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Custom Audio Filter designs for Ham AM - Dynamic Bandwidth Control




 
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Author Topic: Custom Audio Filter designs for Ham AM - Dynamic Bandwidth Control  (Read 11419 times)
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w8khk
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« Reply #25 on: February 12, 2020, 11:16:59 PM »

Over a year ago I experienced the same shortcomings with the LM386, and removed it from my design.  Instead, interfacing to balanced 600 ohm input and output via op amps proved to be superior.

It would be interesting to see a comparison of the SCAF implementation to your multi-pole brick-wall filter solution, and this could be analyzed using the existing MAX processor PCA with minimal effort.  I am not sure there would be an advantage in using both the SCAF and the passive multi-pole filter.  It will be enlightening to see results of the two filter methods in tandem!  In any case, it is crucial to remove all remnants of the clock signal, post SCAF, from the remaining stages of the processor.  

One consideration is that the passive approach is keyed to a single cutoff frequency per filter, whereas the SCAF is continually adjustable, and may be preset to the desired bandwith(s) by switch selection of multiple clock frequency trim-pots.  Our testing previously indicated that a single SCAF was insufficient, but two SCAF filters in cascade provided an exemplary brick wall.  Further confirmation of this assumption would be invaluable.

Our goal is to produce a complete solution to the AM processor challenge, in a single device, and that may be based upon SCAF or passive filter approach,  determined by lab instruments and on-the-air performance testing.  Discussion thus far has focused on just the filter; but the processor includes all the necessary elements to produce a strong, clean, high fidelity signal, from the microphone all the way to the modulator input.  

It is certainly the right time to get more folks involved; so far we have been a team of two working for several months to arrive at the present configuration.  Results thus far have been very encouraging, but there is always opportunity for improvement!

I will be in touch and provide some beta hardware for your continuing analysis.  

73, Rick
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« Reply #26 on: February 13, 2020, 12:58:16 AM »

Hi Rick -

Well, I just ordered the parts for the 10th order 5.5 KHz Butterworth LP filter. The bill of materials for Mouser is below if anyone wants to build it. Or let me get the arrows in my back first...  Grin  It's only 5 inductors and 5 capacitors. A few parts needed to be paralleled to get exact near values.  $10 plus shipping ain't a bad price to pay for a tighter, more controlled signal, right?

I should be testing it in a few days.  I desperately need  control of my extreme highs and hope to leave it inline for all my rigs until we make progress on the Maxim chip.
Yes the 10th order filter in cascade may just be too much adding excessive phase shift that is not a good trade-off compared to a smaller LP filter when using the SCAF. We'll have to see.

And yes, we should focus on the other areas of the Max shortly and maybe by the time it is ready for production we will all understand it enough to feel comfortable running it.

I do like the possibility of getting rid of some redundant audio processors in the shack. For instance I have a noise gate for ssb blower noise in one box and a desser in another box. The 31 band EQ is necessary. My favorite box is a CRL PMA-300A limiter  (compliments of my Secret Santa, W2NBC) which limits the positive peaks at 150% while the negative peaks at -99%. I love the way it is invisible until the highest peaks are hit and then it's soft knee gets to working. Hopefully the Max's limiter will work this way.  I don't like audio compression much and have only about 3 DB in... it brings up blower noise and is always working.  So as you can see the possibility of getting rid of the clutter with one Max box and an EQ sounds nice.  Who knows, maybe a later version will have an optional Desser and noise gate module.

You mentioned other modes. Does this have some utility on ssb or essb?

T

5.5 KHz Filter Mouser Electronics Bill of Materials below:

* FILTER 5.5 KHz Audio.pdf (31.11 KB - downloaded 74 times.)

* Filter 5.5.png (10.69 KB, 535x187 - viewed 130 times.)
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« Reply #27 on: February 13, 2020, 11:04:06 AM »

Tom, that is good information to help us move forward. I will await test results on your passive filter.

The goal of eliminating much of the rack processor equipment has been a priority in the design  of this "all-in-one" processor, but we realize that folks will still want to use multi-band equalizers and other tools.  This is the main reason we provide several line in and line out circuits compatible with balanced 600 ohm devices.  Our inputs are higher, and outputs are lower impedance than 600 ohms, which is the norm today for inter-operability.  All the modules fit together like an erector set, such that you may use an input/output pair for insert send and return functionality.  I believe this "feature" could be leveraged to interface your passive filter, using resistive loading so that it "sees" the proper input and output impedances, and let the existing op-amp circuits bridge to your existing chain with balanced 600 ohm links for noise and RFI immunity.  Your single-ended filter would lie inside the "circle-of-protection" afforded by these interfaces.

Each module in the processor includes a four-pin header to allow the user to include or exclude that stage from the signal path.  Pin one is ground, such that shielded cable may be used, but so far we have found that to be unnecessary.  Pin two is the signal into the current module, and pin three is the "send" to the next module, while pin four is the output of the current module.  To include the current module, connect pin four to pin three, or to exclude the current module, connect pin two to pin three, either with an on-board jumper or a panel-mounted SPDT switch.  

With this modular approach, you will find it very easy to insert your passive filter array in place of the existing SCAF, or in cascade, your choice.

The Maxim chips include an additional op amp.  On the first Maxim chip, we use the op amp as an input interface to avoid loading the previous stage, and to provide unity gain through each module, as necessary to avoid level shifts when inserting or removing a module on the fly.  (Due to the operation of the internal voltage controlled amplifier, the Maxim compressor input exhibits a rather low and varying input impedance.)  The op amp on the second Maxim chip is used as an output buffer to avoid loading the  Maxim chip, no matter what module is connected following the SCAF stages.  In this op amp circuit we have added R148 and C148 to attenuate the clock.  This should have minimal impact on the audio, because the clock is 10 to 30 times the frequency of the voice components.

I have attached a few "snips" of the circuit to illustrate these components that remove the SCAF clock from the audio.  You will also notice fixed resistors R145 and R146 in parallel with trim-pot R147.  This pot is used to fine-adjust the SCAF module to unity gain, and the fixed resistor pads may be used to lock down the attenuator, thus eliminating the pot in the final design.  You will see this combination in several other areas of the device.  The extraneous components will be removed from the final artwork, thus reducing the setup and calibration steps to a minimum.  EDIT: For clarity, I should mention that either the pot, or the fixed resistors will be present, but both will not be present at the same time.  The fixed resistor values (ratio) may be determined based upon the required pot setting during fine tuning.

Two additional optional LPF for clock elimination are in the main output driver stage, R221 and C222 and the post-clipper stage, R209 and C201.  All these are optional, and the goal with this board rev is to determine the bare minimum of filtering with total removal of the clock signal.

As it is obvious we will have at least one more revision of the artwork to nail down all fine-tuning issues, it is certainly possible to add more features needed to eliminate other external components in the audio chain.  

Currently the features following the input buffers include a low-cut filter to reduce the amount of bass power and thus improve readability in heavy QRM or QSB situations, but this is done "gently" to maintain good voice quality.  Following this filter is an optional phase inverter, to allow selection of high positive peaks.  The all-pass filter, AKA phase rotator is the next optional module.  Prior to compression and limiting, we added pre-emphasis, and when used in conjunction with the low-cut filter, the signal is significantly more understandable  when band conditions are rough.  A separate op amp and comparator are used to visually indicate normal voice level and high peak level, as an aid in adjusting microphone level.  The comparator also controls LED indicators to identify whether positive or negative peaks are predominant.  The compressor/limiter is so tight, a clipper might seem unnecessary.  But for the occasional spike that might not be caught by the limiter, a soft clipper is provided.  The clipper may be set off, or 100% for pos and neg peaks, or 125% pos and 100% neg.  You mention using 150% pos, and this could easily be added as a third switch position.  With soft clipping, minimal LPF is required.  As described above, this LPF may double as a clock attenuator as well.  The compressor response time may be adjusted for open or dense performance, and the novel "gain gate" allows the level of compression to remain static in the absence of voice.  This might preclude the necessity for the noise gate you are now using. Lastly, several output circuits are offered to drive multiple transmitters, monitoring functions, metering, etc.  This allows the input level to each modulator to be set for each transmitter, thus simplifying band QSY.

Your suggestions on documenting voltage and signal levels, as well as troubleshooting information and circuit functions, are all in the works and will be available on the web as the project is finalized.

While we are primarily focusing on a device for the avid AM'er, I see absolutely no reason why it would not work for SSB, FM, or even digital voice.  LPAM is another possible application.  After all, we are simply providing a clean, consistent signal to a modulator, nothing unique to AM-only whatsoever.  

One area where this little box might shine is in the schools when hams set up for students to chat with the ISS astronauts.  In cases I am aware, the teachers ask students to submit the questions they would like to ask the astronauts.  Then students vote on the questions, and the persons who submit the most popular questions get the opportunity to actually talk to space!  Some are shy, and barely whisper, while others may be so excited they almost yell.  

I recall when testing, Clark tried speaking softly, then he yelled loudly into the studio mic.  The compressor and limiter performed admirably.  Then he took a long pause.  He told me the noise I was hearing was not a bug, but it was his dog snoring in the living room.  Hence the development of the "gain gate", adjusting compressor gain only when voice is detected.  Might this preclude the need for a traditional noise gate?

I will coordinate the non-technical details of getting test hardware to you offline.



* SCAFlpf.JPG (19.43 KB, 327x268 - viewed 118 times.)

* MAINlpf.JPG (18.14 KB, 441x183 - viewed 114 times.)

* CLIPlpf.JPG (22.95 KB, 383x220 - viewed 123 times.)
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« Reply #28 on: February 13, 2020, 02:38:19 PM »

Hi Rick,

That's a lot of FB info. I read it over a few times to absorb it all.

I see instead of one LP filter for the clock you have scattered filter components around to work as an overall system. I also like the module and in/out approach to keep things versatile and also give users the chance to bypass things they don't need or add in things they do.

I made my first mistake on this prototype passive filter by ordering five ceramic capacitors. It turns out film polystyrene caps are better for audio. I will get the right ones for the real circuit boards assuming the tests work out reasonably well. I plan on using a 600 ohm resistor shunt on both the input and output to match the filter's requirements.   Turns out my own insertion point is high impedance in/out, so this will work to reduce the ripple.

Let's look forward to testing your SCAF version soon and later on to the whole unit.

You obviously have invested a lot of expertise, time and money into the project so far and a successful outcome will surely be coming.

  T
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« Reply #29 on: February 13, 2020, 03:49:05 PM »

Tom:

The Maxim Elliptical SCAF is just about as as sharp as the sharpest DSP filter. My dual-SCAF was something like 36dB/octave after the cutoff point. Really you don't need that sharp of a filter.

Rick: The LM386 suffers from high amounts of crossover distortion, and noise. Have you considered using a second order Sallen-Key filter for your Main LPF?

https://www.edn.com/a-sallen-key-low-pass-filter-design-toolkit/
http://sim.okawa-denshi.jp/en/OPseikiLowkeisan.htm

Enjoying watching both of your developments!

73,
Dan
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« Reply #30 on: February 13, 2020, 08:13:34 PM »

Dan, Thanks for your input and interest.

The LM386 has many known limitations and is not used in the audio processor.  Thanks for your suggestion on the LPF.

I attach a snip of the LPF section we used in version two.  It is no longer included, because the SCAF performs the required bandwidth filtering, while a simple RC added to an op amp following the SCAF eliminates the clock remnants.

I also attach a snip of the current low-cut filter, which is the complement of what you suggested, with the resistor and capacitor positions exchanged to provide a high pass filter instead of a low pass filter.

73, Rick


* LPFearlyRev2.JPG (91.8 KB, 1015x345 - viewed 112 times.)

* LowCut.JPG (17.11 KB, 265x273 - viewed 97 times.)
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« Reply #31 on: February 13, 2020, 09:36:23 PM »

Rick,

In your Max trials using the Maxim SCAF, did you or Clark notice that the natural asymmetry of the voice was stripped to become  more symmetrical; something similar to what an all-pass filter (phase rotator) does to audio?

A friend of mine suggested that a brickwall filter may do this to a signal in the form of phase distortion.  Did you do any phase shift in/out measurements?

I know some guys like symetrical dense audio. I prefer high dynamic range and assemtrical positive peaks that naturally form.  So I will be looking closely at any phase distortion effects.

My friend felt that a more mild roll off would preserve the asymmetry.   Until we run some tests, what do you think?

T

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« Reply #32 on: February 13, 2020, 10:01:04 PM »


A friend of mine suggested that a brickwall filter may do this to a signal in the form of phase distortion.  Did you do any phase shift in/out measurements?

Tom, you bring up an interesting point.  I have observed the phase distortion in the all-pass filter, as expected.  I have not specifically tested the SCAF for phase distortion.  I will do this test soon and report back.  Perhaps the resulting signal is more symmetrical because some of the higher frequency components are the major contributors to the asymmetry?

If the SCAF proves to cause phase distortion and creates a more symmetrical output, there certainly may be cases where a passive multi-pole brick wall filter is optimal.  In this case, perhaps the new processor could include both the SCAF and the passive filter as operator-selectable options.  Time will tell....
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« Reply #33 on: February 13, 2020, 10:15:17 PM »

Hi Rick,

OK on all.

Unfortunately all filters including passive filters like mine will cause some degree of phase distortion.  That's why I am worried about my large L/C filter.  I have tried an all-pass filter (phase rotator) before and hated the effects cuz I lost my big positive peaks. But I know guys who run an all-phase filter deliberately to get symmetrical audio and they love the big carrier and dense audio at 110% positive.

It this happens with mine, I will cut it to half size and try again.

A DSP filter module for the Max is do-able, though not easy without some sophisticated work. My bet is a DSP filter will have little to no phase distortion, but I am not sure.  

I'm very interested in what you see with the Max SCAF.  To test it, looking at your peaks with the filter in and out will show the difference in peaks.   Measuring the phase distortion with tones is something I'd have to think about.  Maybe a dual trace scope looking at the the input and output phase difference would do it.  

T
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« Reply #34 on: February 14, 2020, 06:10:05 PM »

Well, Tom, your assumptions appear to be valid regarding phase shift.

I ran a test today with two MAX296 chips in cascade, with a clock freq of approximately 285 kHz, which should yield a bandwidth of around 1/50 of the clock rate, or 5.7 kHz.  The two SCAFs seem to behave a bit like the phase rotator, with the following shifts recorded:

Starting at 0 degrees phase shift at 210 Hz, I recorded

degrees      frequency
00                210 Hz
90             1.42  kHz
180           2.60  kHz
270           4.10  kHz  
00             5.60  kHz
90             6.60  kHz
180           8.40  kHz
270           9.60  kHz
00           12.40  kHz

That is just about 1.5 revolutions of phase within the desired audio bandwidth.  I would assume one stage would generate half the phase shift, but unfortunately a single stage does not provide a steep enough cutoff characteristic.

We have a couple MAX295 chips available for test, but no MAX294 elliptical devices in stock.

I do not know how many degrees our four-pole phase rotator adds.  I can measure it as well, if that information would be of interest.  Perhaps another test of value could be to pass some asymmetrical audio into the SCAF, and see how symmetrical it comes out the other side.

It would be very interesting to see a comparison looking at the phase shift you record, per pole, of your passive L/C bandwidth filter.
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« Reply #35 on: February 14, 2020, 07:58:11 PM »

Much of the MAX audio processing is based on the same principles as broadcast processors. Why reinvent the wheel with over half a century of proven R&D that's implemented in the field? Here are the main points:

The All Pass Filter (APF) reduces asymmetry as a means of increasing loudness. The problem with "as-is" microphone audio is that there is no proper polarity, only majority polarity. It is quite common for lower vocal frequencies to have the *opposite* asymmetry of higher vocal frequencies. You can see this on the polarity indicators when saying "aaaahhhh" and "eeeeeeee" which will light opposite polarity LEDs.

Even when an air chain is carefully checked for polarity it is only correct most, but not all, of the time. Many have noticed that adding a low-cut filter dramatically increases positive modulation. What is actually happening is that the lower vocal components with opposite polarity are getting suppressed.

Another benefit of the All Pass Filter is that it increases average modulation without an increase in compression. This happens because gain limiting stages such as compressors and limiters act on the highest peak of audio regardless of polarity. More gain reduction and lower modulation will occur if the audio is highly asymmetric.

Finally, the use of All Pass Filters yields more consistent asymmetric modulation since the clipper stage acts on each half of the waveform to different degrees, re-creating asymmetry regardless of originating content.

Users who need to be mindful of component ratings greatly benefit from the use of an All Pass Filter since it decreases the chance of arc-overs in output stages or modulation transformer winding damage.

Despite all this, the APF, like every function, can be bypassed in the MAX processor.
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« Reply #36 on: February 14, 2020, 09:03:49 PM »

I've used the Maxim elliptical filter chip in the Rx channel of a couple homebrew transceivers and have never noticed any "phase distortion" effects.  Is it really fair to call it phase "distortion" when its audible effects are unnoticeable? 

I've read several times that phase shifts through audio filters produce inaudible effects.  The all-pass filter (phase rotator) seems proof of that.

What's the big worry??

Rod
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« Reply #37 on: February 14, 2020, 10:19:39 PM »

FB on the tests, Rick.  Thanks.   Good info that we didn't have before.

Clark: I'm still open to trying an all-pass filter again and seeing the benefits you described. I have seen MOST of these effects, especially the lower freqs having the opposite polarity as the highs. And it will even change as the day goes on. Early morning deep voice phase morphs into late day regular voice phase. Trying to optimize polarity can be like chasing our tails sometimes. But once we randomize it, I can see how it is more controllable.

I'll see what my passive filter does to the audio.  I think it will rotate the phase probably as much as the Maxim SCAF did for you, Rick. If so, I'll try a different approach and see what more symmetry does to the overall signal.

Right now I am seeing 150% positive peaks with -99% negative. Maybe it's just an illusion with low average power. It even required switching in the lows cut on the mic as Clark mentioned. Overall, big audio peaks seems too fragile and temperamental to me.  I could be swayed very easily when I try symmetrical audio again.  It would solve my filter roll off problem if I didn't care about phase shift in the filter.

There are guys who do very well like Steve QIX and several of the class E rigs using big asymmetrical peaks. And my good friend Chuck K1KW put in a big effort to get rid of polarity and run denser audio as a result. And he uses a DSP filter system in the ANAN and could have chosen "no phase shift" and high peaks if he wanted. There are several ways to get there.  It's not an easy decision for me but I'm still flexible and could end up with a totally different audio processing approach a month from now..

Clark: When you use an all-pass filter and lose the big peaks, is there still an optimum phase to use? Does it become more stable thruout the audio spectrum and as our voices change?   My guess is if it is really symmetrical, it will be difficult some times to determine the proper phase, if any.

The REA QIX mod monitor makes it very easy to see the proper phase right now for me. I wonder if the REA is still reasonably clear with more symmetrical audio.

**Clark said:   "Another benefit of the All Pass Filter is that it increases average modulation without an increase in compression. This happens because gain limiting stages such as compressors and limiters act on the highest peak of audio regardless of polarity. More gain reduction and lower modulation will occur if the audio is highly asymmetric." **   Very interesting.


T
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« Reply #38 on: February 15, 2020, 09:27:34 AM »

Clark: When you use an all-pass filter and lose the big peaks, is there still an optimum phase to use?

An APF will greatly reduce asymmetry but not eliminate it. Whatever is left will be slight and you could optimize it so that negative clipping is minimized.

A big advantage to broadcasters is ease of handling a wide variety of audio content. Many different mics from different studios, different announcers, tape recorders in the field, ad agency recordings... it's a mish-mosh of inconsistent audio that is "made right" by an APF.

Does it become more stable thruout the audio spectrum and as our voices change?

Yes. Using an APF thoughout adolescence will help stabilize your modulation during those formative years and avoid shame and embarassment.

**Clark said:   "Another benefit of the All Pass Filter is that it increases average modulation without an increase in compression.

Clark says a lot of things, but at least this stuff is useful. Not only do you lose modulation but the extra recovery time adds to the effect. It's important to note that every broadcast processor made in the past 40 years includes an APF. Success leaves clues.


Some important APF usage notes:

1) All Pass Filters must be installed *before* any compression stages.

2) Those that like to listen to themselves with headphones while transmitting will hear a significant difference after switching an APF in-line. This is due to bone conduction where you are hearing your own voice partly through the headphones and partly though your upper jaw. These two paths may be in or out of phase.

This is why switching mic phase makes a HUGE difference in headphones, and an APF will have a similar but less pronounced effect. It's interesting to note that only the op hears the phase effect and no one else.


The SymmetraPeak unit was the original All Pass Filter developed for the broadcast industry decades ago. I've attached a PDF of the brochure. For those wanting a more technical description, the patent can be read here:

http://www.w3am.com/SymmetraPeakPatent.pdf





* SymmetraPeakBrochure-1.jpg (604.01 KB, 1275x1650 - viewed 115 times.)
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« Reply #39 on: February 15, 2020, 10:02:34 AM »

K1JJ,
What happened to your simple passive fiter idea to NARROW BANDWIDTH for AM transmitters??
Did they talk you into working on digital audio processing instead?
I know from the guitar-p.a.-studio world that it is a never ending-subjective-expensive pursuit.
The guitar world has gone to very expensive NEW tube equip. designs, tube compressors etc. etc. etc.
So, be prepared to sell the house, if your going down that road...(it is Fun though).
ANYWAY, what happened to your simple design to narrow bandwidth, I thought that was interesting.
73's to all,
AG5UM
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« Reply #40 on: February 15, 2020, 12:13:32 PM »

Scope creep. Happens all the time without proper project management and firm requirements/goals...

"Does it become more stable thruout the audio spectrum and as our voices change?"

Yes. Using an APF thoughout adolescence will help stabilize your modulation during those formative years and avoid shame and embarassment.

ROFLMAO!
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« Reply #41 on: February 15, 2020, 01:09:41 PM »

The deeper I dive into this subject, the more I realize how heavily divided amateur AMer opinions are regarding using an all-pass filter (symmetrical audio) or letting the natural voice positive peaks go where they may. (asymmetrical audio)  I have received several emails from some very infomed AMers that lean both ways.

I have no skin in this debate, so I will be trying several approaches at this point:

1) The natural approach:  No LP or all-pass filters at all.  I have never heard this done before, but I have a DSP Berhinger 9024  six band compressor-limiter that will be put to use in a unique way. I was playing around with it the other day to VERY heavily limit and compress the 5KHz to 12 KHz band, thus producing a brick wall at those freqs. The other lower audio bands (< 5KHz) would have little to no 9024 band processing. It was not quite DSP brick walling, but close... I saw some very desirable look-ahead DSP effects to limit my extreme highs while at the same time preserving the positive peaks that my own voice generated.  I should know the results of these tests soon.

2) Natural with mild LP filter rolloff:  Same heavy Behringer 9024  5KHz highs limiting/compression but in conjunction with my L/C passive filter using only 5th order filtering, a smaller filter - slower roll off, less phase shift.

3) All-pass:  Try my all-pass Chinese filter board again, right after the 528E mic preamp to randomize the rotation, looking for symmetrical audio.

4 All-pass and LP filters:  Use the all-pass filter in conjunction with my 10th order passive L/C filter (parts to arrive next week) and see what symmetrical audio AND a sharp roll-off does.


Again, personally, I have been looking for a natural sound with high dynamic range where people in person might say they recognized me by my voice. (with a sharp 5 KHz roll off)  I don't want to be in a morning drive used car commercial with tiring audio..

The last time I used an all-pass filter, (10 years ago)  I was disappointed. I read the two articles posted above, which is what got me interested. But for whatever reason, with symmetrical audio I felt like I was fighting soft modulator tubes or I had the linear amp loaded too lightly or I was always out of phase.  It may be due to old habits - because I am used to seeing and hearing high positive peaks over the years. So I switched back to the more natural sound since.

But I have an open mind and will be trying the four techniques listed above and will report back my findings.  I may find the more agressive set of parameters are better for nightime QRM condix and the other smoother sound better for daytime quiet condix.  It may mean there is a hard rock AM radio audio and a lighter ham rag chew audio that is the issue.

In these kinds of discussions we all learn very quickly and cover a lot of technical ground, which is a good thang.

T
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Use an "AM Courtesy Filter" to limit transmit audio bandwidth  +-4.5 KHz, +-6.0 KHz or +-8.0 KHz when needed. 

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« Reply #42 on: February 15, 2020, 03:53:41 PM »

The deeper I dive into this subject, the more I realize how heavily divided amateur AMer opinions are regarding using an all-pass filter (symmetrical audio) or letting the natural voice positive peaks go where they may. (asymmetrical audio)  I have received several emails from some very infomed AMers that lean both ways.

I'll bet (Hi Dave!). Try this topic on-air and all hell will break loose. Have your pitchforks and torches ready. As an example of eliciting angst...

I was playing around with it the other day to VERY heavily limit and compress the 5KHz to 12 KHz band, thus producing a brick wall at those freqs.

Multiband processors were the solution to music and mixed content audio where bass notes from drums, etc, were causing mid and higher frequencies to be "ducked"  (Optimod 8000). Separating the lows from everything else eliminated this problem (Optimod 8100). In contrast, studio voice processors are all wideband.

Compressing the snot out of the sibilance frequencies will guarantee angst on the bands because it ensures that these frequencies are at 100% modulation at all times whether you have vocal energy there or not. Hiss, background noise, harmonics, anything that exists in that spectrum will be maximized regardless of the rest of the vocal frequencies. Band gain coupling goes a long way to reducing this effect as does careful use of multiband processing for voice processing.
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« Reply #43 on: February 15, 2020, 04:15:03 PM »


FWIW, the folks who have built up switched capacitor filters and that I have heard on the air
were all quite audibly noticeable in a negative way... ymmv.

K1JJ, passive audio filters require stable input and output impedances in order to get the
desired response. So, buffers and load resistors may be mandatory.

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« Reply #44 on: February 15, 2020, 04:54:21 PM »

Compressing the snot out of the sibilance frequencies will guarantee angst on the bands because it ensures that these frequencies are at 100% modulation at all times whether you have vocal energy there or not. Hiss, background noise, harmonics, anything that exists in that spectrum will be maximized regardless of the rest of the vocal frequencies.


Yes, I was just told the same thing about compression.  Thanks.   So heavy limiting of the extreme highs band instead will probably work well?  The 9024 gives me this choice.


Bear:  I have 600 ohm shunt resistors across the filter input and output, so shud be close.

T
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Use an "AM Courtesy Filter" to limit transmit audio bandwidth  +-4.5 KHz, +-6.0 KHz or +-8.0 KHz when needed. 

We die three times; when our body expires, when we're buried and when our name is uttered for the last time.  So, all my dogs are named Yaz.
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« Reply #45 on: February 15, 2020, 05:08:21 PM »

Compressing the snot out of the sibilance frequencies will guarantee angst on the bands because it ensures that these frequencies are at 100% modulation at all times whether you have vocal energy there or not. Hiss, background noise, harmonics, anything that exists in that spectrum will be maximized regardless of the rest of the vocal frequencies. Band gain coupling goes a long way to reducing this effect as does careful use of multiband processing for voice processing.
I use two compressor/limiters.

The first compressor/limiter offers gently compression, and hard limiting only of excessively high levels (cough, sneeze, bumped mic). It is connected post EQ so as not to respond to frequencies outside of the bandwidth I wish to transmit. The EQ offers a gentle rise from 1kHz to 5kHz and rolls-off above that.

The second, and final compressor/limiter, has an EQ connected as an insert. That EQ is set to cause the compressor to heavily compress/limit a only peak around 6.3kHz. Connected in this way, this compressor operates without make-up gain and, in effect, is a de-esser. A form of multi-band compression. It causes gain reduction only... no angst.
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« Reply #46 on: February 15, 2020, 05:27:53 PM »

It comes down to how much gain reduction (in each band) occurs with normal program content.

Limiters are fast acting compressors. In an oversimplified example, compressors deal with average levels while limiters handle peaks. The UREi BL-40 Modulimiter is a great example of each function put together.

Unless there's a platform function that separates compression (gain reduction) and expansion (gain increase), all compressors and limiters reduce gain when an input level exceeds the set threshold.

What goes down must come up...

Compression and limiting will reduce gain when needed, but when a high input signal is lowered, the circuits will "follow" this change with a corresponding increase in gain back to unity. The "heavier" the action of compression or limiting, the more dB this gain change will be. That's why heavy limiting (or compression) at all times will result in dramatic increases in whatever range of input frequencies that circuit is fed *unless* there is no gain reduction with normal content.

This is how the arrangement sounds with KK4YY's de-esser, that is, no gain reduction of sibilance frequencies unless needed.
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« Reply #47 on: February 16, 2020, 02:19:08 AM »

Now I know another thing that doesn't work...   I got the Berhinger 9024 multiband DSP audio box to roll off the extreme highs, but it was not sharp enough. As hoped it did not effect the positive peaks, but I still need something sharper.

So nix the 9024 idea.  I've also tried two high cut modules in the EQ and the 528E with poor results. I also tried the de-esser in the 528E and not sharp enuff to kill the 6-7 KHz extreme highs.  The sss's are killing me even though the rigs are clean.  

I'll try the passive filter next week when the parts come.  I will focus on seeing how many poles of the 5.5 KHz filter I can use before it affects the positive peaks phase excessively.

There is a solution that I will eventually find to effectively control bandwidth one way or another.



** "The second, and final compressor/limiter, has an EQ connected as an insert. That EQ is set to cause the compressor to heavily compress/limit a only peak around 6.3kHz. Connected in this way, this compressor operates without make-up gain and, in effect, is a de-esser. A form of multi-band compression. It causes gain reduction only... no angst."  

Don / KK4YY, could you elaborate more on the practical setup and adjustments of this idea?   I might like to try it...

T
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Use an "AM Courtesy Filter" to limit transmit audio bandwidth  +-4.5 KHz, +-6.0 KHz or +-8.0 KHz when needed. 

We die three times; when our body expires, when we're buried and when our name is uttered for the last time.  So, all my dogs are named Yaz.
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« Reply #48 on: February 16, 2020, 08:15:46 AM »

Don / KK4YY, could you elaborate more on the practical setup and adjustments of this idea?   I might like to try it...

T
The compressor/limiter I'm using is the Symetrix 425 with a Symetrix 532E graphic EQ as an insert (sidechaining). This is how the Symetrix 425 operating manual explains it:



8.7  Sibilance Control

Patching an equalizer into the sidechain can cause the 425 to respond more or less to selected frequencies, giving it the ability to make sibilance problems less apparent. Fine tuning between the compressor threshold, ratio, and the EQ boost applied in the sidechain will have to be made to arrive at premium results.

To find basic settings, start with a fairly high ratio (5:1 or so), and a compressor threshold setting between -20 and -5. Cut the low frequencies on the equalizer and give a 15 dB broadband boost to the EQ at around 5 or 6kHz. Now, carefully "tweak" the threshold setting as you count "four, five, six." What you're looking for is no compression on "four,five," and somewhere around 9 dB of gain reduction on the word "six."

You can refine the setting by listening to the equalizer output and adjusting the EQ to emphasize the sibilance in the source. Remember that you're equalizing the signal to emphasize the sibilance, not to sound groovy. Let the 425 do that.

Do you have a recording where the cymbals drive you nuts?...try the same technique on the overall mix.

Set the peak limiter threshold for 6 dB of gain reduction when Sam Screamer is on the system.

All normal signals will be slightly compressed, and really loud signals will activate the peak limiter. With these settings a shy person will be audible, and the guy who thinks he has to shout won't be too loud, or cause distortion.




BTW, the Symetrix 425 is not my favorite compressor. I used it because I had one. But I wouldn't recommend it to a friend.
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« Reply #49 on: February 17, 2020, 11:06:46 AM »

Tom,

Be glad to whip out a PCB for you. You going to relay switch the 3 filters?

John
Sounds like you are proficient in making pcbs? Have you ever considered making a pcb for Steves audio peak limiter? I tried, but there were a few things about the pcb software that confused the heck out of me. I bet several people would be interested. Ive already collected all the parts.
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