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Get the lowdown on common digital-audio fallacies.

The Information Age is a wonderful time, isn't it? With global media and the Internet, you can find data on just about any topic. Unfortunately, a lot of conflicting information is floating around out there, and it's often hard to tell fact from fiction. This article attempts to clear the air by addressing some common misconceptions about digital audio.

Myth No. 1: Copies of files aren't always perfect.

Dubs between analog tape decks aren't perfect; every time you make an analog copy, the signal degrades. It's therefore natural to assume that all copying methods share that characteristic. Copying an audio file on a computer, however, is completely different from making an analog copy.

When you copy a file on a computer-- whether it's an audio file, a Microsoft Word document, or a shareware program--the operating system has to ensure that every byte of data copies correctly. If one byte in a Word document goes astray, you might get spelling errors, formatting problems, or worse. If one byte in a copy of a shareware program goes south, the software might not run at all.

Because of this situation, accurate copies of any file type are crucial, and digital-audio files are no exception. To prevent problems, the operating system uses a verification scheme to establish that all copies are byte-for-byte perfect. In the unlikely event that an error appears in the copy, the computer lets you know.

So when you copy an audio file from one hard drive to another or back up data to a tape drive or CD-R drive, rest assured that you're creating a perfect duplicate.

Myth No. 2: All file compression degrades audio. Compressed audio formats, such as MP3, have truly changed the face of recorded media by letting music be exchanged easily over the Internet. The MP3 format shrinks audio files using "lossy" compression, which means that not all of the musical data is actually stored in the MP3 file. The more important data is maintained while less important data is thrown away. The audio file is then reconstructed on playback with varying results in audio quality (see Fig. 1). In any event, MP3 audio quality is degraded somewhat with respect to the original file.

Because MP3 is one of the most widely known audio-compression formats, many people assume that all methods of compressing audio files work the same lossy way. However, not all of them do. Some programs, such as Emagic's Zap and Waves' TrackPac, are specifically designed for lossless audio compression (see Fig. 2). Those programs can't shrink files as much as MP3 does, but they do retain all data while compressing files to about 50 percent of their original sizes.

Also lossless by design are general-purpose compression programs such as PKWare's PKZIP, WinZip Computing's WinZip, and Aladdin Systems' Stuffit. To these programs, an audio file is just like a Microsoft Excel document; every byte of data must be retained. Again, the file-size reduction isn't as dramatic as with MP3 compression (and it's often less effective than audio-specific compression programs), but you can be sure that the quality of any zipped or stuffed audio file is completely unaffected by the compression.

Myth No. 3: CD quality. What the heck does "CD quality" mean, anyway? My cumulative annoyance at the misuse of this phrase leaves me feeling like a cranky old curmudgeon when I hear it. Sure, I'll accept the description for any device that operates at 16 bits and 44.1 kHz--a CD player, for example-- as long as its real-world performance measures up to the potential implied by those specs.

Unfortunately, the term is often used to describe almost anything that can spit out a tune. I've seen a $30 sound card with a 65 dB signal-to-noise ratio boast CD quality, even though 16 bits should offer a signal-to-noise ratio closer to 90 dB. Moreover, I've seen MP3 and MiniDisc players claiming CD quality, though those devices start with CD-quality audio and then shrink it using lossy compression, reducing both file size and fidelity. Soon we'll have 4-bit digital toasters claiming their beeps are CD quality. Give me a break!

Even worse is the phrase "near CD quality." For those unfamiliar with marketing doublespeak, "near" is the same as "virtually." In plain English, both words translate to "not". So what was once a technical term is now simply advertising gibberish.

Finally, I have to ask: is CD quality still supposed to be a good thing? At one time, 16-bit, 44.1 kHz audio was synonymous with state-of-the-art digital technology. But that was then. In today's 24-bit world (with 96 kHz sampling rates gaining in popularity), those CD specs are looking a bit long in the tooth. Maybe instead of CD quality, the industry can agree on a more appropriate term like "real old-fashioned CD goodness." It's just a thought.

Myth No. 4: 24-bit is 24-bit is 24-bit.

Resolution is an easy way to specify a digital device's quality. Unfortunately, it is not a reliable benchmark. I remember a meeting with a representative from a major digital-audio--chip manufacturer in which two of the manufacturer's models of 20-bit D/A chips were evaluated. When asked why one of the chips was abnormally noisy and performing more like a 14-bit D/A than its 20-bit spec suggested, the representative responded that it was "20 bits--with 6 bits of marketing."

So what's the moral of the story? Just because two devices are both "24-bit" does not mean they exhibit the same audio quality. In fact, fidelity can vary so widely that a well-designed 16-bit device may sound better than a poorly designed 24-bit instrument.

One variable is the quality of the D/A or A/D chip. The major manufacturers of these chips may have a line of parts with the same general specifications (such as 24-bit, 44.1 to 48 kHz) but with widely diverging noise amounts and differing prices. The clock-circuit quality is also important for minimizing jitter. (For more on jitter, see Myth No. 5.) In fact, several high-end A/D/A manufacturers specifically cite the their clocks' stability as an important selling point (see Fig. 3).

Finally, remember that the A in D/A stands for "analog." You know that there are good and bad sounding analog mixers, preamps, and other gear, so it should come as no surprise that a "digital" device's analog parts can make a real difference in its overall sound quality. High-quality analog parts and clever analog design are absolutely essential for a digital device to realize its true potential.

Myth No. 5: Jitter is recorded during digital dubs. In a perfect world, each digital-audio sample is recorded and played back at exact, even intervals derived precisely from each tick of the digital-audio word clock. For instance, a 44.1 kHz system should sample the incoming audio exactly 44,100 times per second. Real-world clocks aren't quite perfect, however, and each tick of the clock may be slightly behind or slightly ahead of where it's supposed to be. That difference between the ideal timing and the actual timing is called jitter.

Jitter causes distortion in digital audio, but it's different from what you generally think of as distortion. Instead of distortion in amplitude, such as overdrive in a guitar amp, jitter is distortion in time that causes slight variations in the audio waveform's shape. In a sine wave, for example, varying each sample's timing causes the waveform to bulge out and cave in at different points, as opposed to following the ideal smooth curve (see Fig. 4).

Every digital-audio device produces some amount of jitter, but some devices exhibit much more than others do. Jitter can also be cumulative: as a signal passes through multiple signal processors, mixers, and so on, the jitter may get progressively worse. Jitter becomes "frozen" when you record an analog source with a digital system. In other words, every time you play back the audio, you hear the effect of the jitter that was present during the recording. You also hear the jitter produced by the digital-to-analog converter.

However, recording from one digital device to another is different; as long as the data stays in the digital domain, jitter is not recorded. The only thing actually captured is a sequence of amplitude values; digital media simply have no provisions for storing information about individual sample timing. The timing is based implicitly on the sampling rate and is freshly re-created by the digital-to-analog converter's clock every time the audio is played back.

Even digital-audio tape systems don't play audio directly from the tape. Instead, they pass the data through a RAM buffer from which a clock pulls individual samples and sends them to the outputs. As a result, variations in tape speed or data spacing aren't reflected in the output data.