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.

12 thoughts on “QoBuz and the Hi-Res Audio Logo

  • Chris B

    For what it’s worth, I believe that QoBuz have some kind of arrangement with Sony these days: when I received one of the new (Hi-Res) Walkmans at the end of October, it came with a fairly generous QoBuz voucher.

    Nevertheless, that logo (which is in itself a great piece of graphic design, IMHO) does look out of place on the above image—too large, for one thing.

    The end of your post makes me think if “Open Source” recordings could ever catch on: the opportunity for the listener or radio station to set their own reverb and EQ !?

    • Admin

      Thanks Chris…coupons and bundles abound in the world of high-resolution downloads. I’m pushing the idea to some of the very same manufacturers…except my stuff is real HD. I love the idea of metadata in files to modify them on the fly. We’ll see.

  • I was just wondering why any orchestral music sounded so..ghm..extremely displeasing to the ears, even at 192 kHz sample rate.

    Here comes the answer: this music type {unlike the best one, typically referred to as ‘avant-garde’} is very sensitive to system’s impulse response so that it must be recorded at the highest possible sampling rate o otherwise it will never sound correct !

    That also implies that although music can be recorded at 768 kHz sample rate to avoid filtering, yet prior to getting into the loudspeakers it must be astronomically upsampled by DAC’s reconstruction filter to provide decent impulse response.

    • Dave Griffin

      Jay wrote: “it must be astronomically upsampled by DAC’s reconstruction filter to provide decent impulse response”
      Maybe Jay’s using Google translate, just guessing here…

    • Nik Razis

      Last time I’m sucked in in replying to Jay. The reason orchestral music very often doesn’t sound right is usually the capabilities of one’s speakers in tandem with one’s amplification power, combined, of course, with one’s listening room which, more often than not, does not resemble a concert hall…

    • My comment – which actually was the first one on todays subject – has now been tugged in between a couple of other postings.
      What wanted to say is, that I see this use (or misuse) of the logo as the first example of labelling anything, that is markered and sold as higres with a ‘trustworthy’ tag – no matter what the provenance might be.
      It is simply easier to draw attention, if you have a fancy logo/sticker on your website.
      I am sure we will see more of this in the future.

  • Andrea

    Isn’t the file at a 48 kHz sampling frequency? So, why bother doing any analysis? We know it has a steep antialiasing filter before 24 kHz, so can’t meet the JAS definition.

    As for dynamic range, a distantly miked recording is never going to measure as well as your close miking techniques. Like most classical music lovers, I’ll trade the better numbers for a blending of sound and sense of space more akin to what I hear in the concert hall.

    • I’m not like most classical music lovers, I guess. Listening to a live performance from the best seat in the house can be utterly captivating but it pales to standing next to the conductor. When I played the Pine of Rome for the music director/conductor of the NJSO years ago in full immersive 5.1, he absolutely flipped out. He told me, “finally, finally a recording that sounds like real music!” and the faculty of the Colburn School for Performing Arts here in Los Angeles (the “Julliard” of the west coast) insisted on coming to the studio after one of their members told them about my recordings of chamber music. They were equally impressed.

      I don’t think many classical music lovers would switch back to a traditional stereo CD after experiencing an immersive 5.1 surround recording in real HD. There certainly haven’t been any hold outs in my experience. But to each his or her own…I also include stereo mixes on my discs.

      • Dan Lavry: ‘DA’s do not generate narrow pulses (RZ signals – return to zero), because there is not much energy in narrow pulses. Instead, the DA output signal (before the analog anti imaging filter) is a NRZ signal (NRZ – not return to zero). The signal looks like “steps” (not like narrow pulses), and such signal does contain the needed energy to drive the analog filter. However, when you use NRZ signals without up-sampling, you lose some high frequencies. The flatness response is compromised.

        You can see a plot in my web site, in a paper “Sampling, Oversampling, Imaging and Aliasing”.

        Look at page 3 for a plot titled Sin(X)/X plots for X2,X4,X8 and X16.


        You can see that at X2 up-sampling you still lose around .8dB at 20KH. With X4 up-sampling, the loss is around .2dB at 22KHz. At X8 it is less then .1dB, at X16 the loss in a non issue…

        The horizontal axis is frequency (0-320KHz). Your interest is mostly 0-22KHz (audible range), shown by the red line marked “22”. The vertical axis is dB loss in amplitude. The curve that drops fastest is the X2 up-sampling. At X16, you 3dB loss is all the way up near 320KHz, with hardly any loss at 20KHz.

        True, one can compensate for the loss, but it takes a lot of DSP (signal processing). It is not possible to compensate well for the Sin(X)/X curve with analog circuits.

        So the bottom line is: you do not want a DA without any up-sampling. You may not need to up-sample a lot, but you need some up-sampling to enable good filtering and flat response. Most DA’s today up-sample between X64 to X1024. This is overkill for flat response, and is very good from an analog filter standpoint. The reasons for up-sampling so much are is due to modern DA converter architectures such as sigma delta designs.’

        • Jay, I’ve read Dan’s paper a number of times.

          • Nik Razis

            I’m sure there are people who would argue the earth is flat AND quote an Einstein paper to prove it…

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