My only question to anybody,
How would HQ player relate to any of this?
Kenny.
How would HQ player relate to any of this?
Kenny.
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.
67 responses Add your response
In a normal digital or analog signal, there is no reason to have analog ringing prior to the signal making a transition. Ringing always occurs after the signal transition and only if the signal is not critically damped by some means, such as impedance matching. The energy of signal transition is what causes the ringing. That is what I meant by natural. Steve N. Empirical Audio |
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 |
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. I have heard several DAC’s at shows that were touting their apodizing filters, which virtually eliminate pre-ringing, but always at the expense of adding higher amplitude post-ringing. Never liked the sound of any of them. I believe I can hear the post-ringing. The more important aspect of impulse and step response of a DAC IMO is whether the impulse actually achieves the maximum amplitude or not. This is my problem with Jon Atkinsons impulse measurements in Stereophile reviews. The impulse plot never shows the amplitude scale. I suspect that the power subsystems of most DACs don’t allow the impulse to get to full amplitude. The Overdrive SE and SX DACs do. The power subsystem is instrumental in achieving this and this is what sets some DACs apart from others. Here are some plots of the Overdrive SE. The SX is even better. http://www.empiricalaudio.com/images/products/overdrive/Graph_OutputBalXLR_ImpulseResponse96k.png The 96K plot is with the digital filter set for 192. I can manually select them on the Overdrive DAC. http://www.empiricalaudio.com/images/products/overdrive/Graph_OutputBalXLR_ImpulseResponse192k.png Steve N. Empirical Audio |
@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. |
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? |
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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. |
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 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. |
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@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”. |
@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. |
"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. " And in return, cj1965, suggest you contact the engineers at RME and tell them that they cannot control filtering on their DACs, produce response curves as shown on page 55 of their ADI-2 DAC manual, and offer the kind of real-world audio responses that allows the listener to choose between filtering options. |
"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. |
"
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." Regardless, the new RME ADI-2 DAC allows you to select between 5 different Filtering settings that change DA Impulse Responses. These range from traditional oversampling ringing to NOS which is essentially perfect as regards impulse response (no ringing at all). The 5 alternates sound audibly different and can be chosen to improve the musical genre: e.g., the "Slow" option opens up orchestral textures and allows more "breathing" room to make the texture more realistic. The NOS option is extremely accurate, almost painfully clear, and not to my taste at all for longer listening sessions. It does provide a shockingly realistic sound for popular recordings. BTW your system must be audiophile quality to distinguish between the various options. Try it and see -- and, oh, it will be hard to find a ADI-2 DAC because they are new and so popular. But the same facility is available in the RME ADI-Pro 2 which has been available for several years. Perhaps other DACs on the market have this option and let me know if you know of others. |
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. |
"
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. |
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. |
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. |
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!' |
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. |
cj1965"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..." I understand completely there is one level of proof required for one thing and another level of proofs required for another and in BOTH cases you don't provide the proof but demand others do it for you and insult those who question your motives which you still have not revealed so yes I can see why you say faulty premise! |
@ 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 |
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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". |
cj1965"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..." hey cj1965 here’s you’re very answer to someone else who asked for some proof. Do you recognize this??? "cj1965Oh puhleeeeeze. Spend five minutes of your time Googling...Nobody is going to waste time giving you proof of the obvious. If you’re that lazy...don’t expect any of us to serve it up to you on a silver platter. So it is only fair to now ask you to Google for yourself! |
Charles Hansen of Ayre Acoustics (now deceased, R.I.P.) 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.
Here is what Charles described as one of his accomplishments in audio design: "World’s first disc players to provide user-selectable “Minimum Phase” digital filter responses, including both “slow roll-off” algorithm with improved transient response and “apodizing” algorithm for removal of ringing from digital filters used to produce the disc." Read more at https://www.audiostream.com/content/qa-charles-hansen-ayre-acoustics#8mhJtBVPIS8iPIDQ.99 |
As posted above, there has lots of positive and negative comments on
Audiogon about MQA coded albums. Of
course, everyone is entitled to their own opinion that I enjoy reading. I listen to Tidal MQA coded albums since I
feel they sound slightly better (sometimes) than regular albums. The music seems clearer and with less background noise. I do NOT LIKE the Audio Manufacturers having to pay for the MQA license and noticed that some support this extra cost and many others do not. In addition, I know that some people “believe that supporting MQA means handing over the entire recording industry to an external standards organization”. If this statement is true (is it?), it is also a concern to me. I own the Aurender N10 Music Server (it decodes the first MQA layer) and has 10,000+ MQA Coded albums from Tidal that I enjoy. However, I would NOT PAY extra for MQA coded albums. I also know that some Manufacturers want nothing to do with the MQA coding process. I continue to believe we have too many audio formats that make it very confusing for people new to audio and maybe everyone else (such as who does the unfolding of the different MQA layers?). Some DAC's support MQA and many others do not and will NOT. There is much confusion, concern and debate around MQA coded albums, so I am waiting to see what happens. |
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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. |
OP You made some very good salient and interesting points that had me reading deeper. BUT( sorry could not resist) The use of caps to "shout" and your general "take no prisoners" attitude will not win you any favours here. Members will unfortunately soon look past the meat of this thread and just concentrate on the flesh (insults, attitude, shouting etc) That will not help your cause of the one of this thread one little bit and that would be a shame. It is hard I know when one feels they are being personally attacked but may I suggest a modicum of patience? |
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 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. |
cj1965, Please calm down. If MQA is a hoax, it will not become the industry standard. So far it is only being used to stream hi-res on Tidal. Tidal always offers the albums it has encoded with MQA in 16/44.1 also. I don’t know how familiar you are with this forum, but we try to keep discussion low key and friendly. There is another audio forum known for angry arguments, insults and hostility. If you prefer to express yourself that way, I think you would find that forum suits that style better. So go ahead and present your arguments against MQA, but without the anger towards people who haven’t come to the same conclusion. |
cj1965 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 delusionalPlease don't threaten contributors here. Please. 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 acceptableIf you find behavior here unacceptable, please alert the moderators rather than threaten contributors here. 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.Please, stop with the insults and the threats. |
@ 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 |
cj1965 m MQA ... IS A PROPOSED STANDARD FOR AN ENTIRE INDUSTRY.Yes, we understand that. Shouting is not necessary. 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.Why so angry? THE ENTIRE THING IS SHROUDED IN MYSTERY, DOUBLE SPEAK, AND THE USUAL "VEILS HAVE BEEN LIFTED" CLAIMS ...I'm beginning to think you don't care for MQA. THIS IS WHAT PARTICULARLY STINKS ABOUT MQA. THE ONLY NOVEL OR PATENT WORTHY ASPECT TO IT PERTAINS TO THE SECURITY/ENCRYPTION FEATURES BUILT IN ...Okay, you don't like MQA. I'm not much interested in it, either. It's just a puzzle that you'll go to such lengths - the ALLCAPS, the insults - to oppose it. It suggests you have as much at stake as those you criticize. |
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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! |
More propaganda from a truther. Why do you need instruments when you have 2 ears to hear if it sounds good or not. This thread is no different than a vinyl addict claiming analog is better while we know the issues with analog. I have good equipment to hear what I think the best formats are: vinyl, MQA, dsd, hi res, redbook. I don’t need somebody that claims to know something to persuade me 1 way or another. |
Cj1965. I think you’re on the money all round. The military build up is scary-and wastes a lot of resources. But recent wars have improved 1st aid knowledge and techniques-so they can argue those benefits.. The major record company investors “partners” must be licking their lips thinking of reissuing everything since Al Jolson- all ‘new n improved’ jon2020-scary! |
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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. |
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