Erik,
See my recent response to you in the MQA "Philosophy" thread regarding Mr. Craven's "apodizing" filters.
Best Regards
cj
See my recent response to you in the MQA "Philosophy" thread regarding Mr. Craven's "apodizing" filters.
Best Regards
cj
MQA and the "Pre Ring - Post Ring" Hoax
There's been a lot of misinformed babble on various audio forums about impulse response, digital filters, "time errors", "time correction", "time blurring", and similar pseudo science clap trap to convince audiophiles that suddenly in the year 2018, there's something drastically wrong with digital PCM audio - some 45 years after this landmark technology was developed by Philips Electronics engineers. Newsflash folks - it's a scam.
First, let's take a close look at what an impulse or discontinuity signal really is. The wikipedia definition actually is pretty accurate thanks to a variety of informed contributors from around the globe. It is a infinite aperiodic summation of sinusoidal waves combined to produce what looks like a spike (typically voltage for our purposes) in a signal. Does such a thing ever occur in nature or more importantly in our case - music? Absolutely not. In fact, the only things close to it are the voltage spikes that occur when a switch contact is thrown or an amplifier output stage clips because supply voltage to reproduce the incoming signal waveform has been exceeded. So if this freak of nature signal representation doesn't exist in nature or music, of what good is it in measuring the accuracy of audio equipment? The answer might surprise you.
In fact, impulse response, or an audio system's response to an impulse signal, is one of the most useful and accurate representations in existence of such a system's linearity and precision - or its fidelity to an original signal that is fed to it. A lot of focus has been placed on the pre and post ringing of these "discontinuity signals" but what you have to understand is that the ripple artifacts are nothing more than an analog system's (all electronics is analog -digital is just a special subset of analog) limitation in attempting to construct the impulse or discontinuity signal waveform. They are a result of the impact produced by the energy storage devices themselves in creating the signal. To create a large energy peak, you need large storage devices. The larget the capacitor for example, the longer in time it takes for it to absorb and discharge electric field energy. This is the same with inductors. One type stores electric field energy - the other magnetic. Smaller value capacitors can react to voltage changes very quickly but are limited in the peak value of energy that can be stored and dissipated. But if you combine a large number of high value and low value devices in a circuit and apply a voltage spike, you wind up with the kind of oscillations you see in an impulse response graph. Small capacitors for example, rapidly reach their charge capacity and can discharge into larger capacitors that are much more slowly building up charge in the transition from no input voltage to full spike value. This "sloshing around", if you will, or oscillation is what happens in circuits built to provide extreme voltage attenuations. In a linear, time invariant system, any rapid change in frequency response or time response - has these characteristics.
So effectively the entire debate about ringing in digital audio is a misnomer - a hoax. The impulse response ripple is not something that happens in real world sounds or in a properly designed audio reproduction chain. Ever since digital oversampling was developed in consumer products in the early 1980s, there has been no need for steep analog filter circuits with their attendant ringing. The problem very simply DOES NOT EXIST. The ringing generated artificially in an impulse signal is useful in that it provides a very high frequency stimulus to linear audio systems as a means of measuring high frequency and transient response. IT IN NO WAY BY ITSELF, REPRESENTS THE TIME DOMAIN BEHAVIOR OF THE AUDIO REPRODUCTION CHAIN. An accurate audio reproduction system should fully render the impulse signal in all its pre and post ring glory without alteration. Any audio system that eliminates or significantly alters this pre/post ringing present in the signal that is fed to it is not truly "high fidelity" and is thus bandwidth limited.
First, let's take a close look at what an impulse or discontinuity signal really is. The wikipedia definition actually is pretty accurate thanks to a variety of informed contributors from around the globe. It is a infinite aperiodic summation of sinusoidal waves combined to produce what looks like a spike (typically voltage for our purposes) in a signal. Does such a thing ever occur in nature or more importantly in our case - music? Absolutely not. In fact, the only things close to it are the voltage spikes that occur when a switch contact is thrown or an amplifier output stage clips because supply voltage to reproduce the incoming signal waveform has been exceeded. So if this freak of nature signal representation doesn't exist in nature or music, of what good is it in measuring the accuracy of audio equipment? The answer might surprise you.
In fact, impulse response, or an audio system's response to an impulse signal, is one of the most useful and accurate representations in existence of such a system's linearity and precision - or its fidelity to an original signal that is fed to it. A lot of focus has been placed on the pre and post ringing of these "discontinuity signals" but what you have to understand is that the ripple artifacts are nothing more than an analog system's (all electronics is analog -digital is just a special subset of analog) limitation in attempting to construct the impulse or discontinuity signal waveform. They are a result of the impact produced by the energy storage devices themselves in creating the signal. To create a large energy peak, you need large storage devices. The larget the capacitor for example, the longer in time it takes for it to absorb and discharge electric field energy. This is the same with inductors. One type stores electric field energy - the other magnetic. Smaller value capacitors can react to voltage changes very quickly but are limited in the peak value of energy that can be stored and dissipated. But if you combine a large number of high value and low value devices in a circuit and apply a voltage spike, you wind up with the kind of oscillations you see in an impulse response graph. Small capacitors for example, rapidly reach their charge capacity and can discharge into larger capacitors that are much more slowly building up charge in the transition from no input voltage to full spike value. This "sloshing around", if you will, or oscillation is what happens in circuits built to provide extreme voltage attenuations. In a linear, time invariant system, any rapid change in frequency response or time response - has these characteristics.
So effectively the entire debate about ringing in digital audio is a misnomer - a hoax. The impulse response ripple is not something that happens in real world sounds or in a properly designed audio reproduction chain. Ever since digital oversampling was developed in consumer products in the early 1980s, there has been no need for steep analog filter circuits with their attendant ringing. The problem very simply DOES NOT EXIST. The ringing generated artificially in an impulse signal is useful in that it provides a very high frequency stimulus to linear audio systems as a means of measuring high frequency and transient response. IT IN NO WAY BY ITSELF, REPRESENTS THE TIME DOMAIN BEHAVIOR OF THE AUDIO REPRODUCTION CHAIN. An accurate audio reproduction system should fully render the impulse signal in all its pre and post ring glory without alteration. Any audio system that eliminates or significantly alters this pre/post ringing present in the signal that is fed to it is not truly "high fidelity" and is thus bandwidth limited.
Showing 20 responses by cj1965
In the real world, any pre or post ringing associated with digital pulses in pulse code and delta sigma converters is at such a low level and frequency outside the working bandwidth of the circuit as to be of no consequence whatsoever. The comparators, op amps and output filters used are not sensitive in any meaningful way to any ringing unless you are feeding the circuit a Dirac Delta type or Impulse response signal. In that case, yes - low level pre and post ringing could be high enough in amplitude and low enough in frequency to register in a typical converter output. In the real world, however, no such signal containing extreme bandwidth and amplitude will ever be encountered with normal signals. Again, the issue of pre and post ringing is a red herring - a hoax propagated to make a bogus sales pitch for a solution in search of a problem. The manner in which normal signals are digitally deconstructed and reconstructed prevents low level switching noise in digital circuits from becoming audible. |
@geoffkait The ancient Benz wagon I've had for many years has a six cd changer in it. I has traveled over some of the roughest roads in New England and has never missed a beat. Perhaps I should drive it out to California and wait for an earthquake. Maybe then I'll witness this "stubborn vulnerability , especially to seismic vibration" that you speak of..As for the litany of other ills you mentioned, I guess I'm lucky to have never experienced any of those problems. I must be one very lucky guy never to have seen evidence of pre or post ringing originating from digital audio circuitry or any of the maladies you speak of....But for you, my friend, there's always vinyl ... |
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I think MQA will just give the government a cut of the profits...
" - jl35 If you want to know who is being paid off, look for the loudest supporters of this "new innovative technology" which features claims of "improvement" that cannot be tested for or measured. You will also find that these loud supporters are the quietest among the crowd of "faithful" when it comes to discussing the proprietary and security control aspects which represent the downside for consumers. |
"
More propaganda from a truther. Why do you need instruments when you have 2 ears to hear if it sounds good or not.
" - rbstehno Typical misinformed "audiophile" response to any objective fact finding. It's one thing for a manufacturer to buddy up with a few audiophile press "journalists" - offering practically free equipment, tickets to major sporting events, or just simply buying large amounts of advertising from their "honest, trustworthy, and principled" employer in order to get favorable, totally subjective "press" to help move product. The potential negative impact to consumers who might wind up buying a totally over hyped product is going to be limited. You simply can't fool everyone all of the time. MQA, however, represents a different thing entirely. IT IS A PROPOSED STANDARD FOR AN ENTIRE INDUSTRY. So the same old tired stupid behavior that might be tolerated regarding audiophiles and their complete adherence to the value of subjective criteria over objective, provable criteria won't cut it. Any time an industry standard is proposed, it is appropriate and actually ENTIRELY NECESSARY for objective criteria to be provided that establishes concrete, measurable characteristics with which "qualifying products" can be measured against.The problems with MQA is that it doesn't provided any concrete characteristics or criteria that can be measured to allow anyone to prove that a particular quality standard is being met. THE ENTIRE THING IS SHROUDED IN MYSTERY, DOUBLE SPEAK, AND THE USUAL "VEILS HAVE BEEN LIFTED" CLAIMS you sometimes get with manufacturers pushing new products. And the justification for all the secrecy surrounding what it does and how it does it boils down to intellectual property claims. THIS IS WHAT PARTICULARLY STINKS ABOUT MQA. THE ONLY NOVEL OR PATENT WORTHY ASPECT TO IT PERTAINS TO THE SECURITY/ENCRYPTION FEATURES BUILT IN - AND THAT ASPECT WASN'T EVEN CREATED BY STUART ET AL., ITS WAS SUBBED OUT TO ANOTHER COMPANY! |
@ Cleeds "More propaganda from a truther" Did you say insults? If people think they can contribute to a thread like this with ignorant insults and not get them thrown back in their face, they're delusional. It's' not a personal attack to acknowledge a basic simple fact: A high percentage of audiophiles completely disregard the value of ANY objective criteria when it comes to the sound reproduction industry. That is simply not acceptable or even rational. For those who are so inclined, there are plenty of "vote for or against" threads in which to participate. This is not one of them. With regard to the use of CAPS. Unfortunately, a lot of people don't read or attempt to reasonably comprehend others comments before putting forth stupid, misinformed attacks or snarky comments to help boost their fragile egos. CAPS are sometimes appropriate to get some individuals to actually pay attention to SALIENT FACTS. Best Regards |
@tomcy6 I'm very calm. Use of capitals doesn't mean I'm not very calm. I'm not sure how to highlight or boldface anything for emphasis using this interface. Unlike most of the participants here, I don't bother posting to such forums very often. I just don't have the time to amass 5 or 6 thousands posts on an audio forum so I'm a little green when it comes to finding features like emojis or text editing buttons. As for anger or hostility towards others - if someone pops into this or any other thread with an insulting comment like: you're a "truther" spreading propaganda - without anything to back up this statement or factually contradict anything presented in the thread, they will be responded to in kind (see above comment from rbstehno). Essentially all of that users comment was personal attack and nothing was focused on the actual information presented in this thread which centers around ignorance of what a Dirac Delta or Impulse function really is. Whether or not you or others have thin skin and don't like it when someone points out the complete lack of knowledge or understanding on the part of members talking about "pre ring" - I can't help that. The facts are what they are. Impulse signals are what they are - special signals that have absolutely nothing to do with what audio circuitry deals with on a regular basis. All forums such as this strongly encourage participants to focus on the subject matter and avoid attacking the messengers. Just because you don't like the message, it doesn't give you or rbstehno the right to personally attack another member's character with unsubstantiated "truther" or "propagandist" labels. |
"
Your point that preringing and postringing do not exist is not generally
accepted as fact among the people who design and build audio
equipment. If you want to present that opinion to us, fine, but to say
it's fact and that anyone who doesn't agree is ignorant is not helpful
to your cause. I'm just suggesting that you lower the intensity of your
posts a little. We don't need an MQA war here. As I said, there are
forums where such a war would be welcome." - tomcy6 Please cite some examples of "people who design and build audio equipment" who claim that music reproduction involves "pre and post ring" signals. I use impulse tests on a daily basis. I would like to see evidence of ultra broad band high intensity, ultra short duration peaks in music or any other signal source to back up what you appear to be saying. If you read the thread carefully, you will find that no one is saying pre and post impulse ripple doesn't exist. What you will find if you actually read the beginning of the thread is that these phenomenon only exist in circuits that utilize extreme attenuation of signals. Filter theory is very old. There are no special new filters that violate the laws of physics. Every steep filter produces oscillations that can easily be calculated for mathematically in the transfer function and represented in a Bode plot displaying stability or lack thereof. The problem with the entire foolish MQA promotional exercise is that a handful of "Industry people" are using impulse response data to infer that such signals are routinely present in music and their special characteristics (pre/post ripple) must somehow be dealt with. This is a complete farce that exposes widespread ignorance in the "audiophile" community - or at least that subset of the community that embraces ridiculous postulations about impulse signals that aren't true for the majority of music data fed to digital circuits. Again, if you know anyone in the industry who can demonstrate that pre and post impulse ripple is encountered in music playback, I for one would like to see evidence of it. The anecdotal " I know some people" isn't sufficient. Evidence is required. If you want further citations for what an impulse signal is, I can provide links and you can actually generate them yourself using readily available software and a modest equipment setup. You can contrast the signals you produce with that of sharp pulses produced by a square wave generator - another signal type that doesn't exist in the real world of music but is used for equipment testing nonetheless. There's an old saying about "a little information" being dangerous. That is precisely what is going on with MQA and signal ringing. The folks that are being misguided know just enough about the subject matter with which to make fools of themselves. Take the time to at least generate some impulse response signals yourself before you go on an internet forum to "educate" others about it. I have done it. You can too. |
"
Charles Hansen of Ayre Acoustics and many other audio manufacturers use
Minimum Phase filters in their digital components. It is my
understanding that the purpose of these filters is to eliminate
preringing. I know very little about electronics so
I may be wrong about that. With that, I'm out of this discussion.
" - tomcy6 Ok. Let's be clear about what minimum phase filters are. By definition, a minimum phase filter is a filter that remains causal and stable when used in a linear, time invariant system. All that really means is that phase and gain changes cannot result in oscillation (unstable). For this to occur, the poles and zeros of the transfer function plot have to stay inside the unit circle. A common example of violation of this characteristic happens when an amplifier's overall phase shift is 180 degrees at unity gain. Under those circumstances, a typical feedback amplifier will oscillate. In terms of filters, it really has nothing to do with impulse response. As noted previously, to generate a very rapidly increasing and rapidly decreasing voltage spike in a linear, time invariant system - requires and extremely sharp (steep) filter. That's where the ringing (instability) comes in. A minimum phase filter in Charlie's case is simply a gradual rolloff (low phase shift with frequency) filter that is causal and stable - minimum phase. We are talking apples and oranges here. There is absolutely nothing whatsoever about an impulse signal that is "minimum phase". |
@ clearthink The problem with your suggestion is that it is based on a faulty premise. The Audiophile press's push to back MQA is "common knowledge" as a quick 5 minute Google search will reveal. The problem with the MQA pre ring post ring hoax is that a "common misconception" nurtured by a handful of industry people exists with respect to the pertinence of pre ring/post ring phenomena in digital audio circuits. There aren't enough people exposing the common misconception promoted by Bob Stuart in forums like this to counter the smoke screen. However, the "common knowledge" of MQA promotion in the audiophile press - that's easy to find. You don't have to corner an electrical engineer or delve into signal mathematics on wikipedia to figure that out. BIG DIFFERENCE, my friend. Cheers |
"
The way this expertise is simply thrown into the wind in these
discussions, flooded by arguments that are, put in diplomatical words,
two or three floors below the level set by Craven and Stuarts, makes me
cringe! " - pegasus Really? Two or three floors below level set by Craven? TRANSLATION: Just another useless "audiophile" comment aimed at attacking the messenger's credibility without any factual or objective basis whatsoever. This thread is very straight forward and simple - Craven et al. are using a phony argument about impulse response ripple to try to insinuate that such a phenomenon is present in everyday digital sound recordings. It is very clear from the Stereophile impulse response graphs that MQA is doing nothing more than adding dither noise to hide the pre and post ripple associated with the impulse input signal. Additionally, the "origami fold, unfold, deblurr " BS does nothing but add phase delay (distortion) to the primary impulse peak (see negative going pulse just after MQA enabled DAC response that doesn't exist in the non MQA Brooklyn DAC response). If you have anything to say about the technical facts presented here, please direct your comments to those facts - possibly citing some facts of your own. Otherwise, spare us the "Mr. Craven et al are several levels more brilliant than anyone who is participating here in this thread". Your unsubstantiated insults are not welcome. Play the ball - not the man. As for critiques of the original Sony/Philips PCM approach with steep cut off filters 35 years ago - no duh. It was clearly pointed out at the beginning of this thread that oversampling solved the "ringing problem" in digital audio before many of the readers who come here were even born. And no, Mr. Craven's "appetizing" filter (pun intended) doesn't resolve the distortion problems created in those early recordings. There is no need for any of Mr. Craven's security encryption schemes disguised as sonic improvements. The only potential need in the industry that exists is to take the current lossless standard and make it more efficient - some scheme to detect the dynamic envelope of every file that is to be streamed and apply only the bit depth necessary to transmit the particular file. It's a very simple concept but because it doesn't involve "protecting the family jewels" and dramatically increasing profit, no one in the recording industry is bothering. |
"
Of which my main point: I wonder why no real, total AD/DA loop measurements are shown anywhere. The other being the measurement with "correct impulse responses", ie. measuring a DAC not only a not existing, abstract sequence of (one) sample. " - pegasus The above statement clearly demonstrates that you don't yet understand what an impulse response test really is. The folks at MQA have been banking on this problem to assist them with the smoke screen. Again, read the beginning of this thread. For emphasis ( I don't know how to use bold type on this interface) IN ITS TOTALITY, AN IMPULSE RESPONSE IS THE FULL CHARACTERIZATION OF THE TIME AND FREQUENCY DOMAIN BEHAVIOR OF ANY LINEAR, TIME INVARIANT SYSTEM UNDER TEST. Please read the above over in your head several times. If there is any term contained therein that is unclear or confusing, please let me know and I will do my best to try explain it to you. Audio systems are considered by most engineers who build them to be "linear, time invariant" systems - or at least - that is the goal. The impulse response plot posted by Stereophile of the MQA and non MQA DACs show latency distortion as well as added noise in the MQA file. Whether or not this is audible or audibly pleasing/objectionable to the average listener is and likely always will be a matter of endless debate. What is not in debate is that it IS DISTORTION. Any distortion you want to talk about in these kinds of linear system approximations has its origins in energy storage - whether its a standing wave in a speaker cavity or a simple phase delay in a first order crossover network. When a signal's voltage and current go out of phase, distortions result and are typically detected in the form of even and odd ordered harmonics. The more rapidly and intensely energy is stored, the more harmonics are produced regardless of the level of damping (resistance/loss) applied between the storage elements. LATENCY = ENERGY STORAGE = DISTORTION. Simple phase delay networks that involve linear phase changes may appear to be "distortion free" but that only depends on the "working bandwidth" or frequencies of interest. In a linear, time invariant system, time and frequency distortions are derived from one another - different representations of the same thing. So your subsequent statement - " Since when is latency a distortion...?It may be a limiting factor for practibility reasons, or a simple inconvenience. But in replay audio it is (AFAIK) of no concern at all. " represents further proof that your knowledge level is lacking. There are plenty of filtering tricks one can apply to reduce undamped oscillation in a circuit. Linkwitz-Riley crossovers come to mind. There is a faint reference to this technique in the original Sound on Sound BS article put out to promote Mr. Craven's "apodizing filters" - essentially cascading buffered linear phase filters to achieve rapid rolloff without some of the deleterious affects of single stage steep crossovers. ( I found no reference to Linkwitz in the original "Craven's a genius" article, btw.). But if you have actual experience with these types of circuits and have done distortion measurements on them, you will find that total harmonic distortion creeps up as the amplitude of the signal drops off in the transition band of the filter - buffered Linkwitz-Riley or not. There is no free lunch. And it looks like others are waking up to the fact that what Stuart and Craven are offering is more like reheated left over meatloaf than a miraculous "free lunch". |
"Read cj1965’s original post several times and not sure I understand or
agree with his central premise -- that "The impulse response ripple is
not something that happens in real world sounds or in a properly
designed audio reproduction chain."
- craigl59 Impulse signals are neither causal nor stable. No amount of filtering can make them causal or stable - "apodizing", "apetizing", "deblurring", minimum phase, linear phase, etc...etc.... Please Google Dirac Delta or Discontinuity Signal and do some reading. You might learn something about what they actually are, what it takes to produce them, and how they fit in the context of sounds that are recorded for playback in music and broadcast. They have special features not present in any other form of "signal content" and only have purpose/usefulness in testing the response of linear systems to stimuli. |
@craigl59 You can choose all of the filtering options you want. I can guarantee your DAC doesn't have any triggering circuits that detect the onset of an impulse or Dirac Delta signal and magically filter out the pre and post ripple of said signal. As with the MQA garbage, simply raising the noise floor by adding dither eliminates that ripple from the impulse response graph. The DAC isn't "filtering" out anything as far as ripple goes. It's called "masking" - just bury the minor noise no one can really here anyway with more noise. And the problem magically disappears from the response graph. If you're going to quote someone, produce the full adequate context of the quote otherwise it just looks like you're trolling. No amount of filtering can make IMPULSE SIGNALS causal and stable. You left out (I'm guessing intentionally for trolling purposes) the primary subject of the misquoted sentence that happens to be the primary subject of the thread. |
Pegasus said: ' However... a few points about MQA are IMO brillant: - the "information density" in the range above 22kHz is *way* below that in the midrange or audio range. To double, quadruple or "octuple" (;-) the sampling rate for *objecively* (measured and sampled) very small amounts of information is not elegant. It is in a certain way an idiocy.Thinking about how to "underfeed" this information into normally sampled digital files is a brillant idea (IMO). " Actually, no. It's not brilliant. The brute horsepower behind digital audio has always lied in three distinct areas: 1) the precision of high speed switching circuits that affords greater bandwidth and linearity 2) the low noise that is possible with high bandwidth low voltage logic signaling 3) the accuracy (repeatability) of a high resolution (precision) discrete time and discrete amplitude system Your comment above demonstrates a complete lack of understanding as to what actually has given digital audio the strengths it has always had over traditional analog approaches. Bandwidth (high sampling rate) for digital audio is an indispensable tool that serves as the foundation for high levels of linearity and accuracy. It essentially represents the point of the spear in the fight to overcome human hearing's ability to detect error. The fact that human hearing is limited to 20khz is what makes digital audio sound good. If we could hear at frequencies above the sampling rate - it would sound like the ones and zero trash that it truly is. Without a sampling rate well beyond human hearing, it would be impossible to create digital audio that appears to us to be completely linear and accurate. If there is anything that can and should be sacrificed in terms of improving efficiency of the standard - it is at the amplitude precision end. There has never been a need for playback dynamic range to far exceed the threshold for pain and rapid hearing damage/loss. Ask any physician and they will tell you - 145db is insane. I've heard a lot of stupid arguments saying we need well over 100db in dynamic range. In my experience however, even very elaborate well constructed audio systems struggle to produce full bandwidth dynamic peaks in excess of 120 db. In the real world that means at 120 db, sound you hear is about 80db above what is barely detectable in a completely silent room. Does anyone in this forum think they will be able to detect someone whispering right next to them if blindfolded and listening to music blaring at 120 db? This is just one example of how impractical the desire for 24 bit resolution really is. |
"
In principle keeping the whole bandwith but coding it more efficiently into the "data container" *is* an intelligent idea
" - pegasus Ok, at least we can agree on that basic principle. The reason preserving high bandwidth (sampling rates) is more critical has always seemed obvious to me - since steep filters operating below Nyquist were creating problems for achieving reliable sound quality. This was known as far back as the early 1980s and was the primary reason my first CD player 34 years or so ago was the first generation Philips 4X oversampling unit. The laws of physics governing filter stability and distortion still haven't changed since those days of the first space shuttle flights. It is much easier to avoid signal degradation with a gradual roll off filter that effectively wipes out the signal well before the Nyquist frequency is reached. You don't need to employ multiple stage linear phase filters and the end result has been universally praised as being "superb" for the most part. On the amplitude precision side, I have never heard a cogent argument for dynamic range that significantly exceeded the original format - 16 bits. |
@ejr1953 It's hard to say exactly what is causing perceived "glare" by some vinyl fans with respect to digital. In the early days of digital as we've acknowledged above, steep filters were used to accommodate the sampling rate that was marginally above the frequency limit of human hearing. These filters were vulnerable to component tolerance changes over time - an even greater source of potential sound quality problems beyond the large phase shifts they introduced. With the advent of widespread oversampling in the industry - that problem essentially disappeared. But the underlying "improvements" of digital technology I believe may be more the cause of the alleged "glare" some complain about. By virtue of its extremely high precision, bandwidth, and linearity capabilities, digital audio has the ability to accurately render extreme high and extreme low frequency source material like never before. Vinyl encoding - although pretty wide bandwidth, never could provide the same dynamic range -especially at the frequency extremes. The signal had to be compressed to keep distortion generated at the stylus from skyrocketing - particularly at high frequencies. There were a host of other problems that CD technology ameliorated like the high frequency loss created when the tone arm approached the center of the vinyl album - due to a substantial reduction in effective stylus-over-groove speed. Baked in tonearm tracking error was another problem fixed. CD's went way beyond what most perceive to be the primary advantage - no contact laser light eliminating wear and tear degradation altogether. If you want to learn more about the myriad of headaches and limitations of vinyl, you can read about them here: https://www.emusician.com/how-to/mastering-vinyl The bottom line to my theory about perceived differences is that when you grow up listening to a technology that has all these limitations built in, when they are suddenly removed, the new changes (full capacity to render all dynamic high frequency content without measurable distortion) can be unsettling or "unwelcome". We tend to be creatures of habit that like what we're used to. Compounding this problem in the early days was that recording industry techniques were well established - you might even say entrenched. Added high frequency bias built into the recording approach could easily appear "hyped" in the new technology format. So it was important for recording engineers to find a new balance with the new technology and not stick with the same old mic /mixing techniques that worked before. This clearly didn't happen in all cases. |
Thanks Steve for your input. I am a little confused by what you meant to say here, though: "Pre-ringing is certainly unnatural and contrived. Whether it is audible depends on the amplitude and the quality of the system IME. Post-ringing is natural and expected if a system is not critically damped. Certainly better to minimize it though." My understanding with an ideal Dirac Delta signal approximation is that the harmonics prior to the signal peak have the same spectral content as those appearing post peak - demonstrating symmetry. If pre pulse harmonics are missing from the response via either filtering or the addition of masking noise, then the post peak ripple or harmonics should likewise be missing - leaving only the expected "natural" ringing from the device under test. I'm assuming that was what you were saying above. Thanks again for your input. Best Regards cj |