Don't believe all the myth BS about "oh its digital, a copy is exact".
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8 Digital Audio Myths Exploded
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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.
Although jitter causes distortion on playback — and can certainly generate unalterable distortion during the A/D process when recording from an analog source — it is not recorded when making a digital dub or when recording between digital devices. On the other hand, if the jitter of one device involved in a digital dub is bad, it can cause problems of a different sort, as I will discuss in the next myth.
Myth No. 6: Digital dubs are perfect copies
If you copy audio between DAT decks, CDs, or modular digital multitrack (MDM) tape machines, you might expect the copy to be perfect. After all, digital audio is just ones and zeros, right? As I described previously, copies of audio files on computers are indeed perfect. However, when you use tape and CD-audio media, those zeros and ones are being assisted, and sometimes created, by error correction.
Here's the problem: If an error occurs when a computer reads a file from a hard drive, the computer can go back and try again. A backup tape drive can do the same thing, rewinding the tape as necessary. But with an audio DAT and other digital media, the tape or disc must always keep moving; otherwise, you hear a pause in playback. When errors occur, and they always do, going back and trying again simply isn't an option. Instead, the DAT and CD formats include several methods for real-time error correction.
The first and most effective error correction is recorded onto the tape in the form of Reed-Solomon error-correction codes. These codes take up more than 25 percent of the data on a DAT tape or CD, and they allow most errors to be completely corrected, yielding data that is byte-for-byte perfect.
Occasionally, even this sophisticated error-correction mechanism can't recover the data, resulting in one or more completely blank samples. In that case, the second level of error correction comes into play. This technique, called interpolation, considers the data before and after the blank sample or samples and then makes a guess as to what value might have been in that blank space.
Say you have the following sequence of samples: 2, 6, <error>, 14, 15. Simple interpolation might draw a straight line between the samples just before and just after the error so that the sequence becomes 2, 6, (10), 14, 15. More sophisticated interpolation constructs a curve based on two or more samples on either side of the error. This procedure recognizes that the two samples after the error have closer values than the two before the error, and it tries to reflect that curvature in the corrected data. Its reconstruction of the data might look more like 2, 6, (12), 14, 15 (see Fig. 5). Generally, the more points you look at during interpolation, the more accurate of a guess you can make as to the error's original value, though it's still just a guess.
Audio is almost always a continuous waveform, so the results of interpolation are good enough for musical use; it's nearly impossible to hear a single error. However, if a tape is in not in perfect condition or the deck's heads haven't been cleaned recently, you might have enough errors to cause a general degradation of audio quality, which passes on to any copies you make — digital or analog. Some DAT machines and MDMs display the current error rate. If yours does, monitor it from time to time.
So far I've only covered minor imperfections in digital dubs. Some problems can be more severe. For instance, if a DAT tape has too consecutive many errors, the error correction may not work at all, and you'll hear digital noise instead of your recording.
Jitter is another issue. As I discussed previously, it shouldn't affect purely digital connections. AES standards specify maximum allowable jitter on output and minimum jitter tolerance on input, which defines the greatest amount of jitter that the input signal can include and still be received properly. The current standard specifies a minimum jitter tolerance of several times the maximum output jitter to allow for chains of devices and maximal interoperability. Devices that conform to these standards shouldn't have jitter-related difficulties. Not all devices meet the necessary specs, however, and some older devices may have been built prior to the adoption of the current standards.
If a device has extremely high jitter at its digital output or if the device receiving the data can't handle much jitter at its input, their communication can fail, causing pops, clicks, and other artifacts. That holds true for connections between all digital devices, including digital audio workstations (DAWs), sound cards, and effects processors.
Likewise, digital cables can fall prey to most of the same gremlins as their analog counterparts, including loose connections, defects, and impedance mismatches. You can always get errors in DAWs from disk-related problems (such as buffers set too small), or processor spikes caused by brief overloads in system activity.
Finally, many digital-audio glitches — clicks, pops, low-level “fizz,” and the like — can be caused by a simple problem: improper word-clock settings. Two devices that are connected digitally but are not in agreement on the same word clock form a recipe for real trouble. In my experience, this is the source of many erroneous complaints that digital transfers are error prone. If you merely ensure that all connected devices agree on a single word-clock master, you can eliminate this common headache.
In summary, to ensure premium digital dubs, keep tape heads clean, preserve media (such as DAT tapes) in safe storage environments, set word clocks correctly, pay attention to error rates, take the same care with digital cables as you would with analog cables, make sure that your equipment has reasonable jitter characteristics, and set your DAW buffer sizes conservatively. As long as you take those precautionary steps, you can expect digital dubs to work well.
One other point you should keep in mind is to always monitor while you dub, just as you would with analog tape. Your ears should be your final reference. Remember, the recording you save may be your own.
Myth No. 7: All digital synths and effects sound the same
This myth is also known as: All digital EQs sound the same, All virtual analog synths sound the same, and All digital compressors stink. This is as true for digital gear as it is for analog gear, which is to say, not at all.
With analog devices, you have great-sounding EQs and lousy-sounding EQs. What's more, a couple of great-sounding EQs (or compressors or synthesizers) may sound totally different from each other. It should be no shock that the same situation exists for hardware and software in the digital domain.
What makes a good or bad digital EQ, compressor, filter, or oscillator? Many issues of digital-audio quality arise from one source, and it happens to be one of the major differences between analog and digital audio: frequency range. Analog audio has a theoretically infinite frequency range, whereas digital audio (software and hardware) has a hard limit on high frequencies, as determined by the sample rate.
Many analog processes take “infinite” frequency range for granted, but they can't do the same in the digital domain. For example, a standard analog peaking EQ has a bell curve that is symmetrical around the center frequency; one side slopes to 0 Hz, and the other slopes toward infinite Hz.
You can implement the same EQ in the digital domain, but infinity is suddenly much closer. In fact, what was infinite Hz is now the Nyquist frequency (half the sampling rate) or 22.05 kHz at a sampling rate of 44.1 kHz. This difference in the proximity of infinity results in an EQ curve with a dramatically lopsided shape (see Fig. 6).
Things can get stranger as the EQ's center frequency approaches the Nyquist frequency. At those high frequencies, I've seen digital EQs that started to take on weird globular shapes and even go down in actual frequency as I turned up the frequency knob. I've even encountered a peaking EQ that looked much more like a resonant highpass filter.
Those problems can be avoided or at least minimized by clever programming. The degree to which the programmer is successful defines, to a great extent, the differences between good and bad digital EQs.
Besides the creation of a more-or-less correct EQ curve, there are also matters of taste and personality, just as in analog EQs. The way a programmer chooses to approach these infinite-frequency quandaries affects the overall sound. Moreover, some products emulate the more esoteric sonic distinctions among classic analog EQs, such as slope and overshoot characteristics.
Compressors and limiters also have frequency-related issues. You're probably familiar with aliasing — it causes audio artifacts when sampled audio contains frequencies higher than the Nyquist frequency. Aliasing doesn't occur only during sampling, however; it can also happen entirely within the digital domain.
For instance, compression and limiting work by modulating one audio-rate signal (the input) with another audio-rate signal (the compressor or limiter's automatic gain control, which operates in the audio range when the attack and release envelope times are fast). When you modulate one audio-rate signal with another, it has the effect of adding the two signals' frequencies; if the total exceeds the Nyquist frequency, you'll get some aliasing.
A full-bandwidth audio signal processed with a limiter or a compressor with fast attack or decay times falls into that category; the faster the attack or release and the greater the compression or limiting amount, the more aliasing you hear. That is the cause of the crunchiness many people hear in digital-dynamics processors. Again, clever programming, especially oversampling, can minimize these aliasing artifacts.
You'll find similar predicaments in synths. Resonant filters suffer from the same infinity-is-much-too-close syndrome as digital EQs, and various synths differ widely in their success at addressing the problem. For instance, standard Chamberlin digital filters (the most common type) only work correctly to about one-sixth of the sampling rate. For a synthesizer running at 44.1 kHz, that means the resonance tops out at about 7 kHz.
Oscillators have problems similar to compressors. For example, a square wave at, say, 4 kHz is actually generating frequencies well above the Nyquist limit, because of the waveform's sharp edges. Untamed, that can cause excruciating aliasing, especially toward the top of the keyboard (as you can hear in some popular products). Similar aliasing can also happen when samples are transposed above their original pitches. Techniques for dealing with these complications vary and account for some of the sonic differences among synthesizers.
Finally, it's worth noting that some solutions are common knowledge and in the public domain, whereas many others are protected by patents or kept close to the vest as trade secrets. In short, different products use different techniques for dealing with frequency-related obstacles, and some are simply more successful and pleasant sounding than others.
Myth No. 8: Hardware sounds better than software
Defining an audio process as a mathematical equation is essentially what digital hardware and software is all about. Whether you're using a synth or effects processor with dedicated hardware, an algorithm running on a DSP chip, or a plug-in on a Pentium or PowerPC, it's still just a numbers game.
The thing that creates sound quality in a digital synth or effect is the math itself, or the algorithm. As long as the math stays the same, it can run on custom hardware, off-the-shelf DSPs, Pentiums, or PowerPCs and still produce the same output. To look at it another way, the method that converts the math to a form you can use (a software plug-in or a hardware box, for example) is unimportant; only the math matters (see Fig. 7).
So hardware should sound the same as software, right? In general, yes, though several factors complicate the matter. Some issues are technical; others are simply practical.
On the technical side, you may not always be able to run exactly the same math on different hardware. You have two common approaches to performing math on computers: one is called fixed point, and the other is called floating point. Without getting overly specific, I'll just say that the same mathematical operation, such as adding two numbers, may produce slightly different results depending on which method you use.
Many popular DSPs, such as the Motorola 56000 series, perform only fixed-point math; others, such as the SHARC from Analog Devices, practice floating-point math. Desktop processors, such as the PowerPC and Pentium, can handle both calculation types but, in some cases, may do one better than the other. As a result, it may not always be practical — or even possible — to perform the same math on two different machines. In that case, the algorithm designer must write the algorithm specifically for each processor. With careful work, the algorithm may sound exactly alike on each machine, even to a golden ear. In some cases, however, slight differences may remain.
Technical issues aside, there are also practical reasons why hardware and software products may sound different. The underlying algorithms of two products may be very different, for example. If you compare a hardware product from one manufacturer with a software product from another, chances are that the algorithms will not be the same.
The factory presets are important factors in synthesizers' and effects processors' overall sound. Without good sound design, even the best algorithm may not sing as sweetly as it could. Conversely, talented sound designers can sometimes make a mediocre algorithm sound surprisingly good. So even if the two products' underlying algorithms are similar, the talent of the factory sound designers can make all the difference in the world, and that varies from one factory to another.
There is no theoretical reason why hardware should sound different from software. Any differences you encounter are most likely the results of comparing apples and oranges, because few products are offered in identical hardware and software forms.
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Dan Phillips is a Bay Area-based composer and producer, and he is product manager at Korg R&D. Check out his Web site at
www.danphillips.com. Thanks to Andy Leary, Rudy Trubitt, and Benny Rietveld for their assistance with this article .
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