BEHRINGER FX2000 3D EFFECTS VIRTUALIZER

VIRTUALIZER 3D FX2000
High-Performance 3D Multi-Engine Effects Processor
71 breathtaking new algorithms—true RSM (Real Sound Modeling) stereo and 3D effects
Wave-adaptive virtual Room reverb algorithms for natural reverb and delay
Awesome modulation, dynamic, psychoacoustic and EQ algorithms
Authentic amp simulation, distortion and special effects
11 effect combinations with selectable serial / parallel configuration
Up to 7 adjustable parameters plus 2-band EQ per effect
24-bit A/D and D/A converters with 64/128-times oversampling
True stereo processing for realistic channel separation in stereo image
100 factory presets plus 100 user memory locations
Extensive MIDI implementation
Accurate LED level meters for perfect level setting and optimum performance
Servo-balanced XLR and ¼'' TRS inputs and outputs
High-quality components and exceptionally rugged construction ensure long life
Conceived and designed by BEHRINGER Germany

 Reverbs & Delays

Because it is one of the most desired effects, the FX2000 provides 12 different reverb programs, so you always have the ideal reverb for your live performance or studio needs.

Reverbs include:

    CATHEDRAL
    GOLD PLATE
    SMALL HALL
    ROOM
    STUDIO
    CONCERT
    STUDIO
    STAGE
    SPRING REVERB
    AMBIENCE
    EARLY REFLECTIONS

Specialty reverbs include:

    REVERB – reverb is synthetically turned off after a predetermined amount of time
    REVERSE REVERB – reverb envelope is reversed—it slowly gets louder

Delays include:

    STEREO DELAY – delay is processed across the entire stereo image
    TAPE ECHO – simulates classic tape echo devices, pre-dating the advent of digital delays
    PING PONG – delay signal is "bounced” from left to right at an adjustable tempo

Much More than Just Reverb and Delay

The FX2000 has some of the best-sounding reverb and delay programs, but thatʼs not all. It also provides excellent modulation effects (such as chorus, flanger and phaser), including special variations like musical pitch shifter, tremolo and even a rotary speaker simulation.

Modulation and Pitch Shifter FX:

    STEREO FLANGER (originally generated by playing back two synchronized "reel to reel” tape recorders with a finger rubbing on the flange of one of the take-up reels)
    VINTAGE FLANGER simulates a guitar fl anger stomp box
    JET STREAM FLANGER sounds like a classic analog flanger
    STEREO CHORUS (combines a slightly detuned signal with the original)
    ANALOG CHORUS simulates a guitar chorus stomp box
    VINTAGE CHORUS imitates a classic analog studio chorus
    ULTRA CHORUS creates the sound of an eight person chorus
    STEREO PHASER (combines a second, phaseshifted signal to the original)
    VINTAGE PHASER represents a guitar phaser stomp box
    DUAL PHASER processes the left and right channels separately
    ROTARY (simulation of the rotating speakers typically used on an organ) – Slow or Fast
    PITCH SHIFTER (changes the pitch of the original signal) – can be used to create harmonies with the original signal or replace it entirely with the altered pitch. Choices include stereo, two and three vocal pitch shifter
    VIBRATO – the peak frequency of the tone is periodically and uniformly changed (quickly or slowly)
    TREMOLO (common vintage guitar amplifier effect) – a fast or slow periodic variation in volume
    AUTO PANNING – signal is automatically sent from one side of the stereo image to the other, either once or multiple times

Dynamic FX:

    COMPRESSOR – reduces the dynamic range of the signal, maintains consistent signal level and thus avoids distortion associated with excessive input levels
    EXPANDER – effectively broadens the dynamic range of source signals while reducing background noise
    GATED REVERB – helps reduce background clutter by turning reverb off below a predetermined threshold level. Particularly effective on drum mics and vocals
    ANA. KOMPR/LIM. — similar to COMPRESSOR but with Limiting functionality
    ULTRAMIZER – analyses incoming signal and automatically applies compression across two independent frequency bands
    DENOISER – eliminates or reduces noise and other interference
    DE-ESSER – reduces or removes sibilance (Ssss sound) from signal
    WAVE DESIGNER – allows you to influence the envelope by adjusting Attack and Release of the signal

Psychoacoustic FX:

    EXCITER – adds artificially generated overtones to the original signal, increasing presence and perceived loudness without significant increase in signal level
    ENHANCER – functions much like a dynamic pitch equalizer
    ULTRA BASS – sub-harmonic processor combined with bass exciter and limiter
    STEREO IMAGER – divides the signal into middle and side signals, allowing individual signals to be amplif ed when desired and placed on the stereo image
    ULTRA WIDE – creates a broader stereo image
    BINAURALIZER – also creates a broader stereo image and compensates for crosstalk between both speakers

Filter/EQ FX:

    AUTO FILTER – influences the frequency response of a signal. Two filters are included: low pass allows low frequencies to pass and suppresses high frequency content; high pass does the exact opposite
    LFO FILTER - Controls the rate of oscillation effects
    PARAMETRIC EQ – allows you to control the bandwidth, frequency and amplitude of a signal
    GRAPHIC EQ – the sound spectrum is divided into eight (8) adjacent frequency bands, which can be cut or boosted, bandwidth is predetermined

Distortion FX and Amp Simulations

The FX2000 is also equipped with distortion, amplifier and speaker simulation including VOCAL DISTORTION, TUBE DISTORTION, GUITAR AMP, TRI FUZZ, SPEAKER SIMULATION, RING MODULATOR and LO-FI.

Special FX:

    VINYLIZER – adds clicks and/or noise to the signal, reminiscent of old vinyl records and tape machines
    SAMPLER – allows you to record and playback up to five (5) seconds of program material
    VOCODER – allows the input signal to modulate another signal (usually a synthesizer sound), creating the familiar "talking synthesizer” effect
    VOICE CANCELER – removes mono vocal parts from stereo recordings for "instant Karaoke”
    RESONATOR – simulates and oscillating system that amplifies a specific frequency

Our FX Combinations Go to 11

Sometimes you want to add a little (or a lot) of color to your reverb and delay patches. The FX2000 allows you to layer modulation effects like chorus, flanger, pitch or tremolo with your reverb or delay selections.

FX Combinations include:

    Chorus & Reverb
    Flanger & Reverb
    Leslie & Reverb
    Pitch & Reverb
    Delay & Reverb
    Tremolo & Reverb
    Phaser & Reverb
    Chorus & Delay
    Flanger & Delay
    Pitch & Delay
    Tremolo & Delay

Enhanced User Editability
Logical grouping of parameters, along with the combination of encoders, buttons, LEDs and an easily readable LED display, make operating the FX2000 a breeze. You are free to edit up to seven parameters per preset, and then save them for future use in the 100 provided memory locations.
Value
With its extremely powerful processing capability and versatile array of features, the FX2000 will become the busiest tool in your audio arsenal. Stop by your BEHRINGER dealer today and find out why more professional sound engineers are turning to the FX2000 as their primary FX processor— both in the studio and on the road.

Psychoacoustic Bass Enhancement Unit with Renaissance Compressor, MaxxBass, and L2

Bass Boost for Live Sound, Broadcast, Mastering & Post Production
They say that in electronics there's no new technology - the only changes
will be in size and applications. Things will simply keep getting smaller
while doing more. This is certainly true in music (just think about what an
iPod can do), but as we design smaller playback systems with small speakers,
there are also some tradeoffs, particularly in music. Clever cabinet designs and
high-excursion drivers can get you only so far. Any audio engineer trying to extract
bass from real-world components must eventually face the laws of physics:
All things being equal, small boxes and speaker cones can't move enough air
to produce a room-shaking low end. When working with systems tiny enough to
toss in a shoulder bag, the challenge is even more formidable. Interestingly enough,
Waves MaxxBCL has a new solution to this problem using a very old concept.

Waves MaxxBCL at a Glance:
MaxxBass? Bass Enhancement adds deep bass sound without adding bass frequencies
Waves Renaissance Compressor
L2 Ultramaximizer Peak Limiter
24-bit/96kHz resolution with 48-bit, double-precision processing
Supports optical, coaxial S/PDIF, AES/EBU, balanced and unbalanced line-level analog signals

MaxxBass Bass Enhancement
Waves' patented algorithm adds stunning bass sound without adding bass frequencies,
delivering a bigger bottom end. MaxxBass? allows your audience to hear bass frequencies
lower than physically present by applying psychoacoustic principles: Even though we
can hear a bass guitar from a small speaker, we don't actually hear the fundamental
frequency because the speaker can't produce a pitch that low. Instead, we hear the
harmonics that the speaker can produce, and this causes the brain to create the "missing
fundamental." MaxxBass? takes this well-known psychoacoustic phenomenon to the
maximum, giving you the ability to extend the perceived frequency response of a system
about two octaves below its physical limitation.
To accomplish this, the signal is split: high frequencies are passed to the output (to be
added back to the bass). The bass signal is analyzed and a specific series of upper
harmonics are created. Because the dynamics of the original bass are duplicated in these
harmonics, the result is the most natural sounding bass enhancement available. The
MaxxBass? harmonics and the original bass can be mixed in any proportion at the output.
To provide more control, a high-pass filter can be switched in, allowing the harmonics only to
be passed along as an "image" of the original bass frequencies, which is useful when working
with a system with known low-frequency limitations to avoid over-excursion of the speaker drivers.

Renaissance Compressor
Designed to provide the classic warm sound of analog compressors, the legendary
Renaissance Compressor controls dynamics with studio-style warmth that flatters
full mixes, vocals, or instruments. Controls include selection of vintage-style Opto
or modern Electro compression, and threshold, ratio, and attack controls. Waves
ARC? (Automatic Release Control) algorithm dynamically optimizes the compressor's
release value for a wide-ranging input. ARC? reacts much the way a human ear
expects, and can produce increased RMS level with greater clarity.
In general, the release is faster for peak transients and slower for the overall RMS
level. The ARC? system varies the release time to fit the ear's expectations while
increasing RMS, and without creating distracting artifacts. In this way, the
Renaissance Compressor can serve as a leveler plus a fast compressor simultaneously.

L2 Ultramaximizer Peak Limiter
Heard on countless hit records and soundtracks, the L2 Ultramaximizer puts sound
up-front with breathtaking transparency. The L2 is capable of a very fast, overshoot-free
response. Once the limiter threshold has been set, you can define the actual peak level
that the processed signal will reach. Once set, limiting and level re-scaling becomes a
one-shot process. The L2 can significantly increase the average signal level without
introducing any audible side effects. Yet there is plenty of range to recreate "vintage"
effects such as level pumping or severely limited dynamic range if you like.

Pristine Sound Quality
The MaxxBCL offers 96kHz, 24-bit resolution with a 48-bit, double precision internal
processing path and a dynamic range of ~125dB. The totally passive analog input
path to the ADC uses Jensen transformers, while the output path also features Jensen
analog output transformers. The unit is galvanic-isolated, which prevents ground loops
and allows it to operate in electrically unstable environments. Waves MaxxBCL combines
the highest quality converters available with unequalled processing algorithms to offer you
astonishing new power in bass enhancement, dynamics processing, and format conversion
(analog-to-digital, digital-to-analog, and digital re-quantization).

Versatile Connections
The MaxxBCL supports optical, coaxial S/PDIF, AES/EBU, balanced and unbalanced
line-level analog signals. MaxxBCL features a unique set of input and output trim settings:
analog input headroom can be set from 9dB above 0dBu to 24dB in six precise steps
(using the rear-panel selectors). Analog output level is similarly set with a separate
rear-panel trim pot offering you the ability to connect between devices with a wide range
of input and output levels.

Waves MaxxBCL Features:
User Interface:
Clear backlit displays, meters, and buttons are visible in all lighting conditions
THD + Noise: ? 0.0006 % @ 1kHz @ -1dBFS
Precision metering with resettable peak hold options (2 sec, infinite)
Accurate, wide-range metering covering 90dB for input and output
and 12dB for compressor and limiter attenuation
Quick access independent bypass on each processing block
Tactile feedback knobs
Four easy store/recall user presets
Input / Output:
Analog, AES/EBU, S/PDIF
Independent rear-panel input and output headroom calibration in six steps
(+9, 12, 15, 18, 20, and 24 dBu)
Passive analog inputs to the A-to-D converters using Jensen analog input transformers
Jensen analog output transformers
High-performance IDR dithering to 16- or 24-bit output
Precise input level setting using 1% resistor networks
Processing:
48-bit end-to-end internal processing path
Switchable compressor/MaxxBass? order
MaxxBass:
Adjustable processor frequency from 25 to 120Hz
Adjustable harmonic mix percentage from 0 to 100%
High-pass Filter "harmonics only" option
Compressor:
Opto/Electro mode selection
Threshold (0-60dB), Ratio (1:1-12:1), and Attack (0.5, 1, 2, 5, 10, 20, and 50 ms) controls
Proprietary ARC? Automatic Release Control
Automatic gain makeup
Limiter:
Overshoot-free look-ahead processing
Adjustable Threshold (0-18dB)
Adjustable Output Ceiling (0-18dBFS)

Description

The alphaton Feedback-Controller FC-100 is a revolutionary product for every microphone
application including wireless types. It can provide effective feedback control to any sound
reinforcement situation. It can also eliminated string feedback in guitars and bass instruments
which are equipped with piezo or condenser pickups.
The FC-100 provides 6 to 10 dB of feedback protection by analysing the incoming signal,
detecting the presence of feedback and then notching out the feedback frequency.
The FC-100 relieves the sound engineer automatically. It works dynamically and eliminates a
feedback signal before it can grow into a peak. Internally a discriminator circuit detects the
presence of regenerative feedback and triggers the automatic equaliser to insert a notch where
the feedback peak is coming up. This normally prevents other peaks from developing but if they
do, the process is repeated at high speed, thus keeping feedback under control.

Technical Data
In- and Outputs transformer balanced
max. Inputlevel + 3 dB
Gain (RS=200 Ohm, RL=2kOhm) 0 dB
Frequency Response 20 Hz ... 20 kHz, +/- 1,5 dB
Feeback Cancellation  6 ... 10 dB
Power Requirement  30 ... 48 V Phantom
Current drain ca. 8 mA
External Power Supply 9 ... 20 V DC, 3,5 mm jack, tip +
Housing Aluminium Diecast
Weight  285 grams
Dimensions  112 x 60 x 32 mm

Shure DP11EQE Dynamics Processor Specification:

    GENERAL:
        Frequency Response: 20 to 20k Hz +/- 1.0 dB re 1 kHz
        Dynamic Range: 104 dB minimum, A-weighted, 20 Hz to 20 kHz
        Sampling Rate: 48 kHz
        Digital-to-Analog, Analog-to-Digital Conversion: 20 bit resolution
    Voltage Gain:
        1 dB +/- 1dB (power off)
        12 dB +/- 2 dB (input 10 dBV, output +4 dBu)
        12 dB +/- 2 dB (input +4 dBu, output 10 dBv)
        0 dB +/- 2 dB (equal input and output sensitivities)
    Impedance:
        Input: 47 kilohms +/- 20% actual
        Output: 120 ohms +/- 20% actual
    Input Clipping Level
        +18 dBu minimum (at +4 dBu setting)
        +6 dBV minimum (at 10 dBV setting)
    Output Clipping Level
        +18 dBu minimum (at +4 dBu setting)
        +6 dBV minimum (at 10 dBV setting)
    Total Harmonic Distortion
            LED Signal Indicators
        Signal: 40 dB
        Clip: 6 dB down from input clipping
    Propagation Delay from Input to Ouput
        0.8 ms (all filters flat, no dynamics processing, 0 ms delay), up to 2.1 ms (all processing enabled)
    Polarity
        Input to output: inverting optional (default: non-inverting)
        XLR: pin 2 positive with respect to pin 3
        1/4-inch TRS: tip positive with respect to ring
    Operating Voltage
        DP11EQ: 120 Vac, 50/60 Hz, 50 mA max
        DP11EQE: 230 Vac, 50/60 Hz, 25 mA max
        DP11EQJ: 100 Vac, 50/60 Hz, 50 mA max
    Temperature Range
        Operating: 0 degrees to 60 degrees C (32 degrees to 140 degrees F)
    Fuse
        DP11EQ: 120 Vac. Fuse: 100 mA, 250V time delay
        DP11EQE: 230 Vac. Fuse: 50 mA, 250 V time delay
        DP11EQJ: 100 Vac. Fuse: 100 mA, 250 V time delay
    Dimensions
        219 mm x 137 mm x 44 mm (8-5/8 in x 5-3/8 in x 1-3/4 in)
        Weight: 930 g (2.05 lbs)
    DYNAMICS PROCESSOR
        Gate and Expander:
        Threshold: 72 to 1 dB, 0.5 dB resolution
        Attack: 1.0 to 200 ms
        Decay: 0.05 to 1 second
        Gate Hold Time: 0 to 0.5 seconds
    AGC Leveler
        Threshold: 72 to 1 dB, 0.5 dB resolution
        Attack: 0.2 to 3 seconds
        Decay: 0.5 to 5 seconds
        Hinge: Threshold to 1 dB, 0.5 dB resolution
    Compressor and Limiter
        Threshold: 72 to 1 dB, 0.5 dB resolution
        Attack: 1.0 to 200 ms
        Decay: 0.05 to 1 second
        Knee: Hard or Soft selectable
    No Overshoot Peak Limiter
        Propagation delay: 1 ms
        Threshold: 72 to 1 dB, 0.5 dB resolution
        Attack: 0 msDecay: 100 ms
    PARAMETRIC EQUALIZER
        Parametric Filter Frequency Bands
        Up to 9 bands, variable frequency, variable Q
    Boost/Cut Range
        +6 dB to 18 dB per band
        Q Range
        1/40-octave to 2 octave
    High and Low Shelf/Cut Filters
        Shelf, +6 to 18 dB per filter
        Cut, 12 dB per octave nominal
    DELAY
        Up to 1.3 seconds, 21 microsecond resolution

Features
Four Independent Compressor Channels
Channels 1 and 2, 3 and 4 are stereo linkable
7 LED Gain Reduction metering
Balanced XLR Inputs and Outputs
Sidechain Insert points
Intuitive, user friendly layout
Flawless performance
 
The sheer amount of effects and signal processors necessary for today's standards
of audio production puts a great strain on the available space in equipment racks,
both in the studio and on the road.So, ARX would like to introduce the ARX Quadcomp.
An all-new upgrade of the classic award-winning ARX Quadcomp?, the very first four
channel 1 RU compressor.You?ll notice that we haven?t strayed very far from our
original design concept. The Quadcomp  is still four variable ratio Compressor/ Limiters
neatly housed in a compact all steel 1 RU package. However, all new low noise/low
distortion circuitry, better metering plus XLR inputs and outputs put the Quadcomp
in a class of its own.
More Control, Less Space
On the front panel, each channel has the 'industry standard' individual controls for Threshold,
Compression Ratio and Output, plus a new 7 LED Gain Reduction display for accurate visual
indication of the amount of gain reduction being applied to the program, and an IN/OUT
hardwire bypass switch.In addition to this, each channel has a blank numbered panel to write
on for easy confirmation of compressor assigns.
Precision Circuitry
Internally, each compressor uses Class A VCAs and true 2 pole averaging RMS/DC converters
for low distortion and accuracy, plus program dependent attack and release time, which
automatically determines optimum compressor response.
Balanced Inputs and Outputs
The rear panel has true differential Balanced XLR Inputs and Outputs for each compressor.
Each compressor has a TRS jack Sidechain access insert point, for frequency sensitive
compression, De-essing, etc. in conjunction with an external equalizer (such as the ARX EQ260).
Compressor pairs 1 and 2, 3 and 4 also have rear panel mounted Stereo Link switches for accurate
stereo tracking. When switched IN, the first channel becomes the master and controls all the functions
of the second except Output gain.
Universal AC Power
AC power range is switchable 100 to 120V, or 220 to 240V, and is connected to the unit via
a standard IEC connector, with built-in fuse and voltage switch.
Whatever the application ? Channel insert, Bus insert, Master Outputs, Monitor outputs,
Crossover outputs, the Quadcomp II?s unique High Density design, precision Low Noise circuitry
and clean uncluttered layout make it a truly useful audio tool for all applications.

T E C H N I C A L S P E C I F I C A T I O N S
Input Impedance
Balanced 20 Kohms, Unbalanced 10 Kohms
Input Headroom + 20dB
CMRR >50dB, 20 Hz?20 KHz
Output Impedance
Balanced 300 ohms Unbalanced 150 ohms
Output Level (Max) + 20 dB
Frequency Response 20Hz-20KHz ±0.5dB
Signal to Noise ratio
-93 dB Unweighted, -98 dB ?A? weighted
Distortion .015% THD @ 0dB,1KHz
Dynamic Range 108 dB
Attack and Release Times Program dependent
Metering
7 LED display: ?1, 2, 3, 6, 12, 18, ?24 dB
Sidechain Insert Impedance 10 Kohm
Power Requirements
100/120 V AC or 220/240 V AC
Weight 5 lbs/2.2 Kg
Dimensions
19“W x 1?“H x 6“D, 482 x 44 x 155mm
Input Connector type Balanced XLR
Output Connector type Balanced XLR
Sidechain Insert Connector TRS Jack

Front Panel Controls
Hardwire bypass IN/OUT switch
Threshold, Ratio and Output Gain controls
7 LED Gain Reduction display
Marker panel for labelling compressor assigns
Stereo link status LEDs
Rear Panel Connectors
Input and Output balanced XLR connectors
TipRingSleeve Sidechain insert connectors
Channels 1 and 2, 3 and 4 Stereo link switches
IEC AC mains connector with inbuilt fuse and
voltage change

OPTIMOD-FM 5500 puts coveted five-band and two-band OPTIMOD processing into a single rack unit package and brings it to you at the most affordable price ever.

Quality sound is what 5500 is all about-sound that attracts audiences by providing a polished, professional presentation regardless of format and source material. Exceptional versatility allows you to adjust the processor's audio texture to brand your audio, knowing that the resulting signature sound will remain consistent, cut-to-cut and source-to-source. Branding builds businesses and no other processors have the consistency to brand your sound like OPTIMOD.

With the 5500, your signature sound is just a preset away. An easy, one-knob Less/More adjustment allows you to customize any factory preset, trading cleanliness against processing artifacts according to the requirements of your market and competitive environment. Full Control gives you the versatility to customize your audio further. And, if you're a hard-core processing expert, you can explore Advanced Control to tweak presets at the same level as Orban's factory programmers. This versatility makes the 5500 a superb choice for any format. Its five-band processing is ideal for any pop music format (even the most competitive and aggressive CHR), while phase-linear two-band processing yields ultratransparent sound for classical, classic jazz, and fine arts formats. Regardless of your choice, 5500's optimized technology ensures unusually high average modulation and coverage for a given level of subjective quality. Unlike many lesser processors, the 5500 handles speech particularly well. It's always clean, even when you process for loudness.

If you're concerned about latency because you need to feed live talent headphones off air, be assured that the 5500's ultra-low-latency (5 ms delay) processing will keep the most finicky talent happy. Or use optimum latency (15 ms delay) processing for the most competitive sound with delay that's still low enough to satisfy most any talent.

The 5500 can also be used as a superb stand-alone stereo encoder with latency as low as 2 ms and full overshoot limiting in both the left/right and composite baseband domains.

When used in this mode, the 5500 must be driven (usually via an STL) by a full-featured FM audio processor (like Orban's 8600) that incorporates pre-emphasis aware HF limiting and peak control. In both modes, the 5500's stereo encoder helps deliver a transmitted signal that's always immaculately clean and perfectly peak limited, with full spectral protection of subcarriers and RDS/RBDS regardless of the amount of composite limiting.

It is the ideal choice for network broadcasters who process with Orban's flagship OPTIMOD-FM 8600 at the network origination point and who need a processor at every transmitter to eliminate STL overshoots (using the 5500's stand-alone stereo encoder mode) and/or to process local insertions while also eliminating network STL overshoots (using the 5500's audio processor / stereo encoder mode). Moreover, the 5500's two modes make it easy for large government and network broadcasters to manage its inventory of spares because any 5500 can be used as a stereo encoder with or without audio processing.

Available in both modes, the built-in, defeat-able ITU-BS412 multiplex power controller allows the 5500's output to meet even the most stringent European government regulations.

A 10 MHz frequency reference input allows the stereo pilot tone frequency to be locked to GPS or another high-accuracy frequency standard. This improves the performance of single-frequency networks in areas where coverage of the transmitters overlaps.

The 5500's built-in stereo encoder, AES/EBU digital inputs and outputs, and analog I/O permit hasslefree interfacing to any broadcast plant, whether the 5500 is located at the studio or the transmitter. Tight band limiting to 15 kHz means you can use any uncompressed digital STL to pass 5500- processed audio from studio to transmitter without compromising on-air loudness - there's no need to use STL's having 44.1 or 48 kHz sample-rate.

The stereo encoder's stereo sub-channel modulator can operate in normal double sideband mode and in an experimental compatible single sideband mode (SSB/VSB) that is offered to enable users to compare and assess the two modes.

Analog Fallback to Digital control that allows Silence Sense to switch the active input from Analog to Digital if silence is detected in the analog input signal but not on the digital input signal. This function works vice versa as well on both analog and digital AES input. The silence sense parameters apply to both simultaneously and both detectors are available to drive the 5500's tally outputs and sending SNMP Traps/Alerts.

If you want to locate the 5500 away from the studio, you'll be pleased by its three separate remote control ports - GPI contact closures, RS232 serial and built-in Ethernet for TCP/IP networks. The serial and Ethernet ports are supported by the supplied 5500 PC Remote Control application. This Windows® 2000/XP/Vista/7/8 application allows you to do even more with the 5500 than you can do through its front panel, making remote control a pleasure.

5500 PC Remote software allows you to access all 5500 features and allows you to archive and restore presets, automation lists, and system setups (containing I/O levels, digital word lengths, GPI functional assignments, etc.)

Built-in clock-based automation lets you automatically day-part the processing. You can control many other 5500 operating parameters too.

The 5500's feature set fully exploits the processor's DSP and computer-based control architecture. To ensure absolute accuracy, you can automatically synchronize the clock to an Internet timeserver. It has a cool-running, energy-efficient switching power supply and uses the latest dual-core DSP chip technology from Freescale Semiconductor.