Real Time Analyzer

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

In order to set up a basic measurement following steps need to be performed:

• Open the sound card settings dialog by hitting the “Advanced” button on the ribbon bar
• Choose the “Sound In device” which has a microphone attached for Channel 1 + 2
• Switch to the Analyzer tab
• Click on the Source for CH 1 and select SoundIn1 from the context menu
• Close the Settings Dialog by a click on “Done”
• Hit the Start Analyzer on the ribbon bar
• The RTA graph shows now the incoming microphone signal in the time domain
• In order to display the spectrum of the signal open the Analyzer window by a click on the Analyzer settings button in the ribbon bar
• In the Analyzer window set the Mode to Spectrum

The graph shows now the spectrum of the incoming microphone signal:

 

Remark: since only one channel is active, the lower graph has been minimized by dragging the middle line and moving it to the bottom of the window.

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

• In the “Analyzer Settings” window, click on “Advanced Settings”
• Set CH 1 Source to Generator1
• On the ribbon bar click on “Generator”
• Hit the Play button

The graph shows now the spectrum of a 1kHz sine signal:

Integrated Virtual Processing can be started by clicking the “Integrated Virtual Processing” button in the ribbon bar.

At any given time either of IVP/RTA or MM-IR is allowed to be opened.

Integrated Virtual Processing has 4 buttons

• Generator- used to start/stop generator. This is a shortcut to existing generator start/stop
• Plugin Host-used to start/stop Plugin Host
• Mimo Convolver – used to start/ stop Convolver
• Analyzer- used to start/ stop analyzer.

The Sound Card Settings dialog can be either accessed through the Settings button on the ribbon bar or via the Channels window. One can select the audio driver (Host API), the sample rate and the block length of the audio processing.

RTA supports two Host API’s:

• MME: This is the standard windows audio driver. It allows operation of multiple audio devices at the same time. Sample rates are handled by the operating system. RTA can be set to any sample rate. If the sample rate of the physical audio device is different the OS takes care of the sample rate conversion. This mode is recommended if multiple devices are running at the same time, e.g. measuring with an USB microphone while playing back generator signals with an internal sound card. It is recommended to keep the block length at the maximum value of 4096 and the sample rate at 44.1kHz or 48kHz.
• ASIO: This driver is being used with multi-channel sound cards. It enables low latencies and ensures that all input and output channels are in sync. Depending on the latency requirements of the audio signal processing provided by a loaded plugin (future feature) the block length can be reduce down to 64 samples. The RTA sample rate setting has to be equal to the audio device driver sample rate.

Once a Host API has been selected the Sound In and Sound Out devices have to be selected. If no device has been selected RTA runs in a silent mode. This can be used to check the generator modes or to analyze a prerecorded measurement provided by a .wav file.

All devices are provided as two channel devices. The Channels column in the Streams area show to which channel pair a Sound In or Sound Out device is mapped to. These channel pairs will show up at the analyzer and routing settings as SoundIn1..16 and SoundOut1..16 channels. These channels can be selected from contect menu in order to connect sound card channels to RTA processing blocks.

 

In the example above the mapping is configured as follows:

• SoundIn1, SoundIn2: Analog (1 + 2)
• SoundIn3, SoundIn4: Analog (3 + 4)
• SoundIn5, SoundIn6: Analog (5 + 6)
• SoundOut1, SoundOut2: Analog (7 + 8)

 Device has to be reconnected if Sample Rate / Block Length/ Host API is modified

ASIO sound card is preferred over Windows drivers. Use blocklength greater than 1024 while using Windows drivers to avoid noise distortion effect which is a limitation of Windows driver (even seen in audio mulch)

Overview

The Plugin Host is a host for virtual amp dll. The Plugin Host supports up to 3 instances of plugins (virtualAmp.dll in 64 bit), which are executed in series. The block sizes and sample rate will be defined by the sound card settings and used for all plugins. The plugin needs to handle the block size conversion internally in case it is not matching the block size of the device/instance. A sample rate conversion is currently not supported by the virtual Amp and will cause a error message and stop the processing.

Configure plugin host

Open RTA-> Settings-> Plugins

Click Browse to select the xAF library

Set Port number in the port number textbox

If Bypass is selected, then input is fed to the next plugin/output without processing

On click of Apply, updated no. of inputs, no. of outputs and plugin type appears based on the sent signal flow.

On click of Reset all the values of particular row will be set to default values

Similar settings can be applied to all 3 plugins

Switch to Routing tab

Set inputs to Plugin Host (Eg:Generator1 and Generator2)

Set inputs to SoundOut in order to route the PluginHost output channels to the sound card outputs

Optional: Switch to Analyzer tab, to select plugin host output as channel source if  output of PluginHost shall be displayed in RTA.

Set Channel source (Eg:Generator1, Generator2, PluginHost1, PluginHost2) to display in chart

Click Done, once settings are updated

By default (no flash file available next to the virtualAmp.dll), number of in-/outputs in plugin host is 2

Default Port Number starts from 25001

Connect to device through Plugin Host

Click Plugin Host

 

Switch to SFD window, configure signal flow. Click on Send Signal Flow

A pop-up appear which asks to reboot device

Click Reboot

Go Device Designer tab.

Click Connect Device to connect to device.

If AmpSrv cannot establish a connection, close AmpSrv and retry

User can now do Tuning and see the graph in RTA.

Set the graph using Channels window in RTA

When clicking on the Analyzer settings button in the ribbon bar the analyzer window opens:

With “Mode” the analyzer mode can be selected.

Following modes are available:

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

The “Banding” can be adjusted in Spectrum or Multiplexer mode. With banding Off all calculated frequency bins of the spectrum are displayed. In this setting a very detailed analysis is possible. The only drawback is that the higher the FFT size, the more data have to be calculated and displayed, and hence more CPU power is being used. When banding is switched on, frequency bins are combined to groups. The width of such a group can be set by fractions of an octave, e.g. Oct12 i.e, one band has the width of a 12th of one octave.

Depending on the test signal, smoothing of the spectrum over time is required.

This can be set by the “Multiplexer” mode:

• 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 with ms which takes input

Detailed settings can be done with the “Analyzer Settings” Dialog:

 

With the channels list, following adjustments can be made:

• Source: This defines the input of a certain analyzer channel. By clicking on the control (default: “None” for no input), a context menu pops up from which the desired source can be chosen.
• Name: here the name of an analyzer channel can be set. This name appears in the channel viewer and will be set as a default name when storing measurements as traces.
• Calib: 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: here the unit of the analyzer source can be set. 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. With „0“ the channel is omitted, „1“ or „2“ will add the channel to „Average-1“ or respectively „Average-2“.
  • Channels 17 and 18 are reserved for the „Average“ channels. Here only the name can be edited.
• Delay : Add/subtract time delay in milliseconds. In Phase measurement we can add/subtract time delay to compensate HW and/or acoustic delay.

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..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 amount of calculations and data to plot.

By choosing a FFT window the user can define the way how a finite data set, determined by the FFT length is cut out of the more or less infinite input data stream.

See https://en.wikipedia.org/wiki/Window_function for more details on windowing.

Default value of FFT window will be “Hann”

Weighting

Here the user can choose how the input signal is weighted in the frequency range. This can be done independently for the time domain (Freq Weight RMS), and the frequency domain (Freq Weight FFT) measurements.

See https://en.wikipedia.org/wiki/A-weighting for more details on weighting.

Peak Hold

Here the user can adjust the peak hold feature independently for the time domain (Peak Hold RMS) and the frequency domain (Peak Hold FFT) measurements.

• 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

 

Clipping

Clipping occures 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. Modes:

• 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.

 

Other Settings

The time constant for the RMS calculation can be selected under „Sound Level Meter“. It is similar to the Averaging control in the Analyzer window:

• 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

 

The analyzer mode “Multiplexer”, where multiple channels are added to a single “Average” channel can be set to “Time” and “Freq”. 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 on or two spectrums are calculated it can be used when CPU load is an issue.

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 artefacts from switching between channels as it would occur in the “Time” mode.

 

Multiplexer

When the multiplexer mode is set to “Time” the multiplexer gets active. Here the user can set the length of a time slice (“Period Time”). “Fade Time” sets the time within one channel is faded into the next one.

 

In order to perform audio measurements certain measurement signals are necessary. These can be generated by a built in signal generator. The generator can be accessed by the “Generator” button in the ribbon bar:

Different signals can be selected from the Mode control. Following signals are available:

• Sine
• DualSine
• Square
• Noise
• Dirac (Dirac Pulse)
• SinePulse
• SineBurst
• LinSweep
• ExpSweep
• File

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

The generator is started by the Play button, the selected signal is being played back continuously. The signal is stopped by the Stop button.

Incase of LinSweep, ExpSweep, File mode if the loop is checked,  and the generator is started by the Play button, the selected signal will be played back continuously.  If the loop is unchecked, time limited signals will be triggered by Play button.

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

The different modes can be configured within the Generator Settings:

Below all modes and their parameters are described more detailed:

Sine

A single sine wave adjustable in the audible range between 20Hz and 20kHz 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

A stereo noise generator. In the Random mode a standard noise signal is generated. When the NoiseMode is set to Pseudo, a multi sine signal is being generated where on each frequency bin of the selected analyzer FFT, a sine wave is generated. The phases of all sine waves are randomly distributed to achieve a noise like signal. This mode is being used for the spectrum analysis of static transfer functions. The window function of the analyzer has to be set to Rectangle. This results in very smooth spectrums.

The NoiseColor can be changed between Pink (-3dB 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

In this mode the selected wav file is played back.

 

Graph display settings can be done by clicking Graph tab in RTA settings menu

• Min and Max are used to set the Minimum and Maximum value of the Time/Amplitude/Frequency/Spectrum axes
• Interval sets the delta of horizontal/vertical labels
• Frequency axis can be set to Logarithmic or Linear via combo box
• Curve Line thickness can be changed for Measured data, Traces and Target Traces.
• Graph settings will be saved on click of Done and persist values on Import/export of project

Interval will be disabled for Log Scale

Overview

Convolution is the process of multiplying the frequency spectra of our two audio sources—the input signal and the impulse response. By doing this, frequencies that are shared between the two sources will be accentuated, while frequencies that are not shared will be attenuated. This is what causes the input signal to take on the sonic qualities of the impulse response, as characteristic frequencies from the impulse response common in the input signal are boosted.

Configure Mimo convolver with Plugins

Open RTA-> Settings-> Plugins

Click Browse to select the xAF library

Set Port number in the port number textbox

If Bypass is selected, then input is fed to output without processing

On click of Apply, updated no. of inputs, no. of outputs and plugin type appears

Switch to MIMO Convolver tab

Set No. of Inputs/ Outputs

Set Filter taps

User can configure coefficients in 2 ways

• Configure coefficients in Panel
• Configure coefficients through Virtual prediction

Configure coefficients in Panel

Click on Configure Coefficients

Convolver panel launches with 2*2 filters

Tune coefficients by setting them to Flat or by importing via csv/xml files.

Click on Tune, after assigning coefficients

A toast message “Tuning applied” appear

Configure Coefficients through Virtual prediction

Click on Measurement Coefficients

Virtual Tuning window appears

Click Apply after selecting Measurement session from drop down

Switch to Routing tab

Set inputs to Plugin Host

Eg:Generator1 and Generator2

Set inputs to MIMO Convolver

Eg1:PluginHost1 and PluginHost2, if output of Plugin Host is fed to MIMO Convolver

Eg2:Generator1 and Generator2, if generator is fed to MIMO Convolver

Switch to Analyzer tab, to select plugin host, MIMO convolver output as channel source

Set Channel source

Eg:Generator1, PluginHost1, MimoConvolver1 to display in chart

Click Done, once settings are updated

Analyzer View

 Connect to device through Plugin Host. Follow steps as given here.

Open Channels, assign channels to Graph A and Graph B

Chart with outputs of Genertor1, PluginHost1 and MimoConvolver1 appears

Recorder is a sink type in IVP, supporting mono/stereo mode with configurable number of channels to be recorded. The recorder supports append and overwrite mode and can be synced with the generator signal.

Recorder settings are configurable under Recorder tab in Settings window. If recorder is configured with 16 channels with stereo mode, Recorder tab looks as follows :

Mono mode: 1 channel will record per file.

Stereo mode: 2 channels will record per file.

Recording can be appended to same file or overwritten using Record mode 

Sync with Generator: With this option Recorder and Generator  will be sync. When Generator will start, automatically recording will start and same for stop.

On click of Close button , selected file for channel will be cleared from tab settings and closed for recording once settings are applied using DONE button.

 

Max supported recording channels are 64 and supported recording file format is .Wav

Once done with recorder configuration, use Ribbon buttons  for Recorder start/stop and pause/resume.

Recorder started state

Recorder paused state

Recorder stopped state

In this dialog the connections to the sound out devices, plugin host and file recorder can be set.

The Sound Out channels 1..16 correspond to the channel pairs which appear at the Streams area in the Sound Card Settings. A click on a control in the Sound Out column opens a context menu where a source can be selected. Currently player as a source is not supported.

On Mouse hover on “Routing Preview”, use case overview will be displayed for better understanding of Routing

Cursor Measurements

Mouse hovering on any of curve/plots on graph will display horizontal and vertical values of the X and Y positions pointed to. It is important to know that while X value will follow mouse pointer, Y value will show closest trace (as in image below).

 

Markers for quick curves values inspection:

• By pressing CTRL+Click, a marker is created. Such marker will display values of traces (see image).
• Marker will display all trace’s values as tooltips on top of charts.
• A maximum of 5 Markers can be placed on the chart.
• To remove a Markers, select marker by clicking on top of it (line will become wider, as seen in Marker@287.9Hz on image), and press DEL.

Delta Markers (Differential values) for selected curves can be  identified as follows : 

• To enable Delta Markers, press ALT+Click on chart.
• Enable measurements from traces where delta values are desired.
• Selected traces will display: value on first marker, value on second marker and delta between markers (see image).
• Values are in color of trace and underlined with color of marker. Delta markers can be dragged to desired X position.
• To disable Delta markers, press ALT+Click again.

Refresh Spectrum

The spectrum refresh button will refresh all curves (not the traces) on display in the spectrum and multiplexer modes. This feature can be used to restart the averaging time periods (especially important for “forever” averaging).

A click on the Channels button in the ribbon bar opens the Channel Viewer window. In this window the numerical measurements are displayed for each channel.

The channel viewer list contains following columns

•  The first column  indicates channel color. This provide an ability to change the color of channel by clicking on the color box.
• Name: This is the name set in the Analyzer Settings dialog
• Peak: The peak amplitude of the current block of analyzed audio samples
• Rms: The sound level meter value, unit as set in the Analyzer Settings (dBFS, dBV, dBSPL)
• Thd: Total harmonic distortion in %
• Thd+N: Total harmonic distortion plus noise in %
• Peak Freq: The frequency of the maximum level in the measured spectrum in Hz
• Delay : This value is calculated if Analyzer mode is ‘Delay’. The delay measurement is done by cross correlation between a reference channel and a channel which contains the reference signal which went through a certain path (e.g. amp – speaker – microphone). From the position of the maximum within the correlation result the delay can be calculated.
• Offset : +/- Db shifting of measured and math operated channels.
• Graph: Radio buttons allow to quickly select the graph where that channel is displayed.

With the buttons Peak, Rms, Thd, Thd+N and Peak Freq the user can select which values are shown in the list.

Besides the individual channel assignment to a specific graph, a bulk assignment can be performed. If no channels are checked, you can use the button on the “Move all channels to: A, B, Both” to move all channels to desired graph. This will also move Calculated channels. If one or more channels are checked, those same buttons will move only selected channels to desired graph.

To check/uncheck all channels, buttons “Select All” and “Select None” can be used.

The selector control in the top left of the window allows to select which group of channels are shown in the list: all channels (Graph A & B), Graph A channels only or Graph B channels only

The channel window is configured to stay on top and to be re-sizeable to conveniently be kept open for value observation during normal RTA opration.

 

Math operation on Measured channels

Select any two channels

Click on ‘Calculate’ button to get math operation result

Resulted math operated channel is listed on same view

User can delete Math operated channel and as an tool tip you can find which channels selected for math operations.

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

User can easily view the real time values of RMS, THD, Peak, Peak-Frequency, THD+N of selected two channels in ribbon bar as shown below.

At a time two channel’s live data can be seen in Ribbon bar. User can select any of the two analyzer or average channels. There are four live data columns which user can select among RMS, THD, Peak, Peak-Frequency, THD+N.

 

RTA settings “Save” and “Load” options are available on ribbon bar.

On clicking “Save” button  a file save dialog will open with .rta filtered files, on clicking “Save”in the dialog all the following settings will be exported to a file with .rta extension in Json human readable format.

• Generator
• Analyzer
• Audio driver
• Display

Editing a file and importing it back is currently not supported by GTT

On clicking “Load” file open dialog will be displayed. On selecting .rta file RTA settings of the file will be restored. The loaded settings will be applied on the fly.

On Load settings, if sound card settings are invalid, then settings window will be launched. User is allowed to continue only after fixing the sound card settings issue

On click of Apply, sound card settings will be applied, user can then modify other settings

On Click of Cancel, import settings will be cancelled

User has to reconnect device after Import settings

 

Microphone calibration is done to convert the soundcard input values, which are in the range of +/- 1 (full scale FS) to the actual sound pressure level captured by the connected microphone. Therefore a calibration device e.g. a pistonphone is connected to the microphone. This device generates a constant sine tone with a defined level (e.g. 94dB or 114 dB). RTA measures the level of the microphone signal. The user can then store the calibration level so that it will be applied for all measurements.

Prerequisites

• in the Analyzer Settings connect the SoundIn channels to the Analyzer inputs
• set “Peak Hold RMS” to “Off”
• set “Freq Weight RMS” to flat (important if the calibration frequency is not 1 kHz)
• set “Sound Level Meter” to Slow

The actual calibration process

• open the calibration dialog by hitting the “Microphone Calibration” button in the ribbon bar.
  
• adjust the channel which has to be calibrated.
  
• press “Reset”
• select the desired unit (usually dBSPL, but for electrical measurements dBV is also provided, the unit selected here will be applied to the following channels)
• attach calibration device
• wait until the “Measured” value has settled
• press “Apply”
• Compensation file: the File provided by mic manufacturer, which will used for magnitude curve correction. Usage of this can be On and Off by using adjacent checkbox.

The calibration result is automatically stored in the analyzer settings. They can be modified manually at any time.”

“Apply” button click not required for Compensation file selection.

A trace in RTA is a captured measurement curve.

Traces allow for the display of multiple captures measurement curves on the same plot for the quick comparison of measurements. The controls for traces are located on the right side of the spectrum graph.

There are six different kinds of traces:

• Spectrum – Complete data set from measurement without phase
• Phase – Complete data set from measurement with phase data
• Txt – Data imported from a text file (tab separated frequency – spectrum value pairs per line)
• Ovl – Data imported from an Overlay file format
• Eq – Target curve, a spectrum curve described by biquad filter parameters
• Peak Hold – Peak hold trace with three time constants

Traces view is available only in Frequency Domain analyzer modes like Spectrum, Phase i.e, it is not available in Time Domain.

The trace menu is comprised of a trace toolbar and a trace list

Every trace list entry has a button for re-capture, a checkbox for selection and a trace label. The selection checkbox marks the trace for math operations.

A double-click on the trace label opens the trace property dialog:

In this dialog, the name, the offset, and the color of the trace can be set. There are next and back buttons in the dialog so that user can navigate through multiple traces to edit them at a single time.

The Trace Toolbar consists of several buttons:

The Capture button provides two options:

• • Click on button: Capture all traces
  • Drop-down menu: Capture individual traces

The Import button offers a drop-down menu to import single traces (*.trace) or multiple traces (*.trclist).
The Export button offers a drop-down menu to export the highlighted trace (*.trace), the selected trace(s) or all traces (both to *.trclist).
The Add Math Operation button offers a drop-down menu to generate an unweighted average, a difference or a sum trace from the selected traces.
The Add Target Curve button opens the target curve design dialog.
The Delete button offers a drop-down menu to delete the highlighted, all traces. The deletion has to be confirmed.

 

A Target Curve can be added by pressing on the “+” button on menu.

Once target curve is active, its offset will change by 3dB for every jump on octave banding configuration, to follow the behavior of the energetic sum of the octave banding.

 

The Peak Hold Trace can be activated using a checkbox in the Advanced Analyzer menu. It’s time constants Forever, Slow, Fast can be selected in the normal Analyzer Settings Menu. The peak hold trace is reset by choosing Delete for the corresponding trace in the trace list.

     

The Link Mode button is present for the Average channels, Multiplexer mode. This button enables the user to link the measurements from both upper and lower graphs in the RTA screen by allowing trace capture and other operations on traces simultaneously on both graphs. On clicking this button, the user will be presented with an option to provide the name of the charts from 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 lower graph will refer to Average channel 2.

A-weighting, B-weighting, and C-weighting are different frequency weightings that approximate the human ear’s sensitivity to different frequencies. They are used to adjust measurements to better align with the perceived loudness by human listeners.

The Traces view has new “Trace Settings” button, which will open new view where user can select desired Weighting (Flat(Unweighted), A, B, C).

Based on desired selection weighting will be applied to all captured traces. Each trace RMS SPL value will be displayed in traces view as shown below.

 

Smoothing a technique that reduces variations in plotted curves to improve the visual perception of trends or patterns in frequency response or level measurements. It is commonly used in audio analysis and equalization tasks to enhance clarity while considering the trade-off between noise reduction and preservation of important details.

The Traces view has new “Trace Settings” button, which will open new view where user can select desired octave banding for smoothing. Based on desired selection smoothing will be applied to all captured traces.

With selected smoothing option, smoothed curve looks as follows:

At the top of the list of traces in the Traces view, there is a checkbox that allows you to select or deselect all traces.

By selecting the checkbox at the top, all traces in the trace window are automatically selected.

When the top checkbox is unselected, it unselects all the traces in the trace window.

When you are in linking mode and selects the top checkbox in the upper trace window, it automatically selects all the traces from both the upper and lower trace windows.

When you unselects the top checkbox in the upper trace window, it unselects all the traces from both the upper and lower trace windows.

On click of Reboot Plugin host will be restarted implicitly and then set to previous state

 

The Probe Point feature has been added to support streaming data from any point of the signal flow back to GTT to be able to analyze, record or re-use the data inside of IVP. The main idea behind this feature is to give the user and/or developer a way to receive  data from an Audio object and analyze audio input in Real time analyzer view.

Configure and Use probe point for analysis

Enable probe points per core – Use Virtual core property ‘Probe Points: Enable/#” in device view to enable and set number of probe points per core. Only the configured number of probe points can be enabled in signal flow per core.

The probe points configurations will send to the device using “Send Device Config” feature. This configuration can be fetched from device using “Load Device Config” feature.

To run streaming of state variables this feature has to be enabled, hence this is a high configuration feature and will be skipped to safe MIPS and memory. The number of probe points refers to audio streams only. Setting this value to zero (checkbox enabled) will automatically allocate memory for the state variable streaming and support up to 16 state variables.

If Probe Point is disabled for the core, number of probe points input field will be disabled and streamable state variables will be excluded for that core in streaming window.

Add Probe points on Audio Object connection point–

 Probe Point context menu for selected connection has below options:

• Add– Add probe point on selected connection source point and default probe point state set to enabled state.
• Remove – Probe point will be removed from selected connection.
• State – Can alter the state of probe point on selected connection.
  • Enable -State change to Enabled and Pin will highlight with  icon with bright purple Color
  • Disable- State changed to Disabled and Pin will highlight with grey dark Purple Color

 

Manage Probe points 

Open Probe point management window through ‘Manage probe Points’ ribbon button.

In this window, you  can enable/disable probe point and edit probe point name in probe point management window.

Probe points are organized in Device[Device address]->Core->Instance[Sample Rate]->Audio Object Name[Block-Id] order.

State, Pin Label and Probe point name will displayed for each probe point. This window will always be in sync with Signal Flow designer probe point states.

Click Save to persist edited probe point names

Configure probe points in RTA/IVP

• Open Advanced settings window
• Select Probe Point as Analyzer / Recorder / Sound out sources as shown in below image.
• Click Done

To record the probe point signal

Start Probe-Point streaming

Pre-requisites:

• Probe Point feature is enabled for the core and number for active probe points has been set correctly.
• Probe Points are used in the IVP config as a source (Analyzer, Recorder, etc.)
• Plugin Host is started.
• Connection to device is established.

Use IVP Block Length <= 512 for probing to avoid frame dropping

Once above pre-requisites are done, start probe point using Probe Point Ribbon button.

Example of Streamed data