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The LD DS2.1 is a universal 3-channel DSP controller for 2.1 loudspeaker systems with 24-bit processing and convenient front panel operation providing easy setup and high quality audio performance.
Designed for live sound and fixed applications the rack mountable unit offers extensive parameter control of its EQ, crossover, delay and limiter functions. The LD DS2.1 features a 3-band parametric EQ with adjustable Q and high shelf / low shelf selection. The crossover includes Butterworth, Bessel and Linkwitz-Riley filters with 6 dB to 24 dB slopes. Delay time is adjustable to a maximum of 12.5 ms for each of the three outputs and the limiter provides independent threshold and hold time setting for the L / R and subwoofer channels.
A backlit 2 x 16 LCD and jog wheel allow for quick parameter adjustment and recall of up to 6 presets. Also provides real time level meter. The rear panel features XLR connectors for left and right input, left and right plus sub XLR outputs and a USB port. Software and a power cord is included.
Product type
Signal Processors
Type
3-channel
Version
3-channel
AD/DA converter
24 bit
Sampling frequency
48 kHz
Frequency response
20 - 20.000 Hz
Filter type
Bessel , Butterworth , Linkwitz-Riley
Delay
max. 12.5 mS
Filter slew rate
12 dB/oct , 18 dB/oct , 24 dB/oct , 6 dB/oct
Dynamic range
> 100 dB
Signal-to-noise ratio
87 dB
THD
0.005 %
Crosstalk
> 80 dB
Max. input level
+ 13 dBu
Max. output level
+ 13 dBu
Input impedance (kOhms)
22 kOhm
Output impedance
300 Ohm(s)
Controls
Edit , Parameter , Power , Recall
Indicators
multifunction LC display
Line inputs
2
Line input connectors
XLR
Line outputs
3
Line output connectors
XLR (balanced)
Interfaces
USB
Mains connector
IEC socket
Operating voltage
90 V AC - 250 V AC, 50 - 60 Hz
Width
482 mm
Height
44.5 mm
Depth
160 mm
Weight
1,8 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)
| Advanced Feedback Suppression Processor with Full LCD Display
The AFS2 Dual-Channel Advanced Feedback Suppression Processor from DBX protects your ears and audio equipment from annoying and potentially damaging audio feedback without altering your sound. The AFS2 features application-specific filter types for speech, music low, music medium, and music high. The 24 fixed or live narrow-band notch filters are capable of removing frequencies at 1/80 of an octave, ensuring an optimal frequency response without the possibility of feedback.
The front panel features the wizard function to take the guesswork out of setting up any room by incorporating an advanced feedback suppression module and providing a full LCD display for system settings and menus. Additionally, the AFS2 offers a five-segment LED for accurately monitoring input levels for each channel, while the notch filter indicator allows you to monitor the number of filters applied. Dedicated bypass buttons are included to help isolate problem channels. The rear panel offers XLR and 1/4" inputs and outputs, both switchable between +4 and -10 dB operations. A USB port has been provided for firmware updates. The AFS2 ships with a standard IEC power cable.
Number of Channels |
2 |
Inputs |
Connectors: Female XLR and 1/4" TRS
Type: Electronically balanced and unbalanced, RF filtered
Impedance: Balanced 50 kohm, unbalanced 25 kohm
Max Line Level: +20 dBu |
Input CMRR |
>40 dB, typically >55dB @ 1 kHz |
Outputs |
Connectors: Male XLR and 1/4" TRS
Type: Electronically balanced and unbalanced, RF filtered
Impedance: Balanced >120 ohms, unbalanced >60 ohms
Max Level: +20 dBu |
A/D/A Converter |
A/D Converter: DBX type IV conversion system
A/D Dynamic Range: >113 dB A-weighted, >110 dB, unweighted, 22 kHz BW
Type IV Dynamic Range: >119 dB, A-weighted, 22 kHz BW; >117 dB, unweighted, 22 kHz BW
A/D Conversion: 24-bit
Sample Rate: 48 kHz
D/A Dynamic Range: 112 dB A-weighted, 109 dB unweighted
D/A Conversion: 24-bit |
Dynamic Range |
109 dB A-weighted, 106 dB unweighted, 22 kHz BW |
THD + Noise |
0.003% typical at +4 dBu, 1 kHz |
Frequency Response |
20 Hz to 20 kHz, ±0.5dB |
Crosstalk |
Interchannel Crosstalk: >80 dB typical
Crosstalk Input to Output: >80 dB |
Power |
US: 100 to 120 VAC 60 Hz
EU: 220 to 240 VAC 50 Hz
9 W |
Dimensions (HxDxW) |
1.8 x 5.8 x 19.0" / 4.4 x 14.6 x 48.3 cm |
Weight |
4.5 lb / 2.0 kg |
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Based on a two-in, six-out topology, the DC-One is designed primarily for users of small-to-medium sized sound systems in both mobile and installed applications. While the unit is an all-new development based on a powerful SHARC processor, it is 100 percent compatible with all settings from other Electro-Voice signal processors. A large part of DC-Oneʼs development was aimed at ensuring that users can easily handle its serious digital horsepower and high-end audio performance. Free. PC-Editor software is available, or the DC-One can be operated via the front panel.
A/D Conversion 24-Bit Delta-Sigma A/D Conversion 24-bit/sigma-delta (linear phase) 128 times oversampling Analog Inputs 2 XLR (Electronically Balanced), 2 XLR THRU OUT (Electronically Balanced) Analog Outputs 6 XLR (Electronically Balanced) Control Protocol Front Panel USB Connector D/A Conversion 24-bit/sigma-delta 128 times oversampling D/A Conversion 24-Bit Delta-Sigma Data Format 24-Bit Digital Inputs 1 XLR AES/EBU (2 Ch) Dynamic Range 111 dB (unweighted, band limited 22 Hz - 22 kHz) Electronics Type Processor FIR-Drive No Frequency Response 10 Hz - 22 kHz (±0.5 dB) Input Impedance (Balanced) 10 kΩ Internal Processing 32-Bit Floating Point Mains Voltage 100-240 VAC Maximum Input Voltage 8.7 V / +21 dBu (Without -6 dB Analog Pad Engaged) Maximum Output Voltage 8.7 V / +21 dBu Nominal Input Voltage 1.23 V / +4 dBu Nominal Output Voltage 1.23 V / +4 dBu Output Impedance (Balanced) 50 Ω Power Consumption 25 W Sample Rate 48 kHz THD+N < 0.01% (band limited 22 Hz - 22 kHz) Height 1RU 44.45 mm (1.75“) Width 482.6 mm (19“) Depth 355.6 mm (14“) Weight Net 4.6 kg (10.14 lbs) |
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
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Full-featured Speaker Processing in a 1U Space
Speaker processing can be a complicated business requiring a substantial array of equipment, but the Yamaha SP2060 offers everything you need in a single rack space. This innovative 24-bit, 96-kHz digital speaker processor delivers excellent sound quality, an impressive variety of processing functions — gain, delay, PEQ, comp, crossover, limiter, and an all-pass filter for phase adjustment — and intuitive programming from Yamaha's DME Designer application running on a personal computer. It has two analog inputs and six analog outputs, plus two AES/EBU format digital inputs for connectivity with a broad range of systems. And if you use Yamaha Installation Series Speakers, all you need to do is select one of the many optimized presets provided for great sound with minimum measurement and setup time. The SP2060 is a compact, portable 1U unit that is ideal for live sound or installations.
Sampling Frequency | Internal Clock | 96kHz | External Clock | Normal Rate | 44.1, 48kHz (±0.1%) | Double Rate | 88.2, 96kHz (±0.1%) | Signal Delay | 761µsec Input to Output, fs = 96kHz | Frequency Response | 20Hz -40kHz (TYP 0dB, MAX +0.5dB, MIN -1.0dB), fs=96kHz, RL=600ohms | Total Harmonic Distortion | 0.007% (+22dBu@1 kHz),
0.05% (+4dBµ@20 Hz - 40kHz)
fs=96kHz, RL=600ohms;
measured with 18dB/octave filter @80kHz | Hum & Noise | TYP -82dBu,
fs=96kHz, RL=600ohms, Rs=150 ohms;
measured with 6dB/octave filter @12.7kHz;
equivalent to a 20kHz filter with ∞ dB/octave attenuation. | Dynamic Range | 106dB AD+DA
fs=96kHz, RL=600ohms;
measured with 6dB/octave filter @12.7kHz;
equivalent to a 20kHz filter with ∞dB/octave attenuation. | Crosstalk @1kHz | -80dB INPUT to Output
fs=96kHz, measured with 18dB/octave filter @80kHz | Power Consumption | 30W | Dimensions (W x H x D) | 480 x 44x 360.2mm | Net Weight | 4.2kg |
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