Spectral X-Contamination: Problems in Op-Amp Chips Page 2

Conventional acoustic and audio knowledge has little enough to say about the sonic effects of harmonics when they're gross (ie, 50% to 0.5% of the fundamental). At lower levels, almost nothing is documented, and extrapolation of the high-level effects would seem presumptuous. I have witnessed experiments in which the effects of introducing at below –95dB a spray of harmonics, including the seventh, ninth, and eleventh, by loading-down an IC op-amp, are: a) perfectly audible, but b) perceived more as a dulling loss of detail and air than as dissonant in the usual sense of "anti-melodic" or "gritty." There is much yet to discover.

Spectral X-Contamination
When I looked at crosstalk with the spectral-analysis test setup, the results were surprising: Unlike everyday capacitive leakage, in which high frequencies dominate (see a typical channel-separation plot in a Stereophile amplifier or preamplifier review), the crosstalk spectra were generally higher in level at stimulus frequencies below 1kHz. I therefore chose to use a 50Hz test tone as being the frequency of lowest stimulus that would provide easily identifiable integer harmonics while also allowing reasonable resolution. This is because the goodness of the AP test set's 3Hz selectivity disintegrates progressively with reducing frequency; eg, 3Hz is nicely selective relative to a 1kHz fundamental, but gets to be much wider down at 10Hz.

For easy identification in the results for individual op-amps (figs.5–22), the harmonics all fall halfway between the vertical grid lines. The generally leftmost spikes at 100Hz show the second harmonic, those at 150Hz the third, etc. The harmonics seen in the crosstalk spectra had no relation to those previously logged (in more ordinary spectral tests) at the ch.1 output, nor to the AP generator's own residue.

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Fig.5 Analog Devices AD 823, recent design, dual. 600 ohms: second @ –134dB. 4k: All low.

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Fig.6 Elantec EL 2244, recent design, dual. 600 ohms: second @ –109dB. 4k: second @ –110dB.

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Fig.7 Harris HA5102, 1984 design, dual. 600 ohms: second @ –126dB. 4k: second @ –145dB.

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Fig.8 Harris HA5104, 1984, quad. 600 ohms: second @ –135dB. 4k: Low.

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Fig.9 Harris HA5222, ca 1990, dual. 600 & 4k ohms: Low.

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Fig.10 Motorola MC 33079, ca 1990 quad. 600 ohms: second –117dB, third –133dB. 4k: All low.

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Fig.11 National Semiconductor LM 833, 1984, dual, made specially for audio. 600 ohms: second –124dB, fourth –143dB. 4k: All low.

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Fig.12 National Semiconductor LM 837, quad of 833. 600 ohms: second –, third to 11th spray, –114dB to –143dB. 4k: second @ –128dB.

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Fig.13 Signetics (Mullard/Philips) NE 5532, 1977 design, dual, and used throughout most recording chains. 600 ohms: second @ –134dB, fifth @ –145dB. 4k: fifth @ –144dB.

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Fig.14 Analog Devices OP 275, ca 1992, dual. 600 ohms: dominant second, low. 4k: Low.

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Fig.15 Analog Devices OP 279, recent, dual. 600 ohms: Spray –112 to –120dB. 4k: second.

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Fig.16 Burr-Brown OPA 2604, recent, dual. 600 ohms: second @ –115dB. 4k: All low.

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Fig.17 Burr-Brown OPA 2132, dual, made for audio. 600 ohms: second @ –142dB. 4k: Low.

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Fig.18 Texas TL052, 1989, dual. 600 ohms: second @ –128dB. 4k: Low.

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Fig.19 Texas TL054, quad of TL052. 600 ohms: second at @ –127dB. 4k: Low.

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Fig.20 Texas TL072, 1976 design (made ca 1990), dual, and used throughout most recording chains. 600 ohms: second @ –111dB, third @ –140dB, sixth @ –143dB. 4k: Low.

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Fig.21 Texas TL074, 1976 (made ca 1988), quad. 600 & 4k: All noise @ –130 to –135dB, second about –132dB.

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Fig.22 Texas TLE2142, ca 1992, dual. 600 ohms: second @ –115dB, third @ –130dB. 4k: Low.

In the plots, spectral components that lie below –145dB, or one part per 20 million, are ignored in the captions. "Low" means "below –145dB," "Dom" means "dominant harmonic." In each case, "4k" means with just the 4.3k feedback resistor loading, while "600" means with an overall 600 ohm load. Remember, what's seen is not the op-amp channel own spectra, but harmonic contamination from a channel that might be doing another job altogether.

I was surprised to see that nearly all the ICs showed a predominant second-harmonic spike. However, the crosstalk harmonic spectra of Motorola's MC 33079 (fig.10) and the Texas TLE2142 (fig.22) differ from most in having a large third harmonic alongside the second with the harsher, 600 ohm load. Note also the distinct fifth harmonic, which appears even in the lightly loaded "4k" condition.

With a dual chip, there are only two channels to crossfeed to one another. However, the test permutations are rather greater with quad ICs. For simplicity, only a quad's ch.1 was driven, and only ch.3 was sensed—the one diametrically opposed to ch.1 on the package. This delivered two surprises: the continuing occurrence of a strong second harmonic; and, in a rather old TL074 sample (fig.21), while noise is rather high, there are no discernible spectra, leading to the conclusion that there are at least two wholly separate chips inside. As a quad die costs far less to produce, the TL074 probably isn't made this way any longer. This shows that, just as with tubes, long-lived IC models of a given type, even from the same maker, can perform and sound quite different over a period of years as their production "evolves."

Thermal Modulation
Derek Bowers, an IC designer who has designed some of the world's best-sounding audio ICs for SSM, PMI, and AD in the US, agreed with my impression that Stereophile readers are among the first in the world to clearly see, in figs.5–22, the effects of "thermal modulation," aka "thermal distortion," in IC op-amp chips. The existence of this as an unconventional mechanism in solid-state audio sonics, whether for good or ill, has been strongly denied by Hard-Line Objectivists, who cite the convenient absence of measured proof. One HLO even performed futile tests with inadequate equipment to "prove" the nonexistence of thermal modulation.

Yet, in listening tests, good audio designers have picked parts that were later discovered to be more thermally stable. And in order to make ICs used for purposes even more demanding of speed and settling times than audio, IC makers have not only had to learn to control and minimize thermal distortion, but have also had to learn to recognize four different kinds, according to their widely varying timescales. The longest-lived of these thermal distortions would explain the benefit of warming up solid-state gear over hours—yet another effect that perceptive users notice and that HLOs ridicule.

How can such tiny results matter?
Hard-Line Objectivists will claim that the tiny levels being measured are utterly insignificant alongside, say, the grosser distortions of loudspeakers. This assessment, too, is too simplistic. The size of something, enumerated in one dimension alone, does not necessarily describe its potency. Homeopathy, biological warfare, holography, and military intelligence, to name but a few, are fields in which exceedingly small amounts of an active constituent can act very powerfully for good or ill. Just so, signals below the floor of various flavors of noise and distortions picked up on the way are by no means automatically masked or made inaudible.

The audio masking theory on which the development of data-reduction codec is based is not the last word, in my opinion. Consider how someone using a noisy lawnmower will still be able to hear their name being distantly called; or how Morse code is still preferred for some jobs because, unlike comms-grade telephony, it can be heard from beneath 20dB or more of static and hash.

First, we should ask whether we are measuring the right things to best describe what we hear. We might not be able to change a test method overnight, but the "weak" measurement of a quite audible effect can then be understood. Most audio measurements are acknowledged by those who listen and test to be of only marginal musical relevance; while better than nothing, it is only a fraction of what we might measure. The late Richard Heyser anticipated the dimensions of audio testing in his essays of a quarter century ago [15,16,17,18]. The equally masterful late David Bohm's "theory of everything"—Implicate Order [4]—explains neatly how it is possible for human senses to perceive effects that are almost too slight to identify within the current dimensions of measurement.

This is particularly true of audio. Bohm wrote, "In listening to music, one is...directly perceiving...an implicate order." In a sense, he means music transmits or permits higher-dimensional events to be played out in our everyday "3 (space) +1 (time) dimensional manifold." Anyone who's experienced musical reproduction that seemed to create a space that was "more real" than ordinary reality will know this.

Second, the effect of the spectral breakthrough becomes more significant than simple noise breakthrough when the channel being contaminated is passing a signal that's either that of another music channel, or a same-channel signal that's later or earlier in the process. When considered in the context of the music signal, the added second harmonic content is almost wholly benign, at least in itself. But having this phantom, pitch-doubled signal (one way to think of the second harmonic!) leak into a later or earlier stage of an amplifier—RIAA phono equalization, for example—and with random phase changes, would seem to risk causing peculiar sonic effects.

Brave Conclusions
With good playback equipment, my experience suggests that you can clearly hear harmonics below –95dB changing in just one IC of the hundred preceding it in the total record-replay chain (see sidebar, "Acid Tests for Musically Refined Electronics"). The measurements presented in this article surely demonstrate, therefore, that the use of almost any one of today's dual and quad op-amps (let alone those of yesterday that are still in use) in any high-quality musical signal path must at best add another layer of unnecessary chaos or dithery noise. At worse, we're talking about serious grunge. Could this be a cause of some of the nastiness that afflicts so many modern recordings?

The only dual op-amps of those tested that would appear to be acceptable for large-scale use in music recording and high-end reproduction paths would appear to be some made by Harris (fig.9) and Burr-Brown (footnote 5) (figs.16 and 17).

If these dark ages of audio reproduction are at some future date recognized as such, we can be sure that the arrogance of some large makers of pro-audio recording equipment ("audio quality is longer an issue"), and the collateral damage their cheapskate engineering (footnote 6) has done to musicians in the past 28 years, might be a cause for regret for more musically aware future generations. The lesson for high-end audio and the recording-equipment industry, if it ever wakes, is that we must make our own parts, or at least learn to trust no one to look after audio interests.

Acknowledgments
The author is indebted to Russ Andrews and audio professionals Tony Andrews, David Cole, Lawrence Dickie, Neil Grant, and John Newsham for their helpful input on listening and higher audio perceptions; and to David Heaton of Audio Synthesis, who helped make the testing possible.


Footnote 5: A small, well-loved, dedicated US analog IC maker, bought out in September 2000 by Texas Instruments.

Footnote 6: Every time an engineer specifies a dual IC of the NE5532/TL072 ilk, he saves about 20 cents in parts, making an end-user price savings of typically $400 (0.15%) in a $250,000 premium recording console. Savings in space are more difficult to quantify. But if the recording industry could grasp the concept of minimal signal paths, then the PCB real estate needed to use the better-sounding single-channel ICs might be less costly.

COMMENTS
CG's picture

Very interesting reading.

There's far too much to comment upon here. Not the time or place... (Not to correct, but to ask for expansion. More of this would be great!)

One comment I will make. As Mr. Duncan pointed out, IM products tend to rise and fall over real time as the relative phases of the contributing signal components change. Other effects, such as thermal changes, also create different distortion patterns over time. This is why averaging the spectrum can be misleading.

If you look at the distortion spectrum over time with a more or less real time spectrum analyzer, you can see these peaks changing. The distortion products when averaged look like bumps in the noise floor. But, if you zoom in, they're more like fields of grass swaying in the wind. This isn't unique to audio systems - you can see the same thing with various RF systems as well.

What you really hear in an audio system are the peaks of each distortion product as they change over time, not so much the averaged signal power. Listening to plain old noise is a constant change of spectral content and levels. You can hear that.

One immediate burning question - why do these design guys disappear into the aether so often?

John Atkinson's picture
CG wrote:
One immediate burning question - why do these design guys disappear into the aether so often?

From an email I received from Ben Duncan this afternoon: "I am gathering quite a list of Electronic Engineers who have been broken by the mental ordeal of designing and putting perfectionist products into production."

John Atkinson
Technical Editor, Stereophile

CG's picture

And, I'm not even in the audio biz.

But, the glory, heart felt appreciation of the customers, and the money makes up for it. Oh - wait...

Bogolu Haranath's picture

.... and all those 'electronic engineers who have been broken by the mental ordeal', are now listening to and designing, single ended, class-A, zero feedback, triode tube amps :-) .......

tonykaz's picture

I made an attempt at selling perfectionist audio products.

The perfectionist aspect is the consumer, not the Product.

Audiophiles are neurotic & psychotic.

Perfection is a moving target that Guru types review and approve ( this month ).

The headphone world seems to have an accessible and stable True-North relating to Sound Quality ( perhaps because there isn't a $50,000 Headphone Amplifier just being released )

Tony in Venice

Bill Whitlock's picture

The late Deane Jensen co-authored an AES paper (see https://www.aes.org/e-lib/browse.cfm?elib=4719) on this very subject and coined the term "spectral contamination" back in 1988. Available arbitrary waveform generators of the day had limited resolution, which limited the sensitivity of the relatively simple test. I'm hoping to revisit the subject and take advantage of the newest generation of test gear. I'm convinced that such a test is the missing link between what we hear and what we can measure because it reveals the non-harmonically-related distortion spectrum that's so irritating to our ears. For me, it also explains the widespread love of vacuum-tube and transformer containing audio gear! - Bill Whitlock, AES Life Fellow and owner/chief engineer of Jensen Transformers 1989-2014

CG's picture

First, thank you for the reply!

I *think* the problem isn't so much the generator of such signals as it is the spectrum analyzers used to measure the results. Or, at least, how those spectrum analyzers are used.

When you average a zillion sweeps to minimize the noise floor you get what we all see. Averaging reduces that noise because noise is random. But, essentially most of the potential problems then get averaged out, too. Even a 1 dB change in the measured floor because of a bunch of pseudo-random events just gets ignored. Plus, this noise floor is usually very much below other basic noise sources found in a typical listening room, or even out in the desert. So, how relevant is this averaged noise floor then, really? Humans listen to discrete events, not the average.

Now, if a measurer used a "Peak Hold" or "Max Hold" display function, I suspect there'd be more detection of audible problems. That certainly is true in communication systems. (I bet the main effect of investigating this sort of testing would be to create yet another path for decades of arguments between audiophiles...)

Of course, a suitable generator would make life as a measurer easier.

BTW, Brockbank and Wass had some thoughts on the subject back in 1945. I guess some ideas just never catch on.

Ortofan's picture

... for the mill:

http://nwavguy.blogspot.com/2011/08/op-amps-myths-facts.html

http://nwavguy.blogspot.com/2011/08/op-amp-measurements.html

https://www.edn.com/negative-feedback-in-audio-amplifiers-why-there-is-no-such-thing-as-too-much/

https://www.edn.com/negative-feedback-in-audio-amplifiers-why-there-is-no-such-thing-as-too-much-part-2/

John Atkinson's picture
Ortofan wrote:
http://nwavguy.blogspot.com/2011/08/op-amp-measurements.html

From that article: "At gains of 5X and higher nothing could beat the $0.65 NE5532 except by a few dB in noise performance."

The NE5532 is the dual version of the NE5534, an op-amp that, as described by Ben Duncan on this article's "The Pink Floyd Connection" page, started life as the high-audio-quality, metal-can Mullard TDA1034. I first encountered the NE5534 in the phono stage of the Meridian 101 preamplifier and used it in a microphone preamplifier circuit I designed in the early 1980s.

Also from that article: "The older TL072, despite being used in thousands of audio devices, is way out of its league against newer op amps."

The Quad 34 preamp (or was it the 44) used TL072s. I asked Peter Walker why, when even then there were better-performing op-amp chips available? He felt that saving his customers money outweighed any measured advantage of higher-performance chips.

John Atkinson
Technical Editor, Stereophile

CG's picture
What most people measure is but a subset of the actual operating performance of devices. That's true whether it's an opamp or an amplifier in a snazzy box. There are practical limits of what a for-profit company can and will do. Really dedicated amateurs might measure more if they had the resources, which they almost always do not.

In addition, those for-profit companies have budgets for every bill-of-material used in each product. They all cut costs where they think they can, unless they can find a good reason not to. Usually, that comes down to the marketing of a product and what the market is willing to pay.

That's not a knock on these companies. For every potential customer who might say, "That's great! Exactly what I want! The price is OK, too. Sold!", there's probably 873 who will consider the price first and foremost. (That could go either direction - Inexpensive might be a negative in many cases.) I don't envy the decisions and trade-offs they have to make.

What Ben Duncan's article highlights to me is that in general, we really don't fully understand what makes a sound system sound good. There are products with relatively high distortion compared to others that are beloved based on listening. Others that would seem to be great based on the traditional measurements, but don't sound as good. If you dig deeply enough, you can usually discover why one product sounds more appealing than another. It might not be obvious, and the conclusion will almost certainly be contentious. But, you can often find it.

So, who is going to do that? They don't give out many grants for hifi performance research these days. Magazine reviewers and their editors? I don't think they have the resources. One resource being time. I also don't envy the reviewers or editors for the choices they need to make...

Ortofan's picture

... to sing the praises of the 5532/5534. His 4-part article in EE Times, entitled “Op amps in small signal audio design”, includes a survey of various devices.
He does suggest, though, that the LM4562 might finally be an improved replacement for the 5532.
Likewise, the OPA2134 is suggested as an upgrade for the TL072.
https://www.eetimes.com/op-amps-in-small-signal-audio-design-part-1-op-amp-history-properties/

The QUAD 44 preamp used many TL071 op amps – the single version of the TL072.

One advantage of the single 5534 is that the compensation pins are brought out, which enables the possibility of bypassing the input stage for applications where lower input bias current is needed.
An old Siliconix FET databook includes an application note illustrating such a configuration. The normal input pins are connected to the negative supply rail and the bipolar input stage is replaced with a cascode connected pair of 2N5912 dual FETs. The balance and compensation pins then become inputs to the second stage differential transistor pair.
While the 2N5912 no longer seems to be readily available, the Linear Systems LS5911 is a substitute.

One wonders if it ever occurred to Peter Walker to offer a “deluxe” version of his preamp with better performance, from the use of higher performance devices, at a commensurately higher price?
He might have responded that the existing design already performed as a “straight wire with gain” and asked what more than that did you need?

invaderzim's picture

"This has left the world with a legacy of smeared, mangled recordings"

It is so sad that many of us will never know what these performers really sounded like. The true performers from when our equipment just wasn't up to the task of capturing their sound.
Paintings can be restored, manuscripts can be repaired and duplicated but we can't clean or recreate the sound and have it be correct.

CG's picture

Perhaps - some day.

A bigger concern might be the care, or lack thereof, the keepers of the master tapes have shown over the decades. A lot has been thrown out and a lot has been lost to fire. There's some indications that the recording companies have invested a bit more to keeping the recordings - the ones they make their money from, which is crazy if you think about it - stored better. But, an incredible amount has been plain lost forever. I'm sure that has been the case with written documents and other forms of art, but the record companies haven't even been willing to protect their own assets.

CG's picture

I wrote a considered reply to a comment above. Nothing remotely derogatory, inflammatory, or impolite. No bad words.

Gone! It was saved, as shown by the blank text box.

Oh well. This may be a message from the cosmos that posting comments on the internet is not for me.

John Atkinson's picture
CG wrote:
I wrote a considered reply to a comment above. Nothing remotely derogatory, inflammatory, or impolite. No bad words. Gone! It was saved, as shown by the blank text box.

I don't know why the text disappeared but I inserted it from the email I was sent alerting me that there had been a response to one of my postings.

John Atkinson
Technical Editor, Stereophile

CG's picture

Thanks!

I still think the cosmos is trying to tell me something though...

tonykaz's picture

it happens, from time to time.

You can go back a page, in your computer, to retrieve it and re-submit.

It only happens in Stereophile Comments.

Tony in Venice

sdunbar's picture

As an Applications Engineer for Texas Instruments, I can tell you that with advancements in process (smaller die) and packaging (smaller packaged devices) there is very little reason to use dual or quad packaged op-amps in pro-audio applications.

Yes, one can still save some money using dual/quad packages, but the need to use these packages to save space is relaxed.

The OPA1641 that is now already 10 years old and available in a small VSSOP package could be a good start. Other very capable devices are in smaller packages, such as SOT-23. There are literally 100's of devices to choose from, and there is no need to use exactly the same op-amp throughout an entire signal chain. Part of the discretion of design is to use the right part in the right place. Some areas maybe you can cut corners if you know what you're doing, others you cannot.

Linearity is of course the key to good reproduction. I have yet to see a widely accepted definition of a test for characterizing this parameter. Something based on a comb of multiple, yet non-harmonically related tones and then measuring the residual harmonic and IM distortion and expressed as an RMS value might be a good start. It would be necessary to reduce the tone amplitudes so that the peak/average ratio of the combination is still within the boundaries of the op-amp. But clearly there is plenty to argue about here.

CG's picture

There is this:

https://www.ap.com/technical-library/using-multitones-in-audio-test/

A test like this lets you examine the odd and even order IMD products. The fundamentals aren't eliminated, though. That probably could be handled through software trickery.

I'd think that an even more complete test would be to weight the levels of the applied test tones spectrally, so as to simulate some facsimile of the music/sound track/spoken word power distribution. There's higher levels in the lower frequencies than at 20 KHz. Why not shape the test signal? At the analysis end, you could also potentially use A-weighting to approximate the human's ear's sensitivity.

In addition, using some kind of peak hold measurement so that the maximum value in each FFT bin is saved would be desirable. Although averaging helps you look at very low levels because of the randomness of true noise, it also misses distortion peaks that occur when the relative phases of the distortion tones line up to add. You'd hear these peaks, but an averaging scheme would, well, average them them out to be less representative of the reproduced sound.

As for whether it would be widely accepted, I'd doubt it. You can't get anybody to agree on anything. Don't you agree? :8^)

sdunbar's picture

If one knows where the fundamentals are (because the test gear is generating them) and there are no FFT leakage artifacts, it should be simple matter to arithmetically subtract out the fundamentals, leaving the HD and IMD products behind.

Notice how far they have backed-off the magnitude of each tone? No doubt to manage the crest factor/peak-to-average ratio. I would think there are diminishing returns to adding more and more tones; you can argue that more tones better represents real audio, but at the same time you still need to be able to accurately extract the fundamentals and resolve the FFT.

Real audio tends to have lower amplitude content at higher frequencies so using the same amplitude for every tone is probably not the right thing to do, either.

All of this needs to boil down to a single "number" that can be used for comparison purposes. So, RMS of residual.

Surprised that IEEE hasn't standardized something like this decades ago. In a sense, audio is such a commodity now it really SHOULD have industry standards. Since audio is so dependent on linearity, seems like an obvious parameter to quantify.

But your right... it's hopeless.

CG's picture

In the CATV world they've long relied on multi-tone testing to determine the linearity of their systems. That analog testing has been done for at least 40 years. There is CTB - Composite Triple Beat - for odd order IMD products and CSO - Composite Second Order - for even order products. Plus SNR. Three numbers, but people understand what they mean. Today, that's gone away a bit, since the digital TV channels are QAM modulation, so the distortion artifacts are a little different than they were for plain ole analog TV channels. But, the new tests are similar.

In ye olde analog telephone days, the multiple analog channels were frequency division multiplexed so that you could get more than one phone call per line. In fact, they used groups that would let them get hundreds. There, the testing was done with white noise, with some small frequency bands notched out. During testing the noise power in the notched-out bands was measured and compared to the overall power, giving a single performance number called NPR - Noise Power Ratio. IMD of white noise is more white noise, so this gave them a sophisticated way to measure not only the distortion of a group of channels, but also the basic noise level. They pretty much stopped using analog phone channels in the 80's sometime. They were performance testing way before that.

I don't know what the answer is. Perhaps somebody ought to just put a stake in the ground for some sort of test and see how it progresses. The problem then becomes, just what can we hear and what is bad for sound reproduction?

sdunbar's picture

It’s impossible to argue with someone else about THEIR hearing ability. Certainly doesn’t work with my elderly mother-in-law living with us.

I'm the first to admit that my own ears are probably not that attuned. While my own mother had a degree in music and was a piano teacher, I completely squandered that opportunity... nevertheless, I did listen to quite a bit of audio growing up.

Not to be an "HLO", but my general assumption is that if someone/anyone can hear some difference, then that difference can be measured since we can build test equipment that is far more sensitive/perfect than our ears. This article seems to support that assertion.

What’s aggravating is when such a claim is made, but then we’re just expected to take the listener’s word for it- show me on an analyzer, please.

This is further obfuscated when certain distortion products are considered "pleasing" to some people. It's fine if you like the sound of second harmonic distortion, for instance, but from a recording and reproduction perspective, it's still distortion.

CG's picture

Two things...

One is that being even a virtuoso on an instrument, or many, is no guaranty that you're a good listener. My wife went to a fancy music school, played with and was tutored by several recognizable musicians there, and so on and so on. She tells me that most serious musicians aren't concerned with fidelity when they are listening. They've certainly listened to a lot of lousy sounding music in bad halls, just as they've listened to great sound. For them, they hear the musical intent in their head. They fill in the blanks. So, don't feel badly about that part.

(Fun fact: After getting critical, umm, "compliments" from their teachers, most of the voice majors would go sing to themselves in the porcelain lined rest rooms, just to make themselves feel better. Most everybody sounded glorious in there due to the natural reverb. Not sure if the tuba players did that. The pianists certainly didn't.)

Second thing is that just because it's probably possible that we could measure whatever distortion if we only tried, that's not what's done. Test equipment is good in that it often measures one characteristic at a time. But, to do that, it often trades off sensitivity for some of the bigger picture. A good example is the use of averaging and very narrow FFT bins to be able to capture teeny tiny harmonic products. As you pointed out, that not only doesn't exercise the DUT with regard to crest factor considerations, but the distortion products that are subject to the moving peaks are deliberately ignored. Can your hearing system do that?

sdunbar's picture

Certainly this has been discussed ad nausium elsewhere, but whether or not something "sounds good" or "sounds xxx" where xxx is your favorite audio related adjective, will always be a subjective matter.

For it to be objective, it would have to be time and space invariant, and since we cannot transplant ears, etc... from one person to another, it will remain subjective. Even the most attuned, demonstrably astute, refined listener is still engaging in an experience that by its very nature is subjective, and their descriptions of the experience must then be so as well.

Everyone also has their own tastes that can be tempered only so much, and it just becomes more subjective.

There is nothing wrong with this; that's just how it is. There are things about an audio signal chain that may be objective. Linearity performance, cross channel contamination (hence the article), noise floor, power consumption, etc... all of these things can be objectively compared through experiments that can be reproduced at will.

While nothing can be "perfect" we can certainly compare two specimens to see which is better in any particular dimension. My fear is, and this is probably why so many of these audio electronics engineers have vanished, is that despite desirable and measurable improvements in signal chains, there's still some "audiophile" who just doesn't like how it sounds.

When it become irrational, it's no longer fun for the engineer.

CG's picture

I think that when it gets in the way of making a living, that detracts from the fun in a big way.

Audio is not my livelihood, so that's easy for me to say.

All that said, I think there's still more that could be measured in a way that would help give better definition of what is better. Obviously, we're not there yet.

For me, it's far easier to make adjustments and then measure through listening. Jumping to the end is probably not right, but it's hobby.

Herb Reichert's picture

And bravo (!) Ben Duncan for such a broad, knowledgeable, cultured view of some very complex (and thorny) topics.

I am envious of your skills.

hr

Ortofan's picture

... "Hard-Line Subjectivists"?
By not doing so, is he revealing his own biases?

Worth reading is the referenced letter from Bob Orban (split over two pages):
https://www.stereophile.com/content/scientists-ivsi-audiophiles-1999-more-letters
https://www.stereophile.com/content/scientists-ivsi-audiophiles-1999-more-letters-part-2

John Atkinson's picture
Ortofan wrote:
Worth reading is the referenced letter from Bob Orban (split over two pages): https://www.stereophile.com/content/scientists-ivsi-audiophiles-1999-more-letters . . .

Also in the referenced article, a superbly written letter from the late Harvey Rosenberg: www.stereophile.com/content/scientists-ivsi-audiophiles-1999-more-letters-part-4.

John Atkinson
Technical Editor, Stereophile

CG's picture

Dr. Gizmo sure was great!

I'm not sure that digital can't sound great, but an awful lot of recordings that were digitized were done in a hurry, with not much care as to the sound. Or, maybe worse - some, ahh, creative engineer (probably coked up to keep going at 3AM in order to get the releases out quickly) decided to add his or her own special interpretation of the original artists' and producers' work.

Ortofan's picture

... the "Digital Pact" neglected to ensure that the agreement included the recipe for second harmonic sauce.

Maybe Dick Burwen and Bob Carver should collaborate on a device to add transient and background noise into digital recordings to make them sound more like analog.

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