Stereo multi-effects including Ultrabass, synth, delay/loop sampler, chorus, flanger, rotary speaker, voice box, auto wah, phaser, ambience and reverb
125 memory locations including original artist presets
Intuitive user interface with direct display of all essential settings
Additional effect parameters directly accessible on the unit; Tap-tempo function allows real-time adjustment of effects speed parameter
4 renowned distortion and overdrive stomp boxes with adjustable Drive, Tone, Boost and Split
Dedicated wah pedal and studio compressor effects
Effective Presence, Deep and sweepable Shift/Shape controls for all amp models
Sweepable 24 dB Butterworth frequency crossover for bi-amping operation
Stereo Aux input lets you play along with a CD, drum computer or MIDI playback for practice, teaching and home-recording applications
Balanced stereo Line output with virtual speaker simulation for recording and live applications
Gig bag and dual footswitch for preset selection and tuner control included
Adjustable auto-chromatic tuner
MIDI implementation includes program changes, control changes and SysEx, allowing complete MIDI automation through our free Windows editor software downloadable at www.behringer.com
High-quality components and exceptionally rugged construction ensure long life
Conceived and designed by BEHRINGER Germany
Any Classic You Like The BASS V-AMP's 32 amp models are organized into three groups: ACOUSTIC, KEYBOARD and GUITAR. Just twist the dedicated dial to plug into unbelievably authentic simulations of the greatest bass sounds from the 1950s to today. When you select an amp model, a speaker cabinet model is automatically selected. But if you'd prefer a different model, just go to EDIT > CABINETS, then use the BANK UP/DOWN buttons to navigate through all 15 options.
Just like on an actual bass amp, you've got a three-band EQ, GAIN, VOLUME and MASTER VOLUME dials. There's even a COMPRESSOR dial for tight, smooth tone with a variable attack. By holding the TAP button, the TREBLE dial becomes a PRESENCE control. This enables the adjustment of a high-frequency filter that can add or remove a "shimmer" from the BASS V-AMP's tone
Top Tone—In Full Effect The BASS V-AMP gives you flanger, chorus, reverb, delay and 12 other beautiful digital effects to apply to your sound. Adjust the effect's level in the mix via the EFFECTS dial. To adjust effect parameters, you can rhythmically press the TAP button to alter delay times or LFO speeds, or hold down the TAP button and twist the EFFECTS knob to access e.g. feedback or depth. By using a MIDI pedal, such as the BEHRINGER FCB1010, you can access an additional Wah Wah effect. You can adjust the filter characteristic by entering EDIT mode, selecting the DRIVE menu and turning the EFFECTS knob while holding the TAP button. Presenting the Presets. All of these elements have been combined into 125 different presets divided into 25 banks, accessible through the BANK UP/ BANK DOWN buttons. Within each bank, you can press buttons A through E to explore its different presets. You can also create and save your own presets. Start by selecting a preset, then making any changes you wish: amp, EQ, effect, etc. The preset LED will begin to flash, indicating that the BASS V-AMP is ready to save your settings. To do this, hold the preset button down for about two seconds. Once the LED stops flashing, you have saved the modified preset. On top of all this, the BASS V-AMP also features a built-in tuner and noise gate. In essence, it's a Swiss Army Knife for the home recording enthusiast or gigging bassist who can't seem to settle on one amp. Stunning in the studio The BASS V-AMP features several configurations that make it an invaluable asset in the studio. Access the configuration menu by pressing the B and D buttons at the same time. Then use BANK UP and BANK DOWN to select a configuration.
When using headphones, the BASS V-AMP will switch into Studio Mode 1 (S1). This stereo mode works well for both monitoring and recording with effects, amp and speaker modeling. It doesn't apply the additional three-band EQ to the signal, but most recording programs—like energyXT2.5— have their own EQ. In Studio Mode 2 (S2), the BASS V-AMP sends a signal with amp and speaker simulation, but only the right output has effects. You can either record both left and right outputs to separate tracks, or record the "dry" left output while monitoring the right output. Stellar on Stage The BASS V-AMP has three live configurations. When performing live with this device, you can either run it to an amp on stage, run directly to the mixing board, or both. Live Mode 1 is the best choice for running the outputs directly to the mixer, because it applies amp and speaker simulation, effects and an additional 3-band EQ.
Use Live Mode 2 to run the BASS V-AMP into the effects return inputs of 2 different guitar amps. This configuration does not feature speaker simulation, allowing the amplifier's natural speaker sound to go unaltered. In this scenario, the amps would be mic'd on stage to get the sound to the house speakers, and having a dedicated monitor would be less important. In Live Mode 3, you can send the left output to an amp's effects return for onstage volume, while the right output sends signal to the main mixer for inclusion in the main house mix. Also note that you can press the TAP key while turning the GAIN knob to adjust the input sensitivity to match the output of your guitar's pickups. If the CLIP LED lights, the input's sensitivity should be reduced. Veni, MIDI, Vici The BASS V-AMP comes fully equipped to fit right into a MIDI (Musical Instrument Digital Interface) setup. It can be controlled in real time from a computer, or via a MIDI foot controller during a live performance. Its MIDI connectors are international-standard 5-pin DIN connectors. You will need dedicated MIDI cables (not included).
The MIDI IN jack receives MIDI controller data. It can be adjusted in EDIT mode by pressing the A button and then using the arrow keys. MIDI OUT/THRU sends data to a computer. You can transmit both preset data and parameter changes. Some people even use the BASS V-AMP as a remote control for parameters of their DAW or VST instruments, which is more handy than using the mouse alone. If set to MIDI THRU, the BASS V-AMP does not send its own MIDI information, but passes on the signal received at the MIDI IN connector. So Much Sound, So Little Hassle It's hard enough finding another virtual bass amplifier with this much versatility, let alone one as economical as the BASS V-AMP. This go-anywhere, do-anything wonder will leave you reeling in your own creativity, with cash to spare for other gear you'll need along the way to becoming a sonic titan both live and in your home studio.
Want to perform like the stars? Consider what adding an easy to use DSP effect unit to your system could do for your live sound or recording. VocoPro is proud to present the VSP-M1, the World's First effect processor designed for both KJs and DJs. Both will finally get the best of both worlds in just one rack space. In addition to the seven fully controllable digital effects and versatile Vocal Harmonizer, there's also a BPM Auto Synchronization feature to align effects with the beat of your source music. The ability to mix Harmony effects or Reverb with Digital Echo is a must have for any singer, which makes the VSP-M1 perfect for both performing or recording artists. Take the VSP-M1 for a spin and see how VocoPro can turbo chagre your sound.
Features: Clean, professional quality reverb for vocals or music Vocal Harmonizer adds choral effects to any Mic Input 8 DSP Effects For Processing Microphone Or Source Music Including Pan, Filter, Flange and More! 3 XLR/Phono jack Mic Inputs with individual preamps Digital Echo can be mixed with Harmonizer Effects for great sounding vocals Great effects for KJs and DJs, live sound and recording! Balanced XLR outputs
The Dual Deltafex is like having two multi-effects units in one. Providing powerful digital signal processing with flexible dual audio connections in a single rack space makes this unit perfect for studio or live applications. Dual inputs and outputs allow for four operational modes; Series, Parallel, Dual Mono and Sum Mono. Both Series and Parallel modes process a stereo input signal into a stereo output signal. Dual Mono mode configures the unit exactly like two separate stereo effects processors. Sum Mono works the same as Dual Mono, except it combines the two effects units into one pair of outputs. Featuring 16 programmable effects including multiple reverbs and delays, compressor, pitch shifter, chorus, flanger, phaser, tremolo, rotary speaker, distortion, exciter and karaoke (vocal eliminator), the Dual Deltafex is truly two units in one. You'll save money and space with the Dual Deltafex.
Features:
24 bit Stereo inputs and outputs
Stereo, Parallel, Series and Dual Mono modes
16 different effects types for each engine including multiple reverbs, delays, compressor, pitch shifter, chorus, flange, phase, tremolo, rotary speaker with morphing speed control, distortion, exciter, karaoke and 2 adjustable parameters for each effect
Reverb and delay tails continue when effect is switched off
Mix control
Input and output level controls
Defeat jack doubles as speed control for rotary speaker
The KC-300 Pro's Sonic Enhancer will dramatically improve the sound quality of your vocal system, allowing you to fine tune the dynamics of the high and low ends of your audio signal. There is also a studio quality 17-step DSP Key Controller to raise and lower the natural musical key by 2 full octaves, and key changes can be made to any compatible input source. Also on-board are Vocal Cancel and Vocal Elimination features that remove lead vocals from music tracks. Vocal Cancel removes the vocals from Multiplex tracks, while Vocal Elimination removes the vocals from standard CD's! The KC-300 Pro is solidly constructed into a sleek 1RU case, which means you'll hardly have to make room for it! It's large and easy to read Multi-Display will never have you squinting, even in those near dark situations. It's rear panel houses Inputs for two A/V devices and one line-level Auxiliary device, and one A/V Output with dual Video Outputs. There is even a set of XLR output jacks in case you need them! So, if you are seeking to unleash the potential of your vocal system and experience professional level control and sound quality, go with the KC-300 Pro by VocoPro, your choice for ultimate vocal entertainment!
Features:
17-Step DSP Key Controller
Vocal Cancel to Remove Vocals from Multiplex Karaoke Software
Vocal Reducer to eliminate vocals from standard CD/CD+G's
Sonic Enhancer optimizes high/low frequencies
Full function remote control
115V-230V switchable
Shipping Dimensions: 16 1/2" (L) x 21" (W) x 5" (H)
Take control of every facet of your sound reinforcement with Peavey's new VSX loudspeaker management systems. The VSX 26 and VSX 48 feature exclusive Peavey digital processing technology, upgradeable software and functionality for use in singleor multi-zone applications-an all-in-one solution in a single 1U rack space! Any combination of input to output is user configurable to match your system needs. Symmetrical and asymmetrical crossover configurations are user selectable.
Features:
48 kHz sample rate, 24-bit 256x over-sampled Delta-Sigma AD, DA
USB A & B ports for memory storage and computer interface (PC and Mac)
XLR inputs and outputs, AES-EBU digital input
+24 dBu inputs and outputs
Phantom power +48 volt
Built-in signal generator (white, pink, sine)
Easy preset and system updates via Internet
Fully adjustable crossover points
Easy-to-read LCD screen
Features per input: 27-band Graphic EQ Autograph algorithm for automatic EQ settings, Compressor/Limiter, Delay (340 ms)
Features per output: Crossover/Bandpass filter (Butterworth, Bessel, Linkwitz-Riley), Parametric EQ (5-band, HP, LP Notch, Horn EQ, All-Pass), Compressor/Limiter, Delay with polarity inversion
2 input/6 output loudspeaker DSP management system
The SP260 uses high-end 24-bit AKM® AD/DA converters with 120dB dynamic range for incredible sound quality. It boasts 24 user-definable presets and flexible I/O for easy routing and configuration. The SP260 is an ultra-flexible processor, ideal for many portable and permanent applications.
Easy Input Possibilities: The Mackie SP260 has two inputs that instantly connect your mix into a whole host of processing and routing capabilities. There is a mute switch and variable gain control on each input. It doesn't matter how many boxes you have; itt's simple to control the whole system. Turn these down and the entire system turns down as well.
With a 5-band EQ on both inputs, you can choose between parametric or hi/lo shelving, which allows for a flexible approach to room tuning. It has a 600ms delay available on each input. This is good when you have multiple Mackie SP260s driving delay stacks in huge venues. You could also time align the stage with the main speakers if you wanted.
Adjustable Individual Outputs: Along with the inputs on the Mackie SP260 are six individual outputs, meaning many connection possibilities, depending on your system. There is a mute switch and individual gain adjustment on each output, meaning you can turn up seperate outputs without having to adjust power amps. Each output also has a dedicated 7-band EQ, complete with selectable parametric or shelving control per band. When you want to properly voice a loudspeaker for the appropriate material or venue, this proves critical, . By adjusting the voicing with this comprehensive EQ control, you can dial in the ideal response curve for the system and application at hand.
Sound Quality of the Professional: Designed for the professional audio engineer the Mackie SP260 you get total control over every single parameter. With high-end AKM AD/DA converters with 120 dB dynamic range and a sophisticated DSP engine and algorithms, you can be assured of superior sound quality.
An Easy Setup: Plug the Mackie SP260 in and fire away. There's already a choice of pre-configured system setups to assign the correct inputs, name outputs and give the starting crossover point for your system. Once that's done, you can use the available input and output processing to optimise your system for the venue. After that, you can save the setup in one of the 24 available presets. Talk about convenience. If you use your system in the same place often or switch between different sets of gear, this can prove very handy and fuss-free.
Mackie have even made downloadable presets for their popular systems to give you further easy setup possibilities. Since you can also link inputs or any combination of outputs, this saves time adjusting parameters that you want to repeat on other channels.
Control the SP260 From Your Computer: The Mackie SP260 Control Application, is an easy-to-use interface. Just plug in your Windows laptop via the convenient front panel USB port. The software auto-detects the SP260 and shows the current settings. You can control each and every SP260 feature via this app, which features a high-contrast design which is great for low-light areas. You can view and control all processing elements for any input or output. You can even manage and store your presets on the PC for use later, meaning you can setup for shows ahead of time or change things you'd like to improve for the next show.
Features:
Professional 2-input, 6-output system processor for passive and powered PAs
Sophisticated, intuitive EQ, crossover, delay and dynamics processing optimises and protects your system
High-end 24-bit AKM AD/DA converters with 120dB dynamic range
Two balanced XLR inputs with level control, mute, 600ms delay and 5-band parametric/shelving EQ
Six balanced XLR outputs with level control, mute, polarity invert, dedicated high and low pass filters, 7-band parametric/shelving EQ, 600ms delay, and limiting
Powerful limiter on every output for ultimate protection
Convenient front-panel controls for parameter editing and input/output muting
USB port for simple programming via PC
Linkable inputs and outputs for easy stereo configuration
7-segment LED metering for input/output level or limiter activity
5 operating modes: 2 x stereo + sub, 3 x stereo, 2 x 2-way + sub, 2 x 3-way and 1 x 6-way
24 presets for storing your Mackie or other system parameters
After three decades of developing pioneering audio products and systems, QSC raises the bar once again, setting a new standard in signal processing with the new DSP-3 Digital Signal Processing Module.
The DSP-3's two channels of independent signal processing deliver more power, flexibility, and features-in short, more of everything you really need in a DSP-for less. Simple to install and compact, the DSP-3 is the advanced and affordable digital signal processing solution for your audio system.
If you would like to purchase a DSP-3 module, please call our sales department at: 800-854-4079.
Powerful The DSP-3’s powerful processor allows you to perform a wide range of signal processing functions. Whether you need speaker crossovers, EQ, time delay, or subsonic filters, the DSP-3 is as flexible as your systems's needs.
Each channel includes:
Crossover filtering
Multiple parametric EQs
Shelf filtering
Tone and noise generation
Precision attenuation
Multiple delays (up to 910 ms)
Mixing
Compression and limiting
Configurable The DSP-3’s processing horsepower is dynamically assignable so you’re not locked into a fixed configuration. Simply plug your laptop into its RS-232 serial interface and use QSC's powerful configuration software. You can easily configure multiple processing functions and signal flow with "click-and-drop" simplicity. The DSP-3 also allows real-time control or set-and-forget convenience. You can even save each configuration on disk for future recall.
Versatile Getting attached to the DSP-3? Then you’ll be glad to know how easily it attaches to most two-channel DataPort-equipped QSC amplifiers, including the CX, DCA, and new PowerLight 2—without taking up extra rack space. Or use multiple DSP-3s with our rack-mount panel kit to create a stand-alone DSP solution that interfaces with any amplifier or system.
Features:
Hardware
Two independent channels of DSP
48kHz, 24-bit converters
No turn on pops or zipper noise
If the memory or hardware fails, unit turns on muted
Host interface via RS-232 or QSControl Audio Network System* via CM16a Amplifier Network Monitor
VSX 48 Digital Loudspeaker Processor The VSX 48 is a fully progammable 32-bit audio processing and louspeaker management control system. Considerably more powerful than similarly priced units, the VSX 48 provides a versatile and economical alternative for system designers.
Features:
5 inputs/8 outputs
48 kHz sample rate 24-bit 256x over-sampled Delta-Sigma AD, DA
The AO-1 is an eight-channel analog audio input module for the NetMax N8000 and N8000-1500 System Controllers. Audio signals are connected via screw-lockable Euroblock connectors. A-to-D conversion is performed by high-performance, linear, 24-bit converters. The signals are processed internally at 48-bit kHz
Features:
Designed for Cinema, Club Sound, Commercial Sound/Life Safety, Concert Sound, House of Worship, Performance & Sports Venues, Stadium Systems, and other applications.
Eight line-level outputs on Euroblock connectors
Electronically balanced outputs
100 output impedance
118 dB dynamic range provides superior sonic quality
Automatic configuration-Indication of installation and removal in <a href="http://irisnet.electrovoice.com/">IRIS-Net</a>
DSP (100 MIPS) on board
Analog Outputs: 8 x 3-pole Euro block connectors, electronically symmetric D/A Conversion: 24 Bit, Sigma-Delta, 128 times oversampling Data Format: 24 Bit linear D/A conversion, 48 Bit processing Electronics Type: Processor Frequency Response: 20 Hz - 20 kHz (- 0.5 dB) Internal Processing: 100 MIPS Maximum Output Voltage: +21 dBu / 8.7 V Nominal Output Voltage: +6 dBu / 1.55 V Output Impedance (Balanced): 100 Ω Sample Rate: 48 kHz Signal-to-Noise Ratio (A-weighted): 118 dB THD+N: < 0.005 dB THD+N: < 0.005 dB Height: 1RU 33 mm (1.3") Width: 114 mm (4.49") Depth: 258.5 mm (10.18") Weight Net: 9.17 oz (260 g)
The DO-1 is an eight-channel digital output module for the NetMax N8000 and N8000-1500 System Controllers. Digital audio signals in AES/EBU format are connected via screw-lockable Euroblock connectors. Signals are processed internally in 48-bit word length.
Features:
Designed for Broadcast, Cinema, Club Sound, Commercial Sound/Life Safety, Concert Sound, House of Worship, Performance & Sports Venues, Stadium Systems, and other applications.
Four outputs for eight channels of AES/EBU digital audio output
48 kHz sampling rate
+21 dBu maximum output level
Automatic configuration-Indication of installation and removal in IRIS-Net
DSP (100 MIPS) on board
Data Format: 24 Bit linear, 48 Bit processing Digital Outputs: 4 x 3-pole Euro block connectors Frequency Response: 20 Hz - 20 kHz (±0.1 dB) Output Formats: AES/EBU Professional Format Output Impedance (Balanced): 110 Ω Sample Rate: 48 kHz Signal-to-Noise Ratio (A-weighted): 144 dB THD+N: < 0.00002% Height: 1RU 33 mm (1.3") Width: 144 mm (5.67") Depth: 258.5 mm (10.18") Weight Net: 6 oz (170 g)
The NAC-100 is a network audio controller allowing the system designer to create custom, graphical user interfaces using QSControl.net version 3.1 (or higher) systems. Bridging the gap between simple switch closure or potentiometer controls on one hand and sophisticated (and expensive) touch panel or computer-hosted control on the other, the NAC-100 will be at home in hospitality, convention, transportation, entertainment, worship and educational facilities. Since the NAC-100 operates via PoE (Power Over Ethernet), wiring consists of a single CAT-5 cable terminating to an RJ45 connector.
Operation and screen navigation is accomplished by means of five push-buttons arranged in an intuitive left/right, up/down configuration plus a rotary encoder. When multiple NAC-100 controllers control a single system, changes made from one unit will be reflected on the displays of others. System control can range from simple control over a single audio channel to recall of global presets. Creation of the custom user screens is accomplished within the QSControl.net environment. For applications in which room diagrams, corporate logos or other graphical elements are desired, the NAC- 100 can support up to 16 full-color, bitmapped images. Graphics and text are displayed on a 3.5 inch (8.9 cm), backlit, color display with 320 x 240 pixel resolution. Available in black or white, the NAC-100 may be mounted directly to a wall surface. The rear panel of the unit includes holes conforming to the hole patterns of numerous domestic and international electrical boxes.
Features:
Easily create intuitive user interfaces from within QSControl.net™
Capable of displaying custom bitmapped graphics
16-bit, 3.5 inch (8.9 cm), backlit graphics display
Five navigation/ control switches and rotary encoder
Simple, single-wire connection using PoE (Power Over Ethernet)
Available in white
Display: 3.5" (89 mm); backlit, 16-bit color LCD; 320 x 240 pixels; CSTN Controls: 5 x switches, 1 x rotary encoder Software Compatibility: Requires QSControl.net version 3.1 or higher Dimensions (HWD): 5.2" (132 mm) x 5" (127 mm) x 1.6" (40 mm) Power Requirements: Compliant with IEEE 802.3af-2003, works with any PoE compliant switch or PoE injector Construction: ABS with UV protection additive, available in black (NAC-100-WH)
The NAC-100 is a network audio controller allowing the system designer to create custom, graphical user interfaces using QSControl.net version 3.1 (or higher) systems. Bridging the gap between simple switch closure or potentiometer controls on one hand and sophisticated (and expensive) touch panel or computer-hosted control on the other, the NAC-100 will be at home in hospitality, convention, transportation, entertainment, worship and educational facilities. Since the NAC-100 operates via PoE (Power Over Ethernet), wiring consists of a single CAT-5 cable terminating to an RJ45 connector.
Operation and screen navigation is accomplished by means of five push-buttons arranged in an intuitive left/right, up/down configuration plus a rotary encoder. When multiple NAC-100 controllers control a single system, changes made from one unit will be reflected on the displays of others. System control can range from simple control over a single audio channel to recall of global presets. Creation of the custom user screens is accomplished within the QSControl.net environment. For applications in which room diagrams, corporate logos or other graphical elements are desired, the NAC- 100 can support up to 16 full-color, bitmapped images. Graphics and text are displayed on a 3.5 inch (8.9 cm), backlit, color display with 320 x 240 pixel resolution. Available in black or white, the NAC-100 may be mounted directly to a wall surface. The rear panel of the unit includes holes conforming to the hole patterns of numerous domestic and international electrical boxes.
Features:
Easily create intuitive user interfaces from within QSControl.net™
Capable of displaying custom bitmapped graphics
16-bit, 3.5 inch (8.9 cm), backlit graphics display
Five navigation/ control switches and rotary encoder
Simple, single-wire connection using PoE (Power Over Ethernet)
Available in black
Display: 3.5" (89 mm); backlit, 16-bit color LCD; 320 x 240 pixels; CSTN Controls: 5 x switches, 1 x rotary encoder Software Compatibility: Requires QSControl.net version 3.1 or higher Dimensions (HWD): 5.2" (132 mm) x 5" (127 mm) x 1.6" (40 mm) Power Requirements: Compliant with IEEE 802.3af-2003, works with any PoE compliant switch or PoE injector Construction: ABS with UV protection additive, available in black (NAC-100-BK)
With more than three decades of experience pioneering cutting edge audio products, QSC raises the bar once again with the DSP-4 Digital Signal Processing Module. This compact module offers two channels of independent DSP and attaches to the back of most 2-channel DataPort-equipped QSC amplifiers—without occupying any additional rack space.
Capitalizing on the success of our DSP-3, the second-generation DSP-4 provides several enhancements in functionality and performance while also incorporating the universally popular XLR balanced connectors. These enhancements include new A/D and D/A converters for improved signal-to-noise performance and upgraded software that significantly increases the unit’s operational characteristics.
If you would like to purchase a DSP-4 module, please call our sales department at: 800-854-4079.
Powerful Simple to install, compact, and featuring "set-and-forget" convenience, the DSP-4’s powerful processor enables you to perform a wide range of signal processing functions; and with its new A/D and D/A converters, its noise floor improves. Whether you need speaker crossovers, EQ, time delay, or subsonic filters, the DSP-4 is as flexible as your system’s needs.
Each channel includes:
Crossover filtering
Multiple parametric EQs
Shelf filtering
Tone and noise generation
Precision attenuation
Multiple delays (up to 910 ms)
Mixing
Compression and limiting
Configurable The DSP-4’s processing horsepower is dynamically assignable so you are not limited by a fixed signal chain. Simply use QSC’s powerful PC-based Signal Manager software to easily configure multiple processing functions and signal flow with “drag-and-drop” tools.
Cost-effective The power and flexibility of the DSP-4 eliminates the need for expensive outboard processing gear, reducing cost and installation time for almost any application. The compact DSP-4 also plugs directly into the back of most QSC DataPort-equipped amplifiers for use in systems where rack space is a premium.
Features:
Hardware
Two independent channels of DSP
48kHz, 24-bit converters
No turn on pops or zipper noise
If the memory or hardware fails, unit turns on muted to prevent driver damage
Host interface via RS-232 or QSControl Audio Network System via CM16a Amplifier Network Monitor
DSP processing power and memory is dynamically assigned to signal processing functions - eliminating the limitations imposed by fixed signal chain designs
Graphical representation of DSP resources
Firmware upgrades via RS-232
Hard copy printout of signal flow layout or parameter settings
System Requirements
Windows 98, NT4 (SP6), and 2000 (SP1)*
SVGA monitor @ 800 x 600 (min.); 1024 x 768 recommended
CD-ROM drive
32 MB RAM (min.)
10 MB free hard disk space (min.)
Available RS-232 COM port
Male-to-female 9-pin serial cable (for programming)
Featuring intuitive PC system configuration combined with "set-and-forget" convenience, the DSP-30 unites easy-to-use, customizable, two-channel digital signal processing (DSP) with a simple preset selection interface that requires only two buttons. It can be used with all amplifiers and is housed in a 1RU, 19-inch rack-mount steel chassis. Sampling frequency is 48 kHz with 24-bit resolution. Dynamic range is greater than 95 dB. Rugged and dependable in the spirit of all QSC professional audio products, the DSP-30 is well suited to a variety of applications including mobile DJ, club PA, and pro touring.
Powerful The DSP-30's powerful processor enables a wide range of signal processing functions. Whether you need speaker crossovers, EQ, time delay, or subsonic filters, the DSP-30 is as flexible as your system's needs.
Each channel includes:
Crossover filtering
Multiple parametric EQs
Shelf filtering
Tone and noise generation
Precision attenuation
Multiple delays (up to 910 ms)
Mixing
Compression and limiting
Configurable The DSP-30's processing horsepower is dynamically assignable, so you are not limited by a fixed signal chain. Simply use QSC's powerful PC-based Signal Manager software to easily configure multiple processing functions and signal flow with "drag-and-drop" tools. The DSP-30 provides eight fully configurable user presets, selectable from front-panel switches.
Cost-effective The power and flexibility of the DSP-30 eliminates the need for individual outboard signal processors—reducing cost, space, and installation time for almost any application. Housed in a 1RU, 19-inch rack-mount steel chassis, it can be used with all audio systems.
Features:
Hardware
48kHz, 24-bit converters
No turn on pops or "zipper" noise
If the memory or hardware fails, unit turns on muted to prevent driver damage
Easy PC connection with front panel RS-232
Balanced Neutrik® Combo (XLR and 1/4") inputs and XLR outputs
Power and signal present LEDs with signal level
Numeric display indicates current preset
Eight fully configurable user presets
Preset Browse and Accept buttons with lock-out feature
Hard copy printout of configuration layout or parameter settings
DSP processing power and memory dynamically assignable to signal processing functions— eliminating the limitations imposed by fixed signal chain designs
The AI-1 is an eight-channel analog audio input module for the NetMax N8000 and N8000-1500 System Controllers. Audio signals are connected via screw-lockable Euroblock connectors. A-to-D conversion is performed by high-performance, linear, 24-bit converters. The signals are processed internally at 48-bit kHz.
Features:
Designed for Cinema, Club Sound, Commercial Sound/Life Safety, Concert Sound, House of Worship, Performance & Sports Venues, Stadium Systems, and other applications.
Eight line-level inputs on Euroblock connectors
Electronically balanced inputs
10 k input impedance
117 dB dynamic range provides superior sonic quality
Automatic configuration-Indication of installation and removal in <a href="http://irisnet.electrovoice.com/">IRIS-Net</a>
DSP (100 MIPS) on board
A/D Conversion: 24 Bit, Sigma-Delta, 128 times oversampling Analog Inputs: 8 x 3-pole Euro block connectors, electronically ballanced Data Format: 24 Bit linear A/D conversion, 48 Bit processing Frequency Response: 20 Hz - 20 kHz (- 0.5 dB) Input Impedance (Balanced): 20 kΩ Internal Processing: 100 MIPS Maximum Input Voltage: +21 dBu / 8.7 V Nominal Input Voltage: +6 dBu / 1.55 V Phantom Power: no Signal-to-Noise Ratio (A-weighted): 117 dB THD+N: < 0.005 dB THD+N: < 0.005 dB Height: 1RU 33 mm (1.3") Width: 114 mm (4.49") Depth: 258.5 mm (10.18") Weight Net: 7.05 oz (200 g)
MuseBox takes advantage of virtual instrument and effects software for
cutting-edge sounds and unprecedented flexibility. Portable, compact,
and completely self-contained - just plug in, and play
Simple Sound-Finder based user interface lets you find that just right sound or effect FAST.
Tweak and program instruments and
effects by simply connecting a mouse, monitor and keyboard directly to
the MuseBox. Or edit and program remotely by connecting the MuseBox to
your computer and running the exclusive Muse Remote software.
Two pro-grade mic/instrument inputs
on the front panel allow you to connect your mic and guitar so you can
handle any performance situation.
MIDI jack and line inputs on the
back let you compliment your existing keyboard sounds with exciting new
software-based instruments.
The super portable, super flexible MuseBox
Musical Instrument and Effects Box from Muse Research and Development
uses virtual instruments and effects technology in a brand new way, so
you can easily take them to your rehearsals, gigs, to the practice room,
or to the studio. Using technology derived from its powerful big
brother, the award-winning RECEPTOR Hardware Plug-in Player, this
compact 2U half rack design is ultimately portable, incredibly
versatile, and built for the road.
The key to MuseBox is its custom software environment that takes
computer technology and turns it into a musical instrument that works in
any musical situation for any kind of musician. Now you can harness the
world's best sounding virtual instruments and most versatile,
tone-laden effects and take them anywhere - and everywhere.
It's Anything You Want It To Be
MuseBox features incredibly flexible I/O, with front panel guitar and
microphone inputs, MIDI, and USB. So you can play through it, sing
through it, process your keyboards, play along to backing tracks, and of
course, create lush keyboard sounds with your MIDI controller.
Guitarists can use it to run their favorite amp modeling software,
multi-effects libraries and more. Singers can enhance their vocals with
reverb, compression, de-essing and pitch correction in real time.
Keyboardists can run virtual pianos and delicious sounding virtual
instruments with super-low latency and superb stability.
Sounds Great Right Out Of The Box
MuseBox comes pre-loaded. Simply turn it on, press the PLAY button,
select the sound of your choice and off you go into the superior
sounding world of virtual instruments and effects. It's got all the
"meat and potatoes" keyboards to cut any gig. Guitarists will dig
Peavey's ReValver HP Guitar Amp Modeler for killer sounds. And of course
there is an ample supply of essential effects, including Reverbs and
Chorus.
Find Sounds Fast
MuseBox's SoundFinder preset architecture lets you quickly sort through
thousands of presets to find the sound you are looking for... and it
automatically adds new presets as you add additional virtual instruments
and effects to the MuseBox.
Easy And Intuitive
You'll find MuseBox easy to use right from the get-go. Our front panel
allows quick and intuitive access to whatever you have loaded in your
MuseBox, so you can call up sounds, tweak parameters, edit effects
chains, whatever you need.
Features: Hardware
Intel Dual-Core Processor
2GB of DDR-2 RAM
8GB Flash IDE disc module
2 channels line level balanced inputs OR 2 channel mic/instrument inputs
2 channels line level unbalanced outputs
Passive mix mode for connecting external CD or MP3 player, drum machine, etc.
Front panel headphone output
Full Size MIDI input
Four USB ports
VGA video output for viewing / editing software interface of plug-ins
Phantom power
Dedicated mic/instrument input volume control and 3 segment LED meters
Ethernet port
Software
New, two channel "DUO HOST" VST plug-in host environment running on Linux
SoundFinder preset structure for easy selection of presets, "Advanced mode" for using as a normal RECEPTOR
A variety of essential high quality keyboard sounds are included
Peavey ReValver HP guitar amplifier modeling software
Assortment of effects, including Reverb and Chorus
RECEPTOR Remote control for control from a computer
MuseControl software interface for controlling individual plug-in GUIs
The DI-1 is an eight-channel digital input module for the NetMax N8000 and N8000-1500 System Controllers. Digital audio signals in AES/EBU or S/PDIF format are connected via screw-lockable Euroblock connectors. Four Toslink. connectors are available for optical transmission of digital audio signals. All inputs are equipped with high-quality, sample-rate converters. Signals are processed internally in 48-bit word length.
Features:
Designed for Broadcast, Cinema, Club Sound, Commercial Sound/Life Safety, Concert Sound, House of Worship, Performance & Sports Venues, Stadium Systems, and other applications.
Four inputs for eight channels of AES/EBU or S/PDIF digital audio input
Euroblock or TOSLINK optical input connectors
Accepts sample rates of 32-192 kHz
Independent sample rate converters allow inputs of different sample rates on each DI-1 input
Lock indication LED
DSP (100 MIPS) on board
Data Format: 24 Bit linear PCM inputs, 48 Bit processing Digital Inputs: 4 x 3-pole Euro block connectors 4 x Toslink optical connectors Frequency Response: 20 Hz - 20 kHz (±0.1 dB) Input Formats: AES/EBU, S/PDIF, Optical Input Impedance AES/EBU: 110 Ω Input Impedance S/PDIF: 75 Ω Sample Rate Conversion (SRC): High End Sample Rate Converter per channel: 32-192 kHz input to 48 kHz output Signal-to-Noise Ratio (A-weighted): 128 dB Height: 1RU 33 mm (1.3") Width: 114 mm (4.49") Depth: 258.5 mm (10.18") Weight Net: 7.05 oz (200 g)
The Finalizer Express is the fast and efficient way to turn your mix into a Professional Master! Based upon TC's Multi-Award winning Finalizer Mastering Technology, it delivers the finishing touches of clarity, warmth and punch to your mixes, putting the world of professional mastering within your reach.
Insert the Finalizer Express between the stereo output of your mixer or workstation and your mastering recording media to refine your tracks with powerful mastering tools, adding real energy to your mix without worrying about "overs".
Punch up your mix with the fast intuitive user-interface and deliver the ultimate sound quality you deserve-quick and clean! Spectral balance is improved, bass is tightened, the level is optimized and your mix sounds like a final master!
Features:
Punch Up Insert the Finalizer Express between the stereo output of your mixer or workstation and your master recording media to refine your tracks with powerful mastering tools, adding real energy to the mix without worrying about "overs". Punch up your mix with the fast, intuitive user interface and deliver the ultimate sound quality you deserve - quick and clean! Spectral balance is improved, bass is tightened, the level is optimized and your mix sounds like a finished CD...it's that simple.
Bring your mixes to life with TC's unique Multiband Compressor and Limiter algorithms
Boost and cut over three bands with the Spectral Balance Controls
Prevent "overs" from occurring with Soft Clipping
Foresee incoming peaks with Look Ahead Delay, allowing for faster, more accurate response
Use the Finalize Matrix for 25 variations in style and rate
Optimize your overall level with the Automatic Make-Up Gain
Add extra compression in each band by using the Emphasis keys
Record your fades from the built in Digital Fader or the optional TC Master Fader via MIDI
24 bit resolution A/D & D/A converters
16 and 20 bit ditherin
Industry standard connectivity: AES/EBU, S/PDIF, Optical Tos-link & MIDI I/Os
High Resolution LED Metering of I/O & multi-band gain reduction
Really Fast Acquiring a piece of high quality 24 bit digital equipment often means hours of manual reading, keeping you away from the main focus point of your work. With the Finalizer Express, you plug it in and you're on the fast track to a great sounding mix.
Express Yourself The Finalizer Express combines the advantages of analog and digital equipment to bring intuition and creative flow into your work. With the aid of an analog style interface and TC's proprietary dynamics processing technology inside, your ideas come to life at the very moment you create them.
Advanced Technology With TC Electronics' award winning multi- band compression technology, you are actually equalizing your track with selective compression, adding real energy and clarity into the mix without having to worry about "overs". The Finalizer Express delivers the "no compromise" audio quality your mixes deserve. If you are working either in the analog or the digital domain the Express has you covered with: 24 bit A/D & D/A's, AES/EBU, SPDIF and Tos-link I/Os.
Fast and Simple Power Power up the Finalizer Express by pressing the Power button. Holding it down for more than three seconds turns the Express off and prevents sudden accidental power downs that could damage your precious equipment.
Digital Inputs and Outputs Connectors: XLR (AES/EBU) RCA Phono (S/PDIF) Optical (Tos-link) Formats: AES/EBU (24 bit) S/PDIF (20 bit) EIAJ CP-340 IEC 958 EIAJ Optical (Tos-link) Processing Delay: 0.2 ms @ 48 kHz Frequency Response DIO: DC to 23,9 kHz ± 0,01 dB @ 48 kHz Max. input level: +22dBu (balanced) Dynamic Range: >103 dB (unweighted), >106 dB (A) Frequency Response: 10 Hz to 20 kHz: +/- 0.2 dB Crosstalk: <-80 dB, 10 Hz to 20 kHz typical -100 dB @ 1 kHz Max. Output Level: +22 dBu (balanced) Frequency Response: 10 Hz to 20 kHz: +0/-0.5 dB EMC Complies with: EN 55103-1 and EN 55103-2, FCC part 15, Class B, CISPR 22, Class B Environment Operating Temperature: 32° F to 122° F (0° C to 50° C) Humidity: Max. 90 % non-condensing Weight: 5.2 lb. (2.35 kg) Mains Voltage: 100 to 240 VAC, 50 to 60 Hz (auto-select) Power Consumption: <20 W Backup Battery Life: >10 years
The DLMS4080 is a self-contained, fully programmable, Digital Loudspeaker Management System capable of handling full control of the Elevation Series or any other loudspeaker system.
The four inputs and eight outputs can be routed in multiple configurations to meet any system requirement. Full system setup and configuration can be done in real time from the front panel or with the intuitive PC based graphic user interface (GUI) software via the on-board RS232, Ethernet or USB port when more complex system control is required.
Features:
4 XLR inputs and 8 XLR outputs can be configured in any fashion
Top sound quality in its class
8 band Parametric EQ and 31 band Graphic EQ on all inputs and outputs
Input and Output Crossovers with Bessel, Linkwitz-Reilly and
Butterworth slopes
Input Compressors and Output Limiters
Precise frequency control is achieved down to a 1Hz resolution
PC based editing software for onscreen control of all parameters
The bass world's response to our introduction of award winning RH450 and Classic450 has been overwhelming. Now, with the help of a true world-class bass legend, we introduce our third amp to deliver instant world-class bass tone - we are proud to present our cooperation with, Tower Of Power's Rocco Prestia on Staccato'51
Think of legendary bass players and Francis Rocco Prestia features on every list. For over 40 years now Rocco's bass has been underpinning the groove for Californian funk legends Tower of Power. In that time Rocco has defined his own style and become synonymous with the much-imitated 'Finger Style Funk' that he pioneered back in the '60s and '70s.
Laura Clapp & Rocco Prestia Laura Clapp presents the new Staccato'51 bass amp to Francis Rocco Prestia of Tower of Power. Afterwards, they talk about how it all came together and how it was to develop the amp together with TC Electronic. We also hear the response from other members of ToP including Rocco's bass tech.
Now TC and Rocco have joined forces to make Staccato'51 - a custom-tuned version of the RH450 that's been made to Rocco's exact specification and ear. It may look like the RH450 bass head, but the sound design is completely different and meets the demands of a legend like Rocco Prestia - both on stage and in the studio.
Rocco's sound design on Staccato'51 fits his legendary finger-muting technique like a glove. The full-bodied tone with plenty of highs and lows as well as the ultra fast precision response enhances the percussive elements of Rocco's playing to perfection.
In addition to the sheer tone shaping, Rocco has added a touch of our SpectraComp™ per-string compression to tighten the sound even more. And finally a touch of TubeTone™ brings out the growl of Staccato'51 when really digging into those low-end notes with his powerful right hand technique.
To achieve perfect tone, Rocco pairs his Staccato'51 with a stack of our RS cabinets - An RS410 and two RS212s - giving him both punch and enough low end to stand out on stage with the 10-piece Tower of Power band.
Uffe Hansen presents Staccato'51 Head of the TC Electronic bass division, Uffe Hansen, presents the new Staccato'51 bass amp. Uffe gives a complete rundown on the new specifications on Staccato'51
BassIda compares Staccato'51 with RH450 TC Electronic’s own bass demonstrator, the very lovely and talented Bass Ida, compares the new Staccato'51 bass amp with RH450. Ida is using RH450 and 2 x RS210's as her main rig and she is trying out Staccato'51 here for the first time.
How Rocco Prestia and TC Electronic developed the Staccato'51 Staccato'51 was created by combining the unique 'Bass Amp 2.0' capabilities of our RH450 and Rocco's decades of experience as a true bass legend. To get the sound exactly as Rocco wanted, we went through thorough tuning sessions along with countless hours of real life gigs on the road with Rocco and Tower of Power. Here's a rundown of how this unique bass amp came into being...
Rocco had been on the lookout for the ultimate bass amp. Hearing rumors about a new and very different bass amp concept – 'Bass Amp 2.0', Rocco arranged to meet with us on the Denmark leg of Tower of Power's 2009 European tour.
Rocco tried out the RH450 rig prior to the Tower of Power sound check - and liked what he heard so much that his usual rig was instantly removed from the stage the RH450 put in its place.
Shortly afterwards the idea of the customized Staccato'51 was born - Staccato because of Rocco's style of playing, '51 to represent his birth year. Next we meet with Rocco in Hollywood, to start the work on the custom tuning.
During the days of tuning, hours of intensive playing, listening and comparing was undertaken. We tested different playing styles in addition to Rocco's to make sure that the amp would work perfectly - not only for Rocco, but for any player.
Features:
Spectra Comp Most bass players know that integrated compressors generally don’t sound that good. A regular compressor tends to be dominated by the lower string, but Staccato’51’s SpectraComp technology allows virtually ‘per string’ multi-band compression that evens out the compression across all strings - Rocco adds a touch of SpectraComp to tighten up his sound.
TubeTone TubeTone emulates all characteristics of both the pre- and power amp to deliver the full tube sound. Staccato’51 uses the TubeTone to add the growl when really diggin’ in
3 User memories With the 3 user memories on Staccato’51 you can easily switch between, for example, your 4- and 5-string basses, or your Rickenbacker and Fender Jazz. Just plug in the bass and use the ‘car radio’ style Store & Recall controls to easily switch between presets for your chosen model. There are three bass user memories, remote control access and visual indicators for Mute and the Tuner.
Flexible Tone Control Straightforward controls like you know them from traditional amps. Voiced by Rocco Prestia to give you plenty of tweaking options to nail the perfect tone. Chances are that this is all you’re ever gonna need. However, if you’re not fully satisfied with the default frequency areas, fear not - we’ve given you the option to change the focus of each of the tone controls through semi parametric access through the shift button.
Integrated Chromatic Tuner Every bass player needs a tuner, so we built one in. Simply push the Mute button to get a full-resolution tuner ‘light ring’ to help you tune your instrument quickly and accurately. You can also use the RC4 for easy remote access to this functionality. Once you’ve tried it, you won’t go back….Rocco couldn’t!
Rehearsal input Playing any instrument on its own can be a little dull, but everyone needs time to practice, which is why Staccato’51 has rehearsal inputs. This rear panel-mounted feature allows you to plug in your iPod, phone, computer or MP3 player via the supplied mini-jack to RCA cable adapter for easy practicing and playing along with recorded music.
Balanced output: Transformer Balanced XLR, Pre/Post Pre-amp Max. Output: +0dBu Optimal Load Impedance: 600 Ohm Preamp out: 1/4" Jack, Balanced Output, Max Output Level = +8dBu Power amp in : 1/4" Jack, balanced input, impedance = 10 kOhm, Max Input Level = +8dBu Rehearsal input: RCA, Left/Right input, fits to iPod ® Digital Recording out: Balanced XLR, AES/EBU, (24-bit) Remote connection: 5 pin DIN, Cable with shield Dimensions: 275 x 290 x 66 mm / 10,8” x 11,4” x 2,6” Weight: 4kg / 8.8 pounds Finish: Sandblasted die-cast aluminum
Mongoose can replace the analog mic and line level portions of an audio system with digitized audio over regular CAT 5 cable. Mongoose can be used with or without CobraNet.
The Mongoose and its Tracker software work with Rane’s Remotes Audio Devices (RADs) and your CobraNet network to deliver digital audio to the “last mile” of installations – between the equipment room/rack and remote spaces. OK, it’s not a mile, we lied: it’s actually 150 meters (492 feet to those in Liberia, Myanmar and the USA).
The Mongoose’s 32-by-32 digital audio matrix router receives its first 16 audio channels from up to eight RADs via the eight rear panel 8P8C (RJ-45) Remote Audio Device ports. The second 16 matrix input channels come from two eight-channel CobraNet receive (Rx) Bundles via standard CobraNet Primary and Secondary/backup ports. The 32 matrix router outputs transmit 16 channels to eight RADs and 16 more channels to two CobraNet transmit (Tx) Bundles.
A family of RAD models connect to the Mongoose. Each converts analog audio to or from 24-bit, 48 kHz digital audio. Standard CAT 5 cable and termination transport four digital audio channels – two channels in each direction – as well as power, ground and a communications channel via Rane’s proprietary RAD Network. Status indicators are at the RAD signal entry points, at the front and back of the Mongoose, and in Mongoose Tracker software.
Mongoose’s rear panel Ethernet port provides direct or network connection to a computer running Rane’s Mongoose Tracker software. Inexpert users are assured easy network communications with Zeroconf (Link-local/mDNS) and DHCP support. Gone are the days of installers requiring intricate IP knowledge. Yet, facility network managers can configure Mongoose like any other IP network device. The Ethernet port also supports Auto MDI/MDIX which automatically detects and permits either an Ethernet crossover cable (included) or a standard Ethernet cable to be used when directly connecting to a computer.
The NM 1 Network Mic Preamp is a very versatile single channel CobraNet I/O box that finds use in many applications. The NM 1 presents matchless features in a compact, reliable, easy to install and maintain package. It has a single studio-grade microphone input with +48 volt phantom power, and a single amplifier output for connection to an external loudspeaker. The microphone signal can be transmitted over CobraNet and the amplifier input can be driven by any CobraNet audio channel. The NM 1 design is based on the Cirrus Logic CS18101 CobraNet chip and CM-2 reference design including the secondary CobraNet port for data and power supply redundancy (more on this later). It also has logic I/O on a DB-15 connector for reading external switches and driving indicator LEDs so it can be connected to a custom switch panel to implement microphone enable, busy indication and similar functions.
The feature that adds the most versatility to the NM 1 is Power Over Ethernet (PoE). It is fully compliant with the IEEE 802.3af standard as a Powered Device (PD). This means it’s powered through the CAT 5 cable that connects it to an Ethernet switch. Of course, the Ethernet switch used must comply with the 802.3af standard as Power Supply Equipment (PSE); these switches are available from all the major Ethernet equipment manufacturers.
Think about this: the NM 1 does not need a power outlet close by. No AC line voltage wiring. No additional electrical box in the wall. No wall wart, batteries, solar panel, pedals or water wheel. The only wiring needed is the data cable back to the Ethernet switch and the local analog audio, making planning and installation much easier and much less expensive.
Rane’s engineers have designed the NM 1 to have a redundant PoE supply along with the redundant data connections. It can be powered from either Ethernet port independent of which one carries the CobraNet data, so an independently redundant power system comes along with the redundant data system with no redundant effort. Switching between ports for PoE is almost seamless and audio interruption is minimal. If the primary port fails the secondary port takes over very quickly, typically 10 ms, without loss of programmed settings.
All parameters are controllable via standard SNMP messages including microphone gain and muting, amplifier output level and muting, and CobraNet Audio channel and Bundle assignments. Four exterior switches assign the NM 1 MIB’s SysName variable to uniquely identify each unit on the network.
All NM 1 audio and data connectors are metal with locking, annular-ring shields for maximum durability, security and immunity from electromagnetic interference, be it radio, static or legislative hot airwaves. And all this fits in a very rugged (but good looking) extruded aluminum box.
Features: Power Over Ethernet (PoE) with built-in redundancy. Audio over CobraNet. Dual CobraNet interfaces for hardware redundancy. Microphone Preamp with 48V phantom power. SNMP Software-controlled preamp gain. Audio power amplifier for monitor loudspeaker. Switch/LED interface. Switch selectable address. All connectors are metal with captive features. UL/cUL/CE listed.
SysName Switches On the rear panel are four rotary switches used to create a four digit identifier that becomes part of the SNMP variable, sysName.
sysName is then used to uniquely identify a CobraNet device on the network. The condition of being unique requires that each device on the network have a different setting. Looking at the unit with the switches facing you, the identifier reads from left to right.
Thus, setting the switches to 1, A, 3, 7, respectively, sets the sysname variable to “NM1-Sw1A37.”
The SC28 is a 2 input, 8 output digital system controller containing pre-programmed tunings for QSC loudspeaker systems (see list of supported loudspeakers at right). In addition to the preset tunings, the SC28 also offers user-adjustable equalization and delay.
Audio inputs and outputs are via balanced, line-level, analog XLR connectors. Four outputs for each of the two inputs are provided for use with 2 or 3-way systems plus subwoofers. Simple to operate yet uncompromised in audio quality, the SC28 uses 48 kHz, 24-bit A/D and D/A conversion with 32-bit, floating point DSP for wide dynamic range and low distortion. Advanced DSP is capable of implementing tunings that incorporate IIR (Infinite Impulse Response) as well as FIR (Finite Impulse Response) filters.
QSC system engineers have employed the power of the SC28's FIR (finite impulse response) filters to implement "Intrinsic Correction™" of the loudspeaker. The goal of Intrinsic Correction™ is to provide a drive signal to the amplifiers and loudspeakers that will result in the most accurate acoustical magnitude, frequency and phase domain performance possible. When properly implemented, Intrinsic Correction™can compensate for many causal phenomena such as the effects of waveguide acoustical impedance and loudspeaker cone resonance. The result is a loudspeaker system with excellent power response and extremely natural, uncolored sound across its frequency band and coverage area. An array of loudspeakers with Intrinsic Correction™ tunings will also be very responsive to the use of equalization to compensate for array configuration, acoustical conditions or user preferences.
Selection of tunings is accomplished by simply scrolling through a list of QSC loudspeakers and selecting the configuration on the LCD panel. A similar process is used to select the QSC amplifiers being used and to configure amplifier input sensitivity for proper dynamic protection gain structure. Settings for new QSC products may be loaded into the SC28 via a rear-mounted USB port.
Once the processor settings have been matched to the system, the user or installer can take advantage of an integral, 6 band parametric equalizer, high and low shelving filters and signal delay to optimize for acoustic, environmental or aesthetic considerations. Password protection is included to prevent unauthorized tampering. The SC28 is easy to use and delivers superb audio performance at a very affordable price.
The VSS3-reverb technology inherent in the M3000 is an industry standard. The M3000 has some of the best sounding, most versatile and easy-to-use professional reverbs for music and film/post applications. Combining ultimate control of directivity in the early reflections with a transparent and harmonically magnificent diffused field, the art of reverberation is brought to a new and higher level.
The M3000 enables you to add the softest and cleanest ambience to your work that you have ever heard. Furthermore the processor boasts a wide range of tools and presets to give you the exact vintage or post production reverb you are looking for.
With Dual-engine structure enabling two simultaneous reverbs or effects, award winning intuitive user-interface, 600 factory presets and multiple I/O formats, the M3000 is one of the best reverb choices for high-end music- and film/post-production.
Reverbs for Music M3000 contains the VSS-3 and VSS-3 Gate reverb algorithms, which are specifically designed for music production applications.
Since the mid-eighties a lot of effort in the high-end effects companies have gone into reverb technology. However, these technologies have all been built upon variations over the same theme, and thus only marginal improvements were created. Some of the main complaints relating to existing reverbs, are that they have unrealistic imaging, sound too metallic and have a tendency to impose unwanted modulation on the signal. With VSS™3 these drawbacks all belong to the past.
The main sonic advantages of VSS™3 are:
Ultimate realism Combining a perfectly smooth diffuse field with detailed Early Reflections, VSS™3 is the ultimate reverb technology in the search of sonic realism.
Accurate early reflections By employing a large number of directional taps - between 40 and 100 - VSS™3 is capable of simulating the Sonics of actual rooms with an unprecedented precision.
Smooth diffuse fields VSS™3 gives you the highest degree of smoothness and clarity to the late reflections - diffused field.
Pitch-accurate The gentle signal processing of VSS™3 keeps the signal in 100% correct pitch, even when engaging extensive effects processing.
Modulation free VSS™3 is the first reverb technology that has the ability to keep the signal 100% free of sound deteriorating modulation, yet still provides additional modulation as an option that can be added.
All this gives you the ability to experience a completely new approach to reverb while still preserving the virtues of the old style effects.
Features:
The M3000 is a dual-engine reverb-unit, where both software algorithms and hardware performance is optimized to the highest level. This leaves you with a box that is a no compromise solution solving all your reverb requirements - and in the easiest way thinkable.
Presets M3000 holds one of the most versatile sets of application specific factory presets on the reverb-market. 250 Single Engine plus 50 Combined dedicated Music production presets, and 250 Single Engine plus 50 Combined Film&Post-production presets, gives you a total of 600 factory presets to choose from.
All of the presets are made by industry professionals and they are ready to use out of the box. Using the M3000 presets is easy: Select a preset, adjust Decay, Predelay and perhaps the Hi-color and you have the desired reverb sound.
The latest progression in TC’s Multi-Award winning mastering technology - the Finalizer 96K - delivers unprecedented levels of clarity, warmth and punch to your mix.
The Finalizer 96K's all new set of advanced features and enhancements, including 96kHz internal and external processing truly puts the world of professional mastering within reach of every studio - large or small.
Inserted between the stereo output of your mixer or workstation and your master recording media, the Finalizer 96K dramatically enhances your material, creating that "radio ready" sound - previously unattainable outside a professional mastering house.
True 24 bit/ 96 khz Performance New features have been added to the Finalizer 96K. It has been updated with new true 24bit resolution A/D and D/A converters. Sample Rate Converter translates AES/EBU, S/PDIF and Tos input to the Finalizer’s internal or external rate. The Finalizer 96K is now able to perform full Up- and Down Sampling, and the Sample Rate Conversion rate is e.g. 32, 44.1, 48, 88.2 and 96kHz.
The ADAT interface enables you to freely choose 2 ADAT channels and direct them to other channels on the ADAT. This enables you to bounce tracks while processing the sound with the Finalizer’s unique features. Finalize your mix and turn it into a masterpiece.
Simple Versatility The Finalizer 96K gives you all the features you need to add the finishing touch to your mix. The Finalizer 96K’s multi-band processor will make your mix sound punchier and crispier. With the Finalizer 96K you can easily turn your demo tapes into finished recordings. You just decide what your recordings should sound like and the Finalizer 96K will make it happen.
If you prefer using an external device and insist on using it, the Finalizer 96K gives you the ability to insert it, using the analog inputs or Digitally via AES/EBU, S/PDIF or Tos-link. The Finalizer 96K also gives you the choice of inserting internal effects, this if useful for you when you e.g. want to refine the overall EQ before compressing the signal.
Analizer Functions The Finalizer 96K features a wide range of extremely useful analyzer functions: Phase Correlation Meter, Level Flow Meter, Peak Hold Meter and Digital I/O Status. Even a Calibration Tone Generator is included in the Finalizer 96K.
Making it Easy The successful Wizard function is naturally a part of the Finalizer 96K and it will immediately calculate an optimal setting for your material. The more experienced user may tweak the signal path extensively with more than 90 parameters to choose from. All of the Finalizer’s functions are easily monitored on the graphic LCD and on the seven precision LED meters. Dithering The use of dithering is very important when interfacing digitally between audio equipment. When resolution is changed from higher to lower, dithering is necessary to minimize distortion. In the Finalizer 96K you have the user friendly TPDF Dithering. The output resolution of the Finalizer 96K is 24bit, when the receiver of the Finalizer signal has a lower resolution, output Dither must be applied to equal the resolution of the receiver. You can Dither from 8-22bits. If your low level signals are in Stereo, Uncorrelated Dither is the best choice, if they are in Mono, Correlated Dither will be the most unobtrusive. If the low level signals are unfocused, Inverse Dither may be your best choice.
Features:
Convert 24bit/96kHz resolution A/D and D/A converters, Format conversion (AES/EBU, S/PDIF and Tos-link) ADAT interface (track bounce)
Sample Rate Conversion Enter and exit at any Sample Rate e.g. 32, 44.1, 48, 88.2 and 96kHz and make even asynchronous conversion possible
Shape Five band 24bit Stereo Digital Equalizer Enhance Choose between Dynamic EQ, Stereo Adjust, Digital Radiance Generator™ and Spectral Stereo Image
Normalize Real-time Gain Maximizer
Expand Variable Slope multiband Expander
Squeeze Multiband Compressor
Trim Variable Ceiling multiband Limiter
Fade Manual or Auto Fade Tool
Dither Maintain the highest resolution at the digital AES/EBU and S/PDIF outputs, Tos-link and ADAT
Balance L/R balance in 0.1dB steps
In short, all the tools you need are provided in the Finalizer 96K and makes it essential in your studio.
When you use the Finalizer 96K you can easily enhance the Dynamics of your mix while you mix. Inserted between the Stereo output of your mixer or DAW and your DAT or CDR, it enhances the dynamics and ensures optimal Analog to Digital conversion.
Digital Inputs and Outputs Connectors: XLR (AES/EBU), RCA Phono (S/PDIF), Optical (Tos-link, ADAT) Formats: AES/EBU (24 bit), S/PDIF (24 bit), EIAJ CP-340, IEC 958, EIAJ Optical (Tos-link), ADAT Lite pipe Processing Delay: 0.2 ms @ 48 kHz, 0.1 ms @ 96 kHz Frequency Response DIO: 10 Hz to 20 kHz: +0/-0.2 dB @ 48 kHz 10 Hz to 45 kHz: +0/-1 dB @ 96 kHz Crosstalk: <-60 dB, 10 Hz to 20 kHz, typical -90 dB @ 1 kHz Max. Output Level: +22 dBu (balanced) Frequency Response 10 Hz to 20 kHz +0/-0.5 dB @ 48 kHz 10 Hz to 45 kHz +0/-3 dB @ 96 kHz EMC Complies with: EN 55103-1, EN 55103-2, Class B of FCC rules/part 15, Class B of CISPR 22 Environment Operating Temperature: 32° F to 122° F (0° C to 50° C) Storage Temperature: -22° F to 167° F (-30° C to 70° C) Humidity: Max. 90 % non-condensing LCD Dimensions: 56 x 128 dot graphic LCD-display 19" x 1.75" x 8.2 (483 x 44 x 208) Weight: 5.2 lb. (2.35 kg) Mains Voltage: 100 to 240 VAC, 50 to 60 Hz (auto-select) Power Consumption: <20 W Backup Battery Life: >10 years Warranty: Parts and Labor 1 year
The DSP 322ua meets the processing needs of today's demanding audio systems. Controlled over an Ethernet network using QSC's Venue Manager software, this newest generation of DSP devices are fully configurable. Storing up to 8 signal flow design configurations with nearly unlimited snapshot parameter recall, the DSP 322ua can fit any application. And with all data stored in the device memory, files can be uploaded from the unit to multiple computers without complications.
Through QSControl.net, QSC's BASIS and next-generation RAVE and DSP products can be networked together and controlled from a single software interface. In addition, multiple networked computers can be set up to control and monitor all of the units simultaneously.
Fixed Latency DSP Users of most other configurable DSP systems are familiar with a variable latency inherent in the processing configuration. Add more processing blocks and you also add delay, whether you want it or not. QSC's DSP engine is unique in having a short and fixed processing latency through the DSP subsystem. When the A/D and D/A converters are included, the total analog-to-analog latency of a single unit is a negligible 2.354 milliseconds. QSC's fixed latency DSP is configurable DSP that stays fast and predictable from one configuration to the next.
INPUTS: Analog: 8 universal mic/line
DSP: 24x24
OUTPUTS: Analog: 8 line level
Features:
Universal inputs – mic/line with pre-amps and phantom power
Configurable DSP functions and signal paths
Fixed latency DSP engine
Ethernet controllable
Each unit can store eight design configurations that can be changed on the fly
Snapshots can recall config or block and/or parameter settings
THX Approved
DSP functions include, but are not limited to: Matrix mixer – any size, up to 24 x 24 Automixers – gain sharing Routers – any size, up to 24 x 24 Gain controls – any channel count, up to 24 Graphic equalizers Filters – high-pass, low-pass, all-pass, shelf, parametric, parametric shelf, Butterworth high and low-pass, Linkwitz-Riley high and low-pass, Bessel-Thomson high and low-pass Crossovers – Linkwitz-Riley, Butterworth, Bessel-Thomson in-phase, Bessel-Thomson symmetrical, 2-way, 3-way, and 4-way general purpose adjustable Compressors, peak limiters, AGC's, gates, dynamics processor Duckers – up to 8 channels, up to 60 seconds fade in and fade out times, priority mix Pink noise, white noise, sine generators Delays Macros – user-definable custom blocks with password protection
The N8000-1500 represents that state of the art for pro audio signal processing. At the heart of the N8000-1500 is the new DSP-2 engine. Composed of three dual-core processors, the DSP-2 expands the total processing power of the N8000-1500 to 1500 MIPS. The N8000-1500 matrix can be configured IN THE FIELD to one of six matrix layouts. The N8000-1500 can be configured and controlled with IRIS-Net.
Features:
Designed for Cinema, Club Sound, Commercial Sound/Life Safety, Concert Sound, House of Worship, Performance & Sports Venues, Stadium Systems, and other applications.
1500 MIPS internal processing
Up to 1900 MIPS of processing power available per unit
Full CobraNet Audio Transport Support
Supports Ethernet, RS-232, USB and CAN Communications Protocols
Up to 1000 MIPS of processing power available per unit
Extensive range of DSP functions
Modular hardware chassis
Integrated supervision, scheduling and auto-compiling DSP
Fully-programmable analog and digital GPIO support
Analog Inputs: Modular Analog Outputs: Modular Audio Network: CobraNet CM-1 Module (optional) Cooling: Left-to-right, 3-stage fan Data Format: 24 Bit linear A/D and D/A conversion, 48 Bit processing Digital Inputs: Modular Digital Outputs: Modular FIR-Drive: Yes GPIO Control Port: 2 x 6-pole Euro block4 Control Inputs (analog 0 - 10 V / logic control) 3 Control Outputs (Relay contact to ground) 1 Fault Output (NC Relay contact)3 Reference Outputs (+5 V / +10 V/ GND) Internal Processing: 2 DSPs Standard (150 MHz, 300 MIPS), DSP-2 Extension Module (3 dual core processors-1200 MIPS total) , 1 DSP per Audio Module (100 MHz, 100 MIPS) 1500 MIPS total without I/O cards Network Control (IRIS-Net): Yes Network Interface: Ethernet-10/100 MBit/s, RJ-45 Power Consumption: 90 W max. (incl. 2 x AI-1, 2 x AO-1, 1 x CM-1 modules) Power Supply: 100 - 240 VAC, 50/60 Hz Sample Rate: 48 kHz internal Serial Interface: 2 Ports, 9pin D-Sub female (Remote Control) Signal-to-Noise Ratio (A-weighted): 115 dB Total Harmonic Distortion: 0.01 % Height: 2RU 88.1 mm (3.47") Width: 483 mm (19.02") Depth: 381 mm (15") Weight Net: 7.37 kg (16.25 lbs)
The BASIS platform meets the control, monitoring, signal transport and processing needs of amplification and loudspeaker systems over an Ethernet network. The BASIS 922dz units combine three distinct QSC technologies within a single hardware unit. Amplifier and loudspeaker control, monitoring and protection, configurable DSP, and CobraNet audio transport are seamlessly integrated into one powerful single RU package.
Through QSControl.net, QSC's BASIS and next-generation RAVE and DSP products can be networked together and controlled from a single software interface. In addition, multiple networked computers can be set up to control and monitor all of the units simultaneously.
Fixed Latency DSP Users of most other configurable DSP systems are familiar with a variable latency inherent in the processing configuration. Add more processing blocks and you also add delay, whether you want it or not. QSC's DSP engine is unique in having a short and fixed processing latency through the DSP subsystem. When the A/D and D/A converters are included, the total analog-to-analog latency of a single unit is a negligible 2.167 milliseconds. QSC's fixed latency DSP is configurable DSP that stays fast and predictable from one configuration to the next.
INPUTS: AES/EBU: 8 digital CobraNet: 16 of 32
DSP: 24x24
OUTPUTS: DataPort: 4(8 channels) CobraNet: 32
Features:
Amplifier and loudspeaker control, monitoring and protection
Configurable DSP functions and signal paths
Fixed latency DSP engine
Ethernet controllable
CobraNet audio transport with new intuitive GUI
Two Ethernet ports – CobraNet and control can be run over a single cable or be divided between the two ports. The CobraNet port is 100Base-T. The control port is 10Base-T.
Each unit can store eight design configurations that can be changed on the fly
Snapshots can recall config or block and/or parameter settings
THX approved
DSP functions include, but are not limited to: Matrix mixer – any size, up to 24 x 24 Automixers – gain sharing Routers – any size, up to 24 x 24 Gain controls – any channel count, up to 24 Graphic equalizers Filters – high-pass, low-pass, all-pass, shelf, parametric, parametric shelf, Butterworth high and low-pass, Linkwitz-Riley high and low-pass, Bessel-Thomson high and low-pass Crossovers – Linkwitz-Riley, Butterworth, Bessel-Thomson in-phase, Bessel-Thomson symmetrical, 2-way, 3-way, and 4-way general purpose adjustable Compressors, peak limiters, AGC's, gates, dynamics processor Duckers – up to 8 channels, up to 60 seconds fade in and fade out times, priority mix Pink noise, white noise, sine generators Delays Macros – user-definable custom blocks with password protection
The BASIS platform meets the control, monitoring, signal transport and processing needs of amplification and loudspeaker systems over an Ethernet network. The BASIS 922az units combine three distinct QSC technologies within a single hardware unit. Amplifier and loudspeaker control, monitoring and protection, configurable DSP, and CobraNet audio transport are seamlessly integrated into one powerful single RU package.
Through QSControl.net, QSC's BASIS and next-generation RAVE and DSP products can be networked together and controlled from a single software interface. In addition, multiple networked computers can be set up to control and monitor all of the units simultaneously.
Fixed Latency DSP Users of most other configurable DSP systems are familiar with a variable latency inherent in the processing configuration. Add more processing blocks and you also add delay, whether you want it or not. QSC's DSP engine is unique in having a short and fixed processing latency through the DSP subsystem. When the A/D and D/A converters are included, the total analog-to-analog latency of a single unit is a negligible 2.354 milliseconds. QSC's fixed latency DSP is configurable DSP that stays fast and predictable from one configuration to the next.
INPUTS: Analog: 8 line level CobraNet: 16 of 32
DSP: 24x24
OUTPUTS: DataPort: 8(16 channels) CobraNet: 32
Features:
Amplifier and loudspeaker control, monitoring and protection
Configurable DSP functions and signal paths
Fixed latency DSP engine
Ethernet controllable
CobraNet audio transport with new intuitive GUI
Two Ethernet ports – CobraNet and control can be run over a single cable or be divided between the two ports. The CobraNet port is 100Base-T. The control port is 10Base-T.
Each unit can store eight design configurations that can be changed on the fly
Snapshots can recall config or block and/or parameter settings
THX approved
DSP functions include, but are not limited to: Matrix mixer – any size, up to 24 x 24 Automixers – gain sharing Routers – any size, up to 24 x 24 Gain controls – any channel count, up to 24 Graphic equalizers Filters – high-pass, low-pass, all-pass, shelf, parametric, parametric shelf, Butterworth high and low-pass, Linkwitz-Riley high and low-pass, Bessel-Thomson high and low-pass Crossovers – Linkwitz-Riley, Butterworth, Bessel-Thomson in-phase, Bessel-Thomson symmetrical, 2-way, 3-way, and 4-way general purpose adjustable Compressors, peak limiters, AGC's, gates, dynamics processor Duckers – up to 8 channels, up to 60 seconds fade in and fade out times, priority mix Pink noise, white noise, sine generators Delays Macros – user-definable custom blocks with password protection
The BASIS platform meets the control, monitoring, signal transport and processing needs of amplification and loudspeaker systems over an Ethernet network. The BASIS 922uz units combine three distinct QSC technologies within a single hardware unit. Amplifier and loudspeaker control, monitoring and protection, configurable DSP, and CobraNet audio transport are seamlessly integrated into one powerful single RU package.
Through QSControl.net, QSC's BASIS and next-generation RAVE and DSP products can be networked together and controlled from a single software interface. In addition, multiple networked computers can be set up to control and monitor all of the units simultaneously.
Fixed Latency DSP Users of most other configurable DSP systems are familiar with a variable latency inherent in the processing configuration. Add more processing blocks and you also add delay, whether you want it or not. QSC's DSP engine is unique in having a short and fixed processing latency through the DSP subsystem. When the A/D and D/A converters are included, the total analog-to-analog latency of a single unit is a negligible 2.354 milliseconds. QSC's fixed latency DSP is configurable DSP that stays fast and predictable from one configuration to the next.
INPUTS: Analog: 8 universal mic/line CobraNet: 16 of 32
DSP: 24x24
OUTPUTS: DataPort: 4(8 channels) CobraNet: 32
Features:
Universal inputs – mic/line with pre-amps and phantom power
Amplifier and loudspeaker control, monitoring and protection
Configurable DSP functions and signal paths
Fixed latency DSP engine
Ethernet controllable
CobraNet audio transport with new intuitive GUI
Two Ethernet ports – CobraNet and control can be run over a single cable or be divided between the two ports. The CobraNet port is 100Base-T. The control port is 10Base-T.
Each unit can store eight design configurations that can be changed on the fly
Snapshots can recall config or block and/or parameter settings
THX approved
DSP functions include, but are not limited to: Matrix mixer – any size, up to 24 x 24 Automixers – gain sharing Routers – any size, up to 24 x 24 Gain controls – any channel count, up to 24 Graphic equalizers Filters – high-pass, low-pass, all-pass, shelf, parametric, parametric shelf, Butterworth high and low-pass, Linkwitz-Riley high and low-pass, Bessel-Thomson high and low-pass Crossovers – Linkwitz-Riley, Butterworth, Bessel-Thomson in-phase, Bessel-Thomson symmetrical, 2-way, 3-way, and 4-way general purpose adjustable Compressors, peak limiters, AGC's, gates, dynamics processor Duckers – up to 8 channels, up to 60 seconds fade in and fade out times, priority mix Pink noise, white noise, sine generators Delays Macros – user-definable custom blocks with password protection
The BASIS platform meets the control, monitoring and processing needs of amplification and loudspeaker systems over an Ethernet network. The BASIS 914lz units combine three distinct QSC technologies within a single hardware unit. Amplifier and loudspeaker control, monitoring and protection, configurable DSP, and CobraNet audio transport are seamlessly integrated into one powerful single RU package.
Through QSControl.net, QSC's BASIS and next-generation RAVE and DSP products can be networked together and controlled from a single software interface. In addition, multiple networked computers can be set up to control and monitor all of the units simultaneously.
Fixed Latency DSP Users of most other configurable DSP systems are familiar with a variable latency inherent in the processing configuration. Add more processing blocks and you also add delay, whether you want it or not. QSC's DSP engine is unique in having a short and fixed processing latency through the DSP subsystem. When the A/D and D/A converters are included, the total analog-to-analog latency of a single unit is a negligible 2.354 milliseconds. QSC's fixed latency DSP is configurable DSP that stays fast and predictable from one configuration to the next.
INPUTS: Analog: 4 XLR line level CobraNet: 16 of 32
DSP: 24x24
OUTPUTS: DataPort: 8(16 channels) CobraNet: 32
Features:
Amplifier and loudspeaker control, monitoring and protection
Configurable DSP functions and signal paths
Fixed latency DSP engine
Ethernet controllable
CobraNet audio transport with new intuitive GUI
Two Ethernet ports – CobraNet and control can be run over a single cable or be divided between the two ports. The CobraNet port is 100Base-T. The control port is 10Base-T.
Each unit can store eight design configurations that can be changed on the fly
Snapshots can recall config or block and/or parameter settings
THX approved
DSP functions include, but are not limited to: Matrix mixer – any size, up to 24 x 24 Automixers – gain sharing Routers – any size, up to 24 x 24 Gain controls – any channel count, up to 24 Graphic equalizers Filters – high-pass, low-pass, all-pass, shelf, parametric, parametric shelf, Butterworth high and low-pass, Linkwitz-Riley high and low-pass, Bessel-Thomson high and low-pass Crossovers – Linkwitz-Riley, Butterworth, Bessel-Thomson in-phase, Bessel-Thomson symmetrical, 2-way, 3-way, and 4-way general purpose adjustable Compressors, peak limiters, AGC's, gates, dynamics processor Duckers – up to 8 channels, up to 60 seconds fade in and fade out times, priority mix Pink noise, white noise, sine generators Delays Macros – user-definable custom blocks with password protection
The BASIS platform meets the control, monitoring and processing needs of amplification and loudspeaker systems over an Ethernet network. The BASIS 904zz units combine three distinct QSC technologies within a single hardware unit. Amplifier and loudspeaker control, monitoring and protection, configurable DSP, and CobraNet audio transport are seamlessly integrated into one powerful single RU package.
Through QSControl.net, QSC's BASIS and next-generation RAVE and DSP products can be networked together and controlled from a single software interface. In addition, multiple networked computers can be set up to control and monitor all of the units simultaneously.
Fixed Latency DSP Users of most other configurable DSP systems are familiar with a variable latency inherent in the processing configuration. Add more processing blocks and you also add delay, whether you want it or not. QSC's DSP engine is unique in having a short and fixed processing latency through the DSP subsystem. QSC's fixed latency DSP is configurable DSP that stays fast and predictable from one configuration to the next.
INPUTS: CobraNet: 24 of 32
DSP: 24x24
OUTPUTS: DataPort: 8(16 channels) CobraNet: 32
Features:
Amplifier and loudspeaker control, monitoring and protection
Configurable DSP functions and signal paths
Fixed latency DSP engine
Ethernet controllable
CobraNet audio transport with new intuitive GUI
Two Ethernet ports – CobraNet and control can be run over a single cable or be divided between the two ports. The CobraNet port is 100Base-T. The control port is 10Base-T.
Each unit can store eight design configurations that can be changed on the fly
Snapshots can recall config or block and/or parameter settings
THX approved
DSP functions include, but are not limited to: Matrix mixer – any size, up to 24 x 24 Automixers – gain sharing Routers – any size, up to 24 x 24 Gain controls – any channel count, up to 24 Graphic equalizers Filters – high-pass, low-pass, all-pass, shelf, parametric, parametric shelf, Butterworth high and low-pass, Linkwitz-Riley high and low-pass, Bessel-Thomson high and low-pass Crossovers – Linkwitz-Riley, Butterworth, Bessel-Thomson in-phase, Bessel-Thomson symmetrical, 2-way, 3-way, and 4-way general purpose adjustable Compressors, peak limiters, AGC's, gates, dynamics processor Duckers – up to 8 channels, up to 60 seconds fade in and fade out times, priority mix Pink noise, white noise, sine generators Delays Macros – user-definable custom blocks with password protection