Efraimidis Vassilis

AD/DA Converter 24 bit/192 kHz 8 x Balanced jack I/O 4 x AES / EBU I/O 2 x ADAT I/O Word clock sync MIDI I/O Optional MADI I/O Size: 19"/ 1 U Full remote control via MIDI Remote Controllable 8-channel AD/DA Converter RME converters and interfaces are among of the most popular on the planet. The reason being, their excellent sound quality, extraordinary feature sets, and universal compatibility. In fact, most product specialists who come to Sweetwater to demo audio software of all types use RME interfaces since they will work with just about any system you plug them in to. Now RME adds a remote controllable 8-channel AD/DA conversion to their repertiore of products with the new ADI-8 QS AD/DA Converter. The ADI-8 QS is a new highly flexible and fully featured 8-channel AD/DA converter. The ADI-8 QS combines excellent analog circuit design with outstanding low latency AD/DA converter chips. RME ADI-8 QS AD/DA Converter at a Glance: Reference class 8-channel, 24 bit/192 kHz remote controllable AD/DA converter New digital filters reduce the conversion delay to just 8 samples 2 x ADAT I/O, 25-pin AES/EBU, expansion slot for MADI option card Full remote control, intelligent Clock, SyncCheck, and SyncAlign 8-channel AD/DA Along with SteadyClock, the QS surely redefines the reference class of analog/digital converters. The compact 19"/ 1U device has a host of extraordinary features, including analog and digital limiters, 4 hardware reference levels up to +24 dBu, AES/EBU and ADAT I/O up to 192 kHz, optional MADI I/O, remote control via MIDI, remote digital input trimming for full input calibration, remote volume control for all 8 analog outputs, either separately, globally, ganged and much more. Serious Low-latency The analog inputs and outputs feature the same circuitry as RME's well-known ADI-8 series, ensuring excellent signal to noise ratio and low THD. On the digital side, SteadyClock, RME's unique clock technology and jitter suppression, further enhances the list of features, ensuring the best sound quality regardless of the quality of the reference clock signal. The QS uses the latest AD- and DA-converter circuits with up to 192 kHz, with S/N ratios of 120 dBA. New digital filters reduce the conversion delay from over 40 samples to just 8 samples. Now that's some serious low-latency! Numerous Interface Options The ADI-8 QS also offers an unsurpassed choice of interface options. In addition to 2 ADAT I/Os that support up to 192kHz (S/MUX), the device also features a 25-pin AES/EBU interface for all inputs and outputs. And that's not all - The ADI-8 QS features RME's latest technological achievement - an expansion slot that makes the device even more flexible. The slot can host a MADI option card for direct connection to MADI systems. All interface options can be used simultaneously under all conditions; analog (balanced TRS and/or D-Sub), ADAT, AES/EBU and the MADI option card. Remote Control Within the ADI-8 series of converters, the full remote controllability of the QS allows the device to be placed anywhere on stage or in the studio as a remote controllable SuperStageBox with full access from the control room or FOH mixer. Other typical RME features like Intelligent Clock Control (ICC), SyncCheck, and SyncAlign have also been implemented. RME ADI-8 QS AD/DA Converter Features: 8-channel AD converter, fully symmetrical design, 120 dBA S/N 8-channel DA converter, double balanced output, 120 dBA DA Low latency conversion: only 8 samples of delay! 2 ADAT optical inputs, 24 bit, with RME's unsurpassed Bitclock PLL, up to 192 kHz 2 ADAT optical outputs, 24 bit, fully compatible, up to 192 kHz 8 AES/EBU I/Os, full channel count up to 192 kHz, 24 bit, connected via D-sub 8-channel Digital Input Trim over a range of 6 dB Full remote controllability via MIDI and MIDI via MADI All settings are stored permanently Included Remote Control for store/recall of presets, volume and dim Optional MADI module

The RME HDSP AES-32 PCI Audio Card provides eight AES inputs and outputs at 192 kHz sample rate. It is also equipped with two MIDI I/O ports, word clock I/O and can be used with the optional TCO module for synchronization to LTC and video. The RME HDSP AES-32 PCI Audio Card is RME's reaction to requests from audio professionals for an AES-based solution with the typical RME features and quality. This audio card is a perfect all-in-one solution for professional users in the fields of broadcast, TV, theater, stage/PA - and in any professional studio. The RME HDSP AES-32 PCI Audio Card's main board contains word clock I/O and a 25-pin D-sub connector, providing four AES inputs and four AES outputs via standard digital breakout cable (Tascam pinout). The word clock input is galvanically isolated via transformer and operates on Single, Double and Quad Speed signals automatically. The expansion board adds audio channels 9-16 and a connector for a MIDI breakout cable with two MIDI I/Os (four 5-pin DIN connectors). It does not require a PCI slot. Thanks to the DSP-based TotalMix mixer, all 16 inputs and outputs can be routed to 16 physical outputs freely, offering extraordinary monitoring capabilities. Up to 8 fully independent stereo submixes can be created. Routings can be copied and pasted, faders ganged and grouped, which, along with the amazing matrix window, turns the AES-32 into a powerful and easy-to-use redistributor, patchbay, router, converter, and splitter. The hardware-based mixer is fully MIDI-controllable. Furthermore, the DSP hardware calculates RMS and peak level meters for all 48 audio channels, with no measurable CPU load. The Time Code Option of the RME HDSP 9632 and RME HDSP MADI can also be used with the AES-32 to sync to LTC and video. Thanks to SteadyClock, the TCO not only extracts absolute positions from these signals, but also a very clean low-jitter word clock. The RME HDSP AES-32 PCI Audio Card's most important settings are stored on the card itself. Rather than operating in a default mode from the moment the conputer is switched on until the driver is loaded, the RME HDSP AES-32 PCI Audio Card will immediately activate the last used sampling rate, master/slave configuration and AES format, thus eliminating any startup noise or problems in the clock network, both at system start and reboot. An optional 19" breakout box is also available. Its innovative design features XLR I/Os that can be accessed from the front or rear in a rack, according to the user's choice. BOB-32 provides D-sub connectors with Tascam and Yamaha pinout, therefore can be used with other devices as digital rack breakout box as well. RME HDSP AES-32 PCI Audio Card Connectivity 16 Input Channels/ 16 Output Channels 8 x AES/EBU I/O 2 x MIDI I/O Word Clock I/O TotalMix HDSP TCO (Optional) BOB-32 (Optional) RME HDSP AES-32 PCI Audio Card Features Support for 192 kHz at full channel count Native support for Double and Quad Wire transfer Direct conversion between these formats SteadyClock for highest jitter suppression and clock regeneration Intuitive and clearly laid-out user interface TotalMix: 512 channel mixer with 42 bit internal resolution 48 level meter Peak/RMS, hardware-computed RME HDSP AES-32 PCI Audio Card Specifications 8 Buffer Sizes/Latencies Available: 1.5 ms, 3 ms, 6 ms, 12 ms, 23 ms, 46 ms, 93 ms, 186 ms ASIO Zero CPU Load Technology: 0 (zero!)% CPU load when using ALL channels! All Settings: Changeable in real-time Clock Modes: slave and master Clock Control: Automatic and intelligent master/slave clock control Zero Latency: Enhanced Zero Latency Monitoring for latency-free submixes and perfect ASIO Direct Monitoring Sync Sources: 8 x AES, word clock, internal Varipitch: By input signal or word clock Sample Rates: 32 kHz up to 192 kHz, variable (Sync/word clock) Sample Rate Range: AES: 28 kHz - 204 kHz, word clock: 28 kHz - 204 kHz Jitter: < 1 ns, Internal and all inputs Jitter Sensitivity: PLL operates even at 100 ns Jitter without problems Comes With DIGICheck: the ultimate measurement, analysis and test tool Input: 8 x AES, 2 x MIDI, word clock (BNC) Output: 8 x AES, 2 x MIDI, word clock (BNC) Input Word Clock: BNC, Signal Adaptation Circuit (functional from 1.2 Vpp input signal), termination via jumper Output Word Clock: BNC, low-impedance driver stage, 4 Vpp into 75 Ohms, short-circuit-proof MIDI Input & Output: Via four 5-pin DIN jacks

Feature Perfect As in the original Hammerfall, the 9652 product number refers to the fact that the card supports sampling rates up to 96kHz and audio inputs and outputs for up to 52 channels — 26 inputs and 26 outputs, from three ADAT I/O pairs and co-axial S/PDIF I/O. Each ADAT channel supports 48kHz/24-bit operation, and S/MUX mode is also available to pair ADAT channels in order to provide 12 96kHz/24-bit channels instead. Like its predecessor, the HDSP 9652 also implements what RME refer to as 'ASIO zero CPU load' technology, meaning that you can have 52-channel operation without placing any burden on the host processor. An ADAT 9-pin sync connection is also available for sample-accurate transfers (with a suitable ASIO 2-compliant application, such as Cubase), and BNC word clock I/O connections are still provided. On the subject of clocking, the DSP 9652 card excels at keeping you aware of the clock status of your audio and sync ports via the Sync Check area of the control panel window, and thanks to RME's Intelligent Clock Control technology, the card can automatically sync to the port supplying the most stable clock signal — very handy. Co-axial S/PDIF I/O is supplied on the same breakout cable as the ADAT 9-pin connection, and you can assign the first pair of ADAT ports to transmit or receive S/PDIF via optical connections instead. As existing Hammerfall users know, it's also possible to set this so the first ADAT output transmits S/PDIF and the first ADAT input still receives ADAT data, or vice versa, which is rather neat. The original Hammerfall was purely an audio card, but the HDSP 9652 adds two pairs of MIDI ports for 32 MIDI input and output channels via another supplied breakout cable. This is a perfect touch since there are few situations where audio workstations don't require MIDI ports — even if you never use MIDI instruments, hardware control surfaces such as Mackie Control still connect over MIDI. Although the HDSP 9652 uses one PCI slot in your computer, many of the connections are supplied on a separate expansion card that needs to be fitted in an additional backplate, in the place of another PCI card. And if you find the I/O options of the DSP 9652 not to your liking, a range of analogue and TDIF expansion boards are available separately (see The Hammerfall DSP Family box for more information), and you can fit a second HDSP 9652 card to the same system to double the number of available inputs and outputs. The only extra I/O-related feature I'd perhaps like to see on the HDSP 9652 card is a headphone output, which is present on every other Hammerfall DSP product (see the Family box), even if it just mirrored a digital output pair for test and monitoring situations. Who's Gonna Drive Your Card? The HDSP 9652 is compatible with pretty much every major operating system: for Windows users there are multi-client MME drivers, ASIO 2 and GSIF (for Gigastudio) drivers on Windows 98/ME, and full multi-client MME, ASIO and GSIF drivers for Windows 2000/XP users. The multi-client ASIO drivers in this latter set, which RME added during the course of the review, deserves a special mention since they make it possible for two ASIO-compatible applications to be loaded and use the s ame soundcard and drivers simultaneously, so long as both applications are using different channels. This makes it possible to use Cubase and Wavelab at the same time via ASIO, or even Cubase and Nuendo, for example; and this new multi-client support will have engineers salivating when they see it's now possible to run RME's highly regarded Digicheck analysis software alongside an ASIO application, as described in the Digicheck box below. Mac users are catered for as well, with ASIO 2 and OMS drivers for Mac OS 9, and Core Audio support under Mac OS X. However, Mac users won't benefit from any multi-client functionality, and, unfortunately, there are no Core MIDI drivers for OS X users to make use of the MIDI ports at present. Hammer To Fall RME have always been pretty good at supporting their products with driver updates, and the HDSP 9652 card is no exception. During the review, for example, I needed to update the drivers to try out some of the latest features, and in order to use the newer drivers, it was also necessary to update the firmware on the HDSP 9652 card itself. However, I got nothing but errors when I tried to use the firmware update program downloaded from RME's web site, which worried me that the firmware update process might have damaged the card, or that it was faulty in the first place. Fortunately, the HDSP 9652 card features what RME describe as Secure BIOS Technology, which prevents the card from being damaged if there's a problem when updating the firmware. This works by providing a second, read-only BIOS on the card that's automatically activated if the firmware upgrade fails, enabling you to check your system and redo the flash update. After emailing RME's Matthias Carstens, my problem was quickly identified. Apparently, the update procedure for the FPGAs (Field-Programmable Gate Arrays) from Xilinx that RME use on the Hammerfall DSP card is incompatible with certain motherboards, and the only solution if the user encounters a problem when updating the firmware is to put the card in another computer. RME are working to improve the situation, and it should be stressed that it's not the HDSP 9652 card itself that's incompatible with certain motherboards, only the firmware update process. Once the firmware has been upgraded, you can move the card back to the first computer and it should work fine. The motherboard I was using that caused the problems was an Asus AV7M266, a Socket A motherboard featuring an AMD 761 chipset. I tried putting the card in a similar Athlon-based system with an Asus A7S333 motherboard, which has a SiS 745 chipset, and this time the firmware update process worked perfectly. I shouldn't imagine this will be a problem when using the card in a Power Mac, but if you have an incompatible Windows-based system and no additional machine to perform the firmware upgrade, you can send the card to your friendly RME distributor to do it for you. The Hammerfall DSP Family If the all-digital nature of the HDSP 9652 card doesn't suit you, RME offer the EXB range of expansion boards, including the AEB4I, AEB8I, AEB4O and AEB8O, which offer four or eight analogue inputs or outputs, and the TEB TDIF board. The HDSP 9652 supports two EXB boards, which connect to the main card via a ribbon cable and require an additional backplate in your computer, with any combination supported, including two of the same type. It's worth emphasising that these expansions don't offer additional channels, though, just alternative ways of getting audio into and out of the soundcard from the ADAT connections. If the HDSP 9652 still doesn't sound like the card for you, RME's HDSP 9632 might be more appropriate where Total Mix is still desirable, but a smaller number of channels is required. It features a single pair of ADAT I/O ports, co-axial S/PDIF, a balanced stereo analogue input and output, a single pair of MIDI I/O connections, and a headphone output for monitoring that mirrors the other analogue output. Alternatively, there's the forthcoming HDSP MADI, which will be Windows-only initially, offering a pair of MADI connections for 64 inputs and 64 outputs via optical or co-axial connections, with two pairs of MIDI I/O, word clock I/O, and a stereo analogue output for monitoring. While MADI is usually only found on high-end professional equipment, RME have also recently announced the ADI648 converter box, featuring eight pairs of ADAT I/O to a MADI pair. And, finally, there's the original Hammerfall DSP Multiface and Digiface systems as mentioned at the start of the review. DSP Without DSP The DSP appendage to the Hammerfall name refers to the card's ability to carry out certain DSP functions such setting up monitor mixes. Oddly, however, it turns out that the HDSP 9652 doesn't actually have any DSP chips on board; instead, RME provide hardware DSP functions using FPGAs, as mentioned earlier. A Field Programmable Gate Array (FPGA) can be thought of as a reprogrammable processor, where the behaviour of the logic elements can be configured by the designer at any time, even via a downloadable update from the user's computer. So instead of using an off-the-shelf DSP chip or having custom chips manufactured, RME have used an FPGA to provide the specific behaviour required for the DSP functionality the company wants to implement. This means that later, rather than being limited by the fixed behaviour of a DSP chip, RME can simply update the FPGA if its design needs to be changed to accommodate additional functionality. One of the big questions asked by new users of audio workstations is whether they need an additional mixing console, and while my personal response is always an emphatic 'no', there are certain tasks, such as setting up monitoring, that are often made easier by adding a digital mixer when recording groups of musicians in a studio. However, to overcome the need for an additional mixer for monitoring duties, the Hammerfall DSP series of products all offer Total Mix, which provides complete control over the input, playback and output channels of your workstation. The second version of Steinberg's ASIO driver API technology introduced a technology called ASIO Direct Monitoring, which allowed the soundcard to route an incoming signal directly to an output, rather than having it pass through the ASIO host application, such as Cubase, and incur a latency that would make it difficult for the musician to play along with the rest of the song. Total Mix basically takes the concept of ASIO Direct Monitoring many steps further by letting the user decide which inputs and playback channels are mixed to which outputs, and all of this happens independently of your ASIO host application, without affecting the way it's set up to record the incoming signals or play back the outgoing signals. As a simple example of how Total Mix can be useful, say you're recording a guitarist in your live room and you want to give them a different monitor mix to the one you're hearing in the control room. The guitarist's headphones can be connected to a different output pair to the control room, and using Total Mix you can route the input from the guitarist to both monitor mixes on the different output pairs simultaneously. Now, say the guitarist wants to hear more of himself than the playback (for example!), and the engineer wants to hear a balanced level of both the playback and the guitarist — no problem with Total Mix. Simply turn the playback level down on the playback channels routed to the guitarist, and turn the guitarist's input down on the engineer's monitor mix. To finish, the guitarist's mix might be overloading, so you can simply turn the output channels down on that monitor mix. It's a simple example, I know, and one that's really quite easy when a separate mixer is used with your audio workstation — but that's the point: Total Mix facilitates this type of situation without the user needing a hardware mixer. And as I mentioned before, Total Mix works independently of your ASIO host, so the guitar would still come in on the same input channels at unity gain from your software's perspective, regardless of the routing and levels in Total Mix. To give you some idea of Total Mix's power, any input and playback channel can be routed to any number of output channels, with each routing having its own independent level setting. This means, to quote the figures on RME's web site, with the 26 input and 26 playback channels being mixed to the 26 output channels, the mixer must be able to do the sums for (26+26) x 26 = 1352 channels simultaneously, which is rather impressive. Conclusion As I mentioned at the start of this review, I've been a fan of RME's products and technology for several years now — the attention to detail in every aspect of development is exceptional. Having experienced nothing but rock-solid performance from any audio workstation I've used with an RME soundcard, these products are usually what I recommend if no additional features are required along the lines of Creamware's SCOPE cards, for example. Trust isn't always something you have can have with a product or manufacturer, especially in the technology market, but I trust RME and the HDSP 9652 maintains their tradition of great, yet affordable technology. information infop.gif £399 including VAT. infot.gif Synthax Audio UK +44 (0)1664 410600. infow.gifwww.rme-audio.com Test Spec Asus A7S333 motherboard with Athlon XP1800 processor, 1GB DDR (PC2100) memory and an ATi Radeon 7500 64MB dual-head graphics card, running Windows XP Professional. Published July 2003 SOUND ON SOUND

Straightforward "knob per function" mixer layout based on Harrison's renowned music consoles. Precise emulation based on Harrison's famous analog console circuit designs. Precision algorithms for EQ, Filter, Compression, Analog Tape Saturation, and Summing. The Mixbus engine is internally dithered, ramped, and gain staged so that sound quality is preserved as close to analog as possible. Unlimited stereo or mono input channels, each with unlimited plug-ins, sends, and hardware inserts. (limited only by CPU and disc speed). Unlimited MIDI tracks, each with virtual instrument support, unlimited plug-ins, sends and inserts. Polarity, Input trim, Sweepable 4-band EQ with second order HP and LP filters, Compression, and 12 mix bus sends on every track Mixbus32C has an optimized signal flow with 12 stereo mix buses and a master bus, all featuring Tone controls, Compression, Sidechaining, and Analog Tape Saturation simulation Stereo Master Bus that features Tone controls, Analog Tape Saturation, K-meter, Stereo Correlation Meter, and Limiter. Automatic plug-in delay compensation to support effects such as parallel compression without time misalignment. Comprehensive "at-a-glance" metering with peak, peak hold, and compressor gain reduction visible on every track and bus. Extensive DAW editing features including "Smart Tool", Ripple Edit, and configurable keybindings for nearly every operation. Industry standard plug-in support: Mixbus loads AudioUnit (AU) plug-ins on OSX; and both VST and LV2 plug-ins on all platforms. Mixbus provides professional features such as Loudness Analysis (dBTP+LUFS+LU Range), LTC (SMPTE) generation and sync; video pullup/pulldown. Persistent undo (undo remains possible after closing and reopening a session). Fourth-generation Harrison "True Analog Mixing™" processing engine with enhanced compressor/limiter algorithms; built-in sidechain bussing, and increased plug-in flexibility. Support for video timeline, video window, and audio+video exports. Recording operates like a tape machine: Punch-in on the fly, and use per-track input monitoring just like a multitrack tape machine. Extensive export features including multitrack stem export, simultaneous multi-format exports, multiple export ranges, CD track markers, silence trimming, normalization, and command-line arguments to trigger third-party encoders or file-management scripts. Audio regions are stacked in "layers", and every region fadein/out is a crossfade to the region(s) beneath it. Plugin "pin connection management" provides rich editing of signal-flow through each channel's path, including full support for dual mono, mono-to-stereo, sidechaining, and other advanced flows. The user interface scales to fit your monitor; larger monitors will get bigger knobs and sharper text. Refined workflow and graphics, including scaling and support for HiDPI (Retina) displays. Generic MIDI devices can use "MIDI Learn" to assign MIDI controls directly to on-screen knobs or use MIDI profiles to select many pre-configured device layouts from a pull-down menu. Mackie control protocol is provided for motorized fader controllers, and the Mackie protocols have been extended to ergonomically control the Mixbus channelstrip EQ, compressor, and 8 mixbus sends. Dedicated support is provided for popular devices like the Presonus Faderport, Mackie MCU, and Behringer X-Touch controllers. OSC (Open Sound Control) protocol supports ethernet-based wireless tablet controllers such as TouchOSC and Lemur for iOS and Android to remote-control Mixbus. Lua Scripting: Use the Lua scripting language to automate tasks such as track creation, renaming tracks, adding plugins, or creating "macro" operations. VCA Faders: Use VCA faders to control multiple channel faders at once. A fader can be assigned to multiple remote VCA faders, adding another powerful dimension of control and organization to your mixes. Tempo Ramps and Tempo Maps: Use advanced features to map your live recording's tempo and structure (no need to record to a click!) – or import a tempo map from MIDI. Included Plugins: General-MIDI synthesizer and 2 multi-output sampled drum instruments help you arrange a song immediately, without any other plugin instruments. Over 25 realtime MIDI-filter plug-ins for transposing, velocity scaling, delay effects, channel changing, etc. "SetBfree", a lovingly modeled Tonewheel Organ by x42. Harrison's Dyno-Mite envelope-designer plug-in for drums.

Occasionally, a manufacturer offers a product that is not only innovative, but has multi-use applications, ease-of-use, loads of functions and great sonic characteristics. Aphex's new Model 1788, a quality eight-channel remote-controlled microphone preamplifier, is just such a product. The Aphex 1788 is 3.5" high, 19" wide and 13.25" deep. It weighs in at 14 lb and occupies two rack-spaces. The front panel of the Model 1788 is arranged in five separate blocks (local, control, digital, monitor and test). The local block includes the individual channel status readouts. At the top of each channel is a two-digit gain readout LED that shows the input or output gain setting, depending on the status of the gain select button. Below that, on the right side, is a vertical 12-segment LED meter that indicates the available peak headroom, in 4 dB increments from 0 to 44. On the left of the LED meter are several key functions: polarity reverse, 75 Hz low cut, 26 dB pad, 48 V phantom power and microphone output limiter. An associated LED lights when any of these functions are being used for each channel. Buttons and a rotary knob on the right of the eight-channel display engage or adjust these indicators. Under each of the eight channels is a channel select button. Located to the right of the local block is the control block, which selects communication ports and data rates. The Aphex 1788 using MIDI protocol via the RS-232, RS-422 or MIDI control ports controls up to 16 units at once. The RS-232 port is a three-wire, full duplex connection that can connect to a PC with runs between 50 and 100 ft. A Windows 95/98 remote program provides easy and maximum control of the Model 1788. The RS-422 is a full duplex differential transmission capable of runs up to 4,000 ft. A total of 128 channels of mic preamplification can be used with one control link. The adjust button in this section selects options for the functions (MIDI, set port, local control). A six-digit global status readout display is located next to an adjust button. The digital block is an optional module that includes an AES/EBU dB-15 digital output. It is possible to send all eight channels directly to a digital recorder or console. An ADAT digital connector is also included. A TASCAM TDIF DB-25 connector is available as an option ($1,000) for direct connection to digital recorders. This option also sets the synchronization source and sample rate. A clock button determines either the internal or an external clock as a synchronization reference. Internal clock rates can be set to 32 kohm, 44.1 kohm and 48 kohm. The monitor block makes it possible to monitor each channel separately through headphones. The test block is used to adjust system levels with a 700 Hz test tone. The Aphex 1788 rear panel consists of a power receptacle that works with a voltage range of 80 to 280 VAC. The eight XLR male input connectors are transformer-balanced with Jensen transformers. The output section provides two separate analog outputs per channel, eight servo-balanced XLR-type transformer-less connectors and a DB-25 multipin connector, transformerless and balanced. The analog DB-25 connector is marked "aux output" on the rear panel. This allows the inputs to be split and used in live sound for monitors and front of house (FOH) from an optimized microphone preamp, a very nice feature. All XLR-type connectors are wired 2-pin positive (signal). The output levels of the Aphex 1788 (main and aux) are adjustable from +27 dBu to 0 dBu, with sensitive 1 dB increments. The Aphex 1788 is a full-featured, 8-channel, remotely-controllable microphone preamplifier offering excellent sound quality plus two inventions: continuously variable gain control without "zippering" noise and a peak limiter that offers additional dynamic range without clipping the preamp's input. In addition to 26 to 65 dB of variable gain, a 26dB pad, polarity reverse, a 75Hz low-cut filter and 12-segment LED headroom meter, each channel includes a unique microphone limiter (MicLim[superscript]TM) before the preamp, preventing clipping altogether and allowing higher gain settings. The main and auxiliary outputs are individually adjustable to match the limiter's operation to the input headroom of connected analog devices. The limiter also assures that the optional digital outputs never hit 'Over.' All controls can be accessed from the front panel. Channels can be grouped to adjust the parameters or gain of several at once. A headphone amp allows local monitoring. A 700Hz test tone at either -20 or 0 dBfs can be assigned to any channel. Remote control of up to 16 units (128 channels) can be handled with a single MIDI, RS-232 or RS-422 control line via MIDI protocol. The controller can be anything that talks MIDI or a personal computer. An optional dedicated remote controller, the 1788-R, duplicates the front panel controls and adds clip/limit indication and device selection for 16 units, as well as preset loading and editing. Multiple 1788-Rs can be used in serial with or without a PC, allowing control from various remote locations. 1788 Program Software for Windows 95 is included, providing screens for control and status, or saving and recalling scenes. Its main window duplicates the front panel's control and metering for each device, including grouping functions. In addition to real-time metering, the software displays either peak-hold or absolute maximum levels, updated every half second. It allows up to 20 scene changes, each linked to presets that are named in software, and each snapshot is a MIDI dump containing all device parameters. As the preamps offer glitchless gain changes, scenes are smoothly ramped without muting. The System Setup window is an alternative to the master window that simplifies comparing the settings or programming multiple units. All settings of four 1788s, including channel and device names, can be seen at once, and for larger systems this window scrolls. Each channel has two independently adjustable balanced outputs. A digital option-24-bit sampled at 32, 44.1 or 48kHz-has three outputs simultaneously: AES/EBU, ADAT optical and TDIF. Connections are eight XLR inputs, eight XLR outputs and DB-25 (Tascam) for the auxiliary outputs. The digital option has AES/EBU on a DB-15, ADAT optical, TDIF on DB-25 and Word Clock in/out on BNCs. The 1788 is a cost-effective solution for touring acts that want to record their shows as it offers direct digital connection to recorders while the main out is connected to the FOH and the Aux out routed to the monitors. It can substitute for conventional passive splits and offers enhancement over active splits. Placing the preamp closer to the microphone reduces mode noise because gain does not have to be taken at the end of the run. Frequency response effects of capacitance in long mic lines are reduced. Also, loading a mic into a single preamp allows higher quality than loading that mic into several preamps through a splitter. Remote control allows use of fiber systems without an engineer babysitting the preamps. An internal jumper allows users to "unload" this transformer for a more colorful character, though I prefer the honest transparency offered by the factory-default setting. I used it for several vocal applications, from choir mics to handhelds, where the 75Hz highpass was convenient and the mic limiters were a luxury. It was also used on a Pavarotti show for the two Schoeps MK-4 vocal mics, substituting for the usual active splitter, where it offered an open, clear sound, earning compliments from both the artist and the audience. The auxiliary output allowed separate connection to two more channels in the console, used strictly for foldback. It was also used at field level on shotgun mics used at the SuperBowl for crowd ambience. 8-channel preamp with 60 dB gain, 8 input transformers JENSEN 48V,-20dB pad, Phase Reverse, low cut, limit, mute - 8 XLR outputs with separate gain, - 8 analog outputs on DB-25 with separate gain also - 8 AES outputs on DB15 ABB - 8 ADAT - 8 TDIF outputs on sub-D 25 sync I / O rs 232, 422, noon and MLAN

The B100 is designed for use when high quality stereo recording or editing is required. A compact unit, it is equally suited to studio or mobile work. Standard features include a built-in loudspeaker, a phase meter and insert points on mono and master outputs. A totally analog broadcast desk.Perfect for recording/monitoring needs, or to use as a summing mixer in our studio.Very quiet.

The EQ MAGIX is a single-channel equalizer system consisting of a para metric and a psychoacoustic equalizer. Both are combined with a superior microphone preamplifier and an active two-channel panorama output stage. The EQ MAGIX is ideally suited for both a demanding instrumentalist or a singer's rack. Existing equipment can be looped into the signal path INSERT. The unit's outputs are either connected directly with the recording medium or with one to four (2x balanced and 2x unbalanced) mixing channels. From there, the signal is fed into the Insert Return after the section (provided that appropriate Insert points are available), so as bypass as much desk circuitry as possible. As you know, a signal is influenced if all the switches are on »off« and all controls are in the centre position or on »0«. That is why it is advisable to feed the signal in after desk equalizer. The technology in the microphone preamplifier is used in the very latest generation of mixing desks made by Harrison and SSL. The parametric equalizer of the EQ MAGIX is based on SVF design (state-variable-filter) and provides a operation frequency range between 14 Hz and 58 kHz. The integrated VITALIZER was developed by SPL and has achieved a significant degree of popularity over the years. The filter concept works on the basis of psychoacoustic sound principles and is therefore ideal to achieve creative sound effects. The combination of a flexible parametric EQ and VITALIZER means that the EQ MAGIX is not only predestined to process the sounds of acoustic instruments but also to post-process samples or poor quality Sync audio tracks, ranging from the elimination of aliasing artefacts or other digital interferences to the addition of sub basses or comprehensive sound impro- vements via the VITALIZER. The EQ MAGIX provides you with equalizer technology which is unri- valled in comparable units. You will find that this equalizer can not only be set quickly and accurately at the desired frequencies but, with the aid of loudness sound control, can also form sound patterns which cannot be achieved with conventional equalizers. The section below contains a brief description of the EQ MAGIX's func- tion groups: 1. INPUT: Microphone and line level signals can be preamplified in the INPUT section. Microphone signals can be preamplified up to 65 dB. Activating the LINE switch attenuates the preamplifiers stage by -30 dB. There is a phantom power supply (48 V) for condensor microphones. The PHASE REVERSE switch is used to change the polarity of the microphone. The PEAK LED provides adequate warning of potential overdriving of the output stages in the INPUT section. 2. INSERT: Additional sound processing equipment such as a compressor, de-esser or reverb can be looped into the EQ MAGIX via the INSERT jacks. The INPUT section can be used separately for one signal, whilst the equa- lizer sections are available for a second signal. Use the INSERT SEND as the output for the INPUT section and the INSERT RETURN as the input for the second signal. 3. PARAMETRIC: The PARAMETRIC equalizer is to boost or cut different frequencies or to cut out specific interference noise signals. It is, in a manner of speaking, a corrective instrument in the EQ MAGIX chain of sound-processing tools. The PARAMETRIC works according to the constant-Q principle. The frequency ranges between 14 Hz and 58 kHz. The control characteristics of the bandwidth (Q) control ranges from Q0.6 to Q3.6 and can be sensitized via the Q-FINE switch to set more subtle Q values. The boost or cut range is max. +/-15dB. The integrated NOTCH filter can be controlled in frequency, bandwidth and also in the filter slope (ROLL-OFF) from gentle decreasing or increasing (GENTLE) to steeply decreasing or increasing (STEEP). This significantly increases the tonal flexibility of the NOTCH Filter. 4. VITALIZER®®: The VITALIZER is an equalizer concept which makes use of scientific psychoacoustic technology to process audio signals. The unique combination of dynamic equalizers, amplitude-controlled phase correction and harmonic filtering in the VITALIZER opens up a whole new listening dimension by adapting the sound pattern to the non-linearities of the human ear. The VITALIZER is thus the creative element amongst the EQ MAGIX's sound processing tools. The VITALIZER gives the mid frequencies accurate transparency with a soft, unobtrusive sound. The treble and harmonics range is reworked with broadband shelving filters of 2nd and 4th order, which focus on achieving a smooth, silky sound pattern. High frequencies are livened up without them sounding hard or agressive. The VITALIZER uses a newly developed filter network to process the low frequency range. The bass is accentuated without any risk of emphasizing the lower mid frequencies unnaturally. You can choose between a dry, percussive bass (TIGHT) or a punchy, soft and very deep sound character (SOFT). The result is a pleasanter and livelier sound pattern with an unrivalled wealth of detail. You will be amazed by how transparent and powerful almost every audio signal sounds after it has been processed with the VITALIZER! 5. OUTPUT: The OUTPUT section enables you to create a two-channel output signal from a one-channel input signal. You can return the signal to two desk channels or feed it directly into a multi-track recording medium or effect unit. Balanced outputs available via XLR sockets and unbalanced outputs available via jacks can be used in parallel. A signal can therefore be recorded on two media simultaneously. On the front panel, two bargraph meters each featuring 10 LEDs provide information about the output level of each channel. The indication of PPM values (peak values) provide more information than a VU display (average values). The MUTE switch can be used to mute the output signal altogether. The PANORAMA control allows active panning between the two output channels, compensating the loss in volume (- 3dB) which arises in conjunction with panning to one channel. The FADER switch alters the characteristics of the PANORAMA control, making it an output level control. The output level can be reduced for both channels to compensate diffe- rences in the levels of the processed and the dry signal. The EQ MAGIX features a PARAMETRIC constant-Q equaliser combined with the VITALIZER equaliser system. This combination of filters is based on the assumption that after succes- sful microphoning, you only have one problem zone, if at all, that needs to be equalized. The VITALIZER provides all the necessary circuitry for creative sound post-processing. The EQ MAGIX provides a total of five processing ranges: the PARAME- TRIC section provides an extremely variable and efficient filter for selective intervention in a frequency band between 14Hz and 58kHz. The VITA- LIZER interferes in 4 frequency ranges, bass, mid and treble, along with the harmonical frequencies pertaining to even and odd overtones. The PARAMETRIC section is designed to clean up one »problem zone« in the frequency spectrum of the recorded instrument or voice perfectly, without prejudicing the sound and without being inhibited by technical limitations of standardized filter circuits. The equalizer used in the EQ MAGIX is based on the state-variable filter type (SVF). Its function is phase-stable and neutral right down to 14 Hz in the infra-sonic range. The frequency range extends right up to breath- taking 58 kHz! This initially provides a platform for experimentation as to whether interspersing digital components affects into the tonal perception range (16 Hz to 20 kHz) or whether the removal thereof can remedy sound problems directly or indirectly. We have visited sound engineers and producers in European and American audio and video studios and they pointed out to us that inter- spersing the sample rate frequencies (44.1 kHz and 48 kHz) into the audio path is a well-known phenomenon, whereby subharmonic elements can be transposed right down into the audible frequency spectrum. Furthermore, this can result in audible phase shifts and distortions. With the EQ MAGIX, you can now influence these frequencies actively. The sample rate frequencies are inscribed on the scale (44, 48). All frequencies can be boosted or cut with 15dB throughout the entire frequency range from 14 Hz to 58 kHz. The PARAMETRIC equalizer of the EQ MAGIX is based on the constant-Q principle and provides an »inversible Q«, i.e. when filtering out a interfering frequency, inverting the boost value to a cut value results in precise inversion of the bandwidth (Q) values. Notching is also possible with a variable ROLL-OFF to select a STEEP or GENTLE filter slope. Extremely flexible bandwidth defintion allows Q values between 0.6 and 3.6. The VITALIZER is the creative sound component in the EQ MAGIX. It works »intelligently«, provided the appropriate parameters are preselected. This means that it does not just raise or lower a certain frequency range but analyses the signal and reacts to it, influencing the frequency and phase response in a sophisticated fashion. Independently of the potentiometer settings on the front panel, the VITALIZER filter system can correlate the phase and amplitude relations of adjacent frequency bands in such a way that the increasing amplitude of a frequency is given a growing phase drift which, however, can never endanger the signal's mono compatibility. The maximum phase drift is +/- 20° which equals a time difference of around 1 ms to 2.5 ms. This time span is sufficient to trigger two hearing perceptions. Quiet frequencies which are normally masked by louder ones are effectively demasked and thus become audible. The front panel also accommodates controls to define the bass sound colour (SOFT/TIGHT), the starting frequency of the MID-HI TUNE filter, the PROCESS intensity and the addition of the harmonics. The BASS control provides up to 20 dB gain, thus affecting the direction of the phase drift by the bass sound colour selected. The fact that the phase drift is controlled by the amplitude means that even high bass amplifica- tion remains free from negative sound colouration in the lower mid range. The mid-high frequency filter (MID-HI TUNE) works with shelving filters which provide »linear« broadband amplification. This rules out any comb filter effects which create sound distortions similar to those produced by graphic equalizers. The PROCESS control determines the addition of the bass sound selected and the MID-HI TUNE filter to the dry signal. In addition, dominant mid frequencies are lowered by phase shifts which means that as the PROCESS value increases, they are adapted automatically to curves of equal loudness (Fletcher-Munson curves). The PROCESS is, in a manner of spea- king, an »intelligent« loudness control which can be used to set the typical »bath-tub« curves on graphic EQ's by turning one potentiometer. The HARMONICS circuitry provides a silky sounding, refreshed version of the original signal. This proves particularly beneficial for voices and acoustic instruments. No distortions are added to the original as there is no Exciter involved. The filters merely amplify the harmonic tones contained in the original. The EQ MAGIX is constructed like a mixing desk channel. Direct recor- dings can be made on to a recording medium (multi-track machine or hard disk) which means that one less mixer channel is required.

High quality reverberation... The Lexicon PCM 91 gives you Lexicon’s renowned high-end reverb effects with a powerful new interface that provides easy access to superbly crafted presets as well as a wealth of programming capabilities for the sound designer. The Lexicon PCM 91 contains a built-in library of 450 reverb effects that simulate realistic halls, rooms and plates, and let you create completely natural, or other-worldly spaces. The presets are organized into 9 Banks of 50, and are functionally grouped for different applications. Be sure to experiment with all 450 presets to get a feel for the full range of Lexicon PCM 91 capabilities. A program sorting function allows you to tag programs with KeyWords and display only programs which have been tagged. (Press Program Banks or Register Banks repeatedly to step through all available banks and then to the KeyWord display.) The default KeyWord selection, A to Z, allows you to view all of the presets in alphabetical order. Others allow you to view, for example, only Acoustic or Spatial effects. Each preset has already been assigned from 1 to 4 KeyWords – you can easily change these assignments in Edit mode. The selection of the KeyWord you want to use for sorting is accessed in Control mode. Each preset has one or more of its parameters patched to the front panel ADJUST knob, giving you instant control over the primary aspect of the effect – without going into Edit mode. As many as four additional Custom Controls can be created for any effect, allowing you to tailor presets for specific applications. We’ve created some interesting Custom Controls in the presets, and assigned them descriptive names. You can change both the parameter assignments and the names in Edit mode. The Lexicon PCM 91 uses 5 stereo algorithms to create different types of reverb effects and 10 Dual Reverb algorithms which offer superb dual reverb and cascade configured stereo effects. Each single algorithm includes an uncompromised stereo reverb effect with selected “tools” for ambience, post-processing, compression/expansion, as well as modulation and patching parameters which are common to each algorithm. Each dual algorithm contains two independent reverb blocks, as well as the full set of modulation and patch features in the single effects. The Lexicon PCM 91 gives you a unique set of tempo controls.Tempos can be tapped in with the front panel Tap button (or an assigned controller) or “dialed-in”, in BPM (beats per minute) on the display. The Lexicon PCM 91 also lets you generate MIDI clock from your tempo, as well as receive MIDI tempo from an external sequencer or drum machine. In the Lexicon PCM 91, tempo can control LFO speeds and Time Switch controls, as well as all delay parameters, ensuring that all of your modulations are in tempo with your music. You can even set independent rhythmic values for different parameters within a single program. Tempo can be set and displayed in either rhythmic value or time values. Many presets have delay times assigned to Tap tempo. Try loading some of these and pressing Tap twice in rhythm to change tempo. An enormous range of editing control is provided for each algorithm, with parameters organized in an edit matrix. In addition to providing this powerful sound design capability, the Lexicon PCM 91 also allows you to customize these controls for your day-to-day editing needs, or to use a subset of controls specially designed for each preset. The Lexicon PCM 91 has two levels of Edit Mode control called Go mode and Pro mode. In Go mode, the most useful parameters within an effect are grouped for instant access via the front panel Edit button. Parameters can even be grouped for control by a single master control. These master parameters, called Custom Controls, can be labeled with names that describe their function. Each preset has a specially selected set of Go mode parameters which let you make value changes to the effect without losing the character of the sound. Pro mode gives you access to the full parameter editing matrix for the algorithm of any loaded effect when you press Edit. In this mode, you can access a complete set of Modulation and Patching parameters, create your own ADJUST knob patch, create Custom Controls, and assign your own Go mode parameters. A unique Patching and Modulation system provides unprecedented control over your effects, with a versatile set of internal modulators: two LFOs, AR Envelope, Envelope Follower, Latch and Time Switches, MIDI Delay and Sample and Hold. These allow you to create modulation sweeps which move in time with music, or animated effects. You can create as many as 10 patches per effect, each with as many as 8 pivot points. You can patch multiple parameters to a single controller, or patch multiple sources to a single destination. For all of its programming power and flexibility, you’ll find the Lexicon PCM 91 simple to use. The large, 2-line fluorescent display is easy to see from any angle whether the surroundings are bright or dark. Separate SELECT and ADJUST knobs make program loading and editing quick and easy. We’ve even designed in a special Info mode – press and hold any button to find out what its function is, or to get status information such as the name of the running effect, current tempo rate, etc. Analog Audio Input XLR and 1/4" balanced (T/R/S) Input Level -2dBu to +20dBu, balanced;<br>-22dBu to 0dBu, unbalanced Input Impedance 100k&Omega;, balanced;50k&Omega;, unbalanced Analog Audio Output XLR and 1/4"T/R/S balanced Output Level +18dBm, balanced;+4dBm, unbalanced Output Impedance 125&Omega;, balanced Digital Audio Input XLR, AES/EBU;<br>Coaxial RCA, S/PDIF Format AES/EBU, balanced;<br>S/PDIF, unbalanced Sample Rate 44.1kHz and 48kHz Digital Audio Output XLR, AES/EBU;<br>Coaxial RCA, S/PDIF Format AES/EBU, balanced;<br>S/PDIF, unbalanced Sample Rate 44.1kHz and 48kHz Conversion 24-bit A/D;24-bit D/A Internal Audio DSP 24-bit (PCM 81);20-bit (PCM 91) Frequency Response 10Hz - 20kHz &plusmn;.5dB Crosstalk -55dB 10Hz to 20kHz THD &lt;0.006%, 10Hz - 20kHz Dynamic Range D/A:&gt;98dB typical, 10Hz - 20kHz<br>A/D:&gt;105dB typical, 10Hz- 20kHz<br>A/A:&gt;96dB typical, 10Hz - 20kHz MIDI Interface 5-pin DIN connectors for MIDI IN, THRU and OUT Footswitch 1/4"T/R/S connector for two independent momentary footswitches (system detects normal-open, or normal-closed on power up) Foot Controller 1/4"T/R/S connector for foot pedal(100&Omega;- 10k&Omega;impedance) Memory Card Connector PCMCIA type l cards, 68 pins Standards PCMCIA 2.0 and JEIDA 4.0 Card Format Up to 1MB of SRAM (attribute memory not required) Power Requirements 100 - 240 volts AC;50 - 60Hz;35 watts<br>Automatic switching (3-pin IEC connector) Remote Power In 2.5 mm 9 volts AC (not included) Dimensions 19" W x 1.75" H (1U) x 12" D<br>(483 x 45 x 305 mm), rack mount standard Weight 6.4 lbs.(2.9 kg) Operating Temperature 32&deg;to 104&deg;F (0&deg;to 40&deg;C) Maximum Humidity 95% without condensation

The Lexicon 300 not only contains the finest sounds, as you’ve come to expect from Lexicon, it incorporates new functions that satisfy the needs of today’s audio production. Analog and Digital Audio Interfacing For both analog and digital use the Lexicon 300 takes full advantage of recent advances in converter technology and combines them with flexible digital interfacing. The A/D and D/A converters use oversampling techniques to minimize low-level distortion and provide linear phase characteristics. 64x oversampling Delta/Sigma conversion is used in the A/Ds; 8x oversampling is used in the D/As. The result is sonic transparency without the artifacts normally encountered in conversion. The digital I/O simplifies interfacing with both the AES/EBU professional and the EIAJ consumer format. XLR, RCA and Optical connectors are provided for both input and output. The digital inputs of the Lexicon 300 will automatically accept and lock to any AES/EBU/SPDIF format; the Lexicon 300’s digital output format is user-selectable for feeding either professional or consumer equipment. This allows for format conversion as well as using the high quality A/D converters to feed digital recorders. Control of additional Channel Status information is provided as well. Dual DSP Architecture The Lexicon 300 utilizes two proprietary high performance DSP engines. The configu- ration of these engines can be determined by the end-user to suit a particular audio application. The Single Setup configuration unites both DSP engines. Two “split” configurations: Dual Mono and Cascade, allow each DSP engine to run a special program especially designed for “Split “ applications. The Single algorithms include: Random Hall, Random Ambience, Rich Plate, Stereo Pitch Shift, and Stereo Adjust. The “split” algorithms include: Dual Delays, Chamber, Mono Pitch Shift, Compressor, a special mastering dither program called PONS (Psychoacoustically Optimized Noise Shaping), and a small version of Stereo Adjust. Comprehensive Effects Automation The 300 makes effects automation available in three different ways. If you use time code, the 300’s Event List and Time Code Reader add a new twist to effects processing. Each entry in the 50-item list contains a time code value and a 300 effect preset/register number, allowing incoming time code to trigger setup and effect changes, as well as setup and effect parameter glides. For changing effects in music or changing rooms and environments for film and video, the Lexicon 300’s time code capability opens new avenues to all time code users. Dynamic MIDI is included for real-time performance control and effects automation. All parameters in the Lexicon 300 can be patched to most MIDI controllers, allowing full control from remote devices. For full MIDI automation all parameter changes can be recorded on any sequencer. For additional control, complete System Exclusive information is available. Lexicon Sound Even with the best functions and features, the heart of any signal processor is its sound. The 300 contains the very best of the Lexicon Sound – with new refinements such as dynamic size parameters in the Reverb and Ambience algorithms. We have also included stereo and mono pitch shifting, delay effects, a mastering algorithm with precision level/balance controls and equalization. These sounds, combined with analog and digital audio I/O, time code based effect change, an internal LFO, and full MIDI automation result in a system that provides new creative options each time you use it.

MAX.INPUT LEVEL (XLR): +22 dBu MAX. OUTPUT LEVEL (XLR): +22 dBu MAX.INPUT LEVEL (JACK): +14 dBu MAX. OUTPUT LEVEL (JACK): +10 dBu INPUT IMPEDANCE (JACK/XLR): 1 MOhm unbal. / 20 KOhm bal OUTPUT IMPEDANCE (JACK/XLR): 750 Ohm unbal./ 50 Ohm Electronically bal. FREQUENCY RESPONSE: 20-20 KHz, +0 / -0.5 dB, 25 KHz, -3dB TOTAL HARMONIC DIST.: < 0.05% 1 kHz, 0 dBm DYNAMIC RANGE: >100 dB GROUP DELAY: <30 microseconds DELAY RESOLUTION: 0.1 mSec to 999.9 mSec. then 1 mSec DIGITAL-CONVERSION: Dynamic Differential SAMPLING RATE: 1 MHz ENVIRONMENT: Operating, 0¡C to 50¡C STORAGE: -20¡C to 60¡C POWER REQUIREMENTS: Selectable: 100/120/ 220/ 240V, 50-60 Hz, 30W DIMENSIONS: 3 1/2 x 19 x 9.5 inches (89 x 482 x 242 mm) FINISH: Black anodized aluminum face PLATE: Black painted steel chassis NET WEIGHT: 12 lbs (5.7 kg) SHIPPING WEIGHT: 19 lbs (6.4 kg

I love using this on groups of vocals or anything that ''breaths''.When used with wisdom and fineness the unit is capable of creating effects you’ve never heard before.True stereo image presentation - highly futuristic!

Eclipse Transformer Connections: XLR inputs, D-Sub outputs, in addition the inputs and outputs are available as "Phoenix" installation terminals Format: 19"/ 1U Rear pannel D-sub and Phoenix screw terminals Ideal for live recording and signal distribution Jensen transformers Choice of 19" rack or frame mounted designs

The Radial Phazerbank is a 4-channel analogue phase adjustment tool that enables the engineer to effortlessly adjust the phase between two microphones when double mic'ing a single source. Once applied, the result is fatter sounding guitars, a more solid bass, tighter kick and more realistic sounding piano. ''I honestly didn’t think I’d have a whole lot to say about the Phazer when I started this review. But now – different microphones, different arrangements, different instruments (can’t wait to try drums), different polar patterns, direct and mic combinations, and on and on. But isn’t that the point? Just when you thought you knew your mic collection and had worked everything out, something like the Phazer comes along to tell you that just isn’t so. I also admit to pursuing the idea that ‘every tweak must have a predictable effect’ a little too far. I tried more metering than should be legal, to see if I could find method. In the end I had to concede that the Phazer is simply a little box that can make a big difference to your sound. It’s an effect, an EQ, a correction tool, and more besides; but the most important concept is that it works on two channels by working on one… Because phase is relative… Sort of. Anyway, you should get one'' PAUL MAC is the Editor of Audio Media magazine.

The AKG K 121 Studio are entry level professional headphones for various applications, as Project Studios and Home Recording.These semi-open, supraaural headphones are a true allround performer. The AKG self-adjusting headband ensures good comfort.The K 121 Studio are equipped with a 3.5/6.3 mm stereo adapter and is designed for ambitious and rugged use. The leatherette ear pads are easy to clean. General Information Manufacturer Website Address http://www.harman.com Brand Name AKG Product Type Headphone Manufacturer Harman International Industries, Inc Product Line Studio Manufacturer Part Number K-121 Product Name K 121 Studio Product Model K 121 Technical Information Sound Mode Stereo Minimum Frequency Response 18 Hz Connectivity Technology Wired Impedance 55 Ohm Cable Length 3 m Maximum Frequency Response 22.50 kHz Earpiece Earpiece Design Over-the-head Earpiece Type Binaural Physical Characteristics Form Factor Supra-aural Weight Approximate 220 g Colour Black, Gold Interfaces-Ports Host Interface Mini-phone

Active 2-Way Nearfield Monitor 8" Woofer (75 W) + 1" Tweeter (45 W) Power: 120 W biamp Bass reflex system Frequency range: 38 Hz - 30,000 Hz Inputs: balanced XLR and 6.3 mm balanced jack Adjustable input level Room control and high trim response control Dimensions (WxDxH): 250 x 390 x 334 mm Weight: 10.2 kg

The SuperNova II (2000) was available in 24, 36 and 48-voice models with additional 12 or 24-voice expansion boards. Available again in a 3U rack-mount format, the SuperNova II was also available in a 61-note performance keyboard version with velocity and aftertouch, enabling sound tweaking during live performance. FM synthesis capability was included along with ring modulation, dual analogue inputs and a 42-band vocoder. An 8-part arpeggiator was also on-board and in its full version the product offered 8-part multitimbrality and 48-voice polyphony, with 57 and 2304 oscillators running simultaneously. Architecture Type: Digital Synthesis: Subtractive, Virtual Analog Oscillators (+) Open Manual Page Oscillators: 3 Osc Modulation: Envelope, Keyboard, Knob, LFO, Mod Wheel, Oscillator, Ring Modulation, Sync Hard, Velocity Envelopes Envelopes: 3 Evelope Paramerters: Delay, Attack, Decay, Sustain, Release, Freerun, Looping, Time Filters Filters: 2 Types: 12dB Slope (2-pole), 18dB Slope, 24dB Slope (4-pole), Band Pass, High Pass, Low Pass Filter Modulation: Envelope, Knob, LFO, Mod Wheel, Velocity Filter Notes: + 9 Special dual type filters with modulatable frequency separation. + Overdrive control capable of subtle “Saturation” LFO LFO: 2 LFO Parameters: Sample & Hold, Saw Up, Saw Down, Sine, Square, Triangle, Delay Polyphony & Tuning Polyphony: 24 - 48 Timbrality: 8 Tuning: Standard Modes: Mono, Polyphonic Patches Patches RAM: 1 Storage: Internal Arpeggiator Built in Arpeggiator with 128 Preset Polyphonic, 128 Preset Monophonic and 192 User Programmable Polyphonic and 192 User Programmable Monophonic Patterns with 8 Arpeggiators available simultaneously in Performance mode. Effects Distortion, Comb Filter, EQ, Reverb, Chorus/Flanger/Phaser/Ensemble/Rotary Speaker, Delay and Panning effects for each Program. 57 Effects available simultaneously - 7 Effects for each of the 8 Parts of a Performance or Program, Plus 1 Global 42 Band Vocoder. Yes you can have a different Reverb on each part, a different Delay on each part, a different Chorus etc. Vocoder 42 Band Vocoder with Sibilance modes and Spectum Analiser. 18 different Effects Configurations available on each Part of a performance or Program. Realtime Morphing between Effects Configurations is also possible. Chips and Operating System 1.4 Case Case: Rack Rack Size: 19", 3U, Full Controls: Buttons, Knobs, Pedal - Sustain, Tempo Knob, Velocity, Pedal - Volume Display Type: LCD, LED, Backlit Display Count H: 20 Display Count V: 2 Dimensions (WxDxH): Width 483 mm. Depth 222mm. Height 133mm Weight: 5kg Connections Audio Output Connections: 1/4" Phone Jack, Stereo Main, Stereo 2, Stereo Headphone Inputs: 2 assignable Intputs with high quality ADCs. MIDI Ports: IN, OUT, THRU Power: 90V to 260V. 40 Watts. Production Year Released: 2000 Year Discontinued: 2001