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Jim Austin  |  Jan 06, 2018  |  27 comments
Loss is nothing else but change, and change is Nature's delight.—Marcus Aurelius

Master Quality Authenticated (MQA), the audio codec from industry veterans Bob Stuart and Peter Craven, rests on two pillars: improved time-domain behavior, which is said to improve sound quality and what MQA Ltd. calls "audio origami," which yields reduced file size (for downloads) and data rate (for streaming). Last month I took a first peek at those time-domain issues, examining the impulse response of MQA's "upsampling renderer," the output side of this analog-to-analog system (footnote 1). This month I take a first look at the second pillar: MQA's approach to data-rate reduction. In particular, I'll consider critics' claims that MQA is a "lossy" codec.

Jim Austin  |  Dec 12, 2017  |  213 comments
I don't think I've ever seen an audio debate as nasty as the one over Master Quality Authenticated (MQA), the audio-encoding/decoding technology from industry veterans Bob Stuart, formerly of Meridian and now CEO of MQA Ltd., and Peter Craven. Stuart is the company's public face, and that face has been the target of many a mud pie thrown since the technology went public two years ago. Some of MQA's critics are courteous—a few are even well-informed—but the nastiness on-line is unprecedented, in my experience.
Jim Austin  |  Apr 19, 2018  |  47 comments
In an article published in the March 2018 Stereophile, I wrote that critics have been attacking MQA, the audio codec developed by J. Robert Stuart and Peter Craven, by accusing it of being lossy. The critics are right: MQA is, in fact, a lossy codec—that is, not all of the data in the original recording are recovered when played back via MQA—though in a clever and innocuous way. For MQA's critics, though, that's not the point: They use lossy mainly for its negative emotional associations: When audiophiles hear lossy, they think MP3.
Jim Austin  |  May 17, 2018  |  160 comments
The right thing at the wrong time is the wrong thing.—Joshua Harris

The sampling theory formulated by Claude Shannon in the late 1940s had a key requirement: The signal to be sampled must be band-limited—that is, it must have an absolute upper-frequency limit. With that single constraint, Shannon's work yields a remarkable result: If you sample at twice that rate—two samples per period for the highest frequency the signal contains—you can reproduce that signal perfectly. Perfectly. That result set the foundation for digital audio, right up to the present. Cue the music.

J. Robert Stuart  |  Aug 11, 2016  |  116 comments
Author's Note: We are grateful to Stereophile for the opportunity to address some frequently repeated technical questions appearing in comments to articles. Recently this has included misunderstandings about noise calculation, dynamic range, resolution, definition, music spectra, channel capacity, lossless processing and temporal aspects of digital channels.

To simplify this document we have grouped the topics and set them as questions and answers either as response, tutorial or axiom. Some months ago we published a comprehensive Q&A for an online forum and to avoid repetition we occasionally refer to topics already discussed there (see [37] in the "References" sidebar).—J. Robert Stuart

Jim Austin  |  Aug 29, 2017  |  66 comments
Nelson Pass is a consummate engineer, but he got his start in physics, earning a bachelor's degree from UC Davis. As he worked on his degree, he was already an audio designer, focusing on loudspeakers—great training for a designer of audio amplifiers. Soon, in 1974, he cofounded Threshold Audio with René Besne, of audio and folk-dancing fame; their goal was to build electronics, partly because the field is less competitive—it's harder than building speakers.
Keith Howard  |  May 02, 2004  |  First Published: Apr 01, 2004  |  0 comments
Looked at from one viewpoint, DVD-Audio and SACD appear to be exercises in sheer profligacy. In the case of DVD-A, why provide a maximum bandwidth almost five times what is conventionally taken to be the audible frequency range, and couple it to a dynamic-range capability far in excess of that achievable by the microphones used to record the sound? In the case of SACD, why provide a potential bandwidth in excess of 1.4MHz, only to fill more than 95% of it with quantization noise?
Robert Harley  |  May 30, 2004  |  First Published: Feb 01, 1992  |  0 comments
At the February 1991 Audio Engineering Society Convention in Paris, Audio Precision's Dr. Richard Cabot (see my interview in January 1991, Vol.14 No.1) proposed a new technique for measuring noise modulation in D/A converters (footnote 1). The method, based on psychoacoustic principles, attempts to predict the audible performance of D/A converters. Now that Stereophile has added digital-domain signal generation and analysis to our Audio Precision System One, we can employ Dr. Cabot's technique and see if there are any correlations with subjective performance.
Martin Colloms  |  Nov 24, 1992  |  0 comments
Martin Colloms (footnote 1) suggests that the traditional ways of assessing hi-fi component problems overlook the obvious: does the component dilute the recording's musical meaning?
Peter W. Mitchell, Barry Fox, Peter van Willenswaard  |  Jul 05, 2016  |  First Published: Apr 01, 1991  |  2 comments
Editor's Note: In the 21st Century, lossy audio data compression, in the form of MP3 and AAC files, Dolby Digital and DTS-encoded soundtracks, and YouTube and Spotify streaming, is ubiquitous. But audiophiles were first exposed to the subject a quarter-century ago, when Philips launched its ill-fated DCC cassette format. What follows is Stereophile's complete coverage on both DCC and its PASC lossy-compression encoding from our April 1991 issue.—John Atkinson
Peter van Willenswaard, John Atkinson, Peter W. Mitchell  |  Jun 28, 2016  |  First Published: May 01, 1989  |  3 comments
Editor's Note: One-bit DAC chips in the 21st century, where the anlog output signal is reconstructed from a very high-rate stream of pulses, are ubiquitous. But a quarter-century ago, those chips were only just beginning to stream from the chip foundries. In this feature, we aggregate Stereophile's 1989 coverage of the then-new technology, starting with Peter van Willenswaard on the basics.—John Atkinson.

1989 may well become the year of the D/A converter (DAC). CD-player manufacturers have, almost without exception, launched research projects focusing on this problem area of digital audio; many of these projects are already a year old. This is, however, by no means the only imperfection keeping us away from the high-quality sound we have come to suspect is possible with digital audio media, and maybe not even the most important.

Jim Austin  |  Nov 01, 2018  |  10 comments
Virtually all of the active components in your system—DACs, preamplifiers, power amplifiers—work by modulating the DC output of their power supplies with an AC music signal. Surely, then, the more perfect your household AC is, the more perfect your audio system's output will be. Analogies abound—to dirty water used in distilling good whiskey, to inferior thread used to weave fine fabrics—and all amount to the same thing: you can't make a silk purse out of a sow's ear.
Robert Harley  |  Mar 29, 1995  |  0 comments
Time to 'fess up: How many of you actually read the "Measurements" sections of Stereophile's equipment reports and understand what's being measured, and why? I suspect that many readers skip over the technical assessment of the reviewed product and make a dash for the "Conclusion."
Thomas J. Norton  |  Jan 06, 1994  |  0 comments
On a number of occasions we have commented on the effects of an amplifier's output impedance on a system's performance. A high output impedance—such as is found in many tube amplifiers—will interact with the loudspeaker's impedance in a way which directly affects the combination's frequency response. The Cary CAD-805, for example, has a lower output impedance than most tube amplifiers, and should be less prone to such interaction. Some months back—before the CAD-805 arrived—I investigated this phenomenon in conjunction with measurements for a forthcoming review of the Melos 400 monoblock amplifier. Since the Melos 400 also had a relatively low output impedance for a tube amplifier (at 0.43 ohms at low and mid frequencies, rising to 1.2 ohms at 20kHz, from its 8 ohm tap), I took that opportunity to run some frequency-response measurements using an actual loudspeaker as the load for the amplifier.
John Atkinson  |  Aug 19, 1995  |  1 comments
As mentioned by two readers in this month's "Letters," amplifiers are used to drive loudspeakers but are almost exclusively measured into resistive loads. The reasons for this are twofold: 1) real loudspeakers both produce neighbor-annoying sound levels and tend to break when driven with typical amplifier test signals; and 2) the question as to which "standard" loudspeaker should be used is impossible to answer---at least the conventional resistive loads are consistent and repeatable.

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