| Content | The introduction of the Drawmer DL241 Dual Channel \'Auto-Compressor\' highlights the beginning of a new generation of dynamic processors which set new standards of sonic excellence. Its combination of features, displays and well designed control functions make it an outstanding product for a vast range of broadcast, sound reinforcement, and recording applications. The DL241 \'Auto-Compressor\' incorporates many automatic functions allowing the engineer to achieve optimum results in the minimum set up time but does not compromise skilled users with limited controls. Its sophisticated design produces a smooth, transparent and warm sonic performance.
Facilities available on the DL241 include:
Expander/Gate : Drawmer\'s unique "Programmed Adaptive Expansion" circuitry allows effective \'single control operation\' producing a fast smooth response and eliminating \'chatter\' on or around threshold. This design differs from that of conventional Expanders in that the Expansion Ratio, Hold Attack and Release continually adapt themselves to the dynamics of the incoming signal. The Expander/Gate provides 80dB range and variable threshold from +20dB to 70dB with an OFF position. Switching is provided for \'Fast\' or \'Slow\' release characteristics. The \'Fast\' mode can be used for instruments with rapid decay times or for additional tightening of the gating effect. The \'Slow\' mode is more suitable for vocals and instruments with longer decay times or can be used to smooth over rough fade outs. A single LED indicates gating action.
Compressor : A full Auto Attack/Release Compressor with variable Threshold and Ratio successfully combines the smoothness of the \'Soft Knee\' principle with the precision of a \'Ratio\' control. The advantage of this combination is that instead of the onset of compression being sudden at the threshold point the ratio increases over a 10dB input level range until it reaches the ratio set on the front panel control, thus achieving a more natural sounding compression. Selectable manual or fully automatic Attack and Release times dramatically extend the application range and user scope of the DL241. Selecting the \'Auto\' mode removes the need for manual adjustment and is particularly valuable when the program being treated does not have predictable or consistent dynamics, often the case when processing a complete mix or a virtuoso bass guitar performance. Under manual control the wide range of adjustment available gives the user the ability to create dynamic effects as well as normal compression.
Peak Level. \'Zero Response Time\', \'Zero Overshoot\' circuitry : This provides an absolute limit to the peak level of the output signal, adjustable from OdB to +16dB above system level. When set to the required level, any excessive peaks will be stopped and the gain momentarily decreased if necessary. The \'Zero Response Time\' is essential to avoid unpleasant distortion when mastering for CD or sending any signal to digital equipment. \'Zero Response Time\' means that the DL241 provides complete protection from even the fastest and shortest signal \'spikes\' (overloads), making the unit also ideally suited for speaker system protection in sound reinforcement applications.
Metering : Comprehensive bar graph displays give \'at a glance\' indications of gain reduction, and output signal level Gain reduction from 1dB to -30dB is shown simultaneously with signal level from -20dB to +15dB Other single LEDs indicate function status of Gating Acton, Auto Mode, Bypass, Peak Limiting, Stereo Linking, and Power On.
Stereo Link : When processing stereo signals, linking should be used to prevent image shifting. This facility ties both channels together. Using channel 1 (left) as the master channel removes the need to duplicate control settings for both channels.
Bypass : A fully balanced hard-wire bypass connects the input directly to the output allowing signal to pass through the unit with no power applied. In bypass the VU level meter displays the Input signal.
Operating Level : The operating level of the DL241 can be instantly switched, independently on each channel, for either +4dBu or -10dBu operation using a rear panel push switch.
Input/Output : The DL241 is available with either XLR connections and fully balanced inputs and outputs, or with stereo jack (TRS) connections, fully balanced inputs and single ended outputs. |
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.
| 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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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)
|
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
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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 |
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