Jim Roseberry

Intel vs. AMD

Mon, 01 May 2000 00:00:00 GMT

PIII or Athlon,
That is the question.

Well Shakespeare, I'm here to tell you that in many cases it just doesn't make a damn bit of difference. I'll pull out some numbers in a moment, but the bottom line is that both the Intel PIII Coppermine and AMD's K7 Athlon make exceptional DAW CPUs.

Lineage

Since the birth of the PC DAW, Intel CPUs have been synonymous with top-performance and stability. Having no serious competition over a several year period, end users were forced (for better or worse) into a Wintel world. Windows applications were optimized and tested for use with Intel CPUs - and those Intel CPUs provided FPU (Floating-Point Unit) performance that was heads and shoulders above units from Cyrix and AMD. And so it was…

Here and Now: Serious Competition

After losing the FPU battle for years, AMD went back to the drawing board. The result is their K7 Athlon which packs some serious FPU muscle. Early reports of the Athlon having 2-3 times the FPU performance of an equally clocked PIII were a wee bit exaggerated, but it easily holds its own.

Finally! Intel's competition has a CPU that's a real competitor!

So… how do they stack up???

Stability

I've logged many hours behind both PIII and K7 based systems and find them to be equally stable.

I'll share a couple of tips. If you choose to run a PIII system, use a BX Chipset Motherboard. They have been in use for quite a while now, so they're mature and refined to near perfection.

The AMD K7 Athlon can reportedly be a bit finicky about power-supply and RAM. For what it's worth, I've never experienced any problems, and I've built quite a few machines. If you choose to run an Athlon system, use a UL Listed power-supply and good quality RAM. These are built to tighter tolerance and will reduce the odds of related problems.

Flexibility

The first generation of K7 Athlon motherboards were little more than prototypes (ie: the original FIC SD-11). This board would not allow both ‘master' and ‘slave' devices (on the same IDE channel) to function in DMA mode. You had to choose one or the other. Also, the ‘Super Bypass' option was not available on first generation motherboards. This option results in about a 10% speed improvement. Newer 2nd and 3rd generation motherboards (ie: Asus K7m) don't suffer these limitations.

The UDMA Hard-Disk controllers on (PIII) BX chipset motherboards use about 2% less CPU than (K7) VIA chipset motherboards. For example, a 40 Gig 7200 RPM Maxtor UDMA Hard Drive will sustain 28MB/Sec on both BX and VIA chipset motherboards. A BX board will use about 2.5% of the CPU for disk I/O (as opposed to a VIA board using about 4.5%).

Thus… the PIII has a minute disk I/O advantage.

Audio Performance: to Each One's Own

I wanted to create the definitive Intel vs. AMD audio test using real-world applications. To make the tests as equal as possible, I used identical hardware (other than the CPU and motherboard) in both machines.

The following hardware was present in both systems:
Matrox G400 video card
256 MB PC100 RAM
40 Gig Maxtor UDMA Hard Drive (dedicated to audio files)
Frontier Dakota audio card

I ran what I call a ‘stress-test' in several audio applications running under Win98SE (Cakewalk Pro Audio v9.03, Logic Audio Platinum v4.2.2, Cubase VST/24 v3.71r2, and Samplitude 2496 v5.57a.

The ‘stress-test' consisted of:
- 24 solid 24 Bit channels of 44.1k audio (32Bit Float files in Samplitude 2496 and Sequoia)
- 48 bands of Waves' Renaissance EQ (12 instances - each with 4 active bands)
- Timeworks' 4080L reverb (Aux with 4 channels feeding)
- TC Native Reverb (Aux with 4 channels feeding)
- As many Waves' Renaissance Compressors as remaining CPU would allow

Cakewalk Pro Audio v9.03

PIII 600e – ran above stress-test with 4 additional Waves' Renaissance Compressors

CPU @ 71-78%

Disk @ 10-44%

Athlon 600 - ran above stress-test with 4 additional Waves' Renaissance Compressors
CPU @ 71-76%
Disk @ 10-45%

In both cases, latency was set to 104ms (using Dakota's MME driver) and I/O buffer size was 256.

Cubase VST/24 v3.71r2

PIII 600e – ran above stress-test with 7 additional Waves' Renaissance Compressors

CPU @ 95-98%

Disk @ 90%

Athlon 600 - ran above stress-test with 7 Waves' Renaissance Compressors
CPU @ 95-98%
Disk @ 90%

VST was able to sustain the above load using the Dakota's MME or ASIO drivers.

When using MME: latency was set to 464ms and Disk-block size was set to 256k
When using ASIO: latency was set to 12ms and Disk-block size was set to 48k

In both cases: Audio was given highest priority and the number of output channels was set to 64.

Logic Audio Platinum v4.2.2

PIII 600e – ran above stress-test with 2 additional Waves' Renaissance Compressors

CPU @ 95-98%

Disk @ 30-33%

Athlon 600 - ran above stress-test with 4 additional Waves' Renaissance Compressors
CPU @ 95-98%
Disk @ 30-33%

In both cases, Latency was set to 60ms (using Dakota's MME driver). ASIO isn't quite up to snuff in Logic at this time.

Samplitude 2496 v5.57a

PIII 600e – ran above stress-test with 12 additional Waves' Renaissance Compressors

CPU @ 77-78%

Disk @ 31%

Athlon 600 - ran above stress-test with 12 additional Waves' Renaissance Compressors
CPU @ 62-64%
Disk @ 27-28%

In both cases, Latency was set to about 250ms (using Dakota's MME driver).

Pay the Price… Count the Cost

Athlon 600 w/heatsink-fan is about $170

PIII 600e w/heatsink-fan is about $250

Athlon 800 w/heatsink-fan is about $320

PIII 800 w/heatsink-fan is about $569

Note: Celeron 533a @ 800 MHz is about $189 and will very closely approximate the PIII on all audio applications.

As you can see by the numbers… both the PIII and Athlon have strengths and weaknesses. There isn't much difference between the two CPUs when running Cakewalk, Cubase, or Logic. However, if you are running Samplitude 2496 (or Sequoia), the Athlon is about 14% faster than the PIII.

It was interesting to note that Samplitude2496 was able to run the stress-test (with 32Bit Float audio files) AND *12* instances of Waves' Renaissance Compressor at 62-64% CPU use. That leaves 40% CPU left for more tracks and effects… and this is on a 600MHz CPU. Move up to an Athlon 800MHz or 1GHz… and open the floodgates.

NOTE FOR ATHLON USERS: If you run an Athlon system… avoid installing DirectX 7.0 (stay with 6.1 – already a part of Win98SE). Installing DirectX 7.0 will reduce system performance by 5% when running the stress-test.

Swissonic AD96 and DA96 Converters

Wed, 01 Mar 2000 00:00:00 GMT

When working with digital audio, your system's analog-to-digital (A/D) and digital-to-analog (D/A) converters will ultimately determine the fidelity with which you can capture and play back audio.

This is an area where many people getting started with digital audio get confused. "My converters are the latest 24/96 converters," they say. "Isn't 24/96 what I want?"

Converters: What Goes In Must Come Out

The simple fact is that not all converters are created equal.

True, inexpensive 24-bit converters are improving all the time. True, many newer audio interfaces like MIDIMan's Delta 1010 provide very respectable performance (109dB dynamic range). True, for most purposes, this is more than adequate. However, if you wish to attain top performance, you'll have to go with a dedicated set of A/D D/A converters.

So what's available? Many studio owners want world-class performance, but can't justify spending $3000 for a pair converters - especially when their existing converters are already pretty darned good.

So what's a typical studio owner to do?

For the past couple of months, I've been using a set of outboard Swissonic AD96 and DA96 converters, and wanted to share the experience.

Yodel Ley Eee Who?

Swissonic is a company based in Uznach, Switzerland, with Swissonic America being the US branch. Many folks are probably not familiar with the Swissonic name, but are actually using units designed and built by Swissonic.

When you look at several of the Swissonic products like the AD24 and DA24, you'll probably notice that they look vaguely familiar. This isn't coincidence. Swissonic is the company that actually manufactured A/D D/A converters sold under the Sonorus name.

Swissonic AD96

The Swissonic AD96 is a half-rack 4-channel 24Bit A/D converter with balanced XLR inputs and both ADAT lightpipe and AES/EBU outputs.

The first thing you'll notice about the AD96 is that is has a sturdy steel chassis. It's actually fairly heavy for a half-rack unit. The only moving parts are seven push-buttons and a rocker power-switch. There's not much that can wear out, although the push-buttons feel just a tad fragile.

The AD96's user interface is about a simple as it gets. On the left side of the unit, you'll find four LED peak-meters, each with 16 steps. To the right of the meters, there are push-buttons to select the various options for Meters, Clock Source, Wordclock Output, Output Resolution, Sample Rate, and ADAT Format. There's also a button to Calibrate the converters. Pushing each button cycles through its options, indicated by a green LED.

Let's Run the Numbers

The AD24 offers impressive features and performance. Here's a quick run-down of what you can expect:

True 24Bit A/D conversion
7th order, tri-level delta-sigma converter architecture
4 balanced XLR inputs
LED meters for input monitoring (w/multiple modes)
Low-noise L/C PLL system
BNC Wordclock I/O (1x, 2x, Superclock)
Conversion between Wordclock formats
44.1kHz, 48kHz, 88.2kHz, and 96kHz sample rates
Lightpipe optical output (4-channels; switchable between 1-4 and 5-8)
S/MUX and B/MUX format options
Noise-shaped Dither (switchable between 16, 18, and 20Bits)
Two AES/EBU outputs (XLR)
Dynamic Range: 118dB A-weighted, 113dB unweighted
THD + Noise (20Hz to 20kHz): -100dB @ -1dBFS input, -95dB @ -20dBFS input, -53dB @ -60dBFS input
THD: 0.001% @ -1dBFS input

Using the AD96

One great thing about using the AD96 is that it offers very flexbile and informative metering. When working with digital audio, there's nothing more frustrating than not knowing where the "top" is. You're cruising along, happily recording you next opus, and SPLAT go the converters. Here's where an excellent set of meters are worth a lot more than just pretty eye candy.

The AD96 offers four metering options that allow you to really get the job done right.

-60 / 0dB: From –60dB to -21dB… the meters provide 5dB steps, from –21dB to 0dB… the meters provide 3dB steps

-15 / 0dB:The first green LED (@-60dB) is a ‘channel active' indicator, from –15dB to 0dB… the meters provide 1dB steps

-25 / -10dB: The first green LED (@-60dB) is a ‘channel active' indicator, from –25dB to –10dB… the meters provide 1dB steps, the Red LED is an ‘over' indicator

Overloads: The first green LED (@-60dB) is a ‘channel active' indicator, the Red LED is an ‘over' indicator, the middle LEDs show the number of ‘overs' using a logarithmic scale.

In order to be useful in a variety of situations, the AD96 must be able to lock onto many different clock sources. Four clocking options are available on the AD96 to ensure that it will work in virtually any digital studio: Internal, 1x Wordclock, 2x Wordclock, and Superclock (256x). The AD96 will also present any of these clocks (1X, 2X, and Superclock) on its Wordclock outputs.

The AD96 fully supports all popular bit depths and sample rates. Bit depth is selectable from 24, 20, 18, and 16 bit depths. Dithering and noise shaping is applied to the output when using any of the lower bit depths (20, 18, and 16) to ensure the best possible sound quality. Available sample rates include 44.1 KHz, 48 KHz, 88.2 KHz, and 96 KHz.

So How's It Sound?

In a word, excellent!

The perfect acid test for any set of A/D converters is recording acoustic drums. With their powerful attack and long decay, acoustic drums will highlight any problems with digital audio. I just happened to be in the middle of a massive acoustic drum project when the AD96 arrived, so I had a great opportunity to try out the AD96 in a truly demanding application. For these listening tests I would use a SEK'D 2496 D/A converter unit driving a pair of Mackie HR824 monitors.

I set the AD96 for 24Bit / 44.1k operation and recorded for about 10 minutes. Upon playback, it was quite clear that the AD96 is a world-class unit. The initial transient is where you immediately hear the difference. Lesser units tend to blur or smear the initial transient, reducing the impact of the drum, but transients recorded through the AD-96 are clear and well defined.

As a point of reference, I decided to compare drum recordings made on several sets of 24Bit A/D converters including the SEK'D 2496DSP and 2496s units, the MIDIMan Delta 1010, and the Swissonic AD96. I expected the more expensive units to produce better results, and that was pretty much the case. But what surprised me was that the AD96 actually outperformed the $3000 2496DSP, and was the best of the bunch!

Having determined that the AD96 is a first-class set of converters, I was anxious to record vocals and other instruments so I could further evaluate the character of the converters. After several tracking sessions including vocals, acoustic guitar, electric bass, and electric guitar, I feel confident in comparing the AD96's character to that of Mackie mic preamps. The AD96 won't fatten the bottom end, or enhance the mids or top end, but it will capture pretty much any source with great accuracy.

A Word About B/MUX and S/MUX

B/MUX, better known as bit-splitting, allows 16 bit lightpipe-equipped gear to record 24 bit audio. The 24 bit data is spread across two 16 bit tracks. When used like this, a standard 16 bit ADAT can be used to record 4 24 bit tracks. Of course, when using B/MUX, you'll need a B/MUX compatible D/A unit like the DA96 to decode the B/MUX audio.

S/MUX allows the AD96 to record 88.2kHz or 96kHz audio to typical 44.1 / 48KHz machines. Similar to the way B/MUX spreads 24 bit data across two lightpipe channels, S/MUX outputs 88.2 KHz or 96 KHz audio across two output channels.

I'm not too thrilled with the S/MUX method of recording 88.2/96kHz audio. S/MUX (occupying two recorded channels) is a pain in the ass to deal with when recording into a DAW for editing / mixing, unless you have an S/MUX compatible audio card and audio application. If your audio card and audio application are S/MUX compatible (ie: Sonorus Studi/o and Logic Audio) the two channels are managed (merged) ‘behind the scenes' and appear as a single 88.2/96kHz recorded channel in the application. I much prefer the AES/EBU outputs for 88.1 / 96 KHz audio! But this is a limitation of the ADAT Lightpipe format. Lightpipe doesn't have the bandwidth to transmit 88.2/96kHz audio over a single channel, and S/MUX was developed to circumvent this limitation.

To be fair, S/MUX can breathe new life into an ADAT. Used together with B/MUX, S/MUX will allow an old blackface ADAT to record two channels of 24Bit 88.2/96kHz audio. Audio recorded via S/MUX (at 88.2kHz or 96kHz) sounds exceptional. If you have an old blackface ADAT that you hardly ever use anymore, B/MUX and S/MUX can turn that unit into a first-rate 24/96 mastering deck.

Swissonic DA96

The DA96 is a half-rack 4-channel 24Bit D/A converter. Like the AD96, the DA96 has a sturdy steel chassis and features the same push-button/LED user interface.

The DA96's user interface is laid out exactly like the AD96. On the left side of the unit, you'll find four "activity" LEDs (signal present), four Peak LEDs, "Lock" LEDs for each digital input (ADAT, AES/EBU 1, AES/EBU 2), and error LEDs for each digital input.

To the right of the LEDs, there are push-buttons to select the various options for Wordclock Input, Clock Source, Wordclock Output, Data Source, Sample Rate, and ADAT Format. There's also a button to mute the DA96's output. Pushing each button cycles through its options, indicated by a green LED.

Features and Specs

Like its counterpart, the DA24 offers impressive features and performance:

True 24Bit D/A conversion
Low clock jitter sensitivity
4 balanced XLR outputs (capable of driving over 300m of cable)
Activity and Peak LEDs
Low-noise, dual PLL clocking system
BNC Wordclock I/O (1x, 2x, Superclock)
44.1kHz, 48kHz, 88.2kHz, and 96kHz sample rates
Lightpipe optical input (4-channels; switchable between 1-4 and 5-8)
S/MUX and B/MUX format options
Two AES/EBU inputs (XLR)
Automatic detection of input resolution
Clickless muting
Dynamic Range: 112dB A-weighted, 109dB unweighted
THD + Noise (20Hz to 20kHz): -94dB @ 0dBFS output, -88dB @ -20dBFS output, -49dB @ -60dBFS output
THD: 0.002% @ -1dBFS output

Hands-On

Like the AD96, the DA96 will work easily with a variety of Wordclocks: 1x, 2x, and Superclock. The DA96 can lock to clock sources on the Wordclock input, the ADAT input, or either of the AES/EBU inputs. It can also present any of these Wordclocks on its built-in Wordclock output. This capability allows the DA96 to easily integrate into pretty much any digital studio. The DA96 also incorporates B/MUX and S/MUX decoding, making it the perfect partner for the AD96 when using ADATs. The DA96 also includes a Mute button - great when making connections or when things go wrong. The unit will self-mute when it cannot lock to an input source.

To evaluate the DA96, I listened to recordings of acoustic drums, vocals, and acoustic guitar - all recorded at 24 bit 44.1kHz via the AD96. This was an easy test. All tracks sounded excellent through my Mackie HR824 monitors.

As a point of reference, I listened to the same recordings on several sets of 24 bit D/A converters: SEK'D 2496DSP and 2496s, MIDIMan Delta 1010, and the Swissonic DA96. In this scenario, it was very hard to tell much difference between the four units. They all sounded good. If I had to pick a favorite, it would be a three-way tie between the 2496DSP, 2496s, and DA96. A quick look at the specs reveals that, unlike the A/D units, there is less difference between the various D/A units.

And now for the obvious question: so how do the AD96 and DA96 compare with top-flight Apogee converters?

A/D: I've mixed audio recorded with an Apogee PSX-100 and a Swissonic AD96, and I'd rate the AD96 a single notch below the PSX-100, or Rosetta. The primary difference between the units is that the Apogee units offer UV-22 dithering. The dither on the AD96 sounds fine… but it's not UV-22. In any case, it's very unlikely that you would be dissatisfieed with the sound of the AD96. At this level of sound quality, we're talking about fine shades of grey.

D/A: Simply put, the DA96 easily holds its own against any top-rated D/A converters. It has all the features you'll need and expect, and truly excellent sound. As previously mentioned, if you already have a good set of 24 bit converters in your I/O system, it's unlikely that you'll hear worlds of difference between them and the DA96. On the other hand, if your frame of reference is a blackface ADAT, please prepare to be amazed.

If you're looking for a set of world-class A/D and D/A converters, and can't stretch to the cost of Apogee, the Swissonic AD96 and DA96 should be high on your list.

Riding the Wave/8*24

Thu, 01 Apr 1999 00:00:00 GMT

Drum-roll please…

Gadget Labs (http://www.gadgetlabs.com) redefined the standard of "More-for-Less" with their rock-solid Wave/4, so we've ALL been waiting for the scoop on this piece of gear. A 24-bit audio card with eight channels of balanced +4 analog I/O, built-in MIDI… and a MSRP of $499!

Ladies and Gentlemen: I welcome you to the Overview/Review of Gadget Labs' Wave/8*24.

Features

· Full Length PCI card
· Rackmount patchbox - connects to PCI card via included 2-meter cable
· Eight channels of analog I/O (individually switchable between balanced/unbalanced, and +4/-10) via ¼" TRS connections… Channels 1&2 also feature XLR connectors.
· Supports 8, 16, or 24Bit audio (When recording 16Bit audio, the Wave/8*24 samples at 24Bits and dithers down to 16Bits)
· Supports the following Sample Rates: 11.025, 16, 22.05, 24, 32, 44.1, and 48kHz
· Up to three Wave/8*24 cards can be clock-synced via internal connectors
· Monitor incoming audio from the outputs
· MIDI I/O
· Win 95/98 and NT drivers (PC ASIO coming soon, Mac ASIO planned for 2nd quarter of 1999)
· Includes Cool Edit Pro SE

Options

· Longer (7 meter) cable - connects patchbox to the PCI card
· S/PDIF (coax) daughter card ($129.95)
· ADAT Lightpipe daughter card (also provides BNC Word-Clock input) – available around Summer

System Requirements

· One free PCI slot that can accommodate a full length card (it's a long one)…
· A single IRQ
· Gadget Labs recommends at least a 166MHz CPU (Intel recommended – but not required).

Specs

A/D converters – 24Bit 128x over-sampling – 105dB dynamic range
D/A converters – 24Bit 128x over-sampling – 106dB dynamic range
Frequency Response – 10Hz to 20kHz +/- 0.1dB

Package

I don't normally comment much on the appearance of a piece of gear. But in this case, I'll make an exception.
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Gadget Labs Wave/8*24
The first thing you'll notice when you unpack the Wave/8*24 is the striking turquoise color of the rackmount patchbox. Besides attracting attention, the florescent background contrasts well with the white text... and makes patch chores easy in dim light. The patchbox brings to mind an Alesis QuadraVerb in both size and construction.

Installation:

The Wave/8*24 is VERY easy to install. Pop in the card… connect the patchbox… load the driver… restart your machine… and you're done!

To verify compatibility, I installed the Wave/8*24 on an older P2/266 LX system… and a Celeron 300a (450MHz) BX system. Installation was quick and painless in both cases.

NOTE: If you have IRQ Steering disabled, you'll need to enable it to get the Wave/8*24 to install and operate properly. I discovered this the hard way.

Operation

Driver Panel: The Wave/8*24 doesn't have a bunch of useless bells and whistles, so it is spot-on easy to use. (ie: You won't find a half-assed useless software mixer or other confusing options.) Simply open the driver panel (click on the W8 icon in the lower right corner of your desktop)… set the input/output levels to match the rest of your gear… and get on to the business of making music. Kudos to Gadget Labs for keeping it simple!

Patching: Patching to/from the patchbox is SUPER easy because ALL the audio input/output jacks are on the FRONT of the patchbox. No need to crawl behind the rack with a flashlight to setup your next session! The MIDI input/output jacks are located on the rear of the patchbox.

Monitoring: The Wave/8*24 permits ‘Input Monitoring'… which simply routes (echos) audio from an input channel to its corresponding output channel. This is useful if you wish to monitor signals off the Wave/8*24 while they are being recorded. Input Monitoring can be enabled/disabled independently for each of the Wave/8*24's eight channels (via the Driver Panel).

NOTE: An output channel cannot be used to play back existing audio while it is set for Input Monitoring. In other words, if you have output channel 1 set to monitor a vocal you are feeding to input channel 1 (for recording) you cannot use output channel 1 to play existing audio tracks.

Use with Audio apps: I used the Wave/8*24 with Samplitude 2496 and Cakewalk Pro Audio 8.0. In both cases the Wave/8*24 performed flawlessly. MIDI/Audio sync was tight… and there were no ‘hiccups' during record or playback. The Wave/8*24's drivers are rock-solid…

NOTE: At this time, there is a MIDI/Audio sync problem while running Cakewalk Pro Audio 8.04 and the Wave/8*24. The timing seems to ‘lurch' a bit here and there. Previous versions of Cakewalk (including 8.03) work fine. Gadget Labs is aware of the problem… and are currently working with Cakewalk to solve it.

Documentation: The Wave/8*24 comes with a short (easy to digest) manual. It contains all the info you need… from setup… to tweaking your audio app for proper sync/performance. It amazes me that Gadget Labs is one of the FEW companies that actually documents these settings. This is particularly handy for novice users.

Yeah, Yeah, Yeah. But how does it SOUND?

I wanted to spend some quality time evaluating the Wave/8*24's converters before passing judgement.

I measured the noise-floor of the Wave/8*24 (recorded silence and ran Statistics in Cool Edit Pro) at –104dB.

Sure… you can get an eight-channel set of 24-bit converters with a lower noise-floor, but you're gonna have to open your wallet to the tune of about $1100-$1200 to do so. Considering that –104dB is quieter than most project studio mixers… and the cost of the Wave/8*24 is $499… the noise-floor is acceptable for all but the most demanding applications.

Subjectively, the Wave/*24's converters sound exceptionally good! The bottom end is ‘round' and well defined… The top end is clear and detailed without sounding harsh… Want a point of reference? The Wave/8*24's converters sound subtly yet audibly superior to those on a Yamaha O1v. The bottom end is less 'boomy.' And the top end has more clarity/detail.

Mackie uses the same A/D D/A converters on their Digital 8-Bus Mixer… so it's no big surprise that they sound darn good.

Note: When recording 16-bit audio, the Wave/8*24 actually samples at 24 bits… then dithers back down to 16 bits using an algorithm licenced from Pow-r consortium. This yields better fidelity than simply sampling at 16 bits.

How Did Gadget Labs Pull This Off?

Intelligent Engineering!

The converters are actually on the PCI card, BUT (and this is a BIG but)… the Patchbox uses active electronics to send very low impedance audio signals to the PCI card. Thus, the audio is MUCH less susceptible to noise (hum or hiss) than with a conventional balanced cable. This is the same VLZ technology used in Mackie mixers.

Instead of using conventional DSP (Motorola, etc) to manage audio data flow to/from the PC, the Wave/8*24 uses a custom designed SoundCache chip. Gadget Labs describe their SoundCache chip as being an "audio processing accelerator."

Gripes

The patchbox is powered by a wallwart. Given the cost of the unit, I'm sure many users can live with that.

Bottom Line

If you're a project studio owner who's looking for a 24-bit card with eight channels of analog I/O, I strongly encourage you to consider the Wave/8*24!

For those who need more channels of I/O, you can clock sync up to three Wave/8*24 cards… (24 inputs/outputs!)

As they did with the Wave/4, Gadget Labs have once again set a new ‘Bang for the Buck' standard. It's a More-for-Less winner!

Hey Rob… you aren't getting this unit back!

Frontier Dakota, Montana, and Sierra

Mon, 01 Feb 1999 00:00:00 GMT

It's been a while since Frontier released the world's first Lightpipe audio card, the WaveCenter. Without much hype, the WaveCenter managed to establish itself as a solid ‘workhorse' digital multi I/O audio card… and quietly helped to usher in a new era in PC Audio. At last, PC DAWs could digitally record/transfer 8 simultaneous channels of audio!

That was roughly two years ago. So what has Frontier Design Group been up to lately?

Well, I can say this much, they've done their homework and have come up with a powerful new ‘modular' series of audio hardware.

Dakota

Dakota is Frontier Design Group's new PCI Lightpipe audio card that comes stock with the following features.

Two Optical Lightpipe Input ports - providing 16 simultaneous channels of digital audio input (software switchable to Optical S/PDIF)

Two Optical Lightpipe Output ports - providing 16 simultaneous channels of digital audio output (software switchable to Optical S/PDIF)

Coaxial S/PDIF digital I/O - via an included breakout cable (switchable to AES/EBU)

Two independent MIDI Input and Output ports (32 total channels of I/O) – via an included breakout cable

ADAT sync input

Hardware based Chase-Lock to Timecode!

Incorporates SoDA (SMPTE on Digital Audio) technology – explained below

Supports 8, 16, 20, and 24Bit audio

Supports Sample Rates of 44.1 and 48kHz (dynamically resamples 8, 11.025, 16, 22.05, and 32kHz to both 44.1 and 48kHz - for playback through an ADAT or external converters) – can also Vari-speed and lock to digital audio input at Sample Rates from 39-51kHz

Windows 95/98 drivers (PC and Mac ASIO drivers will follow)

SoDA (not pop) in Dakota: SoDA (SMPTE on Digital Audio) allows any audio Input to be used for receiving SMPTE Timecode and any audio Output to be used for sending SMPTE Timecode. SoDA eliminates the need for dedicated hardware.

Montana - double your pleasure

If you need more Lightpipe channels or more advanced sync options than Dakota provides, add the Montana. Montana merely requires a PCI or ISA slot to sit in (you read that correctly)… and requires NO additional resources (IRQ, DMA, etc)!

Frontier Dakota (back) and Montana (front)
Note the cool switchable PCI / ISA configuration on Montana

Montana adds two additional Optical Lightpipe Input ports - providing 32 simultaneous channels of digital audio input (software switchable to Optical S/PDIF). It also adds two additional Optical Lightpipe Output ports - providing 32 simultaneous channels of digital audio Output (software switchable to Optical S/PDIF). Word-Clock or Video sync Input – both external and internal connections are provided. And, ADAT sync output!

Sierra - Eight is enough

If two MIDI ports aren't enough, swap the stock 2x2 MIDI breakout cable for the rackmount 8x8 Sierra. Installation involves connecting the Sierra to the Dakota via the included 15' cable. NO additional resources are required!!

Eight independent MIDI Input ports (128 channels)

Eight independent MIDI Output ports (128 channels)

SMPTE Timecode I/O – via ¼" jacks

Front panel LED activity indicators for each Input/Output

So let's tally this up. A single IRQ buys you 32 channels of Lightpipe I/O, S/PDIF I/O, ADAT sync I/O, Word-Clock or Video sync, and a rackmount 8x8 MIDI interface with SMPTE I/O.

WOW! Talk about future proof…

Driving Miss Dakota

Installation – Installation of the Dakota couldn't be easier. Pop the card in… install the driver, re-boot the machine… and your DAW is ready to use. To check for potential problems, I installed the Dakota on two different systems. The first system was a P2/266 with a LX motherboard… and the second system was a Celeron 300a (@450MHz) with a BX motherboard. In each case, the Dakota installed immediately. No muss, no fuss. This is the way ALL audio card installations should be!

Manufacturers listen up! Audio cards that are a PITA (pain in the ass) to install are destined for failure.

Installation of Montana is as simple as placing the card in an empty PCI or ISA slot (Montana has two sets of edge connectors - allowing it to be installed in either a PCI or ISA slot.) and connecting a ribbon cable between the Montana and Dakota.

Installation of Sierra? This doesn't even qualify as ‘installation' in my book. You just plug the Sierra's 15' cable into the Dakota (replacing the 2x2 MIDI breakout cable). That's all there is to it!

Mixing in Montana

Being a busy kinda' guy, I like to evaluate new gear by USING it on current projects. I spent the past few weeks doing a lot of mixing… and in this scenario I used Dakota/Montana to connect a digital console, a Tango, and two ADATs to my PC. Sierra was used for Automation and SysEx dumps.

I had Samplitude 2496 running about 18 hours a day editing and mixing… and the Frontier hardware never let me down. Amazing! I never thought I'd need 32 simultaneous ins/outs on my current DAW, but once I had em'… they were ALL used.

Man… just two years ago we were ALL using stereo audio cards! (cue Bob Dylan) "Oh the times… they are a changin'…"

Sequencing in the Sierra

After finishing the mixing projects, it was time to open Cakewalk and give Sierra a REAL workout. When opening Cakewalk for the 1st time (after installing the Dakota) you have to run Wave Profiler to insure proper sync between Audio and MIDI. No big deal, Wave Profiler did its thing and I was ready to roll.

Frontier Sierra

To test sync between Audio and MIDI, I decided to load an existing sequence that contained both audio and MIDI tracks. Upon playback, sync sounded tight. So, I decided to record a couple of additional Audio and MIDI tracks. Again, sync sounded tight. With 8 independent MIDI inputs/outputs (one I/O on the front panel) and dedicated SMPTE I/O, the Sierra is a first rate Pro level MIDI interface.

It's about TimeCode

Dakota provides TRUE hardware based Chase-Lock to Timecode.

In other words, Dakota can lock its Sample Rate to Timecode input. Feed a 30 fps (frames per second) SMPTE signal into Dakota, and Dakota will LOCK to the Timecode source by ensuring that there are 44,100 audio samples for every 30 frames of incoming Timecode.

Thus, you have SMOOTH Chase-Lock sync without increasing the load on your system's CPU. If you've ever experienced the distortion and glitches caused by software based Chase-Lock (The audio app varies the audio's Sample Rate on the fly… or adds/removes samples), you'll REALLY appreciate this feature!

ADAT's a Cool Card

If you've paid close attention to this review, you saw that the Montana adds ADAT sync output to the Dakota…

But what's so special about that?

Well, can you say VIRTUAL BRC? Frontier Design Group can… and that is exactly what they have planned!

Conclusion

This is gonna be a short one.

The Dakota/Montana/Sierra combination worked well straight out of the box. There's not a lot of hype, just rock-solid performance.

If you are in the market for a Lightpipe audio card, Dakota is one of the best.

When you factor in the Montana and Sierra expansion options, you have a complete audio hardware system that can anchor virtually ANY studio.

I predict that Dakota/Montana/Sierra will elevate Frontier Design Group to a new level in the PC DAW world. Even the manuals are excellent. All the information you need is provided in a logical and organized fashion. (Frontier even includes system optimization tips.)

The only ‘Cons' I can think of are that Sierra is powered via a wall-wart… and Dakota currently does not have NT or ASIO drivers.

If you deal heavily with ADATs and or Timecode, the Dakota/Montana/Sierra system is an absolute MUST HAVE!

Visit Frontier Design on the Web at http://www.frontierdesign.com

24Bits 96KHz

Thu, 01 Oct 1998 00:00:00 GMT

A couple of months ago, I detailed Samplitude 2496 (from SEK'D). Since this DAW software supports 24Bit recording at Sample Rates up to 96kHz, and all major DAW software is soon to follow... it makes sense to also review the hardware side of a 24Bit 96kHz system. That brings us to this month's review of the Prodif 96 and the SEK'D 2496s A/D D/A converters (AKA the little guy).

Prodif 96 - The Audio Card

The Prodif 96 is a Stereo Digital I/O card that supports 24Bit recording at Sample Rates up to 96kHz. (To my knowledge, the Prodif 96 is currently the only card supporting Sample Rates up to 96kHz.)

Features

Short PCI card (one of the shortest I've seen)
Plug and Play
Bit Depths of 16/20/24 are all supported
AES/EBU I/O via XLR connections (on breakout cable)
S/PDIF I/O (optical) via Toslink
S/PDIF I/O (electrical) via female RCA connections (on breakout cable)
Onboard 20Bit analog D/A converters
Supports professional and consumer digital formats
Ignores SCMS copy protection
Uses a single IRQ *Requires no DMA channels or I/O Addresses*
Full Duplex (record and playback simultaneously)
Can use multiple (synced) cards
Includes drivers for Win95/98 and WinNT (Mac OS drivers also available)
Red error light - lights up when there is a digital clock problem

Installation

I say this in all honesty when I say that the Prodif 96 is absolutely the EASIEST audio card I've ever installed. No muss no fuss... just pop the card in, turn on your machine, load the drivers, restart your machine. The card is installed! The fact that this card requires no DMA channels or I/O address means that it is a breeze to install in just about any machine. All you need is a single IRQ. One other point worth mentioning is that the Prodif 96 is content to occupy IRQ 9 (some cards have a problem there).

Operation

When the Prodif 96's drivers are installed, an icon is placed on your desktop. Double click on this icon and you will jump to the Device Manager (under Control Pannel). Now... click on the Prodif 96 entry (under Sound, Video and Game controllers). Click on the Settings tab... From here you can select the input source, determine the clock source for the output, select the output format, and verify that the card has locked to an external (input) source.

If for any reason there is a digital clocking problem (usually due to incorrect settings), the Prodif 96's red LED will light up. If all is well and the Prodif 96 is locked to the input source, you should not see the red LED.

I've been using the Prodif 96 with Samplitude 2496, Cakewalk Pro Audio, and Cubase VST. As long as the input and clock settings are configured properly, the card works flawlessly.

Subjective

On the digital side, there isn't much subjective (sound wise) to talk about. Transfers via AES/EBU and S/PDIF (both optical and electrical) produced the expected results. What goes in... is what comes out...

The onboard 20Bit D/A converters are capable of playing audio with Sample Rates up to 96kHZ (88.2 kHz is also supported). These converters sound surprisingly good. Usually, the onboard D/A converters included with digital cards are a bit noisy and are weak fidelity wise. This is not the case with the Prodif 96's onboard D/A converters... I should also mention that the output level can be toggled between -10 and 0dB.

Audio played back from the Prodif 96 appears at ALL of its outputs simultaneously! Thus, multiple devices can be fed audio simultaneously. This is a very flexible little card!

IE: I have a pair of SEK'D 2496s A/D D/A converters patched into the Prodif 96 via AES/EBU, a DAT deck is connected via optical S/PDIF, a Lexicon MPX1 is connected via electrical S/PDIF, and a Tascam cassette deck is connected to the onboard 20Bit D/A converters. Any of the above items can input audio to the Prodif 96 (by simply selecting the input source), and the DAT and cassette decks can simultaneously dupe the Prodif 96's output.

SEK'D 2496s - A/D D/A converters

The SEK'D 2496s is a stereo pair of 24Bit A/D D/A converters that can sample and playback at up to 96kHz. (The 2496s is currently one of only a couple of "24Bit 96kHz" converters, and it is also the most affordable.)

Features

Half Rack enclosure
24Bit simultaneous stereo (two independent channels) A/D D/A
S/PDIF I/O (electrical) via female RCA connections
AES/EBU I/O via XLR connections
Balanced Analog I/O via XLR connections
Analog input levels can be independently adjusted from -10 - +10dB
Analog output levels can be independently adjusted from -10 - +10dB
Supports Sample Rates of 44.1, 48, and 96kHz (A/D can also be synced to the D/A)
Supports professional and consumer digital formats
Red LEDs for the analog inputs - light up 3dB before clipping occurs
Red error light - lights up when there is a digital clock problem

Specs

A/D
Dyamic Range: 113dB (A Weighted)
THD+N: <-100dB @ 0dBFs (20Hz - 20kHz, 1kHz)

D/A
Dyamic Range: 106dB (A Weighted)
THD+N: <-90dB @ 0dBFs (20Hz - 20kHz, 1kHz)

Operation

There aren't too many parameters/settings, so using the 2496s is pretty simple.

In my case, I had the 2496s connected to a Prodif 96 (digitally) via AES/EBU. The 2496s' balanced Analog inputs were being fed by the Direct outs (straight off the mic pres) from 2 channels of a Mackie board, and the 2496s' balanced analog outputs were connected to two spare input channels on the Mackie.

Once things are wired up, all you need to do is set the input and output levels, select the desired Sample Rate (via a pair of buttons), and make sure your digital card is setup to accept digital input (via AES/EBU in this case).
If everything is OK, the Prodif 96's red LED will NOT be lit. (If the Red LED is lit, you need to go back and check the input/clock settings.) Obviously SEK'D planned for the 2496s converters to be used with the Prodif 96...

Wishlist

I wish the 2496s offered a Sample Rate of 88.2kHz. Sample Rate conversion from 88.2kHz to 44.1kHz is a LOT simpler process than converting from 96kHz to 44.1kHz. However… don't let this deter your interest. (see below)

Subjective

The listening tests I conducted were not scientific, but rather a common studio monitoring environment... using Yamaha NS10s.

The 16Bit extraction test - To start things off, I extracted material off of 3 different CDs (Jazz, Rock, and Pop). I then proceeded to play the material thru the Prodif 96/2496s combination... and then thru a DAL Card D+ (a solid 16Bit audio card). The results were as I expected. Playback from the Card D+ sounded good, but playback from the 2496s sounded better. The audio was just a little more detailed, had more punch, and the soundstage was more 3-dimentional.

Please understand... This is not a knock in any way against the CardD+!! I would expect a $999.00 (list price) set of converters to sound better than the converters included on ANY audio card. That's the law of economics.

Tracking at 24Bits 44.1kHz - I recently did a session tracking a solo performer playing acoustic guitar and singing. I has just gotten the 2496s converters, so I was itching to use them! My oh my what a WONDERFUL thing it was to track with 24Bits... No compression or limiting on the way in and the results sounded fantastic. The sound almost leapt from the speakers and nada single worry about digital overs. (Signal massive stadium applause!)

Next on the list was recording some acoustic drums... Ever struggled with trying to get a drum sound with BALLS (without digital overs) when recording to 16Bit digital? Again, the extra dynamic range when recording with 24Bits is a HUGE advantage! I was smacking the living hell out of my snare drum (mind you no compression/limiting on the way in) and the resultant sound had some MEAT... but not a single digital over. YAHOO!!!

Ahemm.... I haven't even mentioned the advantage of 24Bits on low level signals (s m o o t h). Fades and Reverb tails are smoother,
quantization noise is greatly reduced, etc.

Whats all the Hubbub about recording at 96kHz?

I know... I know... I can hear many of you saying there is absolutely NO need for recording with a 96kHZ Sample Rate. Two weeks ago, I would have agreed with you! I emphasize *would have* agreed with you! Let me state this very clearly... YOU CAN INDEED HEAR THE DIFFERENCE when recording with a 96kHz Sample Rate!

I wouldn't have believed it myself if I hadn't heard the results. Bottom line is that the highs sound more open and detailed. By the way... two other folks here in my studio could pick the 96kHz track EVERY time in a blind listening test (when compared with a 44.1kHz version). To hell with theory, my EARS tell me there is a difference.

Want a real dose of Blasphemy? I compared recording at 96kHz and Sample Rate converting down to 44.1, to simply recording at 44.1kHz. I couldn't believe my ears! The track originally recorded at 96kHz and Sample Rate converted down to 44.1kHz had much better sounding highs, maintaining much of the character from recording at 96kHz.

This goes against everything that I have learned over the years... and goes against accepted practice. So I don't make this statement lightly! You CAN hear a difference... anyone who tells you otherwise hasn't tried recording at 96kHz! Period.

The Prodif 96 and 2496s converters that I used for this review are staying in my studio. I'm not selling them, and they sure aren't goin' back to SEK'D.

I want to qualify that when test recording at 96kHz, don't record sample- playback units. They will have already rolled-off frequencies above 24kHz to prevent aliasing.

I suggest recording some real (not virtual) analog synth. (I used a Mini Moog for some of my tests.) Sawtooth waves with the filter wide-open will work nicely. Also try recording vocals, acoustic guitar, and acoustic drums.

Samplitude 2496

Mon, 01 Jun 1998 00:00:00 GMT

About the reviewer: As many of you know, I'm currently writing the US manual for Samplitude 2496. I work closely with SEK'D as an independent contractor, but I'm not an employee. So… as a professional DAW user with quite a bit of experience with Samplitude Studio 4.0 and Samplitude 2496, I'll do my best to provide facts and opinions that are truthful and accurate. I'm calling this an Overview / Review because I've included a lot of technical information about the program.

Projects - The DNA of an editing session

Samplitude 2496 is unique in that it can record audio to hard disk or your system's RAM.

1. HARD DISK PROJECT: A recording made to hard disk is called an HDP (Hard Disk Project). In most cases, you'll want to record to hard disk due to the large storage requirements of digital audio.

2. RAM PROJECT: A recording made to your system's RAM is called an RAP (RAM Project). RAP recordings are useful for short segments of audio such as Impulse Responses (for use with the Room Simulator) or drum loops. Used sparingly, this feature can help stretch your system's performance.

3. VIRTUAL PROJECT: Objects representing HDP and RAP recordings are manipulated in a window called the VIP (Virtual Project).

The VIP window is where you'll do most of your work in Samplitude 2496.

Some folks get confused by the use of Projects, and wonder why Samplitude doesn't just record audio as Wav files. Well… the audio IS recorded as Wav files! Samplitude just uses the HDP and RAP files for internal organization (to distinguish between recordings made to RAM and Hard Disk). So… audio recorded in Samplitude 2496 can be used in other audio applications (and vice versa).

Editing – Destructive vs. Non-destructive

Using Samplitude 2496, audio editing can be performed in two ways: DESTRUCTIVE or NON-DESTRUCTIVE.

DESTRUCTIVE EDITING is the process of applying cuts/splices, fades, effects, etc., directly and permanently (destructively) to Hard Disk (HDP) and RAM (RAP) recordings.

NON-DESTRUCTIVE EDITING is the process of applying cuts/splices, fades, effects, etc., to a recording, without physically altering it. How is this possible?

In the VIP window, Samplitude 2496 uses OBJECTS to represent your Hard Disk (HDP) or RAM (RAP) recordings. You can cut, splice, or apply fades to the Objects, but since they are merely representations, you aren't changing (destroying) your original recordings. One of Samplitude 2496's most powerful features is its ability to perform Non-destructive Edits WHILE you are listening to the audio playback.

Non-destructive editing if FAR more flexible and powerful than its Destructive counterpart. You are NEVER locked into an Edit decision, and you don't have to worry about destroying the original audio.

Now that you understand the concepts, it's time to take a detailed look at the VIP window.

VIP window

Since all Non-destructive Editing is done in the VIP (Virtual Project) window, lets take a look at some its features.

Samplitude 2496 VIP Window (click to enlarge)

Each track in the VIP window has a row of buttons.

Track controls in Samplitude 2496

? (Track Properties) button - Click on this button and a Track Info window will open allowing you to: name the Track, select the Record and Playback device, enable Surround Mode (stereo files only), etc.

M (Mute) button - Click on the M button to mute the Track (the button should now appear brown). Right click on this button and a list of all installed audio cards will appear. (This enables you to quickly assign a Play Device to the track.)

NOTE: The number after the "M" refers to the order in which the selected audio card is listed. In other words, if three audio cards show up in the list and you select the second audio card, the mute button would show M2. This provides a quick way of verifying which audio card is assigned to playback the Track.

S (Solo) button - Click on this button to solo the Track (the button should now appear green). All other tracks are muted.

L (Lock) button – By default, clicking on this button will prevent Objects in the Track-Slot from being moved horizontally (the button should now appear gray). By going to the OBJECT MENU and selecting LOCK OBJECTS > LOCK DEFINITIONS, you can choose the functions that will be Locked.

V (Volume Curve) button – Click on this button to enable the Custom Volume (vector mixing) Curve. The button should now appear yellow, and in the track-slot you'll see a yellow line representing the track's Volume.

CREATING A CUSTOM VOLUME CURVE:

1. After clicking on the V button to enable the Custom Volume Curve, double click anywhere on the yellow line (at the top of the track-slot) and a HANDLE will be created.

2. Handles represent points on your Custom Volume Curve. By creating and clicking and dragging these points (HANDLES), virtually ANY Volume Curve can be achieved.

NOTE: If you wish to temporarily disable the Custom Volume Curve, simply click on the V button. The yellow line representing the track's Volume will disappear and the Custom Volume Curve will be ignored. Click on the V button again and the Custom Volume Curve will be restored.

P (Pan Curve) button – Click on this button to enable the Custom Pan (vector mixing) Curve. The button should now appear blue, and in the track-slot you'll see a blue line representing the Track's Pan Position.

CREATING A CUSTOM PAN CURVE:

1. After clicking on the P button to enable the Custom Pan Curve, double click anywhere on the blue line (middle of the track-slot) and a HANDLE will be created.

2. Handles represent points on your Custom Pan Curve. By creating and clicking and dragging these points (HANDLES), virtually ANY Pan Curve can be achieved!

NOTE: If you wish to temporarily disable the Custom Pan Curve, simply click on the P button. The blue line representing the track's Pan Position will disappear and the Custom Pan Curve will be ignored. Click on the P button again and the Custom Pan Curve will be restored.

R (Record Enable) button – Click on this button to arm the Track for recording (the button should now appear red). Right click on this button and a list of all installed audio cards will appear. (This enables you to quickly assign a Record Device to the Track.)

NOTE: The number after the "R" refers to the order in which the selected audio card is listed. In other words, if three audio cards show up in the list and you select the second audio card, the Record Enable button would show R2. This provides a quick way of verifying which audio card is assigned to record the Track.

Just below the row of buttons, each track has a pair of LED peak-meters.

LED Peak-Meters – The LED Peak-Meters reflect the track's peak level during playback. A "Peak Hold" function is also employed which allows you to quickly see the highest peak that was reached during playback. For stereo tracks, the top LED Peak-Meter represents the left channel and the bottom LED Peak-Meter represents the right channel. For mono tracks, both LED Peak-Meters function together as a single unit.

Below the track's LED Peak-Meters, you'll find Volume and Pan sliders.

Volume Slider – This slider provides a quick way to increase or decrease a track's Volume. If you've enabled the Custom Volume Curve or used the Mixer window to record Fader movements, this Volume Slider will SCALE the track's overall Volume.
In other words, say you've applied a Custom Volume Curve to the Track, but later decide the Track is too loud. No need to redo the entire Custom Volume Curve, simply use the Volume Slider to reduce (scale back) the Track's overall Volume!

NOTE: Double click on the VOLUME SLIDER to quickly return it to detent (0db) position.

Pan Slider - This slider provides a quick way to alter a track's Pan Position. If you've enabled the Custom Pan Curve or used the Mixer window to record Pan movements, this Pan Slider will SCALE the track's overall Pan Position. In other words, say you've applied a Custom Pan Curve to the Track, but later decide the Track needs to be SKEWED further left. No need to redo the entire Custom Pan Curve, simply use the Pan Slider to skew (scale to the left) the Track's overall Pan Position!

NOTE: Double click on the PAN SLIDER to quickly return it to center detent position.

In the lower left corner of the VIP window, you should see four "S" buttons and four "Z" buttons. These are referred to as SCREEN FORMAT buttons.

S (Screen Position & Zoom Level) buttons – The four S (Screen Formatting) buttons can each store the current Screen Position and Zoom Level. This allows you to quickly switch between up to four different (stored) VIEWS. To store the current Screen Position and Zoom Level to one of the S buttons, press the SHIFT key and click on one of the S buttons.

Z (Zoom Level) buttons - The four Z (Screen Formatting) buttons can each store the current Zoom Level. This allows you to quickly switch between up to four different Zoom Levels. To store the current Zoom Level to one of the Z buttons, press the SHIFT key and click on one of the Z buttons.

In the upper left corner of the VIP window, there is a button labled MULTI CARD MODE.

Multi Card Mode button – Click on this button to enable Multi Card Mode (the button should now appear green). Multi Card Mode was designed to support audio cards (such as the Analog Arc88) that offer multiple inputs and outputs.

NOTE: Currently, Windows can only address Stereo audio devices, so these cards must present themselves as multiple (virtual) Stereo audio devices. IE: In Windows, the Analog Arc88 (8 analog inputs and outputs) will appear as 4 (virtual) Stereo audio devices.

When Multi Card Mode is active, a track can be assigned to playback through any of the (virtual) stereo audio devices. This is perfect when you want to use an external mixer to mix down your audio tracks.

NOTE: When using Multi Card Mode, the Master section of Samplitude 2496's Mixer window will be removed. It is not needed in this scenario.

Mo' Bits

High resolution audio (or Mo' Bits as it has been eloquently put) is available now.

Bit depth: Samplitude 2496 can record and playback audio using either 32Bit Floating Dynamic Range or 16Bit Integer formats.

Floating Dynamic Range means that no matter what the level of the audio (IE: -20dB or +20dB), you still have 32Bits of resolution (the audio is scaled). Also, it is virtually impossible to create clipping internally. IE: When applying additive EQ with a large boost, you don't have to worry about the EQ causing clipping.

Sample Rate: Samplitude 2496 can record at sample rates up to 96kHz.

What does this increased audio resolution mean to you?
1. The storage requirements of 32Bit Floating Dynamic Range audio are twice that of ‘standard' 16Bit audio. Also, your system will have to literally move twice as much data to achieve the same track count. If you plan on multi-tracking at 32Bit Float., you MUST have a fast CPU/Hard Disk. If you intent to primarily Master with Samplitude 2496, a more modest CPU/Hard Disk will do fine.

2. 32Bit audio provides a LOT more resolution than 16Bit audio. (Each added bit literally doubles the resolution. Check out this month's EM for more info on this subject.) Bottom line… Fades will sound smoother, you have a wider dynamic range to work with, when applying DSP – the cumulative effect of Rounding Error will be MUCH less than when using 16Bit resolution, etc. One thing I'd like to mention… you will NOT hear the difference of 32Bit Float audio when a tune is blasting away (unless you are talking about avoiding clipping), instead – you'll notice the difference on the lower level material! When it's time to burn your CD, Samplitude 2496 will dither (user selectable type and amount) the 32Bit Float. audio down to 16Bits maintaining much of the original quality.

3. If your project is destined for CD, I'd stick with 44.1 as your Sample Rate for the time being. This will avoid having to put your material through a Sample Rate conversion. Samplitude 2496 has a good Sample Rate conversion algorithm (realtime - while recording), but it's better to avoid any Sample Rate conversion process if possible.

User Interface

Samplitude 2496 has a slightly different ‘feel' than many audio programs. (Probably because the program originated from behind the Iron Curtain.) But… in my opinion, this difference is part of what makes Samplitude 2496 stand out from the crowd.

IE: Let's say you want to apply a Fade-In to an Object. Select the Object (click on it in the lower half of the Track-Slot). Five Handles will now appear. Click on the upper left Handle and drag it to the right. The length you drag this Handle will determine the length of the Fade. What's that you say, you want to alter the curve and taper of the Fade? No problem, click both mouse buttons together on the Object (right button slightly before the left) and a window will open allowing you to adjust the Fade's curve and taper. Sinus and Cosinus curves are available for super smooth fades. Virtually ANY Non-destructive Edit (including the above Fade-In example) can be performed WHILE the audio is playing.

With its realtime capabilities, Samplitude 2496 has a ‘just reach out and grab it and change it' kind of feel. Keep in mind that this performance is from a host-based DAW, not one relying on dedicated DSP hardware! In my opinion, that puts Samplitude 2496 in a class by itself. This my friends is why I have been so excited about Samplitude. Not because I work with the company, but because (as a user) the program has impressed me and enabled me to do things that would otherwise be a nightmare.

Want another quick example?

Let's say you want to Split an Object into two separate Objects.

Select the Object (click on it in the lower half of the Track-Slot), place the cursor where you want to Split to occur, and press the T key. What's that you say, you decided you don't like where the Objects were Split? No problem! You can fix this in a couple of different ways.

1. Press CTRL + Z to undo the Split. Then repeat the process making sure to place the cursor at the desired location.

2. You can Select one of the two Objects and drag either its lower left or lower right Handle (depends on which way you wish to ‘move' the Split) to Re-size it. Now, (making sure Snap is enabled), simply repeat this process for the other Object. When you get close to the Edge of the first Re-sized Object, this Object's Re-sized Edge will automatically Snap to it.

Again… you can perform this Edit WHILE the audio is playing. If you first mark a Range surrounding the Edit (click and drag in the upper half of a Track-Slot), Samplitude 2496 will continuously loop playback through the marked Range. This way, you don't have to keep pressing the Spacebar to start and stop playback. This makes Editing VERY fast.

To end this section, I'd just like to add that no matter what audio application you use, once you've experience true Non-destructive Editing (of audio or MIDI), you'll NEVER go back to Destructive Editing. I guarantee it!!!

Attention all audio software creators: Software (audio and or MIDI based) that doesn't catch on to the power of Non-destructive Editing will be left behind.

Recording Audio

To record a mono or stereo Track:

1. Arm the desired Track for recording by clicking on its "R" button (the button will appear red).

2. Press the "R" key. This will open the Record Parameter window.

3. In this window you can: set the Bit Depth and Sample Rate, name the File, monitor incoming audio level, drop Markers ‘on the fly', and start and stop Recording.

To record multiple Tracks simultaneously:

1. For each Track you wish to record, click on the Track's "?" (Track Properties) button and select the desired Record Device. Also, select whether the Track should be recorded in Stereo, Mono, or whether the Track should be recorded ONLY by the Left or Right input of the Record Device.

2. Arm the desired Tracks for recording by clicking on their "R" buttons (the buttons will appear red).

3. To check the incoming levels for each Track, go to the FILE MENU and select MULTI INPUT MONITOR. The LED Peak Meters for each Track will show the level of the incoming audio. When you have finished checking levels, click on STOP.

4. To begin recording, go to the FILE MENU and select RECORD MULTIPLE FILES.

Mixer window

Press the "M" key to open Samplitude 2496's realtime Mixer window. This is where you'll apply realtime EFX to your Tracks, as well as being able to use the Volume Faders and Pan Knobs to control your mix. ALL functions in the Mixer window are realtime. The Mixer window shows eight channel-strips at a time (for the sake of screen clarity), plus the master section.

Channel-Strips: Each channel-strip contains a Volume Fader, Pan Knob, Three EQ Knobs, Dynamics Processor Knob, Delay Knob, two Aux. Send Knobs, Mute Button, Solo Button, Automation Button, DirectX Button, and a Link Button. Yes Virginia, each Track has its own independent realtime EFX.

Volume Fader: To adjust the Volume of a Track, click and drag its Fader either up (louder) or down (softer). A numeric display at the bottom of the Fader will (temporarily) show the amount of volume increase or decrease in db units. The Volume Fader can be adjusted WHILE audio is playing.

Pan Knob: To adjust the Pan position of a Track, click on its PAN KNOB and drag the mouse left or right. The text "Pan" (beside the Pan Knob) will temporarily turn into a numeric display and show the Track's Pan position in db units. The Pan Knob can be adjusted WHILE audio is playing.

EQ Knobs – These Knobs control the Track's REALTIME three-band fully Parametric EQ. Each of the three bands can overlap the others allowing maximum EQ flexibility. The EQ can be adjusted WHILE audio is playing.

Click on any of the EQ Knobs and drag the mouse left to Cut or right to Increase. The text (beside the EQ Knob) will temporarily turn into a numeric display and show the amount of EQ decrease or increase in dB units.

For finer control, right click on any of the blue EQ KNOBS and a Filter Adjustments window will open allowing complete control of ALL parameters for that Channel's Parametric EQ. You can use the display (bottom half of the window) to help visualize the EQ you are applying.
Dyn Knob: This Knob controls the Track's REALTIME Dynamics Processor (Compressor, Limiter, Gate, etc.). The Dynamics Processor can be adjusted WHILE audio is playing. Click on the DYN KNOB and drag the mouse right to increase or left to decrease Dynamics Processing (Compression). The text "Dyn" (beside the Dyn Knob) will temporarily turn into a numeric display and show the Track's Compression Ratio. IE: If the display shows 4.0, the Compression Ratio is 4.0 to 1.

For finer control, right click on the light purple DYN KNOB and a window will open allowing complete control of ALL parameters for that channel's Dynamics Processor (see below).

Delay Knob: This Knob controls the Track's REALTIME Delay Effect. The Delay Effect can be adjusted WHILE audio is playing. Click on the DELAY KNOB and drag the mouse right to increase or left to decrease the level of the Delay Effect. The text "Delay" (beside the Delay Knob) will temporarily turn into a numeric display and show the level of the Delay Effect in dB units.

For finer control, right click on the dark purple DELAY KNOB and an Echo/Delay Effect window will open allowing complete control of ALL parameters for that channel's Delay Effect.

If you are using a multi input/output audio card, you can use Samplitude 2496's Aux 1&2 Sends to send audio (in realtime) to an outboard processor.

Aux 1&2 Send Knobs: These Knobs control the Track's REALTIME Aux 1&2 Send Levels. The Aux 1&2 Send Levels can be adjusted WHILE audio is playing. Click on either AUX KNOB and drag the mouse right to increase or left to decrease the Aux Send Level. The text (beside the Aux Knob) will temporarily turn into a numeric display and show the Track's Aux 1 or 2 Send Level.

NOTE: The Master Aux 1 & 2 Send Levels and the Audio Device used to output the Aux 1 & 2 Sends are selected in the Master Section of the Mixer window.

LED Peak-Meters: These meters function just like the meters in the VIP window.

Mute button: Click on this button to Mute the Track (the button will appear red).

Solo button: Click on this button to Solo the Track (the button will appear green).

NOTE: Multiple Tracks can be Solo'd (simultaneously) from the Mixer window.

Auto button: This button enables the Track's REALTIME Volume and Pan Automation. Volume Fader and Pan Knob movements can be recorded WHILE audio is playing.

Dir X button: This button enables DirectX compatible Plug-Ins to be used as REALTIME Channel-Insert effects for the Track. The Plug-Ins can be manipulated WHILE audio is playing.

NOTE: It is important to understand that your computer's CPU is bearing the load of ALL Realtime Effects (including DirectX Plug-Ins). Therefore, the faster the system's CPU, the more Realtime Effects it will be capable of running.

Link button: Click on this button to Link two adjacent (odd, even) Mixer Channels. The button will appear light blue indicating that the two channels are currently Linked. At this point, either Channel can be used to control both simultaneously. IE: If you move Channel 1's (Track 1) Volume Fader, Channel 2's (Track 2) Volume Fader will follow.

So how do the stock Samplitude 2496 channel EQ, Compression, and Delay EFX sound? In my opinion, the EQ and compression are amongst the best soft-processors available. The Delay sounds good, but is monaural. Don't forget, Samplitude 2496 supports DirectX, so you can use ANY available Plug-In as a channel insert effect.

Master Section:

In addition to the channel-strips and their EFX, Samplitude 2496's Mixer window offers (realtime) Master: Volume Faders, LED Peak-Meters, Non-destructive Normalization, level control of Aux Sends 1&2, Multi-Band Dynamics processing, separate broadband Dynamics Processor, 3-Band fully Parametric EQ, Dehissing/FFT Filter, Multi-Band Stereo Enhancer, and any DirectX compatible Plug-In as a master insert effect. For those who don't know, the Master Section of a Mixer controls the combined Stereo output of ALL channel-strips.

Master Volume Faders: To adjust the Master Volume (combined Stereo output of ALL Channel Strips), click and drag the Left or Right Master Volume Fader either up (louder) or down (softer). A numeric display at the bottom of the Fader will (temporarily) show the amount of volume increase or decrease in db units. Notice that the Left and Right Master Volume Faders are Linked by default (when one is adjusted, the other follows). The Master Volume Faders can be adjusted WHILE audio is playing.

Master EQ Knobs – This section functions just like the EQ in each channel-strip.

Master (Multi-Band) Compressor Knob: This Knob controls the Master Section's REALTIME Multi-Band Dynamics Processor (Compressor). The Multi-Band Dynamics Processor can process up to four separate Bands, and can be adjusted WHILE audio is playing. But… check this out. Any of the Frequency Bands can be soloed!

Samplitude 2496 Multiband Dynamics Processor

For those unfamiliar with Multi-Band Dynamics Processors, they basically combine the frequency selection of a Parametric EQ, with the ability to apply Dynamics Processing independently to each selected Frequency Band.

Click on the MASTER COMPRESSOR KNOB and drag the mouse right to increase or left to decrease Multi-Band Dynamics Processing (Compression). The Activate/Bypass button (just right of the Master Compressor Knob) now appears green to indicate that the Multi-Band Compressor is active. Also, a numeric display (just above the Activate/ Bypass button) will now temporarily show the Compression Ratio. IE: If the numeric display shows 4.0, the Compression Ratio is 4.0 to 1. By default, ALL Frequency Bands are initially Linked (share the same settings) and thus share the same Compression Ratio.

For finer control, right click on the light purple MASTER (MULTI-BAND) COMPRESSOR KNOB and a Multiband Dynamics window will open allowing complete control of ALL parameters for the Master Section's Multi-Band Dynamics Processor.

Typical uses for Multi-Band Dynamics Processors are: adding PUNCH to the low end of a mix without disturbing the rest of the frequencies, and de-essing vocal tracks that are too sibilant. The Multi-Band Dynamics Processor in Samplitude 2496 is extremely powerful.

Master Limiter Knob: This Knob controls the Master Section's REALTIME Dynamics Processor (Limiter). The Dynamics Processor can be adjusted WHILE audio is playing (just like you'd experience with a "real" outboard Limiter)! Click on the MASTER LIMITER KNOB and drag the mouse right to increase or left to decrease the Threshold of the Limiter. The Activate/Bypass button (just left of the Master Limiter Knob) now appears green to indicate that the Master Limiter is active. Also, a numeric display (just above the Activate/ Bypass button) will now temporarily show the Threshold in dB units.

For finer control, right click on the light purple MASTER LIMITER KNOB and a window will open allowing complete control of the Master Section's Dynamics Processor. This window is identical in form and function to the window used to manipulate a channel-strip's Dynamics Processor (outlined above).

Dynamics Processor Screen

The Master Section's Dynamics Processor can not only function as a Limiter, but also as a Compressor, Expander, Gate, and Distortion device.

Dehissing (FFT Filter) Knob: This Knob controls the Master Section's REALTIME Dehisser/FFT-Filter. The Dehisser/FFT-Filter can be adjusted WHILE audio is playing! Click on the DEHISSING KNOB and drag the mouse right to increase or left to decrease the Hiss Reduction. The Activate/Bypass button (just right of the Dehissing Knob) now appears green to indicate that the Dehisser/FFT Filter is active. Also, a numeric display (just above the Activate/ Bypass button) will now temporarily show the amount of the Hiss Reduction (in dB units).

Right click on the black DEHISSING (FFT FILTER) KNOB and a Dehisser/FFT-Filter Mixmaster window will open allowing complete control of ALL parameters for the Master Section's Dehisser/FFT-Filter.

Tip: In most cases, you want to adjust the Absorption parameter to be just aggressive enough to reduce the Hiss. Extreme settings will most likely cause artifacts!

When using the Dehisser, click on (enable) the INVERSE DEH. option (just below Byp. Filter) to hear ONLY the "Hiss" that is being removed. This is a great way to make sure that you aren't causing side effects by using the Dehisser! If you hear any significant amount of signal besides Hiss, you should probably readjust the Absorption and Reduction parameters.

If the concepts of Dehissing and FFT-Filters seem complex, you don't need to think about the science of the technology. What you have here is a Hiss Reduction unit combined with a Graphic EQ that lets you DRAW the EQ curve (instead of using sliders).

Stereo Enh. Knob: This Knob controls the Mid Frequency Band of the Master Section's REALTIME Multi-Band Stereo Enhancer. The Multi-Band Stereo Enhancer can process up to three independent Bands and can be adjusted WHILE audio is playing. Also, any of the Frequency Bands can be solo'd!

For those unfamiliar with Multi-Band Stereo Enhancers, they basically combine the frequency selection of a Parametric EQ, with the ability to Expand or Collapse the Stereo Image independently for each Frequency Band.

Click on the STEREO ENH. KNOB, and drag the mouse right to Expand or left to Collapse the Stereo Image of the Mid Frequency Band. A numeric display (above the Knob) will temporarily show the amount of Expansion (settings above 100) or Collapse (settings under 100). Notice that the Activate/Bypass button (just left of the Stereo Enh. Knob) now appears green to indicate that the Multi-Band Stereo Enhancer is active.

For finer control, right click on the black (MULTI-BAND) STEREO ENH. KNOB and a Multi-Band Stereo Enhancer window will open allowing complete control of ALL parameters for the Master Section's Multi-Band Stereo Enhancer.

NOTE: If Multi Band Mode is NOT enabled, the Multi-Band Stereo Enhancer becomes a Single-Band Enhancer that affects the Mid Frequency Band only.

Master Aux Sends (1, 2): Samplitude 2496's Master Aux Sends (1, 2) can be routed Internally to any DirectX compatible Plug-In/s installed on your system, or Externally to an outboard processor.

MASTER AUX 1&2 KNOBS - These Knobs control the Master Section's REALTIME Aux 1&2 Send Levels (combined output of ALL Channels' Aux 1 and Aux 2 Sends). The Aux 1&2 Send Knobs can be adjusted WHILE audio is playing. Click on either the AUX 1 or AUX 2 KNOB and drag the mouse right to increase or left to decrease the Aux Send Level. The numeric display (just left of the Aux Knob) will temporarily turn into a numeric display and show the Track's Aux Send Level.

Again, it is important to understand that your computer's CPU is bearing the load of ALL Realtime Effects (including DirectX Plug-Ins). Therefore, the faster the system's CPU, the more Realtime Effects it will be capable of running.

You must have an audio card with multiple outputs (at least 4) to route the Master Aux Sends to external outboard processors.

Master LED Peak-Meters: The pair of Master LED Peak-Meters reflect the combined Stereo (peak) output level of ALL Tracks during playback. Otherwise the meters are identical in form and function to the other LED peak-meters in Samplitude 2496.

Master Norm (Normalize) button: This button (in between the Master Volume Faders) can be used to quickly (Non-destructively) Normalize the Master Stereo Output.

To Normalize (Non-destructively) the Master Stereo Output

1. Press the SPACEBAR to start playback of your project. Make sure that the loudest section of audio has been played.

2. Once the loudest section of audio has been played, press the SPACEBAR to stop playback. The Peak-Meters (LED and Numeric) will now display the level of the loudest section.

3. Click on the NORM button. The Master Stereo Output Level will be (Non-destructively) Normalized to 0dB. Notice that the Normalization is accomplished by automatically adjusting the Master Volume Faders.

NOTE: Since the (Non-destructive) Normalization is accomplished by automatically adjusting the Master Volume Faders, you can undo the Normalization by simply double clicking on one of the Master Volume Faders.

Master Dir X button: This button (between the two Master Volume Faders) enables DirectX compatible Plug-Ins to be used as REALTIME Master-Insert effects (to process the Master Stereo Output). The Plug-Ins can be manipulated WHILE audio is playing.

Now might be a good time to interject that Samplitude 2496 can perform a realtime (while the audio plays) Mix Down using the Mix to File option in the lower right corner of the Mixer window. This function is similar to the realtime mixdown feature in Cakewalk Pro Audio (what you hear is what you get). An offline Bounce Down (mix down) of the Master Stereo output also available by selecting TRACK BOUNCING… under the TOOLS MENU. In both cases ALL realtime processing is included in the Bounce Down.

So how do the realtime Master EFX sound?

The Multi-Band Dynamics processor is simply the best soft-processor of its kind available right now. In fact, I personally feel that this processor alone is worth the price of admission. Incredible sound and incredible control! Want to hear the downside? This processor uses a considerable amount of CPU power. If you plan to use it in a multi-tracking situation, buy the fastest P2 CPU/Hard Drive that you can afford.

The broadband Master Limiter uses the same algorithm as the Dynamics processor in each channel-strip. As previously mentioned, it is amongst the best soft-Limiters available.

The Master EQ uses the same algorithm as the EQ in each channel-strip. As previously mentioned, it sounds excellent.

The De-hisser does an excellent job of reducing Tape Hiss. The fact that you can enable the INVERT DEH. option and hear EXACTLY what is being removed is a tremendous feature. Use this option to ensure that you aren't degrading the desired audio. The FFT Filter takes some getting used to, but offers a powerful way to control the overall Frequency Contour of the Master Stereo Output. Again, think of this processor as a sort of Graphic EQ that lets you draw the Frequency Contour. Sound quality wise, the results sound very good. SEK'D has done an excellent job in creating ‘musical sounding' soft-EQ.

The Multi-Band Stereo Enhancer works well, but I advise you to use the Phase Correlator to make sure that you aren't throwing things too far out of phase.

Currently I know of no other host-based DAW that offers realtime audio processing this advanced. But… don't take my word for it, give Samplitude 2496's realtime DSP a thorough workout and let your ears be the judge.

Realtime CD burning

Ever wished you could burn a quick ‘one off' CD of your latest rough mix without having to first perform a Bounce Down and creating an Image File? Samplitude 2496's realtime CD burning can do just that. All realtime processing is included while burning the CD.

To make a quick ‘one off' CD from a current rough mix in the VIP window:

1. Click on the AUTO TRACK MARKERS icon (in the upper toolbar). This will automatically set all PQ codes.

2. Click on the MAKE CD icon (in the upper toolbar).

3. Select BURN ON THE FLY (under Mode).

4. Click on OK. Samplitude 2496 will now burn your rough mix directly to CD.

Realtime CD burning isn't as flashy as some of the other features of Samplitude 2496, but it's most definitely a feature that you will use constantly. Anything that makes life quicker/easier in the studio scores high marks with me. If you are extremely busy, realtime CD burning is a valuable time saver.

Get to the end already - Geesh

I'm going to cut this Overview/Review short now. I could literally write a Book about Samplitude 2496. ;-) You've probably already realized that if you're still reading.

I just want to quickly mention a list of other items that Samplitude 2496 offers:

1. Punch-In recording (manual or automated)

2. Graphic Noise-Print style Noise-Reduction (a la SoundForge).

3. Room-Simulator (very similar to Acoustic Modeler) for creating truly World-Class reverb.

4. De-Clipper for fixing digital overs. I've used this function MANY times. It really works and it doesn't leave artifacts.

5. Waveform generator for creating test-tones, etc.

6. Take Manager - allows you to quickly swap Objects in the VIP window.

7. You can Link a VIP to an AVI file and have the frames and Video displayed and controlled by the VIP. Great for those creating/working with audio for AVI.

8. User Defined Shortcut keys for ALL Menu Options.

9. Live Input mode for processing/mixing of external audio tracks (IE: from an ADAT) without first recording them in Samplitude 2496! You need an audio card with multiple audio inputs to use this feature.

10. Markers can be stored and recalled (even dropped ‘on the fly' as the audio is playing).

11. Nudge key combinations - for nudging Objects or Crossfades left or right (in either of two user-defined increments).

12. Object based realtime EFX: Each Object can have its own REALTIME 3-Band fully parametric EQ and or Dynamics Processor! These are completely separate from the processors available to each Track.

13. Multi-Level undo, up to 100 levels

14. Vector style Volume and Pan curves.

I'm sure I'm leaving some things out, but I think you get the picture here. Samplitude 2496 is ushering in a new era in host-based DAWs. With its realtime nature, high-quality DSP, DirectX support, etc. you are easily looking at the most advanced host-based DAW currently available. Samplitude 2496 is simply a monster. If you are in the market for a powerful host-based DAW, I urge you to check this one out.
2496 - A Big Step in the Right Direction
Joel Braverman

Samplitude 2496, SEKD's new product shows some promising improvements in the area of user interface, and has some fantastic new features. I believe the product still has a long way to go in conforming to "windows standards", but if you are willing to spend the time learning the interface, it can pay off quite well. If you are already a user of Samplitude, then you have the option of continuing to work in the old ways, or to use some of the new features. 2496 is, as its name implies, a recording package capbable of recording 24bits at 96khz, if you have hardware that supports those frequencies.

In terms of user interface improvements, the most notable is the new "Universal Mouse Mode" which alters the way the mouse works depending on which half of a recorded waveform the mouse is over. If the mouse is over the upper half, it works in 'range' mode - where you can select a range of the wave to edit or process. If its over the bottom, its in object mode, which looks at the recording as an object which you can move and manipulate as a whole - sliding it forward in time, or performing a non-destructive fade, or overall volume change.

The coolest new features of 2496 are the Multiband Compressor, and the Multiband Enhancer. The enhancer is a stereo expansion effect, which is quite dramatic - you can apply separate enhancement settings to the Bass, Mid and Treble, with selectable frequency ranges. The compressor also functions in a similar fashion. The coolest in both of these is the ability to SOLO any band, and tweak the compression or enhance, then un-solo and hear how it sounds in the entire mix. Extracting the soloed band to a file and using it soloed for a few bars is a KILLER effect for Techno and Rap. Both effects need to be applied sparingly, when used on the whole as the effect on the overall mix can be negative if either of them are overused.

Another new feature is the Dehisser. This is a special function that removes tape and other hiss type noises, without killing off too much of the high end. As in Samplitude Studio, there is also an extremely cool acoustic modeler for capturing the ambience of any space and applying it to a sound.

I used 2496 to Master a demo for a Rapper known as "Tha Gabba", to send to a record company in New York. I processed two versions, one with minimal alteration and one with a good amount of experimentation with the Multiband compressor, and enhancer, and a tiny amount of modeled reverb. Then i burned a cd with the !!!BUILT IN REALTIME CD BURNING!!!. Previous versions required you to create a Table of Contents file, and use an external CD burning package. (One came with the old 24bit version of Samplitude Master)

Everyone who has heard the processed version liked it much more than the unprocessed version. I spent several hours tweaking the compressor and enhancer settings. I only used a small amount of enhancement, on the mid, and more on the highs. Enhancement can kill your bass end really easily, so it must be used sparingly.

I did suffer several crashes, trying to use DirectX plugins, but for the most part 2496 worked quite well. Since I only used 2496 heavily for a week, I don't feel that I can give a truly critical review - it has some problems, mostly stemming from the user interface's inital obtuseness. However the UI also allows one to work faster once you become familiar with it - almost every function is on or two keystrokes away - you might never need to touch a mouse if you don't want to, and if you do use it heavily, it is MUCH easier to deal with than in previous releases. The one-keystroke functions tend to work against you when you first start using 2496 though. - be careful with your precious projects - back them up before converting them to 24bits and processing them.

I recorded an interview into 2496 to transcribe while it played. I discovered that the space bar, which is used to start and stop playback, is active even when you are in a different app, typing text. This interrupted the playback so that many times I had to go back to using my micro-cassette recorder in order to quickly get it done. The realtime pitch shifting function did help me to listen to words that were semi-unintelligable on the tape, and understand them, + to record the tape in quickly - I set the microcassett to play back at high speed, then tranposed (non-destructively) down 16 semitones to play it back.

There are many more features, and 2496 is a comprehensive package, with tons of features, new and old, and an attempt to further improve the user interface that is a step in the right direction.

Yamaha DSP Factory

Wed, 01 Apr 1998 00:00:00 GMT

One of the most exciting products to make headlines recently is the Yamaha DSP Factory DS2416.

The Yamaha DS2416 offers the mixing power of the Yamaha 02R digital mixer, complete with 24 channels of digital mixing, on-board digital effects and dynamics processors -- along with everything else professionals need - plus 16 tracks of hard disk recording with up to 32 bit resolution.

Unlike most other audio cards, the DS2416 relies on its own processing power and not the computer's CPU. This arrangement makes much better use of your existing hardware.

DSP Factory equipped with 2 expansion bays

The feature list is impressive:
- 24 channel, 32-bit digital mixer
- 10 bus outputs and 6 aux sends
- 104 bands of 4-band parametric EQ
- 26 dynamics processors
- 2 effect processors equal in quality to Yamaha's REV500
- Channel delay on 20 channels
- Comprehensive metering
- Digital cross-patching for channel inputs and outputs
- 2 channel 20-bit AD/DA converter
- Stereo DIGITAL(Coaxial) input and output
- ALL the above features are available all the time.

The DS2416 has five Yamaha proprietary DSP chips right on the card, which are dedicated to performing all the above mixing functions simultaneously, making it far more powerful than other systems. What's more, all the major audio software companies seem to be jumping on board the DSP Factory, so it won't feel like a proprietary system. This is a VERY BIG plus in my opinion! See the list of vendors who have already committed to supporting the DSP Factory:

- Cakewalk
- Canam Computers
- C-mexx
- Emagic
- IQS (Innovative Quality Software)
- Musicator
- SEK'D
- Sonic Foundry
- Steinberg

SEK'D and Cakewalk have already announced new controller interfaces for the DSP Factory. Cakewalk's interface is an integral part of the new Cakewalk Pro Audio 7.0 software package:

Cakewalk Pro Audio 7.0

SEK'Ds solution is called the Samplitude Studio / C-Console, and provides an intelligent controller interface for all features of the DSP Factory.

Samplitude Studio / C-Console

Installed in the 1 PCI card slot of a standard personal computer the card uses only 1 IRQ and no DMA or port address lines (the card uses intelligent memory mapped I/O). Using only 1 slot, 1 IRQ and no DMA makes the DSP Factory one of the most "polite" cards available.

The AX44 expansion unit (pictured below) adds 4 additional inputs and outputs to the DS2416. This unit installs directly into the 5-1/4" drive bay of your computer. Due to negative feedback from customers a seperate breakout box is planned.

AX44 Expansion unit

Yamaha is also planning digital interface solutions which will allow connection of multi-channel digital equipment such as outboard digital multi-track recorders and signal processors. The first planned is the optional AX16-AT Audio Expansion Card which will provide 16 digital inputs and outputs in ADAT format. This will enable easy transfer of digital multi-track audio to and from the DSP Factory recorder.

Targeted shipment of the DS2416 card and AX44 is Summer 1998. The estimated price of the DS2416 is under $1000, the AX44 is estimated at about $300.

If any company has the resources to make this happen, it would be Yamaha (by the way, Yamaha and Korg are held by the same company). They were the first with affordable digital synthesis, affordable digital mixers, and they have tremendous R&D and capital compared to smaller companies.

I'm not saying that Yamaha makes the absolute BEST gear; but for a company that makes darn near EVERYTHING musical, they haven't made too much "junk." If this card lives up to the hype and actually sees the light of day, what a *Glorious* day for the Project Studio! Dare I say, the "Studio in a Box" might then REALLY exist.

But... I've gotta agree with Lionel, those inputs on the drive bays have to go! Give me a breakout box! Those of us who want I/O expansion options would be willing to pay slightly more for a breakout box. (ed. - agreed)

Links to more information about the DSP Factory:
Yamaha Japan
Yamaha America
Yamaha UK