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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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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 |
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)
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Interactive Reference-Class 2-Channel Expander/Gate/Compressor/ Peak Limiter with
Integrated De-Esser, Dynamic Enhancer, Tube Simulation and Low Contour Filter
Whether you?re recording, mixing or mastering, the COMPOSER PRO-XL MDX2600
gives you total dynamic control. It includes all of the features you would expect from
a reference class compressor?plus a voice-adaptive de-esser, a new dynamic
enhancer, authentic tube emulation and much more. These are some of the indispensable
tools for producing clean, consistent audio with virtually no loss in signal quality.
Features:
Switchable IKA (Interactive Knee Adaptation) program-adaptive compression circuitry
combines the advantages of hard-knee and soft-knee characteristics
Integrated de-esser with switchable male/female voice recognition removes excessive
sibilance from your vocal tracks
IGC (Interactive Gain Control) peak limiting circuitry combines clipper and program
limiter for reliable and inaudible protection against signal peaks
Switchable dynamic enhancer for brilliant, lively audio even with heavy compression
IRC (Interactive Ratio Control) expander/gate circuitry for virtually inaudible
noise suppression
Switchable tube simulation for the extra warmth and transparency of classic
tube circuitry
Automatically or manually adjustable attack and release times
Switchable low contour filter prevents "pumping" due to low-frequency
dominated compression
Stereo couple function with independent output level settings
Switchable side chain input with side chain monitor function
Ultra low-noise 4580 operational amplifiers and state-of-the-art THAT¬ VCA's
Separate 12-segment LED meters for input/output levels and gain reduction
Dedicated "traffic light" threshold and de-esser level displays
High-quality detented ALPS¬ potentiometers and illuminated switches
Servo-balanced inputs and outputs with ? in. TRS and gold-plated XLR
connectors
Relay-controlled hard bypass switch with auto bypass function in case
of power failure
Manufactured under ISO9000 certified management system
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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)
| Bring a more professional sound to your mix
Adding a dbx® 266xs 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.
dbx knows compressors...after all we invented them! The 266xs 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 266xs 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.
Features
- Error proof operation to smooth uneven levels, add sustain to guitars, fatten drums or tighten up mixes
- New gate timing algorithms ensure the smoothest release characteristics
- Program-adaptive expander/gates
- Great sounding dynamics control for any type of program material
- Separate precision LED displays for gain reduction, compression threshold and gate threshold allow quick, accurate setup
- Stereo or dual-mono operation
- Balanced inputs and outputs on 1/4" TRS and XLR connectors
- Side Chain insert
- Classic dbx® "Auto" mode
Specifications
Input Connectors |
1/4" TRS and female XLR (pin 2 hot) |
Input Type |
Electronically balanced/unbalanced, RF filtered |
Input Impedance |
40kΩ balanced/unbalanced |
Max Input |
>+22dBu Balanced or Unbalanced |
Output Connectors |
1/4" TRS, female XLR (pin 2 hot) |
Output Type |
Impedance-balanced/unbalanced, RF filtered |
Output Impedance |
+4dBu: 100Ω balanced, 50Ω unbalanced; -10dBu: 1kΩ balanced, 500Ω unbalanced |
Max Output |
>+21dBu balanced/unbalanced into 2kΩ or greater; >+18dBm balanced/unbalanced (into 600Ω) |
Sidechain |
1/4" TRS Phone, Normalled: Ring = Output (send); tip = Input (return) |
Sidechain Impedance |
Tip = >10kΩ (Input), Ring = 2kΩ (Output) |
Sidechain Max Input Level |
Tip = >+22dBu (Input) |
Sidechain Max Output Level |
Ring = >+20dBu (Output) |
Compressor Threshold Range |
-40dBu to +20dBu |
Compressor Threshold Characteristic |
Selectable OverEasy® or hard knee |
Compressor Ratio |
1:1 to Infinity:1 |
Compressor Attack Time |
Scalable Program-Dependent AutoDynamic™ |
Compressor Release Time |
Scalable Program-Dependent AutoDynamic™ |
Expander/Gate Threshold Range |
-60dBu to +15dBu |
Expander/Gate Ratio |
1:1 to 4:1 |
Expander/Gate Attack Time |
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