean if you did an internal "loopback calibration" of the calculation process then the symptom would be accounted for, and the DAC-->ADC loopback would then only show the effective loopback error of the hardware and not the systematic behaviour. Still better would be to address the root cause, not the symptom, of course.
No, but I have of course made a numerical test to verify. I also have an external soundcard where I can make a digital loopback to very every layer in the software.

"Keep In/out stream alive" ?
Could you try the "Setting for device & signal" > "Keep In/out stream alive" ?
Check the box - then wait apprx. 10 seconds and then measure again. (While the stream is still alive)

Please zoom out, so I can see your timewindow.

Regarding save as a HLM-file
HOLMImpulse > File > Save Measurements
This will save all 50 measurements (If not empty) in a custom format (file.hlm)
Which can again be opened in HOLMImpulse File > Open
(Or simply by Drag'n'Drop onto HOLMImpulse)

My laptop soundcard
Attached shows the response of my laptop soundcard using the same settings as you.
Attached Images
File Type: png new picture (1).png (45.1 KB, 715 views)
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Old 8th June 2009, 01:37 PM   #72
soongsc is offline soongsc  Taiwan
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Everything looks fine here. Audigy 2ZS Notebook card.
I loop one channel output back to both input channels.
Once I accidentally selected "mic input" in the Windows volume control. This did result in the funny low frequency response.
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File Type: gif holm impulse example.gif (28.7 KB, 685 views)
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Old 8th June 2009, 01:39 PM   #73
KSTR is offline KSTR  Germany
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>> (Samplerate converters + Nyquist limit in the DAC-ADC chain is a much worse problem)

A strong point. Practically all DAC/ADC chips do alias in the top freq range, due to "incorrect" filters (since correct ones consume to much real estate on the chips). This also could be the problem. Bruno Putzeys on these matter:
http://recforums.prosoundweb.com/ind...and#msg_278132
http://recforums.prosoundweb.com/ind...and#msg_265660

(way above my head, BTW)
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Old 8th June 2009, 02:10 PM   #74
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Default Re: Log-file

Quote:
Originally posted by askbojesen


When the window has closed then
attach the following file:

C:\Documents and Settings\All Users\Application Data\HOLM Acoustics\HOLMImpulse\HOLMLogging.txt

(Or similar path for C:\Documents and Settings\All Users)
Just ran it for measurement, unfortunately, it terminated abnormally.
I heard the log sweep sound around 1 second, then the software quit.
See the attach file for log.

My system configuration:

OS: Windows XP SP2 traditional Chinese
Sound card: 1) south bridge integrated SiS7012, 2)USB 1.1 connected Creative E-MU 0404|USB

Tried both sound cards and 44.1/48KHz, got identical result.
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File Type: txt holmlogging.txt (829 Bytes, 25 views)
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Old 8th June 2009, 02:33 PM   #75
KSTR is offline KSTR  Germany
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Hi Ask,

"Keep Alive" didn't fix it, it seems.

Click the image to open in full size.

I also tried MME drivers, and different DAC/ADC-Channels (but *not* accross chips -- there a four of them on the card).

The zipped .HLM (we can't attach files to eMails sent via DIYaudio)
http://www.freefileserver.com/304137

It's also not a big deal if we can't fix this, since with higher start freq, calibration and the response of real DUTs (drivers/speaker) the problem isn't that relevant, more or less an academic issue.

BTW, when I start at 10Hz the plot stays the same (keeping the 1Hz left border), only THD goes up at <20Hz. Looks like the input AC-coupling hasn't settled fully -- the usual problem, but it can be minimized with a raised cosine fade-in envelope a few periods long (did you use this, or is a linear slope?)


- Klaus
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Old 8th June 2009, 02:48 PM   #76
KSTR is offline KSTR  Germany
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This is what happens when I start the sweep at 1kHz:

Click the image to open in full size.

Looks like we are getting down to where the dog is buried, as we Germans say
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Old 8th June 2009, 03:14 PM   #77
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Thumbs up Time window

Quote:
Originally posted by KSTR
Looks like we are getting down to where the dog is buried, as we Germans say
No that's not were the dog was buried, because there has never been a dead dog - and who wants to bury a dog alive

Actually it is a feature rather than a bug.
It is the timewindow, which was not large enough.

It seems that your soundcard (DAC-ADC) has very long lowfrequent post ringings. (See attachment)

NB: Notice the little triangle in at the frequency axsis corresponding to the wavelength of the timewindow
(It moves when you move the timewindow)

So what now?
When I measure the DAC-ADC I will enable a larger timewindow to adapt your soundcards with these low-freq postringings.

Until the new Release then modify the time window from the measurement > options > window

THD at very low frequencies
Yes, I should disable showing THD at frequencies lower than the log-sweep frequency start

The bottom line: The calculated Impulse Response is correct

again, Thanks for the feedback
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Old 8th June 2009, 03:38 PM   #78
JohnPM is offline JohnPM  United Kingdom
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It is a good idea to provide a log amplitude view of the impulse, it then becomes much easier to see when the window is truncating the response too early.
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Old 8th June 2009, 04:09 PM   #79
KSTR is offline KSTR  Germany
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OK Ask, correct windowing fixed it (bounds: -44100 and 44100 or greater), thanks! Now I get results consistent with all other softwares I have, which are quite a few (plus the sanity check of measuring the ADC-->DAC response -- not DAC-->ADC -- from the "outside" with a hardware test rig)

--> Please autoscale the window length to the period of the start frequency by default ("Auto-Detect" doesn't seem to do anything in my system).

But I'm unclear on how to interpret a measurement when I start at 1Khz (as shown), I mean how do you get the data that you display below that point when there is no data (assuming the correct spectrum of the sweep: no content below 1kHz)? Regarding not only the "false" distortion data but also the basic response?

EDIT:@all: Ignore data that is below the higher value of either the start freq or the freq indicated by time window marker.

OK, no-one would use display boundaries that are beyond the assumed bandwidth of measurements, but, again, autoscaling or limiting of that would be nice, to prevent users from falling into these traps.


And my card doesn't ring at LF, believe me. It has other issues, though

- Klaus
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Old 8th June 2009, 04:49 PM   #80
KSTR is offline KSTR  Germany
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>> It is a good idea to provide a log amplitude view of the impulse,
>> it then becomes much easier to see when the window is
>> truncating the response too early

It's already there, the "dB" button

@Ask: what does the "Reverberation" button do, exactly?
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