1. Opening the ISA / AISA and settings
The ISA can be accessed via the IO menu item by selecting the sub-item of the input EQ:
This is now the interface of the Input Signal Analyzer:
You can choose between the conventional Input Signal Analyzer (ISA) and the Advanced Input Signal Analyzer (AISA).
If you want to use the Advanced Input Signal Analyzer, you will need an external USB sound card as an additional accessory.
This USB sound card must be connected to the last RCA output of the DSP or DSP amplifier via an RCA cable and, of course, to the PC. The output level of the measurement signal is 1 Volt.
After selecting the AISA, you will automatically be shown which RCA output you need to connect via the USB card.
In the example, we are using a HELIX DSP.3S. You will be shown that you need to connect output H via RCA cable for this DSP.
As you want to use this output as a measurement output, you must select this in the checkbox.
A window with a warning will appear immediately.
This indicates that output H is now switched to measuring. All settings that may have been made previously are thus deactivated.
If you agree, please simply confirm this.
You can find the basic settings for the AISA by clicking on the highlighted cogwheel symbol.
You can now select your USB sound card via the drop-down menu. This is usually selected automatically.
You can also, for example, display curve information "Display Graph Information" or delay calculations "Display Delay Estimation" for the signals by ticking the two corresponding boxes.
The resolution of the X-axis can be changed using the arrow buttons. A resolution of +10 to -15 dB can be switched to a resolution of +15 to -35 dB.
By clicking on the "A", you can switch from automatic adjustment of the measurement curve to the zero curve to manual adjustment. The curve can be shifted manually in the range from -10 to +40 dB.
2. Operation
The channel to be measured is selected or a sum of several channels is formed using the "drag & drop" operation. The mixing ratio can be adjusted by double-clicking, just like in the IO configuration. The polarity of individual channels can also be inverted. The main difference compared to the normal IO configuration is that no DSP channel is configured in this "drag & drop" configuration, but rather the input signal of the analysis tool.
The measurement method works with correlated pink noise. It is important to only use this signal for the measurement, as otherwise measurements of channel summations and other analysis methods will not work and will lead to incorrect measurement results.
The AISA measurement starts by clicking on the "Start Analyzer" button and can be stopped by clicking on "Stop Analyzer". As soon as no signal to be measured has been selected, the measurement is automatically terminated and the message "No Input selected" appears.
In the same window, you can also choose between fast and slow smoothing. As a rule, we recommend slow smoothing.
A reset button can also be pressed to start a new measurement.
As soon as the measurement is started, you will see a green measurement curve. This measurement curve is slowly smoothed and shows the frequency response of the input signal(s). In our case, you can see a measurement of the summed left and right full-range signal.
If two signals with different runtimes are recorded, these differences in runtime can be calculated by activating the "Display Delay Estimation" function. The calculated delay time is then displayed in the top left-hand area. Slight corrections to the calculated value may be necessary.
The delay time error can be recognized by the so-called comb filter, a kind of ripple in the frequency response.
Of course, the frequency response of the individual signals does not show any transit time.
The calculated value should now be entered on one of the two pages and the measurement restarted. As you do not know on which side the value must be entered, please simply try one of the two sides. If you have chosen the wrong side, the comb filter will move to the left. Otherwise, the comb filter will move significantly to the right or even disappear completely.
In our case, a runtime addition of 1.19 ms to the right front signal led to a perfect result.
If there are all-pass filters in the input signal, these can only be detected if two input signals are measured at the same time. It is always advisable to measure one of the two signals first and to save this measurement using one of the 10 memory buttons. To save, right-click on one of the memory locations (M1, M2, ...). The color of the saved curve corresponds to that of the respective memory location (M1 = red, M2 = blue, ...)
In our case, we measure the left signal first and save this curve in memory location 1 (red) before measuring the right signal and saving it in memory location 2 (blue).
We can see that both signals are identical and almost perfect.
Now we measure both signals together, whereby all-pass filters (phase shifts) become recognizable. In our case, two 2nd order all-pass filters can be recognized. One at approx. 240 Hz and one at approx. 1500 Hz.
To neutralize the all-pass filter, an all-pass filter must also be set on the opposite signal. Try and error to find out which signal the all-pass filter must be set to. Once you have set the filter to the correct signal, the all-pass filter is corrected.
Adding the all-pass filter to the wrong signal results in more dips in the frequency response.
If the signal is an already filtered signal, you can display the respective high-pass and low-pass filters by displaying the "Display graph information". The frequency and also the slope of the filter are displayed.
In our example, this is a high-pass filter at approx. 255 Hz with a 24 dB slope.
If required, the frequency response of the input signal can be evened out using manual equalizing or the "TuneEQ" function.
The example shows the frequency response of an input signal that has already been processed in the OEM amplifier or radio and thus reaches the input of our DSP.
Now move the measured curve as centrally as possible to the red zero curve using the left slider. You must first click on the blue "A" (for automatic averaging) to access the manual averaging "M".
Please note that we can shift an equalizer by a maximum of +6 dB, but by up to -15 dB!
Now you can use the TuneEQ.
It is up to you to decide how many EQ bands TuneEQ can use in parametric mode. The default setting is to use 6 bands (band 2 to band 7), as band 1 may be required to use the DLC function (Dynamic Loudness Control). If you do not need or want to use DLC, you can of course also use the first band for TuneEQ. In our case, we are using band 2 to band 6, as DLC is activated. Now select the frequency range in which TuneEQ should be used. In our case, it is the range from 125 Hz to 6300 Hz, as the remaining range does not show any problems.
Clicking on the TuneEQ button immediately displays a thin grey line showing the equalizing calculated by TuneEQ.
The calculated equalizing is then transferred to the EQ bands by selecting SetEQ.
By starting a new measurement (or a reset), you can immediately determine how TuneEQ affects the input signal.
In our case, all the problems with this channel were corrected perfectly.
Advanced Input Signal Analyzer & Input EQ
With the introduction of the ACO platform, all DSP devices have the ability to analyze the amplitude response of the signal sources connected to the analog inputs.
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Der ISA ist über den Menüpunkt IO erreichbar, indem man den Unterpunkt des Input-EQs auswählt:
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