As an 'SSB Audiophile', for the past 6 years, I've done much experimenting with audio compressors, limiters, splitband processing, EQ's and the like. The goal is to produce the most articulate, pleasant sounding and cleanest SSB signal as possible. Here are a few rules of thumb that seem to work in practice:

1. Always strive for maximum linearity in the exciter and linear amplifier and allow yourself 3dB of headroom. That is what produces a clean and narrow signal i.e., a signal that takes up no more RF bandwidth then it has to, for a given baseband bandwidth. 2. ALC should be used to throttle back the drive level when it approaches say 90% of maximum power. It should warn the operator that he needs to back off the drive level! ALC should not be used to increase average power! 3. IMHO talkpower is best achieved with a Split-band compressor or Peak limiter, like the broadcasters use. Don’t be deceived! You can have LOUD without distortion. 4. Audio equalization can greatly increase articulation by emphasizing the higher frequencies that carry speech silibants, again without that ‘crunchy’ odd-harmonic distortion we commonly hear on the air.


Maintain linearity in the RF chain i.e., the exciter and the final amplifier, by driving it to 50% of maximum power. Allow yourself 3dB of headroom. The 3dB loss buys you significantly lower 3rd and 5th order IMD distortion products, thereby narrowing the signal. I’ve yet to see a 100 watt RF amplifier that does not benefit from this procedure!

ALC should only be used as a Peak Limiter with the Threshold set about 1dB below maximum output. ALC should only come into play when the exciter is in danger of being overdriven, no other time. IMHO. it should not be used to increase talkpower because there’s no way to filter out distortion products produced with ALC. The Attack time should be AS FAST AS POSSIBLE, otherwise loud peaks will briefly overload and produce “buckshot”. The Release time should be about 200 to 600mS or the ‘syllabic’ rate of the speech signal.

RF clipping was devised to provide instantaneous attack time by clipping or hard limiting the low level SSB signal and then filtering it to remove most of the odd harmonic distortion. This was SOTA 40 years ago, but there are better methods today that produce talkpower without significant distortion. However, while IMD products outside the speech passband are greatly reduced by RF Clipping, inband distortion is not. Although more than 6dB of RF Clipping is IMHO the point of diminishing returns, most operators use much more then that! In fact, many of the Dx'ers and contester's believe that one cannot develop talkpower without distortion and that the distortion actually improves the readability! This is not true of all methods! It is possible to produce loud, in your face audio with minimal distortion. Haven't you ever turned down the TV because the advertisements always seem much louder then the show? The broadcast boys do this by using Split-Band-Processing (and other methods) and they produce plenty of 'loudness' AND low distortion. Let’s get past RF clipping and be innovative with the SDR!

In split-band processing for example, we divide up the audio passband into SIX 1-octave wide bands e.g., (100-200, 200-400, 400-800, 800-1.6k, 1.6k-3.2k, 3.2K-6.4K ) so that no 3rd order or higher distortion can pass thru any particular band, then we can Clip each filtered signal 6 dB e.g., and post filter each band. Each filter output should have no harmonic distortion and if we sum the filter outputs of all bands to produce the full bandwidth signal the result will be loud and clean.. This is nothing new in broadcast audio, but until the interest in 'HiFi SSB' very few phone operators used these techniques!

Another very important method for increasing ‘articulation’ i.e., the ability to be understood in the presence of noise - is using equalization to shape the audio passband. Although frequencies under 300Hz are not essential for articulation, frequencies above 2600Hz will greatly improve articulation! Narrowing speech bandwidth does not concentrate the signal and cut thru the QRM – that is false. What does cut thru the QRM are the sibilant frequencies from approx. 2000 to 3600Hz. If we pre-emphasize our speech about 3-6 dB/octave starting at about 800Hz, the resulting speech spectrum will cut thru the din because we have placed more energy in frequencies carrying sibilants. The receiving station can narrow his receiver bw somewhat (de-emphasis) decreasing the hiss and noise and still understand this processed signal. In 2.8 – 3.3kc bw.s of course, this signal would sound tinny, strident and so forth, but when you wish to sound natural and maybe HiFi, you can switch to a different EQ profile say from 90Hz to 3600Hz and decrease by 6dB the frequency band ranging from 400 to 1800Hz - as an example.

a couple of suggestions -

1.      Please keep the SDR as linear as possible.
2. Provide a method for switching to different transmit audio ‘profiles’ ie., save the EQ settings, compressor settings and so forth so the operator can switch from ‘DX-Contest audio’ to ‘HiFi SSB audio’ to something else audio. 3. Provide an ALC THRESHOLD control that is relative to the RF Power Setting. E.g., if RF Power=50watts and ALC Threshold is set for 90%, then ALC commences at 45watts.


73, Alan Davis   K2WS



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