Manual
Table Of Contents
- Overview
- Chapter 1: Introduction
- Chapter 2: PRO Series Live Audio Systems
- Chapter 3: About The Control Centre
- Getting Started
- Chapter 4: Setting Up The System
- Basic Operation Of The PRO Series
- Chapter 5: Before You Start
- Chapter 6: Working With The Control Centre
- Chapter 7: Navigation
- Chapter 8: Patching
- Chapter 9: Basic Operation
- Setting a mic amplifier’s input gain
- Setting the high and low pass filters
- Input equalisation (E zone)
- Input dynamics processing (D zone)
- Output processing
- Using VCA/POP groups
- Setting up a mix
- Using fader flip
- Setting up the effects rack
- Simple routing to master stereo outputs
- Scene and show management (automation)
- Configuring the inputs and outputs
- Using copy and paste
- User library (presets)
- Surround panning
- Two-man operation
- Saving your show files to a USB memory stick
- Security (locking mode)
- Security (locking mode)
- Advanced Operation And Features
- Chapter 10: Stereo Linking
- Chapter 11: Panning
- Chapter 12: Soloing
- Chapter 13: Muting
- Chapter 14: Monitors And Communications
- Chapter 15: Graphic Equaliser (GEQ)
- Chapter 16: Internal Effects
- Overview of the internal effects
- About the effect window
- Working with the effects
- Effect configuration
- Effect programs
- Delay effect
- Virtual DN780 Reverb effect
- Flanger effect
- Phaser effect
- Pitch Shifter effect
- SQ1 Dynamics effect
- 3-Band Compressor effect
- Submonster
- DN60 Spectrum Analyser
- Tape Saturation
- Variable Phase
- Dual Stereo Delay
- Ambience Reverb
- Vintage Room Reverb
- Chamber Reverb
- Hall Reverb
- Plate Reverb
- Stereo Graphic EQ
- Dynamic EQ
- Matrix Mixer
- Stereo Chorus
- UNCL.D
- Loudspeaker Processor
- De-esser
- TC M350
- MIDAS Spectrum Analyser
- MIDAS Automixer
- Chapter 17: Control Groups
- Chapter 18: Copy And Paste
- Chapter 19: Assignable Controls (I Zone)
- Chapter 20: Scenes And Shows (Automation)
- About automation
- Automation controls
- Automation screen
- Using the right-click menu
- Scene contents
- Point scenes
- Numbering and navigation
- Global scene
- Initial snapshot scene (safe scene)
- Date and time
- Scene cue list
- Editing scene properties
- Adding a new scene
- Copying and deleting scenes
- Changing the order of the scenes
- Overriding store scope
- Using patching in automation
- Using zoom
- Show files
- Rehearsals
- Safes
- Chapter 21: Scope (Automation)
- Chapter 22: Events (Automation)
- Chapter 23: Crossfades (Automation)
- Chapter 24: User Libraries (Presets)
- Chapter 25: File Management
- Chapter 26: Using Other Devices With The PRO X
- Chapter 27: Changing The User Settings
- Setting the meter preferences
- Configuring a virtual soundcheck
- Restoring the PRO X defaults
- Checking the PRO X build information
- Setting the configuration preferences
- Changing the user interface preferences
- Configuring the channels, groups and internal units
- Changing the default input/output names
- Adjusting PRO X illumination
- Setting the time and date
- Chapter 28: Delay Compensation (Latency)
- Description
- Chapter 29: Panel Connections
- Chapter 30: Inputs
- Mic amp input gain (preliminary input processing)
- Chapter 31: Outputs
- Chapter 32: GUI Menu
- Appendices
- Appendix A: Application Notes
- Appendix B: Technical Specification
- Appendix C: KLARK TEKNIK DN370 GEQ
- Appendix D: KLARK TEKNIK DN780 Reverb
- Technical Specifications
- Appendix E: I/O Modules
- DL443 analogue Jack I/O module
- Appendix F: Replacing A Module
- Appendix G: Troubleshooting
- Appendix H: Updating PRO X Host Software
- Appendix I: Documentation
- Appendix K: Parameters Affected By Scope
- Appendix L: Parameters Affected By Automate Patching
- Appendix M: Parameters Protected By Safes
- Appendix N: Parameters Affected By Copy And Paste
- Appendix O: Parameters Affected By Stereo Linking
- Appendix P: Parameters Copied Through Scenes
- Appendix Q: Service Information
105 PRO X User Manual
work like two limiters in series with the PS (Passive Split) button deactivated.
A crossover parameter is provided for controlling the dynamics of the signal in
two dB regions.
Finally, the brick wall limiter gives a control over the look-ahead size to be
processed. The latency introduced by the look-ahead is automatically calculated
in the delay section, and updates the display in that section.
De-esser
The de-esser is designed to reduce sibilance in human voices, such as excessive
presence of s or f sounds.
Due to the special sibilant detection algorithm, the de-esser is completely input
level independent. This means that the reduction of sibilants will not change if
the input level is changed. For example, changing the microphone amplier gain
does not aect the detector.
Although the de-esser is primarily designed to be applied on human voices, it can
also be used creatively on other instruments.
• crossover knob Part of the detection algorithm is a matched lowpass/
highpass crossover lter. The crossover frequency is adjusted by the
crossover knob. When a normal, non sibilant, sound is present in the input
the energy will be mostly focused in the lower section on the frequency
spectrum. On the other hand, when a sibilant sound is present most of the
energy will be present in the higher section of the frequency spectrum.
Therefore, the crossover knob is useful for tuning the detector so that the
de-essing mechanism is only triggered by high-frequency energy content,
as it would occur with an s sound.
The optimal crossover frequency value is indicated by the high pass lter being lit
blue only when a sibilant sound is present.
• de-esing knob When a sibilant is successfully detected the amount
of reduction can be adjusted by using the de-essing knob. At minimum
position, the reduction is 0 and equivalent to bypass, i.e. the audio is not
aected. As the user turns the knob clockwise, some MIDAS magic will
reduce the amount of sibilants.
• Enable listen In order to prevent accidental pressing of any of the listen
buttons during a live performance, the Enable Listening button must be
activated before any of the listen buttons can be pressed.
• Listen button When the listen button is pressed (after switching Enable
listen on) the signal from the high pass lter is routed directly to the output
of the de-esser, enabling ne-tuning of the crossover. Note that this happens
in place, i.e. the main output is ltered, therefore the user should be very
careful when using it.
• Ch name The user can add a specic name to each channel of the de-esser.
This will be completely scene and preset recallable.
• Bypass When pressed the input signal is unaected.
• Assignable control pages The de-esser has two dierent assignable
control pages. The rst one allows to quickly access the de-essing knobs
and bypass buttons, whereas the second page provides access to the
crossover knobs
Controls:
• Bypass
• De-essing
• Listen
• x-over
• Enable
listen
Smart Dynamics Processor
The Smart Dynamic Processor is the rst example of a new generation of Fxs
based on adaptive signal processing techniques which are able to simplify the
user workow, without compromising sound quality and versatility.
In particular, the Smart Dynamics is an intelligent dynamic range processor based
on the MIDAS Channel Compressor and is therefore is characterized by the same
unique sound signature. It can operate in two dierent processing modes: smart
transient mode or smart loudness mode.
The Smart Transient mode is designed to work on the envelope of transient
sounds, like drums or slap bass, and modify its dierent components in order to
achieve the desired sonic result.
When operating in this mode, 4 dierent Semantic Controls associated with the
desired sonic result are available:
• the Snap control, insert space which can be used to amplify or to reduce the
attack component of a transient signal;
• the Sustain control, which can be used to amplify or to reduce the sustain
component;
• the Balance control allows the blending of the dry signal with processed signal
in order to get either a more natural or enhanced sound;
• the Output level controls can be used to correct the nal output level and
avoid clipping.
The Smart Loudness mode is instead designed to increase naturally the loudness
of any type of sound, without introducing distortion. This is achieved through a
combination of intelligent upward compression and adaptive limiting.
When operating in this mode, the overall perceived loudness of the track can
be increased by operating the Loudness control, The output level can then be
adjusted to avoid clipping.
Meters:
• Gain reduction
• Output level
• Spectral balance