Dr. AIX

Mark Waldrep, aka Dr. AIX, has been producing and engineering music for over 40 years. He learned electronics as a teenager from his HAM radio father while learning to play the guitar. Mark received the first doctorate in music composition from UCLA in 1986 for a "binaural" electronic music composition. Other advanced degrees include an MS in computer science, an MFA/MA in music, BM in music and a BA in art. As an engineer and producer, Mark has worked on projects for the Rolling Stones, 311, Tool, KISS, Blink 182, Blues Traveler, Britney Spears, the San Francisco Symphony, The Dover Quartet, Willie Nelson, Paul Williams, The Allman Brothers, Bad Company and many more. Dr. Waldrep has been an innovator when it comes to multimedia and music. He created the first enhanced CDs in the 90s, the first DVD-Videos released in the U.S., the first web-connected DVD, the first DVD-Audio title, the first music Blu-ray disc and the first 3D Music Album. Additionally, he launched the first High Definition Music Download site in 2007 called iTrax.com. A frequency speaker at audio events, author of numerous articles, Dr. Waldrep is currently writing a book on the production and reproduction of high-end music called, "High-End Audio: A Practical Guide to Production and Playback". The book should be completed in the fall of 2013.

19 thoughts on “Shoveling Something…But Not Snow!

  • February 14, 2015 at 5:23 pm
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    I have always been mystified by the idea of 1-bit sampling. PCM, at 16 or 24 bit, encodes how loud the signal is at that moment in time. What can 1 bit encode? The package inserts with early SACDs explained that it asked, millions of times a second, “Is it louder than it was last time?”

    If this is the case, the system allows for two answers: “Yes, it’s louder” or “No it’s not.”

    How much louder? What if it’s quieter?

    That explanation is clearly nonsense, and I’ve never figured out how a two-valued sample – which is all that a 1-bit system can encode – can provide anything but square waves, which is probably why there’s so much supersonic noise that the decoder needs to include a “brick wall filter” to protect the power amplifier and speakers (and our ears).

    Can you hear me now?

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    • February 15, 2015 at 9:55 am
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      Phil, using 1-bit samples at very high sample rates does work…in fact, virtually all modern ADC and DACs use Delta Sigma in their designs. The two states essentially monitor the direction of the waveform (with a feedback loop). Because the samples are taken so frequently, there question of amplitude…or how much…is moot. The system simply describes the original waveform. That’s why advocates for DSD often say…it’s almost like the analog wave. DSD is a completely impossible method for producing commercial recordings. Most (85%) of the DSD/SACDs that exist are not native DSD.

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  • February 14, 2015 at 5:51 pm
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    Curious as to your thoughts on this post by Charles Hansen @ Pono:

    “Hello All – FLAC is an efficient way of storing PCM (Pulse-Code Modulation) files. PCM samples the analog signal at a rapid rate, and assigns a value (the higher the number of bits, the more resolution in that value). When played back, it can regenerate the analog signal.

    The higher the resolution, the more closely it can track the value. The sampling rate is a bit more nebulous. Everyone knows that a higher sampling rate will allow higher audio frequencies to be captured.

    The question is, “Why capture frequencies beyond the range of human hearing?” There are two answers:

    1) We are just finding out now THAT THE TRADITIONAL UNDERSTANDING OF HUMAN HEARING IS WOEFULLY INCOMPLETE. It turns out that throughout the animal kingdom that there are QUANTUM-BASED sensors are used extensively.

    This for example is what allows birds to migrate thousands of miles and arrive at their precise destination — in other words, THINGS THAT OUR NORMAL “SCIENCE” CANNOT EXPLAIN. The exact same is true for our hearing.

    Don’t believe anybody that flat-out claims “Humans simply cannot hear this or that.” You all know what you can hear…

    2) The real benefit of high sampling rates is that it gives us much more flexibility in designing the filters required for digital audio. Traditional filters that have been used in ALL digital audio equipment “smear” the time information.

    The human ear/brain is EXQUISITELY sensitive to time information, as this has what has allowed us to survive. Identifying the location of either prey or predator by sound alone is key. Humans can differentiate time delays as small as 1 to 10 micro-seconds.

    Normal digital filters completely smear this time information, destroying the naturalness of the sound. Ayre has worked for years developing digital filters that preserve as much time information as possible. These went straight into the PonoPlayer.

    ~~~~~~~~

    DSD is not an engineering term. It is a marketing term invented by Sony. So it means whatever Sony wants it to mean. They have defined it as a 1-bit digital signal recorded at 64x the CD sampling rate (later expanded to include 128x CD also).

    DSD uniformly sounds very good because there is NO FILTERING on the record side, and very little filtering on the playback side. The difficulty with DSD is that it requires the recording and mastering studios to purchase expensive new equipment and learn new ways of using it.

    ~~~~~~~~
    With the PonoPlayer, the advanced digital filters used allow for OUTSTANDING sound from PCM, even at the single sample rate. Double sample rates are better and quad is the best (virtually no time smear).”

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    • February 15, 2015 at 10:02 am
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      Charles get most of this right and I agree with his sentiments on ultrasonics. However, he’s not completely on the mark when talking about DSD. His comment, “The difficulty with DSD is that it requires the recording and mastering studios to purchase expensive new equipment and learn new ways of using it,” makes not sense. The problem with DSD is no matter what equipment professional studios acquire, there are no tools that allow one to work natively with the format.

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    • February 15, 2015 at 12:28 pm
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      DSD uniformly sounds very good because it has got much better impulse response as sampling rate hugely increased

      Actually, until quite recently, it was still unknown that oversampling recreates impulse response after the event {recording}. Thus far, any fully digital CD track is potentially a much more natural sound than master tape, LP, etc., and Infinite Oversampling should purport the Ultimate Sound Quality. Hence, it is insane to record at higher than 44.1 kHz as it would always mean less quality increase than that of software oversampler.

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      • February 15, 2015 at 12:42 pm
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        Jay…I’m going to stick with other professionals and keep recording at higher sample rates…call me insane.

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  • February 14, 2015 at 5:53 pm
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    I keep wondering if DSD is really just a delta modulator. I built one as a senior project in 1979, and although mine worked to a certain extent, the voltage comparators of the day didn’t allow for a very high clock speed (I don’t remember what mine maxed out at, but I did pass recognizable voice).

    A delta modulator is basically an up/down counter with its output going to a DAC, and that output going to one input of a comparator with the other input being the signal being digitized. There’s only one bit doing the controlling, telling the counter to go up or down based on how the input and output compare. It seems like this is a similar scheme, but I haven’t been able to dig up anything that tells me how DSD compares to a delta modulator.

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    • February 15, 2015 at 10:04 am
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      It is Pulse Width Modulation, a delta sigma modulator spewing out a stream of digital ones and zeros that can accurately represent an analog waveform. The problem comes when you try to do anything with the native stream.

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  • February 14, 2015 at 6:46 pm
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    I’m huddled up inside this evening in Boston, while we are getting yet ANOTHER 12-15 inches of snow. Because apparently the first 3 storms weren’t enough. MAKE IT STOP!!

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  • February 14, 2015 at 8:34 pm
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    I wonder how much more rediculous the lies will become over the next 10 or 15 years as the snake oil peddlers look for ever new make believe technology to sell us recordings that sound little improved over what my beloved 60-80s rock does on the CDs I bought 30 some years ago. MFSL did some good sounding remasters back in the day, but today’s remasters really are only super compressed horrible sounding loudness war releases, or some snake oil BS tech who’s only reason for existence is to get deep in your pockets selling you some greatly overpriced POS.
    Mark here and a few others due some incredible sounding recordings that are well worth the price if you can find anything in their catalogs that meets your musical taste.

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  • February 14, 2015 at 8:44 pm
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    Mark, It never fails to amaze me how damaged my brain must be that I actually spent 60 years living in the frozen hell known as Chicago. The last 5 years here in Central Florida have been like paradise. Tonights NASCAR 75 in Daytona was a gas. WOOT

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  • February 15, 2015 at 7:33 am
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    I lived in Southern California for 10 years. I actually ended up getting sick of all the sunshine and really missed the change of seasons. Ended up moving to central Ohio and never looked back. LA was always this bizarrely surreal place where nothing ever seemed quite as it was. Probably explains why all the big studios and record labels are there! 😉

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    • February 15, 2015 at 10:05 am
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      Sick of all the sunshine? Not me. I live near the ocean and love the foggy morning…knowing it will clear off later. I can appreciate the beauty of the seasons but I choose to visit the seasons not experience them everyday.

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  • February 15, 2015 at 9:33 am
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    A.S. ‘On July 21st, 2012, at 8.20 a.m., the city of Jask {Farsia} experienced an unprecedented heat in known history: +34°C air temperature at 100% relative humidity .’

    “why continue to suffer through terrible winters when you can move to a city with year round great weather?”

    The warmest place on Earth is Pulo Anna Island {Palau} .

    Now, to the point:

    * 2048+ Oversampling should be applied during recording at 44.1 kHz to completely avoid anti-alias filtering

    * Extreme Oversampling can be applied to a properly noise-shape-dithered recording during playback in order to not only exclude anti-alias filtering again but make the sound incomparably more natural than any analog source could ever deliver

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  • February 15, 2015 at 4:36 pm
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    Seems that Paul of PS Audio has even one-upped Sharp in his forever novelistic techno-blurb marketing.

    Writing about his new DSD DAC: ” talk about providing multiple versions of the Pikes operating system for DirectStream. Good conversations. Stimulating talk. The idea centers around tuning the OS to each person’s system. Create multiple OS releases, some warmer, some cooler, others with stronger top ends, still others with better bass. It’s an enticing idea but one with a number of problems.¨

    Obviously no concept whatever of high fidelity – just personal tweaking with factory incentive, but OMG, of the OPERATING SYSTEM of a DAC !?! This is just so hilarious if not sad – but reading Paul always makes me re-quote Billy Joel on Pianoman:

    “Paul is a real estate novelist” => “Paul is an audiophile noveiist” , though he certainly seems to have the knack of capturing clients money in exchange for snake oil… meanwhile writing science fiction all round.

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    • February 16, 2015 at 9:17 am
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      Oh boy…tweaking the operating system? Hard to imagine how far some will go to pitch their products. I’ve seen articles on different sounding “NAS”.

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  • February 17, 2015 at 8:29 am
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    Speaking of shoveling, off topic: what do you think about this from Mr. Hansen at Pono:

    “Cardas is my personal favorite cable. Just be aware (that like the PonoPlayer) Cardas cables take a good week (or two!) of playing on repeat to break-in.”

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    • February 17, 2015 at 9:39 am
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      I’ve never bought in to the “break-in” thinking for equipment and/or cables.

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      • February 17, 2015 at 2:13 pm
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        Hooray! As an audiophile, I have to bite my lip while fellow-audiophiles go on about cable break-in (and capacitor break-in, preamp break-in, DAC break-in) incessantly.

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