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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
ptss
1,964 posts
 
@pegasus. I find your comment “Contrary to many here on this thread I doubt that the ‘lost bits’ below bit 18 or 19 and below the already nicely sampled noise from the recording chain are audible at all.” to reflect many current attitudes. But, it misses entirely the point under discussion in this thread and others related to MQA ; which is that MQA claims to yield a recording “equal to” the Master Tape. That “Master Quality Authentication” claim has been proven to be a lie. That’s a “fact” as opposed to your “doubt”. 
As far as audibility of differing bit lengths; today, this is very system dependant, but 50 to 100 years from now it may not be as improvement in playback equipment will yield hopefully more transparency to the recorded music. Now we’re back to the thrust of the criticism of MQA. It simply does not provide “more transparency”. 
Post removed 
cj1965 OP
124 posts
 
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.
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.

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