DRAWMER DS501 – Power Gate

The DS501 is a sophisticated dual channel noise gate with fully tuneable \'Peak Punch\' incorporating a number of features pioneered by Drawmer, which are invaluable to the sound engineer, and not found on conventional noise gates.

• \'Tuneable\' Peak Punch.
• Variable high pass and low pass filters for \"frequency conscious\" gating.
• Comprehensive envelope control, attack, hold, decay, and range.
• Key input for external triggering.
• \"Key Listen\" facility.
• Extremely fast attack time, to preserve the natural attack of the sound.
• Balanced inputs and outputs.
• Stereo linkable.
• Can be used for \"Gating\" or \"Ducking\".
• High audio specification.

Positioned in the Drawmer range over the industry standard DS201 Dual Gate, the DS501 Power Gate features a new dual mode ‘tuneable’ Peak Punch processing section on each channel. In tuneable mode a fully variable frequency selector with a range from 75Hz to 16kHz allows the user to shape the transients of the gated signal and ‘tune-in’ to the particular area of the audio spectrum where the Peak Punch is to be active. In addition a variable ‘more’ control allows the user to tailor the amount of processing. A secondary Full Band Peak Punch mode is also available.

Designed principally for drum and percussive gating applications, tuneable Peak Punch makes it possible to add transient punch to frequencies lacking in individual drum sounds, adding greater definition and presence to the gated signal. Using low frequency Peak Punch adds depth to thin drums, whilst higher frequency Peak Punch can dynamically emphasise rim shots or the ‘crack’ of a snare drum. Setting up noise gates can be tedious and unrewarding. For example, when attempting to separate a signal from unwanted noise or crosstalk that is relatively high in level, spurious triggering of the gate by unwanted components within the sound can be a serious problem. This can often be experienced in the studio when recording a drum kit. If a gate is used to clean up the snare drum sound it is quite likely that the nearby hi-hats will spill into the snare drum microphone and cause the gate to open. Increasing the threshold level may cure this problem, but then there is a very real danger that any quieter snare drum beats may not cause the gate to open at all and the performance can easily be ruined.

The DRAWMER solution to this problem is the inclusion of two variable filters, one high-pass and one low-pass, which act upon the side-chain keying circuitry. By setting the output switch to key listen the user can hear the action of the filters and adjust them to reject high frequency spillage from the hi-hats. This now enables the gate to only open on the lower frequencies present in the snare drum.

The very fast attack of the DS501 means that it can open in a matter of micro seconds, thus preserving the natural attack of whatever sound is being gated and the comprehensive envelope controls mean that the gain can be changed at whatever rate best suits the material being processed. With vocals for example, a fairly fast attack is needed so as not to clip the leading consonants but a slower release time will prevent the end of words being clipped off and will also fade out any noise gradually rather than cutting it off abruptly. This latter point is very important as the human ear is far more sensitive to rapidly changing noise levels than to a constant low level noise.

Another important role played by the gate is in the reshaping of existing sounds. An example of this might involve a simple sampling delay line used to store a sound to be re-triggered later and added to a mix. A gate can be used to impart a slower attack or faster decay to the sound and careful setting of the decay envelope can effectively hide any noise present at the end of the sample. When using a sampler or drum machine the gate can be considered a triggered envelope shaper, and the wide range allowed by the envelope controls make the DS501 ideal for this application.

The unit can be switched to accept a key input, which allows the gate to be triggered externally. An example of this would be to use a snare drum signal to open the gate on a separate ambient microphone to create a natural alternative to gated reverb.

Each channel can be individually switched from Gating to Ducking for \'voice over\' applications or the removal of \'clicks\' and \'pops\'.

The latest addition to the 10 Series product line, the dbx® 1074 QuadGate™, is the perfect companion to the 1066 and 1046. The 1074 offers 4 channels of gating with threshold, depth and release controls on each channel. The 1074, like the rest of the products in dbx 10 Series, is based on the legendary dbx V2 VCA and offers XLR inputs and outputs, and 1/4” side-chain input. In addition to an external key input per channel, the 1074 also has an internal filter that can be independently activated and controlled on a channel-per-channel basis. This filter allows the 1074 to not only clean up tracks, but gives you frequency selective control on each gate, to open exactly when you want it to. These features combined, make the 1074 an absolute must for those who require quad gating on applications including: Gating dry and percussive sounds such as snare drum and kick drum,gating sounds that have longer decay times such as cymbals and Pianos, gating hum or buzz from live instruments or recorded tracks or eliminating Headphone leakage into microphones. Gating instrument microphones to eliminate amplifier bleed. 

From the company that has been redefining the standard of signal processing for more than 30 years, dbx® Professional Products comes a bulletproof product that caters to the scrutinizing needs of those who require the precision noise gating capabilities. Once again, the engineering staff at dbx® Professional Products have taken technology that has been over three decades in the making and have created a product that is flexible, versatile and rock solid in construction design. The 1074ʼs independent channel design with key filtering capabilities make it the perfect tool for countless applications.

• Four independent channels of gating
• Independent key filtering
• Independent Threshold and Release controls
• Differentially balanced gold-plated XLR and 1/4" inputs and outputs
• True RMS level detection
• Stereo Coupling mode
• Switchable +4dBu or -10dBv operation per channel
  
  

  Input Connectors	XLR (Pin 2 and tip hot)
  Input Type	Electronically balanced/unbalanced, RF filtered
  Input Impedance	Balanced > 50 kOhm, unbalanced >25 kOhm
  Max Input	> +22 dBu balanced or unbalanced
  CMRR	40dB: Typically >55 dB at 1 kHz
  Key Input	Electronically balanced/unbalanced, RF fitered Balanced 50kOhm, unbalanced > 25 kOhm
  Output Connectors	XLR (Pin 2 hot)
  Output Type	Servo-balanced/unbalanced, RF filtered
  Output Impedance	Balanced 60 Ohm, unbalanced 30 Ohm
  Max Output	> +22 dBm balanced, > +20 dBm unbalanced
  Bandwidth	20 Hz to 20 kHz, +0/-0.5 dB
  Frequency Response	0.35 Hz to 200 kHz, +0/-3 dB
  Noise
  Dynamic Range	> 115 dB, unweighted
  THD+Noise	0.008% typical at +4 dBu (1 kHz unity gain), 0.08% typical at +20 dBu (1 kHz unity gain),
  Interchannel Crosstalk
  Stereo Coupling	True RMS Power Summing
  Operating Voltage	100-120 VAC 50-60 Hz or 200-240 VAC 50/60 Hz
  Power Consumption	30 Watts
  Fuse	100-120 VAC:250 mA Slow Blow, 200-240 VAC:125 mA Type T
  Power Connector	IEC receptacle
  Unit Weight	6.9 lbs.
  Shipping Weight	9.3 lbs.

  
  

Innovation

The ARX AFTERBURNER? is a unique Multi Mode compressor/limiter
designed for use in any professional audio dynamics control application.
Multiple Modes
The Afterburner can be set up and used in three different ways:
In Two channel mode, it performs as two independent compressor/limiters,
with 'industry standard' variable Threshold, Ratio and Output gain. In stereo
mode, our New Adaptive Stereo Link circuitry provides increased stereo
imaging accuracy when linking both channels as a stereo pair. A single front
panel switch puts the Afterburner into its alternative Mono mode, setting it up
as a Single channel, Dual Band compressor/limiter, with separate dynamics
control of both Low and High frequencies, opening up a whole new range of
gain control techniques.
Enhance Switch
In any mode, the Afterburner features an 'Enhance' switch, which provides
frequency restoration to preserve the spectral balance of the audio signal,
compensating for the sagging Low and High frequency response of compressed
program material. Think of it as a 'smart' loudness control.
 
Features
Switchable Modes - Stereo; Dual Channel Single band; or Single Channel
Dual Band (Low and High)
Low and High frequency compression in Single Channel mode
'Enhance' switch restores spectral balance of compressed signal
New Hard/Soft knee compression switch
New Adaptive Stereo Link circuit for accurate stereo imaging
New Above/Below Threshold LEDs enable at-a-glance compression
confirmation
Balanced XLR and Jack Inputs and Outputs
Sidechain Insert points
Intuitive user-friendly layout
Flawless performance
 
New Above/Below Threshold LEDs enable at-a-glance compression confirmation
New Hard or Soft knee compression option
New LED metering provides accurate Level status, with separate Gain Reduction
metering in easy to read 'wide scale' meters. There is also comprehensive LED
indication of all operating functions and status in any mode.
Balanced Inputs and Outputs
On the rear panel, each channel has true differential Balanced inputs and outputs,
on both XLR and TRS jack connectors. As well, each channel has a TRS jack
Sidechain access insert point, for applications such as De-essing (when used with
an external equalizer such as the ARX Multi Q or EQ 260).Other features include a
true Hardwire bypass switch for each channel, and passive RFI filters on the inputs.
Universal AC Power
AC power range is a universal 100 to 120V or 220 to 240V AC, and is connected to
the unit via a removable power lead and standard 3 pin IEC connector, with built-in
fuse and voltage change switch.
With its smooth compression, intuitive user friendly layout, high density precision
circuitry, and extensive user-variable operating parameters, the unique ARX Afterburner
is equally at home in Studio, Installation, Broadcast and Sound Reinforcement
environments. It can provide great sounding dynamics control effects that are not
available with any other device.
 
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 + 22 dB
CMRR >60 dB, 20 Hz-20 KHz
Output Impedance Balanced 300 ohms, Unbalanced 150 ohms
Output Level (Max) + 22 dB
Frequency Response 20Hz to 20KHz ±0.2dB
Signal to Noise ratio -93 dB Unweighted, -99 dB 'A' weighted
Distortion .02% THD @ 0dB,1KHz
Dynamic Range 115 dB
Sidechain Insert Impedance 10 Kohm
Filter Section
Filter Type Phase corrected 6dB/octave
Summed Filter Response ±0.2dB through crossover region
Dividing Frequency 250 Hz
Enhance Section
Low Enhance 50 Hz, High Enhance 10 KHz
Power Requirements 100/120 V AC, 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/OutputConnector type XLR, Balanced Jack
Sidechain Insert Connector TRS Jack

Front Panel Controls
Hardwire bypass IN/OUT switch
Threshold, Ratio and Output Gain controls
Above/Below Threshold LEDs
12 segment LED Output Level display
Numbered marker panel for labelling compressor assigns
7 segment LED Gain Reduction display
Enhance switch and status LED
Adaptive Stereo link switch and status LED
Dual/Single channel mode switch and status LEDs
Rear Panel Connectors
Balanced Inputs and Outputs, on both XLR and TRS jack
connectors. In Single channel (Mono) mode, use Channel
1 Inputs and Outputs only
Sidechain Insert TRS connector on each channel
AC input connector, with voltage switch and fuse.
 

The AD2055 combines 100% discrete, pure class A signal amplifiers with state-of-
the-art passive and active filter topologies. The AD2055's unique circuitry delivers
very high resolution transient detail at the operational extremes of real world equalization
demands. The AD2055 auto-bias DC servo loop control eliminates the need for all
interstage capacitor coupling. The AD2055 breathes life into all musical performances!
 
AD2055 FEATURES
Pure Class A, 100% discrete design
Smooth musical detail and sonic excellence
Minimum audio signal path
Dual mono operation
Transparent active and passive filter design
High headroom +30dB Very low noise -94dB
Fully balanced inputs and outputs
DC coupled, no transformers in audio path
Switched frequencies in high and low bands
Mid bands use X10 frequency for extended range
Wide bandwidth -3dB 1Hz to 450kHz
Low distortion less than 0.5% THD and IMD
All signal routing with sealed silver relays Conductive
plastic potentiometers for low noise
External 150W toroidal power supply 100% discrete
power supplies for audio path
Long lasting, stainless steel hardware
 
SONIC EXCELLENCE
The Avalon AD2055 Pure Class A music equalizer is the most powerful, low noise
parametric equalizer available today. Designed to optimize absolute signal integrity
and musical performance, the AD2055 combines the best of active and passive filter
topologies with sonic excellence unequaled by lesser designs. The AD2055 is the
perfect solution for two buss music-program equalization, special instrument EQ and
FX applications and ultra high performance mastering studio's.

Features include state-of-the-art, balanced 100% discrete, Pure Class A signal amplifiers,
practical user features and rugged hardware designed to deliver true high performance
audio for many years. 

PASSIVE-ACTIVE FILTER DESIGN
Avalon equalizers feature the unique combination of both active and passive filter EQ designs.
This special combination enables the AD2055 to deliver high speed transient detail at the
operational extremes of real-world equalization demands. The passive high and low bands offer
alternate musical tone range to the full function active parametric mid bands. Passive equalizers
have long been a favorite with music lovers around the world.  Full bodied, powerful (up to +/-32dB,
64dB range !) and sweet frequency selections are the benefits of the passive high-low EQ bands.
The full bandwidth twin mid bands provide variable frequency selection (X10 frequency multipliers
for very wide range), variable Q (width) and amplitude control.

MINIMUM SIGNAL PATH DESIGN
Avalon's advanced true symmetry design offers high-voltage, large headroom, extended bandwidth
and very low noise. The use of 100% discrete, Pure Class A signal amplifiers give the serious music
professional unlimited sonic character and a natural harmonic detail that enhances the program
material and becomes one with the music itself. The Avalon AD2055 breathes life!

AD2055 SPECIFICATIONS
Circuit Topology  High-voltage 100% discrete, balanced and symmetrical
Class A 
Output Gain Range  Unity Gain 
Maximum Input Level  +30dB balanced XLR pin 2 hot 
Maximum Output Level  +32dB balanced 600 ohms, DC coupled, high-current
discrete Class A 
Input / Output Type  XLR type, pin 2 hot balanced 
Noise 20kHz Unweighted  -94dB (EQ in) 
Distortion THD, IMD  0.5% (typical 0.05% at +6dB 1kHz) 
Frequency Response -3dB  1Hz to 450kHz (input band limited) 
Equalizer Type  Passive high and low bands plus two fully parametric mid
bands 
Bypass  Hard-wire relay bypass for equalizer in-out 
Low Band F1  Passive, amplitude to -32dB to +32dB shelf or peak-dip curve 
F1 Frequency Range  Switched 10 position 18Hz, 25, 30, 50, 72, 100, 150, 215,
300, 450Hz 
Mid Band F2  Active, amplitude to -16dB to +16dB peak-dip curve 
F2 Frequency Range  Variable 35Hz to 450Hz (x10) 350Hz to 4k5Hz, Q (width)
0.3 to 3.0 
Mid Band F3  Active, amplitude to -16dB to +16dB peak-dip curve 
F3 Frequency Range  Variable 160Hz to 2k0Hz (x10) 1k6Hz to 20kHz, Q (width)
0.3 to 3.0 
High Band F4  Passive, amplitude to -26dB to +26dB shelf or peak-dip curve 
F4 Frequency Range  Switched 10 position 1k5Hz, 2k5, 3k5, 5k, 7k2, 10k, 12k5,
15k, 20k, 25kHz 
AC-DC Power  External AC supply, 150w toroidal transformer, 4 pin cable 90v
isolated, 
(B2T Power Supply Included)  100-240v selectable 50/60Hz, 150w max 
Dimensions  19 x 3.5 x 12 in (482 x 88 x 305mm) 
Shipping weight  30lbs (13.6 kg) 

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)

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.