The DS201 is a sophisticated dual channel noise gate incorporating a number of features pioneered by Drawmer, which are invaluable to the sound engineer, and not found on conventional noise gates. Features: Variable high pass and low pass filters for “frequency conscious” gating. Comprehensive envelope control, attack, hold, decay, and range. Key input for external triggering. […]
The DS201 is a sophisticated dual channel noise gate incorporating a number of features pioneered by Drawmer, which are invaluable to the sound engineer, and not found on conventional noise gates.
Features:
Variable high pass and low pass filters for “frequency conscious” gating.
Comprehensive envelope control, attack, hold, decay, and range.
Key input for external triggering.
“Key Listen” facility.
Extremely fast attack time, to preserve the natural attack of the sound.
Balanced inputs and outputs.
Stereo linkable.
Can be used for “Gating” or “Ducking“.
High audio specification.
These features and its “user friendliness” have made it the “industry standard” throughout the world.
Setting up noise gates can be tedious and unrewarding. For example, when attempting to separate a signal from unwanted noise or crosstalk that is relatively high in level, spurious triggering of the gate by unwanted components within the sound can be a serious problem. This can often be experienced in the studio when recording a drum kit. If a gate is used to clean up the snare drum sound it is quite likely that the nearby hi-hats will spill into the snare drum microphone and cause the gate to open. Increasing the threshold level may cure this problem, but then there is a very real danger that any quieter snare drum beats may not cause the gate to open at all and the performance can easily be ruined.
The DRAWMER solution to this problem is the inclusion of two variable filters, one high-pass and one low-pass, which act upon the side-chain keying circuitry. By setting the output switch to key listen the user can hear the action of the filters and adjust them to reject high frequency spillage from the hi-hats. This now enables the gate to only open on the lower frequencies present in the snare drum.
Drawmer pioneered the concept of frequency conscious gating, and the Drawmer DS201 was the first noise gate to include this feature.
The very fast attack of the DS201 means that it can open in a matter of micro seconds, thus preserving the natural attack of whatever sound is being gated and the comprehensive envelope controls mean that the gain can be changed at whatever rate best suits the material being processed. With vocals for example, a fairly fast attack is needed so as not to clip the leading consonants but a slower release time will prevent the end of words being clipped off and will also fade out any noise gradually rather than cutting it off abruptly. This latter point is very important as the human ear is far more sensitive to rapidly changing noise levels than to a constant low level noise.
Another important role played by the gate is in the reshaping of existing sounds. An example of this might involve a simple sampling delay line used to store a sound to be re-triggered later and added to a mix. A gate can be used to impart a slower attack or faster decay to the sound and careful setting of the decay envelope can effectively hide any noise present at the end of the sample. When using a sampler or drum machine the gate can be considered a triggered envelope shaper, and the wide range allowed by the envelope controls make the DS201 ideal for this application.
The unit can be switched to accept a key input, which allows the gate to be triggered externally. An example of this would be to use a snare drum signal to open the gate on a separate ambient microphone to create a natural alternative to gated reverb.
Each channel can be individually switched from Gating to Ducking for \’voice over\’ applications or the removal of \’clicks\’ and \’pops\’.
Weight
3.50 kg
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The DS201 is a sophisticated dual channel noise gate incorporating a number of features pioneered by Drawmer, which are invaluable to the sound engineer, and not found on conventional noise gates.
Features:
Variable high pass and low pass filters for "frequency conscious" gating.
Comprehensive envelope control, attack, hold, decay, and range.
Key input for external triggering.
“Key Listen" facility.
Extremely fast attack time, to preserve the natural attack of the sound.
Balanced inputs and outputs.
Stereo linkable.
Can be used for "Gating" or "Ducking“.
High audio specification.
These features and its "user friendliness" have made it the "industry standard" throughout the world.
Setting up noise gates can be tedious and unrewarding. For example, when attempting to separate a signal from unwanted noise or crosstalk that is relatively high in level, spurious triggering of the gate by unwanted components within the sound can be a serious problem. This can often be experienced in the studio when recording a drum kit. If a gate is used to clean up the snare drum sound it is quite likely that the nearby hi-hats will spill into the snare drum microphone and cause the gate to open. Increasing the threshold level may cure this problem, but then there is a very real danger that any quieter snare drum beats may not cause the gate to open at all and the performance can easily be ruined.
The DRAWMER solution to this problem is the inclusion of two variable filters, one high-pass and one low-pass, which act upon the side-chain keying circuitry. By setting the output switch to key listen the user can hear the action of the filters and adjust them to reject high frequency spillage from the hi-hats. This now enables the gate to only open on the lower frequencies present in the snare drum.
Drawmer pioneered the concept of frequency conscious gating, and the Drawmer DS201 was the first noise gate to include this feature.
The very fast attack of the DS201 means that it can open in a matter of micro seconds, thus preserving the natural attack of whatever sound is being gated and the comprehensive envelope controls mean that the gain can be changed at whatever rate best suits the material being processed. With vocals for example, a fairly fast attack is needed so as not to clip the leading consonants but a slower release time will prevent the end of words being clipped off and will also fade out any noise gradually rather than cutting it off abruptly. This latter point is very important as the human ear is far more sensitive to rapidly changing noise levels than to a constant low level noise.
Another important role played by the gate is in the reshaping of existing sounds. An example of this might involve a simple sampling delay line used to store a sound to be re-triggered later and added to a mix. A gate can be used to impart a slower attack or faster decay to the sound and careful setting of the decay envelope can effectively hide any noise present at the end of the sample. When using a sampler or drum machine the gate can be considered a triggered envelope shaper, and the wide range allowed by the envelope controls make the DS201 ideal for this application.
The unit can be switched to accept a key input, which allows the gate to be triggered externally. An example of this would be to use a snare drum signal to open the gate on a separate ambient microphone to create a natural alternative to gated reverb.
Each channel can be individually switched from Gating to Ducking for \'voice over\' applications or the removal of \'clicks\' and \'pops\'.
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
>+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
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
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.
Versatility doesn't stop with sound.
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.
VIRTUALIZER 3D FX2000
High-Performance 3D Multi-Engine Effects Processor
71 breathtaking new algorithms—true RSM (Real Sound Modeling) stereo and 3D effects
Wave-adaptive virtual Room reverb algorithms for natural reverb and delay
Awesome modulation, dynamic, psychoacoustic and EQ algorithms
Authentic amp simulation, distortion and special effects
11 effect combinations with selectable serial / parallel configuration
Up to 7 adjustable parameters plus 2-band EQ per effect
24-bit A/D and D/A converters with 64/128-times oversampling
True stereo processing for realistic channel separation in stereo image
100 factory presets plus 100 user memory locations
Extensive MIDI implementation
Accurate LED level meters for perfect level setting and optimum performance
Servo-balanced XLR and ¼'' TRS inputs and outputs
High-quality components and exceptionally rugged construction ensure long life
Conceived and designed by BEHRINGER Germany
Reverbs & Delays
Because it is one of the most desired effects, the FX2000 provides 12 different reverb programs, so you always have the ideal reverb for your live performance or studio needs.
Reverbs include:
CATHEDRAL
GOLD PLATE
SMALL HALL
ROOM
STUDIO
CONCERT
STUDIO
STAGE
SPRING REVERB
AMBIENCE
EARLY REFLECTIONS
Specialty reverbs include:
REVERB – reverb is synthetically turned off after a predetermined amount of time
REVERSE REVERB – reverb envelope is reversed—it slowly gets louder
Delays include:
STEREO DELAY – delay is processed across the entire stereo image
TAPE ECHO – simulates classic tape echo devices, pre-dating the advent of digital delays
PING PONG – delay signal is "bounced” from left to right at an adjustable tempo
Much More than Just Reverb and Delay
The FX2000 has some of the best-sounding reverb and delay programs, but thatʼs not all. It also provides excellent modulation effects (such as chorus, flanger and phaser), including special variations like musical pitch shifter, tremolo and even a rotary speaker simulation.
Modulation and Pitch Shifter FX:
STEREO FLANGER (originally generated by playing back two synchronized "reel to reel” tape recorders with a finger rubbing on the flange of one of the take-up reels)
VINTAGE FLANGER simulates a guitar fl anger stomp box
JET STREAM FLANGER sounds like a classic analog flanger
STEREO CHORUS (combines a slightly detuned signal with the original)
ANALOG CHORUS simulates a guitar chorus stomp box
VINTAGE CHORUS imitates a classic analog studio chorus
ULTRA CHORUS creates the sound of an eight person chorus
STEREO PHASER (combines a second, phaseshifted signal to the original)
VINTAGE PHASER represents a guitar phaser stomp box
DUAL PHASER processes the left and right channels separately
ROTARY (simulation of the rotating speakers typically used on an organ) – Slow or Fast
PITCH SHIFTER (changes the pitch of the original signal) – can be used to create harmonies with the original signal or replace it entirely with the altered pitch. Choices include stereo, two and three vocal pitch shifter
VIBRATO – the peak frequency of the tone is periodically and uniformly changed (quickly or slowly)
TREMOLO (common vintage guitar amplifier effect) – a fast or slow periodic variation in volume
AUTO PANNING – signal is automatically sent from one side of the stereo image to the other, either once or multiple times
Dynamic FX:
COMPRESSOR – reduces the dynamic range of the signal, maintains consistent signal level and thus avoids distortion associated with excessive input levels
EXPANDER – effectively broadens the dynamic range of source signals while reducing background noise
GATED REVERB – helps reduce background clutter by turning reverb off below a predetermined threshold level. Particularly effective on drum mics and vocals
ANA. KOMPR/LIM. — similar to COMPRESSOR but with Limiting functionality
ULTRAMIZER – analyses incoming signal and automatically applies compression across two independent frequency bands
DENOISER – eliminates or reduces noise and other interference
DE-ESSER – reduces or removes sibilance (Ssss sound) from signal
WAVE DESIGNER – allows you to influence the envelope by adjusting Attack and Release of the signal
Psychoacoustic FX:
EXCITER – adds artificially generated overtones to the original signal, increasing presence and perceived loudness without significant increase in signal level
ENHANCER – functions much like a dynamic pitch equalizer
ULTRA BASS – sub-harmonic processor combined with bass exciter and limiter
STEREO IMAGER – divides the signal into middle and side signals, allowing individual signals to be amplif ed when desired and placed on the stereo image
ULTRA WIDE – creates a broader stereo image
BINAURALIZER – also creates a broader stereo image and compensates for crosstalk between both speakers
Filter/EQ FX:
AUTO FILTER – influences the frequency response of a signal. Two filters are included: low pass allows low frequencies to pass and suppresses high frequency content; high pass does the exact opposite
LFO FILTER - Controls the rate of oscillation effects
PARAMETRIC EQ – allows you to control the bandwidth, frequency and amplitude of a signal
GRAPHIC EQ – the sound spectrum is divided into eight (8) adjacent frequency bands, which can be cut or boosted, bandwidth is predetermined
Distortion FX and Amp Simulations
The FX2000 is also equipped with distortion, amplifier and speaker simulation including VOCAL DISTORTION, TUBE DISTORTION, GUITAR AMP, TRI FUZZ, SPEAKER SIMULATION, RING MODULATOR and LO-FI.
Special FX:
VINYLIZER – adds clicks and/or noise to the signal, reminiscent of old vinyl records and tape machines
SAMPLER – allows you to record and playback up to five (5) seconds of program material
VOCODER – allows the input signal to modulate another signal (usually a synthesizer sound), creating the familiar "talking synthesizer” effect
VOICE CANCELER – removes mono vocal parts from stereo recordings for "instant Karaoke”
RESONATOR – simulates and oscillating system that amplifies a specific frequency
Our FX Combinations Go to 11
Sometimes you want to add a little (or a lot) of color to your reverb and delay patches. The FX2000 allows you to layer modulation effects like chorus, flanger, pitch or tremolo with your reverb or delay selections.
Enhanced User Editability
Logical grouping of parameters, along with the combination of encoders, buttons, LEDs and an easily readable LED display, make operating the FX2000 a breeze. You are free to edit up to seven parameters per preset, and then save them for future use in the 100 provided memory locations.
Value
With its extremely powerful processing capability and versatile array of features, the FX2000 will become the busiest tool in your audio arsenal. Stop by your BEHRINGER dealer today and find out why more professional sound engineers are turning to the FX2000 as their primary FX processor— both in the studio and on the road.
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
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