Date   

Re: Speech Processor

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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Re: Speech Processor

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 -


--

- + - + -

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.


Re: Speech Processor

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 -


--

- + - + -

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.


Re: Speech Processor

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 -


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


Re: Speech Processor

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.


Re: Speech Processor

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

 

 

 


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


Re: Speech Processor

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

 

 

 


Re: Speech Processor

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 -


--

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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.


Re: Speech Processor

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.


Re: Speech Processor

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.

 


Re: Which version of OmniRig?

Rick Ellison
 

Go to www.dxatlas.com then go to the download section and you can find it there..

 

73 Rick N2AMG

 

From: main@SDR-Radio.groups.io [mailto:main@SDR-Radio.groups.io] On Behalf Of Dustin
Sent: Tuesday, August 31, 2021 8:06 AM
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Which version of OmniRig?

 

Hi all!

Do any of you have a download link for 1.19?

Thanks for any help!

Dustin - KD9QGP


Virus-free. www.avast.com


Re: Speech Processor

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.

 


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


Re: Which version of OmniRig?

Dustin
 

Hi all!

Do any of you have a download link for 1.19?

Thanks for any help!

Dustin - KD9QGP


Re: Speech Processor

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.

 


Re: Speech Processor

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 -


--

- + - + -

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

 


Re: Speech Processor

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.

 


Re: Speech Processor

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 -


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


Re: Speech Processor

kb3cs
 


Re: Audio Selection

Conrad, PA5Y
 

I think that VB Audio is a component but the routing is different. VB Cabe does not support ASIO which is very significant and the routing is much less sophisticated. I use ASIO when possible. Sadly WSJT-X does not use ASIO due to licensing restrictions.

 

So no it is not merely a graphical overlay.

 

Regards

 

Conrad

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of f1hdi via groups.io
Sent: 30 August 2021 11:29
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Audio Selection

 

Hi Conrad,

You lost me, is voice métier potatoo not à graphical overlay mixer of vb audio cable (and physical ones)?

 

 

 

Envoyé de mon Samsung Galaxy Note10+ Orange

 

 

-------- Message d'origine --------

De : "Conrad, PA5Y" <g0ruz@...>

Date : 30/08/2021 10:25 (GMT+01:00)

Objet : Re: [SDR-Radio] Audio Selection

 

Hello Jean-Marc.

 

I think that we are not comparing the same thing. Perhaps this is entirely due to overhead and the fact that you are running a virtual machine. Also vanilla VBcable is not the same as Voice Meter Potato. I can assure you that on EME Voice Meter Potato (and probably other weak signal modes) completely out performs VAC.

 

My findings are contrary to yours.

 

Regards

 

Conrad PA5Y

 

 

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of f1hdi via groups.io
Sent: 30 August 2021 08:31
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Audio Selection

 

Hi Conrad,

I have counter experience regarding vac and vbcable on a 24/7 spyserver clients.

The setup with sdr# (connected to the spyserver running an R2 ) -> vac -> wsjt-x (running wspr decode on 2m) is rock stable whereas

sdr# (connected to the spyserver running an HF+) -> vac -> wsjt-x (running wspr decode on 2m) is running few days (then sdr# stuck)

I tried to substitute vac with vb with poorer results both cases.

So this is no evidence on my side which virtual cable is performing the best.

I will try a c"client" running sdrcv3 instead of sdr# but the remote system is a windows VM under linux on an i7 small mini pc, so the horse power is minimal.

73 Jean-Marc

Le 29/08/2021 à 13:11, Conrad, PA5Y a écrit :

Voice Meter Potato (VMP) also gets my vote. I use it for many tasks including running demodulated audio from SDRCv3 or Linrad (for EME)  to WSJT-X.  I always found Sample Rate Conversion artefacts when using VAC. It has a similar effect to poor phase noise on weak signals with up to 2dB degradation when compared to a radio (K3S or TS-890) and a soundcard. This does not happen with VMP. I don’t know this for sure, but I think if you use soundcard with a decent clock such as a UMC202HD as the master output device this becomes the reference clock. I could be wrong as this is not made clear. However, I am sure that VMP is much better than any other virtual soundcard solution that I could find 18 months ago. It also has ASIO on the virtual inputs which has much lower latency and does not get mangled by Windows.

 

You can monitor any signal via the main output device. Like a real mixer it as solo buttons.

 

Regards

 

Conrad PA5Y

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien via groups.io
Sent: 28 August 2021 21:21
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Audio Selection

 

i use potatoe from the same company ... works superb

also downloaded vac plus vac_a and vac_b and hificable

so now have 4x vac besides the 3 that potatoe brings with .. means 7 virtual cables used here together with the full blown mixer

(and additionalli installed lighthost and a few vst plugins to give my audio a better punch and sound)

so .. i can highly recommend the mixer and the virtual cables

greetz sigi dg9bfc

 

 

 

Am 28.08.2021 um 18:30 schrieb Max Mucci, N5NHJ via groups.io:

Hello Gedas,

A very helpful software mixer I use for almost everything:

VOICEMEETER by VB-AUDIO – The world´s most used Virtual Audio Mixer & Sound Tool 

 

Friday, August 27, 2021, 5:23:20 PM, you wrote:

 

Using a RSPdx with SDRC V3.1 and Laptop with W/10.

 

All works aok but then I had the bright idea to install VB Audio to get some VAC going so I could run WSJT etc.

 

That went smooth and was able to make many FB decodes overnight.

 

1. I wish there was a way to be able to hear what the RSPdx is tuned WHILE it's audio is being ported over to WSJT via the VAC. Is there an easy way of monitoring what the VAC is transferring?

 

2. The next part has me scratching my head......I am done with WSJT and just want to do some listening on this W10 laptop with the RSPdx. When I go to SDRC/OPTIONS/AUDIO/PLAYBACK/DEVICES I can see my Realtek PC speakers on one line and the VAC on the 2nd line.

 

I select the Realtek PC speakers, hit OK then re-start SDRC hoping to be able to hear the receiver via the speakers but get zip. Is there an easy way to switch from one to the other? I thought the OPTIONS tab is where I make this selection but apparently not. Thanks

 

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.

 

 

 

 

 

 

 

 

 

----------------------

73 de Max, N5NHJ (I8NHJ)


Re: Audio Selection

Conrad, PA5Y
 

Ah OK.

 

I should have explained this better. The document was related to VAC and not VMP. I think that VMP is more sophisticated but this is an assumption based on my observations. One produces artifacts, the other does not.

 

Regards

 

Conrad

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of jdow via groups.io
Sent: 30 August 2021 12:27
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Audio Selection

 

I scanned the link you provided.


When a Virtual Cable is used in a path containing another audio device (a hardware device, a software device other than Virtual Cable and even another Virtual Cable device), clock difference effect occurs. To minimize it, change cable clock correction multiplier until cable internal clock speed becomes closest to another device clock.

...


VAC uses relatively simple linear resampling algorithm and does not use dithering or other advanced smoothing features. Therefore, conversion results may sound worse than an original signal. To prevent quality degradation, you need to pay attention to a format matching. In an ideal case, all three formats (the playback client's format, the cable format and the recording client's format) are the same.


{^_^}

On 20210830 01:22:58, Conrad, PA5Y wrote:

Hello where did you read that? I had always assumed that it did and was rather better at it than VAC.  I am very interested to learn how this stuff works.

 

Conrad

 

 

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of jdow via groups.io
Sent: 30 August 2021 00:41
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Audio Selection

 

I see, VMP does not even try to match clocks dynamically. I guess its author figures that's a bear and a half, too.

{o.o}


On 20210829 15:14:16, Conrad, PA5Y wrote:

That is the best-case scenario for virtual soundcards, but it is dependent on the routing, whether you are using MME, WDM/KS or ASIO . But virtual soundcards have a buffer, to help with under runs or over runs and therefore data may not arrive when expected. To some extent there will be some sample shifting due to timing errors. This means that degradation can occur.

 

But VMP handles all of this very well and certainly produces the best results.

 

If you read this:

 

https://documentation.help/Virtual-Audio-Cable/advanced.htm

 

It is all explained, but you must read the whole thing as well as some hyperlinked pages to get a picture.  I wish that I could find a better reference, there is a lot that I don’t understand, and it really affects my EME operating. If your signals are well above the noise floor then this issues are a lot less important, or at least that is what I observe.

 

Conrad

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien via groups.io
Sent: 29 August 2021 23:21
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Audio Selection

 

What if the soundcard is just a virtual card.....and you have a handfull of them all calculated from your main cpu clock...?!? 

How "exact" are they?? 

 

Dg9bfc sigi 

 

Am 29.08.2021 21:34 schrieb "Conrad, PA5Y" <g0ruz@...>:

Exactly!

 

Conrad

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of jdow via groups.io
Sent: 29 August 2021 21:34
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Audio Selection

 

Dream along with me.....

You'd think that, wouldn't you; but, only when you figure crystals are all perfectly on frequency including your sound card crystals....

{^_-}

On 20210829 09:30:34, Siegfried Jackstien wrote:

Set all sampling rates to same setting 48000 then no resampling problems...

Greetz sigi dg9bfc 

 

Am 29.08.2021 17:36 schrieb Gedas <w8bya@...>:

Hiya Conrad, long time. Thanks !

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 8/29/2021 7:11 AM, Conrad, PA5Y wrote:

Voice Meter Potato (VMP) also gets my vote. I use it for many tasks including running demodulated audio from SDRCv3 or Linrad (for EME)  to WSJT-X.  I always found Sample Rate Conversion artefacts when using VAC. It has a similar effect to poor phase noise on weak signals with up to 2dB degradation when compared to a radio (K3S or TS-890) and a soundcard. This does not happen with VMP. I don’t know this for sure, but I think if you use soundcard with a decent clock such as a UMC202HD as the master output device this becomes the reference clock. I could be wrong as this is not made clear. However, I am sure that VMP is much better than any other virtual soundcard solution that I could find 18 months ago. It also has ASIO on the virtual inputs which has much lower latency and does not get mangled by Windows.

 

You can monitor any signal via the main output device. Like a real mixer it as solo buttons.

 

Regards

 

Conrad PA5Y

 

From: main@SDR-Radio.groups.io <main@SDR-Radio.groups.io> On Behalf Of Siegfried Jackstien via groups.io
Sent: 28 August 2021 21:21
To: main@SDR-Radio.groups.io
Subject: Re: [SDR-Radio] Audio Selection

 

i use potatoe from the same company ... works superb

also downloaded vac plus vac_a and vac_b and hificable

so now have 4x vac besides the 3 that potatoe brings with .. means 7 virtual cables used here together with the full blown mixer

(and additionalli installed lighthost and a few vst plugins to give my audio a better punch and sound)

so .. i can highly recommend the mixer and the virtual cables

greetz sigi dg9bfc

 

 

 

Am 28.08.2021 um 18:30 schrieb Max Mucci, N5NHJ via groups.io:

Hello Gedas,

A very helpful software mixer I use for almost everything:

VOICEMEETER by VB-AUDIO – The world´s most used Virtual Audio Mixer & Sound Tool 

 

Friday, August 27, 2021, 5:23:20 PM, you wrote:

 

Using a RSPdx with SDRC V3.1 and Laptop with W/10.

 

All works aok but then I had the bright idea to install VB Audio to get some VAC going so I could run WSJT etc.

 

That went smooth and was able to make many FB decodes overnight.

 

1. I wish there was a way to be able to hear what the RSPdx is tuned WHILE it's audio is being ported over to WSJT via the VAC. Is there an easy way of monitoring what the VAC is transferring?

 

2. The next part has me scratching my head......I am done with WSJT and just want to do some listening on this W10 laptop with the RSPdx. When I go to SDRC/OPTIONS/AUDIO/PLAYBACK/DEVICES I can see my Realtek PC speakers on one line and the VAC on the 2nd line.

 

I select the Realtek PC speakers, hit OK then re-start SDRC hoping to be able to hear the receiver via the speakers but get zip. Is there an easy way to switch from one to the other? I thought the OPTIONS tab is where I make this selection but apparently not. Thanks

 

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.

 

 

 

 

 

 

 

 

 

----------------------

73 de Max, N5NHJ (I8NHJ)

 

 

 

 

 

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