SAMSON S-GATE 4 FOUR CHANNELS GATE-DUCKER

The S gate 4 is a versatile four channel Expander/Gate/Ducker that can be operated as
four discrete gates or linked into two stereo pairs. The unit's discerning circuitry provides
precise frequency dependent triggering with two bands of highly accurate, selectable filters.
Its handling of audio is fast and precise offering subtle control, and can be sonically invisible
if so desired.
The S gate 4's powerful Ducker feature allows the user to automatically lower the signal of
a channel or stereo pair. You can also lower or "duck" the signal level by inserting an external
source into the Key Input. This is especially useful for utilizing another audio source, such
as a solo instrument or voice-over to "duck," or regulate, a rhythm track or stereo program
track. A Key Listen switch enables monitoring of the key signal.
The S gate 4's Link switch allows the settings on channels 1 and 3 to control the processing
of channels 2 and 4. In this mode, stereo Gating and Ducking can be accomplished easily.
Each of the gate channels has High and Low Pass Filters, Threshold, Attack, Release,
and Range Controls for precise manipulation. An 8-segment Gain Reduction LED allows
the user to monitor the circuit's effect. Ducker, Link, In/Out, Filter and Key Listen switches
add increased versatility. 

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)

OPTIMOD-FM 5500 puts coveted five-band and two-band OPTIMOD processing into a single rack unit package and brings it to you at the most affordable price ever.

Quality sound is what 5500 is all about-sound that attracts audiences by providing a polished, professional presentation regardless of format and source material. Exceptional versatility allows you to adjust the processor's audio texture to brand your audio, knowing that the resulting signature sound will remain consistent, cut-to-cut and source-to-source. Branding builds businesses and no other processors have the consistency to brand your sound like OPTIMOD.

With the 5500, your signature sound is just a preset away. An easy, one-knob Less/More adjustment allows you to customize any factory preset, trading cleanliness against processing artifacts according to the requirements of your market and competitive environment. Full Control gives you the versatility to customize your audio further. And, if you're a hard-core processing expert, you can explore Advanced Control to tweak presets at the same level as Orban's factory programmers. This versatility makes the 5500 a superb choice for any format. Its five-band processing is ideal for any pop music format (even the most competitive and aggressive CHR), while phase-linear two-band processing yields ultratransparent sound for classical, classic jazz, and fine arts formats. Regardless of your choice, 5500's optimized technology ensures unusually high average modulation and coverage for a given level of subjective quality. Unlike many lesser processors, the 5500 handles speech particularly well. It's always clean, even when you process for loudness.

If you're concerned about latency because you need to feed live talent headphones off air, be assured that the 5500's ultra-low-latency (5 ms delay) processing will keep the most finicky talent happy. Or use optimum latency (15 ms delay) processing for the most competitive sound with delay that's still low enough to satisfy most any talent.

The 5500 can also be used as a superb stand-alone stereo encoder with latency as low as 2 ms and full overshoot limiting in both the left/right and composite baseband domains.

When used in this mode, the 5500 must be driven (usually via an STL) by a full-featured FM audio processor (like Orban's 8600) that incorporates pre-emphasis aware HF limiting and peak control. In both modes, the 5500's stereo encoder helps deliver a transmitted signal that's always immaculately clean and perfectly peak limited, with full spectral protection of subcarriers and RDS/RBDS regardless of the amount of composite limiting.

It is the ideal choice for network broadcasters who process with Orban's flagship OPTIMOD-FM 8600 at the network origination point and who need a processor at every transmitter to eliminate STL overshoots (using the 5500's stand-alone stereo encoder mode) and/or to process local insertions while also eliminating network STL overshoots (using the 5500's audio processor / stereo encoder mode). Moreover, the 5500's two modes make it easy for large government and network broadcasters to manage its inventory of spares because any 5500 can be used as a stereo encoder with or without audio processing.

Available in both modes, the built-in, defeat-able ITU-BS412 multiplex power controller allows the 5500's output to meet even the most stringent European government regulations.

A 10 MHz frequency reference input allows the stereo pilot tone frequency to be locked to GPS or another high-accuracy frequency standard. This improves the performance of single-frequency networks in areas where coverage of the transmitters overlaps.

The 5500's built-in stereo encoder, AES/EBU digital inputs and outputs, and analog I/O permit hasslefree interfacing to any broadcast plant, whether the 5500 is located at the studio or the transmitter. Tight band limiting to 15 kHz means you can use any uncompressed digital STL to pass 5500- processed audio from studio to transmitter without compromising on-air loudness - there's no need to use STL's having 44.1 or 48 kHz sample-rate.

The stereo encoder's stereo sub-channel modulator can operate in normal double sideband mode and in an experimental compatible single sideband mode (SSB/VSB) that is offered to enable users to compare and assess the two modes.

Analog Fallback to Digital control that allows Silence Sense to switch the active input from Analog to Digital if silence is detected in the analog input signal but not on the digital input signal. This function works vice versa as well on both analog and digital AES input. The silence sense parameters apply to both simultaneously and both detectors are available to drive the 5500's tally outputs and sending SNMP Traps/Alerts.

If you want to locate the 5500 away from the studio, you'll be pleased by its three separate remote control ports - GPI contact closures, RS232 serial and built-in Ethernet for TCP/IP networks. The serial and Ethernet ports are supported by the supplied 5500 PC Remote Control application. This Windows® 2000/XP/Vista/7/8 application allows you to do even more with the 5500 than you can do through its front panel, making remote control a pleasure.

5500 PC Remote software allows you to access all 5500 features and allows you to archive and restore presets, automation lists, and system setups (containing I/O levels, digital word lengths, GPI functional assignments, etc.)

Built-in clock-based automation lets you automatically day-part the processing. You can control many other 5500 operating parameters too.

The 5500's feature set fully exploits the processor's DSP and computer-based control architecture. To ensure absolute accuracy, you can automatically synchronize the clock to an Internet timeserver. It has a cool-running, energy-efficient switching power supply and uses the latest dual-core DSP chip technology from Freescale Semiconductor.

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

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)