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'deaf' above 14kHz, appear to hear above that frequency -- hypothesis as to how.


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12 minutes ago, GregWormald said:

Individually: maybe the way I inadvertently tested a set of cables. I put in the new cables and loved the new clarity. A month later I noticed that I was turning the volume down or off often and not enjoying the music as much as previously. I went back to the old cables and my listening returned to *my* normal.

 

So, record listening habits, headaches, fatigue, etc. for a month or so, then swap the library for an MP3 version and record again. This could be arranged to be 'blind' for those that can't already tell the difference.

 

That's been done with equipment. A test where a randomly selected component (of two) was put into the circuit each time the system was turned on. The listener didn't know which one it was. The test circuit also tracked how much time the listener spent with each component over a period of a few months. The expectation was that a better sounding component would make it easier to listen to, and the listener would spend more time with it. In the end, the result wasn't very conclusive. I posted on this test before, but will need to locate the description for more of the details. It wasn't a true experiment, just a simple test by a single individual, but does point to a possible way to test these types of explanations.

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12 minutes ago, GregWormald said:

I wonder if the hearing but not consciously noticing might be responsible for: tiredness, headaches, less listening to music for enjoyment rather than music as a background, etc.

mmm, don't experience that with mp3 any more than I do with PCM rates...improvements in source to DAC chain are what have gradually whittled my

digital fatigue down

 

But prefer to only listen to mp3 if its music I don't know well or fairly simple piece; seems like a lot of musical detail gets blenderized/lost if I play familiar

complex music in reduced data density mp3 format on main system. Its like settling for generic vs artisan prepared food, obvious quality reduction.

Regards,

Dave

 

Audio system

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A remarkably simple test, which I find extremely reliable ... put on a piece of music, and it immediately feels 'right'; then the system is OK. But if you're not quite sure, and you think, "I'll have to try something else, and see how that goes ..." - then, the system has got a problem, a 100% guarantee of it, 😉.

 

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6 hours ago, davide256 said:

mmm, don't experience that with mp3 any more than I do with PCM rates...improvements in source to DAC chain are what have gradually whittled my

digital fatigue down

 

But prefer to only listen to mp3 if its music I don't know well or fairly simple piece; seems like a lot of musical detail gets blenderized/lost if I play familiar

complex music in reduced data density mp3 format on main system. Its like settling for generic vs artisan prepared food, obvious quality reduction.

You must have better hearing than I do -- but that isn't very difficult to achieve nowadays :-).   However, that almost 'blenderized' sound is a good way to describe the temporal distortions that I hear on MP3, WHEN I CAN HEAR THEM.   More often than not, especially on less dense material, at most I detect only a purely subjective and probably non-repeatable difference.

 

Where I do hear a true difference with MP3, again -- very seldom a casually noticeable difference -- is on complex material with intense detail.  I have NEVER detected anything more than a 'feeling' (unrepeatable) difference on non-dense recordings.   (Hear it more on moderately intense pop-rock, less-so on typical classical recordings.)

 

For casual listening, I hear NO difference, but again, my hearing is much poorer than when I was 30-40yrs old or younger.

 

I respect the amazingly acute hearing that some people DO have.

 

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8 hours ago, firedog said:

For the interested ones I point out the differences. Like other things in listening, once you are aware of them, you can suddely hear them. The downside is that I've ruined mp3 listening for them.😃

 

I assume you are talking about 320 Kbps MP3s (and especially not 48 or 96 Kbps as found on some internet radio streams)?

 

As an aside, I always wondered why most audio engineers and mixers are mature/old people. Why is this if they can’t hear as good?  Maybe they can hear "good enough"?

mQa is dead!

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14 hours ago, John Dyson said:

Where I do hear a true difference with MP3, again -- very seldom a casually noticeable difference -- is on complex material with intense detail.  I have NEVER detected anything more than a 'feeling' (unrepeatable) difference on non-dense recordings.   (Hear it more on moderately intense pop-rock, less-so on typical classical recordings.)

John

 How did you find the 2 examples I posted ?

I tried these on Headphone Reviewer Dale Thorn several years ago. Up till then he really hadn't noticed any differences

Alex

 

https://www.dropbox.com/s/z1c8tx8gf2khacq/Touch Yello - 04 - Bostich (Reflected) (Yello).mp3?dl=0

https://www.dropbox.com/s/f9idn860qudm4rq/Touch Yello - 04 - Bostich (Reflected) (Yello).wav?dl=0

 

How a Digital Audio file sounds, or a Digital Video file looks, is governed to a large extent by the Power Supply area. All that Identical Checksums gives is the possibility of REGENERATING the file to close to that of the original file.

PROFILE UPDATED 13-11-2020

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On 3/24/2021 at 4:48 PM, John Dyson said:

I don't know what this means, but before making any assertions -- my hearing is blind to tones > 14kHz.

Note exactly the language that I am using -- 'blind to tones',  not cannot hear >14kHz!!!

Your surprise suggests that you are assuming your hearing is linear. There is every reason to believe that human hearing, like every type of  neural processing, is highly  nonlinear. There is no way to characterize your nonlinearlity in a mathematical fashion without extensive testing, if its even possible. 

Custom room treatments for headphone users.

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4 hours ago, jabbr said:

Your surprise suggests that you are assuming your hearing is linear. There is every reason to believe that human hearing, like every type of  neural processing, is highly  nonlinear. There is no way to characterize your nonlinearlity in a mathematical fashion without extensive testing, if its even possible. 

I didn't suggest that hearing was linear, but 'someone' is forgetting about masking also.   Actually, with additional tones, hearing should become MORE blind to lower levels assuming the log relationships and masking.   I am hearing the opposite.  Also, the effect that I am hearing is more like 'pumping' or 'tape bias'. (which is what I had stated earlier here or somewhere else). which is NOT linear and also NOT what one would normally expect.   (Masking is used to hide some of the defects in mp3 encoding, for example.)

 

Again, NO WAY did I suggest that hearing is LINEAR?!?!?!?   Where did that come from?

 

 

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4 hours ago, The Computer Audiophile said:

In addition, frequency sensitivity depends on sound levels. At lower levels, our ear sensitivity in the bass and treble areas, drops significantly.

We are more we have wider response for loud tones, but we become more blind to tones nearby -- that is, masking is what should be happening, but I am hearing 'pumping' (an effect similar to tape bias).

 

 

ADD-ON:  Whoops, I forgot about the term 'pumping' in the audio processing sense, because I have my EE/math hat on when I am speaking up 'pumping' here.  Best way to describe it -- look up 'pumping' in the sense of parametric amplication.   THAT IS DIFFERENT than AGC pumping...   Sorry if there is confusion in the usage of terms -- seems like we re-use words in odd and confusing ways at times.

 

 

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7 hours ago, John Dyson said:

Again, NO WAY did I suggest that hearing is LINEAR?!?!?!?   Where did that come from?

You have an assumption that your hearing of a complex signal should be predicted by pure tone response, or that masking should behave a certain way etc.

Custom room treatments for headphone users.

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10 minutes ago, jabbr said:

You have an assumption that your hearing of a complex signal should be predicted by pure tone response, or that masking should behave a certain way etc.

 

* I won't mark 'disagree' in the previous message, because sometimes that also suggests 'disapproval', which I do not.   I believe that there is some misinterpretation going on here.

 

Interpeting what I was trying to say, in less technical terms...   Pumping a nolinear system can result in amplification, it can also just result in distortion -- but the amplification side of things is what I think that I was seeing.   I don't think that I was really specifying pure tones, and if I was -- then I didn't intend to.  (Parmetric amplification can be a lot like using an AC signal for a 'power supply' instead of 'DC'...   Well, in a way.)   Paramps were used in early microwave receivers because the gain wasn't really based on a resistive type transconductance (noisy) but instead of a reactive kind of transconductance -- lower noise.

 

Anyway -- my vision of things is somewhat complex, so communicating is sometimes tricky -- do I start writing more deeply technical things in my long paragraphs that are already too boring, or try to explain in more reasonable terms -- perhaps taking shortcuts?

 

 

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15 minutes ago, John Dyson said:

 

* I won't mark 'disagree' in the previous message, because sometimes that also suggests 'disapproval', which I do not.   I believe that there is some misinterpretation going on here.

 

Interpeting what I was trying to say, in less technical terms...   Pumping a nolinear system can result in amplification, it can also just result in distortion -- but the amplification side of things is what I think that I was seeing.   I don't think that I was really specifying pure tones, and if I was -- then I didn't intend to.  (Parmetric amplification can be a lot like using an AC signal for a 'power supply' instead of 'DC'...   Well, in a way.)   Paramps were used in early microwave receivers because the gain wasn't really based on a resistive type transconductance (noisy) but instead of a reactive kind of transconductance -- lower noise.

 

Anyway -- my vision of things is somewhat complex, so communicating is sometimes tricky -- do I start writing more deeply technical things in my long paragraphs that are already too boring, or try to explain in more reasonable terms -- perhaps taking shortcuts?

 

 

 

As an engineer you are tempted to analyze the human hearing system using tools in the engineering toolbox. We can analyze an electronic circuit pretty well using SPICE. 

 

The field of computational neuroscience is an attempt at understanding the brain in a similar fashion. Humanity isn't there yet. Thats for the normal human brain. You aren't normal. Elon Musk's Neurolink aims to record from 1000 neurons simultaneously but 100 billion?

Custom room treatments for headphone users.

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19 minutes ago, jabbr said:

 

As an engineer you are tempted to analyze the human hearing system using tools in the engineering toolbox. We can analyze an electronic circuit pretty well using SPICE. 

 

The field of computational neuroscience is an attempt at understanding the brain in a similar fashion. Humanity isn't there yet. Thats for the normal human brain. You aren't normal. Elon Musk's Neurolink aims to record from 1000 neurons simultaneously but 100 billion?

I use Spice when I need it.   All too often, if I need to emulate a nonlinear LF circuit, I can do it straight away in C++.

 

Spice doesn't do the kind of parametric nonlinearities like paramps very well.   It can, but not so good.

I used to buiild my own spice 3e2 back in the 80's, but I don't jump on spice every time I run into something.   A quick run for an amplifier of some kind, or a lower frequency nonlinear circuit - it is great.

 

For higher frequency stuff -- it is hit or miss.   However, it is super-duper good for a 1st order design, getting an idea of what is going on with things.

 

For using Spice for other things, like the DA decoder, you need reasonable models.  There are NO models for the selected components, so it is a cr*pshoot.   Instead of worrying about spice modeling, I simply translated the circuit into C++ software (easy for me -- been doing semiconductor circuits since the '60s) -- and then did a bisection scheme to find the correct parameters.   Frankly, I'd have to desolder a few of the diodes to get an idea of its magic characteristics for direct modeling (is it a low or high is diode?)  Obviously, they have good exponential characteritics, because that is a critical design feature of the attack/release circuit.   (I did a straight  exponential model -- it worked great, but curve-fitting was a b*tch, becuase there are 3 circuits where the  diode parameters are criticial, and they interact!!!)   Reallly messy.

 

Also, imagine modeling (fully) a DolbyA HW unit -- yikes!!!   Need-a-supercomputer time -- that is, if you have the models for the parts (which I NOW DO, after lots and lots of work -- I know some secrets!!!) :0>   yea, that'd be fun -- run a Spice emulated DolbyA unit to decode commercial recordings -- YEA -- that's the ticket!!!

 

So -- spice is really good for certain things, but there are better packages for certain nonlinear analysis -- e.g. for parametric nonlinearities.  IT CAN DO IT -- but I don't have any projects like that right now.

 

However -- Spice can be a lifesaver when a circuit might be hard to conceptualize...

 

John

 

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On 3/24/2021 at 8:48 PM, John Dyson said:

Here is an interesting datapoint.   This is about HF hearing, not about the software that I am working on...

 

I don't know what this means, but before making any assertions -- my hearing is blind to tones > 14kHz.

Note exactly the language that I am using -- 'blind to tones',  not cannot hear >14kHz!!!

 

During testing, I found that I made a mistake on one of the equalizers in my SW -- caught by hearing an anomaly in the sound...

I was wrongly using an equalizer that is an HF shelving filter of 21kHz with a rolloff of -3dB.  (Yes, such small details are important.)

Instead, I should have used 18kHz, with a gain of 18kHz/21kHz.or -1.34dB.  (  I happened to choose the  wrong

filter spec because I was not thinking clearly, and should have chosen the 18kHz EQ to begin with...)

 

Even though when considering the normal hearing rolling off at 18-20kHz, and my hearing rolling off at 14kHz, I could hear a profound difference

between the 18kHz, -1.34dB 1st order EQ vs. 21kHz, -3dB.   The 21kHz, -3dB rolloff seemed to leave more highs in the sound, but the highs seemed distorted.

These kinds of distortions or 'tells' are why I can do EQ in certain cases in frequency ranges where I cannot hear well. Oddly, this scheme doesn't

appear to work at freq <1kHz or so.)

 

Now -- about this strange HF hearing.   A 1st order EQ of 18kHz, -1.34dB  has a small effect at 14kHz, and even at 14kHz, I my hearing of

tones is more weak than at 10kHz.   The difference that I am hearing is more significant than the sub-dB change that technically is happening

in my normal range of hearing.

 

This is a guess, and for EEs they will understand this hypothesis:   I believe that my hearing, blind at 14kHz is sometimes getting the effect of parametric

amplification, perhaps the nonlinear effects at frequencies above my normal range of hearing.   So, with the co-incedence of energy in the HF region, a kind of parametric amplification

or biasing of my hearing can boost my ability to hear transients above 14kHz.   I don't think that I  can ever hear 'tones' at 18kHz, but I am hearing some kind of

distortion as a result of co-incedental energy.

 

This biasing/parametric effect thing is the REAL reason why analog tape bias works.  It makes me wonder if the same kind of thing is happening, but

with a different result or purpose, for hearing >14kHz?

 

For those into 'measurements' -- this might be an interesting project, and one reason why some 'golden ears' honestly complain about differences when

BTW is wider/narrower than 20kHz, yet most of 'em cannot hear much above 18kHz or even less....   I really don't think that this is much about

'filter skirts', but might be a contributing factor.   Also, there can be issues associated with time delay (e.g. high order analog filters or not-linear

phase digital filters causing time delay differences.)

 

Just an observation -- but also a helpful odd artifact of human hearing (or at least, my own human hearing.)

 

ADD-ON:  I noticed the 'distortion' on processing the Supertramp recordings this morning, but heard it again when processing 'Olivia's' recordings.   I did a review and found the bug described above.  Once the 'bug' was corrected, the distortion went away.

Another ADD-ON:  This has manifest on different headphones & different situations in the past.  I just finally realized that this might be an interesting issue for someone to research.   I am sure that at the levels that we are speaking, this isn't a headphone nonlinearity matter, and I am using high sample rates in the 88.2k/96k range for the ultimate D/A conversion.

 

 

 

 

 

 

 

John, what speakers or headphones are you using for monitoring your work?

"Science draws the wave, poetry fills it with water" Teixeira de Pascoaes

 

HQPlayer Desktop / Mac mini → Intona 7054 → RME ADI-2 DAC FS (DSD256)

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11 minutes ago, semente said:

 

John, what speakers or headphones are you using for monitoring your work?

Speakers are an anathema for the kind of subtle distortions that I have to listen for.

I don't even care about speakers -- I just don't like speakers for the highly technical work -- but for listening, they are great.

 

I use the most evil headphones available and selected for-purpose.  Beyerdynamic DT770/80.

You might claim -- there are prettier sounding headphones, yes that is correct.

 

Take a look at the characteristics and what I am doing -- the DT770 is perfect.   A peak near the 7-10kHz range, and bass down to DC with almost no rolloff.   This gives me a picture of two of the critical parts of the spectrum.

 

OTOH -- I found that I need 'pretty' headphones that are closer to what people normally use.  I am thinking about some Sennheisers with a ltitle more flat HF, and the nice LF rolloff that keeps the air pressure from hurting my ears.  (My LF hearing disappears, but I can still feel the air-pressure.)   When headphones have response that go down as low as the DT770, one needs to be VERY careful about the LF EQ.   This does give me an indication of too much LF, but also takes away from the enjoyment when the flat LF sometimes causes some misjudgment.

 

So, I am NOT mastering -- I am hunting for 'tells' in the signal that show that there are messed up joins across the frequency range.   Until I gained this experience, I didn't even realize that there is a LOT to EQ that I was clueless about.   I had only encountered multi-pole EQ, but most of this DolbyA and pro-stuff is single pole, and it acts WAY WAY WAY different than multi-pole.   There is no concept of adding bass by increasing the single pole response  below, say 50 or 100Hz.   The single pole EQ is about attaining a flat response without any 'tells'.

 

If you want to EQ and master for listening purposes, except for some very limited cases, 2nd order, parametric are the best bet.   I have added some EQ modes to the decoder that change the 'balance' between the highs and lows, but not 'bass boost' or treble boost/cut per-se.   BTW, the decoder has some really high tech 2nd order EQ with the anti-distoriton attribute.   I doubt that most people would want to dial in the EQ forumulas that cancel distortion artifacts -- and it really DOES work.   With my 2nd order EQ technique, it is so easy to control 'rough' sibilance by even a 0.375dB tweak.   (It is about RELATIVE levels, you boost/cut.)   OTOH, sibilance IS a problem on some older recoridngs -- Simon & Garfunkel recordings DO have too much sibilance (as mentioned by someone directly hearing the tapes, and myself dealing with them.)   So, there is some dynamic automatic parametric EQ going on if you ask for the anti-sibilance.   (There is also a fixed ant-sibilance -- mostly useful if there was a mastering peak in the recording.)

 

SO -- speakers -- NAY.   They are mostly an anathema to 90% of what I need to do.   I even now know the tells for stereo image issues -- and since I am only replicating a master tape, and not changing the stereo image -- I just don't need speakers in my living room or bedroom.   (Sorry, I know that it is disappointing, but if I did get some speakers, I'd get some powered monitors -- but without mastering, why bother?)

 

QUESTION:  do you want the decoder adjusted for my listening environment.   WIth the headphones, there is a direct coupling - no room effects, not dependent on seating position.   The headphones DO have personality, but I have quantified them.

 

 

 

 

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