Content |
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
| High-Definition 31-Band Stereo Graphic Equalizer with FBQ Feedback Detection System
- Professional 31-band stereo graphic equalizer for live and studio applications
- Revolutionary FBQ Feedback Detection system instantly reveals critical frequencies and can also be used as audio analyzer
- Ultra-low noise 4580 operational amplifiers for highest signal integrity
- Dedicated mono subwoofer output with adjustable crossover frequency
- Additional sweepable high and low-cut filters for each channel remove unwanted frequencies, e.g. floor rumble, hiss etc.
- Highly accurate 12-segment LED input/output metering and input gain controls for easy level setting
- Relay-controlled hard-bypass with an auto-bypass function during power failure (fail-safe relay)
- Servo-balanced inputs and outputs with 1/4" TRS and gold-plated XLR connectors
- “Planet Earth" switching power supply for maximum flexibility (100
- 240 V~), noise-free audio, superior transient response plus low power consumption for energy saving
- 3-Year Warranty Program*
- Conceived and designed by BEHRINGER Germany
|
Features
Unique High Density design puts SIX gates in
minimum rack space
Intuitive, 'user friendly' layout
Balanced Inputs and Outputs
Fast, low noise opto-coupler circuitry
Sidechain/Key Input points on each channel
Hardwire Bypass switch on each gate
Flawless performance in any audio environment
The sheer amount of effects and signal processors required 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, in order to give engineers more control in less space, we've created
the ARX Sixgate; Six full function noise gates neatly housed in a compact,
all steel 1 RU package, without sacrificing features or quality.
Why Six gates?
Research shows that the majority of noise gates get used on drum kits,
which means using typically 5 to 6 gate channels. Compare this to the
two or maybe 4 channels that most other units have, and you'll see the
reason for the Sixgate's success.And, by using a minimalist approach to
drum gates, the ARX Sixgate has enough gates left over for backing vocals
and other instruments.
Industry Standard controls
Front panel controls for each gate consist of the 'Industry Standard' controls
for Release, Depth and Threshold, plus Red and Green LED displays to indicate
Gate Open or Closed status, and an IN/OUT hardwire bypass switch.
In addition to this, each gate has a blank panel to write on for easy confirmation
of gate assigns. No more pieces of masking tape stuck everywhere!
Ultra Low Noise
Internally, each gate has proprietary ARX ultra low noise high-speed opto-coupler
circuitry with program dependent Attack time, which tracks the incoming signal to
automatically determine optimum gate response.
Balanced Inputs and Outputs
The rear panel features true differential Balanced Inputs and Outputs for each gate,
on insulated TRS jack connectors.Each gate channel also has Individual Key Inputs/
Sidechain access insert points which can be used either for gate control by an external
signal, or for frequency sensitive gating (when used in conjuction with an external
equalizer such as the ARX EQ260).
Universal AC Power
AC power range on the Sixgate is a universal 100 to 120V or 220 to 240V, and is
connected to the unit via a standard 3 pin IEC connector, with built-in fuse and voltage
switch.
The Sixgate's unique combination of High Density design, intuitive 'user friendly' controls,
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 + 22 dB
CMRR >60 dB, 20 Hz-20 KHz
Output Impedance
Balanced 300 ohms, Unbalanced 150 ohms
Output Level (Max) + 20 dB
Frequency Response 20Hz to 20KHz ±0.2dB
Signal to Noise Ratio
Gate Closed: -95 dB Unweighted, -105 dB 'A' weighted
Gate Open, Depth Minimum: -93.5 dB Unweighted -98dB
'A' weighted
Distortion 0 .01% THD @ 0dB,1KHz
Dynamic Range 125 dB
Attack Time Program dependent using high-speed
opto-couplers
Release Time User variable
Input and Output Connector type Balanced Jack
Sidechain Insert Connector TipRingSleeve Jack
Sidechain Insert Impedance 10 Kohm
Power Requirements 100/120 V AC, 220/240 V AC
Weight 5 lbs/2.2 Kg
Dimensions
19“Wide x 13/4“High x 6“Deep; 482 x 44 x 155mm
Front Panel Controls
Individual Release, Depth and Threshold controls
Hardwire bypass IN/OUT switch
Open/Closed gate status LED
Marker panel for labelling gate assigns
Rear Panel
Balanced Inputs and Outputs, on Balanced jack
connectors
Sidechain Insert/Key Input TipRingSleeve jack on
each channel
AC input connector, with voltage switch and fuse
|
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)
| SHARK FBQ100
Automatic Feedback Destroyer with Integrated Microphone Preamp, Delay Line, Noise Gate and Compressor
High performance single channel Feedback Destroyer with integrated microphone preamp, delay line, noise gate and compressor
Automatically and "intelligently" locates and destroys up to 8 feedback frequencies
Narrow FBQ filters for extremely effective feedback suppression, while keeping highest sonic quality
Ultra-low noise Mic/Line input with Gain control and +48 V phantom power
Delay line with up to 2.5 seconds of delay, adjustable in meters, feet and msec
Noise gate with automatic and manual parameter settings
Automatic compressor with variable density
Subsonic filter with adjustable cut-off frequency
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 |
Features Enhance switch to restore spectral balance of compressed signal Switchable Mode - Dual Channel or Stereo linked Above or Below Threshold LEDs Balanced XLR and Jack Inputs and Outputs Intuitive, user friendly layout Flawless performance
Innovation The ARX COMPO? is a unique compressor/limiter designed for use in any professional audio dynamics control application, packaged in a compact 1 RU chassis.The COMPO operates as two independent compressor/limiters, with 'industry standard' variable Threshold, Ratio and Output gain controls. A Stereo Link switch on the front panel lets both channels track as a stereo pair in a Master/Slave configuration. Enhance Switch The COMPO 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. The best way to think of it is as a 'smart' loudness control. Wide Scale metering There is comprehensive LED indication of all operating functions and status. Above or Below Threshold LEDs on each channel give you instant visual confirmation of compressor activity, as well as easy to read LED Gain Reduction meters. Balanced Inputs and Outputs On the rear panel, each channel has true differential Balanced inputs and outputs, on both XLR and TRS jack connectors. Other features include an Active/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. ARX Quality With its smooth compression, intuitive user friendly layout, high density precision circuitry, and extensive user-variable operating parameters, the ARX COMPO is the ideal installation 'set and forget' compressor, and is also equally at home in Studio, Broadcast and Sound Reinforcement environments.
Every day, ARX Audio interface products solve audio problems for thousands of people around the world.
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 ±0.4dB Signal to Noise ratio -85 dB Unweighted, -99 dB 'A' weighted Distortion .02% THD @ 0dB,1KHz Dynamic Range 107 dB 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
Front Panel Controls Channel A Active/Bypassed IN/OUT switch Threshold, Ratio and Output Gain controls Numbered marker panel for labelling compressor assigns 5 segment LED Gain Reduction display Above/Below Threshold status LEDs Enhance switch and status LED Stereo link switch and status LED Channel B controls identical to Channel A Rear Panel Connectors Balanced XLR Input and Output connectors. Balanced TRS jack Input and Output connectors. AC input connector, with voltage switch and fuse. |
Reviews
There are no reviews yet.