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These are a few of my published articles that I have converted to PDF format. They are all copyright James A. Moorer. Feel free to use these with appropriate referencing.
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Hard-Disk Recording and Editing of Digital Audio.  Presented at the 89th AES convention, September 21-25 1990, Preprint Number 3006 (K-6)

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Whither Dither: Experience with High-Order Dithering Algorithms in the Studio.  with Julia C. Wen. Presented at the 95 AES convention, October 7-10 1993, Preprint Number 3747 (B3-AM-3)

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Breaking the Sound Barrier: Mastering at 96 kHz and Beyond. Presented at the 101st AES Convention, November 8-11 1996, Preprint Number 4357 (I-2)

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Music Recording in the Age of Multi-Channel. Presented at the 103rd AES Convention, September 26-29 1997, Preprint Number 4623 (F-5)

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Towards a Rational Basis for Multichannel Music Recording. (with Jack H. Vad) Presented at the 104th AES Convention, May 16-19 1998

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A Native Stereo Editing System for Direct-Stream Digital. (with Ayataka Nishio and Yasuhiro Ogura) Presented at the 104th AES Convention, May 16-19 1998

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48-Bit Integer Processing Beats 32-Bit Floating-Point for Professional Audio Applications. Presented at the 107th AES Convention, September 24-27 1999, Preprint Number 5038 (L-3)

Scanned Copies of Older Papers



For my older papers, I don't have machine-readable copies. A lot of them were done on text editing and formatting engines that don't exist any more. I am starting to scan them in so I can offer PDF copies over the net.



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The Synthesis of Complex Audio Spectra by Means of Discrete Summation Formulas Journal of the Audio Engineering Society, Volume 24, Number 9, November 1976, pp717-727. "A new family of economical and versatile synthesis techniques has been discovered, which provide a means of controlling the spectra of audio signals, that has capabilities and control similar to those of Chowning's frequency modulation technique. The advantages of the current methods over frequency modulation synthesis are that the signal can be exactly limited to a specified number of partials, and that 'one-sided' spectra can be conveniently synthesized." That was the abstract of the paper.

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Linear-Phase Bandsplitting: Theory and Applications (with Mark Berger) Presented at the 76th Convention of the Audio Engineering Society, October 8-11, 1984, New York, Preprint 2132 (session A-1) "There are a number of applications for banks of bandpass filters in professional audio studios, both for film and music production. In this paper, we explore digital techniques for bandsplitting that have the property that the spectrum may be separated into a number of bands such that when these bands are added back together, the result is a pure delay. There need be no amplitude or phase distortion other than delay. This allows such applications as linear-phase graphic equalizers, multi-band noise gates, and many other improvements over conventional studio equipment. These algorithms have been implemented on a large-scale audio signal processor and run in real time. They are currently being used in major motion picture production."

This is probably my most-misunderstood paper. Most people pick up on the noise-gate stuff, which led to the NoNOISE work I did at Sonic Solutions (after many refinements, most of which are still proprietary so that I can't disclose them). What I consider the most important part of the paper is that the effect of certain families of window functions such as Hamming, Hanning, Blackman, or any that are finite sums of harmonic sinusoids can be calculated in closed form. I find this a remarkable result (although obvious in hindsite). If I want to divide the spectrum into 5, 10, 50, or 500 bands, I can compute linear-phase FIR impulse responses that will do that by evaluating formula (17). Note that the bands do not have to be equal-width. For instance, I could do something like a wavelet transform by having consecutive filters use wider and wider bandwidths. These will all sum to an impulse (by construction), so that they are guaranteed to be an identity. I know now that it is not limited to those window functions - that this can be done using any window function. That is, you can just write down the coefficients of the impulse responses of a set of filters that will divide the spectrum up any way you want. Maybe I should rewrite this paper sometime. Or maybe not.

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The Use of the Phase Vocoder in Computer Music Applications Journal of the Audio Engineering Society, Volume 26, Number 1/2, January/February 1978, pp42-45. This paper is one of the first (maybe the absolute first) to show how to use short-term Fourier transform as a method of analyzing and synthesizing musical sound, but with the signal-processing rigor necessary to make the system an identity in the absence of modification. Probably the most ignored contribution, and the one I consider probably the most important, is the technique for unwrapping the time-variant phase. Equation (9) represents a largely foolproof unwrapping method that involves no heuristics. This paper led to much of the subsequent work by Dolson and others who have extended and refined the method for time and frequency modification of high-quality musical sound.

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Signal Processing Aspects of Computer Music: A Survey Proceedings of the IEEE, Volume 65, Number 8, August 1977, pp1108-1137. This was an invited paper. Larry Rabiner invited me to write and submit this paper. It still stands as a reasonable survey of signal processing in music. It is interesting that synthesis is so little used today, whereas recording and playback (i.e., sampling) is so common. I guess it's a lot easier. Missing from this paper is any discussion of processing of the signal (aside from analysis). The computation for any interesting processing, except maybe reverberation, was so expensive at that time that we were not able to do much of it.

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About This Reverberation Business Computer Music Journal, Volume 3, Number 2, June 1979. This is a somewhat rambling random walk through some investigations into room reverberation. I had originally submitted it to the Journal of the Acoustical Society of America (JASA). I got a scathing review back that I swear was longer than the paper. The reviewer complained that it was in "an antequated discursive style." Yeah, that's probably correct. The reviewer differed with me on several technical points. I thought about it a while and concluded that the reviewer missed the point and didn't know what he was talking about, and in at least one area was flat wrong. Rather than try to fight with the reviewer, I sent it to CMJ, who was quite happy to publish it the way I wrote it.

I should mention that I believe I got mixed up between feet and meters in the graphs of air attenuation with distance. You may want to check this in a real acoustics textbook if it is important to you.




Random Notes



I read somewhere that papers with a colon in the title attracted more attention and were perceived as more athoratative, so I always tried to put a colon in each and every title for my papers. I dunno if it worked or not.

There are some funny lines in these earlier papers. Since digital audio was in its infancy, I felt I had to work real hard to convince people that these techniques were for real. That accounts for statements like "[digital techniques] are currently being used in major motion picture production." You don't have to say things like that any more. I had other statements like "These techniques will have a major influence in the all-digital studios to come." Today, these statements seem totally gratuitous, since all the studios are digital, and the techniques are so routine that nobody thinks about them any more. I guess that is how it is supposed to be. I can't tell you how many people I had tell me that there is no way that we can do away with all the analog equipment. One by one, I watched the analog pieces of equipment get replaced by digital. Studios stopped asking "whither digital" and began asking "what will our approach to digital be?" Nowdays, every garage band has a digital recorder, whether it is a free-standing device or something running on a PC.

In "Signal Processing Aspects . . .", I describe digital recording and editing on the computer. Although we weren't the first to do it - I believe Tom Stockham and Robert Ingebretson at SoundStream antedated us by a few years - I think we were probably the second. Some digital recording had been done at MIT and Bell Labs a decade earlier, but not for the purpose of music editing. We could lay down up to 5 tracks before the sluggish hard drive of that era started missing transfers.