COMREX VH2 two-line VoIP hybrid for individual broadcast or flawless conferencing

2,765.00 incl. GR VAT

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Hybrid phone two VoIP telephone lines. Supports telephone lines G.711 & G.722. Has ability checks the phone lines via VoIP devices phone. Easy installation via web based environment. In the area of I / O features two analog inputs and two outputs to XLR form configurable and AES / EBU. audio input “on-hold” in order […]

 

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Hybrid phone two VoIP telephone lines. Supports telephone lines G.711 & G.722. Has ability checks the phone lines via VoIP devices phone. Easy installation via web based environment. In the area of I / O features two analog inputs and two outputs to XLR form configurable and AES / EBU. audio input “on-hold” in order to provide music on hold, AGC and network port 10/100/1000.

Any studio that processes phone calls needs hardware to interface with phone lines. This device, traditionally called a Hybrid, filters, separates and provides gain adjustment and call control to more easily allow for recording or broadcasting “phoners”.

As telephone companies are shifting from traditional phone lines to Voice-over-IP (VoIP) systems, the devices that performed this job in the past are becoming outmoded. Broadcasters need a VoIP hybrid to ensure on-air and recorded phone calls sound as good as possible.

A dual-line hybrid, VH2 connects two VoIP lines to a studio for individual broadcast or flawless conferencing. VH2 prevents echo and distortion, and automatically adjusts caller audio to a uniform level, leaving you with a result that’s clean and clear. Plus, VH2 uses only VoIP phone lines, saving you money and increasing functionality. VH2 can even connect to many VoIP PBX systems

  • Prevents echo and other artifacts
  • Supports normal phone calls (G.711) and wideband phone calls (G.722)
  • Allows for caller ID, outgoing calls, screening, and post-air management using companion VoIP phone set
  • Set up via easy-to-use web based configuration page

Easy Hookup

VH2 can be configured in several ways to be compatible in environments where the studio has different telephone connection arrangements. Dual or single input and outputs can be selected, and AES3* or analog audio I/O can be chosen. VH2 can be configured for callers to hear each other, or be isolated depending on the needs of the studio. *Supports 48 KHz sampling rate only for AES3.

On-Hold Inputs

VH2 offers a pair of audio inputs for callers “on-hold”. This allows for listeners to hear your programming while they are waiting to be put “on-air”.

Consistent Audio Levels

Selectable automatic gain control (AGC) maintains a uniform audio output, even when the caller signal varies widely. Also, selectable caller ducking lowers incoming caller audio, so local talent remains in control.

Companion Phone

When configured with its companion phone (the Polycom VVX 201 IP phone), VH2 does even more. Calls can be answered on the handset and easily transferred back and forth to VH2, just like a traditional telephone hybrid. And the companion phone supports caller ID and outbound calling.

Status Indication

VH2 is outfitted with front panel controls and status indicators so it can be used out-of-the-box. Front panel buttons can also be remoted via the rear panel connector, so your console buttons can trigger its functions.

Audio Connections

  • Caller audio out on balanced XLR-M output
  • Send audio in on balanced XLR-F input (clip +20dBu). Switchable to AES3 I/O (48KHz sampling rate only for AES3).
  • On-hold audio in on ¼” TRS jack input (clip +20dBu)

Other Connections

  • 10/1000 Ethernet port
  • Contact closures
  • 9 pin mini DIN
  • Serial port on 8 pin mini DIN
  • Power in on 4 pin mini DIN
  • Universal external power supply +24VDC
  • Compliant with worldwide regulations, including FCC, CSA, and CE

When you want to present, broadcast, or record a telephone conversation, you need a device to process the phone call and present it to the console, as well as to separate “send” audio from the “receive” audio on the call. If send and receive audio aren’t isolated, it will result in an echoey, muddy sound – not to mention, annoyed listeners.

As many major markets are shifting from traditional phone lines to Voice-over-IP (VoIP) systems, the digital hybrids that could have performed this job in the past are becoming outmoded. Radio stations need a VoIP hybrid to ensure on-air and recorded phone calls sound beautiful.

A dual-line hybrid, VH2 connects two VoIP lines for individual broadcast or flawless conferencing. VH2 prevents echo and distortion, and automatically adjusts caller audio to a uniform level, leaving you with a result that’s clean and clear. Plus, VH2 uses VoIP phone lines, saving you money and increasing functionality. VH2 can even connect to many VoIP PBX systems.

Main Features:
Audio Processing and Performance:
• Prevents echo and other artifacts
• G.722 codec support for wideband calls. Also supports G.711
• Receive filter reduces telephone line noise
• Selectable automatic gain control (AGC) maintains a consistent audio output, even when the caller signal varies widely
• Selectable caller ducking lowers incoming caller audio so local talent remains in control of the conversation
• Can be configured to automatically answer and disconnect incoming calls

Operation:
• Easily segue from caller-to-caller
• Separately selectable single-ring auto-answer function for assisted or unattended operation
• Handy front panel controls and status indication
• When used with companion VoIP telephone, calls can be answered on handset and easily transferred back and forth to VH2
• Hybrid on/off controls and status remotable via web or contact closures
• Send and caller level indication
• Easy call conferencing
• Dual “On-Hold” audio inputs to send program to callers on hold
• Auto-Switching External Power Supply
• Compliant with worldwide regulations, including FCC, CSA and CE

Audio Configuration:
• Configure for separate caller outs of single caller mix
• Configure for separate send feeds or single
• Pro level, balanced audio I/O in XLR
• Selectable AES3 I/O

IP Features:
• Web-based configuration for the VoIP phone line setup, making it easy to adjust settings remotely from a browser
• Transfer calls back and forth to many PBXs, or use optional companion extension VoIP phone
• Ability to engage or drop or dial calls via web page
• Companion phone – easy to move calls between handset and hybrid with the touch of a button

Weight 1.00 kg

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COMREX VH2 two-line VoIP hybrid for individual broadcast or flawless conferencing removeAVALON VT-737SP DIRECT SIGNAL PATH removeBSS DPR-504 QUAD NOISE GATE removeAVALON AD2055 DUAL MONO PURE CLASSA PARAMETRIC MUSIC EQUALIZER removeARX QUADCOMP 4CHANNEL COMPRESSOR-LIMITER removeBEHRINGER FBQ1000 FEEDBACK DESTROYER remove
NameCOMREX VH2 two-line VoIP hybrid for individual broadcast or flawless conferencing removeAVALON VT-737SP DIRECT SIGNAL PATH removeBSS DPR-504 QUAD NOISE GATE removeAVALON AD2055 DUAL MONO PURE CLASSA PARAMETRIC MUSIC EQUALIZER removeARX QUADCOMP 4CHANNEL COMPRESSOR-LIMITER removeBEHRINGER FBQ1000 FEEDBACK DESTROYER remove
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Hybrid phone two VoIP telephone lines. Supports telephone lines G.711 & G.722. Has ability checks the phone lines via VoIP devices phone. Easy installation via web based environment. In the area of I / O features two analog inputs and two outputs to XLR form configurable and AES / EBU. audio input "on-hold" in order to provide music on hold, AGC and network port 10/100/1000.

Any studio that processes phone calls needs hardware to interface with phone lines. This device, traditionally called a Hybrid, filters, separates and provides gain adjustment and call control to more easily allow for recording or broadcasting "phoners”.

As telephone companies are shifting from traditional phone lines to Voice-over-IP (VoIP) systems, the devices that performed this job in the past are becoming outmoded. Broadcasters need a VoIP hybrid to ensure on-air and recorded phone calls sound as good as possible.

A dual-line hybrid, VH2 connects two VoIP lines to a studio for individual broadcast or flawless conferencing. VH2 prevents echo and distortion, and automatically adjusts caller audio to a uniform level, leaving you with a result that’s clean and clear. Plus, VH2 uses only VoIP phone lines, saving you money and increasing functionality. VH2 can even connect to many VoIP PBX systems

  • Prevents echo and other artifacts
  • Supports normal phone calls (G.711) and wideband phone calls (G.722)
  • Allows for caller ID, outgoing calls, screening, and post-air management using companion VoIP phone set
  • Set up via easy-to-use web based configuration page

Easy Hookup

VH2 can be configured in several ways to be compatible in environments where the studio has different telephone connection arrangements. Dual or single input and outputs can be selected, and AES3* or analog audio I/O can be chosen. VH2 can be configured for callers to hear each other, or be isolated depending on the needs of the studio. *Supports 48 KHz sampling rate only for AES3.

On-Hold Inputs

VH2 offers a pair of audio inputs for callers "on-hold”. This allows for listeners to hear your programming while they are waiting to be put "on-air”.

Consistent Audio Levels

Selectable automatic gain control (AGC) maintains a uniform audio output, even when the caller signal varies widely. Also, selectable caller ducking lowers incoming caller audio, so local talent remains in control.

Companion Phone

When configured with its companion phone (the Polycom VVX 201 IP phone), VH2 does even more. Calls can be answered on the handset and easily transferred back and forth to VH2, just like a traditional telephone hybrid. And the companion phone supports caller ID and outbound calling.

Status Indication

VH2 is outfitted with front panel controls and status indicators so it can be used out-of-the-box. Front panel buttons can also be remoted via the rear panel connector, so your console buttons can trigger its functions.

Audio Connections
  • Caller audio out on balanced XLR-M output
  • Send audio in on balanced XLR-F input (clip +20dBu). Switchable to AES3 I/O (48KHz sampling rate only for AES3).
  • On-hold audio in on ¼” TRS jack input (clip +20dBu)

Other Connections
  • 10/1000 Ethernet port
  • Contact closures
  • 9 pin mini DIN
  • Serial port on 8 pin mini DIN
  • Power in on 4 pin mini DIN
  • Universal external power supply +24VDC
  • Compliant with worldwide regulations, including FCC, CSA, and CE

When you want to present, broadcast, or record a telephone conversation, you need a device to process the phone call and present it to the console, as well as to separate "send” audio from the "receive” audio on the call. If send and receive audio aren’t isolated, it will result in an echoey, muddy sound - not to mention, annoyed listeners.

As many major markets are shifting from traditional phone lines to Voice-over-IP (VoIP) systems, the digital hybrids that could have performed this job in the past are becoming outmoded. Radio stations need a VoIP hybrid to ensure on-air and recorded phone calls sound beautiful.

A dual-line hybrid, VH2 connects two VoIP lines for individual broadcast or flawless conferencing. VH2 prevents echo and distortion, and automatically adjusts caller audio to a uniform level, leaving you with a result that’s clean and clear. Plus, VH2 uses VoIP phone lines, saving you money and increasing functionality. VH2 can even connect to many VoIP PBX systems.

Main Features:
Audio Processing and Performance:
• Prevents echo and other artifacts
• G.722 codec support for wideband calls. Also supports G.711
• Receive filter reduces telephone line noise
• Selectable automatic gain control (AGC) maintains a consistent audio output, even when the caller signal varies widely
• Selectable caller ducking lowers incoming caller audio so local talent remains in control of the conversation
• Can be configured to automatically answer and disconnect incoming calls

Operation:
• Easily segue from caller-to-caller
• Separately selectable single-ring auto-answer function for assisted or unattended operation
• Handy front panel controls and status indication
• When used with companion VoIP telephone, calls can be answered on handset and easily transferred back and forth to VH2
• Hybrid on/off controls and status remotable via web or contact closures
• Send and caller level indication
• Easy call conferencing
• Dual "On-Hold” audio inputs to send program to callers on hold
• Auto-Switching External Power Supply
• Compliant with worldwide regulations, including FCC, CSA and CE

Audio Configuration:
• Configure for separate caller outs of single caller mix
• Configure for separate send feeds or single
• Pro level, balanced audio I/O in XLR
• Selectable AES3 I/O

IP Features:
• Web-based configuration for the VoIP phone line setup, making it easy to adjust settings remotely from a browser
• Transfer calls back and forth to many PBXs, or use optional companion extension VoIP phone
• Ability to engage or drop or dial calls via web page
• Companion phone - easy to move calls between handset and hybrid with the touch of a button

The Avalon VT-737SP features a combination of TUBE preamplifiers,
opto-compressor,sweep equalizer, output level and VU metering in a
2U space.
PREAMPLIFIER
The VT-737SP has three input selections: 1. The VT-737SP features a
higher-performance microphone input transformer with +48v phantom
selection. 2. Instrument DI high source input with jack located on front
panel. 3. Balanced line input, discrete high-level Class A. The Class A
preamplifier utilizes two cascaded, dual vacuum tube triodes configured
with minimum negative feedback. A high gain switch boosts the overall
gain of the preamplifier. The four high quality tubes are configured as
singled ended anode coupled followers. A passive-variable high pass filter
and hard-wire relay bypass completes the input signal conditioning. The
phase reverse relay is available on all three inputs.
OPTO-COMPRESSOR
The opto-compressor features a minimum signal path design with twin Class
A vacuum tube triodes for gain matching. The optical attenuator acts as a
simple passive level controller. Full dynamic control from soft compression to
hard- knee limiting can be achieved with threshold, ratio-compression, attack
and release controls plus gain reduction selection on the large VU meter. Special
spectral control including de-ess is available with the dual sweep mid EQ to side-
chain switch. The EQ section can be flipped pre or post the opto-compressor
via a front panel switch for alternate effects and tone shaping. Two VT-737SPs
can be linked via a rear panel link cable for stereo tracking. The compressor
bypass is a sealed silver relay for the most direct signal path.
SWEEP-EQUALIZER AND OUTPUT LEVEL
The VT-737SP equalizer utilizes 100% discrete, Class A-high-voltage transistors
for optimum sonic performance. The high and low frequency bands provide the
smooth characteristics of an all passive design, while the dual mid bands include
variable frequency and switched Q-width selection. The enhanced range of the mid
bands is extended into the high and low bands by the use of X10 frequency multipliers.
The bypass switch incorporates a sealed silver relay for the most direct signal path.
When the EQ to sidechain is engaged, the high-low EQ remains in the audio path for
"tone enhancement“. The output level control provides a variable control of the overall
signal path. The output amplifier utilizes another dual triode vacuum tube driving a
100% discrete, Class A, high-current, balanced and DC coupled low noise output
amplifier.

AVALON VT-737SP FEATURES
Vacuum tube triode signal path, transformer balanced microphone input, high-voltage
circuits for maximum headroom to +30dB, low noise -92dB, internal discrete power
supply with toroidal transformer, soft-start tube life extender, stereo link for compressors,
all discrete Class A equalizer with musical passive filter design, sealed silver relays for
signal routing, large control knobs with professional conductive plastic potentiometers,
VU meter for output level monitoring and compressor gain reduction.

VT-737SP SPECIFICATIONS
Circuit topology Four dual triode vacuum tubes (Sovtek 6922), high-voltage discrete
Class A
Gain Range Microphone: Transformer balanced 850/2500 ohm, 0dB to +58dB Instrument:
Unbalanced 1 meg ohm, 0dB to+30dB Line: Balanced Class A 20k ohms, -27dB to 28dB
Maximum input level and connector types Microphone26dB@25Hz, +30dB@1kHz balanced
XLR Instrument+30dB unbalanced front panel jack socket
Line +36dB balanced XLR
Maximum output level +30dB balanced 600 ohms, DC coupled, discrete Class A
Output type and gain XLR connector, output trim gain -45dB to20dB
Noise 20kHz unweighted -92dB
Noise microphone EIN -116dB, 22Hz to 22kHz unweighted
Distortion THD, IMD 0.5%
Frequency response -/2.5dB 10Hz to 120kHz input filter included
Frequency response -3dB 1Hz to 200kHz line in-out
VU meter and gain reduction High quality illuminated OVU =+4dB and gain reduction to -20dB
High cut filter Variable 6dB per octave 30Hz to 140Hz
Compressor type Optical passive attenuator incorporating twin vacuum tubes and stereo link
Threshold - Ratio Threshold variable -30dB to +20dB, ratio-compression variable 1:1 to 20:1
Attack - Release Attack variable 2mS to 200mS, release variable 100mS to 5 seconds
Equalizer type Discrete Class A, variable active and switched passive design
Frequency bands (4) Treble - switched 10kHz, 15kHz, 20kHz, 32kHz, +/- 20dB range, shelf
High mid - variable 200Hz to 2k8Hz and 2kHz to 28kHz, +/- 16dB range, hi-lo Q Low mid -
variable 30Hz to 450Hz and 300Hz to 4k5Hz, +/- 16dB range, hi-lo Q Bass - switched 15Hz,
30Hz, 60Hz, 150Hz, +/- 24dB range, shelf
AC power Internal toroidal 100v to 240v, 50-60Hz selectable, 75 watts maximum
Dimensions 19 x 12 x 3.5 in (482 x 305 x 89mm)
Weight 22lbs (10kg)
Dimensions-shipping carton 21 x 18 x 8 in (533 x 457 x 203mm)
Weight-packed 26lbs (11.8kg)

The DPR-504 is a 4-channel noise gate.

Each channel features:
A parametric key filter with key filter listening.
Simultaneous key level and threshold metering with average and
peak metering.
A gate status LED.
A RELEASE/HOLD control (hold tracks proportionally with release
times).
Attack switchable auto/fast (auto tracks signal dynamics for optimum
setting).
Link (links two channels together).
Key source allows external triggers.

The Range switch selects between 7OdB and 2OdB gating depth to
provide alternative release profiles.
The unit features electronically balanced inputs and outputs. 

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) 

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

FEEDBACK DESTROYER FBQ1000 Automatic and Ultra-Fast Feedback Destroyer/Parametric EQ with 24 FBQ Filters High-performance 2-channel Feedback Destroyer and Parametric EQ Automatically and "intelligently" locates and destroys up to 24 frequencies Ultra-narrow FBQ filters for extremely effective feedback suppression, while keeping highest sonic quality "Set-and-forget" default setting enables immediate and super-easy Feedback Destroyer performance Auto mode continuously monitors the mix and automatically sets filters Single-Shot mode automatically detects and destroys feedback plus locks the filter until you reset it manually Manual mode allows setting of up to 24 fully parametric filters with individual frequency, bandwidth and gain parameters Extensive MIDI implementation and user preset memories for instant program recalls Balanced inputs and servo-balanced outputs with ¼'' TRS and gold-plated XLR connectors High-quality components and exceptionally rugged construction ensure long life Conceived and designed by BEHRINGER Germany BEHRINGER FEEDBACK DESTROYER FBQ1000
Weight1.00 kg11.80 kg5.00 kg13.60 kg5.20 kg1.90 kg
DimensionsN/AN/AN/AN/AN/AN/A
Additional information
Weight 1.00 kg
Weight 11.80 kg
Weight 5.00 kg
Weight 13.60 kg
Weight 5.20 kg
Weight 1.90 kg