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

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

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.

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”.

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.

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.

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.

• 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)

• 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

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)

• 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

Shure DP11EQE Dynamics Processor Specification:

    GENERAL:
        Frequency Response: 20 to 20k Hz +/- 1.0 dB re 1 kHz
        Dynamic Range: 104 dB minimum, A-weighted, 20 Hz to 20 kHz
        Sampling Rate: 48 kHz
        Digital-to-Analog, Analog-to-Digital Conversion: 20 bit resolution
    Voltage Gain:
        1 dB +/- 1dB (power off)
        12 dB +/- 2 dB (input 10 dBV, output +4 dBu)
        12 dB +/- 2 dB (input +4 dBu, output 10 dBv)
        0 dB +/- 2 dB (equal input and output sensitivities)
    Impedance:
        Input: 47 kilohms +/- 20% actual
        Output: 120 ohms +/- 20% actual
    Input Clipping Level
        +18 dBu minimum (at +4 dBu setting)
        +6 dBV minimum (at 10 dBV setting)
    Output Clipping Level
        +18 dBu minimum (at +4 dBu setting)
        +6 dBV minimum (at 10 dBV setting)
    Total Harmonic Distortion
            LED Signal Indicators
        Signal: 40 dB
        Clip: 6 dB down from input clipping
    Propagation Delay from Input to Ouput
        0.8 ms (all filters flat, no dynamics processing, 0 ms delay), up to 2.1 ms (all processing enabled)
    Polarity
        Input to output: inverting optional (default: non-inverting)
        XLR: pin 2 positive with respect to pin 3
        1/4-inch TRS: tip positive with respect to ring
    Operating Voltage
        DP11EQ: 120 Vac, 50/60 Hz, 50 mA max
        DP11EQE: 230 Vac, 50/60 Hz, 25 mA max
        DP11EQJ: 100 Vac, 50/60 Hz, 50 mA max
    Temperature Range
        Operating: 0 degrees to 60 degrees C (32 degrees to 140 degrees F)
    Fuse
        DP11EQ: 120 Vac. Fuse: 100 mA, 250V time delay
        DP11EQE: 230 Vac. Fuse: 50 mA, 250 V time delay
        DP11EQJ: 100 Vac. Fuse: 100 mA, 250 V time delay
    Dimensions
        219 mm x 137 mm x 44 mm (8-5/8 in x 5-3/8 in x 1-3/4 in)
        Weight: 930 g (2.05 lbs)
    DYNAMICS PROCESSOR
        Gate and Expander:
        Threshold: 72 to 1 dB, 0.5 dB resolution
        Attack: 1.0 to 200 ms
        Decay: 0.05 to 1 second
        Gate Hold Time: 0 to 0.5 seconds
    AGC Leveler
        Threshold: 72 to 1 dB, 0.5 dB resolution
        Attack: 0.2 to 3 seconds
        Decay: 0.5 to 5 seconds
        Hinge: Threshold to 1 dB, 0.5 dB resolution
    Compressor and Limiter
        Threshold: 72 to 1 dB, 0.5 dB resolution
        Attack: 1.0 to 200 ms
        Decay: 0.05 to 1 second
        Knee: Hard or Soft selectable
    No Overshoot Peak Limiter
        Propagation delay: 1 ms
        Threshold: 72 to 1 dB, 0.5 dB resolution
        Attack: 0 msDecay: 100 ms
    PARAMETRIC EQUALIZER
        Parametric Filter Frequency Bands
        Up to 9 bands, variable frequency, variable Q
    Boost/Cut Range
        +6 dB to 18 dB per band
        Q Range
        1/40-octave to 2 octave
    High and Low Shelf/Cut Filters
        Shelf, +6 to 18 dB per filter
        Cut, 12 dB per octave nominal
    DELAY
        Up to 1.3 seconds, 21 microsecond resolution

In today's recording and sound reinforcement environments, the need for multiple channels of high quality, easy to use compression is growing rapidly. The new dbx® 1046 is designed to provide the audio professional with just that: 4 channels of great sounding dbx compression for a variety of applications. Incorporating the industry standard dbx designs and the latest available manufacturing techniques, the dbx 1046 provides pristine sonic quality with that classic dbx sound.

The 1046 provides 4 channels of smooth classic dbx OverEasy® or Hard Knee compression that are perfectly suited for use on individual tracks of your multitrack recorder, and in virtually all applications the separate channels can be individually interfaced and used for entirely independent purposes. Additionally the newly developed PeakStopPlus™ is ideal for protecting your system from the oppressive peaks that can take out valuable drivers in your sound reinforcement rig or studio monitors.

All four channels have the following controls:

Threshold - allows you to set the level at which the compressor starts affecting the gain

OverEasy - allows you to select between soft compression for overall gain control or Hard Knee compression based on the characteristics of the original dbx 160.

Ratio - allows you to set the slope of gain reduction affecting the signal over the threshold level.

Input/Output Meter - allows you to set the meter to check the input and output levels for maximum signal-to-noise ratio and best level matching.

Output Gain - allows you to add make-up gain or to adjust the output level for that channel to match the next device's input gain.

Bypass - hard wire bypass allows you to hear how the compressor is affecting your signal.

PeakStopPlus - allows you to set the maximum signal level you want to pass through this channel. While it's virtually impossible to eliminate distortion, the PeakStopPlus circuit does it gracefully and effectively with minimal distortion.

Stereo Link - allows you to link channels 1&2 and 3&4 for two channels of true stereo compression.

All four channels feature balanced gold plated XLR and 1/4" inputs and outputs, and switchable +4dBu or -10dBV operating level to interface each individual channel with any other device.

So whether you need to control the level, placement in the mix, or overall characteristics of 4 independent signals or control the gain leveling on a couple of stereo pairs, the dbx 1046 is for you.

• Four independent channels of operation, stereo linkable in two pairs
• PeakStopPlus™ limiting control for setting maximum allowable level regardless of compressor settings
• Independent Threshold and ratio controls
• Switchable OverEasy® or Hard Knee compression
• Classic dbx® compression
• Differentially balanced gold-plated XLR and 1/4" inputs and outputs
• True RMS level detection
• Precision metering of input level, output level, and gain reduction
• Dual True stereo or quad mono operation
• Switchable +4dBu or -10dBV operation per channel