Author: Debanka

A signal generator is an important feature that generates specific measurement signals. These signals can be sent to the device or system for evaluation.
To conduct audio measurements, it is essential to have specific measurement signals that can be produced using a built-in signal generator. You can generate a signal using the “Generator” button in the ribbon bar.

By utilizing the “On/Off” function, you can activate or deactivate a signal. It is possible to generate multiple signals using this feature. The number of signals visible in the “Generator” window is determined by the number of instances specified in the Generator settings.

The gain of the generator signal can be adjusted in 1 dB steps with the Gain control.

To configure the signal, click on the “Generator Settings” to open the advanced RTA setting dialogue box. Here you can configure different generator modes.

On the RTA setting dialogue box, enter the instance value or use the increase and decrease buttons to change the instance value.

Using the “Mode” option, you can select different signals from the drop-down list. The available modes are listed below.

• Sine: A single sine wave adjustable in the audible range between 20 Hz and 20 kHz by SineFreq. The phase between the two output channels can be set by SinePhase.
• DualSine: Two sine waves mixed together to one mono output. The frequencies can be set via DualSineFreq1 and DualSineFreq2, the mixing gains by DualSineGain1 and DualSineGain2.
• Square: Similar to the standard sine wave but shaped as a square wave.
• Noise: This is a stereo noise generator mode. In the Random mode, a regular noise signal is produced. However, when the Noise Mode is set to Pseudo, a multi-sine signal is generated where a sine wave is produced on each frequency bin of the chosen analyzer FFT. The phases of all the sine waves are distributed randomly to create a signal similar to noise.
  This mode is used for spectrum analysis of static transfer functions, and it is essential to set the analyzer window function to Rectangle for optimal results, producing very smooth spectrums.
  The NoiseColor can be changed between Pink (-3 dB per octave fall off) and White (flat frequency spectrum).
  By adjusting the phase the output can be coded in a way so that surround upmixers can pan the signal according to the adjusted angle. The output changes from mono at 0° to L/R uncorrelated at 90° to out of phase at +/- 180°.
• Dirac (Dirac Pulse): In this mode one sample wide pulses are generated. The time between two pulses is set by SignalLength.
• SinePulse: This mode generates sine squared pulses. The shape of the pulse is set by SinePulseFreq, the interval between two pulses by SinePulseInterval.
• SineBurst: In this mode sine bursts are generated. The frequency is set by SineBurstFreq, the length of the burst by SineBurstLength, and the interval by SineBurstInterval.
• LinSweep: This generates a sine sweep starting from SweepStartFreq and ending at SweepEndFreq. The length of the sweep is set via SweepLength. The frequency progress is linear.
• ExpSweep: Similar to LinSweep only with an exponential frequency progress.
• File: Click on the folder and select the wav file. Based on their selection, the number of channels present in the WAV file will be displayed here. For the selected file, each channel of the selected file will be used as the generator input.
  After selecting this mode the user has to adjust the routing settings, hence number of channels depends on the selected file.
  
  
  After changing the mode from file mode to any other mode, this routing adjustment has to be adjusted according to the user need and will be signaled by GTT as shown below.
  

Use the “Active” checkbox to either select or deselect all traces. When you select the checkbox at the top, all instances in the window will be automatically selected. Similarly, when the top checkbox is deselected, it will unselect all the traces in the trace window.

Additional Configurations

Loop: If you enable the “Loop” option while using Dirac, SinePulse, SineBurst, LinSweep, ExpSweep, or File mode, the generator will play the chosen signal repeatedly when the Play button is pressed. However, if the loop option is disabled, the generator will play time-limited signals upon pressing the “Play” button.

Delay: You can enter the Delay value. The delay represents the time interval between the input and output of the generator, indicating the time it takes for the generated signal to propagate through the system.

Gain: You can enter the Gain value. The Gain setting allows you to increase or decrease the volume or strength of the generated signal.

The signal generator has a stereo output. This is relevant for signals with adjustable inter-channel phase or stereo wav file playback.

Microphone setup tasks such as calibration, channel selection, and mic compensation file selection can be done using the Mic Setup view. If analyzer source is Sound-In, its respective calibration and compensation files will be considered for magnitude curve correction.

The compensation file is only considered for magnitude curve correction; it has no impact on calculated metrics such as sound pressure level (SPL) and total harmonic distortion (THD).

Using an Analyzer, you can measure and analyze various aspects of an audio signal. It can be used to measure characteristics such as frequency response, amplitude, distortion, and noise level.

Settings

Banding

In Spectrum or Multiplexer mode, it is possible to adjust the “Banding”. When the banding is turned off, all frequency bins of the spectrum are displayed, allowing for a highly detailed analysis. However, this setting requires more CPU power as the amount of data that needs to be calculated and displayed increases with the FFT size.

Spectrum mode is shown in the example below when Banding is turned off.

On the other hand, when banding is turned “On”, frequency bins are grouped together. The width of each group can be adjusted by fractions of an octave, such as Oct12, which means that one band has the width of a 12th of one octave.

Spectrum mode is shown in the example below when Banding is turned on.

Mode

Using the Mode option, you can select different analyzer modes from the drop-down list. The available modes are listed below.

• Time: Displays source channels in the time domain (one block of 4096 samples).
• Spectrum: Displays the spectrum of the source channels.
• Multiplexer: Switches the RTA into a multiplexer mode where multiple source channels are combined into two average channels.
• Phase: Displays the magnitude and phase of the source channels. Phase can be wrapped and unwrapped using Graph Settings in the settings window. The phase measurement is done by a dual-channel FFT analysis.
• Delay: Displays source channels in the time domain. The delay measurement is done by cross correlation between a reference channel and a channel that contains the reference signal that went through a certain path (example – amp – speaker – microphone). From the position of the maximum within the correlation result the delay can be calculated. The calculated Delay value is displayed in the Channel viewer in the Delay column.
• IR: Displays the magnitude and IR of the source channels. This is an Impulse response measurement with an exponential sine sweep. When this analyzer mode is selected ‘ExpSweep’
• Generator mode is set, and you are not allowed to change to other modes. The ‘Play’ button is disabled in the Generator view and with the ‘Single’ button he can generate ‘ExpSweep’ once.

 Averaging

Depending on the test signal, smoothing of the spectrum over time is required. This can be set by the “Averaging” option.

Following are the averaging options available.

• Fast: Small smoothing time constant and hence only a small amount of smoothing (time constant 125 ms).
• Slow: Large smoothing time constant and hence significant smoothing (time constant 1000 ms).
• Custom: Custom smoothing time constant and a textbox where the user can enter custom time constant in ms.

Peak Hold

You should be able to select a time constant for peak trace. Depending on the time constant setting, the peak hold trace shall show the maximum value that occurred within the defined moving time window.

Peak Hold settings include:

• Slow
• Fast
• Forever

Advanced Setting

Click on the “Analyzer Settings” to open the advanced RTA setting dialogue box. Here you can configure different analyzer settings.

The following modifications can be made in the Analyzer setting window using the channels list:

• Source: This defines the input of a certain analyzer channel. By clicking on the control a context menu pops up from which the desired source can be chosen.

If there is no input available, None will be shown as the source by default.

• Name: Enter the name of an analyzer channel. This name appears in the channel viewer and will be set as a default name when storing measurements as traces.
• Calib[db]: When a channel is being calibrated for a certain microphone the determined value appears here. It can also be overwritten by entering a desired value. The unit is “dB”; the analyzer input stream will be scaled by this value.
• Unit: Allows you to set the analyzer source unit. This unit appears later in the channel viewer.
• AvgCH: When the analyzer is in “Multiplexer” mode this control determines to which “Average” channel the analyzer source is added.
  When the channel is “0,” it is not included; when it is “1” or “2,” it is added to “Average-1” or “Average-2,” respectively.
• Channels 17 and 18 are reserved for the “Average” channels. Here only the name can be edited.
• Delay: Add or subtract time delay in milliseconds. In Phase measurement, we can add/subtract time delay to compensate for HW and/or acoustic delay.
• Peak Trace: Peak hold trace allows the analyzer to display a secondary live trace for each channel showing the highest amplitude values for each frequency. This feature helps to mark the highest amplitude reached at each frequency.

By default, all the peak traces will be disabled. This can be enabled using the checkbox available in the analyzer settings tab for each channel.

Click “Delete” in the data context menu of the peak trace in the trace list to reset the peak trace. When a peak trace is deleted, the database will also delete the current peak trace and create a new one.

FFT Settings	The length of the FFT which is used for the spectrum calculation can be set between starting from 4096 up to 131072 samples (4k to 128k). The higher the value, the finer the frequency resolution of the spectrum. But with increasing lengths the CPU load will increase due to the higher number of calculations and data to plot. You can specify how a finite data set is extracted from the roughly infinite input data stream using the “FFT Window”. The “FFT Length” determines how the data set is cut out.For more details about windowing, refer to the Window Functions. “Hann” will be the default value for the FFT window.
Weighting	The Weighting function allows you to select how the input signal is weighted across the frequency range. This can be customized separately for time domain measurements (Freq Weight RMS) and frequency domain measurements (Freq Weight FFT). These support A, B, C, and D weightings. For more details about weighting, refer to the A-weighting.
Peak Hold	The Peak Hold function enables you to independently adjust measurements in the time domain (Peak Hold RMS) and frequency domain (Peak Hold FFT). Off: Disables the peak hold feature. Fast: Sets the hold time to 1 sec. Slow: Sets the hold time to 5 sec. Forever: Holds the peak values until the Reset button in the ribbon bar is clicked. Averaging RMS: The time constant for the RMS calculation can be selected under “Sound Level Meter”. Off: No smoothing Fast: Small smoothing time constant and hence only a small amount of smoothing. Slow: Large smoothing time constant and hence significant smoothing. Forever: Extreme smoothing time constant. Average Mode: The analyzer mode “Multiplexer”, where multiple channels are added to a single “Average” channel can be set to “Time” and “Freq”. Time: In “Time” mode the analyzer works as a multiplexer. It combines multiple input audio signals into one audio signal by dividing the input channels into equal fixed-length time slots and mix them into a common output channel with fading between channels. The length of the time slots and the fading characteristic can be configured during runtime. The output signal is the signal of one input channel at a time. If the last input channel is reached, the next input channel will be the first input channel again. Since in this mode only one or two spectrums are calculated it can be used when CPU load is an issue. Freq: In “Frequency” mode the analyzer calculates the spectrum of each individual channel and calculates the average of all spectrums. This method is faster and more precise because there are no artifacts from switching between channels as it would occur in the “Time” mode.
Clipping	Clipping occurs when the input signal exceeds the full-scale range of the input sound device. RTA can detect this condition and signal it. There is also an option to exclude the data packet which contains clipped data from the analysis. Clipping Detection mode includes: Off: Disables clipping detection. On: Enables clipping detection. ExcludeData: Enables clipping detection and excludes clipped data packets from being analyzed. When data are clipped, and the detection is enabled a “DATA CLIPPED” message on the top right corner of the graph is shown.
Multiplexer	Activating the multiplexer mode to “Time” allows you to set the length of a time slice (referred as “Period Time”) and the time duration for fading one channel into the next (referred as “Fade Time”).

In the Channels setting window the numerical measurements are displayed for each channel.

The channel viewer list contains the following columns:

• The first column indicates the color of the channel. This allows you to change the color of the channel by clicking on the color box.
• Name: Display the name of the channel. You can change the name in the Analyzer Settings dialog box.
• Offset: +/- Db shifting of measured and math operated channels.
• Enable: Channel enable and disable allow on display graphs on Analyzer window.
• Peak: The peak amplitude of the current block of analyzed audio samples.
• Rms: The sound level meter value, the unit as set in the Analyzer Settings (dBFS, dBV, dBSPL) with selected Weighting (A B C D).
• Thd: Total harmonic distortion in percentage (%).
• Thd+N: Total harmonic distortion plus noise in percentage (%).
• Delay: This value is calculated if Analyzer mode is set to ‘Delay’. The delay measurement is done by cross correlation between a reference channel and a channel that contains the reference signal that went through a certain path (example: amp – speaker – microphone). The delay can be calculated using the position of the maximum within the correlation result.
• Peak Freq: The frequency of the maximum level in the measured spectrum in Hz.
• Graph: Radio buttons allow you to quickly select the graph that displays that channel

By using the Peak, Rms, Thd, Thd+N, Delay, and Peak-Frequency buttons, you can select which values to display in the list.

In addition to assigning individual channels to specific graphs, you can also perform bulk assignments. If no channels are selected, you can use the “Move all channels to A, B, or Both” button to move all channels, including calculated channels, to the desired graph.

If one or more channels are selected, the same buttons will only move the selected channels to the desired graph.

You can use the “Select All” and “Select None” buttons to check or uncheck all channels, respectively.

The selector control located at the top left of the window enables you to choose which group of channels to display in the list: all channels (Graph A & B), only Graph A channels or only Graph B channels.

The channel window is designed to remain on top of other windows and can be resized as needed, making it easy to keep open for value observation while using the RTA.
Click on the “Advanced Settings” to perform additional configuration. For more details about Advanced configuration, refer to the RTA Advanced Settings.

Math operation on Live Channels

To perform math operations:

1. Select any two channels.
2. Click on the Calculate button to get the math operation result.
   Math operated channel is listed on the same view.
   

You can delete Math operated channel and as a tooltip, you can find which channels are selected for math operations.

Only one Math operated channel can be created for combinations of measured channels.

The Real Time Analyzer (RTA) is a multi-channel analyzer for audio signals. It provides time and frequency domain analysis tools to measure RMS or peak levels, frequencies, THD, delays, magnitude, and phase responses. A built-in signal generator provides sine tones, sweeps, and pulses and various noise signals. Using a file player recorded signals can be analyzed.

Related Topics

• Real Time Analyzer Components
• Real Time Analyzer Settings
• Probe Point Configuration
• Real Time Data View
• Graph Settings and Measurement
• Traces
• Analysing Audio Signal

Below are the settings available for configuration in the Real-Time Analyzer.

• Channels Setting
• Generator Settings
• Analyzer Settings
• Microphone Calibration
• RTA Advanced Settings
• Saving and Loading RTA File

When the RTA or Measurement Module is opened, the main window title bar is updated with status information including sound card and analyzer settings information such as

• Selected HOST API
• Selected device
• Sample rate
• Block length
• FFT length
• Analyzer mode
• FFT window
• Averaging
• Banding

Integrated Virtual Process (IVP) refers to the use of virtualization technology to create a seamless and interconnected environment for analyzing various audio signal processes. It involves following operations.

• Generating virtual signals
• Connecting Plugin Host
• Utilizing Mimo Convolver
• Analyzing audio signal
• Utilizing Probe Points

You can start Integrated Virtual Processing by clicking the “Analyzer” or Play button.

Integrated Virtual Processing is a combination of the following options.

• Generator: Used to start/stop generator.
• Plugin Host: Used to start/stop plugin host
• Mimo Convolver: Used to start/ stop mimo convolver.
• Analyzer: Used to start/ stop analyzer.
• LinkMode:  The Link Mode feature allows you to establish a connection between the measurements in the upper and lower graphs on the RTA screen. This connection enables you to perform trace capture and other operations simultaneously on both graphs.
  

To enable Link Mode, you need to configure the Analyzer settings mode option to Multiplexer.

On clicking Link Mode, you will be presented with an option to provide the name of the charts from the below window. Once the linking is activated, any operation performed on the Traces in the upper graph will be reflected in the lower graph. The upper graph will refer to Average Channel 1 and the lower graph will refer to Average Channel 2.

Configure Basic Measurement

Steps to configure basic measurement for analyzing audio signal.

1. On the IVP RTA tab, click on the Advanced or Advanced Settings. This opens RTA Settings window.
   
   
2. On the RTA Setting window, select the Sound Card tab, and then select the “Sound In device” that is connected to a microphone for channel 1 + 2.
   
3. Switch to the Analyzer tab, click on the Source for Channel-1, and select SoundIn1 from the context menu.
   
4. When you have finished configuration, click Done to close the Setting dialogue box.
5. On the ribbon bar, click on Analyzer. The RTA graph now displays the incoming microphone signal in the time domain.
6. In order to display the spectrum of the signal, click on the Analyzer Settings in the ribbon bar. This opens the Analyzer Setting window.
7. On the Analyzer Settings window, set the Mode to Spectrum from the drop-down list.
   

The graph now displays the spectrum of the incoming microphone signal.

As only one channel is active, the lower graph has been minimized by dragging the middle line and placing it at the bottom of the window.

Analyse RTA without Soundcard Signal

In order to test RTA without a soundcard signal, the test signal generator can be connected directly to the analyzer.

1. On the Analyzer Settings window, click on Advanced Settings. This opens RTA Settings window.
2. On Analyzer tab, click on the “Source” for Channel-1, and select Generator 1 from the context menu.
   
3. When you have finished configuration, click Done to close the Setting dialogue box.
4. On the ribbon bar, click on Generator, and click on Analyzer or “Play” button. The RTA graph now displays the spectrum of a 1 kHz sine signal.
   

The Toolbox contains the core objects that were retrieved from the xAF dll. The objects that can be used within the core to create the device signal flow are called core objects. Each core object has its own purpose and solves parametric issues which block routing within the core.

Core Objects are classes that are part of the Audio Core (virtual core) class and operate at a higher level than audio objects. The audio processing class itself is a core object. The relationship between core objects and Audio core is similar to that of audio objects and the Audio Processing class.

• Xaf Instance
• Buffer
• Splitter
• Merger
• Ssrc lir Int
• Float to Fixed
• Fixed to Float
• Nan Detector

The execution order (or index) of the core object is displayed by Core Object Id. Routing determines the order in which core objects are executed within a core. The core objects that are connected to the core input will be executed first, and the core objects that are connected to the root object after that will be given the next execution order.

The device identification feature is enabled for audio libraries version 13 and higher.

Xaf Instance

The Xaf Instance is the core object inside which the signal flow for that instance can be created.

• Core Object Id (execution order of core-object with-in core) and Instance Id (index of xAF instance with-in core, based on execution order) will be displayed as read-only fields.
• The sample rate and block length of the instance will control signal flow within the instance. You can change the sample rate and block length of the instance in the properties section.
  

Further information on signal flow creation is available in the GTT Signal Flow Designer guide.

Buffer

Buffer core object is used to convert the input block length into the required output block length. The buffer core object has an equal number of input and output channels. It can be used as a pass through core object OR it can be used to, as its name suggests, buffer samples from the input to the output. The object does not change the sample rate (it is the same at the input and the output).

If you want to connect two core objects with different block lengths, you can use a buffer core object. As a result, the input block length will be the Block Length of the first core object, and the output block length should be the Block Length of the other core object.

It can be configured as follows:

• If the input block length is equal to the output block length, then it behaves as a pass through object (so you could have an audio core with a buffer object to connect the core input to the output)
• Input and output block lengths must be integer multiple of each other
• When input and output block lengths are not equal, the object handles taking in input at a lower block length and outputting it at a higher one and vice versa. For example, it facilitates the connection of an object at block length 32 to an object at block length 64

Introducing this object into your signal flow for any case but pass through WILL result in latency at the output.

Splitter

Splitter core object is used to convert one input to multiple outputs of the same sample rate and block length.

• This core object always has a single input.
• In order to make routing from any core object to the splitter both the source core object and splitter core objects sample rate and block length should match.
• Number of output channels for the splitter is configurable.
  

It is not to be confused with the Splitter audio object.

This object operates in parallel to an xAF instance NOT within it.

Merger

Merger core object is used to merge multiple inputs into a single output of the same block length and sample rate.

• This core object always has a single output.
• In order to make routing from any core object to merger both the source core object and merger core objects sample rate and block length should match.
• Number of input channels for the merger is configurable.
  

It is not to be confused with the Merger audio object.

This object operates in parallel to an xAF instance NOT within it.

Ssrc lir Int

Synchronous Sample Rate Converter (SSRCs) is used to convert the input sample rate to the required output sample rate.
SSRCs are core objects that can operate within an audio core. Currently there is one implementation of SRCs in Awx.

Two options are provided to convert the sample rate. Both these options are mutually exclusive.

IIR Integer Multiple SSRC

This core object implements a synchronous sample rate converter whose input sample rate / input block length and output sample rate / output block length are integer multiple of each others. This is also an infinite impulse response implementation (IIR).

The object operates in one of 2 modes:

• User Coefficients mode
• Predefined Coefficients mode

Before we get into the details, there are some common configuration parameters between the two.

• The input block length needs to be set by the user.
• The Biquad filter topology. Currently 2 topologies are exposed.
  • Direct Form I
  • Direct Form II

User Coefficients mode: In this mode, the user has to provide the input and output sample rate. Input and output sample rates should not be equal. The Number of Biquads field is read-only.
User has to import the coefficients by clicking on the button “Import Co-efficients”. Based on the number of coefficients in the file, the Number of Biquads is updated.

Validations for User Coefficients mode: The Input and Output sample rates cannot be the same. Validation is shown when the same values are entered.

After adding a new “Ssrc lir Int” object and selecting “User Coefficients Mode”, if the coefficients are not imported, the following message will be displayed on various operations such as “Save”, “Edit Device”, “Copy Core Objects” and “Paste Core Objects”. After importing coefficients, the user can perform the required operation.

Predefined Coefficients mode: In this mode, the xAF dll is used to read the input sample rate, output sample rate, and the number of biquads. When a value in the combo box is selected, the xAF dll is also used to fetch the corresponding coefficients.

Biquad Co-efficient has to be re-imported whenever the mode is switched between Predefined Co- efficient mode to User Co-efficient mode.

For these pre-defined coefficients, the quality measures are as follows:

• Signal to noise ratio: 80 dB
• Total harmonic distortion: 2e-3f
• Spurious free dynamic Range: 59 dB
• Total harmonic distortion plus noise: -60 dB
• Frequency response flatness: 3 dB

Output block length (Displayed as a read-only field) = (Output sample rate /Input sample rate) * Input block length.

Float to Fixed

Float to Fixed core object accepts audio buffers that are in floating point format and outputs buffers that are in fixed point format (16-bit, 24-bit, 32-bit etc).

• # of Channels is configurable. No of Input channels = No of Output channels
• The user can configure the scalar value to indicate what fixed point format is required. This scalar value is multiplied by the floating point input samples to convert them to fixed point.
  For example, to convert from float to 32 bit fixed point, this scalar value must be:
  (1 << (32-1) – 1) = 2,147,483,647
• In order to make routing from any core object to Float2Fixed object both the source core object and Float2Fixed core objects sample rate and block length should match.

Float To Fixed core object is enabled for audio libraries version 16 and greater.

Fixed to Float

Fixed to Float core object accepts audio buffers that are in fixed point format (16-bit, 24-bit, 32-bit, etc) and outputs buffers that are in floating point format.

• # of Channels is configurable. No of Input channels = No of Output channels.
• The user can configure the scalar value to suit the fixed point format of the input samples. The reciprocal of this scalar value is multiplied by the fixed point input samples to convert them to floating point.
  For example, to convert from 32 bit fixed point to float, this scalar value must be:
  (1 << (32-1) – 1) = 2,147,483,647
• In order to make routing from any core object to Fixed2Float object both the source core object and Fixed2Float core objects sample rate and block length should match.

Fixed To Float core object is enabled for audio libraries version 16 and greater.

Nan Detector

The NaN (Not a Number) detector core object detects NaN from input samples and informs the platform using an xTP command if NaN is found. The xTP command will inform about the core id, core object instance id and channel index, so that platform can react accordingly by muting or resetting states. The input samples are copied to the output without doing any other processing. The number of output channel(s) is always same as the number of input channel(s).

• # of Channels is configurable. No of Input channels = No of Output channels.
• Block length and Sample rate are configurable.
• The number of input channels is user configurable and ranges from 1 to 255.
• This core objects’ block length and sampling rate are the same at both input and output side.
• Block length is configurable in the range of 4 to 4096 samples.
• Sample rate is configurable in the range of 8 kHz to 192kHz.

In order to make routing from any core object to NaN Detector, both the source core object and NaN Detector core objects sample rate and block length should match.

NaN Detector core object is enabled for audio libraries version 19 and greater.

Core Objects Validation

When the GTT is loaded with a version of the xAF library lower than 13. If a user tries to open a device view that contains core objects other than a XAF instance, they will see the following error message.

Aside from the Xaf instance, every other core object will be red.

The Control Generator audio object allows to generate a constant control value at a specified time.

Control Generator Properties

Below table describes the audio object properties and functionalities.

Properties	Description
# of  Control outputs	Enter the number of control outputs. In Single control mode, the number of control outputs is configurable between 1 to 255. In Multi control mode, the number of control outputs is the same as the number of control inputs. Range: 1 to 255 (Min – 1 and Max – 255) By default, the number of output channels is set to 1.
Display Name	Display the name of the Control Generator audio object in signal flow design. It can be changed based on the intended usage of the object.
Object Mode	This audio object can be configured in two operation modes. Single control Multi control

Mode

The object operates in one of the following two modes:

Mode	Description
Single control	In this mode, the object has a single control input and configurable control outputs ranging from 1 to 255. It has a single set of tuning parameters that decide the functioning of all control outputs. Based on the Control value and Trigger time configured, which is applied to all the control outputs,  the constant control value is sent out every “n” seconds i.e. trigger time configured. If its value is 0 the control value is sent out just once. This is the default mode.
Multi control	This audio object can be configured with control channels ranging from 1 to 255. The number of control inputs will be equal to the control outputs. The object has a separate set of tuning parameters that decide the functioning of each control output. Control value and Trigger time can be configured separately for each control output. The default number of control channels shall be 1.

The control input in both modes is triggered only and the value at the control output is set by the Control value tuning parameter.

Additional parameters

There are no additional parameters available for the Control Generator audio object.

Tuning Parameters

The following are the paraments that can be tuned from GTT.

Parameter	Description	Data Type	Range	Default	Unit
Control Value	The value which is sent out to the control output. Setting this value will send the constant control value out.	xFloat32	Min: single precision float minimum -3.402823 x 10^38 Max: single precision float maximum 3.402823  x 10^38	0.0f	NA
Trigger time	It is the rate at which the control value is sent on the control output. Setting this tuning parameter shall send the constant control value out. If its value is 0 the control value is sent out just once. If its value is greater than 0, let’s say a value of N, the control value shall be sent out every N seconds.	xFloat32	0 – 600	0.0f	seconds

Control Interface

The Single-control mode has a single control input and configurable control outputs. The control outputs are configured using the “# of Control outputs” property.  The control output value ranges from 1 to 255.

By default, control input and control output is 1.

The Multi-control mode has configurable control outputs. The control outputs are configured using the “# of Control outputs” property.  The control output value ranges from 1 to 255.
The number of control inputs is equal to the control outputs.

Native Panel

The Control Generator audio object does not support a native panel.