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.
Good to see people coming around to the dog and pony show

As an engineer I have been very concerned about a world of MQA audio, marketed as equal to or better than the master.

It's not.  Equal.  Or Better.   At all.  Period.

Making a playback system is a fine form of creativity for the audiophile, and preference for mp3s over 24 bits has happened in blind tests.  Preference is not important with MQA, all that matters is the integrity of the master.
With all the techno babble being pushed to and fro regarding MQA, the bottom line is that  MQA uses a lossy compression system and once lost ...... 'it ain't coming back'! Think about it! Look what MP3 has done to sound quality. Unless you can provide an audio carrier system that can retrieve musical information in it's entirety 'without loss', we will always be bombarded with claims of a fix. MQA is another form of resolution strangulation. ..... 'We'll do it in a nice way and it won't hurt a bit!'   
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.
No idea what this fellow is going on about...but I do have a lot of recording studio as a musician experience. I have A/B’d MQA vs non MQA countless times on my two high end audiophile systems at home. Any MQA version only sounds to my ears as a slightly different studio mix from the previously available version. It never sounds better ... or worse.
" 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?

 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.