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Why doesn't MiniDisc sound as good as DAT or CD? After all, they're all digital.

MiniDisc or MD uses a loss-y compression algorithm (called ATRAC); crudely, this means that the numbers coming out of the machine are not the same as those that went in. The algorithm uses complex models of the way the ear works to discard the information that it thinks would not be heard anyway. For example, if a pin dropped simultaneously with a gunshot, it may be reasonable to suggest that it isn't worth bothering to record the sound of the pin! In fact it turns out that around 75 to 80 per cent of the data for typical music can be discarded with surprisingly little quality loss. However, nobody denies that there is a quality loss, particularly after a few generations of copying. This fact and others make both MD useful only as a performance or consumer-delivery format. They have very little use in the studio as a recording or (heaven forbid!) mastering format. Recent advances have made the MD much closer to the uncompressed signal. They have also been integrated into a number of lower cost 4 and 8 track recording devices. MD is now also found in a number of radio stations, theatres and performance venues.

What is S/P-DIF and what is AES/EBU?

AES/EBU and S/P-DIF describe two similar protocols for communicating two-channel digital audio information over a serial link. They are slightly different in details, their basic format is almost identical, but there are enough differences that the two are, for all intents and purposes electrically incompatible. Both of these digital protocols are described fully in an international standard, IEC 958, available from the International Electrotechnical Commission. AES/EBU (which stands for the joint Audio Engineering Society/European Broadcasting Union standard) is the so-called "professional" protocol. It uses standard 3-pin XLR connectors and 110-ohm balanced differential cables for connection (no, standard microphone cables, not even good quality cables, won't work, even though it seems they might) and a 5 volt, differential signal. This is now properly called AES-3, and there is a version called AES-3id which allows the same format over coaxial cable. S/P-DIF (which stands for Sony/Philips Digital InterFace, a now obsolete standard superseded by IEC 958) is the so-called "consumer" format. It uses what appears to be standard RCA connectors and cables, but, in fact, require 75-ohm connectors and cables. Good quality video "patch" cables have proven adequate (no, standard "audio" patch cords, even excellent quality versions, have been shown not to work). The signals are 0.5 volts unbalanced. The actual data streams are very similar. Each sample period a "frame" is transmitted. Each frame consists of two "subframes", one each for left and right channels, each subframe is 32 bits wide. In that subframe, 4 bits are used for synchronization, then up to 24 bits are usable for audio (the "consumer mode" format is limited to 16 bits). The remaining four bits are used for parity (the first level of error detection), validity, user status and channel status. 192 subframes are collected, and the 192 user bits and 192 channel status bits are collected into separate 24 8 bit status bytes for each channel. The channel status bytes are interesting, because they contain the important control information and the major differences between the two protocol formats. One bit tells whether the data stream is professional or consumer format. There are bits that specify (optionally) the sample rate, de-emphasis standards, channel usage, and other information. The consumer format has several bits allocated to copy protection and control: the SCMS bits. Now, the notion that all of this is encoded in a standard may be reassuring, but a standard is nothing but a voluntary statement of common industry practice. There is a lot of incompatibility between equipment out there caused directly by subtle differences between interpretations and implementations. The result is that some equipment simply refuses to talk to each other. Even THAT possibility is stated in the standard!

What kind of cable AES/EBU or S/P-DIF cables should I use? How
long can I run them?

The best, quick answer is what cables you should NOT use! Even though AES/EBU cables look like ordinary microphone cables, and S/P-DIF cables look like ordinary RCA interconnects, they are very different. Unlike microphone and audio frequency interconnect cables, which are designed to handle signals in the normal audio bandwidth (let's say that goes as high as 50 KHz or more to be safe), the cables used for digital interconnects must handle a much wider bandwidth. At 44.1 KHz, the digital protocols are sending data at the rate of 2.8 million bits per second, resulting in a bandwidth (because of the biphase encoding method) of 5.6 MHz. This is no longer audio, but falls in the realm of bandwidths used by video. Now, considerations such as cable impedance and termination become very important, factors that have little or no effect below 50 kHz. The interface requirements call for the use of 110 ohm balanced cables for AES/EBU interconnects, and 75 ohm coaxial unbalanced interconnects for S/P-DIF interconnects. The used of the proper cable and the proper terminating connectors cannot be over emphasized. Ordinary microphone or RCA audio interconnects DO NOT WORK. It's not that the results sound subtly different, it's that much of the time, it the receiving equipment is simply unable to decode the resulting output, and simply shuts down. As to how long these cables can be, it's hard to say. However, a couple of general rules apply. S/P-DIF was NEVER intended to be a long-haul hardware interconnect. The relevant specifications talk of interconnect lengths less than 10 meters (33 feet). In fact, many pieces of equipment cannot tolerate cables even that long, due to the excessive capacitance and possibly induced common mode interference. AES/EBU is more tolerant of longer runs because it is balanced (thus more immune to interference) and it's run at a higher signal level (5 volts instead of 0.5 volts). The standards "allow signal transmission up to a few hundred meters in length." The reality is that much is highly dependent upon the actual conditions at hand.

What's all this about 20- and 24-bit digital audio? Aren't CDs limited to 16 bits?

Yes, CDs are limited to 16 bits, but we can use >16-bit systems to produce 16-bit CDs with higher quality than we could otherwise. We are able to record audio with effective 20-bit resolution nowadays. The finest A/D converter systems have THD+N values around -118 dB with linearity extending far below even that. When it comes time to reduce our word-length to 16 bits, we can use any one of a variety of noise shaping curves. The job of which is to mix with our 24-bit audio, shift the dither spectrum of the noise into areas where our ears are less sensitive, thus enabling the noise component to comprise audio information at the spectral areas where our ears are most sensitive. See Lipschitz's seminal papers for fuller detail on this subject. Furthermore, we often perform DSP calculations on our audio, and to that end it is worthwhile to carry out the arithmetic with as much precision as we can in order to avoid rounding errors. Most digital mixers carry their math out to 24-bit precision at the I/O, with significantly longer word lengths internally. As a result, two 16-bit signals mixed together can produce a valid 24- bit output word. For that matter, a 16-bit signal subjected to a level change can produce a 24-bit output if desired (except, of course, for a level change that is a multiple of 6 dB, as that's just a shift left or right). The number of noise shaping curves available today is staggering. Sony SBM, Sonic TBM, Apogee UV-22, Prism SNS, Lexicon, PONS, Waves, and, of course, the classic Lipschitz curve are just a few of the multitudinous options that now exist

What is a digital audio workstation?

Ddigital audio workstation (DAW) is one of our newest audio buzzwords, and applies to nearly any computer system or dedicated desktop audio recording system that is meant to handle or process digital audio in some way. For the most part however, the term refers to computer-based nonlinear editing systems. These systems can comprise a $500 board that gets thrown into a PC, or can refer to a $150,000 dedicated digital mastering desk

What is mastering?

Mastering is a multifaceted term that is often misunderstood. Back in the days of vinyl records, mastering involved the actual cutting of the master that would be used for pressing. This often involved a variety of sonic adjustments so that the mixed tape would ultimately be properly rendered on vinyl. The age of the CD has changed the meaning of the term quite a bit. There are now two elements often called mastering. The first is the eminently straightforward process of preparing a master for pressing. As most mixdowns now occur on DAT, this often involves the relatively simple tasks of generating the PQ subcode necessary for CD replication. PQ subcode is the data stream that contains information such as the number of tracks on a disc, the location of the start points of each track, the clock display information, and the like. This information is created during mastering and prepared as a PQ data burst which the pressing plant uses to make the glass pressing master. Mastering's more common meaning, however, is the art of making a recording sound "commercial." It is the last chance one has to get the recording sounding the way it ought to. Tasks often done in mastering include:

� Adjustment of time between pieces
� Quality of fade-in/out
� Relation of levels between tracks (such that the listener doesn't have to go swinging the volume control all over the place)
� Program EQ to achieve a desired consistency
� Compression to make one's disc sound LOUDER than others on the market

The list goes on.

A good mastering engineer can often take a poorly-produced recording and make it suitable for the market. A bad one can make a good recording sound terrible. Some recordings are so well produced, mixed, and edited that all they need is to be given PQ subcode and sent right out. Other recordings are made by people on ego trips, who think they know everything about recording, and who make recordings that are, technically speaking, wretched trash. Good mastering professionals are acquainted with many styles of music, and know what it is that their clients hope to achieve. They then use their tools either lightly or severely to accomplish all the multiple steps involved in preparing a disc for pressing

What is a near-field monitor?

A near field monitor is one that is design to be listened to in the near field. Simple, eh? The "near field" of a loudspeaker is area where the direct, un-reflected sound from the speaker dominates significantly over the indirect and reflected sound, sound bouncing off walls, floors, ceilings, the console. Generally the center third of a room. Monitoring in the near field can be useful because the influence of the room on the sound is minimized. Near field monitors have to be physically rather small, because you essentially need a small relative sound source to listen to (imagine sitting two feet away from an 18" woofer and a large multi- cellular horn!). The physics of loudspeakers puts severe constraints on the efficiency, power capabilities and low frequency response of small boxes, so these small, near-field monitors can be inefficient and not have the lowest octave of bass and not play too loud.

What is a Plug-in?

We could think of a plug-in as an optional �mini� program that runs inside a required �host� program, to add new features and functionality to the existing host program or device. There are many types of plug-ins now available. There are plug-ins for computer audio software that have standard formats or compatibility, such as Direct-X (Windows platform), VST (Steinberg�s Virtual Studio Technology for Mac and Windows), TDM (Digidesign Pro Tools), and other less popular formats. There are also hardware devices that accept hardware/software combination plug-ins, such as the Mackie d8b (Digital 8 Buss mixer) line of plug-ins that require a DSP card, the Yamaha AW4416 and AW2816 recorders that accept the Waves Y56K plug-in DSP card, and the TC Electronic VoicePrism that accepts the VoiceCraft modeling card to name just a few.

So can I just buy a plug-in and make it work? It if were only that simple. You must first have a compatible �host� program.

Can you give me an example? Sure. Let�s take Cakewalk�s Sonar program for example. Sonar is a program that allows the user to record multiple tracks of both audio and MIDI information. The product information on the box says that it is �Direct-X� compatible, so that means we can invest in a Direct-X effect plug-in, like the Antares Autotune plug-in and use it on an audio track that is recorded in Sonar to correct someone that has sung out of tune (that has never happened to you has it?).

Do I have to use Sonar to use Direct-X plug-ins? No, but you must use a program like Sonar, such as Sound Forge, Cool Edit Pro, or other host program that can use Direct-X plug-ins.

Can I use a VST plug-in with a Direct-X program such as Sonar? Strictly speaking, the short answer is no. There are exceptions to the rules, but you should use VST plug-ins with programs like Cubase and Wavelab for the best results.

Can I use a TDM plug-in with a Direct-X compatible program? No

As you are starting to realize, there are a lot of so-called �standards�, and the staff at IRC Audio is trained to help you through the maze of plug-in possibilities. Please contact us for more information.