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Description of the Operation of ALC in PowerSDR

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Description of the Operation of ALC in PowerSDR

System Dependencies
Minimum PowerSDR Version: 1.X.0
Applicable Hardware: All FlexRadio Systems transceivers

Content provided by: FlexRadio Systems Engineering

The purpose of ALC is to limit AF input beyond a certain defined limit, which in this case is 0 dB; the point where the DSP begins to "clip". By doing so, the secondary affect is that you are not over driving the PA which could induce a decrease the linearity of it causing splatter and RF output is reduced accordingly eliminating RF "spikes" or overshoot.

ALC is a "safety mechanism" built into the radio and is used differently that with traditional radios where driving the radio into ALC was a method of getting the max power out of your transmitter. Software defined radios operate differently where driving the radio into ALC actually creates distortion in your transmitted signal. With a properly configured AF input signal path, you should never exceed 0 dB of AF input gain causing the ALC algorithm to be invoked. You get no appreciable improvement in output talk power by setting your mic gain to the point that it is hitting the 0 dB ALC threshold limit causing the radio to attenuate your input mic gain. By driving the AF input gain too hard to invoke ALC the result will be distorted audio from the time between clipping the DSP and having ALC kick in and attenuate the AF input gain. Peaking your mic input gain at -2 dB results is plenty of "drive" and will result in getting the full 100 watts PEP out of your radio.It is recommended that you use the ALC TX meter rather than the MIC meter to monitor your AF input gain to make sure that it never exceeds -1 dB under any circumstance (voice peaks).

The algorithm that is responsible for controlling the ALC function has to do several things.

First it has to monitor the AF input gain in the TX audio chain. If the AF input gain threshold is exceeded (0 dB), then it has to reduce the AF input gain by a certain amount in order to be below the AF input gain threshold value. Obviously it would do no good, just to turn off the AF input resulting in no RF output and some really choppy audio, so the ALC algorithm has to gracefully ramp down or decrease the AF input gain by an amount (aka the ALC clamping value) that is in excess of the AF input gain threshold point. These operations do not happen instantaneously and there are variables that control this behavior.

The ALC algorithm has three variables that are exposed to the user for adjustment; attack, decay and hang. All of which have units in the time domain (milliseconds (ms).

Here is an explanation of how ALC works using an example where the AF input gain threshold has been exceeded by +3 dB

The ALC algorithm detects that the AF input gain has exceeded the threshold limit by +3 dB

How quickly ALC responds to this event is determined by the ATTACK parameter. The default value for this variable is 2 ms. So when the AF threshold has been exceeded by 2 ms, the ALC clamping or gain reduction of the AF input signal reduces the AF input gain by -3 dB achieving the threshold limit of 0 dB.

Once the ALC has clamped the AF gain "spike", it will continue to attenuate the AF at that clamping value for a fixed amount of time. This is the HANG parameter and the default value for it is 500 ms.

Once the HANG time has been exceeded, rather than setting the clamping value back to 0 immediately, it slowly decreases the AF gain clamping value back to 0 over a specific time period. This is so there isn't an abrupt change in your AF input gain, making audio transitions smoother and more natural sounding. This is the DECAY parameter and the default value for it is 10 ms.

So the net effect is that your AF gain input being sent to the transmitter does not exceed 0 dB, even on multiple voice peaks within the HANG period, making you audio sound better due to not clipping of the DSP and reduces RF spikes or overshoot in excess of the RF drive value you have set.



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