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 […]
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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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
GENERAL DATA
Power Supply 220 / 110V 25 VA
Dimension 434x351x44mm (1 rack unit)
Weight » 4 Kg
MEETING INTERFACE IN / OUT
Connector DB 9 female
Nominal in / out audio level 0 dBm
Input impedance 600 W / 10 KW (selectable)
Output impedance 100 W
* nominal line level: - 6 dBm
** it can vary depending on characteristics of telephone line
All measurements are intended at 1 kHz.
AUDIO PROGRAM IN / OUT
Connectors XLR, electr. balanced
Input impedance 600 W / 10K W (selectable)
Output level * - ¥ ¸ + 16 dBm
Output impedance 100 W
Noise on Receive output
2-Wire separation £ 25 dB**
4-Wire separation £ 70 dB
2-WIRE SECTION
Connector RJ11
Nominal input level - 6 dBm
Nominal output level - 6 dBm
Compensation mode Electronic, transf. decoupled
Impedance 600 W
4-WIRE SECTION
Connector Terminal block
Impedance 600 W
Nominal TX level 0 dBm
Nominal RX level 0 dBm
Product Features:
Ultra-high resolution 24-bit/96 kHz mastering processor featuring 32/40-bit floating-point DSP technology Audiophile 24-bit/96 kHz A/D- and D/A converters offering 113 dB dynamic range 4 concurrently selectable EQ modules (31-band graphic EQ, 10-band parametric EQ, Feedback Destroyer plus 3 Dynamic EQs per stereo channel) Ultra-high resolution 61-band real-time FFT analyzer with additional auto EQ function for room and loudspeaker equalization Unique VPQ (Virtual Paragraphic EQ) option allows parametric control of graphic EQs State-of-the-art compressor/expander with peak limiter per stereo channel, additional stereo imager and stereo delay for delay line applications Multi-functional level meters (peak/RMS, VU and SPL meter with dBA/dBC weighting via RTA/Mic input) 64 user memories for complete setups and/or individual module configurations Separate mic/line input with phantom power for RTA and Auto-EQ applications Balanced inputs and servo-balanced outputs with gold-plated XLR connectors, stereo aux output, AES/EBU and S/PDIF inputs and outputs (XLR and optical) Professional Wordclock input and MIDI connections for full remote control, preset dumps and system updates Open architecture allowing future software updates via MIDI "Planet Earth" switching power supply for maximum flexibility (100 - 240 V~), noise-free audio, superior transient response plus low power consumption for energy saving High-quality components and exceptionally rugged construction ensure long life Conceived and designed by BEHRINGER Germany
Renaissance Compressor
Designed to provide the classic warm sound of analog compressors, the legendary
Renaissance Compressor controls dynamics with studio-style warmth that flatters
full mixes, vocals, or instruments. Controls include selection of vintage-style Opto
or modern Electro compression, and threshold, ratio, and attack controls. Waves
ARC? (Automatic Release Control) algorithm dynamically optimizes the compressor's
release value for a wide-ranging input. ARC? reacts much the way a human ear
expects, and can produce increased RMS level with greater clarity.
In general, the release is faster for peak transients and slower for the overall RMS
level. The ARC? system varies the release time to fit the ear's expectations while
increasing RMS, and without creating distracting artifacts. In this way, the
Renaissance Compressor can serve as a leveler plus a fast compressor simultaneously.
L2 Ultramaximizer Peak Limiter
Heard on countless hit records and soundtracks, the L2 Ultramaximizer puts sound
up-front with breathtaking transparency. The L2 is capable of a very fast, overshoot-free
response. Once the limiter threshold has been set, you can define the actual peak level
that the processed signal will reach. Once set, limiting and level re-scaling becomes a
one-shot process. The L2 can significantly increase the average signal level without
introducing any audible side effects. Yet there is plenty of range to recreate "vintage"
effects such as level pumping or severely limited dynamic range if you like.
Pristine Sound Quality
The MaxxBCL offers 96kHz, 24-bit resolution with a 48-bit, double precision internal
processing path and a dynamic range of ~125dB. The totally passive analog input
path to the ADC uses Jensen transformers, while the output path also features Jensen
analog output transformers. The unit is galvanic-isolated, which prevents ground loops
and allows it to operate in electrically unstable environments. Waves MaxxBCL combines
the highest quality converters available with unequalled processing algorithms to offer you
astonishing new power in bass enhancement, dynamics processing, and format conversion
(analog-to-digital, digital-to-analog, and digital re-quantization).
Versatile Connections
The MaxxBCL supports optical, coaxial S/PDIF, AES/EBU, balanced and unbalanced
line-level analog signals. MaxxBCL features a unique set of input and output trim settings:
analog input headroom can be set from 9dB above 0dBu to 24dB in six precise steps
(using the rear-panel selectors). Analog output level is similarly set with a separate
rear-panel trim pot offering you the ability to connect between devices with a wide range
of input and output levels.
Waves MaxxBCL Features:
User Interface:
Clear backlit displays, meters, and buttons are visible in all lighting conditions
THD + Noise: ? 0.0006 % @ 1kHz @ -1dBFS
Precision metering with resettable peak hold options (2 sec, infinite)
Accurate, wide-range metering covering 90dB for input and output
and 12dB for compressor and limiter attenuation
Quick access independent bypass on each processing block
Tactile feedback knobs
Four easy store/recall user presets
Input / Output:
Analog, AES/EBU, S/PDIF
Independent rear-panel input and output headroom calibration in six steps
(+9, 12, 15, 18, 20, and 24 dBu)
Passive analog inputs to the A-to-D converters using Jensen analog input transformers
Jensen analog output transformers
High-performance IDR dithering to 16- or 24-bit output
Precise input level setting using 1% resistor networks
Processing:
48-bit end-to-end internal processing path
Switchable compressor/MaxxBass? order
MaxxBass:
Adjustable processor frequency from 25 to 120Hz
Adjustable harmonic mix percentage from 0 to 100%
High-pass Filter "harmonics only" option
Compressor:
Opto/Electro mode selection
Threshold (0-60dB), Ratio (1:1-12:1), and Attack (0.5, 1, 2, 5, 10, 20, and 50 ms) controls
Proprietary ARC? Automatic Release Control
Automatic gain makeup
Limiter:
Overshoot-free look-ahead processing
Adjustable Threshold (0-18dB)
Adjustable Output Ceiling (0-18dBFS)
Description
The alphaton Feedback-Controller FC-100 is a revolutionary product for every microphone application including wireless types. It can provide effective feedback control to any sound reinforcement situation. It can also eliminated string feedback in guitars and bass instruments which are equipped with piezo or condenser pickups. The FC-100 provides 6 to 10 dB of feedback protection by analysing the incoming signal, detecting the presence of feedback and then notching out the feedback frequency. The FC-100 relieves the sound engineer automatically. It works dynamically and eliminates a feedback signal before it can grow into a peak. Internally a discriminator circuit detects the presence of regenerative feedback and triggers the automatic equaliser to insert a notch where the feedback peak is coming up. This normally prevents other peaks from developing but if they do, the process is repeated at high speed, thus keeping feedback under control.
Technical Data In- and Outputs transformer balanced max. Inputlevel + 3 dB Gain (RS=200 Ohm, RL=2kOhm) 0 dB Frequency Response 20 Hz ... 20 kHz, +/- 1,5 dB Feeback Cancellation 6 ... 10 dB Power Requirement 30 ... 48 V Phantom Current drain ca. 8 mA External Power Supply 9 ... 20 V DC, 3,5 mm jack, tip + Housing Aluminium Diecast Weight 285 grams Dimensions 112 x 60 x 32 mm
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
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, but as we design smaller playback systems with small speakers, <br>
there are also some tradeoffs, particularly in music. Clever cabinet designs and <br>
high-excursion drivers can get you only so far. Any audio engineer trying to extract <br>
bass from real-world components must eventually face the laws of physics: <br>
All things being equal, small boxes and speaker cones can't move enough air <br>
to produce a room-shaking low end. When working with systems tiny enough to <br>
toss in a shoulder bag, the challenge is even more formidable. Interestingly enough, <br>
Waves MaxxBCL has a new solution to this problem using a very old concept.
</p>
<p>
Waves MaxxBCL at a Glance:<br>
MaxxBass? Bass Enhancement adds deep bass sound without adding bass frequencies<br>
Waves Renaissance Compressor<br>
L2 Ultramaximizer Peak Limiter<br>
24-bit/96kHz resolution with 48-bit, double-precision processing<br>
Supports optical, coaxial S/PDIF, AES/EBU, balanced and unbalanced line-level analog signals
</p>
<p>
MaxxBass Bass Enhancement<br>
Waves' patented algorithm adds stunning bass sound without adding bass frequencies, <br>
delivering a bigger bottom end. MaxxBass? allows your audience to hear bass frequencies <br>
lower than physically present by applying psychoacoustic principles: Even though we <br>
can hear a bass guitar from a small speaker, we don't actually hear the fundamental <br>
frequency because the speaker can't produce a pitch that low. Instead, we hear the <br>
harmonics that the speaker can produce, and this causes the brain to create the "acrobat_reader_logo."){kind=logo}
fundamental." MaxxBass? takes this well-known psychoacoustic phenomenon to the 
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