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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.
cj1965
pegasus
116 posts
 
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.
Does it?

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. 
Did I say something different? MQA is in one aspect based on the fact, that 24 Bit is a de facto standard, but leads to an "excess" dynamic range, ie. "not used bits".  If file dimensions or data transfer rates are an issue. IMO they still matter.
In principle keeping the whole bandwith but coding it more efficiently into the "data container" *is* an intelligent idea. If you look at the (peak) musical signals above 20kHz, their level is extremely low, but for every doubling of the sampling freuquency you double the file dimension, for a very small increase in coded information that might be important sonically.
The whole coding into a lower datarate "container"  has nothing to do at all with actaul sampling bandwidth. It's a form of intelligent lossless data compression basically - if the "only" information "thrown away" is below eg. -108 dB o/ 18 Bit resolution, or lower.My doubts creep in is, if 2 or 3 Bits of 16 Bits are thrown out for a doubling of coded bandwidth.
And really critical listening and testing of different sampling rates / audio formats would have to prove that it really is "lossless".

Your continuing furor is amazing. I hope you can apply it to your daily tasks too :-)
I'll leave it at that.
pegasus
116 posts
 
Forgot to mention:
If MQA sounds worse at simlar file sizes (and it seems to have measurable and sonical issues) the whole process is indeed very questionable.Because an idea is only brilliant if proven in practice.

cj1965 OP
124 posts
 
" 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
918 posts
 
Very interesting post and very interesting responses as well, thank you all for that.

Lots of people who have posted here seem to have a LOT of knowledge about this technology.  So, I'd like to ask, could the reason those people who like vinyl and say they hear "glare" from digital be responding to the process used to take the extremely hi-res digital files from the studio, run them thru some sort of dithering algorithm to reduce their size to produce the resulting size being distributed to the public be causing that?
cj1965 OP
124 posts
 
@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.
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