i do not know it ... (the l3 multimaximizer)
... (one or two minutes later) ... ok found it ...
lots of free vst plugins available so i will definitely not PAY 40 bucks for a plugin ;-)
my used plugins (in order how audio runs) ...
reafir (surpress some noise ... especially some "rumble" on the low end)
reagate (only send audio when you are speaking but low background noise is surpressed)
reaxcomp (compressor with multiple bands) ... seems to be similar to the l3
and finally frontier (self adaptive limiter) ... that combination wortks superb BUT:
such a high number of plugins needs a lot of tweaking till your audio sounds like from the bbc newsreader (grin) so do not ask me how i have set it all ... it would take a much longer mail to describe what i have done here
so finetune your audio (or making a complete mess from it) can be done easy hi hi
the l3 multimaximizer seems to be similar to reaxcomp so i do not need to try out the L3 i guess
Sigi as you are using VST plugins have you tried the Waves L3 Multimaximizer? It is hard to make it sound bad and it certainly increases talk power with minimal distortion, it never overshoots.
the nice thing when using all in software (sdr) you can do things with compression, limiter, etc etc ... that you can NOT do with analogue circuits
your "rf clipper" may produce a more boosted signal ... but i guess when all is made in software you can be a bit better
the sdr has the audio in a chain (buffer) and you can have a look forward agc ... limit every word to the maximum without overshooting or harmonics produced
and with a monitor receiver on the output of your amp you can add predistortion ... result is an even cleaner (but more punchy) signal
no way to do that with conventional circuits
you could try to combine your rf clipper with the processor in sdrc (but do not crank it up tooo much) ... i do a simlar thingy with my mixing software (voicemeeter potatoe) with some vst plugins added in the chain ... first the plugins do a tiny bit of compression and limits the peaks not to overdrive the sdrc input ... and a bit of processing of the sdrc is added afterwards ... the result is a well boosted signal with no splatter (pay also attention not to produce intermod distortion on the final amp or driving stages!!)
simon you have the equalizer ... what if each filter in the eq would have its own processor and limiter?? is that what you mean with half octave filters (and boosting those) ?!?!?
Am 31.08.2021 um 12:01 schrieb MikeC:
Interesting article. I did experiment further with the processor settings and compared the result with an analogue speech processor that was part of a ssb rig I designed about 20 years ago. This processor uses variable rf log clipping of an ssb signal (at 10.7MHz) followed by another ssb filter. The audio is pre-processed by a syllabic compressor (a vogad). The clipping is done with two stages of log amplifiers. In the original design the resulting ssb was converted to the required frequency but I had already added a demodulator back down to audio for something else so here the output is audio. As the article points outs, non dc baseband clipping avoids the serious harmonic distortion that baseband clipping induces at all but very light clipping because the harmonics multiply outside the passband. There are still intermodulation products that eventually limit performance.
Of course this processor is very component hungry by today’s standards and I happened to have it on the shelf. As the article also points out, the the same result can be achieved in dsp by other means that could not be realised in practice by analogue circuits.
I feed the audio output at line level into the pc and have the SDRC processor turned off. The audio processor bandwidth is from about 200Hz to 2.4kHz, determined almost entirely by the analogue ssb filters.
I think that overshoot is a slightly different, but related problem. Interestingly though I had a look at this processor’s performance by putting test sine and square-waves frequencies into it and looking at the audio output on a scope. Probably by more luck than judgement, the overshoot is a few percent unless the processing is set ridiculously high. I suspect that a fortuitous combination of filters and time constants, plus the relatively narrow bandwidth requirement helps.
I have not checked for overshoot on the rf output from the Limesdr that eventually generates the signal for transmission.
My metering indicates that the average to peak power ratio has increased significantly and, although this is pretty unscientific, the “hf pileup breaking index” is markedly improved.
Nothing can be all things to all men (or women). Everything is a compromise of requirements, performance, complexity, time and cost.
Sent from Mail for Windows
I use a single band RMS compressor followed by a very-well designed limiter which avoids overshoot. I’ve tried multi-band compressors but not achieved much benefit. When I get into my main winter project I may look at this again, having a compressor instance for each octave or half-octave.
For winter I want to finally play with my Hermes-Lite 2, with my main amplifier I’ll have ~350 watts so will want to boost the audio.
Simon Brown, G4ELI
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