Discussions on Low Level EME III
These are Contributions to a Discussion on Low Level EME from the Moon-Net.
From: Mike Cook, AF9Y
Subject:Low Power EME Data Collection ExperimentsLow Power EME Data Collection Experiments. de Mike, AF9Y Introduction The recent MoonNet discussions on low power EME has renewed my interest in development of an optimum waveform for the EME channel. Before the community settles on a particular waveform, we should first determine the channel characteristics. Important characteristics such as coherence bandwidth, selective fading and multipath are best determined by direct measurement with averaging over varying conditions. W3IWI, KA9Q and others have made astute theoretical statements concerning the impact of these perturbations on various modulation techniques. I believe there is an opportunity for the larger EME stations to collect data on the channel characteristics using their radio equipment and a PC equipped with sound card. Data Collection Approach - Phase 1, Initial Experiments For the first phase, we need recordings with the strongest signal possible via the EME path. The idea is to transmit various waveforms tailored to allow extraction of key channel characteristics. A one minute transmission is sent by one EME station and the resulting received audio is recorded at the second EME station. Post analysis of the recorded data should provide some interesting insight. VE7BQH and I ran the first attempt at collecting data on April 28, 1996 using a PC program called TX55a. This program uses the sound card to generate audio waveforms that modulate the transmitter in SSB mode. I am releasing TX55a and its derivatives as a free public program. The program and the data collected will be available from my webpage at: http://www.webcom.com/af9y. In it's current form, TX55a provides the following: - 1 Khz Tone (Transmitted every odd 4 second period) This allows centering of the signal in the receiver passband and will be used as a reference signal during the test period. - Narrow Band Spread Spectrum BPSK (Transmitted alternating even 4 second period) This modulation is coded with a 63 bit Maximum Length Sequence (MLS) to allow extraction of channel delays beyond simple phase shifting. It is useful for multipath and channel coherence analssysis. I consider this waveform a candidate for weak signal modulation. TX55a transmits the "R" and "O" characters during the test. The main lobe bandwidth is 250 Hz. - Multi Tone FSK (Transmitted alternating even 4 second period) During this period, three simultaneous tones 1000 Hz, 750 Hz and 1250 Hz are transmitted. This waveform should provide insight into the selective fading aspects of the channel. Multi Tone FSK is a candidate for weak signal modulation and is being considered by W3IWI and KA9Q. The implementation of this modulation mode will likely require an update after W3IWI and KA9Q have had a chance to review it. I may not have captured their intent. Data Collection Procedure Lionel and I used the following procedure for the test: 1) Establish a 2mtr frequency just above 144.100 since this is SSB mode. 2) Using the program generated 1 Khz tone at the transmitter, tune the receiver for a 1 Khz receive signal 3) If available, rotate polarity for maximum signal. During our test, Lionel rotated his array through approximately 60 degrees during a 1 minute TX period of the 1 Khz tone. Using the FFTDSP42 program, the signal strength bar showed the best polarity position. This was an interesting side test that dramatically showed the advantage of polarity rotation. (I have include a picture of the results on my webpage) 4) Agree to a start time and record 1 to 2 minutes of the test transmission from the TX55a program. You can use FFTDSP42 to record the data as a WAV file. Just make sure the "R" button is red (active) and hit "SAVE" at the at the end of the test period before the signal image scrolls off the screen. If you are using another sound program for recording, I recommend you use a sampling rate of 8192 samples per second. The stronger the signal, the better the analysis. I suggest that we strive for S/N ratios of at least 15 to 20 dB (100 Hz bandwidth). Each band will have different characteristics, so we need recordings for 144, 432 and above. If you email your recordings to me, I will post them on my webpage. de Mike, AF9Y WEB page: AF9Y's WEB Page Work: email@example.com Home: firstname.lastname@example.org
From: W3IWI, Tom Clark
Subject: Re: Low Power EME Data Collection Experiments
Thanks to Mike, AF9Y for posting his test plan. The data will be very useful. I would like to ask the 70cm and up EME folks to try to gather some of the same type of data on other bands. From my experience at 70/23/13 cm as compared with anecdotal reports (especially from W2RS) at 2M, I come to the conclusion that the character of the signals can be very different as a function of frequency. On the higher frequencies, the signals seem to be (most of the time) spread with something like a Gaussian scattering function. At 2M it seems that some of the time discrete "glints" cause nearly specular reflections (much like the glints seen on a rotating mirror-ball in a discoteque). The development of an optimum (or at least somewhat optimized) signal processing strategy depends on our having a physical understanding of the reflection/scattering processes. Of particular interest in analyzing the "glint" reflections at 2M will be a detailed examination of small Doppler shifts associated with the received echoes. I therefore ask all of you who participate in Mike's experiments to carefully measure your transmit and received frequencies and pay careful note to the times -- the desire will be to try to account for all things that might contribute to the echoes to the 100 Hz (or if possible 10 Hz) level. I'm anxiously awaiting the factual data! 73 de Tom, W3IWI
From: Kaj Wiik, OH6EH.
On Mon, 29 Apr 1996, Tom Clark -- W3IWI wrote: Of particular interest in analyzing the "glint" reflections at 2M will be a detailed examination of small Doppler shifts associated with the received echoes. I therefore ask all of you who participate in Mike's experiments to carefully measure your transmit and received frequencies and pay careful note to the times -- the desire will be to try to account for all things that might contribute to the echoes to the 100 Hz (or if possible 10 Hz) level. I'm anxiously awaiting the factual data! 73 de Tom, W3IWI If professional data is acceptable, you can get it from "Radar Astronomy" by John V. Evans and Tor Hagfors, McGraw-Hill, 1968. The delay vs. relative power plots at different frequencies are especially revealing, they explain nicely the differences in the spread between low frequencies (e.g. 70 cm) and 10 GHz. Cheers, Kaj email@example.com
Ian, You're right, of course, about the difficulty which many people have (myself included, for the time being) in erecting large arrays in their limited gardens. My property is about 14,000 sq ft but it's heavily wooded and certainly wouldn't accommodate a large EME array. I've defined "QRP" in terms of EIRP rather than transmitter power. W5UN has approximately 2 megawatts EIRP, while I have about 5 KW (37 dBW). The typical AO-13-class OSCAR station, with 100 W or so power output to a 15- or 20-foot cross-Yagi, has about 33 dBW EIRP. 73, Ray
From: Mike, AF9Y My station showed a non linear drift rate of aprox 0.02 Hz/sec during the test. I doubt that many ham stations will be able more stable. At the 2 Hz BW for FFTDSP, the signal would remain in the bin for aprox 100 seconds. 100 seconds is the period of integration for the 200x setting. Conclusion: -174 dBm will be the minimum detectable signal for a ham station using a good preamp (0.5 dB NF) when running the FFTDSP program. (For those of you familiar with the UNKN422 weak signal challenge, as a comparison, the UNKN422 signal was aprox -156 dBm) Here's my calculation for the MR Beacon level: (Someone please correct me if I am wrong.) MR 437.1 Beacon Tx level = 1.3 watt = +31 dBm Distance, Earth to Beacon = 1.0E7 Km Path loss = 92.4 + 20 log (km) + 20 log(fghz) = 92.4 + 20 log(1.0 e7) + 20 log (.437) = 92.4 + 140 - 7.2 = 225 dB Signal level at Antenna =-194 dBm Assumed Antenna Gain = 20 dBi (include coax loss) Signal level at Preamp =-174 dBm I believe this test confirms the possibility of detecting the probe with a 20 dBi antenna (including coax) and a 0.5 dB NF preamp. Assuming 0.5 dB coax/combiner loss, four 436CP30 M2, 10ft antennas would provide 21.5 dBic. I think this is about the minimum antenna system required for a reasonable chance of detection. Also, I would recommend mounting these antennas with non conductive masts and combining the coaxes behind the reflectors. You can easily loose 1.5 dB if conductive masts are used. Doppler Chirp correction would be required if the rate is greater than .02 Hz/sec. If someone can confirm this need, I'll add it to the program. de Mike, AF9Y http://www.webcom.com/af9y Work: firstname.lastname@example.org Home: email@example.com
From: E-mail: James, G3RUH
From: Ian, G3SEK
Subject: Re: RX Experiments at -174 dBm
Mike Cook AF9Y wrote: >As James points out, the antenna array temperature is also >important. Here's what you can expect with different >types of arrays: > > Commercial Dish with Cassegrain = 18 deg K > > Typical Amateur dish using dipole/horn feed = 65 deg K > > Opitmized 16 Yagi Stack = 85 deg K > > Wide spaced 16 Yagi Stack (poor stacking) >170 deg K ======================================================================= At 430-something MHz, the yagi figures are still too pessimistic. An optimized single yagi, such as the DJ9BV series, has the entire 'rear panorama' suppressed by well over 20dB, and mostly more than 30dB. Taking the 20dB figure, that means that when aiming at an object high in the sky, noise pickup from ground at 290K will add only 2.9K to the system noise temperature. Unless the yagis are very badly stacked, the noise pickup from the rear of the whole array should not be significantly different from one yagi. At one time there were quite a lot of poorly stacked arrays of 16 noisy yagis on 432, but thanks to natural selection there are very few of them left today (yes, I could give callsigns :-) DJ9BV has modeled several 4-yagi arrays at 432MHz. Including sky noise of 15K plus noise generated by internal losses, rear pickup and sidelobe pickup at low elevations, the best antennas came out in the 25-30K region. Even the very worst didn't go above 50K. You should of course add contributions from phasing/feedline losses and the RX noise temperature, but with a good system the total from all sources needn't be higher than the 85K quoted above. With the 2Hz effective bandwidth, that looks like a noise floor of -176dBm. ======================================================================= > >Does anyone know what temperature to expect for a 2 or 4 >bay, 10 ft helical array? ======================================================================= Not I, but it's probably going to depend on the size of the backplane and the stacking distance. Unfortunately I don't know of any equivalent to the DL6WU rule for yagis, that would apply to helicals. If the polarization of the Mars Relay beacon is circular, I'd agree with James that a dish would be the most suitable, because a CP feed is so easy to arrange. I'm extremely suspicious about the real-life CP gain of multiple crossed yagi systems because the feed arrangements are so messy. An effective dish for 437MHz would have to be quite large because spillover, feed blockage and other inefficiencies start to become very important at smaller diameters. As a guide, a 12ft TVRO dish is very effective for EME at 1296 and above, but useless at 432, so I'd suggest 15ft as an absolute minimum for receiving the MR. Otherwise you'd do better with yagis or helicals - if you can make them work well. 73 from Ian G3SEK Editor, 'The VHF/UHF DX Book' 'In Practice' columnist for RadCom (RSGB) Professionally: IFW Technical Services Clear technical English - world-wide.
From: Tom Clark W3IWI
Subject: forward From: Ray Soifer <71331.1337@CompuServe.COM> To: "Tom Clark, W3IWI" Cc: "Bill Tynan, W3XO" , "Phil Karn, KA9Q" Subject: Article by SM5BSZ
Tom and Phil, Since I don't think that either of you gets W2CRS' "VHF EME Report," I've typed in the following article from the June 1996 issue, which arrived today (I have no scanner and I'm not within range of a fax machine). Any errors or typos should therefore be blamed on me, not on the author: Optimum Use of EME Channel by Leif, SM5BSZ The use of morse code exactly the way we are used to is in my mind very close to optimum use of the EME channel at 144 MHz. The suggestion by KA9Q and W3IWI, presented in the April issue of the bulletin implicates very poor use of the final amplifier. With single yagis and both ends, something like 500W is needed for each tone, so the idea to transmit four tones simultaneously will require a linear power amplifier capable of delivering peak power outputs in the order 8 kW. The four tones will all be in phase at regular intervals so the output voltage needs to be four times what is needed for 500W which leads to 16 times the power. The best use of the transmitter is to transmit an unmodulated carrier. This means that if we want to use a multi-tone scheme, the coding with N tones only gives us N states if we do not count the state with no tone on. AF9Y points out, in the same issue, that the characteristics of the EME channel will determine the optimum waveform to use, and with the above in mind, this means that the channel characteristics will determine the optimum duration for a single data bit. On 144MHz the libration fading will typically modulate a carrier with a few Hz, so therefore the receive channel needs a minimum bandwidth of something like 5Hz. There are two possible optimum modes of communication, one is to make the bits so short that a significant amount of information can be transferred within one QSB peak. The bandwidth then has to be in the order of 20Hz. This is in fact the mode in use today, and switching from morse code to some other code would not give any significant improvement. Lowering the speed compared to what is common practice today will reduce performance. Getting characters one by one will not give enough information - the information is not only the characters themselves, but also the sequence they come in. With a very slow speed CW one can use 5Hz bandwidth, which gives a 6dB improvement in S/N, but the fading is much deeper than that, so although one can much easier say that the signal is there, it is not possible to get any useful information out of it. Lowering the speed to much below normal CW, using 5Hz bandwidth and post detector integration is a possibility to detect signals not using the QSB peaks, but instead the average signal level. In my experience the average signal level is about 10dB below the peaks (peak value within a one minute period). The bandwidth reduction gives 6dB so we need another 4dB that have to come from post detector filtering. To get that, a post detector time constant of about 1.5 seconds is needed. In a real circuit probably 5 or 10 seconds is needed since in this mode information gets lost if one bit is lost. so the integration time has to be long enough for the probability of the average signal dropping below the threshold to be very small. With the above in mind I would suggest two possible modes of improved EME communication. One is to transmit a continuous carrier, jumping between say 128 different frequencies, staying at each one for say 5 seconds. The first and last 5 seconds could be at frequency 0 and 127 respectively and each transmission would have to be 2 minutes or more. Each frequency would be one character, and decoding would be by means of something like FFTDSP by AF9Y, with some simple routines added. The transmitter could be a SSB rig fed with a single tone generated from a soundblaster. With a channel separation of 15Hz the total signal would fit in the standard bandwidth of our rigs. I think this way of making EME would mean the possibility to work EME using about 3dB lower erp as compared with present methods. This is far below the 10-20 dB improvement suggested by W3IWI so I guess he did not take QSB into account. This communication mode, using the average signal level, will be very reliable once the signal threshold is overcome. (The estimate of 3dB improvement is based on the information that this is the level at which signals are detected with FFTDSP compared to when a random QSO is possible - with a lot of time and effort) The other mode I propose I think would be much more elegant, and it is probably slightly better. The real beauty of it is the compatibility with todays CW, and the concept is simply to define "the standard EME time window." This means we decide for example that the whole message has to be exactly 15 seconds. The computer will calculate the appropriate speed and number of repetitions of various parts within the message. Exactly the same sequence is then transmitted during a full 5 minute period, synchronized to the 15 seconds window. With a computer it is easy to calculate "sliding FFT," with say 256 points. This could give a bandwidth of 20 Hz corresponding to two points in the FFT. Every second we would get 40 such transforms, so in the whole 15 second window there would be 600 transforms. To store the power at each frequency and time would require a matrix of 256*400 points which is no problem with a PC today. By just adding point by point the 20 matrices belonging to the 20 windows within the whole 5 minute transmission period we would have made a post detector integration that improves the S/N by 6.5dB, but this is the average S/N, so using the ears would still probably be better. If we instead store in the matrices 0 if the correspondiing power is below 2* the average noise power and 1 if the the current value is above, the sum of 20 matrices would have a column of values significantly larger than other columns if a weak signal is present. I am quite sure that correct information can be extracted at levels well below those needed today. Both methods require good frequency stability. Of course I do not mean that above suggested times are really optimum - I just want to point out the two different types of modes that are the two possibilities that well use the EME channel according to my understanding of the problem. 73s Leif / SM5BSZ (W2RS note: Leif is very active on 144 MHz EME but to my knowledge he does not currently have an Internet e-mail address. He can be reached through the 2-meter EME Net, weekends on 14.345 SSB. Leif is well known for his development of an innovative polarization diversity system, using 45-degree increments and a voting device at the receiver, which achieves close to optimal results in combating Faraday rotation without the need for mechanical Z-axis rotation. His calculations in the above article appear to assume random operation rather than prearranged schedules. My own work has shown that, at 144 MHz, it is possible to complete a prearranged schedule with a S/N ratio 6-8 dB lower than that required for a random QSO; this results, I believe, from "cerebral signal processing" as the receiving operator's brain makes use of the information it has when trying to copy the incoming signal. 73, Ray)
From: Brian Beezley, K6STI Subject: UNKN422.WAV Mike, I wanted to tell you about my ongoing efforts to crack your UNKN422.WAV EME callsign mystery. I'm motivated primarily by the technical challenge; the $100 prize you offer will barely pay for the electricity I've expended! I think this is a really neat challenge you've come up with, so I'm copying this note to the EME reflector and to the TAPR HFSIG reflector (where there's recently been some discussion of CW decoding) in hopes of spurring others to try tackling the problem (perhaps one more time). Earlier this week I seriously looked at the problem for the first time. I spent the better part of two days on it. Listening to the file without benefit of signal processing, I could just barely hear a tone pop out of the noise for a second or two. (The first time I listened, I heard only noise.) This was so unexpected that I downloaded the file again (http://www.webcom.com/af9y) to make sure I had the right one! I cobbled together a special version of DSP Blaster to process the .WAV file instead of real-time, sound-card audio. I found neither its LMS noise reduction nor 15-wpm matched filter helpful. For quite some time I've wondered whether a matched filter really is optimal for processing CW signals for detection by ear. A matched filter is optimal for threshold detection of signals in noise under certain conditions, but I wonder whether the conditions hold when preprocessing a signal for detection by ear. I don't know what the ear/brain does to detect CW, but it's certainly more complex than simple thresholding. For example, for many years I've preferred to use an SSB-bandwidth filter to recover weak HF CW signals rather than a 500-Hz CW filter. The additional noise somehow makes signal detection seem easier. (I haven't run a test to verify that it actually makes detection more reliable). I find listening through extremely narrow filters somewhat disconcerting because they shape the background-noise spectrum approximately to that of a sine wave at the filter center frequency. So your ear must then distinguish between the desired sine wave and weaker ones that bobble around in the background. Even though I estimate that DSP Blaster's 15-wpm matched filter improves CW detection by perhaps 1 dB for my ears, I sometimes prefer to detect CW in wideband noise (even though listening for long periods that way can be fatiguing). (It's interesting that most ops seem to vastly prefer narrowband CW filters.) Since UNKN422 remained a mystery, I tried implementing a signal-processing technique I had planned for DSP Blaster version 2. I've had my Pentium-100 for one year and this was the first time I ever ran up against its processing-power limits. I had to recode the algorithm using Pentium-specific, floating-point optimizations (pipelining). It immediately yielded clearly audible dots and dashes throughout the .WAV file. Now I was getting somewhere! Right away I discovered I had misestimated the sending speed. I now was sure that recovering the callsign would be easy. I visualized the "K6STI Cracks UNKN422 with Secret Algorithm" headlines. So I listened carefully for some time, wrote down some possible calls, and consulted your list of 1000 EME callsigns, among which is the mystery callsign. Hmmmm...nothing matched very well. After more listening I found that I could mentally transform any given sound segment into my current best-guess callsign with very little wishful thinking! I was familiar with this phenomenon but thought I'd outgrown it years ago. I've heard 160-meter operators talk about it and I imagine it must be old-hat to you EME ops. Twice I was quite sure of the callsign. I was ready to submit my winning entry and thought I'd take one last listen. Then I discovered that I heard completely different letters by altering playback speed and pitch! This was quite a shock to someone who prides himself on his hearing acuity. Although the dots and dashes are readily audible, the signal level is quite variable. (I assume this is due to some kind of moon multipath.) Often the code elements seem to have holes in them or to be smeared together. Sometimes it's not clear whether it's one long dash or a dot and dash. I also discovered that my callsign choices tended to depend critically on one or two crucial letters. It was very easy to persuade myself to ignore other sounds that really didn't fit but clearly weren't important because, after all, surely this was the correct callsign! The fascinating thing about this problem is that it involves both a technical and psychological challenge. In frustration, I tried cascading a semi-matched filter with the processor-intensive technique. It didn't help. I really didn't want to guess at a callsign, so I kept listening over and over, tweaking algorithm constants. More new callsigns. I got so involved with the problem that my sleeping schedule became disrupted. When I reluctantly stopped listening to take a much-needed nap, I fitfully dreamed about weak CW! I decided it was time to return to normal life and finally gave up on the problem late one night. Temporarily. I have one more technical trick to pull. In fact, although it's not implemented in the current version, this trick was the whole reason I started the DSP Blaster project. I plan to implement this technique in version 2 along with the processor-intensive algorithm. I think the combination may provide enough S/N enhancement to yield a solution. It will take some experimentation to get right and a fair amount of programming to implement, so I expect it will be a little while in coming. Meanwhile, maybe someone else will decide to try a concerted attack on the problem. I think your UNKN422.WAV challenge would be a really neat term project for university engineering students. It's a fascinating mystery, involves signal-processing theory, practical engineering, programming techniques, and human perception and psychology. It's really a lot of fun! You might try to locate an engineering professor somewhere who is also a ham and lay it on him. 73--Brian, K6STI firstname.lastname@example.org
From: David, KT1X Subject: Code vs. No code and VHF..inappropriate styles and debates While reading the "debate" on the code and no code stuff, a couple of thoughts came to my mind. FIRST...Why is this being argued as a code vs. no-code issue on a VHF Reflector? The VHF rules were changed years ago...the battleground is really for easy access to HF previlages. The discussion over CW on VHF should clearly be about effectivness (S/N, bandwidth, realities of propagation, cost) alone. Isn't that clear!?? Issues such as its usefulness to hams should dominate our discussion. On HF, the arguments seem to be the following in favor of code: It is historical..with out this OPERATING SKILL (NOTE I DID NOT SAY TECHNICAL SKILL), one is not skilled in a mode used by daily by many HFers for a variety of reasons.....personal, technical-economic etc. and yes, even one that hasn't been mentioned,.....a bit of privacy.... since only good CW ops are likely to eavesdrop on your narrow band converstation. CW is used by many modern HF hams for a variety of reasons, including to escape the brashness and BS of phone modes, with their "elbows out" get on the "soap box" style of operating....not to mention the QRM of all types (this is the same reason I seek out unoccupied bands and operate robust digital HF modes). As, far as the effectiveness of CW goes, it is a COST EFFECTIVE weak signal/low power technology, HF or VHF, unlike many of the more advanced/but very interesting weak signal technologies which generally cost an arm and a leg, if they are available at all to even the technically adept ham. Remember, much modern communications technology of interest to business is almost antithetical to the ham spirit. Business folks want it to be dependable, exploitable, dull, owned, centrally controlled, and other than gadgets to "hook" or sell to the public, they want the technology to be invisible to the average user. Argueing whether or not CW is useful to commericals (I, like many hams was one), or maintaining that a certain mode is/is not "outdated" based on what the current use/training of communications techs demands is IMHO bogus. Ham radio is not some sort of official training ground for telegraph ops or computer/communications techs. It has traditionally been a multifaceted hobby with a few basic enterance requirements to assure capable operating. It seems to me that hams should do things because they are fun, educational, interesting, work for our purposes, not to ape commerical communications, except if it fits a ham oriented agenda. How exciting/cost effective would it be to call up folks on the international telephone circuits just for a brief chat, or to try out your latest telephone gadget? It also seems to me that much of the "debate" on code/no-code is one with some hidden agendas, not relating to the issue as presented. So far nobody has mentioned them, although one might detect their presence in some of the typical hostilities present in the usual code vs. no-code flap. Many of these "arguements" are based on stereotypes and personality, and generational culture clashes, as well as the need of certain individuals to feel superior or to vent their frustrations. They are argued using crude half-truths or veiled insults. It is a common human tendency, exacerbated by the "cocky bullshit oriented American persona". "If I don't know it or use it, what good is it!" This faulty attitude seems to be applied to justify things whether the "it" is CW, digital, HF, packet etc. It is a fault shared by "old farts" and "brash newbies" alike. As a fairly active long-time ham whose had had a lot of experience and fun on a variety of modes and bands, I am quite insulted to be characterized and stereotyped by some soap box extremist from either of the two camps, and I suspect that many others are, no matter what their interests, background and experience, use of modes etc. In case you haven't noticed, these stereotypes have been picked up by the media, and hams are either portrayed as geriatric cases with straight keys, or socially challenged gadget geeks. I personally find many of the styles of operating in ham radio today to be obnoxious....on many bands and modes.....VHF/FM/HF/SSB etc. It's not the mode or the technology, its the style of the ops, the wise-assed know-it-alls, and those with narrow vision and a need to feel superior who bug the be-jesus out of me. Of course, that doesn't surprise me, come to think about it. In many ways I suspect ham radio is just reflecting the general society, and it will become increasingly like it. If we want ham radio to have a worthwhile future, this is where we must work at improving things. If not, it will end up like the CB debacle..a lowest common denominator sewer. David PS...I would be all for replacing the higher speed HF CW requirements with one that was one reasonable speed, and then linking further operating privilages to technical proficiency based on in-depth knowledge about things like propagation, RFI, RF safety, design of filters, advanced modulation techniques etc at a conceptual level other than memorizing terms and regs. I know several long-time hams who were turned off by the lack of technical challenge for even the old style Extra before the current round of "question pool" exam system came in. This challenge was what the original incentive licensing was supposed to do, except that over time they dumbed the tech part down and over did the code. At least if we reemphasized the tech part, we could argue that ham radio has educational value. Problem is, don't know how this could be implemented within the current "get rid of the FCC" mania...and I am sorry to say, it wouldn't sell as many bells and whistles radios to an eager crowd of "gadget consumers".
For comments, typo's and changes: Rein Smit, W6/PA0ZN