Speech Processor


MikeC
 

 

Sorry, another question...

 

I was wondering what sort of processing the processor function on the transmit audio is. It seems to improve the average to peak power ratio but only by a relatively small amount. By the comment that it makes the audio harsh I rather assume it implements a similar function to rf clipping. I can’t find any description for it in the audio help. Am I right that the audio alc is syllabic in time constant?

 

I have had a couple of reports that the audio is a bit thin which set me off on this. Playing with the audio filtering and the equaliser doesn’t quite get me what I want. The headset boom mic is not the most expensive I have to say.

 

Mike

 

 

Sent from Mail for Windows

 


Siegfried Jackstien
 

Near any mic should give a goid audio... Equalizer plus processing... Maybe switch on the mic boost...
Already cranked up mic audio in windows??
If you look in tx tab where the vox slider is.. While talking in the mic (no need to tx!).. How high does the grey bar come up?? (shows raw audio strength that goes IN the console)....

Processing with 30 to 40 should give enough boost... 

Thin audio?? Boost low tones a bit.. 

Greetz sigi dg9bfc 

Am 27.08.2021 19:46 schrieb MikeC <mike.christieson@...>:

 

Sorry, another question...

 

I was wondering what sort of processing the processor function on the transmit audio is. It seems to improve the average to peak power ratio but only by a relatively small amount. By the comment that it makes the audio harsh I rather assume it implements a similar function to rf clipping. I can’t find any description for it in the audio help. Am I right that the audio alc is syllabic in time constant?

 

I have had a couple of reports that the audio is a bit thin which set me off on this. Playing with the audio filtering and the equaliser doesn’t quite get me what I want. The headset boom mic is not the most expensive I have to say.

 

Mike

 

 

Sent from Mail for Windows

 



Simon Brown
 

Mike,

 

The speech processor is quite good, it’s a compandor followed by a limiter. Let’s see a screenshot of the transmit options while you’re talking, the Spectrum shows the range and relative levels. A few tweaks and you’ll be good to go.

 

Simon Brown, G4ELI

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of MikeC
Sent: 27 August 2021 18:46
To: main@sdr-radio.groups.io
Subject: [SDR-Radio] Speech Processor

 

 

Sorry, another question...

 

I was wondering what sort of processing the processor function on the transmit audio is. It seems to improve the average to peak power ratio but only by a relatively small amount. By the comment that it makes the audio harsh I rather assume it implements a similar function to rf clipping. I can’t find any description for it in the audio help. Am I right that the audio alc is syllabic in time constant?

 

I have had a couple of reports that the audio is a bit thin which set me off on this. Playing with the audio filtering and the equaliser doesn’t quite get me what I want. The headset boom mic is not the most expensive I have to say.

 

Mike

 

 

Sent from Mail for Windows

 


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kb3cs
 


Simon Brown
 

Probably.

 

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

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of kb3cs
Sent: 30 August 2021 21:37
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Controlled-Envelope good?
http://www.arrl.org/files/file/QEX_Next_Issue/2014/Nov-Dec_2014/Hershberger_QEX_11_14.pdf

 - 73 -


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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.

 

Mike  

 

Sent from Mail for Windows

 

From: Simon Brown
Sent: 31 August 2021 06:41
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Probably.

 

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

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of kb3cs
Sent: 30 August 2021 21:37
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Controlled-Envelope good?
http://www.arrl.org/files/file/QEX_Next_Issue/2014/Nov-Dec_2014/Hershberger_QEX_11_14.pdf

 - 73 -


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Siegfried Jackstien
 

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) ?!?!?

dg9bfc sigi

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.

 

Mike  

 

Sent from Mail for Windows

 

From: Simon Brown
Sent: 31 August 2021 06:41
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Probably.

 

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

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of kb3cs
Sent: 30 August 2021 21:37
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Controlled-Envelope good?
http://www.arrl.org/files/file/QEX_Next_Issue/2014/Nov-Dec_2014/Hershberger_QEX_11_14.pdf

 - 73 -


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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.

 

73

 

Conrad

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien via groups.io
Sent: 31 August 2021 12:22
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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) ?!?!?

dg9bfc sigi

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.

 

Mike  

 

Sent from Mail for Windows

 

From: Simon Brown
Sent: 31 August 2021 06:41
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Probably.

 

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

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of kb3cs
Sent: 30 August 2021 21:37
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Controlled-Envelope good?
http://www.arrl.org/files/file/QEX_Next_Issue/2014/Nov-Dec_2014/Hershberger_QEX_11_14.pdf

 - 73 -


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Simon Brown
 

One thing,

 

With software we can’t have overshoot, hence a well-designed limiter. In software it’s also easy to write an RF speech processor but when set too high can sound harsh. I get favourable reports on my audio with Pluto and FDM-DUO so am happy.

 

Now to get my HL2 on the air…

 

Simon Brown, G4ELI

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of MikeC
Sent: 31 August 2021 11:01
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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.

 

Mike  

 

Sent from Mail for Windows

 

From: Simon Brown
Sent: 31 August 2021 06:41
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Probably.

 

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

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of kb3cs
Sent: 30 August 2021 21:37
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Controlled-Envelope good?
http://www.arrl.org/files/file/QEX_Next_Issue/2014/Nov-Dec_2014/Hershberger_QEX_11_14.pdf

 - 73 -


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Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.


Siegfried Jackstien
 

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)

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 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

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.

 

73

 

Conrad

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien via groups.io
Sent: 31 August 2021 12:22
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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) ?!?!?

dg9bfc sigi

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.

 

Mike  

 

Sent from Mail for Windows

 

From: Simon Brown
Sent: 31 August 2021 06:41
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Probably.

 

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

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of kb3cs
Sent: 30 August 2021 21:37
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Controlled-Envelope good?
http://www.arrl.org/files/file/QEX_Next_Issue/2014/Nov-Dec_2014/Hershberger_QEX_11_14.pdf

 - 73 -


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Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.

 


Simon Brown
 

Or,

 

Just buy a decent microphone?

 

Simon Brown, G4ELI

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien
Sent: 31 August 2021 16:23
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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)

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 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

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.

 

73

 

Conrad

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien via groups.io
Sent: 31 August 2021 12:22
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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) ?!?!?

dg9bfc sigi

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.

 

Mike  

 

Sent from Mail for Windows

 

From: Simon Brown
Sent: 31 August 2021 06:41
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Probably.

 

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

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of kb3cs
Sent: 30 August 2021 21:37
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Controlled-Envelope good?
http://www.arrl.org/files/file/QEX_Next_Issue/2014/Nov-Dec_2014/Hershberger_QEX_11_14.pdf

 - 73 -


--

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Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.

 


--
- + - + -
Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.


Gedas
 

Or take singing lessons....

Gedas, W8BYA EN70JT

Gallery at http://w8bya.com (under repair)
Light travels faster than sound....
This is why some people appear bright until you hear them speak.
On 8/31/2021 11:45 AM, Simon Brown wrote:

Or,

 

Just buy a decent microphone?

 

Simon Brown, G4ELI

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien
Sent: 31 August 2021 16:23
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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)

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 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

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.

 

73

 

Conrad

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien via groups.io
Sent: 31 August 2021 12:22
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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) ?!?!?

dg9bfc sigi

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.

 

Mike  

 

Sent from Mail for Windows

 

From: Simon Brown
Sent: 31 August 2021 06:41
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Probably.

 

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

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of kb3cs
Sent: 30 August 2021 21:37
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Controlled-Envelope good?
http://www.arrl.org/files/file/QEX_Next_Issue/2014/Nov-Dec_2014/Hershberger_QEX_11_14.pdf

 - 73 -


--

- + - + -

Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.

 


--
- + - + -
Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.


MikeC
 

I feel very disadvantaged ONLY having about a dozen old analogue chips in my processor......

 

Of course if I did it again now it would be dsp which would give much more flexibility as well. I do think the key is avoiding the harmonic distortion problem which dsp can do.

 

In the end its all the same signal processing methods implemented by either an analogue or a digital computer. An old friend of mine (long dead now) who was a tutor at Imperial College said, “the interesting problems getting digital systems to work are actually analogue”. He was a control man and was also heard to say “any frequency above 1Hz is dangerously high”.

 

Mike  

 

Sent from Mail for Windows

 

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien
Sent: 31 August 2021 16:23
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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)

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 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

greetz sigi dg9bfc

 

 

 


Simon Brown
 

I have a working DATONG speech clipper. Over 40 years old, sitting in a cupboard…

 

 

Simon Brown, G4ELI

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of MikeC
Sent: 31 August 2021 17:32
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

I feel very disadvantaged ONLY having about a dozen old analogue chips in my processor......

 

Of course if I did it again now it would be dsp which would give much more flexibility as well. I do think the key is avoiding the harmonic distortion problem which dsp can do.

 

In the end its all the same signal processing methods implemented by either an analogue or a digital computer. An old friend of mine (long dead now) who was a tutor at Imperial College said, “the interesting problems getting digital systems to work are actually analogue”. He was a control man and was also heard to say “any frequency above 1Hz is dangerously high”.

 

Mike  

 

Sent from Mail for Windows

 

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien
Sent: 31 August 2021 16:23
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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)

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 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

greetz sigi dg9bfc

 

 

 


--
- + - + -
Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.


N2CBU
 

On 8/31/21 12:37 PM, Simon Brown wrote:
I have a working DATONG speech clipper. Over 40 years old, sitting in a cupboard…
That's fine as long as there are also tea and biscuits.


Siegfried Jackstien
 

hmm i have a good microphone ... wireless speakermike with ptt ... or in other words a mic with a 600 feet long cable hi hi

barefoot it sound "ok" (with some equalizer tweaking and a bit of compression added) ... but with the vst plugins it is a bit more punchy (without sounding bad)

mainly i installed the mixer cause i wanted to have a few different audio ways for rx in parallel (drive speaker, headphones and wireless mike plus maybe the bluetooth speakers or the speakers fron the tv set that i use as big monitor for sdrc)

the mixer on the tx part is also nice (easy to change levels also for digimodes ...or whatever) ... the vst plugins just added as "nice to have but not really needed"

no i will not take singing lessons (grin)

greetz sigi dg9bfc

Am 31.08.2021 um 17:45 schrieb Simon Brown:

Or,

 

Just buy a decent microphone?

 

Simon Brown, G4ELI

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien
Sent: 31 August 2021 16:23
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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)

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 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

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.

 

73

 

Conrad

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien via groups.io
Sent: 31 August 2021 12:22
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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) ?!?!?

dg9bfc sigi

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.

 

Mike  

 

Sent from Mail for Windows

 

From: Simon Brown
Sent: 31 August 2021 06:41
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Probably.

 

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

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of kb3cs
Sent: 30 August 2021 21:37
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Controlled-Envelope good?
http://www.arrl.org/files/file/QEX_Next_Issue/2014/Nov-Dec_2014/Hershberger_QEX_11_14.pdf

 - 73 -


--

- + - + -

Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.

 


--
- + - + -
Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.


Gedas
 

Gosh and I was so looking forward to hearing you sing fiorentini schnell (I hope I got the spelling right).

Gedas, W8BYA EN70JT

Gallery at http://w8bya.com (under repair)
Light travels faster than sound....
This is why some people appear bright until you hear them speak.
On 8/31/2021 3:24 PM, Siegfried Jackstien wrote:

hmm i have a good microphone ... wireless speakermike with ptt ... or in other words a mic with a 600 feet long cable hi hi

barefoot it sound "ok" (with some equalizer tweaking and a bit of compression added) ... but with the vst plugins it is a bit more punchy (without sounding bad)

mainly i installed the mixer cause i wanted to have a few different audio ways for rx in parallel (drive speaker, headphones and wireless mike plus maybe the bluetooth speakers or the speakers fron the tv set that i use as big monitor for sdrc)

the mixer on the tx part is also nice (easy to change levels also for digimodes ...or whatever) ... the vst plugins just added as "nice to have but not really needed"

no i will not take singing lessons (grin)

greetz sigi dg9bfc

Am 31.08.2021 um 17:45 schrieb Simon Brown:

Or,

 

Just buy a decent microphone?

 

Simon Brown, G4ELI

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien
Sent: 31 August 2021 16:23
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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)

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 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

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.

 

73

 

Conrad

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien via groups.io
Sent: 31 August 2021 12:22
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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) ?!?!?

dg9bfc sigi

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.

 

Mike  

 

Sent from Mail for Windows

 

From: Simon Brown
Sent: 31 August 2021 06:41
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Probably.

 

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

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of kb3cs
Sent: 30 August 2021 21:37
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Controlled-Envelope good?
http://www.arrl.org/files/file/QEX_Next_Issue/2014/Nov-Dec_2014/Hershberger_QEX_11_14.pdf

 - 73 -


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Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.


Siegfried Jackstien
 

do re mi fa so la si doooooooo

Am 01.09.2021 um 00:09 schrieb Gedas:

Gosh and I was so looking forward to hearing you sing fiorentini schnell (I hope I got the spelling right).

Gedas, W8BYA EN70JT

Gallery at http://w8bya.com (under repair)
Light travels faster than sound....
This is why some people appear bright until you hear them speak.
On 8/31/2021 3:24 PM, Siegfried Jackstien wrote:

hmm i have a good microphone ... wireless speakermike with ptt ... or in other words a mic with a 600 feet long cable hi hi

barefoot it sound "ok" (with some equalizer tweaking and a bit of compression added) ... but with the vst plugins it is a bit more punchy (without sounding bad)

mainly i installed the mixer cause i wanted to have a few different audio ways for rx in parallel (drive speaker, headphones and wireless mike plus maybe the bluetooth speakers or the speakers fron the tv set that i use as big monitor for sdrc)

the mixer on the tx part is also nice (easy to change levels also for digimodes ...or whatever) ... the vst plugins just added as "nice to have but not really needed"

no i will not take singing lessons (grin)

greetz sigi dg9bfc

Am 31.08.2021 um 17:45 schrieb Simon Brown:

Or,

 

Just buy a decent microphone?

 

Simon Brown, G4ELI

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien
Sent: 31 August 2021 16:23
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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)

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 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

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.

 

73

 

Conrad

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien via groups.io
Sent: 31 August 2021 12:22
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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) ?!?!?

dg9bfc sigi

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.

 

Mike  

 

Sent from Mail for Windows

 

From: Simon Brown
Sent: 31 August 2021 06:41
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Probably.

 

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

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of kb3cs
Sent: 30 August 2021 21:37
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Controlled-Envelope good?
http://www.arrl.org/files/file/QEX_Next_Issue/2014/Nov-Dec_2014/Hershberger_QEX_11_14.pdf

 - 73 -


--

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Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.

 


--
- + - + -
Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.


Conrad, PA5Y
 

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 long time.

 

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.

 

Conrad

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Simon Brown via groups.io
Sent: 31 August 2021 17:45
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Or,

 

Just buy a decent microphone?

 

Simon Brown, G4ELI

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien
Sent: 31 August 2021 16:23
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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)

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 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

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.

 

73

 

Conrad

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien via groups.io
Sent: 31 August 2021 12:22
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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) ?!?!?

dg9bfc sigi

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.

 

Mike  

 

Sent from Mail for Windows

 

From: Simon Brown
Sent: 31 August 2021 06:41
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Probably.

 

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

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of kb3cs
Sent: 30 August 2021 21:37
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Controlled-Envelope good?
http://www.arrl.org/files/file/QEX_Next_Issue/2014/Nov-Dec_2014/Hershberger_QEX_11_14.pdf

 - 73 -


--

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Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.

 


--

- + - + -

Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.


jdow
 

Reverb? That's the "Good buddy" sound. A relatively flat spectrum with a broad notch at about 1.2 kHz, depending on the voice, concentrates your power where it does good. For DX and contests reduce the bass, too. For ad hoc on the air chats over a virtual cafe table raise the bass no more than 6dB. Going for the voice of God sound with or without the reverb and other "stuff" tell me what I've run across is not people I want to talk with.

{^_-}

On 20210901 08:19:41, Conrad, PA5Y wrote:

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 long time.

 

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.

 

Conrad

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Simon Brown via groups.io
Sent: 31 August 2021 17:45
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Or,

 

Just buy a decent microphone?

 

Simon Brown, G4ELI

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien
Sent: 31 August 2021 16:23
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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)

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 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

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.

 

73

 

Conrad

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien via groups.io
Sent: 31 August 2021 12:22
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

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) ?!?!?

dg9bfc sigi

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.

 

Mike  

 

Sent from Mail for Windows

 

From: Simon Brown
Sent: 31 August 2021 06:41
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Probably.

 

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

https://www.sdr-radio.com

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of kb3cs
Sent: 30 August 2021 21:37
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Speech Processor

 

Controlled-Envelope good?
http://www.arrl.org/files/file/QEX_Next_Issue/2014/Nov-Dec_2014/Hershberger_QEX_11_14.pdf

 - 73 -


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- + - + -

Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.

 


--

- + - + -

Please use https://forum.sdr-radio.com:4499/ when posting questions or problems.