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CESSB Questions

Bob N7ZOBob N7ZO Member ✭✭

In David’s QEX CESSB article, he mentions several times of the value of baseband audio processing applied before his algorithm:

“It is possible to overdrive the algorithm by driving the baseband “RF clipper” into a few dB of clipping beyond what is necessary to remove the Hilbert transform overshoots. This will provide a further increase in average power. I think it is better to do baseband audio processing, however, and let the CESSB system only remove overshoots.  Historically, baseband audio processing was not considered particularly effective for increasing average SSB power — because of envelope overshoots. This is no longer true, with the introduction of CESSB. With baseband audio processing, sophisticated multiband compression and clipping is possible, with better results than a simple single-band RF clipper.”


“The processing used does not produce any significant audible artifacts. It may be used in conjunction with speech processing.


“The system only affects transients that would produce overshoot. Audio com­pression to increase density, compression, peak limiting, and so on should be applied before the audio input to Figure 11.”

This raises the following questions for the 6000 series implementation:

1.       What baseband audio processing is applied before the CESSB algorithm?

2.       What does the NOR-DX-DX+ control do?  Is it simply overdriving the algorithm as mentioned above?



  • Bob N7ZOBob N7ZO Member ✭✭
    edited February 2017
    I should have added, in question #1 above, "...other than the explicit downward expansion"
  • edited February 2017
    For those interested in a historical understanding of speech processing, these articles set the stage for W9GR's work.  The Sabin and Collins articles nicely explain differences between baseband audio and R.F. speech processing:

    Bruene, W., "Notes on Speech Clipping and Filtering," QST, Mar., 1952, pp. 54-56;
    Clegg, E.T. & Squires, W.K., "Speech Clipping for Single Sideband," QST, July, 1965, pp. 11-15;
    Collins, H., "Ordinary and Processed Speech in SSB Applications," QST, Jan., 1969, pp. 17-22;
    Kirkwood, B., "Principles of Speech Processing," Ham Radio, Feb., 1975, pp. 18-34;
    Moxon, L., "Performance of R.F. Speech Clippers," Ham Radio, Nov. 1972, p. 22;
    Sabin, W., "R.F. Clippers for SSB," QST, July, 1967, pp. 13-18;
    Tong, D.A., "Distortion Levels in R.F. Clipping," Wireless World, Oct., 1976, pp. 77-81.

    Paul, W9AC
  • Steve-N5ACSteve-N5AC Community Manager admin
    edited April 23
    The transmit signal chain if fairly complicated with a lot of different stages that account for differing bandwidths of input signal and what is desired for transmit, gain adjustments to correctly position the audio input into the right range for each algorithm, etc.

    The NOR-DX-DX+ control adjusts the level of processing applied to the audio.  The further to the right you slide the control, the more processing is used and potentially the more distortion.  To the folks we tested on, it was difficult to tell anything was different other than being a lot louder in the NOR/DX range.  DX+ added more volume and altered frequency response, also some distortion, although very tolerable.  If you're trying to bust a pileup and can't get through on DX, try DX+ -- you might not get a great report on audio quality, but hopefully the added power and intelligibility will help you make the contact.
  • Jim  KJ3PJim KJ3P Member ✭✭
    edited December 2016

    ...and all of this audio processing is applied before the audio hits the RF clipper?

    -- Single band or multi-band audio processing?

    -- AGC or compression?

    -- Flat audio response except for DX+ and user adjustments?

  • Bob N7ZOBob N7ZO Member ✭✭
    edited February 2017

    Hi Steve,

    Thank you for the comments about the voice processing.  The reason I was asking about the specific baseband processing applied before CESSB is because I am considering buying an outboard processor and would like to know if applying single or multiband compression and other processing common to the hardware voice channel strips would be of benefit or would be redundant to what is already inside of the 6000 series.

    Also, I am pondering what the Waveform API could be put to use for in the area of additional baseband audio processing.

    If this information is considered proprietary, I understand.  Otherwise, it would be helpful to get some more specific information.

    Thanks and 73, Bob, N7ZO

  • Bob N7ZOBob N7ZO Member ✭✭
    edited February 2017

    Can you give us any information at all about the audio processing beyond CESSB, other than the obvious downward expansion and EQ?

    Thanks, Bob, N7ZO
  • Steve-N5ACSteve-N5AC Community Manager admin
    edited February 2017
    Bob, well this all depends on who you are and how much of an audiophile you are.  I am not a professional audio aficionado.  We do have some folks that are that work for us -- Mark, NA6M, for example.  It would probably be better to get a response from Mark to compare and contrast what we are doing now vs. what an outboard processor does.  He brought in a Vorsis M-1 (http://wheatstone-processing.com/index.php/m-1-digital-mic-processor-overview) for us to play with and we currently have it on the shack station playing with it.  That box is an analog in/analog out box that has all DSP guts and is also controllable over Ethernet should you choose to do so.  It also has digital out, but in a format we don't consume.  The box seems to work well (what do I know), but does duplicate some of the functions we have -- it has a parametric EQ whereas ours is currently fixed-band.  It has a LPF/HPF on the MIC input, a downward expander (both of which we have) a De-esser (we do not have this) and a compressor (which we have).  My guess is that our compressor (CESSB) is better than what they have, but that their downward expander works better -- just a guess.

    We also do various audio processing required to make sure our other pieces work well.  As I mentioned, the audio chain is rather complex and it's one of those deals where you could see a high level block diagram, but then you'd ask a question about something and we'd have to open up the block, etc.  It is a "devil is in the details" kind of thing.

    I would probably use an outboard processor on the MIC before I would use the waveform API.  The waveform API is really intended to support new modulation schemes, but it certainly could be used -- you could take it and jack it into an audio plug-in tool on Windows and do all kinds of fancy audio stuff.  You just need to be sure to modulate the signal into I/Q before sending it out of the software -- we expect IQ data out, not real stereo.

    I'll ping Mark to come read this thread and add his two cents.
  • Bob N7ZOBob N7ZO Member ✭✭
    edited February 2017

    Hi Steve,

    I am an audiophile, but I am not a frustrated broadcaster looking for ESSB tones.  I am interested in getting the loudest, most copyable signal, which, I would think, is an interest FlexRadio shares.

    I tend towards audio that might be a bit sharp for rag-chewing, but punches through to the DX stations.  So, for instance, a de-esser might be good for broadcast radio, but it may actually work against intelligibility (as long as there is a clipper downstream).  And I doubt that there is a significant effective difference between a parametric EQ and the 8 fixed-band graphic EQ in the 6000 series for the ham radio application.

    Two areas that are of interest for a loud intelligible signal are compression and voice symmetry.

    Compression: CESSB naturally causes some compression and overdriving it causes more.  Can a single or multi-band baseband compressor add more average power to the signal? (Dave, W9GR, seems to recommend it.)  Would a multi-band compressor be more effective in front of CESSB? (A multi-band compressor has the advantage of also being an adaptive EQ.)

    Voice symmetry:  Many of the commercial voice channel strips use all-pass phase shifters to smear the voice peaks and reduce the natural high crest factor of the human voice significantly.

    I have here a Symetrix 528E voice processor (with a single band compressor and voice symmetry correction, among other things) and am trying to get my hands on a TC Electronic Triple C multiband compressor, but it seems a shame to use external hardware for this.  There are a number of good software based compressors for the PC that could be patched into the audio chain for experimental purposes.  But the best solutions would be internal to the radio.

    Anyways, these are interesting questions and I appreciate your comments on the 6000 series audio chain (and look forward to Mark’s input).

    73, Bob, N7ZO

  • Mark-NA6MMark-NA6M Inactive Employee
    edited June 23
    Howdy folks,

    I spent the last 30 years working in broadcast radio and have seen a lot of technology improvements during that time. One of the areas that just keeps getting better is the art of audio processing. In broadcast radio we process the microphone audio to sound the way we want as well as bring up the average output levels with a microphone audio processor. This mic processor is not the final processing on the sound of a broadcast station but rather a key part of it where talent voices are concerned. It is responsible for that larger than life sound you typically hear when the DJ opens the mic.

    As Steve mentioned, the Vorsis M1 is a DSP based unit that allows the user to save and recall profiles based on their needs. While it is designed for use in a broadcast facility or recording studio, it is certainly right at home in amateur radio as well. By using the parametric equalizer to tailor the audio, I am able to achieve the sound I want for the job at hand. One profile might be wide and open for ESSB while another could be more narrow and punchy for working DX. By letting the Vorsis process my microphone audio I am able to maintain tighter control over the sound of my signal. I use a Neumann TLM-103 Condenser microphone on my station.

    Greg K5GJ and I will be making some audio recordings of the Vorsis in action today. We will record the performance both with CESSB enabled and disabled.

    73 de na6m

  • Alan, K2WSAlan, K2WS Member
    edited November 2015
    I'm impressed with CESSB performance! It develops a high level of average power, without the distortion. I'm an audiophile and use outboard audio gear. Before CESSB I would drive my 6500 with a properly EQ'ed and peak limited audio. While the audio envelope was peak limited, the SSB RF envelope was not because squared off audio waveforms result in higher peak to average RF envelopes! Well, W9GR's CESSB processor at the normal setting resulted in every syllable modulating to full output, with very little variation at maximum level, a real RF peak limiter. The Tx IMD did not change on the Panadapter and the Tx audio was audiophile quality. With CESSB offline the average power dropped off about 6dB! Very impressive. CESSB behaves like an ideal RF clipper, with minimal distortion and canting. The DX and DX+ settings drive CESSB harder and that does increase talk power but with some distortion. On the other hand, if we drive the rig with a more compressed and EQ'ed voice signal, CESSB may provide additional average power at low distortion levels. 73, Alan K2WS
  • edited December 2014
    Wow Alan, I sure would like to get the results you're seeing there!  I am measuring miserable avg/peak performance here on my 6500.  With "PROC" enabled and set to "DX" I am measuring 1500W PEP on my LP-100A and only 100W AVG on my Bird 43.  That's an avg/peak ratio of less than 0.1 (yuck).  What in the world are you doing than I am not ?!?

    73 de W4WMT (Bryan)
  • Ken - NM9PKen - NM9P Member ✭✭
    edited December 2016

    Interesting stuff.  Any results on the recordings?


    Ken - NM9P

  • edited December 2014
    Hey Mark, what kind of latency are you seeing through that Vorsis unit???
    73 Bryan (W4WMT)

  • np2gnp2g Member
    edited November 2015
    WOW a great conversation.

    Now I am not the only one playing with duty cycle and "LOUD Audio"  but there is a difference between increasing the duty cycle  hence the output and just **** audio  coming out of the transmitter.

    Since Flex has marked these 3 levels normal,DX, DX+  one would presume that the efficiency can be seen while running DX  . And with that usage the only problem is the majority of the DX stations out there are not listening 4.0 wide or 3.5 wide.  Most are restricting their bandwidth to 2.8 or less . (The majority of stations out there capability.)

    SO the  ESSB venue  cannot  be  the issue with Bulk compression.

    If you wish higher duty cycle or "LOUD audio" this is not cleanly accomplished by that slider.

    Compression should be added in various spots within the audio spectrum. Not large doses but literally small amounts  and the sum total  of these increases is what makes your audio "Loud" un distorted, clear and clean. Improved duty cycle .

    I have only found multi channel electronic audio processing capable of this feat 

    To really get ahead of the curve there are a few more things to consider .

    Keeping  your transmitter not at the full output but far below this level. Where it has overhead . The IMD mix is far less there  . Is necessary 

    If so inclined must keep your amplifier totally linear and not pushed to the wall. . If you wish to run full legal that amplifier better be capable of far in excess of this 1.5 KW  level. Again Overhead. 

    IF 1.5 KW out is your goal   better get the big boy pants on amp  Not for the watts out but for that reserve capacity  

    And Yes you can achieve full bodied audio . Fantastic duty cycle   running 1.5KW and still be 3DB Better the rest. 

  • edited November 2018
    So how did the Vorsis M1 work out?
  • edited December 2014
    I have been looking forward to those results on the Vorsis M1 too!!!
  • Dale KB5VEDale KB5VE Member ✭✭
    edited November 2017
    In talking to Tim he mentioned he used a microphone preamp. He said the least you use Dsp gain the better. I use a Pr 40 which has worked great on many radios as long as the radio had the ability to eq. I took notice that I was having to use the 20 db boost with the pr 40 or a lot of mic gain, several people who know me could hear a slight distortion on peaks but otherwise great sounding audio. I played with processor on of dx 1 and dx 2. Nothing totally cleaned it up. I knew it was not rfi and everyone agreed, so I went back to what Tim said the more Dsp gain you use the more potential noise would be introduced. I explored mic preamps, not full fledged eq but mainly preamp. As I did this I went back to times I helped to operate the sound system at church. Remembering when we went from the older tube system to the new solid state how we had a had time getting the same warmth in the mic audio we had with the tube system. I had noticed several of the amplifiers had a tube in them. Talking to a local who owns a recording studio he asked what I wanted to accomplish. I told him what the flex would could do and gave him some reading material on it. I told him I did not want to have a "rack system" but to be able to use the Heil pr 40 for radio communication, and I did not want to bust the bank just make me sound a little better. He came back with a pream which had a tube and solid state amplifier, you move from solid state amplification to using the tube which gives you a warmth. Also the unit he chose has a compressor and limiter and a few other items in a very small footprint. Of course you have the ability to adjust gain before going into the balance input on the back of the flex. I spent 95.00 and it was the best 95.00 I have ever spent in ham radio. Ido my eq with the flex, do not use the 20 db boost, set mic gain at 10 and then set my tube preamp to desired amount of warmth and gain to what I need to get audio where it is recommended. I use dx1 all the time. Now I get great audio reports, no distortion and with prophiles I have a rag chew pr40 and a dx pr40 and a obnoxious pile buster dx2 pr40. I contribute the advise Time gave me concerning the dsp gain in getting the results I have obtained. And of course my friend and his knowledge in nudging me on some things to do without sitting down to a full fledged sound board.
  • edited December 2014
    Nice report Dale. Thanks!
  • N7AIGN7AIG Member
    edited June 23

    Hi Guys,

    As mentioned by Bob, N7ZO, a phase rotator is very effective at increasing the average power in speech. When you examine a vocalized phrase you will typically see a highly asymmetric waveform with a high peak to average ratio, or crest factor. Radio stations use the Symmetrix processors to alter the phase relationship between harmonics with the result that the crest factor is reduced substantially. And that permits downstream compressors and limiters to do a more effective job while maintaining high average output power.

    Such processing is really nothing more than a multi-pole (4 to 12 poles) all pass filter, typically tuned down around 300 Hz. It does not introduce any unwanted transient behavior, and the human ear cannot discern the difference between processed and unprocessed sound. We are insensitive to the relative phase between harmonics in sounds, whether musical or speech.

    A Symmetrix box is pretty expensive for what it does. But if you have any coding savvy then it is really easy to produce All-Pass Filters in software for use in a DAW (digital audio workstation) running between your mic and the transmitter. E.g., VST plugins for Reaper.

    73 de Dave, N7AIG

  • edited November 2016
    What you describe. Is an all-pass phase *scrambler* circuit. A phase rotation circuit looks ahead at input asymmetry, then "flips" the input in the direction that yields the most positive peak amplitude. An example of this is the Inovonics 231 MAP II processor.
  • Jim  KJ3PJim KJ3P Member ✭✭
    edited December 2016

    Yes, flipping the audio phase to favor the positive is a standard AM broadcast audio processing feature to take advantage of the FCC’s broadcast AM modulation rules. I have no idea how such asymmetric modulation applies to AM on the ham bands.

    But a phase scrambler (like Leonard Kahn's passive Symmetra-Peak) works to give a symmetric average to the audio waveform, and bring the negative and positive peaks to  the same level. If applied *before* other audio processing, it can yield higher average modulation equal to the difference in asymmetry. Another way to look at it:  if a limiter limits to the highest peak, either positive or negative, you can see that it’s better to have those peaks approximately the same.

    Not all voices need this correction, but I’ve measured some voices that are wildly asymmetric, and would benefit greatly from phase scrambling before any sort of limiting/compression.

    Any smart (mic) audio processing chain should have such a phase scramble, and I'd love to see one added to the 6xxx audio chain.

  • Ken - NM9PKen - NM9P Member ✭✭
    edited December 2016
    Yes, It would be another great "science experiment" for the 6000!  
  • N7AIGN7AIG Member
    edited December 2014

    Heh! Call it whatever you like, but Google each of "Phase Rotator" and "Phase Scrambler" and see what you get.

    BTW... not a science project. Been doing this for years. See W3AM's article at http://www.w3am.com/8poleapf.html  going all the way back to the 1950's.

    73 de Dave, N7AIG

  • Jim  KJ3PJim KJ3P Member ✭✭
    edited December 2016

    The Goog notwithstanding, a phase rotator or "phase flipper" has an output that is asymmetric on purpose (e.g., favoring positive peaks). A phase scrambler aims to have a symmetric output, with both polarity peaks equal.

    I have previously read W3AM's article, and it makes the point accurately. This should be a snap for a good DSP programmer to code (he says, not knowing of what he speaks!).

  • edited March 2017
    I, for one, would be grateful if someone at FRS would just let us know unambiguously when CESSB is on and when it's off.  Is it on all the time during xmit?  Is it on only when "PROC" is enabled.  What's the deal there?

    73 de Bryan W4WMT

  • edited November 2016
    Gary, W3AM calls the all-pass network a phase rotator, but knowing him, I suspect he would agree that the more appropriate term is a "phase scrambler."  Note that Bob Orban (inventor of the Optimod) calls an all-pass network a phase scrambler as well. (ref. p. 1-3, Optimod 8100 Operating Manual).

    Probably the important take-away is that there's a difference between the two circuits.  A multi-section all-pass network removes asymmetry.  The alternate circuit uses no all-pass network, and simply rotates the direction of audio peaks through phase *reversal.*  The two circuits behave differently with AM, but each is designed to maximize loudness.

    Due to the Hilbert transform, an SSB waveform is not related to the audio input waveform in a simple manner and unlike AM, peak control of the SSB audio signal does not guarantee a reduction of SSB's peak-to-average power ratio under all conditions (i.e., modulation density increase).  

    I don't recall W9GR bringing either phase technique into his QEX article.   It would probably be worthwhile to get his opinion on effectiveness with CESSB.

    Paul, W9AC

  • Tim - W4TMETim - W4TME Administrator, FlexRadio Employee admin
    edited December 2016
    Only when PROC is enabled.
  • N7AIGN7AIG Member
    edited December 2014

    I admit I had never heard of the "phase unifier" before. I have a hard time imagining how that works without producing distortion -- a al BPSK. A full-wave rectifier does this too, but I would only use it to enhance bass frequencies by producing harmonics that trick the mind into believing that the bass must be louder than it actually is.

    As for the comment about SSB waveform not being related to audio input amplitude, I also have trouble with that notion. To be otherwise would mean a nonlinear relationship between amplitude loudness levels and the modulated waveform.

    The Hilbert transform merely tricks a system into producing a one-sided spectrum by conjuring up a related imaginary component for a real waveform. But amplitude is preserved, and SSB is formed by the sum or difference of two modulated carriers with positive and negative frequencies. So again, no nonlinear impact on amplitude.

  • edited November 2016
    > "As for the comment about SSB waveform not being related to audio input amplitude, I also have trouble with that notion. To be otherwise would mean a nonlinear relationship between amplitude loudness levels and the modulated waveform."

    That's not how the Hilbert transform works.  The only instance where the input and output are preserved in phase and amplitude is with a single sine wave - but not complex waves that require a Fourier series like our voices.


    See the bottom of p. 2.

    Paul, W9AC

  • N7AIGN7AIG Member
    edited December 2014

    Thanks for sending along that link. I see what the problem is here... one of terminology. In fact, a Hilbert transform does *not* produce harmonic content. And it really does preserve amplitude.

    But with a complex waveform with harmonic content, the resulting phase shifts by 90 deg puts their sum into the very thing that the "phase rotator" was trying to undo.

    So perhaps it won't work with SSB? Not sure I buy that... you see the processing is applied to the audio voice waveform just ahead of compression and limiting. The aim here is to produce even tighter compression of the voice waveform than would be possible otherwise. And RMS compressor would not care either way, but many compressors are peak compressors, and these would benefit from the phase rotation to produce a more symmetrical waveform.

    Then we move on to modulation (of any kind - AM, SSB, digital), and you end up producing a carried waveform that can be received at the other end as generally louder. So my bet (any takers?) is that a waveform symmetrizer ahead of a compressor would benefit our SSB modulations for increased punch, on the side bet that the compressor is a peak compressor / limiter.

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