Debate Thread: Scott's Hi-res Audio Test - Page 86 - AVS Forum
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post #2551 of 2920 Old 07-28-2014, 03:22 PM
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Originally Posted by NorthSky View Post
Everything is possible, in audio.
Including hearing hypothetical differences where there are none

Hires Music formats ..............."Why does it sound like a CD ?" ............. can we make it louder "?
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post #2552 of 2920 Old 07-28-2014, 03:30 PM
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Originally Posted by m. zillch View Post
Not all outboard DACs provide exactly the same, say 2.0 V analog RCA output, so what you're talking about necessitates adding some stage of level matching for the two competing DACs, but yes. Many source devices offer two simultaneous digital outputs.
+ 1 Good point ! Yes precise level matching is very important in any comparative evaluation or differential testing of this type .

Play them in Audacity or another sound editor not WMP that will allow precise level matching ,you can usually drag and drop the unzipped downloaded WAV or any supported files straight in and read them directly from the hdd rather than importing them as a project ensuring no alteration of the original files
ofc matching level of 2 DAC in parallel playback chain might be problematic maybe two simultaneous digital outputs
is a workaround if using 2 DAC's as to why one would do that is a mystery though other than avoiding down sampling
which in this case may not be necessary although in all fairness it does possibly raise some good questions .

OTOH I'm not so sure you could use the foobar 2000 ABX comparative facility in this way unless Audacity supports the plugin also. Maybe some one could enlighten us ? ( I know I'm being lazy ☺☺)

Unless those in the know feel that the playback gain settings in Foobar 2000 are sufficient ?

In any event WMC or most players should probably not be used to evaluate and or test for differentials without level matching in the playback chain

Hires Music formats ..............."Why does it sound like a CD ?" ............. can we make it louder "?
"The wireless music box has no commercial value. Who would pay for a message sent to nobody in particular?"
- David Sarnoff's associates at RCA the 1920's -

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post #2553 of 2920 Old 07-28-2014, 03:30 PM
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Originally Posted by mcnarus View Post
Is it possible for something to be recorded and the signal split such that one part goes to a 44.1 DAC while the other goes to a 192 DAC?

Don't you mean ADC, not DAC?
Yes' my bad.

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post #2554 of 2920 Old 07-28-2014, 04:03 PM
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Originally Posted by tubetwister View Post
Forgot one more question on PC >Xonar STX amplified sound card> into studio phones that can resolve 27 kHz (much more than I can ofc ) . I have the card set to 96 kHz it *can do 192 kHz as that is it's native sample rate my question is which would be the better setting for these hires files which IIRC are 96 kHZ .
You must set the Xonar to 96kHz, the same sample rate as the file, as otherwise there would be real-time sample-rate conversion to 192kHz , which adds a confusing variable. (Most real-time SRC introduces audible degradation that might well obscure any differences.) also, when I reviewed this card, the early ASUS drivers were flawed and would downsample all hi-rez data. Make sure you have the latest driver.

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post #2555 of 2920 Old 07-28-2014, 05:28 PM
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Originally Posted by mcnarus View Post
Is it possible for something to be recorded and the signal split such that one part goes to a 44.1 DAC while the other goes to a 192 DAC?

Don't you mean ADC, not DAC?
Benchmark ADC:

The ADC1 USB has a total of 5 digital outputs (1-XLR, 2-coax, 1-optical, and 1-USB). These can operate simultaneously at up to three independent sample rates. One of the coaxial outputs can be configured for 16-bit TPDF dithered output. This unique flexibility allows simultaneous recording to a CDR, while recording high-resolution to a digital recorder and a DAW. For example the CDR may operate at 44.1/16 while the DAW operates at 88.2/24 while the digital recorder operates at 192/24. Backup and/or reference recordings can be created with ease while high-resolution outputs are fed to primary recording devices. The optical output supports AES or ADAT formats at resolutions up to 192/24. In ADAT mode, high sample rates are supported using SMUX2 and SMUX4.
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post #2556 of 2920 Old 07-29-2014, 01:11 AM
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Originally Posted by stereoeditor View Post
You must set the Xonar to 96kHz, the same sample rate as the file, as otherwise there would be real-time sample-rate conversion to 192kHz , which adds a confusing variable. (Most real-time SRC introduces audible degradation that might well obscure any differences.) Also, when I reviewed this card, the early ASUS drivers were flawed and would down sample all hi-rez data. Make sure you have the latest driver.

John Atkinson
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Yes and thanks ,the early drivers were flawed ...... I might add more than one issue I have the latest ones and have been keeping it set at 96kHz for 96kHz files I use same as file sample rates routinely with few exceptions .That's what I was hoping the answer would be because it aligns with conventional wisdom as I know it anyway .

Just read your review and update of the Zonar STX It's about what I would have expected for this card it's decent on phones within its limitations ofc . I only use it for that and a modest 2.1 set up it's not part of my HT or studio gear. I picked it up on Pay bay NIB for like $79.00 for what they want for it new I would look to spend a Little more on some DAC /HP amp parts maybe from Schiit Audio or someone like that depending. I try not to a pay retail for consumer electronics though IMO they are consumable generally anyway and often not reparable or always worth repairing if you know what I mean .
.


regards.

Hires Music formats ..............."Why does it sound like a CD ?" ............. can we make it louder "?
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post #2557 of 2920 Old 07-29-2014, 04:38 AM
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Originally Posted by Chu Gai View Post
Is it possible for something to be recorded and the signal split such that one part goes to a 44.1 DAC while the other goes to a 192 DAC?
Yes.

You've just described Meyer-Moran when the source was a 192/24 recording.
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post #2558 of 2920 Old 07-29-2014, 04:40 AM
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Originally Posted by stereoeditor View Post
. (Most real-time SRC introduces audible degradation that might well obscure any differences.
Appears to be yet another self-serving unfounded assertion.

I'm very sure that every SRC introduces audible degradation in sighted evaluations of the kind that "Stereophile Truth" is based on.
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post #2559 of 2920 Old 07-29-2014, 06:31 AM - Thread Starter
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Originally Posted by arnyk View Post
Appears to be yet another self-serving unfounded assertion.
Good morning Arny. Seems to this feeble mind that us dismissing the effect of upsampling as having no audible effect is more self-serving to us than anyone else. After all, your test is all about resmampling and whether that is audible or not. Should we put in record that upsampling causes no audible difference, then that helps our case.

Seeing how so far the data says downsampling changes the sound in your test file, then I say there is every motivation to assume for now, that upsampling could also have an audible difference.

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I'm very sure that every SRC introduces audible degradation in sighted evaluations of the kind that "Stereophile Truth" is based on.
Arny, for 14 years you thought your key jingling file showed no one could hear such differences in double blind tests. Now turns out we can. And that you don't have the ears yourself to be able to tell all those 14 years, given the damage done in Army and such. So I think it is fair from here on that we don't trust your statements of fidelity in this regard.

If you have independent listening tests that used trained listeners, let's see that. That would add immensely to our conversation here.

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post #2560 of 2920 Old 07-29-2014, 06:45 AM - Thread Starter
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Originally Posted by arnyk View Post
Speaking of helping people rule IM out, I decided to test my own monitoring system on this PC which I know to be flawed:



This should effectively counter JA's claim that all monitoring systems are free from audible IM and therefore the issue is according to him a red herring.
Hi Arny. I am confident JA has not at all said that "all monitoring systems are free from audible IM." He has said that he has measured his equipment and performed the listening tests to show that your IM theory is false. I have done the same. So with respect the two of us, your theory is a red herring. It would have been prudent to have us run your test before predicting that it is a problem.

As to your crappy gear, a precondition of such testing I thought was to not use crappy systems. My gear is not crappy and neither is JA's. Sad to see you use such crappy systems to make such evaluations. I suspect it has considerably clouded your judgement of such fidelity differences.

This is why you want to invest in quality equipment Arny. You don't get fooled by their poor performance one day. Buy cheap, buy again as the saying goes.

Quote:
Given that Amir claims access to good test equipment it wouldn't be hard for him to provide the same information on the systems he used.
I have no test equipment to measure IEMs. I can measure the electronics of the system but having run your listening tests, I thought we established that there is no problem. Hopefully you are not saying that we wasted our time running those listening tests.

Quote:
BTW, here is an ABX log for a run with the above monitoring system, levels and passage selection fudged err optimized to create maximum "Correct" results:
If you don't mind, it would be good to not introduce such cheats in our testing Arny. I took your files, put them in foobar and ran an ABX test against them. That is how I passed them. Not by taking broken systems, fudging this and that, to get a reading. If you have to do all of that then your listening ability is poor and let's not rely on it one way or the other.

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Sorry about the one I missed, I think I mistakenly hit the "Play A" button instead of the "Play X" button but recorded the result anyway.
Oh? Such mistakes can happen? So the results aren't always representative of what you heard? Must come as a news to many .

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post #2561 of 2920 Old 07-29-2014, 06:57 AM
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Originally Posted by amirm View Post
Hi Arny. I am confident JA has not at all said that "all monitoring systems are free from audible IM."
Right. I was not directly quoting him. The above quote is completely a fabrication.

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Originally Posted by amirm View Post
He has said that he has measured his equipment and performed the listening tests to show that your IM theory is false.
He said that, but the evidence he provided seems adverse to that conclusion. He provided measurements of a number of small but relatively high-priced pieces of equipment that had substandard performance. He also provided measurements of some expensive equipment that seemed to perform well.

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I have done the same.
Really? Please provide a link to the technical tests of your monitoring system that you performed. I must have missed it. If you wish to assert that your entire systems are technically blameless, your testing needs to include the acoustical output of your earphones and loudspeakers. If you wish to indict your systems, testing of the substandard components all by themselves seems to be sufficient.

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I have no test equipment to measure IEMs.
A measurement-grade microphone or even a SPL meter with a line level output and a homemade jig would probably suffice. We're not measuring frequency response, we are measuring nonlinear distortion.

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I can measure the electronics of the system but having run your listening tests, I thought we established that there is no problem.
Oh, so then you admit that you didn't do the testing that John and I did, nothing like it. It's hard to follow you when you are so careless about contradicting yourself.

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Originally Posted by amirm View Post
Hopefully you are not saying that we wasted our time running those listening tests.
What I'm saying is that no single test is the be-all and end-all. The IM listening test is provisional, and itself under test. I've already pointed out that I think it may be flawed on the side of being too accepting of equipment that actually fails in use. The IM listening test is for general audiophiles who lack the resources that John, you and I have.

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post #2562 of 2920 Old 07-29-2014, 07:23 AM - Thread Starter
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Originally Posted by arnyk View Post
Really? Please provide a link to the technical tests of your monitoring system that you performed.
??? I ran your listening test Arny. It showed no problem. So as JA said, it was a red herring.

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I must have missed it. If you wish to assert that your entire systems are technically blameless, your testing needs to include the acoustical output of your earphones and loudspeakers. If you wish to indict your systems, testing of the substandard components all by themselves seems to be sufficient.
My assertion is your assertion Arny. You said to run a test, we ran a test. You told us what we should hear. That is what I heard. Please declare your listening test invalid and explain why and we can see about running other tests.

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A measurement-grade microphone or even a SPL meter with a line level output and a homemade jig would probably suffice. We're not measuring frequency response, we are measuring nonlinear distortion.
Measurement mics can have significant distortions of their own Arny. Here is my USB measurement mic with no signal playing:



You see all those distortion spikes it has generated on its own? Fortunately for room measurements it doesn't matter. In this case it very much does.

Quote:
Oh, so then you admit that you didn't do the testing that John and I did, nothing like it. It's hard to follow you when you are so careless about contradicting yourself.
No it is not hard to follow. I explained that we ran your IM listening tests and we passed. Now you created another fishing expedition for us. After that, there will be another and another.

Quote:
What I'm saying is that no single test is the be-all and end-all.
It has not been one test Arny. It has been a test after test all the while denying all the data prior to that. If we had failed to differentiate the files, that would have been it. But the fact that we have not, and consistently so should change your outlook Arny. Either we are hearing differences or you are terrible at creating tests to show otherwise. Either way, the road needs to end some place.

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The IM listening test is provisional, and itself under test.
You should have positioned it that way in which case I would not have run it until you provided assurance that it was a definitive test.

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I've already pointed out that I think it may be flawed on the side of being too accepting of equipment that actually fails in use. The IM listening test is for general audiophiles who lack the resources that John, you and I have.
You have a theory. So far you have been wrong on all of your theories.

And no, none of us have resources to ascertain distortion in headphones. You might think a crappy fixture does that but it doesn't. If we run it and get negative results, you will come back and say, "oh, the test was provisional. I screwed up. Now run this other test."

No one would have gotten so many chances to prove their point Arny. If an audiophile failed a single such test, we would have ridiculed them and hung them in the public square for good measure. You have put test after test in front of us and no matter how many times we pass them and invalidate your point of view, you come back with "now run this."

As I said earlier, since you can't hear what we can, then you are not in a situation to create appropriate tests to diagnose the audibility differences. It is illogical to continue to be in the driver's seat when you are blind Arny, pun intended .

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post #2563 of 2920 Old 07-29-2014, 07:35 AM - Thread Starter
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Originally Posted by m. zillch View Post
Based on the new test I think Arny already knows what I did, but if he would like a PM from me, I'll provide it.
I trust he did and you did tell him?

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post #2564 of 2920 Old 07-29-2014, 07:35 AM
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Originally Posted by arnyk View Post
Quote:
Originally Posted by stereoeditor
Most real-time SRC introduces audible degradation that might well obscure any differences.
Appears to be yet another self-serving unfounded assertion.
Not at all. Mr. Krueger. My assertion involves real-time SRC by computers when the file being played has a different sample rate from the playback hardware setting. In such cases, I have felt there there was something degrading the sound, which I confirmed by measurement.

Given the poor measured performance of your playback hardware, as typified by the spectral analysis you posted recently, I doubt you would hear any such degradation.

You can find SRC measurements at http://src.infinitewave.ca/

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post #2565 of 2920 Old 07-29-2014, 07:48 AM
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Originally Posted by amirm View Post
??? I ran your listening test Arny. It showed no problem. So as JA said, it was a red herring.


My assertion is your assertion Arny. You said to run a test, we ran a test. You told us what we should hear. That is what I heard. Please declare your listening test invalid and explain why and we can see about running other tests.


Measurement mics can have significant distortions of their own Arny. Here is my USB measurement mic with no signal playing:



You see all those distortion spikes it has generated on its own?
One of these days you might learn how to properly interpret FFTs.

The above FFT most definitely does not show distortion. Two very solid reasons:

(1) There's no evidence of an applied signal. To have distortion you have to have something that gets distorted. Apparently you thought those big spikes you have been criticizing in my tests just came out of nowhere and served no purpose? They are the test signals and their purpose in life is to become distorted when there is nonlinear distortion in the UUT. Just a friendly reminder.

(2) The spikes themselves show a regular pattern: 120 Hz, 180 Hz, 240 Hz, 300 Hz... What might one find in just about every home in the USA and Canada that causes that?

Post again when this is clearer, or please ask more questions.
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post #2566 of 2920 Old 07-29-2014, 07:52 AM
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It seems that finding an audible difference is interesting, but doesn't there need to be some reasonable explanation?

We cannot hear ultrasonics, therefore if ultrasonics, are responsible for audible differences, then that suggests that artifacts could be responsible or ignored.

IMO, 24-bit depth and perhaps the sample-rate may improve accuracy.

Why can't the preamplifier signal be compared to the source to test for accuracy, which I would argue is the goal?

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post #2567 of 2920 Old 07-29-2014, 07:53 AM - Thread Starter
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Originally Posted by arnyk View Post
One of these days you might learn how to properly interpret FFTs.
Yeh, it is on my TODO list .

Quote:
The above FFT most definitely does not show distortion. Two very solid reasons:

(1) There's no evidence of an applied signal. To have distortion you have to have something that gets distorted. Apparently you thought those big spikes you have been criticizing in my tests just came out of nowhere and served no purpose?
Of course there is "applied signal." There is noise in the room represented by the random bits in the waveform. That is the signal. Those correlated spikes are not at all part of the room noise. Therefore they are distortions.

Quote:
(2) The spikes themselves show a regular pattern:
Which is what I explained above. Should I play your clips and see the same spikes, I would not know if they are caused by the playback system or capture/mic.

The correlated spikes by the way are caused by the USB ADC in the mic. Still think digital audio is perfect? I think not .

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post #2568 of 2920 Old 07-29-2014, 07:57 AM
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Originally Posted by amirm
Arny, for 14 years you thought your key jingling file showed no one could hear such differences in double blind tests. Now turns out we can. And that you don't have the ears yourself to be able to tell all those 14 years, given the damage done in Army and such. So I think it is fair from here on that we don't trust your statements of fidelity in this regard.

If you have independent listening tests that used trained listeners, let's see that. That would add immensely to our conversation here.
There might be some sort of Decline Effect going on here?
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post #2569 of 2920 Old 07-29-2014, 08:02 AM - Thread Starter
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Originally Posted by arnyk View Post
(2) The spikes themselves show a regular pattern: 120 Hz, 180 Hz, 240 Hz, 300 Hz... What might one find in just about every home in the USA and Canada that causes that?
You edited the post while I was responding. You theory is false there Arny. Here is the measurement of my mic again:



I specifically put the cursor on the first distortion spike. It says 1 Khz. The rest all line up perfect on multiples of it. They are caused by the USB buffer timing.

60 Hz hum doesn't all of a sudden peak at 1Khz. To say nothing of 1 Khz not being a multiple of 60.

And since this mic is USB powered, it would be illogical to assume that there is a 60 Hz component there lest you think there is that kind of hum on USB bus. Do you?

Maybe I am not so bad at reading FFTs after all .

Quote:
Post again when this is clearer, or please ask more questions.
How could it be more clear Arny? I measured the response of the mic with nothing playing to see what it measures on its own with respect to noise and distortion. This is a "control" that you need to run always before trusting your measurement system. That is, what it shows in the "best case" scenario.

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post #2570 of 2920 Old 07-29-2014, 08:10 AM - Thread Starter
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Originally Posted by 8mile13 View Post
There might be some sort of Decline Effect going on here?
To some extent yes. This part is definitely true: "Add it up and researchers are seeing what they want to see. The New Yorker take makes sense---humans hate being wrong."

In the case of people who literally for decades had convinced themselves that it is impossible to pass these tests, this is an impossible pill to swallow that not just one person but multiple have accomplished it. So they keep creating new tests, not realizing that with everyone that we pass, the case becomes even more bulletproof.

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post #2571 of 2920 Old 07-29-2014, 08:29 AM
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Originally Posted by stereoeditor View Post
Not at all. Mr. Krueger. My assertion involves real-time SRC by computers when the file being played has a different sample rate from the playback hardware setting. In such cases, I have felt there there was something degrading the sound, which I confirmed by measurement.
The problem john is that you made a global assertion of poor performance which my own measurements and those of many others contradict. There are good SRCs out there, and finding them is not a problem for most who diligently seek. Of coure

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Originally Posted by stereoeditor View Post
Given the poor measured performance of your playback hardware, as typified by the spectral analysis you posted recently, I doubt you would hear any such degradation.
Maybe you ought to read my posts more carefully, John. I have repeatedly indicated that the playback system I showed measurements and listening tests for as evidencing poor performance was just one of several that I have.

I should also add that my measurements were taken with the actual headphones that I use attached. I wonder if you are that careful about what you are doing.


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Originally Posted by stereoeditor View Post
You can find SRC measurements at http://src.infinitewave.ca/
I have referenced that source here and many other forums many times as indicating that SRCs with excellent performance abound but that SRCs with poor performance, including some ones that are expensive and produced by "name" organizations, also exist. Most people who review it seem to agree. Sorry that your cup is far less than half empty, John.


Maybe we've got a case of reading what is desired to be read... ;-)
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post #2572 of 2920 Old 07-29-2014, 08:32 AM - Thread Starter
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Originally Posted by arnyk View Post
There are good SRCs out there, and finding them is not a problem for most who diligently seek. Of coure
Did you use a good SRC in the jingling file we have been testing or crappy ones Arny?

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post #2573 of 2920 Old 07-29-2014, 09:07 AM
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I'm more than a bit confused here, Amir, and perhaps others are too, so I'm hoping you can help me out.

1) I readily acknowlege that you and some others have reliably been able to tell the difference between 44.1 and the hi-res version. As I understand it the way this has been done is to derive one from the other. In your estimation, your opinion, could the reason for this have to do with quirks, misbehaviors, converters just not being good enough, etc.? JA has stated that Arny's playback hardware's measurements are poor and one might therefore draw the implication that whatever he's using for conversion is is similarly lacking.

2) Although you've been asked, you've not stated to my knowledge what it was that you picked up on in these evaluations that you've aced. Will you now do so and provide time stamps? It may advance everyone's understanding and the topic in a non combative way.

3) Earlier I posed the question whether one could simultaneously record an event in both 44.1 and hi-res. Given my first question, in your estimation would such an endeavor better determine whether there were reliable audible differences?

4) Since your findings as well as those of others are at odds with M&M, have or will you be contacting them to discuss this?

5) Since you've read the M&M paper as I'm sure the discussions surrounding it, what do see as the core differences in the way you ran the tests vs how they ran it?

Thanks in advance.
4)

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post #2574 of 2920 Old 07-29-2014, 10:43 AM
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Quote:
Originally Posted by Chu Gai View Post
I'm more than a bit confused here, Amir, and perhaps others are too, so I'm hoping you can help me out.

1) I readily acknowlege that you and some others have reliably been able to tell the difference between 44.1 and the hi-res version
[emphasis mine]


Huh? "Some others", plural? If we are talking about Arny's files, only one other person besides him claims to hear a difference between 44.1 and the hi-res version, that being forum member imagic. So "others", plural, is incorrect. Sure, I aced the test too, however instead of claiming it's due to my "superior hearing", "high end and expensive gear", and/or a "certain distinct, pristine shine or clarity to the upper, lower treble that sounds like a veil was lifted", I'm instead stating up front it has nothing to do with the hi-res sound being superior or more detailed, it has to do with a subtle audible difference between the two files, undoubtedly due to some artifacts generated by the different way they were produced:
Quote:
Originally Posted by m. zillch View Post
Here are my log files using Arny's old files [the only version of his files yet posted in this thread, up to now] ...

...19:57:20 : 15/15 0.0%


20:02:51 : 16/16 0.0%


20:03:33 : Test finished.


----------


Total: 16/16 (0.0%)

Today, using his new files, I unfortunately hear a faint IM problem so I can't do that test, however I did want to point out that the data I provide above was accomplished by my keying on a secret, audible "tell", I don't think I should disclose, calling into question anybody else's published data prior to mine, using the same files, even if they truly had no IM problems in their system, just like I didn't.

No dogs, no bats, no children with >22kHz hearing, no analyzer, and no text editor used, nor was I comparing the click noises themselves; it was just me and my headphones listening intently for over an hour in suboptimal conditions.
Tell me Chu, do you hear a superior quality to the hi-res version? If you don't, as I strongly suspect since you are human, then why do you believe others who claim to if I've proven that I can pass the same test by keying off a very subtle sound difference which is actually just an artifact?
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post #2575 of 2920 Old 07-29-2014, 10:57 AM
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Quote:
Originally Posted by m. zillch View Post
[emphasis mine]


Huh? "Some others", plural? If we are talking about Arny's files, only one other person besides him claims to hear a difference between 44.1 and the hi-res version, that being forum member imagic. So "others", plural, is incorrect. Sure, I aced the test too, however instead of claiming it's due to my "superior hearing", "high end and expensive gear", and a "a certain distinct clarity to the upper, lower treble that sounds like a veil was lifted", I'm instead stating up front it has nothing to do with the hi-res sound being superior or more detailed, it has to do with a subtle audible difference between the two files, undoubtedly due to some artifacts generated by the different way they were produced:


Tell me Chu, do you hear a superior quality to the hi-res version? If you don't, as I strongly suspect since you are human, then why do you believe others who claim to if I've proven that I can pass the same test by keying off a very subtle sound difference which is actually just an artifact?
I want to make sure nobody mistakenly thinks that I'm claiming the hi-res version sounds better. All I did was find a way to hone in on a tiny difference in the transient details that acted as a "tell." I don't know if IM distortion is the reason for it, or if it has to do with resampling 24/96 to 16/44.1—a process that I've never been convinced is truly transparent. I'd feel more comfortable comparing 24/88 to 16/44, or 24/96 to 16/48. Perhaps it is something else altogether.

I continue to think it would be beneficial for the thread if you were to discuss the specifics of what you did in order to hear the difference between the files.
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post #2576 of 2920 Old 07-29-2014, 11:12 AM
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^Thank you for your clarification and honesty. Sorry if I might have misrepresented anything about your stated findings/beliefs, either directly, indirectly, or through innuendo. My apologies.

In A/V reproduction accuracy, there is no concept of "accounting for taste". We don't "pick" the level of bass any more than we get to pick the ending of a play. High fidelity is an unbiased, neutral, exact copy (or "reproduction") of the original source's tonal balance, timing, dynamics, etc..


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post #2577 of 2920 Old 07-29-2014, 11:17 AM
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Quote:
Originally Posted by amirm View Post
... I specifically put the cursor on the first distortion spike. It says 1 Khz. The rest all line up perfect on multiples of it. They are caused by the USB buffer timing...
I'll assume the windows recording level was "high", maybe even set to 100. My graph looks very similar to yours.

Averaging will reduce the hash, the spikes are still there.

Now...

Set the recording level to 1.

With no averaging, there is still hash, no spikes visible. Also no tone visible in the hash.

With forever averaging, still no spikes visible, but my test tone emerges - 10.5khz in my picture.

The tone is -60db as created in audacity: create level = 0.001, is played through PC speakers, about 6 feet from the mic facing the other way.

I'm not claiming anything about the sensitivity or what the levels are in the air, just an experiment with the self-noise on USB, and being able to (apparently) get rid of much of it.

File this (setting the record level to 1) away for future reference, maybe.

If the 1khz spike series is buffer related, where did they go?



edit: Well, I guess I do still see them a little, but it's not the mess that it is when the record level is set higher...
Microphone is a newly acquired miniDSP UMIK-1

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post #2578 of 2920 Old 07-29-2014, 11:37 AM
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Quote:
Originally Posted by amirm View Post
Of course there is "applied signal." There is noise in the room represented by the random bits in the waveform.
Not much of a help since the room noise is an incoherent signal. Test signals for distortion measurements generally need to be coherent to facilitate the measurement. Something as simple as a FFT requires the stimulus for the distortion measurement be a coherent signal. Not even close, no cigar! ;-)

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That is the signal.
Repeat, not so much. You said it is random noise, so you disqualified it.

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Originally Posted by amirm View Post
Those correlated spikes are not at all part of the room noise. Therefore they are distortions.
Not necessarily. I guess I need to introduce you to another class of thingies that show up in FFTs. I call them undesired signals. They are coherent as you say. But, since we can line them up with a stimulus, they come from the outside world, not the internals of the UUT.

Let's get back to your experimental data:



To help with the recognition process, I annotated (in hot pink) a set of spikes in your data in order to help clarify what I am talking about. They are annotated 120 Hz, 240 Hz, 480 Hz, and 960 Hz. They are all even harmonics of the USA/Canada power line so I know that this measurement was probably not performed in Europe.

They are all related to the power line signal (60 Hz) distorted in a specific way. But since the original 60Hz signal is not at all obvious in the FFT, it was probably coming in from somewere outside the mic. It may have enterded the mic acoustically, in which their presence does not indict the mic, their presence may indict the room that the mic was in when the measurement was made.

If the mic was open to some room, then it is possible that the room had some source of 60 Hz noise in it. Could be a fan, a light on a dimmer or something else.

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Which is what I explained above. Should I play your clips and see the same spikes, I would not know if they are caused by the playback system or capture/mic.
Analysis of the source file for my clips excludes your playback system. I save you some work and admit that they show spikes at 120, 180, 240, 300 and 420 Hz. I'll spare you how much work it took to get them as low as they are, given that the keys themselves are not the loudest things in the world.

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The correlated spikes by the way are caused by the USB ADC in the mic.
Not the ones at

120 Hz, 240 Hz, 480 Hz, and 960 hz for sure. The source of the separate series (lime green) at 750 and 1500 Hz is unknown. There is also the blue series that you started marking at 1 Khz, and I marked at 2 KHz, 3 KHz, etc., etc.

Quote:
Originally Posted by amirm View Post
Still think digital audio is perfect? I think not .
Digital audio is not perfect, but the source of the artifacts we have been discussing is no doubt in the analog domain and probably acoustical.

Furthermore, they are all like 80 dB or better down (reference not stated). Distortion due to transducers starts out more like 30 dB down, so this has all been a discussion of moot artifacts.

Beyond that, the mic used in this measurement is not the only mic in Washington State, so if you don't like it, get a better one!

It's handy to have measurement mics that require phantom power. If you turn the phantom power off, it cuts the mic off from acoustic noise, and whatever artifacts you see came from the electronic side of the mic, the mic preamp, the dac, etc.
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post #2579 of 2920 Old 07-29-2014, 11:51 AM - Thread Starter
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Originally Posted by m. zillch View Post
Huh? "Some others", plural? If we are talking about Arny's files, only one other person besides him claims to hear a difference between 44.1 and the hi-res version, that being forum member imagic.
Thanks for teaching Chu healthy skepticism. If you let him, he would believe Elvis is still alive and we didn't really go to the moon. The latter is actually right but it would damage his reputation to say it if he is not me.

Back to your comment, I trust you ignored Arny's results for a reason?

Quote:
Originally Posted by arnyk View Post
BTW here is the ABX log for me running an ABX test on just the keys jangling portion of the file:

--------------------------------
*Note - levels and passage selection fudged for best false positives


foo_abx 1.3.4 report
foobar2000 v1.3.2
2014/07/28 07:53:00

File A: C:\Users\client64\Music\AVS\Keys jangling\keys jangling full band 2496 test tones f3 4416.wav
File B: C:\Users\client64\Music\AVS\Keys jangling\keys jangling full band 2496 test tones f3.wav

07:53:00 : Test started.
07:54:38 : Trial reset.
07:56:40 : 01/01 50.0%
07:56:55 : 02/02 25.0%
07:57:15 : 03/03 12.5%
07:57:21 : 04/04 6.3%
07:57:27 : 05/05 3.1%
07:57:35 : 06/06 1.6%
07:57:42 : 06/07 6.3%
07:57:55 : 07/08 3.5%
07:58:10 : 08/09 2.0%
07:58:27 : 09/10 1.1%
07:58:35 : 10/11 0.6%
07:58:52 : 11/12 0.3%
07:59:09 : 12/13 0.2%
07:59:15 : 13/14 0.1%
07:59:22 : 14/15 0.0%
07:59:52 : 15/16 0.0%
07:59:59 : Test finished.

----------
Total: 15/16 (0.0%)
--------------------------------

Obviously, I'm not going to BS anybody, this was a bogus test. I selected the level and the portion of the track that I actually listened to to maximize the audible difference based on nonlinear distortion in the crappy monitoring system in this PC producing more audible IM with the 2496 test file than with the 4416 file.
By ignoring his results altogether you are hinting at us that you told him how to cheat on this test and he didn't really hear the difference due to "audible distortion based on non-linear distortion?"

Quote:
Tell me Chu, do you hear a superior quality to the hi-res version? If you don't, as I strongly suspect since you are human, then why do you believe others who claim to if I've proven that I can pass the same test by keying off a very subtle sound difference which is actually just an artifact?
I have not said anything whatsoever about "superior quality" of hi-res version. As I keep saying, ABX testing is not about qualitative analysis. It is a forced choice, binary decision and data saying there is or isn't a difference.

Even now, I don't know which file is which. As I explained yesterday, I simply grabbed the files, threw them into the playlist in Foobar and ran an ABX test on them. That is the fair way to run the test.

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post #2580 of 2920 Old 07-29-2014, 11:52 AM
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Quote:
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Did you use a good SRC in the jingling file we have been testing or crappy ones Arny?
Why are you asking me? Won't I be a biased source? ;-)

If you read my posts you know that the SRC I use was the one in Cool Edit Pro 2.1 with pre/post filtering, and you know about the SRC reviews at Infinite Wave. Adobe Audition 2.0 is a close descendant of Cool Edit Pro 2.0 which is testsed at Infinnite Wave. I could tell you to go figure.

However, I can also tell you that I've run the tests that Infinite Wave used on all those SRCs on the Cool Edit Pro 2.1 SRC with very similar results. I also did some tests that are far more difficult than what Infinite Wave used, and found it has a weakness. The weakness is that if you upsample or downsample with really high ratios such as upsampling from 44.1 to several, maybe 10 megahertz, its digital filter's bandpass filtering seems to loose some of its precision. The circumvention is to do the upsampling in steps.
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