I agree buy a good microphone, although it
need not be that good for amateur SSB and NBFM radio use.
However a good microphone will not help you have punchy audio.
Having a good microphone AND using high
quality audio processing is not mutually exclusive. If you
want really well behaved punchy audio from DXer to BBC
presenter and are already using VST plugins then the L3
multimaximizer is a very good easy to use one stop solution.
Waves plug ins have been in regular use in broadcasting for a
Sigi you need to compare the L3 to your
free plug ins to appreciate the difference. I would NEVER use
any reverb processor, you really don’t need it, in fact it
almost certainly will reduce intelligibility.
Personally I am perfectly happy with the
processing on my TS-890S or K3S. The 890S sounds very natural
but louder than unprocessed audio.
Just buy a decent microphone?
i do not know it ... (the l3
... (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)
reaeq (shape the input signal a bit)
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
protoverb (just slight delay to give the
audio a more filled sound)
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
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
greetz sigi dg9bfc
Am 31.08.2021 um 13:43
schrieb Conrad, PA5Y:
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
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
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
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:
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
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.
indicates that the average to peak power ratio has
increased significantly and, although this is pretty
unscientific, the “hf pileup breaking index” is markedly
Nothing can be all
things to all men (or women). Everything is a compromise
of requirements, performance, complexity, time and cost.
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
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
Simon Brown, G4ELI
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