Bridging the Gap

One can't help but notice the increasing number of 24-bit, 96 kHz-capable digital audio devices flooding the pro-audio market these days. The unprecedented
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Most people who currently work with high-resolution digital audioprefer using the 96 kHz sampling rate and 24-bit word lengths for theirproductions. Roger Talkov of DVD Labs cites this as the most commonformat for both stereo and surround-sound DVD-A (DVD-Audio) discs.That's mostly because 96 kHz is a little easier to work with if yourproject will also include video content. Read on to learn why this isso.Despite all the concerns a couple of years ago about whether DVD-Adiscs would play in early-model DVD-Video players, the DVD disc hasturned out to be fairly universal. (Of course, a DVD disc won't play ina CD player.) DVD discs can play in either DVD-Audio or DVD-Videoplayers because audio content is stored in a different formatcompatible with each type of player and on separate portions of thedisc. Many DVD-Video discs use the Dolby AC-3 compression format forencoding their audio content, and the Dolby digital encoder wants to"see" a 48 kHz sampling rate to do this work. Converting a 96 kHzsampling rate to AC-3 is a little easier than converting a rate of 88.2kHz because 96 is an integer multiple of 48 (2 × 48 = 96). Thatsimpler processor "math" should theoretically result in a more pristineconversion. That's why you're better off using a 96 kHz sampling rateif your DVD project will include video content.Although the DVD-A spec also includes support for 176.4 and 192 kHzsampling rates, and all DVD-Audio players are capable of playing audioat those rates, they have very rarely been used to date.In some cases, a studio might be planning to release a project onboth DVD and CD. If only one mix will be produced for both releaseformats, the engineer might choose to mix at the 88.2 kHz samplingrate. Because 88.2 kHz is an integer multiple of the CD-format's 44.1kHz rate, downsampling an 88.2 kHz mix for CD release shouldtheoretically cause less degradation than downsampling anon-integer-multiple 96 kHz mix. Accordingly, Talkov sometimes receives88.2 kHz masters for DVD authoring. Gateway Mastering and DVD, on theother hand, reports that it rarely receives 88.2 kHz DVD-A files toaccommodate conversion to Red Book CD format. "In fact," Gateway'sLance Clark says, "I think we've done only one DVD-A disk at 88.2kHz."Downsampling theory notwithstanding, Talkov says that "there areexcellent sampling-rate converters that will take 96 kHz to 44.1 kHzokay." Bob Katz of Digital Domain, however, takes a stronger stance onthis subject. He asserts that, considering the pristine performance ofthe best sampling-rate converters available today, there is nopractical reason ever to use the lower 88.2 kHz sampling rate in lieuof 96 kHz. "We're talking one flea's worth of audible difference"between 88.2 and 96 kHz files, Katz says. "Still, I've been pushing for96 kHz. It seems to be a touch warmer, but only because the filter isjust a little bit farther out." Preferences aside, Katz also regularlyaccepts 88.2 kHz files for CD mastering.

One can't help but notice the increasing number of 24-bit, 96kHz-capable digital audio devices flooding the pro-audio market thesedays. The unprecedented sonic performance that high-resolution digitalmixers, recorders, converters, and DAW I/O boxes promise has manyowners of project studios and small-commercial studios ponderingwhether they should upgrade their gear to the emerging standard. Butbefore you whip out your credit card, there are some major issues toconsider.

For one thing, just because two pieces of equipment are "24/96"(that is, 24-bit, 96 kHz) capable doesn't mean that they can be usedtogether. They may, in fact, each support a different and thereforeincompatible data format, thus preventing them from talking to oneanother. Many products also demand trade-offs in functionality —for example, a halving of the maximum number of available tracks orchannels — in return for working at higher sampling rates. Andbefore you begin production, you'll need to consider which format touse to deliver your high-resolution mixes to the mastering or authoringhouse.

That, in turn, brings up another question: whatend-user/mass-duplication format will your musical masterpiece bereleased on? If your production is destined for release on CD, will youeven hear the sonic benefits of high-resolution recording and mixingonce your masters have been downsampled to the CD's 16-bit, 44.1 kHzformat? Considering all the hoops you'll need to jump through toproduce a 96 kHz recording (not the least of which is monetary), you'llwant to be sure that your efforts will result in a superior-soundingend product.

In this article, I'll explore many of the issues that you'll need toresolve in the course of cobbling together a 96 kHz-capable studio andproducing a high-resolution master. Most of what I'll discuss appliesequally to working at an 88.2 kHz sampling rate, by the way, but forthe sake of simplicity I'll refer mostly to 96 kHz digital audioproduction. (For a discussion about when to use an 88.2 kHz instead ofa 96 kHz rate, see the sidebar "Which SamplingRate Should I Use?")

The touted benefits of working at high sampling rates are clearlycontrovertible. Indeed, some industry notables contend that the emperorhas no clothes. Before rushing headlong into outfitting your studiowith expensive 24/96 gear, you should consider the issues carefully.Let's get started with a quick review of digital audio theory.

RATE HIKE

The Nyquist theory states that the highest frequencies that adigital audio converter can capture or reproduce are equal to half thesampling frequency (also known as the sampling rate). In reality, thecaptured bandwidth is usually a little bit less than half that of thesampling rate, but we're niggling here. For example, a 48 kHz samplingrate should theoretically be able to capture audio frequencies as highas 24 kHz. Thus, a 96 kHz rate should be able to capture twice thebandwidth afforded by a 48 kHz rate, that is, audio frequencies up to48 kHz.

In order to avoid aliasing tones (unmusical frequencies) frombeing generated and reproduced by the converter, a steep digitallowpass filter is typically inserted before the converter's output atapproximately the Nyquist frequency, or half the sampling rate at thetop end of the captured audio band. These steep filters are notoriousfor generating their own artifacts, such as phase shift and passbandripple (low-level echoes or ringing). Using a higher sampling rateallows you to move the filter higher because the Nyquist frequency alsomoves higher. With the filter placed far beyond the audible range, amore gradual roll-off can be used, which results in fewer artifactsbeing generated.

Some proponents of 96 kHz digital audio contend that it's thegentler filter that is responsible for the improvement in sound qualityproduced at high sampling rates. However, Richard Elen, vice presidentof Marketing at Apogee Electronics, maintains that this is a ridiculousargument because oversampling at 44.1 or 48 kHz already accomplishesthe same thing — that is, it also allows for use of a gentlerfilter. "These days," Elen explains, "you never actually sample at yourreal sampling rate because converters do oversampling. Oversamplingruns your clock at a multiple of the actual sampling rate. Because youreffective sampling rate is so high, half your effective sampling rateis also ultrasonic. That means that all your filters can be way upthere, as well. All problems with Nyquist go away. Any ringing orphase-shifting effects caused by the filters will be long gone by thetime you get down to the audible range of the audio band. That said,there definitely are improvements that most people hear when listeningto high-quality 24/96 conversion systems versus high-quality 24/48conversion systems."

YOU ANIMAL, YOU

Though many people assume that the improved sonics of 96 kHz digitalaudio can be attributed primarily to the extended frequency responseafforded by the higher sampling rate, Bob Katz, president of DigitalDomain (a CD mastering house located in Altamonte Springs, Florida),views the situation a bit differently. "Some people think human beingscan hear like dogs and bats," he says, "but in fact we haven't suddenlydeveloped supersonic hearing." While Katz agrees with Elen about thebenefits of oversampling, he points out that high-resolution mastersdestined for CD release will be downsampled to 44.1 kHz rate at somepoint and thus will have a steep filter and sampling-rate converterapplied. "This means there's one more filter in the chain," Katz says,"and it will add its own phase shift, ripple, and other artifacts. Youcan oversample it all you want [during subsequent playback], but stillyou've added an additional filter."

Keith Olsen, corporate director of Global Market and ProductDevelopment for Mackie, recalls attending an A/B listening test thatpitted 96 kHz conversion against 48 kHz, using the same converter.(Readers may recall that Olsen produced many hit records for FleetwoodMac, Foreigner, Pat Benatar, Rick Springfield, Sammy Hagar, Santana,and others before joining Mackie.) "It sounded like the top end [with96 kHz conversion] was not as harsh," says Olsen. "But the amazingthing was that upsampling [that is, converting a previously sampled 48kHz recording up to the 96 kHz sampling rate] sounded just as good! Icouldn't tell the difference."

Katz maintains that Olsen's experience proves his contention thatthe sonic benefits of 96 kHz sampling are due to gentler filtering, andnot to extended bandwidth. He points out that upsampling cannot addfrequencies that weren't captured in the first place by alower-resolution A/D converter. "If 50-year-old ears — whichtypically cannot hear above 15 kHz — can hear the differencebetween 44.1 kHz sampling and 192 kHz sampling," says Katz, "thenobviously what is going on is a reduction in artifacts in the audibleband."

In fact, Katz routinely upsamples 44.1 kHz material to 96 kHz whenmastering material in his studio for CD release. "When using nonlinearprocessing (such as compression) at a higher sampling rate," says Katz,"distortion in the audible band is reduced by at least 3 dB, even iffollowed by a sampling-rate conversion back down to the lower rate." Inother words, if you're going to use digital dynamics processing,upsampling the material will reduce the amount of audible distortion inthe final file.

Despite his love for the sonics produced by 96 kHz sampling, Katzthinks such a high conversion rate might not be necessary if impeccabledigital-filter designs were employed in converters. He also notes thatupgrading the digital filters in converters would be considerably lessexpensive than retooling studios for ever-higher sampling rates.

Elen concurs. "There is a point at which multiplying by largenumbers, whether it's high sampling rates or high numbers ofoversampling, is done to impress people — in terms of sellingthings — and not for audio quality," Elen says. "But after acertain point, you're not going to be able to hear the differenceanymore."

NOT HEARING IS BELIEVING

Though Elen, Katz, and Olsen differ regarding why 96 kHzconverters sound better than lower-resolution ones, they all agree thatthe difference is audible. But Paul Lehrman, a composer, educator, andconsulting editor for Mix magazine, feels that most listenersare unlikely to hear the difference. He contends that "99.999 percentof the people who listen to recordings are not in a position toperceive any difference between a 96 kHz recording and a 44.1 or 48 kHzrecording. I think that whatever advantages you get out of 96 kHz arefar overshadowed by the limitations of the transducers at both ends ofthe signal chain."

In defense of his position, Lehrman points out that the frequencyresponses of most mics and digital musical instruments roll off ataround 20 kHz. Thus, anything recorded above 20 kHz at a 96 kHzsampling rate "is probably junk," claims Lehrman. In response to theargument that it's the digital filter in 96 kHz systems, and not theextended frequency response, that's responsible for the improvedsonics, Lehrman says that, in A/B tests, he has "never been able totell, definitively, the difference between a well-constructed 44.1 or48 kHz oversampling converter and a 96 kHz converter."

Engineer, producer, and EM contributing editor Larry the Oweighs in somewhere between 96 kHz cheerleaders and naysayers. "Thereis some extra sparkle at 96 kHz; it does make a difference," he says."But it's nothing compared to the difference between 24-bit and 16-bitdigital audio."

Naturally, you should listen yourself and draw your own conclusionsabout whether 96 kHz digital audio sounds better than, say, 48 kHz.Assuming you hear enough of a difference to compel you to take theleap, you'll then need to determine which high-resolution digital audioformats are being used by currently available products and whattrade-offs, if any, each format requires.

SPLIT DECISION

The various stereo and multichannel digital audio formats that havebeen in use since the 1980s were not originally designed to pass 24/96audio. In some cases, such as with AES/EBU, the spec had enoughbandwidth that it could be formally rewritten to accommodate highersampling rates over existing cables without much ado. In other cases,such as with the 8-channel ADAT Lightpipe and TDIF formats,sample-splitting schemes had to be developed by manufacturers toallow their equipment's multichannel I/O to pass 24/96 digitalaudio.

Sample splitting is a process that splits, typically, one 96 kHzdigital signal into two 48 kHz signals (or one 88.2 kHz signal into two44.1 kHz signals), thus allowing the split signal to be sent along twochannels rather than one. Because sample splitting uses two channelsfor transmitting each original channel of 24/96 audio, the number ofavailable channels that the ADAT and TDIF I/O can transmit at 96 kHz iscut in half, to four rather than eight channels. If this seemscumbersome, limiting, or confusing, bear in mind that sample splittingis a transitional strategy meant to enable high-resolution digitalaudio production using entrenched connectivity technologies. Ratherthan wait until a better solution is developed, we can employ theseinterim formats to create multichannel 24/96 productions now.

Similar to sample splitting, bit splitting is anencode/decode process that breaks up a 24-bit, low-resolution (44.1 or48 kHz) digital word into two data streams — one 16-bit, theother 8-bit — and records them onto two tracks of a 16-bitrecorder. The two tracks are recombined on playback into one 24-bitstream by a device that decodes the split bit stream. Note, however,that the use of bit-splitting formats is fading as 24-bit devicesbecome the norm.

HELLO, GOOD-BYE

All 24/96 data formats require a chip on the sending and thereceiving end of the transmission chain that recognizes what protocolis being used. Without the chip in both pieces of connected gear, theunits cannot talk to one another. Before outfitting your studio with24/96 gear, make sure all the pieces that you're considering buying canspeak the same language. If they can't, you'll hear nada.

In many cases, 24/96 operation is available only for some of thesupplied I/O on a given piece of equipment. For example, the MackieMDR24/96 modular hard-disk recorder (M-HDR) can record 96 kHz digitalaudio by way of its Lightpipe connections, but its AES/EBU connectionscan handle only 44.1 and 48 kHz sampling rates (at the time of thiswriting, anyway).

The point is, you can't determine that two pieces of gear willprovide inter-operable 24/96 capabilities simply by checking to see ifthey offer the same type of I/O. The best way to determine 24/96compatibility is to ascertain that both pieces of gear support the samehigh-resolution data formats. Unfortunately, a comprehensive summary ofsupported 24/96 data formats is difficult to come by for most currentlyavailable products, and usually requires digging beyond promotionalliterature, spec sheets, and the like. (For a quick-reference guide tosome popular 24/96 products and the high-resolution formats that theysupport, see the table "Resolution Conflicts.")

In addition to checking compatibility of data formats, make sure anyproducts you plan to use together also support the sampling rate(s) youwant to work with and can sync to a common word-clock rate. Somerecorders must receive 48 kHz word clock (which is doubled internally)to record 96 kHz digital audio. In such a case, you will need a devicelike the Swissonic AD96 mk2 4-channel A/D converter (see Fig. 1)that can output half the word-clock rate (44.1 or 48 kHz) to yourrecorder's word-clock input when working with 88.2 or 96 kHz audio-datasampling frequencies.

Finally, if you are working on a DAW, make sure you have tons ofhard-disk storage available: 24/96 data demands a lot of diskspace!

Next, let's take a look at the data formats that currently are beingoffered in 24/96 gear. The following is not meant to be an exhaustivelist of 24/96-capable gear; it is offered simply to give you an idea ofwhat's available and what's compatible.

DIVIDE AND CONQUER

S/MUX

S/MUX is a sample-splitting technology developed by Sonorus thatsplits one channel of 24/96 digital audio into two 24-bit, 48 kHzchannels (or splits one channel of 24-bit, 88.2 kHz digital audio intotwo 24-bit, 44.1 kHz channels) for transmission over ADAT LightpipeI/O. The new MOTU 2408mk3 DAW I/O box (see Fig. 2), theSwissonic AD96 mk2, and the Apogee AD16 and DA16 converters all supportthe S/MUX format through Lightpipe I/O. Additionally, the Yamaha 02R96digital mixer, the Alesis ADAT HD24 M-HDR, and Mackie's HDR24/96,MDR24/96, and SDR24/96 M-HDRs all use an S/MUX-compatible format whentransmitting 88.2 or 96 kHz digital audio over their Lightpipeconnections. (Yamaha, Alesis, and Mackie's high-resolution Lightpipeformats may, in fact, be identical to S/MUX format; exactspecifications were not available.) Let's take a closer look at howsome of these products implement their own form of S/MUX to record andplay back 24/96 audio.

As one would expect from units using a sample-splitting scheme, theabove-mentioned Alesis and Mackie hard-disk recorders all have theirmaximum track counts cut in half when shuttling high-resolution digitalaudio over Lightpipe I/O — from a maximum of 24 tracks at 44.1 or48 kHz to 12 tracks maximum at 88.2 or 96 kHz. The new Yamaha 02R96mixer (see Fig. 3), however, distinguishes itself as one of thefew products on the market that does not lose any channels whenoperating at 88.2 or 96 kHz sampling rates over Lightpipe I/O. Eachrecombined 24/96 track (returning from an MHDR, for example) shows upon one 02R96 fader, and the mixer's 56 simultaneous input-channelfaders are available no matter what sampling rate you use.

MOTU's 2408mk3 also loses half its maximum number of availablechannels on each digital bank when using sample-splitting schemes suchas S/MUX and 96 kHz TDIF. (I'll discuss TDIF in a moment.) So, forinstance, each bank of Lightpipe I/O can transmit only four channels of24/96 audio in S/MUX mode. Thankfully, though, the 2408mk3 records each24/96 (or 24/88.2) channel of S/MUX- or TDIF-format audio to onetrack in Digital Performer (DP). (All MOTU I/O boxes are compatiblewith Mac and PC DAWs and will also work with other applications besidesDigital Performer.) On playback, each DP track becomes just one 96 kHzstream that can be sent out one analog output, in any supported formatyou wish, on the 2408mk3. That keeps operations simple anduser-friendly. Moreover, the 2408mk3 can perform real-timeformat conversion between its inputs and outputs.

For some readers, it's important to note that the Alesis HD24 canonly sync to its Lightpipe inputs when recording digitally at an 88.2or 96 kHz sampling rate. That is, the HD24 cannot sync to its BNCword-clock input when receiving 88.2 or 96 kHz digital audio over itsfiber-optic lines, but must instead sync to the clock embedded in theaudio bit stream. Lightpipe has a reputation for being jittery, andfinicky engineers may wish to evaluate whether the benefits ofrecording high-resolution digital audio over Lightpipe justifies theassumed trade-off in jitter performance. Fortunately, Alesis alsooffers an optional analog I/O board for the HD24 that can record at88.2 or 96 kHz while synced to the unit's internal clock. All three ofMackie's hard-disk recorders (mentioned above) can sync to theirrespective word-clock inputs when recording through their Lightpipe I/Oat any sampling rate, including 96 kHz.

TDIF

As is the case with S/MUX for lightpipe-equipped devices, recording96 kHz digital audio over TDIF lines usually requires two channels ofstorage for every 24/96 channel transmitted. For M-HDRs, that alsoresults in a track count reduced by half. Most mixers that support24/96 operation over TDIF I/O, such as the Tascam DM-24, also loseaccess to half their faders when operating in this mode (but,thankfully, each 24/96 channel shows up on only one DM-24 fader).

Things are a bit less cumbersome when using TDIF in DAW land: theMOTU 2408mk3 I/O box records each 24/96 digital audio channel receivedover its TDIF lines to one track in Digital Performer. Note thatalthough the Apogee AD16, Rosetta 96, and PSX-100 converters are24/96-capable and include TDIF connections, they cannot handle 24/96operation over TDIF lines.

Apogee ABS96

A proprietary format developed by Apogee Electronics, Apogee ABS96combines bit-splitting and sample-splitting techniques to allow you torecord two channels of 24/96 digital audio on an 8-channel, 16-bitrecorder. The Apogee Rosetta 96 A/D and PSX-100 A/D/A converters bothoffer ABS96 mode. You'll need the latter unit to decode any tracks thathave been encoded in ABS96 format.

2-CHANNEL FORMATS

Double-wire AES

Also known as double-wide AES, double-wire AES mode sends onechannel of 88.2 or 96 kHz digital audio down each AES/EBU cable. Thus,two connectors are required for a stereo signal, whereas only oneconnector is needed to send two channels of 24-bit, 44.1 or 48 kHzaudio. Use of the double-wire AES format is not very common anymore,because an AES/EBU spec was formalized for "single-wire/double-speed"(88.2 or 96 kHz) transmission of two channels down one AES/EBU cable.That said, Tascam's DM-24 mixer, DA-98HR 8-track recorder, and MX-2424SE hard-disk recorder; Benchmark's AD2402-96 A/D converter; PrismSound's Dream AD-2 A/D converter; and Apogee's Rosetta 96 and PSX-100converters all support double-wire AES mode in addition to thenow-standard single-wire/double-speed AES mode. (The Tascam MX-2424 SErequires an option card to enable double-wire AES operation.)

Single-wire/double-speed AES

Like the regular AES/EBU format that we've always used, thesingle-wire/double-speed mode sends two channels of 24/96 audio downone connector, but at twice the usual sampling rate. Products thatsupport single-wire/double-speed AES mode include the Yamaha 02R96 andTascam MX-2424 SE digital mixers; MOTU's 896 and 1296 DAW I/O boxes;Digidesign's 192 Digital I/O, 192 I/O, and 96 I/O boxes; Lucid'sAD9624, Swissonic's AD96 mk2, Benchmark's AD2402-96, dB Technologies'AD122-96 MK.II, Sek'd's ADDA 2496 S, Prism Sound's Dream AD-2, andSonifex's Redbox converters; and Apogee's Rosetta 96, PSX-100, Trak2,Mini-Me, and AD16 (with optional card) converters.

Double-speed S/PDIF

Double-speed S/PDIF is an unbalanced version of thesingle-wire/double-speed AES format, except that the S/PDIF flavor alsouses a different voltage and impedance than AES/EBU. Products thatsupport the double-speed S/PDIF format include the Yamaha 02R96; theTascam MX2424 SE; the MOTU 2408mk3; Digidesign's 192 Digital I/O, 192I/O, 96 I/O, and Digi 002; the MAudio Duo USB mic preamp; and the PrismSound Dream AD-2, the Swissonic AD96 mk2, the Sek'd ADDA 2496 S, theBenchmark AD2402-96, the Sonifex Redbox, and Apogee's Rosetta 96,PSX-100, Trak2, and Mini-Me converters. The Edirol UA5 and UA-700 DAWI/O boxes also both support 96 kHz, but not 88.2 kHz, I/O over S/PDIFlines.

COMPUTER CONNECTIVITY

IEEE 1394

Also known as FireWire, IEEE 1394 is an industry-standardspecification developed by the Institute of Electrical and ElectronicsEngineers (IEEE) for connecting consumer audio and video devices toeach other and to computers. However, companies such as MOTU andDigidesign also use IEEE 1394 for connecting their professional I/Oboxes to DAWs. The FireWire interface on the Power Macintosh G3 and G4uses Apple's implementation of IEEE 1394. The original FireWireprotocol, IEEE 1394a, provides 100 to 400 Mb per second (Mbps)bandwidth. A new implementation of FireWire, dubbed IEEE 1394b, is justaround the corner and will provide up to 3.2 Gb per secondbandwidth.

The MOTU 896 and Digidesign Digi 002 both use IEEE 1394a/FireWirefor bidirectional 24/96 digital audio connectivity with acomputer-based DAW.

USB

USB is another type of bus used to get digital audio (and MIDI) datain and out of a computer. All of the USB-based digital audio devicesthat I'm aware of use the original USB spec, which provides 12 Mbpsbandwidth. By the time you read this, however, devices using the newHi-Speed USB 2.0 protocol should be out. USB 2.0 provides 480 Mbpsthroughput, which should dramatically increase track counts as comparedwith current USB capability.

The M-Audio Duo and Quattro and the Edirol UA-5 and UA-700 areexamples of I/O boxes that offer 24/96 audio transmission over USB toand from computers. Due to USB's bandwidth limitations, each of thesefour products can deliver only two simultaneous tracks at 96 kHzsampling rate, and none of them can record and play back trackssimultaneously. Although the USB-based Apogee Mini-Me can output 24/96audio by way of its AES/EBU and S/PDIF jacks, the unit's USB port candish out only 44.1 and 48 kHz rates.

From the above discussion of formats (and referring to the table"Resolution Conflicts"), we can see that the Yamaha 02R96 mixer andTascam DA-98HR recorder cannot work together in 24/96 mode. Neither theMackie HDRs nor the MOTU 896 and 1296 I/O boxes can communicate withthe Tascam DM-24 mixer in 24/96 operation. And neither the DA-98HR northe DM-24 can receive 24/96 digital audio from the dB TechnologiesAD122-96 MK.II, the Lucid AD9624, the Sek'd ADDA 2496 S, the SonifexRedbox, the Swissonic AD96 mk2, or the Apogee AD16, Trak2, or Mini-Meconverters. Clearly, it pays to confirm interoperability of all 24/96gear that you're interested in before you buy.

Now that you have a handle on the formats currently available for24/96 digital audio production, let's examine what you need to knowabout high-resolution delivery formats for your masters.

PREPARING YOUR MASTER

If you decide to dive in to 24/96 digital audio production, you'llneed to know what delivery formats mastering houses and authoringfacilities can accept. Many DVD-mastering and production facilitiesprefer that you send them a copy of your mixes on a hard drive. RogerTalkov, president of DVD Labs (a DVD-Audio and DVD-Video productionhouse located in Cambridge, Massachusetts), notes that delivering yourproject on a hard drive "precludes the need to restore [from themasters you provided] and back up. We also get Retrospect backups onQuantum DLTs." DLT (digital linear tape) can hold up to 80 GB of dataon a single cartridge.

DVD Labs can also accept AIFF or WAV files on CD-ROM, DVD-ROM, orAlesis Masterlink discs. For surround-sound projects, which containmultiple tracks for each mix, Talkov emphasizes the importance ofmaking sure that all of your files are saved on your DAW timeline sothat they play together in sync.

Lance Clark, multimedia engineer at Gateway Mastering and DVD (inPortland, Maine), notes that most of the masters he receives arrive as24/96 AIFF or WAV files on a hard drive. He also gets a lot of mastersdelivered on Tascam HR tape (the format used by Tascam'shigh-resolution, tape-based MDMs). Clark also notes that Gateway canalso accept CD-ROMs, but the company does not receive many CD-Rs inMasterlink format. Although Gateway also masters projects for SACD (acompeting high-resolution format to DVD-Audio), Clark observes thatsuch work is currently "slim." He says that the number of projects thatGateway masters for SACD release are perhaps only 1 in every 10 or 12projects it receives.

Digital Domain's Katz says that he often receives 24/96 and 24/88.2files, even though he masters for 16-bit, 44.1 kHz CD release. Heoccasionally receives mixes on hard disks in Pro Tools format, "but 99out of 100 projects that come to me have been on CD-R." Katz notes thatan entire album's worth of 24/96 stereo mixes will typically fit ontothree or four CD-Rs. He calls Masterlink a "fantastic delivery format"for this purpose, but he is also comfortable using any CD-ROMcontaining AIFF or BWF-type WAV interleaved files. (Broadcast WAVFormat, or BWF, adds time-stamping to regular WAV files.) Because hehas encountered incompatibilities among various DVD writers andreaders, Katz does not recommend putting your files on DVD-R at thistime.

BIT (OF) RESOLUTION

As this article has made clear, many currently available24/96-capable products suffer substantial trade-offs in functionalityin return for promised higher fidelity. If you've had the chance toaudition 24/96 audio gear and you like what you hear, the only questionthat remains (besides affordability) is whether you can accomplish yourgoals with it.

As Larry the O says, "with most of the 96 kHz devices in theEM readership's price range, all of your facilities are halvedas soon as you go to 96 kHz. In the case of mixers, you no longer haveenough channels to do a mix of any level of complexity. Just doing bassand multimiked drums will use up most of your available channels at 96kHz, leaving no room for guitar, vocals, or whatever. If you're doing asmall ensemble like a jazz trio where you're only using one or two micson the drums, then you might be able to do 96 kHz." Of course, as moreproducts such as the Yamaha 02R96 (which retains its full complement ofchannel faders in 88.2 and 96 kHz modes) become available, the O'sconcerns will become moot. Looking even further into the future, TDIFand Lightpipe protocols could easily fade away as more people migrateto DAWs equipped with FireWire.

"Doing bit-splitting techniques and doubling up on things," Elensays, "you end up with a lot of real estate being taken up byconnectors. Today you have one little connector [FireWire] that willhandle 400 mbps." And with the imminent arrival of 1394b, one can'thelp but wonder how long it will be before all the digital audiodevices in our studio will be connected by FireWire into one largepeer-to-peer network.

Make no mistake: our industry is in the midst of a radicaltransition. Only you can decide when is the right time — if ever— to dive in. If and when you do, hopefully this article willhave armed you with the information necessary to make informeddecisions in this brave new world of 24/96 digital audio.

Michael Cooperis anEMcontributing editorand owner of Michael Cooper Recording, located in beautiful Sisters,Oregon.

RESOLUTION CONFLICTS

This table shows a sampling of currently available high-resolutiondigital audio products and the 24/96 data formats they support(indicated by a - mark). As long as they offer compatible samplingfrequencies and word-clock rates, any two products that support thesame format should be able to work together in 24/96 operation (withthe possible exception of products that sport FireWire or USB ports,which are included here only as a handy reference to show which DAW I/Oboxes support 24/96 computer connectivity).

The I/O that a product provides is implicit in its supportedformats. For example, a product that supports S/MUX format will provideADAT Lightpipe I/O. Some products may require an option card to enablea particular format that is noted here as being supported.

RESOLUTION CONFLICTS

This table shows a sampling of currently available high-resolutiondigital audio products and the 24/96 data formats they support(indicated by a - mark). As long as they offer compatible samplingfrequencies and word-clock rates, any two products that support thesame format should be able to work together in 24/96 operation (withthe possible exception of products that sport FireWire or USB ports,which are included here only as a handy reference to show which DAW I/Oboxes support 24/96 computer connectivity).

The I/O that a product provides is implicit in its supportedformats. For example, a product that supports S/MUX format will provideADAT Lightpipe I/O. Some products may require an option card to enablea particular format that is noted here as being supported.

S/MUX or
compatible
format

96- kHz TDIF

Apogee
ABS96

Double-wire
AES

Single-wire/
Double-speed
AES

Double-speed
S/PDIF

Fire
Wire

USB

Alesis ADAT HD24

¯

Apogee AD16

¯

¯

Apogee DA16

¯

Apogee Mini-Me

¯

¯

Apogee PSX-100

¯

¯

¯

¯

Apogee Rosetta 96

¯

¯

¯

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Apogee Trak2

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Benchmark AD2402-96

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dB Technologies AD122-96 MK.II

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Digidesign 192 Digital I/O

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Digidesign 192 I/O

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Digidesign 96 I/O

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Digidesign Digi 002

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Edirol UA-5

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Edirol UA-700

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Lucid AD9624

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M-Audio Duo

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M-Audio Quattro

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Mackie HDR24/96

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Mackie MDR24/96

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Mackie SDR24/96

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MOTU 1296

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MOTU 2408mk3

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MOTU 896

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Prism Sound Dream AD-2

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Sek'd ADDA 2496 S

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Sonifex Redbox

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Swissonic AD96 mk2

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Tascam DA-98HR

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Tascam DM-24

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Tascam MX-2424 SE

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Yamaha 02R96

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MANUFACTURERS

ALESIS
tel. (401) 295-9000
e-mail info@alesis.com
Web www.alesis.com

Apogee Electronics Corporation
tel. (310) 915-1000
e-mail info@apogeedigital.com
Web www.apogeedigital.com

Benchmark Media Systems, Inc./Sonic Sense, Inc.(distributor)
tel. (800) 262-4675 or (315) 437-6300
e-mail sales@benchmarkmedia.com
Web www.benchmarkmedia.com

dB Technologies/Lavry Engineering, Inc. (distributor)
tel. (206) 381-5891
e-mail jeff@lavryengineering.com
Web www.dbtechno.com

Digidesign
tel. (800) 333-2137 or (650) 731-6300
e-mail prodinfo@digidesign.com
Web www.digidesign.com

Edirol Corporation North America/Roland Corporation U.S.(distributor)
tel. (323) 890-3700
e-mail edirol@edirol.com
Web www.edirol.com

Lucid
tel. (425) 742-1518
e-mail info@lucidaudio.com
Web www.lucidaudio.com

M-Audio
tel. (626) 445-2842 or (800) 969-6434
e-mail info@midiman.net
Web www.m-audio.com

Mackie Designs
tel. (800) 258-6883 or (425) 487-4333
e-mail productinfo@mackie.com
Web www.mackie.com

Mark of the Unicorn, Inc. (MOTU)
tel. (617) 576-2760
e-mail info@motu.com
Web www.motu.com

Prism Sound/Prism Media Products Ltd. (distributor)
tel. (973) 983-9577
e-mail sales@prismsound.com
Web www.prismsound.com

Sek'd/plus24 (distributor)
tel. (800) 330-7753 or (323) 845-1171
e-mail info@sekd.com
Web www.sekd.com

Sonifex Ltd./Independent Audio (distributor)
tel. (207) 773-2424
e-mail sales@sonifex.co.uk
Web www.sonifex.co.uk

Swissonic/plus24 (distributor)
tel. (800) 330-7753 or (323) 845-1171
e-mail info@swissonic.com
Web www.swissonic.com

Tascam
tel. (323) 726-0303
Web www.tascam.com

Yamaha Corporation of America
tel. (714) 522-9011
e-mail info@yamaha.com
Web www.yamaha.com

WHICH SAMPLINGRATE SHOULD I USE?