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.

7 thoughts on “Compression vs. Compression? Part II

  • Andy Punch

    Suggestions for greater clarity:
    Title over amplitude waveform graphics: Audio Dynamic Range Compression and Normalization
    Title under center amplitude waveform graphic: Source file heavily compressed
    Title under right amplitude waveform graphic: Source file heavily compressed, then normalized to maximum volume.

    It should be clear what type of Audio Compression is illustrated.

  • Probably this is the reason why we keep pumping money into expensive systems with the hope of getting more out of compressed audio files.

    If the source was left untouched it would be much easier to setup a reasonably priced system.

    I listened to your sample files on my modified PC speaker system – the sound was clear and enjoyable. I imagine the quality of sound on the high end system in the living room. I do not have a system that can playback DASD files yet.

  • Mark, if that is ‘the standard operating procedure in the commercial music business’, it doesn’t explain the large amounts of clipping found on many pop music CDs. The procedure you describe would yield no clipping.

    • Actually, it does. In a standard mastering session there a number of steps that contribute to increasing the volume. Compression and normalization are two of them. There are others such as “ultra maximizer” that push even further…and expanders. A common trick is to push past clipping and then pull back .2 dB…that’s why you see so many tracks that aren’t clipped but have a max of .2 dB below full scale.

  • I felt compelled not to comment on your article but on a recent advertisement which seemed to me to be quite misleading:

    Coming August 15, 2014 pre-order your copy today! Orders with both pre-order and in stock items will have all in stock items shipped immediately!
    This item not eligible for any further discount offers!
    Special Limited Time Sale Price!!!
    Limited Run of Only 1,000 First Edition Pressings Available for Optimal Sound Quality!!!
    Ultra High Definition 32-Bit Mastering Produced Using PureFlection Process!
    This Ultra HD 32-Bit Mastering CD will play on ALL CD players!
    First Impressions Music presents Decca: Supreme Stereophonic Legacy, a narrative and musical account of the Golden Age of Decca stereo recordings. Set of 4CDs with beautiflly illustrated book!
    Format: 4 First Impressions Music Limited Edition Ultra HD 32-Bit Mastering CDs Plus Book Box Set
    sale $224.99


    This advertisement is from the Elusive Disc website and is for the Decca: Supreme Stereophonic Legacy Ultra HD CD Box Set . It seems to me that while 32 bit mastering of the CD is well and good, the standard playback on a routine CD player will be at a 44.1kHz sampling rate and 16 bit word length, correct?

  • Stuart O'Malley

    Another interesting article. If you think it’s appropriate, perhaps in Part III or another future article mention using fixed point versus floating point-based software for mixing and mastering and resampling, with the significance that using floating point helps avoid hard clipping with intermediate processing steps. In other words, in fixed point programs a clipping event will permanently harm the signal, while in floating point programs the overage can be carried all the way through processing without any problems for the most part, and only must be dealt with at the final step of creating the digital file for customers.

    • Great idea…and you’re absolutely right.


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