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User manual XTA SIDD - PC SOFTWARE V2
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User guide XTA SIDD - PC SOFTWARE V2
Detailed instructions for use are in the User's Guide. Contents
CONTENTS THANKS INTRODUCTION INSTALLATION STARTING THE SOFTWARE CONNECTING UNITS TO THE COMPUTER
Using Midi... Using RS232... Using RS485...
2 4 4 5 7 8
8 8 8
TYPICAL INTERFACE SET-UPS
RS232 Connection (Single Unit) RS232 Connection (Multiple Units) RS485 Connection Midi Connection
9
9 9 10 10
SETTING UP COMMUNICATIONS AND GOING ON-LINE BREAKDOWN OF THE MAIN SCREEN. MENU BAR AND TOOL BAR SUMMARY GUIDES SYSTEM BAR AND SYSTEM TOOLBAR FUNCTION MODES EXPERT/NORMAL/BASIC... MODIFIYING THE USER VIEWS CHANGING THE SIZE OF THE EDIT WINDOWS ANALOGUE METERING MODE MODIFYING THE UNIT'S CONFIGURATION
Configuring the Computer and Software Going `On-line'
11 13 14 15 16 19 20 21 22
11 12
WHAT IS DYNAMIC EQ? HOW DO THE DYNAMIC EQ CONTROLS INTERACT? INDICATION OF CLIPPING THE INPUT TO A MODULE EDITING PARAMETERS ON THE DYNAMIC EQ WINDOW NOISE GATE KNOW-HOW
Attack and Release Times. Setting the Range Correctly. Using the Hold Time. What does the edit window show? What does the bouncing ball show?
What is the Difference Between Ganging and Linking? Choosing the Configuration Input/Output Gains Main Delay Parametric Filters, High and Low Pass Filters. Harmonics Generator and ADT/Aux Delay Controls.
30 34
23 25 26 27 27 29
40 41 42
42 44 44
34 39
HOW DO THE NOISE GATE CONTROLS INTERACT? SIDECHAIN EQUALISATION HOW AND WHEN TO USE IT
When would Sidechain EQ be useful? De-essing and de-popping. Maximising loudness without dulling the mix. Suppressing feedback without adding notch filters. Picking out instruments for gating. How would sidechain EQ normally be implemented? What does the edit window show? What does the bouncing ball show?
46 48
46 47
LOOK AHEAD DELAY PRE-EMPTIVE ACTION COMPRESSION WHAT IT IS AND WHAT IT CAN DO
What does a Compressor do? Why are Compressors necessary? How does a Compressor work? What makes the XTA Compressor different?
52 55
55 55 56 57
48 48 48 49 50 51
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SiDD PC Software Version 2.00
EDITING & INTERACTION OF THE COMPRESSOR PARAMETERS
Make-up Gain. Setting the Attack and Release times. The Compressor `Knee' Control.
58
58 60 60
THE EXPANDER MODULE THE COMPLIMENTARY PROCESS
What does an Expander do? Why are Expanders necessary? How does a Expander work? What is the difference between an Expander and a Noise Gate? Setting the Attack and Release times. Main Limiting Methods.
62
62 62 63 65
EDITING & INTERACTION OF THE EXPANDER PARAMETERS LIMITERS USE (AND ABUSE)
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66 67
67 69 70 71
70
EDITING THE LIMITER MEMORY HANDLING AND FEATURES INDEX
Accessing Memories
SiDD PC Software Version 2.00
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Thanks
Thank you for choosing the XTA for Windows TM software.
Please spend a little time reading this manual, so that you obtain the best possible performance from your purchase. All XTA products are carefully designed and engineered for cutting-edge performance and world-class reliability. If you would like further information about this or any other XTA product, please contact us. We look forward to hearing from you in the near future.
Introduction
This manual covers the installation and use of XTA's proprietary software package for the DP324 processor, or SiDD. This software is a remote control application running under Microsoft Windows TM , and is currently compatible with DP324's only. The software is designed to allow all these products to be connected to a computer through a variety of interfaces (RS232/485 or Midi) and will permit full remote control of up to 32 devices. Please note that SiDD processors may co-exist on the same network as other XTA products, such as the DP224/6 Loudspeaker Management Systems, or DP200 Digital Equaliser/Processors. Our other PC control package, Audiocore, may be used to control these devices. The only caveat is that the total number of devices connected must not exceed 32. Audiocore will ignore connected SiDDs, and vice versa. For the latest information about XTA's software and product range, visit the website at www.xta.co.uk.
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SiDD PC Software Version 2.00
Installation
Please note that this software will not run under WindowsTM 3.1 or 3.11. It is designed for WindowsTM 95/98/NT4. Follow these steps to install your copy of for Windows TM .
Insert the disk supplied into floppy disk drive A on your computer. It is recommended that you close all other applications before attempting to install this software. This is true during all software installation, and is not peculiar to this package. Click on the `Start' button on the toolbar, and select `Run' with another click.. Type `a:\setup' in the text box as shown, and then click `OK'.
After a couple of seconds the installation will begin. Follow the instructions that appear on the screen.
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Alternatively, open the control panel by clicking on the `Start' button, selecting `Settings' and then `Control panel'.
From this window, select the `Add/Remove Programs' icon, and double click on it, opening the window shown below.
Click on the button marked `Install...' which will start the software installation wizard. Follow the instructions to install the software.
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Starting the Software
The installation procedure will have created an entry in the `Programs' list off the `Start' menu. This will be called `AudioCore'. Clicking on this followed by the `SiDD PC' that appears will start the software. Each time the software starts no attempt will be made to connect to the unit(s). This should be initiated by the user. Connecting units to the computer is covered in the next section. It's worth noting, at this point, the major difference between this software package and the majority of remote control applications available in the professional sound market. The SiDD processor and software were developed concurrently and designed to offer a complete solution to equalisation and dynamics control. To this end, neither the software, nor the unit(s) assumes responsibility as a master when the two are working together. This means that any changes via the PC will immediately update the relevant unit, and any changes to the unit via the front panel will be immediately reflected on the PC. Changes made to a unit when the software is `Off-line' or the PC is not connected will be automatically uploaded when the computer is next connected. In this way, the computer and units always remain synchronised, and there is no possibility of settings stored in one overwriting settings in the other.
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Connecting Units to the Computer
Three types of remote interface may be used to connect units to the computer. These are outlined below. Using Midi... To output Midi information from the computer, it must be fitted with suitable Midi device and driver software appropriate to that device. A Midiman `Portman PC/P' can be used to convert the parallel port to a Midi port, if there are not enough spare Com (serial) ports available, or if the computer is a laptop. If this device is used, be sure to select the driver with the `w/LEDs' extension in the `Setup' menu, and ensure that the `Enable Device' box is checked. If the Midi device is internal, such as a Soundblaster sound card, an adapter cable is required to `break-out' the Midi connections from the 25-way D-connector on the card. Note that the maximum recommended cable run for Midi is 15 metres. Using RS232... RS232 uses a serial port connection from the computer (normally a 9-pin Dconnector for Com1 ports) and allows remote control of one unit directly. In the case of the DP226, additional units may be cascaded from the RS485 connectors on the rear of the first unit. This may be a pair of XLR connectors, or RJ45 (telephone jack) connectors. Typical set-ups are included in the section following this. Using RS485... To output RS485 from a computer requires either a converter to connect to the serial (RS232) port, available from XTA, or a plug in card along the same lines as a Midi or Sound-card. The main advantage that RS485 has over RS232 is that the RS485 system is a `multi-drop' system as opposed to a `point-to-point' system, so if one unit fails, all other units will continue to operate as no unit is dependant on the one electrically previous to it in the remote control chain for the relay of data. Additionally, the electrical specification of RS485 permits cable rums of up to 1km before any regeneration is required. The basic specification of RS232 only caters for cable runs up to 25m.
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Typical Interface Set-ups
RS232 Connection (Single Unit) A typical interface set-up might involve running an RS232 link from laptop or a desktop computer to one SIDD set up as a master unit. The diagram below shows this method of connection, the required menu options are also given. Note that the RS232 cable must be a 1-1 connection type, NOT a null modem cable (which has connections crossed internally).
RS232 Connection (Multiple Units)
If control over multiple units is required, typically the slaves will be set up to run from the RS485 ports on the master DP324. Note the incremental `Remote ID Num' option in the unit's menu options.
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RS485 Connection To use RS485 communication directly from a computer, a master DP324 must be configured to receive RS485. You must have a suitable RS485 port on your computer, or a converter connected to the serial port in use. This configuration is shown below, along with the required menu options.
Midi Connection To use MIDI communications, the DP324 must be configured to receive as a master via its MIDI port. You must have a MIDI card or interface connected to your computer. The setup is shown below.
Note that two-way communication is NOT possible via the MIDI interface only download to the unit is possible, and no settings are returned to the connected computer.
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Setting Up Communications and Going On-line
Configuring the Computer and Software After starting the software, choose Remote RS232/RS485 as appropriate from the menu bar. A window will appear containing the configuration settings required for the interface. In the case of RS232, this is as shown here, with selectable options for the `Com' (serial) port to use, the speed of this port and options for improving the robustness of the interface, if the port supports them. `Use Acknowledge Cmd' ensures that every message sent from the software reaches its destination correctly, but at the expense of operational speed. It is recommended to enable this option when long cable runs are used in noisy environments. `Hardware Flow Control' uses additional physical lines to control the traffic on the port, and is only available with the RS232 interface. If the cable being used to connect the units to the computer does not physically have these additional lines connected, do not enable this option as communications will not work correctly. Choose the RS485 selection if you plan to connect to the first unit via it's RS485 XLR socket. The window is virtually the same as that shown above, but without the ability to select `Hardware Flow Control' or `Radio Modem'. Important note about interaction between SiDD and other connected units. Remember that SiDD's and other XTA units can co-exist on the same network, so long as the total number of units connected does not exceed 32. Although Audiocore will ignore any connected SiDDs, they must be set to ID numbers not in use by any other unit. If they share an ID number with another unit, niether unit will apppear on the network in either Audiocore or SiDD's software.
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Going `On-line' If the remote interfaces have been configured correctly at both the units(s) end and the computer/software end, it will now be possible to go `on-line'. This is achieved simply by pressing the On/Off line button on the tool bar. The network will be scanned for any connected units this is accompanied by a message window as below. The ID number will increment from 1 to 32, and the entire scanning process typically takes about 4 seconds with a baud rate of 38400 bits/S and only one unit connected. The selected communications speed will obviously affect this time - set to 2400 baud, this period is extended to typically 16 seconds. As units are located, the `Found' figure will increment. Immediately following the scan, all the units will be interrogated for their settings, and the data collected. This is accompanied by the message... Once this process is completed (again the slower the baud rate, the longer this will take), all the units found will be listed, channel by channel in the unit selection window. This is shown overleaf.
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Breakdown of the Main Screen.
Starting at the top there is the menu bar, and the tool bar. The tool bar consists of... Open file: Open a different file of previously saved information. New file: Start a new file from scratch. Save file: Save the current information with the chosen filename. Cut settings: Delete settings, copying them to the clipboard. Copy settings: Keep current settings, but copies them to the clipboard. Paste settings: Overwrites settings with those currently on the clipboard. Print information: Print out details of current settings. Help: Starts main help system, allowing subjects to be searched and printed. "What's This?" help: Instant pop-up help on any screen control. On/off-line: Initiate search, synchronisation and connection to chain of units. Modify views: Change which windows are displayed in user selectable views. Modify configurations: Change expander/gate operation and stereo linkage. Audiocore: Switch to Audiocore software (if running greyed out if not)1 Test button: Reserved for future use and factory test functions.
1
SiDD PC Software Version 2.00
Audiocore version 5.11 or above
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Menu Bar and Tool Bar Summary Guides
The main menu items are summarised below, along with references where greater explanation is given elsewhere in this manual.
New: Start a new system from scratch Open: Open an existing system configuration Save: Save this system configuration Save As: Save this system configuration with another name Options: Various general configuration options Print: Print the current window Print Preview: Show current window printed page layout Print Setup: Choose printer and format of print Exit: Close the software
Undo: Step back one edit stage. Cut: Delete settings, copying them to the clipboard. Copy: Keep current settings, but copy them to the clipboard. Paste: Overwrites settings with those on the clipboard.
Toolbar: Toggle toolbar below menus. Status Bar: Toggle status bar at bottom of screen. System Bar: Toggle main channel and views toolbar. System Toolbar: Toggle main channel and views dropdown menu version of system bar. User Views: Select view in same way as from System Bar or System Toolbar.
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No Connection: Go `off-line' RS232: Choose & configure RS232 settings. RS485: Choose & configure RS232 settings. Note that in addition to pressing `F1' on your keyboard for help, the `What's This?' help system invoked using provides additional hints and tips. In addition, right clicking over many sections of any window will invoke context sensitive help relating to that item.
System Bar and System Toolbar
Navigating around the various units connected and selecting which edit windows to see at any time is accomplished using the System Bar shown here to the left, or the System Toolbar, shown below.
These both do the same thing, allowing you to select from the connected units, channel by channel, and the module editing windows, as set up using the Modify Views option. For more information on modifying the user views see page 16.
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Function Modes Expert/Normal/Basic...
To simplify editing in high-pressure situations, or to hide certain parameters from user tampering, SIDD has three levels of editing complexity, known as the function modes. These are designated `Expert', `Normal', and `Basic'. The differences between each mode are explained in the tables below. The most comprehensive mode `Expert' gives access to all the editable parameters of a module. `Normal' mode dispenses with the less used features. `Basic' mode offers only a single parameter screen showing only the most important parameter(s). Input EQ. etc. Normal Mode Input/Output Gain Main Delay ADT/Split Delay High Pass Filter Low Pass Filter Parametric Filters Dynamic EQ Normal Mode Threshold Ratio
Expert Mode Input/Output Gain Main Delay ADT/Split Delay Harmonics High Pass Filter Low Pass Filter Parametric Filters x 6
Basic Mode Input/Output Gain Main Delay
High Pass Filter Low Pass Filter Parametric Filters
Expert Mode Threshold Ratio Attack Release Dynamic Filter Operating Mode Expert Mode Threshold Ratio (Range) Attack (Hold) Release Sidechain HPF Sidechain LPF Sidechain PEQ x 2
Basic Mode Threshold
Dynamic Filter
Dynamic Filter
Expander Gate (Gate Mode) Normal Mode Basic Mode Threshold Threshold Ratio (Range) Ratio (Range) Attack (Hold) Release Sidechain HPF Sidechain LPF
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Expert Mode Threshold Ratio Make-up Gain Attack Release Knee Sidechain PEQ x 2
Compressor Normal Mode Threshold Ratio Make-up Gain Attack Release Knee
Basic Mode Threshold Ratio Make-up Gain
Expert Mode Threshold Overshoot Attack Release Sidechain PEQ x 2
Limiter Normal Mode Threshold Overshoot Attack Release
Basic Mode Threshold
The function mode of any module is selectable via a drop-down list at the bottom of each editing window. Examining the noise gate window, the `Expert' mode is shown below. All parameters are accessible as detailed in the table above. Selecting `Normal' will grey out some of the parameters as show overleaf.
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Note that the coloured nodes are missing on the sidechain frequency response graph, indicating that these parameters cannot be edited. Selecting `Basic' limits the available parameters to just the threshold, as shown.
These changes are immediately reflected on the unit as well, with the current function mode for any module shown during editing, on the module summary information screen.
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Modifiying the User Views
Pressing the `Modify Views' button on the tool bar - will display the window shown below. There are eight tabs corresponding to the eight buttons on the System Bar (in addition to the fixed `Channel View' button with show all the edit winows for a particular channel and cannot be edited.) Selecting a tab allows the channel of any unit to be accessed using the left hand pane. Below this is the type of window to be included in the view. So, the process consists of selecting the channel, the window and then pressing the `ADD>>>' button. This will include the chosen window in the right hand pane. To remove a window from the view, first select in from the list of those already included in the right hand pane, and press the `REMOVE' button.
Each view can be given a more meaningful name by typing directly in the `View Name' text box above the included windows list. Pressing `OK' will update the user views and their respective buttons and drop-down list in the case of the System Toolbar. Note that the set of views can be saved by pressing the `Save a View File' button. This permits the views for as many systems as required to be recalled and saved as required.
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Changing the Size of the Edit Windows
Before examining the edit windows in detail, it's worth knowing that each window can be displayed in three different sizes. This reduces clutter on the screen and makes it easier to monitor several different modules or units at the same time. Using the in/out window as an example, it may be displayed as below... `Large'...all parameters are directly editable;
`Medium'...all parameters still available, but filters adjustable only by curve dragging; `Small'...only metering available (with threshold adjustment for module if appropriate).
The size is selected by clicking on the icon in the top left hand corner of the window, as shown below. The basic rule is that the `Medium' size window retains full parameter control at the expense of losing relevant filter details, and `Small' retains appropriate metering information, and still allows the threshold of dynamics modules to be adjusted by dragging the appropriate node on the transfer function graph.
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Analogue Metering Mode
The metering mode for the dynamics modules may be swapped between the `bouncing ball' view with adjustable threshold, and an analogue gain reduction meter mode, as shown below. This example shows the dynamic EQ module's metering (in stereo), with the module active. Double click on the `bouncing ball' view will toggle between this view, and the VU meter mode.
Note that the gain reduction will be shown, even if the module is bypassed. The bypassed state is represented by the meter `backlight' going out, as in the compressor window example below. This meter mode is ideal for setting up a selection of views forming a monitoring panel.
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Modifying the Unit's Configuration
Pressing the `Configure Unit' button on the toolbar of the currently selected unit's configuration to be These include the ability to: Change the name of the channel, as displayed in the software and on the unit; Choose either Expander behaviour or Gate behaviour per channel; Choose the signal source for the output; Gang modules on both channels on the same unit together for parameter tracking; Link module sidechains on both channels on the same unit for stereo tracking... ...and, with the preset configurations available, the ability to: Choose ADT or SPLIT format for the unit; Position the gain control pre or post-limiter; Route the sidechain listen signal to the main or auxiliary outputs; The window displayed is shown below. There are several important points to be made about the configuration parameters available here. - allows various aspects adjusted.
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Source: This selects the complete input, which will be processed by the channel. By default, each input feeds its complementary output A to A, B to B. However, the inputs can me swapped if required, or summed (A+B option). Additionally, both outputs can be fed from the same input allowing, for example, a four output delay to be realised.2 Mode: Independent selection of the expander/gate module for each channel allows the behaviour to be changed between a downward expander or a noise gate. Name: The channel's name can be easily changed here. This name appears in the System Bar (and System Toolbar) lists, and all the associated edit windows. It also immediately updates the unit with the channel name. Unit Ganging & Linking: The operation of each module can be adjusted so that it shares parameters with the same module in the other channel fo the unit, and/or shares the sidechain signal. Please read the following section, which explains these concepts in detail. What is the Difference Between Ganging and Linking? These two terms are easily confused and justify a section explaining what each means before dealing with the options themselves. On a traditional dynamics processor, if it has at least two channels, there is often the ability to `slave' one channel to the other. Under these circumstances the left hand channel normally becomes the `master'. The parameter controls for the `slave' channel become redundant with both channels assuming the parameter settings of the master channel. Stopping at this point this is what the `Ganging' on SIDD accomplishes. The parameters are linked from one channel to the other, so that any adjustment of one module will set the same parameter on the other channel's module to an identical value. Adjustments are absolute that is no offset will be maintained between the same parameter on each module. If, for any reason, an offset is required gang the modules, set the one channel, then ungang and introduce the offset value. Parameters will only assume the same value once they have been adjusted in the stereo gang mode. For this reason, it is best to enable the stereo ganging prior to any editing to ensure all parameters are the same. Considering the traditional dynamics processor again pressing the `Stereo' or `Slave' button not only links the controls on the front panel, but connects the sidechains of the two channels internally. The reason for this is to maintain the same degree of processing on both channels when treating a stereo signal. So, for example, if in the case of a stereo limiter arrangement, one channel begins to limit by 3dB, the same
2 Set source to `A' for both channels and select `SPLIT' mode.
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amount of limiting will be applied to the other channel. This prevents shifts in the stereo image that would occur if the level dropped in one channel but not the other. On the unit being edited, stereo ganging is also indicated on each modules' window title bar. The title changes to read... [1:A+B] meaning ID number 1 and channels ganged [A+B]. Note that, in the case of the input/output window, the meters are also displayed in stereo.
On all other windows (i.e. dynamic processors), the `bouncing ball' for both channels will be displayed together, like this. On the unit itself, the summary information screen the channel ID changes from `A' or `B' to `S' for stereo.
Linkage is only indicated in this configuration window, but is also shown on the relevant module's summary information screen, by an "x" with a bar over it, like this.
Summarising the difference between the two functions: (Stereo) Ganging Links Parameters (Function) Linking Links Sidechains.
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Choosing the Configuration SiDD offers five configurations which allow various modal permutations. These are listed in the pane as shown, and each has an accompanying routing diagram to clarify how the signal path changes.
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Editing Parameters on the In/Out Window
This window accesses all the parameters accessible under the channel button on SiDD the `A' or `B' button. These include the input and output gains, the six parametric EQ bands, the high and low pass filters, the delay line and the harmonics generator. Input/Output Gains The input and output gains for the channel (or channels if A & B are ganged see page 21 for details on how to achieve this) are adjustable via the two faders shown above. In stereo, there will be a pair of meters shown for input level and output level. Note that the input metering is pre-processing adjusting the input gain fader will not affect the input headroom, so the input meter will not change. Most likely the output meter reading will rise, unless there is a large amount of compression or limiting being applied to the signal. If the input meter is reading too high, reduce the signal level fed to the unit accordingly. Other points to note about the operation of this section are: Clicking on the gain slider track, rather than the knob will change the gain in 1dB steps. Clicking on the text box below the gain slider allows direct numerical entry. Press ENTER to accept the new value. To reject a value typed in accidentally, delete all text from the box and click on any other box. The gain will revert to that shown by the slider.
Page 26 SiDD PC Software Version 2.00
Main Delay The main delay time may be adjusted using the nudge buttons beside the readout directly below the input and output faders. Clicking in the box below the gain slider allows direct numerical entry. Press ENTER to accept the new value. The units used for the readout may be changed from time (milliseconds) to distance (metres or feet). Options from the menu bar, displaying This facility is accessed by selecting File the window below. Select the `Units' tab as shown and then choose the desired readout units from the lower pane.
Parametric Filters, High and Low Pass Filters. The EQ curve is fully interactive just click and hold a coloured node corresponding to the band to be adjusted, and drag it the frequency and level will adjust in real time as it is moved. To adjust the `Q' (or bandwidth) of a filter using the mouse, select the node and, holding the left mouse button down, move the scroll wheel.3 The keyboard may also be used to adjust filter settings, not just on it's own, but also to enhance the feature available during curve dragging. Holding down `Shift' while dragging locks the frequency of the filter, whilst allowing the gain to be adjusted. Holding down `Ctrl' while dragging locks the gain of the filter, whilst allowing the frequency to be adjusted. If your mouse does not have a scroll wheel, then the page up and down buttons may be used to adjust the `Q' during curve dragging. Whilst a filter is selected, (left mouse button down) pressing `Page Up' will increase the filter `Q' by one step. Whilst a filter is selected, (left mouse button down) pressing `Page Down' will decrease the filter `Q' by one step.
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An `intellipoint' type mouse is required for this operation.
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Selecting File Options and then the `Units' tab allows the units displayed for parametric filter bandwidth to be changed from `Q' to actual bandwidth, in octaves. For accurate adjustments of the filters, use the nudge buttons positions beside each parameter readout. Selecting the associated `Flat' checkbox by a filter will set its gain to zero, but leave the `Q' and frequency unaffected the coloured node will remain in place. Parametric or Shelving Filter Behaviour Any parametric filter may be changed to a shelving type response by setting the `Q' control one nudge past the widest `Q' value. This will select `LoShelf', followed by `HiShelf'. Note that the gain of the filter must be set to 0dB to access these responses types. If the filter is `flattened' using the associated tick box, the gain will be automatically reset to 0dB should the response be changed from parametric to shelf. Holding down a frequency nudge button will start moving the filter in 1/36 octave steps (the highest resolution), and accelerate to 1/6 octave, and finally 1/2 octave steps. Leave go near the desired frequency and nudge in single clicks.
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Harmonics Generator and ADT/Aux Delay Controls. Pressing the button marked `Effects' immediately below the main delay text box accesses additional controls for the harmonics generator and secondary delay line. The second and third harmonics content may be individually adjusted using the two faders shown in red. The entire harmonics generator may be bypassed using the `Bypass' button. The right hand pane will be titled either `ADT Effect' or `Aux Output Control' depending on the chosen configuration settings. (To change configuration settings, see page 21 for details.) The ADT level is adjustable from 60dB up to 0dB; for the auxiliary delay level, the range extends +15dB above 0dB. A bypass facility is only available in ADT mode. Note that if the level fader is pulled right down to minimum it will turn the effect off completely.
The keyboard may also be used to navigate around parts of this screen. Adjust gain up in 0.1dB steps, aux/ADT level up in 0.5dB steps, or harmonics up 1 step. Adjust gain down in 0.1dB steps, aux/ADT level down in 0.5dB steps, or harmonics down 1 step. Adjust gain up in 1dB steps, aux/ADT level up in 3dB steps, or harmonics up 1 step. Adjust gain down in 1dB steps, aux/ADT level down in 3dB steps, or harmonics up 1 step.
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What is Dynamic EQ?
Dynamic EQ is essentially a compressor or expander that can be set to respond and act upon only a certain range of frequencies. Its behaviour is dependant on the operating mode chosen two of these are relatively traditional, whilst two modes offer the possibility to turn the normal action of compressors and expanders on their head to allow innovative adaptive control of the program material. The four operating modes are explained in detail below. Quadrant/Mode I: "Boost Above" This is the other less than traditional mode of operation, offering upward expansion, where the signal is boosted once it reaches the threshold. The example below shows that 1kHz filter again, this time with the threshold at +10dB. As can be seen, as the signal rises above the threshold it is progressively boosted around the 1kHz region.
Uses of "Boost Above" mode. This mode is more useful that it might first appear the ability to add EQ only at higher signal levels allows some very effective emphasis of certain parts of the spectrum to be added, without the side effect of a permanent audible peak. Adding some `top end sparkle' try picking out high-hats and cymbals with a filter at 12kHz , `Q' of 1 Octave, and fast attack and release, typically 5mS and 25mS. This gives a significant boost to the top end, without bringing up noise in the absence of any high frequency content. Similarly, `punch' can be re-introduced to a lacklustre bass drum by setting the filter to about 80Hz and slowing the attack to 49mS and the release to 100mS. As the bass drum causes the EQ only to be applied on peaks, there is no additional muddiness added to the bottom end of the spectrum.
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Quadrant/Mode II: "Cut Above" This is one of the more traditional modes of operation. Having selected the frequency band to work with, the dynamic eq will listen to this band and act upon it by cutting(compressing) any frequencies present in it that go above the predetermined threshold. Consider the example below where the threshold is set to 20dB, and the selected frequency band is centred around 1kHz, with a `Q' of 1.0.
Signals below the threshold will pass unaltered, but as increasing signal is applied, those frequencies centred around 1kHz will be cut or compressed. The ratio in the above example is set at 2:1 so, as with any compressor, the amount of gain reduction applied depends on how much the signal exceeds the threshold. The red line represents a signal at 0dB, which is 20dB above the threshold. At 1kHz, therefore, the signal has been compressed to 10dB or 2:1. Uses of "Cut Above" mode. Traditional use of `frequency conscious' compression is to control or `tame' a certain band of frequencies within the program material. Insertion of EQ into the sidechain will make the compressor respond to the required band, but it will cause broadband compression of the signal, so any peaks will cause the entire signal to be compressed. This produces the familiar problem of dulling the material if it is bass-heavy, or causing unnecessary dips and changes in ambience when attempting to remove sibilance. The difference with dynamic EQ is that only the band selected is compressed. This means that it becomes possible to compress the low frequency content of material without affecting the high frequencies at all. The result is increased volume and perceived level with out sacrificing clarity. Any instance where the desired result is to control a band of frequencies, such as de-essing, or de-popping, without affecting the surrounding frequency ranges is an ideal use for this mode. Try de-essing with the filter centred at 8-9kHz, and a relatively narrow `Q' of 3.2, and a maximum gain of 12dB, attack 1mS, release 100mS.
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Quadrant/Mode III: "Cut Below" Having selected the frequency band to work with, the dynamic eq will listen to this band and act upon it by cutting any frequencies present in it that drop below the predetermined threshold. Consider the example below where the threshold is set to +10dB, and the selected frequency band is centred around 1kHz, with a `Q' of 1.0.
Signals above the threshold will pass flat, but as the level decreases, those frequencies centred around 1kHz will be cut or expanded. The amount of gain reduction applied depends on how much the signal drops the threshold and the ratio set a 2:1 ratio would mean that for every drop of 1dB below the threshold, the band centred around 1kHz would drop by 2dB. Uses of "Cut Below" mode. Reducing the level of high frequency noise can be effectively implemented in this mode. Particularly effective on percussive material, unwanted tape noise and interference can be usefully removed without affecting the signal at normal levels. Try the filter set to a high shelf mode at 4kHz, and a maximum gain of 12dB, attack 25mS, release 100mS. The threshold setting is more crucial in this mode than usual, with the trade-off being effective removal of noise against possible intrusive dulling of the program material.
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Quadrant/Mode IV: "Boost Below" This mode operates in a slightly unconventional manner insofar as behaving as an `upwards expander', as opposed to the more traditional `downwards expander'. What this means is that as the signal drops below the threshold, the selected band of frequencies will be progressively boosted in relation to the rest of the spectrum, offering a perceived `lift' in the band. Consider the example below where the threshold is set to +10dB, and the selected frequency band is centred around 1kHz, with a `Q' of 1.0.
Signals above the threshold pass unaltered but, as the signal drops below the threshold, frequencies around the 1kHz region will be progressively boosted (or expanded). How much boost is applied will depend on the ratio set and how far below the threshold the signal actually is. Uses of "Boost Below" mode. One of the best uses of this mode is in the area of voice levelling and clarification. Placing the filter at about 700Hz (lower to nearer 600Hz for men, up to 800Hz for women/children) with a wide `Q' typically 0.7, a ratio of 2:1, a maximum gain of 12dB, attack 10mS and release 100mS. This will ensure that quiet talkers will have their vocal range boosted, without bringing up system noise or microphone handling noise/room rumble.
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How Do the Dynamic EQ Controls Interact?
The best way to explain how the dynamic EQ controls interact with each other is to adjust the parameters in the edit window and examine what happens to the transfer function graph. In this way the behaviour of the process can be broken down into simple stages. What does the edit window show? In its `large' form, the edit window displays two graphs and a set on controls for all available parameters of the dynamic EQ module. The left hand graph is known as the input/output transfer function and represents how the output of the dynamic EQ module varies with different input levels. The scale along the bottom of the graph shows increasing input level, from 60dB up to +20dB. Correspondingly up the left hand side, the output level scale runs from 60dB to +20dB as well. If the dynamic EQ was bypassed, there would be no difference between the signal going in and that coming out, so the graph would show a straight diagonal line running bottom left to top right. (Following the dotted green line shown.) Pressing the `Bypass' button will demonstrate this the red line is overlaid by a grey line as shown here... The most important relationship is between the ratio and the threshold. Set the threshold to 20dB and the ratio to 1:1 initally. Select the Quadrant to be `CutAbv' (cut above). Make sure the `Bypass' button is not pressed. The graph will still be displaying a straight line, as if the module was bypassed. Effectively, the module is bypassed if the ratio is 1:1 then no matter where the threshold is set, the gain of the output will be 1 x the gain of the input. Confirm this by adjusting the threshold no change to the graph. Remember that the threshold can be adjusted by clicking on the graph at the threshold point and dragging up and down. When the mouse is over the threshold point, the pointer changes to a cross pressing and dragging will then show a doubleended arrow.
Page 34 SiDD PC Software Version 2.00
Setting the threshold back to 20dB (centred in the cyan circle as shown), adjust the ratio until it reaches 2:1. The graph will begin to bend at the threshold point, moving away from the 1:1 position. It should now look like this:
If the signal rises above the threshold now, it will be reduced in level by half (or a ratio of 2:1). So, if the input rises by 10dB (for example from 20dB to 10dB) the output only rises by 5dB (correspondingly from 20dB to 15dB). Hence the slope of the graph. As the ratio is increased, the angle of the line above the threshold point will become ever more horizontal. Try increasing the threshold to 4:1 and note the change of slope. As the ratio was adjusted, the other graph (on the right hand side of the edit window) will have been changing as well. This other graph represents the change of frequency response that the dynamic EQ filter will impose on the output. Until now, the transfer function graph could just have easily been referring to a normal compressor, set to a ratio of 2:1. Introducing the dynamic filter into the equation will now be simpler to explain.
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This graph shows how the output will vary, not only with frequency, but at a selection of input levels. These are represented by the six cyan lines, at 3dB intervals above the threshold. As the lines are above the threshold, adjusting it will not affect this graph. What will have an effect is the ratio. The graph above represents a ratio of 2:1. At the centre frequency of the filter, (1kHz in this case) it is simple to correlate what this graph shows with the transfer function graph. Remembering the ratio of 2:1 if the signal at 1kHz is, for example, 6dB above the threshold (the second line up from the bottom), the output level will be half this 3dB. Similarly, 12dB in will come out at 6dB and so on. Increasing the ratio will have the effect of bunching the lines closer together ever increasing signal levels above the threshold will be cut by more and more examine the graph below where the ratio has been increased to 4:1. Now, an input of 12dB will only come out at ¼ of the input level (ratio is 4:1 remember), so at 3dB. The added element of the filter means that this response will vary with frequency, hence the shape of the curve(s). The most cut is applied at 1kHz, with the amount tailing off on either side of this frequency. Try adjusting the `Q' of the filter this will `tighten' or `loosen' the band of frequencies affected by the compression characteristic. The other parameter that will affect this graph (and the in-out transfer function) is the value of the `Max Gain'. This determines the absolute allowable gain reduction (or boost for the `Boost Above' and `Boost Below' quadrants) that can be imposed by the dynamic EQ. If the ratio is thought of as setting how pronounced the effect will be, then the `Max Gain' sets the ceiling on how much can be applied. This can be useful when operating with very high or very low thresholds and high ratios sudden changes in level might well produce far too much cut or boost, causing clipping or unacceptable loss of level, both of which could be very obtrusive.
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With a ratio of 4:1 and the threshold as before, at 20dB, leaving the `Max Gain' at `Max' means that the gain reduction that can be applied will not `bottom out', and the inout transfer function will look as expected. Similarly, the change of gain with frequency will follow the laws set by the threshold and ratio parameters, as shown.
Reducing the `Max Gain' parameter to 6dB means that even if the signal goes high enough above the threshold to impose more than 6dB of gain reduction, only 6dB will be applied. In this example, anything over 8dB above the threshold will be capped with 6dB of gain reduction; in at 8dB with a ratio of 2:1 would come out at 2dB, giving (8 2 =) 6dB of gain reduction. Over 8dB in, and more than 6dB of gain reduction would normally be applied. The effect of this can be seen in the graphs. The cyan line shows the way the process would behave, had all the gain been made available. The red line shows what will actually happen normal until 8dB over the threshold and then capped at 6dB. As this is fixed, all that will be happening is a constant drop of 6dB, hence the line running parallel to the `straight-through' case (dotted).
This is also visible on the frequency response graph. All the lines 8dB above the threshold become equally spaced as only a maximum 6dB
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of gain reduction can be applied.
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Comparing the Four Quadrants... Remember that the Quadrant can only be changed if the ratio is set back to 1:1. Similarly, the filter response can only be changed from parametric to shelving (of full range) if the ratio is set to 1:1. Quadrant/Mode I: "Boost Above" Above the threshold, the gain would be increased at a ratio of 2:1, so the in/out transfer function shows the curve going over the 1:1 green dotted line. The frequency related curves show progressively more boost about the filter centre frequency as signal level increases over the threshold, which is at the bottom of the graph. Quadrant/Mode II: "Cut Above" Above the threshold, the gain would be decreased at a ratio of 2:1, so the in/out transfer function shows the curve going under the 1:1 green dotted line. The frequency related curves show progressively more cut about the filter centre frequency as signal level increases over the threshold, which is at the bottom of the graph. Quadrant/Mode III: "Cut Below" Below the threshold, the gain would be decreased at a ratio of 2:1, so the in/out transfer function shows the curve going under the 1:1 green dotted line. The frequency related curves show progressively more cut about the filter centre frequency as signal level drops below the threshold, which is at the top of the graph.
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SiDD PC Software Version 2.00
Quadrant/Mode IV: "Boost Below" Below the threshold, the gain would be increased at a ratio of 2:1, so the in/out transfer function shows the curve going under the 1:1 green dotted line. The frequency related curves show progressively more boost about the filter centre frequency as signal level drops below the threshold, which is at the top of the graph. What does the bouncing ball show? The bouncing ball (not shown on the above series of graphs) represents the real-time input level to the dynamic EQ module. The bouncing ball is only shown when the PC is connected to a unit (or units) and on-line. Please see page 11 for details of how to accomplish this. Assuming the PC is connected, and on-line, the ball will move along the in-out curve in real-time, giving a clear indication of the level in relation to the threshold, and the settings of the envelope controls (attack time, hold time where appropriate, and release time). A red ball represents the left channel on any unit (channel A), and the right channel (channel B) with a blue ball. Whilst the input signal to any particular module is below the threshold, the ball will be noticeably darker than when above the threshold. Once the ball crosses the threshold, gain reduction is being applied (or gain boost in two modes of dynamic EQ) and the ball will lighten.
In the case of a stereo configuration, one graph will be shown with both balls appearing on the same curve. In this example, channel A is above the threshold, and channel B is below.
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Indication of Clipping The Input to a Module
Before exploring editing any further, it's worth noting that whilst SiDD possesses very high headroom due to the 40-bit accumulators used for all calculations (as opposed to more usual 24 or 32 bit), it is still possible to reach an overload state internally. Even if the input to the unit appears to be relatively low according to the input meter, if input gain has been added and there is a lot of boosted EQ, the signal level within the unit can be running at a very high level. This will not always be apparent, especially in the case of the dynamic EQ module where a large amount of signal amplification can be introduced, but only on peaks (which by their nature are higher than the usual signal level). If the input to a module is too high, the in/out transfer function graph will flash the message `Overdrive' as shown below. Reduce the level from the offending module to until the warning disappears, or audible distortion will occur. Note that this message refers to the input of the module, so the level needs to be reduced `upstream' from the module displaying the message. The table below outlines possible scenarios.
Overload Condition
Input meter clipping (in the red) Dynamic EQ is being overdriven Gate or Expander is being overdriven Compressor is being overdriven Limiter is being overdriven Output meter clipping (in the red)
Possible Cause
Actual input level to the unit is too high Too much input EQ, or input gain too high. Also, ADT level may be too high4 or harmonics generator level(s) Dynamic EQ is applying too much boost. Either Input section is running too hot, or DEQ is running too hot Input section is running too hot, DEQ is running too hot, or Compressor makeup gain too high. Any previous condition, or output gain too high
Possible Action
Reduce the direct feed to the unit itself (adjusting the input gain control will not work) Reduce EQ boost, input gain, ADT level, or harmonics level Set DEQ threshold higher or reduce the `Max Gain' parameter if the threshold must remain fixed See DEQ action and Gate/Expander action See DEQ action and Gate/Expander action. Lower compressor threshold, or turn down makeup gain if threshold must remain fixed. Check all above actions or turn down output gain
4
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Only applicable in ADT mode
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Editing Parameters on the Dynamic Eq Window
Having explained in detail how the dynamic EQ module works, adjusting the parameters is very straightforward. The threshold may be adjusted by either nudging the value using the arrows, or dragging it up and down directly on the in-out transfer function curve. Remember that to change the Quadrant (Operating mode), the ratio must first be set to 1:1. Attack time covers the range 30uS up to 2.0 seconds. The release time covers the same range. Note that the software will not allow release times to be less than attack times, due to the likely `parameter abuse' involved and subsequent poor performance. Adjusting the filter can again be accomplished by either nudging the frequency with the button or directly dragging the centre frequency, denoted by the red square, along the frequency response curve. Note that, unlike the EQ or sidechain frequency response curves, this square will be fixed in its vertical position, as the gain is the parameter being adjusted dynamically. To adjust the `Q' (or bandwidth) of the filter , use the nudge buttons beside the readout or, using the mouse, select the node and, holding the left mouse button down, move the scroll wheel.5 Holding down `Shift' while moving the scroll wheel locks the frequency of the filter, making adjustment of the `Q' easier.
5
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An `intellipoint' type mouse is required for this operation.
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Noise Gate Know-how
Most engineers know the principle reason for using a noise gate to shut off the output of some device that, in the absence of the desired output signal, produces (or passes on) an undesirable level of background noise. The behaviour of the noise gate is controlled by adjusting, principally, its threshold. This sets the point at which the level of the desirable signal is deemed quiet enough to turn it off, thereby cutting off all the background noise as well. Considering the diagram shown here, the desired signal would be the high level burst and, in between, low level noise. An example of this scenario might be the signal from a bass drum microphone a burst of signal, but perhaps with unwanted pickup from lighting dimmers producing a background buzz. Setting the threshold just above the level of the buzz would keep the gate closed during periods were the bass drum was not being played, only allowing signal through when the output would be sufficient to mask the noise anyway. The output graph above highlights several other points that are particularly important with noise gates. Attack and Release Times. As the gate opens, there will be a finite delay before the signal crosses the threshold, cutting off part of the start of the desired signal's envelope. Similarly, as the signal drops below the threshold, the gate will close, cutting off some of the end of the envelope. The diagram above is not truly representative of how the signal approaching the threshold would be affected.
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Controlling the speed at which the gate opens (attack time) and closes (release time) is crucial to providing transparent operation. Setting the attack time too fast on a signal that naturally has a relatively slow attack will produce a click. Such a case often occurs when gating the human voice. The attack time for typical speech or singing is in the order of several milliseconds setting the gate to faster than this will make the voice `appear out of nowhere' with an audible click. This is due to the sudden change in level from the gate being closed to fully open. The fact that this is a large transient change will make it sound Sudden jump in level as the like a click. Changing the gate opens appears as a `distance' between the gate click at the start of the being closed and open will signal. remove this click, but also defeat the purpose of the gate in the first place if the range is set high enough to prevent the click, it's probably not gating the signal enough anyway.
As mentioned earlier, if the attack is set too slow, the gate will not open in time, and part of the signal will be lost. For most musical instruments, the essential part of the signal is the attack phase, as this part identifies the instrument to the human ear. Hearing only the sustain or Start of signal has been lost due release of a sound makes it to slow opening of the gate. difficult to discern what type of instrument is being played, and sounds very unnatural.
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Setting a sympathetic attack time that is similar to that of the required signal will produce the most natural and transparent results. As the diagram below demonstrates, the gate opens at a similar rate to the signal, allowing the full attack portion to be heard. The same rule applies to the release of the gate but for slightly different reasons. Too slow a release, and some of the background noise will be heard as the gate closes; too fast, and the signal will be prematurely cut off, sounding unnatural.
Setting the Range Correctly. The range control sets how much attenuation is applied to a signal once it drops below the threshold and closes the gate. It is quite often overlooked when setting up a gate, the temptation being to set the range at maximum and leave it. When the gate is fully closed, no signal will be heard when the range is set like this. In some situations, this much attenuation can sound quite unnatural, especially if the threshold is set high to gate a loud signal, due to the stark contrast between the gate fully closed and fully open. It's also worth remembering that if the background noise is quite severe, removing it all will only highlight the noise more when the gate opens and the noise becomes apparent again. It's often better to set the range to around 25-30dB of attenuation so that it can still be heard very slightly. Using the Hold Time. When the gate is being used with transient signals that require fast attack but relatively slow decays, it can be beneficial to introduce a hold time into the gates envelope. This has the effect of prolonging the time the gate stays open, after the signal drops below the threshold. The diagram overleaf explains this concept.
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Adding some hold time to the envelope can help with tricky situations where the gate might open and close erratically, such as when there is a lot of background noise. This problem, coupled with a long release, can cause sporadic bursts of noise to open the gate in situations when it would be better to hold it open slightly longer.
SiDD PC Software Version 2.00
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How Do the Noise Gate Controls Interact?
As with the dynamic EQ module, the best way to explain how the controls interact with each other is to adjust the parameters in the edit window and examine what happens to the transfer function graph. What does the edit window show? In its `large' form, the edit window displays two graphs and a set of controls for all available parameters of the noise gate. The left hand graph is known as the input/output transfer function and represents how the output of the gate module varies with different input levels. The scale along the bottom of the graph shows increasing input level, from 60dB up to +20dB. Correspondingly, up the left hand side, the output level scale runs from 60dB to +20dB as well. If the noise gate was bypassed, there would be no difference between the signal going in and that coming out, so the graph would show a straight diagonal line running bottom left to top right. (Following the dotted green line shown.) Pressing the `Bypass' button will demonstrate this the red line is overlaid by a grey line as shown here... The most important relationship is between the threshold and the range. Set the threshold to 20dB and the range to 0dB initially. Make sure the `Bypass' button is not pressed. The graph will still be displaying a straight line, as if the module was bypassed. Effectively, the module is bypassed if the range is 0dB then no matter what level the input is, the gate cannot apply any attenuation below the threshold. Confirm this by adjusting the threshold no change to the graph. Remember that the threshold can be adjusted by clicking on the graph at the threshold point and dragging up and down. When the mouse is over the threshold point, the pointer changes to a cross pressing and dragging will then show a double-ended arrow.
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Setting the threshold back to 20dB (centred in the cyan circle as shown), adjust the range until it reaches -30dB. The graph will begin to bend at the threshold point, moving dropping vertically by 30dB and then continuing in a diagonal line as shown here. What this shows now is that all signals below the threshold will be dropped by 30dB in level. At the threshold, the gain will jump back to 0dB and the output will be at the same level as the input hence the straight diagonal line above the threshold, following the path as if the gate was bypassed. What does the bouncing ball show? The bouncing ball (not shown on the above series of graphs) represents the real-time input level to the gate module. The bouncing ball is only shown when the PC is connected to a unit (or units) and on-line. Please see page 11 for details of how to accomplish this. Assuming the PC is connected, and on-line, the ball will move along the in-out curve in real-time, giving a clear indication of the level in relation to the threshold, and the settings of the envelope controls (attack time, hold time where appropriate, and release time). Note that the ball will move along the bottom axis of the graph even when the output level should force it off the bottom to follow the lower linear portion. As the ball rises up the vertical part of the curve, it is crossing the threshold, and the gate will open. The ball will continue along the upper linear part of the curve with 0dB of attenuation, as shown. (It will have darkened slightly as it is now over the threshold.)
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Sidechain Equalisation How and When to Use it
Each of SiDD's dynamics modules, with the exception of the dynamic EQ, offers the facility to add EQ to the sidechain signal controlling the gain through the module. This obviates the need for a host of external processors that would have to be pressed into service for this task, using up vital EQ resources that could otherwise be used for normal EQ tasks. When would Sidechain EQ be useful? When it is necessary to adjust the sensitivity of the dynamics module, tailoring the sidechain signal with equalisation is the only way to achieve this without affecting the main signal path. De-essing and de-popping. For example, a de-esser is really a compressor that has had its sidechain made more sensitive to sibilance (sounds such as `s' and `t') which occurs in the range 6kHz to 9kHz. A de-popper will be a compressor sensitised to respond to low frequency plosives (sounds such as `p' and `b') in the range 80Hz to 150Hz. In reality, the attack and release times of the compressors as implemented for de-essing and de-popping will be set appropriately to prevent the compression having too great an effect on the overall signal. Both of these examples illustrate using the sidechain EQ to make the sidechain more sensitive to certain ranges of frequencies. It can be very useful to be able to desensitise the sidechain to certain frequency bands to prevent excessive activity at inappropriate times. Maximising loudness without dulling the mix. Loudness, as opposed to volume, is dependant on the signal density as much as the level and so compressing the program can increase the perceived loudness without actually requiring more headroom. For example, consider the situation where bassheavy program material is to be compressed to maximise loudness in a live situation. The spectrum might look like this. If the threshold was set to 0dB (the dotted line) it can be seen that the compressor will start to act first when the low frequency part of the spectrum crosses the threshold.
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This will have the effect of pulling down the level across the entire spectrum (including the high frequencies) and so causing the familiar `dulling' that can easily occur with full range compression. The solution to this problem is to introduce some complementary sidechain EQ that will remove some of the offending bass from the control signal, making the compression more `even-handed' again. The required effect on the spectrum would be similar to that shown below. Having used EQ to tame the bottom end in the sidechain, the compressor will now be much less sensitive to low frequencies and so won't start applying gain reduction in response to only bass. Traditional sidechain EQ is normally only featured in noise gates, and is limited to high and low pass filters. In this situation, whilst a high pass filter could have been used to remove some of the low frequency weight present, a wide shelving cut band would be better, offering much more accurate control of the amount of bass removed. The curve imposed by a high pass filter is shown in orange, and that of a shelving filter in red. This highlights the advantage to incorporating more flexible EQ into the sidechain than that normally offered. The high and low pass filters may be used to perform quite severe tailoring of the frequency response, but for accurate highlighting of a small band of frequencies, parametric filters are much better. The noise gate (or expander) available in SiDD has two bands of fully parametric equalisation, along with traditional high and low pass filters. The parametric filters may be transformed into shelving filters if required. The compressor and limiter both have two bands of fully parametric EQ in their sidechains. Suppressing feedback without adding notch filters. The two parametric bands (as opposed to the high and low pass filters) allow accurate selection of the two main feedback nodes on a monitor mix. Traditionally the main feedback nodes would be picked out with a couple of notch filters placed in the main signal path. This is not ideal for several reasons. Firstly, whilst this method is used by common feedback eliminations systems, it does produce audible artefacts. The introduction of a multitude of notches that are present regardless of feedback conditions cannot be anything other than detrimental to the overall sound quality.
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Secondly, the inclusion of another processor in the signal path, with its analogue to digital and digital to analogue conversion will only increase system noise and degrade performance - consider the situation with one of these systems in place on each monitor! By sensitising the sidechain to the two main feedback nodes, SiDD can be made to compress (and so attenuate) the main signal, but only when feedback occurs. In this way, the main signal is completely unaffected until feedback is present. Whilst this will not stop the feedback completely, it will reduce it to a level where it will no longer build up. The most important factor in feedback suppression is the avoidance of `system runaway' where feedback continues to build to the point of clipping the amplifier and possibly destroying the speaker. This method of suppression will prevent just that situation and not introduce any audible artefacts in the process. Picking out instruments for gating. Sidechain EQ is most often associated with homing in on a band of frequencies to improve the performance of a noise gate. Being able to exclude extraneous program material will make the triggering of a gate much more reliable and easier to set up. SiDD's noise gate (and expander) offers not only the high and low pass filters, but two additional bands of fully parametric EQ. This allows the high and low pass filters to be used for the removal of extraneous program material as normal, but also the parametrics to `highlight' any particular and (or bands) of frequencies. This can prove very useful in situations where there may be limited difference between the wanted and unwanted signals. Being able to accentuate the wanted band at the same time as attenuating the unwanted material, especially with the added possibility of highlighting more than one area for triggering (there are two parametric sidechain filters!) could prove invaluable.
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