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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
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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)
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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)
| Compressos Expander Gate Kimiter DBX 166XS Stereo
Overview Bring a more professional sound to your mix
Adding a dbx® 166xs Compressor/Limiter/Gate to your live sound rig or studio gives you more dynamic control to help create a more polished, professional sound. Having compression in your audio chain gives you the ability to smooth out uneven levels, add sustain to guitars and fatten up your drums. It also makes it easy to bring vocals to the front of your mix - adding greater clarity and making them stand out from the surrounding instruments. To protect your expensive amps and speakers the PeakStop® limiter provides an absolute ceiling for peak excursions or large transients that could damage your equipment.
dbx knows compressors...after all we invented them! The 166xs is the latest in a long line of the world's most successful compressors from the inventors of the technology. Its patented Overeasy® compression technology provides smooth and musical performance while the AutoDynamic™ attack and release controls, found only on dbx compressors, puts great sound within easy reach. The 166xs can operate in stereo or dual-mono modes, has true RMS power summing and features quality XLR and 1/4" TRS inputs and outputs. It cuts no corners on visual feedback with gain reduction metering and easy-to-read backlit switches.
Specifications
| Input Connectors |
1/4" TRS, female XLR (pin 2 hot) |
| Input Impedance |
>50kΩ balanced, >25kΩ unbalanced |
| Max Input |
>+24dBu, Balanced or Unbalanced |
| Input Type |
Electronically balanced/unbalanced, RF filtered |
| Sidechain |
1/4" TRS Phone, Normalled: Ring = Output (send); tip = Input (return) |
| Sidechain Impedance |
Tip = >10kΩ (Input), Ring = 2kΩ (Output) |
| Sidechain Max Input Level |
>+20dBu (tip/input); >+20dBu (ring/output) |
| Output Connectors |
1/4" TRS, female XLR (pin 2 hot) |
| Output Impedance |
120Ω balanced, >60Ω unbalanced |
| Max Output |
+21dBu balanced/unbalanced into 2kΩ or greater; >+18 dBm balanced/unbalanced (into 600Ω) |
| Frequency Response |
20Hz - 20kHz; +0, -0.5dB, Typical 3dB points are 0.35Hz and 110kHz, unity gain |
| Noise |
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The S com 4 is a compact and versatile single-rack space device that provides
four channels of high quality dynamics processing with an Expander/Gate and
Compressor/Limiter on each channel.
The four channels operate independently or in stereo pairs. Its multi-segment LED
metering displays input/output level as well as gain reduction. The Expander/Gate
section features a continuously variable Threshold control as well as a switch for
fast or slow release times.
The Compressor/Limiter section includes variable Threshold, Ratio and Output levels.
S com 4's Enhancer switch restores high frequencies that are sometimes diminished
by heavy compression. This wide-ranging combination of features makes the S com
4 an efficient and versatile audio tool for a wide variety of applications. Its clean and
quiet audio characteristics make it ideal where high sonic integrity is imperative.
FEATURES:
4 Channel Compressor/Limiter, Expander/Gate with Enhancer
Linkable in two Stereo pairs
Expander/Gate with variable Trigger control and switchable Fast/Slow Release
5 Segment LED meters for Input/Output levels and Gain Reduction on each channel
Advanced circuit design utilizing Low Noise Operational
Amplifiers and High Quality VCA's
Servo Balanced Inputs and Outputs on XLR and 1/4" connectors
SKD (Smart Knee Detector) circuit automatically switches from soft to hard knee
based on the level of Input signal applied
Individual Threshold and Ratio controls on each channel
Three-year extended warranty
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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.
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