Debate Thread: Scott's Hi-res Audio Test - Page 51 - AVS Forum
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post #1501 of 2920 Old 07-03-2014, 10:22 AM
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Originally Posted by stereoeditor View Post
I am in the office right now and don't have access to the files, but I will post the spectrum and level of the background noise in my urban dedicated listening room later.

John Atkinson
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Some places in B'kln aren't so bad. I lived in an apartment in a private house on 60th and 20th Ave. pretty quiet and having a very large cemetary not to far didn't hurt. OTOh, I also lived on the corner of W. 9th and Kings Hwy. traffic, lots of stores, nearby subway, and sanitation trucks at night...not so much. Lots of people have their HT in rooms that aren't dedicated and are multifunctional.

Just sayin...

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post #1502 of 2920 Old 07-03-2014, 10:31 AM
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So long as you focus myopically on Antarctica to the exclusion of everything else, you'll have a tough time understanding why people wear bikinis.

No one has a tough time understanding why people wear bikinis.
I have a tough time understanding why some people wear bikinis.


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post #1503 of 2920 Old 07-03-2014, 10:40 AM
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You wouldn't if you saw my neighbor!

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post #1504 of 2920 Old 07-03-2014, 10:47 AM
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Originally Posted by RobertR View Post
John, I have read the reviews in Stereophile, and no, they do NOT answer my question. Your responses give no indication that you or anyone else at the magazine can.
So it goes.

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post #1505 of 2920 Old 07-03-2014, 11:11 AM
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You wouldn't if you saw my neighbor!
Pictures, Chu, pictures!
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post #1506 of 2920 Old 07-03-2014, 11:16 AM - Thread Starter
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Originally Posted by arnyk View Post
Personal preferences for loud sound are another big area of mystery.
You are confusing dynamic range with loud. Loud says high average level. Dynamic range means allowing both the quietest moments and loudest ones co-exist. A Rock concert is just loud. Listening to keith john johnson's reference recordings on JBL K2 is dynamic. This is one of the aspects of a "reference system" which you asked about.

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Originally Posted by arnyk
For example I've measured my system > 110 dB SPL with several peak reading SPL meters but I rarely even hit 95 dB in actual listening for general pleasure.
Your SPL meter again has no spectrum display. Nor do you know its bandwidth over which it is measuring that "peak." calibration is usually for 1 Khz. You are taking a knife to a gunfight with that SPL meter Arny .

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Originally Posted by arnyk
Very few audiophiles have even invested in the equipment to do this test even though it costs < 35 and takes only a few minutes of effort to use. In fact one audiophile asked whether he could rent it someplace!
Better to not have the tool than to use it for incorrect application!

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Originally Posted by arnyk
If we are advising people to build "Blue Moon" systems whose capability will only be used once every Blue Moon, we probably ought to make that clear.
Just the same, if our standard is as low as it gets with listening to music while the dishwasher is going and someone vacuuming, let's be up front with that too.

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post #1507 of 2920 Old 07-03-2014, 11:38 AM
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Originally Posted by amirm View Post
Here is the critical thing: the algorithm has completely eliminated ultrasonics! Of course that is what it needs to do with respect to limits of our hearing range. If we are to accept this as fact, then we don't need to run Scott's test. We would be saying that nothing above 20 Khz matters as we try to analyze its loudness this way. The perceptual model in Replaygain invalidates the test and vice versa.
I found the level difference of 0.2dB between Scott's testfiles by ear. I can't hear ultrasound, so that value might be different for others who can hear ultrasound, although I doubt it.
What procedure do you propose to match the (perceptual) loudness of two spectrally different audio sources ?
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post #1508 of 2920 Old 07-03-2014, 11:42 AM
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Originally Posted by amirm View Post
Good morning Arny. What I asked for was to not throw voodoo technical arguments at people where we make them worried about going deaf when there should be no concern. And make them think their rooms are noisier than any human ear would perceive.

I am hoping you confirm this in your next reply and we are once and for all are done with these lay arguments that have no foundation in science or reality.
Speaks to knowing very little about science in general - only what is read in just a few papers.

JA is going to publish a spectral analysis of the noise floor of his listening room - you might want to do the same.
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post #1509 of 2920 Old 07-03-2014, 11:53 AM
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Quote:
Originally Posted by Kees de Visser View Post
I found the level difference of 0.2dB between Scott's testfiles by ear. I can't hear ultrasound, so that value might be different for others who can hear ultrasound, although I doubt it.
What procedure do you propose to match the (perceptual) loudness of two spectrally different audio sources ?
There is an easy way to test if you can hear ultrasound. Buy one of those ultrasonic pest repellers. Better yet, buy a four pack and plug them all of them into a power strip. Turn it on and off. Do you hear anything? Do you feel a tingle? It's similar to trying to see the infrared light coming out of your remote control.

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post #1510 of 2920 Old 07-03-2014, 11:56 AM
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Quote:
Originally Posted by amirm View Post

You go on to say that your "SPL meter" told you that. Well, the SPL meter lies in this regard. It doesn't know the spectrum of human hearing system and gives you a single value. That single value is useless with respect to audibility of noise.
False claim. Most modern SPL meters have a feature called "C Weighting" which approximates the spectral response of the human hearing system at low SPL levels.

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Originally Posted by amirm View Post
Even if the SPL meter told you the right number it still is non sequitur in this argument. Here is Fieldler in his paper again:

The above statement is clearly vague because it, to turn your own argument back at you, says nothing about the spectal content of the "audio system noises" and nothing about the spectral content of the room noise. Let's say that the "audio system noises" are composed of a pure tone at 4 KHz (like a high pitched kazoo), and the room noise is well approximated by red noise. Since the spectrum of noise in my living room is different from red noise and I don'y play a kazoo during critical listening sessions, Fielder's vague example is irrelevant to me and most other non-Kazoo-playing audiophiles.


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We are able to hear equipment noise 15 db below room noise.
If the vaguely defined "we" are also Kazoo players, its all irrelevant to most audiophiles including myself.

Please try:

Defining things in a way that is clearly "real world", not vague baseless assertions.


Perhaps you provide some examples from your own listening room since you seem to have an unbeatable pile of expensive measurment equipment at your disposal.
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post #1511 of 2920 Old 07-03-2014, 11:57 AM
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Originally Posted by Kees de Visser View Post
I found the level difference of 0.2dB between Scott's testfiles by ear.
More details, please. A FOOBAR2000 ABX test log would be appropriate.
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post #1512 of 2920 Old 07-03-2014, 12:53 PM
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Re: Gain adjustment to account for the, uh, truncation of high frequencies

Thought Experiment:

Go to a piano with extended range - 136 keys.

Play some low notes with the left hand, while playing some possibly inaudible high notes with the right hand, and measure the SPL.

Continue playing with the left hand, stop playing with the right hand. Measure the SPL - I'll assume it is lower.


Did that change the SPL from the left hand? Do you have to play more forcefully with the left hand now to make up for the loss of the right?

---------

PS: I still don't get why the 24/96 file, after being downgraded to 16/44, is not left in that state, and is upped back to 24/96. Maybe it was explained and I missed it.

I'll be back later...



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post #1513 of 2920 Old 07-03-2014, 01:17 PM
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Quote:
Originally Posted by RayDunzl View Post
Re: Gain adjustment to account for the, uh, truncation of high frequencies
PS: I still don't get why the 24/96 file, after being downgraded to 16/44, is not left in that state, and is upped back to 24/96. Maybe it was explained and I missed it.
It is all about keeping as many things the same as possible other than the simple matter of the downsampling to 44 KHz

Probably the most likely source of extraneous change that is unrelated to the basic resampling is monitoring systems, particularly DACs which may have different frequency response when playing @ 44 KHz versus 96 KHz.

This was a lot bigger problem 15 years ago.
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post #1514 of 2920 Old 07-03-2014, 01:22 PM
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Quote:
Originally Posted by RayDunzl View Post
Re: Gain adjustment to account for the, uh, truncation of high frequencies

Thought Experiment:

Go to a piano with extended range - 136 keys.

Play some low notes with the left hand, while playing some possibly inaudible high notes with the right hand, and measure the SPL.

Continue playing with the left hand, stop playing with the right hand. Measure the SPL - I'll assume it is lower.


Did that change the SPL from the left hand? Do you have to play more forcefully with the left hand now to make up for the loss of the right?

A test with another sample rate converter (SoX) showed no gain difference.
Consensus is that the used sample rate converter applies -0.1dB gain before the resampling.


Quote:
Originally Posted by RayDunzl View Post
---------

PS: I still don't get why the 24/96 file, after being downgraded to 16/44, is not left in that state, and is upped back to 24/96. Maybe it was explained and I missed it.

To rule out differences in the DAC's operation on different sample rates.
and file size would be a giveaway.
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post #1515 of 2920 Old 07-03-2014, 01:32 PM
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Originally Posted by arnyk View Post
More details, please. A FOOBAR2000 ABX test log would be appropriate.
I somehow expected that
OK. Use any DAW, open both stereo files in a 4-track project and do a nulltest (inverse polarity of one pair, mix them and listen to the master output). Now align them to the closest sample, then (optional) use a sub-sample plugin to minimize sub-sample sync difference and adjust level of one pair to find the best null. In this case the best null was with the hi-res version at -0.2dB. I mentioned this to illustrate that no psy model was used to estimate the loudness difference. It was my personal hearing, using the full bandwidth difference signal.
Another method would have been to listen to both Left or both Right channels of the two samples in stereo, swap L/R frequently while listening and adjust the level of one pair to obtain minimum center image displacement.
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post #1516 of 2920 Old 07-03-2014, 01:37 PM
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Originally Posted by urapnes1 View Post
When you wrote what I quoted below, I interpreted it to mean that a bias controlled comparison is required in order to hear an audible difference.
First of all, I never said a DBT was 'required' in order to 'hear' an (actual) audible difference. I wrote about standards of proof. But is one required to verify every possible reported audible difference, including complete absence of output versus normal listening output level? No, of course not, that is absurd. I didn't really think this needed explaining. Clearly a mistake on my part.


You provided a plausible reason why a huge audible difference would be perceived in a normal listening situation (i.e., not a lab test where you are measuring 'just noticeable difference' versus silence) : in one state, there was NO OUTPUT. The poster I was originally addressing said the difference he perceived was 'huge'...but he surely didn't say it was output vs no output. And he too was listening at his normal levels. The cause he cited was that he had upgraded a DAC or an amp. Faced with such a claim , " A caused B", you have a choice to believe
1. A is most likely to cause B
2. Something else is equally or more likely to cause B

Cognitive bias is not likely to cause a person to think one DUT has no output whatsoever, and the other has a normal-level output. You, btw, have not determined what the actual cause of the system failure was..i.e., whether it was the 'fault' of the Pioneer amp, or whether you simply badly mismatched an amp with a pair of difficult-to-drive loudspeakers (in which case, the fault is yours).

But psychological biases are *very* likely to lead a person to believe that an 'upgrade' improved output. Even if there wasn't really a change (a so-called 'phantom switch' experiment). An analogous experiment with taste includes a by now famous study of people given the same wine labelled with different prices. They tended to report that the pricier wine tasted better.


Please note that, as with all real audible difference, you could have 'verified' your difference independently of listening if you wanted to, with measurements. You could verify that your system is *broken*, without doing any listening to it at all...just by watching it shut down.

Last edited by krabapple; 07-03-2014 at 01:51 PM.
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post #1517 of 2920 Old 07-03-2014, 01:45 PM
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The Replaygain filter curve shown in Debate Thread: Scott's Hi-res Audio Test is no longer in use within foobar2000. foobar2000 developers assert Replaygain as currently implemented in foobar2000 uses the EBU R128 specs, which means it uses the following curve that does not truncate high frequencies

source: https://tech.ebu.ch/docs/tech/tech3343.pdf
If anyone has information that the foobar2000 developers didn't do what they claim to have done, please share your sources or analyses.

edit: while the image above stops at around 20k for display purposes, I see no high frequency cutoff in the underlying formulas of the filters that are used to construct that curve.
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post #1518 of 2920 Old 07-03-2014, 01:50 PM
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Quote:
Originally Posted by Kees de Visser View Post
I somehow expected that
OK. Use any DAW, open both stereo files in a 4-track project and do a nulltest (inverse polarity of one pair, mix them and listen to the master output). Now align them to the closest sample, then (optional) use a sub-sample plugin to minimize sub-sample sync difference and adjust level of one pair to find the best null. In this case the best null was with the hi-res version at -0.2dB. I mentioned this to illustrate that no psy model was used to estimate the loudness difference. It was my personal hearing, using the full bandwidth difference signal.
Another method would have been to listen to both Left or both Right channels of the two samples in stereo, swap L/R frequently while listening and adjust the level of one pair to obtain minimum center image displacement.
Not sure I understand...you listened to the *difference* file and perceived that you could hear something?
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post #1519 of 2920 Old 07-03-2014, 02:03 PM
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Quote:
Originally Posted by Kees de Visser View Post
I somehow expected that
OK. Use any DAW, open both stereo files in a 4-track project and do a nulltest (inverse polarity of one pair, mix them and listen to the master output). Now align them to the closest sample, then (optional) use a sub-sample plugin to minimize sub-sample sync difference and adjust level of one pair to find the best null. In this case the best null was with the hi-res version at -0.2dB. I mentioned this to illustrate that no psy model was used to estimate the loudness difference. It was my personal hearing, using the full bandwidth difference signal.
Another method would have been to listen to both Left or both Right channels of the two samples in stereo, swap L/R frequently while listening and adjust the level of one pair to obtain minimum center image displacement.
Listening tests based on electrical difference tests can be fun, and even sometimes diagnostic qualitatively but they are not quantitatively representative.

The representative test is what I recommended - ABX the files as is. If you want to find out how audible 0.2 dB is, take two files, attenuate one by 0.2 dB and ABX your heart out. To me that is easier than working with nulling two files, and its obviously a ton more real world.
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post #1520 of 2920 Old 07-03-2014, 02:05 PM
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Originally Posted by krabapple View Post
Not sure I understand...you listened to the *difference* file and perceived that you could hear something?
Not news.

With good file alignment you can increase the size of the difference by 100s of times, even thousands of times (60+ dB).

While 0.2 dB is generally not audible (but > 0.3 dB can be under ideal conditions) a good nulling scheme can turn that into a 20 dB difference. Sure, you can hear it, but only if you are not deaf! ;-)
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post #1521 of 2920 Old 07-03-2014, 02:13 PM
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Quote:
Originally Posted by urapnes1 View Post
When you wrote what I quoted below, I interpreted it to mean that a bias controlled comparison is required in order to hear an audible difference.
The words bias controlled are wholly gratuitous in this context.

When a piece of equipment turns itself off and flashes a red light at you, no listening test at all should be needed.

The original (ca. 1975) statement of the requirement for bias-controlled testing was:

"Bias controlled listening tests are required for all situations involving subtle differences."

At the time we thought we knew what a subtle difference was, but of course that was about 40 years of dogmatic posturing by the high end press, ago. There barely were high end audio cables back then.

Call me old fashioned, call me dogmatic, call me simplistic but I can't see a piece of equipment that turns itself off and flashes a red light as being "a subtle difference"
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post #1522 of 2920 Old 07-03-2014, 02:14 PM
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I am not sure I understand. Wasn't Kees just using the test as a diagnostic, not that he could pass an ABX test of the two files? Seems to me they were just making sure that the two files were correctly leveled. Otherwise, someone could complain later.. Kees (or ISA), correct me if I am wrong.
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post #1523 of 2920 Old 07-03-2014, 02:14 PM
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Quote:
Originally Posted by krabapple View Post
Not sure I understand...you listened to the *difference* file and perceived that you could hear something?
Not a difference file, but a difference mix: the real time output of the DAW mixer. The advantage is that every adjustment can be heard right away. If two files are identical, the mixer output will be -INF. In this case it wasn't, which is to be expected. But the difference was still too large to be caused by a SRC process. With a DAW and a bit of experience it's possible to minimize the difference in a few minutes time. These differences can then be applied to the source files, so that other listeners don't have to apply any replay-gain or other corrections before starting the test.
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post #1524 of 2920 Old 07-03-2014, 02:23 PM
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Quote:
Originally Posted by amirm View Post
Going back to Scott's level discussion, I ran the same analysis on Arny's "jitter" files:



Focus on the last two: no jitter and max jitter. The difference in "level" is 0.6 db or so.

Is this a problem or not a problem? If it is a problem to have 0.6 db difference then why were we presented with this test? If it is not a problem, why do we make a big deal out of it???
As usual, its a matter of context. The sequence of files that I provided (in the reverse order shown above) were composed of a mixture of training files and testing files. The first two highest jitter files ( the ones marked 0.01 and 0.025) were for training, and only the last two files actually involved a serious challenge. There was one outlier by any reasonable standard and it was the first training file, the one titled "max jitter".

OK, the max jitter file sounded different for a number reasons, and the slight shift in level might have been one of them. Its purpose in life was to convince people that they could hear a difference in an ABX test. And based on my listening the most obvious difference was the rough texture that was added to the music which is characteristic of audible FM Distortion with the modulating frequency that was used in the test. It was very representative.

BTW that first "Max Jitter" training file had so much jitter that it is possible that it may have given a stronger indication in Replaygain than it did with the diagnostic tools that I used.

I didn't label the files as overtly as I might because I didn't want to bias the listeners.
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post #1525 of 2920 Old 07-03-2014, 02:31 PM
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Originally Posted by krabapple View Post
...

That fact that humans are not perfect recorders and interpreters of objective reality is the reason why controls exist in science. That many people don't seem to realize this is one of the failings of science education as it is currently done.
When was the education such that people didn't have a problem properly interpreting their senses?
Human gullibility is very powerful, innate part of us; few can have it under control and even well educated even in the sciences are not immune.
Hard work.
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post #1526 of 2920 Old 07-03-2014, 02:43 PM
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Quote:
Originally Posted by fotto View Post
Quote:
Originally Posted by stereoeditor[B
][/B] I am in the office right now and don't have access to the files, but I will post the spectrum and level of the background noise in my urban dedicated listening room later.
Could you elaborate on what method and equipment is being used to measure this background noise?
I have three spl meters, ranging from an antique RadioShack analog meter to the combination of the Studio Six Digital SPL meter app for the iPad, which is used with Studio Six Digital's iTestMic. (The Studio Six iTestMic is supplied with a correction file, generated by comparison with a B&K lab mike.) I also have an AudioControl SA3050A 1/3-octave analyzer, which has an spl function. These all systems agree within 3dB or so but I haven't checked them against my DRA Labs MLSSA system, which is used with a calibrated DPA 4006 mike but is housed elsewhere in the house.

Using the Studio Six app on my iPad and the iTestMic, the spl of the background noise in my dedicated basement listening room measures 40.5dB(C) and 28.6dB(A), both taken with the meter's ballistics set to Slow. To generate the attached spectrum of the noise in my room, I played a 1kHz tone at 95dB spl at the listening chair, measured with the Studio Six app & mike, and adjusted the preamp gain on an EMU404 USB ADC to give a digital level of -3dBFS. I then turned off the tone and captured the room tone for 30 seconds. Adding 98dB to each FFT bin allowed me to plot the spectrum with an spl vertical scale.

My room is very quiet, with the main source of noise distant traffic. (I don't have central AC.) However, it is not as quiet as some of Stereophile's reviewers' rooms, who live in rural locations. Acoustic treatment uses ASC Tube Traps and RPG Abfussors, and three of the four walls are lined with bookshelves and CD/LP racks. My goal was to make the room articulate and uncolored but not too dead. Rt60 is an even 250ms in the midrange and low treble and increasingly lower than that in the top two octaves.

Now that I have posted the spectrum of my room's background noise, I anticipate that other AVS forum members will do likewise.

John Atkinson
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post #1527 of 2920 Old 07-03-2014, 02:45 PM
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Originally Posted by imagic View Post
There is an easy way to test if you can hear ultrasound. Buy one of those ultrasonic pest repellers. Better yet, buy a four pack and plug them all of them into a power strip. Turn it on and off. Do you hear anything? Do you feel a tingle? It's similar to trying to see the infrared light coming out of your remote control.

Yes, but, are they a reliable source for ultrasonic sound? What frequency? Does it even emit anything?
Or, just another marketing gimmick to the gullible masses.
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post #1528 of 2920 Old 07-03-2014, 03:34 PM
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That was generous and thoughtful of you John, seriously. No central air (not such a big surprise in B'kln) but does the room have a window unit?
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post #1529 of 2920 Old 07-03-2014, 03:49 PM
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The words bias controlled are wholly gratuitous in this context.

When a piece of equipment turns itself off and flashes a red light at you, no listening test at all should be needed.

The original (ca. 1975) statement of the requirement for bias-controlled testing was:

"Bias controlled listening tests are required for all situations involving subtle differences."

At the time we thought we knew what a subtle difference was, but of course that was about 40 years of dogmatic posturing by the high end press, ago. There barely were high end audio cables back then.

Call me old fashioned, call me dogmatic, call me simplistic but I can't see a piece of equipment that turns itself off and flashes a red light as being "a subtle difference"
I agree. If it shuts down something is wrong. However, in my case it took well over 40 minutes before failure. What I don't know is whether or not there were issues present before it died. I thought I heard differences....I know I know... But I guess I have a question. When we sit down to evaluate a component is there a generally agreed on approach as to when we can...immediately upon power up, 10 minutes after power up, 60 minutes? 24 hours? In my case a test conducted at initial power up might have been different than let's say 35 minutes in when the avr was close to giving up? I have seen people mocked for saying something sounded better once it warmed up. Perhaps there is something more to this.

My gut tells me that you want to evaluate it in normal listening, which to me says let it run for an hour to let it get itself to normal operating temperatures.
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post #1530 of 2920 Old 07-03-2014, 03:54 PM
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Perhaps you provide some examples from your own listening room since you seem to have an unbeatable pile of expensive measurment equipment at your disposal.
I know you addressed this to Amir, but I just posted the spl and spectrum of the background noise in my listening room. The spl meter (iPad with $199 Studio Six iTestMic) and mike preamp-A/D converter (an E-MU 0404 for which I paid $180) were inexpensive and while I used a pricey Earthworks QTC-40 omni mike for the spectral analysis, there is an excellent Behringer omni mike, the ECM8000, available for just $60. I performed the FFT analysis with Adobe Audition, which I know you already own.

Look forward to seeing a similar analysis for your room, Mr. Krueger.

John Atkinson
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