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The solution for complex sound-reinforcement challenges and permanent installations.
The DFC is a digital 3-way mono controller designed to be used in an HK AUDIO amplifier rack.
With its specially developed software and matching MIDI interface, it offers a comprehensive
means of managing a complex sound reinforcement system via an intuitive analog-style interface.
PA Remote Management
The distributed sound systems used at today?s large venues have become so complex over the
years that they now practically require a full-blown control center just to set up and manage them
efficiently and maintain a clear overview over them. To ensure that each individual acoustic area is
treated by a dedicated part of the PA, the various sections of the overall system must be divided into
sensible groups,such as PA wing left/right, center cluster, long-throw, near-field, sub-low, delay 1,
delay 2, and so on.
Such a division usually leaves you with a large number of different sound coverage zones, which have
to be treated separately but managed centrally.
A system controlled in the traditional manner would normally need at least one EQ, delay, and crossover
channel per coverage zone. Add to that the necessary standard cabling and rack and truck space,
and you end up with a considerable number of racks and, what?s more, a complicated set up. The
Digital Field Controller offers an easier and more cost-effective solution. The word "controller" usually
refers to functions such as crossover and protective circuitry; the DFC expands the meaning of the word to
include a very different meaning: PA-REMOTE-MANAGEMENT.
To simplify the handling of large system configurations featuring several amplifier racks, you can network
up to 32 DFCs via MIDI interface, controlling them remotely with the aid of a PC.
Using the control program that was developed especially for this very application, you can manage even
extremely large PA systems very conveniently, accessing all installed DFCs individually, if you wish, equalizing
and correcting delay values for each single device separately.
At the same time, you can choose to group several DFCs together in order to control an entire group of
speakers (PA wing, delay tower, subs) that are powered by several amp racks as one. All settings that you
enter for a given group are automatically transmitted to the appropriate controllers. This option for example lets
you effortlessly implement a delay station: All you have to do is assign all of the DFCs that control the speakers
of the delay station to one group, and set one single delay value, which is then automatically transmitted to all
DFCs in that group.
This feature considerably reduces the amount of cables you have to connect and simplifies the entire signal-processing
operation considerably. The group control function lets you comfortably handle PA systems of any size,
from club gigs to major open-air festivals.
The extensive control functions of the DFC system can be operated from the graphic user interface of the
PC software as conveniently and intuitively as you are accustomed to with individual analog equipment.
The graphic user interface of the PC software makes using the extensive control functions of the DFC system
as intuitive as using familiar analog equipment. In fact, you can manage your entire system from just three
screens.
The first screen displays the status of the peak and temperature limiters in the three frequency bands of every
attached DFC. So, the operational status of the complete system can be checked easily from either the FOH
position or the control room.
You can also create groups, and the zones they are intended to cover, in this window.
By double-clicking on the Group, you arrive at a second screen. Here you can assign a name to the group (e.g.
“Long-throw Left“). Additionally you can set basic functions here, such as Mute, Level, Phase Reverse and Delay,
for every one of the three frequency bands, as well as for the Group Master.
From the third screen you can access, for each connected controller and for each created group, a 28-band equalizer
with 15dB of cut or boost, Set-Flat and Bypass functions, and gain control.
Dynamics
The dynamic range of the input section plays a crucial role in a digital audio controller because
* the greater the dynamics of the input section, the better the headroom.
* the greater the dynamic range, the lower the audible noise.
* dynamic reserves in the input stage are the basis of excellent dynamics at the output stage
Through the innovative use of Split-Stack Technology, in which two 20-bit AD converters are stacked together, the
DFC achieves a dynamic range of 117 dB at the input! That exceeds what a high-quality 24-bit converter can do by
4 dB!
Absolute Correction of Frequency and Phase
In contrast to standard digital controllers, the DFC does not use IIR (Infinite Impulse Response) filters that are adjustable
at the unit itself. Instead, at each output it uses one FIR (Finite Impulse Response) filter, which accomplishes both
the dividing and equalizing functions in one.
The difference between the two types of controllers is essentially how the signal is processed in the controller. Most
other digital controllers have a fixed number of filter components which allow you to implement a low-pass filter with a
slope of 12 decibels, or a fully parametric filter. The desired type of signal processing operation is made possible by
a combination of these components. The processed signal is routed to the device's output.
For example, to separate the signal for a woofer with a slope of 48 dB, the device requires eight filter components (four
low and high-pass filters each). Optionally, you also can assign additional parametric filters to equalize the system.
The FIR filters of the DFC, on the other hand, work with a sample of the desired filter impulse response. The sample is
uploaded to the filter using Fourier transformation. It shapes the frequency and phase response of the filter. The sample
is downloaded via the same process. As a result, FIR filters can execute detailed equalization operations with extremely
steep slopes and freely assignable amounts and phases. This type of signal processing operation requires a PC. It
computes the required impulse response for the desired signal processing operation via an inverse Fourier transformation.
Once calculated, the filter can no longer be varied. If you want to change a parameter or value, the entire filter has to be
recomputed. On the other hand, you no longer have to execute a number of tasks manually. For example, you do not
have to adjust the delay and phase corrections for the split signals, since these operations are carried out automatically.
The DFC is not a universal controller, but is set up for a specific system and it knows its specific characteristics. This is
the only way a sound reinforcement system can really be equalized effectively.
Together with frequency response, the phase response of a system plays a very large role in how that loudspeaker
system reproduces sound. Phase errors are rooted in three main causes:
1. Each loudspeaker itself produces phase errors because different frequencies require different travel times
on the membrane.
2. Loudspeaker enclosures always function as high bandpass filters of the second order in the case of sealed
enclosures, and of the fourth order in the case of bass-reflex cabinets. All filters have the effect of making the
phase response less linear. This includes the filters of a passive crossover network.
3. In most systems, the voice coils of individual loudspeaker components are not lined up on one axis. As a
result, sound waves generated from these components are not in sync but out of phase, yielding distortion in
the phase response.
The DFC is the first controller which not only uses time alignment to compensate for any time lags between the
individual components, but goes a step further and compensates for the phase distortion arising from the enclosure,
the crossover networks and each individual loudspeaker.
So now, for the first time, a tool is available that uses a comprehensive approach for truly flattening the phase response
of an entire sound reinforcement system.
For the user this means a previously unattainable level of clarity and sonic purity, more effective use of equalization,
a more natural soundstage, and a greater tolerance among listeners for higher volume levels.
Biamped and 3-Way Active: Best of Both Worlds!
Both two-way and three-way systems have specific advantages and disadvantages. In combination with the two-way
systems of the HK AUDIO R-Series and T-Series, the Digital Field Controller enjoys all the typical benefits of both
two-way and three-way systems without having to suffer their disadvantages. As already mentioned, the DFC is dealing
virtually with three separate ways ? low, mid and high ? but recombines the mids and highs before the output stage.
In other words, the DFC produces a two-way signal, and yet each frequency band can be muted separately, and even
its phase can be reversed separately! In a mid/high loudspeaker module such as the RT 112 (12“+2“), you are free to
control the volume of the 12" speaker separately, reverse its phase, or even mute it briefly altogether without affecting
the high frequency driver at all. And that, mind you, although both are connected to the same speaker cable and both
are running over one and the same passive crossover.
* Comprehensive frequency adjustments
* Quick access to speaker components
* Individual speakers can be muted
* Corresponds to traditional international usage
* Easier to wire up
* Less power amplifiers are required
* Less electrical power is needed
* Less weight
* Less warehouse / truck space needed
* Fewer sources of error
* Higher Return on Investment
Perfect Protection for the System: the Limiters of the DFC
The HK AUDIO Digital Field Controller includes two anticipatory limiter sections per frequency band:
a) Thermal Limiter
Since the DFC has data on the entire sound system, power amps included, it can continually monitor the operating
state of its individual components. Taking into consideration how long the system has been running, it compares the
processed signal with the stored maximum system values, and momentarily reduces the output as needed. In this
way, the PA is protected against the dangers of overload.
b) Peak Limiter
The signal which has been processed by the DFC is checked with a peak detector before it reaches the output. If the
signal is too high, the peak limiter section is directly activated; it calculates how much to reduce the signal before the
peak signal reaches the output stage. In contrast to previous analog and digital controllers, the DFC does not use a
feedback circuit of the sort where the limiter only reacts after the first peak has already gone through. Instead, the signal
between the peak detector and the limiter is delayed by two milliseconds, allowing the needed corrections to have
already been made before the peak reaches the outputs of the DFC and, therefore, the P.A.
Delay Functions
The DFC features both a master delay and a frequency-band delay. The master delay serves to compensate for the time
lag between individual coverage zones, for example between the FOH PA and delay towers farther back in the audience.
Up to 2000 milliseconds of delay is available to sync loudspeakers that are up to 680 meters (2230 feet) apart from each
other, and that should be enough for most applications...
The frequency-band delay offers up to 100 milliseconds to align the individual components of a PA stack with each other,
for example if the mid/high cabs are flown but the bass cabs are resting on the floor. Both types of delay can be entered
in time (milliseconds) or distance (feet or meters).
| Overview
More sound to your amps and speakers
The secret to making speakers sound great lies with the crossover you use. With the dbx® 223xs Crossover you\'ll get great performance, ultra low-noise, rugged reliability and four decades of dbx knowledge and expertise in building the world\'s finest processors. The 223xs uses precision filters to separate the audio signal and direct the correct frequencies to your loudspeaker drivers. By directing only the specific frequency bands to each speaker driver the 223xs leaves your amplifiers free to use their full power on the usable signal eliminating distortion and giving your PA system a cleaner and better sound.
The dbx 223xs is a dual channel crossover with all the features you would expect from a professional product. It features Linkwitz-Riley 24dB per octave filters, independent output gain controls for level matching, output phase inversion, 40 Hz low cut filter, and optional mono summing of the low frequency (subwoofer) output. Everything in the design exudes great precision from the solid "click" controls to the high quality XLR inputs and outputs
Features
- XLR balanced ins and outs
- Mode switch for stereo 2-way or mono 3-way operation
- Low frequency summed (subwoofer) output
- x10 range switch on both channels
- 40Hz high pass (low cut) filter both channels
- Phase reverse switch on all outputs
- Individual level controls on all outputs
- 24dB per octave Linkwitz-Riley filters (the professional standard)
- Stereo/Mono status LEDs indicate the selected mode
- dbx® 2 year parts and labor as standard
- CSA NRTL/C approved
- CE compliant
Specifications
| Input Connectors |
XLR (pin 2 hot) |
| Input Type |
Electronically balanced/unbalanced, RF filtered |
| Input Impedance |
Balanced >50k ohm, unbalanced >25k ohm |
| Max Input |
>+22dBu balanced or unbalanced |
| CMRR |
>40dB, typically >55dB at 1kHz |
| Output Connectors |
XLR |
| Output Type |
Impedance-balanaced/unbalanced, RF filtered |
| Output Impedance |
Balanced 200 ohm, unbalanced 100 ohm |
| Max Output |
>+21dBu balanced/unbalanced into 2k ohms or greater |
| Bandwidth |
20Hz to 20kHz, +/-0.5dB |
| Frequency Response |
90 kHz, +0/-3 dB |
| Signal to Noise Ratio |
Ref: +4 dBu, 22 kHz measurement bandwidth |
| Signal-to-Noise (Stereo Mode) |
>94dB (Low Output), >91dB (High Output) |
| Signal-to-Noise (Mono Mode) |
>94dB (Low Output), >93dB (Mid Output), >91dB (High Output) |
| Dynamic Range |
> 106 dB, unweighted, any output |
| THD+Noise |
< 0.004% at +4 dBu, 1 kHz; < 0.04% at +20 dBu, 1 kHz |
| Interchannel Crosstalk |
< -80 dB, 20 Hz to 20 kHz |
| Crossover Frequencies (Stereo Mode) |
Low/High: 45 to 960 Hz or 450 Hz to 9.6 kHz (x10 setting) |
| Crossover Frequencies (Mono Mode) |
Low/Mid: 45 to 960 Hz or 450 Hz to 9.6 kHz (x10 setting); Mid/High: 45 to 960 Hz or 450 Hz to 9.6 kHz (x10 setting) |
| Crossover Filter Type |
Linkwitz-Riley, 24 dB/octave, state-variable |
| Low cut Switch |
Activates 40 Hz Butterworth, 12 dB/octave high-pass filter, one switch per channel. |
| Phase Invert Switch |
Inverts the phase at the output, one switch per output. |
| x1 / x10 Switch |
Multiplies the low-high(mono: low/mid and high/mid) crossover frequency range of the front-panel markings by a factor of 1 or 10, one switch per channel. |
| Mode Switch |
Selects stereo/mono and 2/3-way operation |
| LF Sum Switch |
Selects normal (stereo) or mono-summed low frequency operation and disables Ch 2ʼs low output phase invert LED to indicate that this output is not operational in the LF sum mode. |
| Indicators |
Stereo Operation (Green); Mono Operation (Yellow); Low Cut (Red-per channel); x10 (Green-per channel); Phase Invert (Red-per output) |
| Operating Voltage |
100 VAC 50/60 Hz; 120 VAC 60 Hz 230 VAC50 HZ |
| Operating Temperature |
32°F to 113°F (0°C to 45°C) |
| Power Consumption |
15W |
| Power Connector |
IEC receptacle |
| Dimensions |
1.75" H X 19" W X 6.9" D (4.4cm x 48.3cm x 17.5cm) |
| Unit Weight |
3.7 lbs. (1.7 kg) |
| Shipping Weight |
5.4 lbs. (2.5 kg) |
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The Soundweb London BLU-160 offers configurable I/O, configurable signal processing and a high bandwidth, fault tolerant digital audio bus.
The BLU-160 has open architecture which is fully configurable through HiQnet™ London Architect. A rich palette of processing and logic objects and a "drag and drop” method of configuration provide a simple and familiar design environment.
This processor features a low latency, fault tolerant digital audio bus of 256 channels which uses standard Category 5e cabling giving a distance of 100m between compatible devices. Fiber media converters can be used to increase the distance between devices to over 40km.
Four card slots which accommodate analog inputs, analog outputs, digital inputs and digital outputs in banks of four facilitate many different device I/O configurations.
Analog Input Cards provide software configurable gain in 6dB steps up to +48dB per channel and software selectable Phantom Power per channel. Digital Input Cards and Digital Output Cards process AES/EBU and/or S/PDIF audio and offer a variety of clocking and syncing options. (Further information about the I/O cards can be found on dedicated datasheets)
Phantom Power, Sync, Signal Present and Clip information per channel is easily accessible, without the requirement for a PC, from clear front panel LED indication. Device-specific information such as Device Name, Device Type, Firmware Version Number, Time, IP Address and Subnet Mask is available from the front panel display. A bi-directional locate function allows devices to be identified both from and within HiQnet London Architect.
12 Control Inputs and 6 Logic Outputs allow the BLU-160 to be integrated with GPIO compatible devices. The Soundweb London Interface Kit, comprehensive documentation which details how Soundweb London systems can be integrated with third party control systems, is included within the installation of HiQnet London Architect.
The BLU-160 and the other members of the Soundweb London family provide the building blocks of the perfectly tailored system solution.
Features - Four Input / Output Card Slots
- Configurable Inputs / Outputs
- Analog Inputs (with Phantom Power per Channel)
- Analog Outputs
- Digital Inputs (AES/EBU and S/PDIF)
- Digital Outputs (AES/EBU and S/PDIF)
- Configurable Signal Processing
- Rich Palette of Processing and Logic Objects
- 256 Channel, Low Latency, Fault Tolerant Digital Audio Bus
- Clear Front Panel LED Indication
- Informative Front Panel Display
- Bi-Directional Locate Functionality
- 12 Control Inputs and 6 Logic Outputs for GPIO Integration
- Soundweb London Interface Kit for Third Party Control System Integration (Documentation)
- HiQnet Device
- Configuration, Control and Monitoring from HiQnet London Architect
| Behringer CX-3400 3-Way Stero/4-Way Mono Frequency Crossover with Limiters
The CX3400 is an active frequency crossover enabling 2 or 3-way stereo or even 4-way mono operation. Each band features two 24-dB Linkwitz-Riley filters, as well as individual IGC limiters, while the integrated delay enables flexible time alignment. Polarity switches for each output allow quick and easy runtime compensation for various driver and horn types.2
Respectively:
3-Way stereo or 4-way mono
24 dB / octave Linkwitz-Riley filter
Mono-Sub switchable
LF / HF gain regular / mutable
CD horn EQ switchable
Vertex frequency tunable
Factor switching (10x)
Multi-band limiter switch / variable (threshold)
Limit display each way
LF Delay (max 2ms / 60cm)
Phase reverse
25 Hz subsonic filter, sym
XLR in / out with multiband limiters
Dimensions: 19", 1U, 147 mm case depth
Weight: 1.7 kg | Behringer DCX2496 Overview
The DCX2496 ULTRADRIVE PRO from Behringer is a digital loudspeaker management system, that provides excellent audio quality, along with a comprehensive feature list. It allows for precise set-up of multiple loudspeaker arrays, suitable for live sound, fixed installations that require control of loudspeaker systems, and other contractor-installed applications.
Each of the three-inputs and six-outputs have fixed processing blocks, including gain, delay, EQ and dynamic EQ on the input channels, and gain, crossover, EQs, limiter and delays on the output side. One of the inputs can accept a stereo AES/EBU digital signal. An integrated sample rate converter assists in accepting various sample rates. Additionally, an internal input-sum signal is easily set-up.
Four different mono and stereo output operating modes are available, all with individual crossover filter types with selectable roll-off characteristics. Input and output delays allow for time adjusting signals to compensate for ambient temperature changes or for adjusting arrival time differences for individual speaker clusters. EQs can be used for both musical enhancement or for level-dependent equalization. Brick-wall limiters guarantee speaker protection.
Remote control of one or several units can be accomplished utilizing downloadable software for a Windows based computer. Set-ups can be saved to the software, or to an optional PCMCIA card for the ultimate in portability. Servo-balanced, gold-plated XLR connectors for all inputs and outputs guarantee excellent connectivity.
- Three-input, six-output loudspeaker management system
- Four different mono- and stereo-output operating modes
- Individual crossover filter types including Butterworth, Bessel and Linkwitz-Riley, with selectable roll-off characteristics from 6dB to 48dB per octave
- Choice of high-, band- and low-pass filters for each input and output
- Dynamic EQ for level-dependent equalization and parametric EQ selectable for each channel
- “Zero" attack limiters on all output channels for optimal signal protection
- Automatic or manual-adjustable delays for all inputs and outputs allow for adjustment due to room temperature, phase and arrival time differences
- Link option via RS-485 network interface allows for linking multiple units for larger installation requirements
- Downloadable software enables single- or multiple-unit remote control utilizing a Windows based computer
- Sixty user-defined internal presets can bestored via software or on a PCMCIA card
- Internal universal power supply allows the unit to operate virtually anywhere
| Frequency Response |
10Hz to 35kHz, -1dB |
| Dynamic Range |
109dB |
| Input Connections |
3 x XLR |
| Output Connections |
6 x XLR |
| Input Impedance |
20kΩ @ 1kHz |
| Output Impedance |
160Ω @ 1kHz |
| Maximum Input Level |
+22dBu |
| Maximum Output Level |
+22dBu |
| Sample Rate |
32kHz to 96kHz |
| RTA Mic Phantom Voltage |
+15V |
| Crosstalk (Main Outputs) |
-100dB @ 10dBu In |
| THD & Noise |
0.007% @ 0dBu In |
| Dimensions |
19" x 8.5 x 1.75" (483 x 217 x 45mm) |
| Weight |
6.6 lbs (3 kg) |
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With the DSP 260 digital 2-in-6 sound system manager, DYNACORD is continuing its proud tradition in the fi eld of innovative digital signal processors.
Based on the most modern hardware, this system manager offers tried-and-tested as well as new algorithms for the simple and swift construction of active multi-way systems. As much importance has been attached here by the developers to high user-friendliness as to the realization of an extremely favorable price.
24-bit sigma-delta AD/DA converters and a 32-bit floating-point signal processor make possible a dynamic range of 111 dB. In addition to the two analog inputs, a digital stereo input in AES/EBU format is available. A -6 dB pad switch in front of the AD converters offers additional security against overload when the device is connected to mixing desks with very high output levels. The 6 individually mutable outputs are electronically balanced and available on XLR sockets. For all inputs and outputs, easily read LED level meters are provided on the front panel.
Users can choose whether to configure their own systems at the device itself using the controls and display provided or else by means of the intuitive editing software.
3-way stereo, 2-way stereo + FR, 4-way mono + FR etc. Furthermore, factory presets compatible with those of the tried-and-tested DSP-244 are available for Dynacord loudspeaker systems.
In terms of DSP, the device offers parametric and graphic EQs as well as a delay function in the input and crossover, channel EQ, channel delay and a level control with limiter in each output channel.
An RS-232 interface allows Master/Slave operation of multiple devices as well as GPI switching of settings. A PC (the editing software runs under Windows) can be connected via the USB interface on the front panel.
A variable parameter lockout function allows you to determine which settings and parameters can be accessed directly at the device. This is just one of many features that make the DSP 260 an ideal choice also for the rental business.
The wealth of features combined with its outstanding audio performance and favorable price commend this audio system manager for a multitude of professional applications.
General Mains Voltage 100-240 VAC 50-60 Hz Power Consumption 25W Audio Analog Inputs 2x XLR IN, electronically balanced, 2x XLR THRU OUT, electronically balanced Digital Inputs 1x XLR AES/EBU IN Nominal Input Voltage 1.23 V / +4 dBu Maximum Input Voltage (Without -6dB Analog Pad Engaged) 8.7 V / +21 dBu Input Impedance 10k ohm Common Mode Rejection -80 dB @ 1 kHz (typical) A/D Conversion 24-Bit Sigma-Delta Outputs 6x XLR OUT, electronically balanced Nominal Output Voltage 1.23 V / +4 dBu Maximum Output Voltage 8.7 V / +21 dBu Output Impedance 50 ohm D/A Conversion 24-Bit Sigma-Delta Frequency Response 10 Hz-22 kHz (+/- 0.5 dB) THD+N < 0.01% (band limited 22Hz-22kHz) Dynamic Range 111 dB unweighted, band limited 22 Hz - 22 kHz Interfaces USB USB Type B on front panel (PC Interface) 9-pin DSUB Software Confi gurable for GPI Preset Recall or Master/Slave Connection to second DSP 260 unit Signal Processing Sample Rate 48 kHz Data Format 24-Bit Internal Processing 32-Bit Floating Point Physical Dimensions (WxHx) 19 x 14 x 1.75 inches (482.6 x 355.6 x 44.45 mm) Weight (Net) 10.1 lb (4.6 kg) Weight (Gross) 13.0 lb (5.9 kg)
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