DME64N_datasheet

Main Features of the New Version 3
This major upgrade significantly enhances the functionality and versatility of the DME series product line the DME64N and DME24N and the DME Satellite models: DME8i-C, DME8o-C, DME4io-C, DME8o-ES, DME8i-ES, and DME4io – which allows advanced audio systems to be designed and implemented via a comprehensive computer interface. The DME Designer software application that provides the programming interface is also being upgraded to Version 3.

Enhanced Automatic Adjustment for Meeting and Conference Rooms
An Auto Gain Control component that automatically compensates for variations in the level of a speakers voice, and an Auto Mixer II component that provides an over-ride function and automatic output level control make it possible to create intelligent systems that provide automated control for applications that dont require the presence of an experienced operator.

Improved Zoning Capabilities
Room Combiner
Room Combiner
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Version 3 includes a Room Combiner component that greatly facilitates the combining and dividing of rooms in conference venues for example, as well as the control of zoning in undivided spaces at shopping centers or theme restaurants. Additional Router and Matrix component variations have also been provided for increasing further the capability and efficiency of system design.

Ambient Noise Compensation
Ambient Noise Compensator
Ambient Noise Compensator
(Click to enlarge)
The new Ambient Noise Compensator component can be used to measure ambient noise levels and automatically and dynamically adjust the program level for optimum balance and intelligibility. This function can be useful, for example, to optimize the level of announcements at shopping malls at which the ambient noise level may vary greatly depending on the time of day or day of the week.

Automatic Audio Input Detection
Audio Detector
Audio Detector
(Click to enlarge)
The Audio Detector component generates GPI output based on whether or not an input audio signal is detected. This capability could be used, for example, to automatically pan a camera to the current speaker during a conference, control lighting, or perform other advanced functions. Conversely, it could be used to detect when an audio input has stopped and change scenes accordingly.

Simple Mixer
A simple but very useful stereo mixer available in 16-input and 24-input variations, both with dual auxiliary sends. This component can significantly enhance system design flexibility in a wide range of applications.

DME64N_datasheet
dme64nv1_es_om

Hardware Features

DSP power that rivals the Yamaha DM1000 digital mixer and expands the boundaries of configuration design

The DME64N and DME24N offer such powerful performance complements of Yamaha’s proprietary DSP7 and DSP6 LSIs, which were designed exclusively for audio processing. The DME64N features processing power to rival Yamahas highly acclaimed DM1000 digital mixer, while the DME24N offers about half that.

This awesome DSP power offers two big advantages. First, it gives a single DSP box enough muscle to build larger, more complex DSP configurations, giving audio designers a free hand in fine-tuning audio signals for systems in concert halls and a wide range of other venues. And second, it makes configuration design easier than ever – because configurations that once required multiple DSP boxes can now be run with one DME64N or DME24N. Which means big time and energy savings in both design and actual installation, making these Yamaha DMEs affordable solutions as well.

Sound-focused circuitry design with 24 bit/96kHz processing support for the ultimate in sound quality

Highly recommended by leading professionals for live performances and studio recording, Yamaha digital mixing engines inherit our cutting-edge audio technologies in mixing console development – including power supply unit placement and proper grounding. And the results speak for themselves. No wonder the DME64N and DME24N can deliver peerless fidelity in reproducing any original sound – to the point of rivalling a large mixing console. The DME24N, for instance, is redefining the boundaries of audio fidelity at 24bit/ 96KHz operation with newly developed head amplifiers onboard. Just one more reason the best sound in the world comes from Yamaha.
Note: Configurations that can be created at 96kHz operation will be half the size of those that can be
created at 48kHz operation.

Exceptional I/O flexibility and expandability. Cascade up to 8 DME64Ns for 512 input/512 output capability

The DME64N features 4 mini–YGDAI slots on its rear panel for optional I/O cards. The DME24N features one slot. These slots support analog inputs and outputs when using high–precision A/D and D/A cards with up to 8 I/Os, as well as AES/EBU–, ADAT–, and TASCAM–compliant digital I/O cards with 16 I/Os. Thanks to these YGDAI slots, the DME64N and DME24N offer additional support for I/Os over a wide range of audio formats.

As a result, the DME64N can support up to 64 input and 64 output channels with 4 MY card slots. The DME24N accommodates up to 24 x 24 – with 8 analog inputs and 8 analog outputs onboard, plus one MY card slot. In addition, up to 8 DME64N devices can be cascaded to support large–scale applications, providing a maximum of 512 inputs/512 outputs with digital I/O MY cards and 128 inputs/128 outputs with analog I/O MY-cards.

DME24N rear panel features 8 analog input and 8 analog output terminals

The DME24N’s rear panel features 8 mic/line analog inputs and 8 output Euroblock terminals. Each terminal offers built-in high-precision 24 bit/96 kHz A/D– and D/A–converters, which in practice, not only makes installation fast and easy, but translates into some of the best cost performance in the business.

MY16-C mini-YGDAI I/O card ensures CobraNet™ compatibility

With the addition of the new 16-I/O MY16-C card that supports the CobraNet™ advanced digital audio network, Yamaha’s mini-YGDAI card portfolio now offers a broader range of digital format I/O cards. These cards enable network construction with other CobraNet™-compliant products – including the NHB32C network hub and bridge and the ACU16C amp control unit. The DME64N can support up to 2 MY 16-C cards. The DME24N, one.

Control Network via Ethernet

Up to 16 DME64N, DME24N and ICP1 intelligent control panel units can be connected via their RJ45 connectors to form an Ethernet network. Making use of existing 100base-T CAT 5 cables and other Ethernet infrastructure components enables quick, easy, cost-effective system construction.

Comprehensive GPI, RS-232C, USB and MIDI control signal support

The DME64N and DME24N offer a wide range of control interfaces for use with different units. The DME64N has 16 in and 16 out GPI ports to link with installed sound systems and other equipment. The DME24N has 8 I/O. What’s more, both include RS232C / RS422 serial ports for interfacing with remote control units and PCs, USB ports for popular PC interfaces and MIDI IN/OUT interfaces for synchronization with electronic musical instruments, sequencers and stage lighting controllers.

Jumbo LCD display with extensive front panel controls

Both DME64N and DME24N offer outstanding ease of operation thanks to their large, easy-to-read backlit LCD displays and comprehensive clusters of front panel controls – LED status indicators, jog wheel and SCENE, Home and Utility keys. A monitoring headphone jack with phone level control is standard. The DME24N also provides SIGNAL and PEAK LED indicators on the front panel for 8 inputs and outputs.

Scene and function names in 5 languages

The DME64N, DME24N and ICP1 intelligent control panel can display names for scenes and function keys in 5 languages – Japanese, English, French, German and Spanish. And all 3 units feature a user-friendly interface for trouble-free operation regardless of level of skill or experience.

SP2060_datasheet

Main Features

* Digital speaker processor with a variety of built-in features and highly accurate speaker processing capability

The SP2060 packs all of the features necessary for speaker processing into a compact 1U-size chassis to fully and flexibly meet a variety of speaker processing demands. Our exclusive LSI features the specialized sound processing expected from Yamaha for even higher accuracy in digital signal processing. Many built-in functions play major roles in speaker processing, including gain, delay, PEQ, comp, crossover, and limiter. The highly accurate All-Pass Filter can minutely control just the phase without changing the gain.

* Equipped with two analog inputs and six analog outputs, plus two AES/EBU format digital inputs

The rear panel is outfitted with two analog inputs and six analog outputs. Plus, two AES/EBU format digital audio inputs are included, allowing direct input from digital devices. It accepts up to 96kHz digital audio signals without the need for additional hardware, providing great compatibility with digital mixers and other devices. All terminals use XLR type connectors.

* Provides excellent sound quality that ranks with the most-advanced Yamaha Digital Mixers, with 24bit/96kHz high resolution

The internal workings utilize audio processing LSI that boasts a superb processing powered mixer, and provides the fantastic sound quality of a tremendous dynamic range with an SN ratio of over 110dB. Also, for the inputs, outputs, internal circuitry, and other aspects of the analog audio stage, we have included technology achieved through our experience in high-end analog mixing console design, such as the professionally lauded PM5000, to maximize sound quality. The SP2060 further contributes to the Yamaha Professional Audio name by providing top-notch sound.

* Capable of detailed programming with DME Designer PC application software

SP2060 DME Designer:
With the SP2060, detailed programming on a PC is enabled with the DME Designer software used for programming Yamaha’s top-of-the-line digital audio devices, the Digital Mixing Engine DME64N and DME24N, which can be used for a wide variety of applications. By loading data programmed with DME Designer into the main unit, detailed settings can be assigned instantly with a computer, enabling more efficient editing. Also, the method of data processing can be changed even more flexibly to meet the user’s needs. This allows programming of large-scale sound systems that use multiple devices such as SP2060, DME64N, and DME24N, and provides the advantage of being able to integrate and collectively manage outputs.

* User-friendly design includes Ethernet port enabling direct connection to a computer and an easy-to-operate front panel

SP2060 is equipped with an Ethernet port, which enables direct connection with a PC and allows users to directly load data created with DME Designer. Using wireless LAN, programs and parameters can be set while listening to the sound as heard from the audience’s receiving points with a tablet computer or other device. Plus, the main unit’s front panel features six output LED meters for excellent operability and visibility, making it user-friendly even if not used in conjunction with a PC.

*Optimized for Yamaha Installation Series Speakers, guaranteeing speaker processing with tremendous accuracy

Data for Yamaha Installation Series Speakers, with a great lineup variety and top-notch sound quality, comes preset in the SP2060. This allows highly accurate settings with just a few easy steps, providing the best speaker processing with the least amount of effort.

*Compact, portable 1U size for smooth setup to meet a variety of needs, from live PA to installation use

The compact 1U size of the SP2060 gives it tremendous installability and portability as well as superior cost performance. It is ideal for a wide range of applications, from installation in halls and theaters to live sound reinforcement. The bold dark blue color of the front panel also offers a new level of class for professional audio devices.

Based on a 2-in, 6-out topology, the DC-One is a rackmountable loudspeaker controller from Electro-Voice designed primarily for users of small-to-medium sized sound systems in both mobile and installed applications. Based on a powerful SHARC processor, it is 100% compatible with all settings from other Electro-Voice signal processors. Includes software for optional control via a PC.

A large part of EV DC-One’s development was aimed at ensuring that users can easily handle its serious digital horsepower and high-end audio performance. Though free PC-Editor software is available, the DC-One speaker controller can be easily operated via the front panel. Direct access buttons, 6 predefined configuration modes (1 being free configuration), straightforward output delay settings, and true professional metering all allow for safe operation.

The DC-One’s unique front-panel access control allows the user to select a personal library of accessible presets (out of 60 factory and 20 user presets). In addition, the user can determine which individual parameters in each preset are locked (e.g. factory predefined output parameters) or editable (e.g. master EQ or delay settings).

The EV DC-One has a 19″ chassis that is deep enough to match typical amplifier depths, a front-panel USB connector for the PC-Editor, and accepts either analog or digital (AES-EBU) input signals; a 6-dB input pad is available. The RS232 on the rear also provides 8 contact closures for preset changes or allows 2 units to be linked for larger systems. The PC-Editor software, as well as firmware updates allowing additional functions, will be available on Electro-Voice’s website.

Electro-Voice DC-One Loudspeaker Controller Features:

Ideal rack controller for small-to-medium sound systems
Compatible with other Electro-Voice signal processors
Easily operable via front panel or through free PC-Editor software
Direct access buttons
Six predefined configuration modes
60 factory, 20 user presets

Electro-Voice DC-One Loudspeaker Controller Specifications:

• Analog Inputs: 2 XLR (Electronically Balanced), 2 XLR THRU OUT (Electronically Balanced)
• Digital Inputs: 1 XLR AES/EBU (2 Ch.)
• Analog Outputs: 6 XLR (Electronically Balanced)
• Nominal Input Voltage: 1.23V / +4dBu
• Maximum Input Voltage: 8.7V / +21dBu (Without -6dB Analog Pad Engaged)
• Nominal Output Voltage: 1.23V / +4dBu
• Maximum Output Voltage: 8.7V / +21dBu
• Input Impedance (balanced): 10kOhm
• Output Impedance (balanced): 50 Ohms
• A/D Conversion: 24-Bit Delta-Sigma
• D/A Conversion: 24-Bit Delta-Sigma
• Sample Rate: 48kHz
• Data Format: 24-Bit
• Internal Processing: 32-Bit Floating Point
• Frequency Response: 10Hz–22kHz (± 0.5 dB)
• Dynamic Range: 111 dB (unweighted, band limited 22Hz–22kHz)
• THD+N: < 0.01% (band limited 22Hz–22kHz)
• Control Protocol: Front Panel USB Connector
• Mains Voltage: 100-240 VAC
• Power Consumption: 25W
• Electronics Type: Processor
• Weight: 11 lb.
• Dimensions: 19″ W x 1-3/4″ H x 14″ D

GENERAL DESCRIPTION
The Samson S•3-Way is a one-space, dual channel electronic crossover capable of managing the frequency control for stereo 2-way, stereo 3-way, 4-way mono and 4-way mono low speaker systems. The S•3-Way is an ideal crossover solution for small and large PA systems, live sound venues, commercial installations, recording studio monitors and DJ setups.
The input gain features +/-12 dB range with LED metering and peak indicators. The S•3-Way features low, mid and high frequency outputs with +/-6 dB of gain control.  Each output incorporates a mute switch for monitoring the individual frequency bands and a phase switch to invert the phase of the output.

The first crossover point is sweepable in three ranges: from 35 Hz to 800 Hz, from 350 Hz to 8 kHz with the 10X-multiplier switch engaged, or from 16 Hz to 400 Hz in 4-way low mode.

The second crossover point is sweepable from 175 Hz to 4 kHz or from 350 Hz to 8 kHz, depending on the setting of the mode switch.
In addition to its flexibility in configuring to different sound systems, the S•3-Way has advanced features such as CD horn equalization, dedicated limiter, high-pass filter and a delay section with up to 2ms of delay to time-align low frequency outputs for improved phase response of any PA system.

The S•3-Way is a complete electronic crossover solution, which will give you precise control and superior sound from your loudspeaker system.
FEATURES
> Full-featured, stereo 3-way, mono 4-way
electronic crossover
> Phase switches invert the phase of individual outputs
> Adjustable low frequency delay for time aligning
speaker stacks
> Mode Indicator LEDs provide quick visual display
of operating mode
> 4-segment LED input meter allows easy gain adjustment
> Precision Linkwitz-Riley, 24 dB/octave filters
> Output mute switches for all individual outputs
> Constant Directivity EQ smoothes the response
of CD horns above 3.5 kHz
> Peak Limiter with independent threshold control
and peak LED.

The benefits of multi-band processing, with the ability to apply different parameters to different sections of the audio bandwidth have been apparent to sound engineers for quite some time.
However, ‘locking in’ your audio to any one specific multi-band unit can vastly reduce the creative options. The desired combination of processing to deal with low frequency energy, the vocal mid range and high frequency detail and enhancement is not necessarily available from any one manufacturer.
The new Drawmer THREE-SUM opens up a whole new set of options allowing the engineer to split the audio into two or three bands and apply his or her own sonic signature to each part of the audio bandwidth.
The THREE-SUM employs a high quality signal path culminating in a variable threshold, brick wall limiter section with bypass facility. To ensure transparency the limiter design is ‘two stage’, applying different processing to the H.F. content of the material. In applications where the dynamics of the material need to be retained to create an open sound, the limiter is essential for catching peaks.
The THREE-SUM has been designed for use in high-end mastering and general recording applications, alternatively, it can transform a project studio, equipped only with ‘single band’ processors, into a serious multi-band facility.
Main Feactures:
• Continuously Variable Band Splitting
• Switchable Mute & Bypass On Each Band
• Variable Threshold, Two Stage Brick Wall Limiter Section
• Two Or Three Band Operation
• VU Metering With +10dB Re-Scale Mode For Hot Output Levels
• Variable Input & Output Levels
• Individual Band Or Multi-Band Processing
• High Quality, Minimal Signal Path With Balanced XLR Inputs/Outputs

Se trata de una aplicación que corre bajo entorno Windows PC, y es la herramienta del instalador para convertir un equipo en una solución integral personalizada que se ajusta exactamente a las necesidades del cliente o usuario final. Controla las unidades iDR-8, iDR-4, las expansiones de entrad o salida, los conmutadores de red iDR y los paneles murales y otros dispositivos de control remoto. Presenta un aspecto muy intuitivo, de estilo “analógico” que facilita la familiarización de manera muy rápida. Una vez realizado el diseño, éste es transferido a la unidad central iDR, pudiendo desconectar el ordenador personal desde ese instante, y funcionando éste como sistema autónomo.

* Comunicaciones:
* Red TCP/IP: Seguramente el modo más potente en la actualidad de controlar cualquier sistema. Conector RJ45 estándar con conmutador MDI/X para seleccionar la conexión directa a un ordenador personal o a una red mediante un conmutador de red. Pueden conectarse varias unidades iDR a la red, cada una con una dirección e identificador únicos. Esto es de especial utilidad en sistemas en red de gran tamaño: el instalador puede acceder y configurar cualquier parámetro desde el software System Manager y el usuario puede controlar localmente el sistema desde paneles virtuales basados en clientes del software PL.
* RS232: Una alternativa a la conexión en red para conectar un ordenador personal desde el panel frontal (iDR-8) o trasero, de manera directa o vía módem.
* SysNet: Otro puerto RS232 sujeto al protocolo SysNet de Allen & Heath, para la comunicación con controles de diversos fabricantes, como los paneles táctiles de AMX o Crestron.
* MIDI (sólo iDR-8): Conector MIDI de 3 contactos opto aislado, sujeto al protocolo SysNet, para el control de parámetros mediante el conocido estándar MIDI. De gran utilidad en aplicaciones de directo y teatro, en donde sonido, iluminación y otros tipos de operaciones técnicas son habitualmente automatizadas.
* DR-Link: Sistema de lógica de control de Allen & Heath para la comunicación con las expansiones iDR y los conmutadores de red iDR. Pude ser encadenado, por ejemplo, para la conexión de varios conmutadores de red iDR en serie.
* PL-Anet: Red serie RS485 para la conexión de multitud de controladores encadenados mediante cable CAT5, sobre distancias de hasta 300 metros. Los controladores se alimentan remotamente mediante la conexión PL-Anet.

Solid State Logic XLogic Alpha Link MADI AX

a gama XLogic Alpha-Link Audio I/O es una nueva familia de conversores multicanal de audio para estudio, de altas prestaciones, diseñados para su uso en aplicaciones de estudio, directo y radiodifusión. Está formada por tres modelos diferentes, cada uno de los cuales ofrece circuitos de conversión AD/DA de 24 canales, de alta calidad, que ofrecen la posibilidad de seleccionar entre diversos formatos de audio digital.
Los conversores de la serie Xlogic Alpha-Link poseen un diseño estilizado, y se presentan en un formato de rack estándar (2U), con un panel de control con una disposición clara de los diferentes controles. Cada una de ellas ofrece una extensa matriz de direccionamiento que se utiliza para realizar la configuración global entre las diversas conexiones de entrada y salida, que hace posible todas las combinaciones. En su panel frontal también se dispone de una conexión accesible para auriculares y una sección de medición para las entradas y salidas analógicas, así como un pulsador de selección AD/DA, LEDs indicadores de modo y 24 LEDs tricolores para indicación de nivel.
Todas las unidades Alpha-Link pueden utilizarse como conversores de formato autónomos, aunque si se utilizan en combinación con la tarjeta Soundscape Mixpander proporcionan una solución altamente flexible para las entradas / salidas en estaciones de trabajo de audio basadas en PC. Cuando la unidad es conectada a Mixpander, es posible dirigir de forma independiente todas las entradas y salidas desde el PC utilizando el software Soundscape Mixer.
XLogic Alpha-Link MADI AX proporciona una solución fiable, versátil y de alta calidad para la conversión MADI a ADAT Lightpipe. Ofrece hasta 64 canales de entradas / salidas digitales MADI a 48 kHz (o 32 canales a 96 kHz), 24 entradas y salidas digitales ADAT Lightpipe hasta 48 kHz (12 entradas y salidas digitales ADAT Lightpipe cuando se trabaja a 96 kHz), 24 entradas y salidas analógicas balanceadas, Bus de Expansión TDM®, y mucho más.

* 56 o 64 entradas y salidas digitales MADI trabajando hasta 48kHz; 28 o 32 entradas y salidas digitales MADI trabajando hasta 96kHz.
* 24 entradas y salidas analógicas balanceadas, con AD/DA SSL hasta 96 kHz, sobre conectores D-Sub (para su uso con adaptador D-Sub a XLR).
* Entrada y salida Word/SuperClock.
* Funcionalidad entrada Video Sync para la entrada WordClock: la señal Video Sync puede ser convertida internamente y enviada a través de la salida WordClock.
* Entrada y salida MIDI.
* Conexión a Mixpander a través del puerto de bus de expansión.
* Todos los canales de entrada y salida pueden utilizarse simultáneamente, en cualquier configuración.
* Completa matriz de direccionamiento de entrada / salidas y selección de modo.
* Medición analógica de entrada / salida, a través de 24 LEDs tricolor.

XTA ELECTRONICS DP 324 SIDD

MultiProcesador Digital de Audio XTA. 24bit/48KHz. 40bit doble precisión interno. Dispone de 2 entradas, 2 salidas principales y 2 salidas secundarias. 6 bandas de ecualización paramétrica por cada entrada; filtros paso-alto y paso-bajo; módulo de ecualización dinámica; compresor con dos bandas de ecualización y codo variable; puerta de ruido o expansor; limitadores de salida con control ataque, recuperación y umbral; procesado “side-chain y retardo de anticipación en cada bloque de proceso; generador de armónicos para simulación de sonido a válvula. Hasta 300 mS de retardo principal. Hasta 100 mS de retardo en salidas auxiliares. MIDI, RS232, RS485. 256 memorias no-volátiles. Tarjeta de almacenamiento de memoria para PC. Lectura independiente de nivel para cada entrada y salida, medidores en cada bloque de proceso. Pantalla LCD. Bloqueo de seguridad multi-nivel.Control vía panel frontal o mediante Software SIDD WindowsTM. Conectores XLR balanceados. Procesador de 1 u. de “rack” y 19” Standard. OPCIONES: transformador para entradas/salidas; AES/EBU; tarjeta PCMCIA de 256K; interfaz de control general.

DBX 1231 1215 EQ GRAPHIC

ECUALIZADORES GRAFICOS SERIE 12

Control Input Gain: Este control ajusta el nivel de señal que va al ecualizador. Su capacidad es de -12dB a +12dB de ganancia. Su efecto es aparente con solo mirar el GRAFICO DE BARRAS DE NIVEL DE SALIDA.
EQ Bypass: Este interruptor elimina la sección del ecualizador gráfico de la ruta de la señal. El interruptor BYPASS, sin embargo, no afecta a la GANANCIA DE ENTRADA, ni a los filtros de CORTE DE GRAVES.
Piloto EQ Bypass: Este LED rojo se ilumina cuando el EQ esté en el modo bypass. Tenga en cuenta que este modo de anulación solo afecta a la sección de ecualizador de los EQs de la Serie 12. Los controles INPUT GAIN y LOW CUT no se ve afectados cuando el ecualizador queda en bypass.
Interruptor y LEDs de selección de rango de realce/corte: Este interruptor elige cual de los dos rangos de realce/corte usará el ecualizador, si el de ±6dB o ±15dB. El piloto rojo se ilumina cuando se selecciona el rango ±15dB, y el amarillo se enciende cuando se elige el rango ±6dB. Observe que el interruptor BOOST/CUT está ligeramente hundido. Esto se ha hecho para evitar la activación del interruptor por accidente, lo que podría producir posiblemente daños en el resto de los componentes de su sistema.
Gráfico de barras de nivel de salida: Estos cuatro LEDs indican el nivel de salida del ecualizador. El piloto rojo está 3dB por debajo de la saturación y está marcado como +18dBu. Este indicador gráfico se ocupa de monitorizar el nivel en la salida del ecualizador después del resto de procesados.
LED Clip: Este piloto se ilumina siempre que el nivel de señal interno llega a los 3dB por debajo de la saturación, lo que puede ocurrir en cualquiera de estos casos: 1) cuando la señal de entrada es “más activa” que +22dBu, 2) cuando se aplica demasiada ganancia con el control de ganancia de entrada, o 3) cuando se aplica un realce excesivo por medio de los mandos deslizantes de frecuencia.
Controles deslizantes de bandas de frecuencia: Cada uno de estos potenciómetros deslizantes realzarán o cortarán la frecuencia correspondiente en ±6dB o ±15dB, dependiendo de la posición del interruptor BOOST/CUT RANGE. Cuando todos estos controles estén en la posición con muesca central la salida del ecualizador será plana. Los centros de las bandas de frecuencia del 1231 están marcados a intervalos de 1/3 de octava en separaciones standard ISO, mientras que los centros de las bandas de frecuencia del 1215 están marcados a intervalos de 2/3 de octava a separaciones standard ISO.
Interruptor de activación de corte de graves: El interruptor LOW-CUT introduce o suprime de la ruta de la señal el filtro Bessel de corte de graves de 40Hz y 18dB/octava. Cuando este interruptor LOW-CUT está pulsado, el filtro de CORTE DE GRAVES está ACTIVADO en la ruta audio.

XTA ELECTRONICS Crossover DS800

Distribuidor de Señal de Audio Activo XTA. 8 canales micro/línea en 2 unidades estándar de “rack” de 19”. Cada canal proporciona 2 salidas activamente balanceadas. +30dBu antes de saturación. Ajuste de ganancia entre +10 y +50dB, conmutador micro-línea, selector de alimentación fantasma a +48v., lectura de nivel a leds de 5 segmentos y escucha enrutada al bus de “solo”. Escucha por auriculares con control de nivel. Bus link. Conectores XLR balanceados. OPCIONES: transformador para salidas de señal balanceadas.

XTA MIC LINE DISTRIBUIDOR

The XTA DS800 Mic / Line Distribution System is a high quality active distribution /splitter unit containing eight channels of processing in 2U of rack space. Each channel features one actively balanced input and a total of four outputs, two of which are individually actively balanced and normally connected to Front Of House and Stage Monitor consoles. The two remaining outputs are transformer isolated and located on the front panel of the unit. These are typically used as broadcasting or recording feeds. A maximum of 50dB gain is available via the five position Gain switch. This allows sufficient gain to be applied to low level signals, such as dynamic microphones, to avoid the poor noise performance often associated with active mic splitters. Phantom power (48VDC) is provided, and is switchable to each input by selecting ‘Phantom Power’ on the front panel. The DS800 is designed to reduce system grounding problems; please refer to the ‘System Grounding’ section further on in this manual for more information.