Audio clippers: Friend or Foe?

[WD5JKO]

   This is a good discussion. A few years back I bought a kit from K7DYY. This was a sophisticated mic compressor that is intended to go into the base of a D-104 microphone, and to set the speech level to drive one of the K7DYY Class D AM power transmitters. There are two designs, and the latest has just one pot. That is what I got. I was astounded to see how the thing would saturate (clip) on voice peaks during the processor "attack time". Apparently the "Super Senior" must have a clipper inside to minimize this effect. I was using this D-104 with my Gonset G-76 and Central Electronics 20A. I ended up adding a clipper to the circuit which only gets hit (limits) on the first voice peak after a long pause. After doing that, I absolutely love the thing.
   
  The following article from old man James Tonne, W4ENE covers this topic of clipping and post clipping processing very thoroughly. I suggest you print this out and read several times since it is choc full of good information. He had a QST article back in 1956 called "Compression and Clipping". His more recent work adds to that earlier article after nearly 60 years have passed by.

http://tonnesoftware.com/appnotes/speech/speechamp.html

Jim
Wd5JKO

[K1JJ]

Yes, the attack time on many music store processors - AND even on some well known AM processor/ limiters can be very slow.     I had to get rid of several units because they would produce a square wave on the first hard syllable.  Try a "tock" sound with your tongue and see what it looks like on the RF scope or output of the processor.  You may be surprised.   A pulse / square wave into a plate modulated system can cause havoc on parts and produce splatter.  

Of course, the newer software and digital boxes have "look ahead" methods that will limit leading edges well. Just be sure to test your complete system together with the fast rising "tock" of the tongue. You never know if some stage in the audio chain is not acting well.

I finally settled on an older CRL  PMC-300A peak modulation processor. The attack time was fast enuff to handle anything. This is used in conjunction with the QIX neg peak clipper circuit.    

I've used another unit called the Aphex Dominator II that is quite fast. It is a later model and often used as a failsafe to limit rogue audio spikes - to save musicians' eardrums when using headphones.

T

[WBear2GCR]

A few thoughts.

I'm unclear why a typical ham would need or want compression.
Seems to me that a soft knee limiter is all that is required.

The chip in the K7DYY mike bottom is a fairly sophisticated unit that has several functions built in, if I am recalling properly it can be set up to do any or all of them with various different parameters. Of course that still doesn't mean that it couldn't simply not be able miss that first attack. Imho, the earlier chip looked better as far as specs on paper compared to the later one. I have not tried them, but I looked very hard at them with an eye to using them in a multi band processor set up. Never did it.

                    _-_-

[AB2EZ]

A question:

I have reasonably normal hearing.

When I listen to people talking (not over the radio, but when they are standing near me in a quiet room), they all sound different.

They don't sound distorted, or "unnatural" in any way... just different.

If I understand, correctly, the very elementary physiological aspects of how humans produce speech, and how hearing works ... there are a combination of linear and non-linear processes taking place that make one "natural" sounding individual's voice sound different from another "natural" sounding individual's voice.

Likewise, voice processing can make someone sound different, but sill sound "natural"

What types of linear and non-linear processes can be applied, between the input of a microphone and the output of a speaker, that will only change the way someone sounds; but, will not be perceived as producing an "unnatural" sounding result?

What types of linear and non-linear processing effects are most objectionable, in terms of making natural sounding voices sound unnatural?

Stu

[K1JJ]

A question:


What types of linear and non-linear processes can be applied, between the input of a microphone and the output of a speaker, that will only change the way someone sounds; but, will not be perceived as producing an "unnatural" sounding result?

What types of linear and non-linear processing effects are most objectionable, in terms of making natural sounding voices sound unnatural?

Stu

Hi Stu,

First, I think all forms of processing, EQing, limiting and hard clipping are non-linear. Put a sinewave thru and look at the departure from the perfect sine.  Hard clipping is probably the harshest, with softer audio limiting cleaner.  EQing produces phase shift problems too. They are all  forms of distortion.

If the audio gear is set up properly, then I think generating an unnatural sound is a matter of degree. Most techniques can sound good if used sparely.  But, almost any processing technique, when pushed past a certain point will make the sound unnatural, thus the added "distortion"  sounds poor. And, it's a matter of opinion where this point is.

For example, 1 dB of compression will not be noticeable.  20dB of compression will make most people cringe.

Positive peaks:  If we use a negative peak clipper and run 110% positive and 95% negative, no one will notice. But run 200% pos by slamming the clipper hard, then many people will notice and some will object. Certainly some diode detectors will complain.

Add 1 dB of bass boast and no one will notice.  Add 15dB with a Big Bottom box and they will beat ya up.

This is why it can be an endless quest to make everyone happy. There will always be a few who will not like the sound.  It's a compromise. If we are lucky, eventually our buddies will like what they hear, we get a few good unsolicited audio reports - and we like what we hear of ourselves on the playback recording.  

T

[ka4koe]

I've noticed several versions of the qix 3 diode clipper. Which one is preferred?

Tnx

Philip

[K1JJ]

I've noticed several versions of the qix 3 diode clipper. Which one is preferred?

Tnx

Philip

Hi Philip -

I'm talking about a low level circuit - that is a high level one.

The 3-diode clipper is a high level circuit that goes between the modulation transformer and before the final.  Personally, I have tried that circuit on at least five plated modulated rigs and later removed them due to IMD problems when hit hard.  They are probably OK as an occasional safety net, however.  

For a NPL clipper, I prefer the one diode low level circuit as found in QIX's PDM generator circuit. This can be found somewhere on his website.  There is also a thread on this BB about it with a circuit too.

If you were to order one of his kit PDM gen boards, it would be contained within it  - and also uses a 6 KHz low pass filter after the clipper stage.   I use one on the output of my AM audio chain (after the audio soft limiting) to handle neg peaks on the various PDM and plate modulated rigs.

I use a touch of two techniques together - the PMC-300A peak limiter that limits both pos and neg peaks  -   and the QIX single diode neg peak clipper with 6Khz LP filter.

T

[N2DTS]

Something often sounds better then nothing.
Some guys who run a standard old rig, with mods or without, with just a microphone into the rig often have very low weak audio.
Back away from the mike and you get 50% or less modulation.
There is someone I talk to from time to time who sort of drifts down in level, down to 20% modulation, I will say something, and its back up for a short while, then back down.
Your standalone rig has only modulator current to tell about the modulation level, and some set and forget compressor can help a lot. It gives lights to look at, and some compensation for the audio level.
A 50 to 100 watt rig needs to have a good amount of audio on it, a weak signal at 50% mod is just annoying.

On lower power rigs, I always used to set the audio to overmodulate a bit.
If a little is good, more is better!
 

A few thoughts.

I'm unclear why a typical ham would need or want compression.
Seems to me that a soft knee limiter is all that is required.

                    _-_-

[Opcom]

For those wanting a 'look ahead' without nasty digital stuff, a 50FT coil of garden hose put neatly inside a box makes a decent delay. It is not flat or perfect and I have not done this on the air, it was before I had a license or radio transmitters. The delay reminded me of it.

A small 1-2" speaker at one end is the driver and any sort of transducer can work at the other end. A crystal earphone worked as a pickup in my experiment, based on an old popular electronics project for making 'delayed' rear channels for a hi-fi system. The article didn't complain about reverberation but it was  a slight issue I overcame mostly by putting some thick felt at the pickup end of the hose, around the ear-canal part of the earphone. It did ok for the intended purpose and YMMV.

50FT is long, 50ms delay, and shorter could be used to get past that first 100-1000us I guess. I didn't try that but thought about resonances at such a short length.

There is an audio recording from someone here, unfortunately for all their effort they allowed distortion that should not have been in there. duh. One channel is 50FT and the other 15 FT. 15FT seems much better. That is what, 15ms? enough.
http://theb-roll.com/garden-hose-stereo-delay/

Anyway there's an analog delay that could be made to work but with a lot of effort.  There is always tape as a delay, but that has moving parts.

[N1BCG]

Well, this has turned into a very interesting and informative posting...thank you all!

I mentioned earlier that I would report on the "final product" for using the Valiant with the original clipper circuit. We all seem to agree that it's undesireable to drive any clipper with raw audio even though that's exactly what the Valiant does. I imagine the EF Johnson designers decided to leave it at that in the interest of simplicity and pricing.

IF the clipper is to stay, something needs to limit the audio before the 6AL5 sees it, and I found a curious little compressor made by Waters that does just that:



Delightfully, the 359 features a choice of 1/4" three conductor jacks or screw terminals (the tip carries the PTT closure) and takes a microphone level input and reduces the dynamic range.

For more flexilbility, I ended up using a combined pre-emphasis circuit & compressor with microphone level output to feed my Valiant, but I wanted to showcase the 359 because it's a neat find.

[N2DTS]

Early in my AM operating, I used some sort of signal corps tube type compressor.
Transformer input and output, and it had a round meter on the front marked gain reduction.
At the time, I thought it worked quite well, it was rack mount and cool looking.
The next thing I tried was a solid state DBX unit that could do expansion or compression, and even had a threshold setting.
I still have it, its pre IC chip stuff.
You were supposed to compress the audio before recording it on tape, then expand it on playback to drop the noise floor.
For keeping the audio in a general range, both worked, but likely with a bit of overshoot.
Back then, I think most guys were crappy sounding and wide, along with other issues, at least by todays standards...


 

[YB1AHY]

Hi

I love this topic.

i would like to have a kind of "confirmation" regarding the allpass filter or sometimes said as a phase scrambler. But still related to the main topic.

As the description of all pass filter which is delaying audio starting from 300hz and more delay on higher audio freq. And my perception when receiving the transmission from AMer who use all pass filter, it seems that the talk punch is increasing. Am i right at this point? my knowledge said that using all pass filter, amplitude are spreading all over the audio, so the waveforms will become "fat" with less "thin" peaks.

So, when the allpass filter is installed do you think it is not necessarily to clip the audio after allpass in order to increase talk power (loudness).

Thank you for replying my email.

Best Regards

Agus



   

[N1BCG]

Hi Agus!  Phase rotators (or All-Pass filters) almost deserve their own thread. Basically, they were developed for broadcasters as a way to reduce the amount of gain reduction needed for a given output level by reducing the amount if asymmetry in program content, particularly voice.

Since the DC voltage used for gain control in compressors comes from the audio signal, higher peaks in one polarity cause greater gain reduction than is needed. The use of a phase rotator gives compressors that follow a more consistent waveform to sample.

They do this by gradually rotating the phase throughout the audio spectrum so that asymmetry in the different frequency ranges tend to cancel each other out. Think of it as zero phase shift at 100 Hz, 180 degrees at 3 kHz, and a full 360 at 10 kHz. That's just an example...

You can only hear a difference if you wear headphones since the audio mixes with the sound of your voice conducted through your jaw, but to anyone else listening, they won't hear any phase or equalization effects. You will, however, be louder since your compressor will not be reducing gain as much as without a rotator.

[Steve - K4HX]

You are mostly correct. The allpass filter will reduce the asymmetry of the waveform. So some peaks will be reduced.