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

Showing 4 responses by pegasus

As we all know, the present times show that every coffee table round is filled with the "real experts". And that the so-called "elite" (be it political, technical or economical) is corrupt and all these are false experts.
Leaving this premise beside:

Reading through previous work of Craven, and some papers of (yes, indeed!) Bob Stuart, everybody could see that both have a very high level of mathematical and engineering skills and training, besides original thinking.
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!

However... a few critical words towards MQA first:
When the Sony PCM recorder was first introduced in the beginning of the 80's, engineer K.L. Breh of "HiFi Stereophonie" measured the Sony PCM machine vs. a tape recoder run at different speeds.It was obvious by looking at the signals after passing the *complete* recording/replay chain, that the PCM recorder had far worse impulse response (then induced by the analogue filtering, the Sony wasn't oversampling digitally).
- I miss such a complete measurement, including frequency response, distortion, impulse response and aliasing artefacts of a complete MQA chain. This would clear up many slightly (at best) "foggy claims" of MQA.
Where are these complete measurements, which are not that complicated to do for a professional reviewer?
- That MQA "messes" with aliasing criteria is something that silently is distilling out of this fog.
- Pretending that there is no aliasing artefact, because there is no information in the frequency range close to the sampling frequency, which would cause aliasing, would make the whole HighRez issue a moot point - if it is, or if it would be true. (To which point I want take a position in this discussion).
If there is no signal that can cause aliasing, the sampling frequency is unnecessarily elevated, no need for HighRez.
- The impulse response that "nicely" shows the FIR filter coefficients of a digital filter (and or the type of used analogue post-DAC filters) generating ringing is an artificial signal, reproduced by playing only half of the recording chain: DA only.
It would look quite different if looped through the complete AD/DA chain.
The ringing seen can't be triggered by a correctly lowpass filtered PCM recorded impulse.It shows - as a semi-abstract picture - what filters are used.

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).
- 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. 20 Bit conversion is all we need for audio. 32 bit resolution is "fake news" or good marketing... :-)

DSD has a lot of fake advantages that are no less marketing driven, like "analogue-like" signal handling (with very high order noise shaping processes necessary not quite true), and has drabacks at least in DSD 64 format, and the other formats are increasingly wasting huge amounts of storage.

The proprietary MQA mastering and decoding process is a thing to critically reflect about but IMO also maybe nice to have – if every other question is openly answered. And if the majority of the claims and promises are kept.Which I will not exclude at this moment.Throwing MQA into the garbage could be a missed opportunity which future generations might be sad about, at least from a quality point of view.
There is no need for any of Mr. Craven's security encryption schemes disguised as sonic improvements. 
I agree (if it would be the only aim of the idea).

Play the ball - not the man.
... and try not to kick balls in a glass house.

Actually I made several technical comments:
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.

Regarding your technical arguments elsewhere: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.
(It is of concern in live electronics, PAs and filtering.)

I can understand that eg. a mastering engineer isn't too enthusiastic about an additional treatment of his files for (at the moment) dubious advantages or dubious advantages in the market place.

I asked MQA for MQA treatment of my recordings and got no answer. :-)

My main doubts when SACD and DVD-A came out, was that
a) the differences between DACs were IME considerably more audible than the difference between the SACD and CD layer on a good quality Sony player. (And I didn't prefer SACD on every aspect). 
b) the user interface of DVD-A was f* up from the beginning.: A sad tale.
The copy right protection of DVD-A was also of Bob Stuart origin...
c) there were some simply amazing PCM recordings on DAD (Classic Recordings), ie. a non copy protected standard format. These sounded better (for me) than any DVD-A (or SACD) I have.
And - funny! - they were transferred from analog tapes. 

I feel (...) that the stereo vs. multichannel, and SACD vs. DVD-A format insecurities helped to move audiophile audio into the back yard of consumer electronics interest.
MQA most probably wan't help either. I'm sorry about that.

And I agree that sonically optimizing CD-replay is much more relevant to 99.9% of the potential public and 99.9% of the recordings, ie. for the music.
And I agree that there are some mind-boggling high quality recordings in the CD-format. So it's the quality of the recording itself that matters most, ie. the microphones, electronics, rooms, setup and mastering.
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.

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.