[Flexradio] audiodg.exe
Just found a nasty little "feature" in Win/Vista and Win/7, named "audiodg.exe". This is a "required" audio driver that attempts to isolate processes audio routing graphs for security purposes. Problem is, the cure is almost as bad as the disease they are trying to prevent. On my Dell laptop, the problem is so bad, that I see frequent long pauses where the GUI's become unresponsive, audio dropouts occur, and this presumably is the main culprit for the F1.5K audio dropouts. It is so bad on the Dell laptop, that even after disabling all audio "features" as indicated on the various complaint web pages for audiodg.exe, I still cannot run Sonar nor Cubase without long periods of slience, lots of stutters, and general unresponsiveness. It probably also adds to the DPC problems. The only audio editing program that works well is Adobe Audition. No doubt Adobe was aware of the problems in Win/Vista and Win/7 that they must have circumvented the Windows audio system with their own code. I can kill audiodg.exe and still run Audition with real audio output. Killing audiodg.exe otherwise disables all audio outptut from programs that "follow the rules". Just Google audiodg.exe and get an eyeful of the problems with this Windows "feature"... Dr. David McClain, N7AIG Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.529.2437 web: http://refined-audiometrics.com ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] audiodg.exe
Hi Tony I just installed an external Lexicon Alpha on my laptop, and downloaded the latest drivers. I told Cubase to be sure to use the Lexicon ASIO driver, not its generic driver. When I do that, I no longer get the audiodg.exe in the task list. And the system does run better, but not yet correctly. There are a few other Win/7 features that cause the system to go silent for several seconds. One may be wmiprvse.exe (Google for its issues), and I notice that whenever Cubase goes to disk, it pauses the sound. So, while there can be some improvements from eliding audiodg.exe, there are many more gotchas in these stupid little systems. The Dell laptop has only one USB hub inside, and *everything* (disk, video, USB, mouse, keyboard, scratchpad, etc.) competes for service. (That's why I referred to it as a stupid little system). I guess for 90+% of folks, the computer is good enough... :-( 73 de Dave, N7AIG On Jun 21, 2011, at 15:01, Tony Estep wrote: Wow, David, that is ugly. Thanks for the heads-up. This apparently is a misguided attempt by Microsoft to provide a process external to the protected kernel where audio developers could link their ASIO drivers etc. without getting into the supposedly secure area of the OS. My Win7 has such a file, but even when I install an ASIO driver and/or run audio programs, no process by that name appears in my Task Manager. I wonder what's different with my system?? 73, Tony KT0NY On Tue, Jun 21, 2011 at 4:19 PM, David McClain audiometrics.com wrote: Just found a nasty little "feature" in Win/Vista and Win/7, named "audiodg.exe". ... ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Images & 1500
Just noticed the same thing happening here while viewing a broadcast station at 7275 kHz. Noticed an image that seems to move in the opposite direction as I tune -- almost what you would expect from a first order alias signal. Running PSDR 2.4 and Win7/64. Switching on Spur Reduction made it go away for me. - de Dave, N7AIG On Oct 30, 2012, at 10:06 AM, George J Molnar wrote: > Brian (G3VGZ), I have noticed the same thing. Occasionally there will be > "upside down" signals several kHz away from the "real" ones, that sound > genuine. Saw them on 15 meters yesterday. I've also seen the backwards > movement on the panadapter, too. > > Sounds like it's not your installation. I have no amplifier, here. Also > running PSDR 2.4, under Win7/64. > > > George J Molnar > Las Vegas, USA > > Amateur Radio: KF2T > Twitter: GJMolnar > > > > > > > ___ > FlexRadio Systems Mailing List > FlexRadio@flex-radio.biz > http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz > Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ > Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/ > Dr. David McClain d...@refined-audiometrics.com ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
[Flexradio] Phase2 Display on Flex-1500
On my Flex-1500 the Phase-2 display always shows up, almost off-screen, in the upper left of the window pane. I understand that it is showing the raw, unfiltered, ADC/L and ADC/R. So when I go to the meter to select those two readings, my Right channel reads about 5-6 dB lower than the left reading -- hence the offset in the Phase 2 display. Is this something that I should be concerned about? as in send back to the factory for adjustment? or is it just fine the way it is? - de Dave, N7AIG Dr. David McClain d...@refined-audiometrics.com ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
[Flexradio] Noise Reduction??
Hi... wondering if some experts out there can verify... In playing around with the Flex-3000 and Flex-1500, it appears that the Noise Reduction (NR) algorithm becomes progressively less effective as the filter bandwidth becomes narrower. That, in turn, indicates to me that the NR algorithm is being applied after the narrow-band filtering. I use these kinds of adaptive LMS filters in audio processing, quite often for removing the sound of room reverb from poorly made recordings of speeches. The idea is that you want to use a pseudo-reference made by delaying the signal just slightly, to effect adaptation and filter tap updates. The reasoning is that a human voice waveform is highly correlated across time durations much less than 1 ms, while the room reverb and other white-ish noise in the recording are not cross-correlated to the speech recording beyond that same short delay. The adaptive filter removes the uncorrelated portion to a high degree (typically around 12-20 dB noise reduction). When you send an audio signal through a narrow filter, you end up increasing the correlation length of the noise component. You can't be very decorellated across intervals shorter than about 1/B for bandwidth B, since that is the approximate duration of the filter impulse response. There are fewer spectral components in the signal to create a decorellated component, and the destructive interference results in a wider wave packet. So it seems that we should be applying the NR adaptive process to audio with the widest possible bandwidth (narrowest possible noise correlation duration), and then sending the NR processed wide-band audio through a narrow reception filter, like the CW 250 Hz filters. What we seem to getting through things like the CW 250 Hz filters is next to no noise reduction with NR, and almost no fiddling with the algorithm parameters to accommodate the narrowed bandwidth seems to help by much. But it works progressively better as the bandwidth grows wider. - 73 de Dave, N7AIG Dr. David McClain d...@refined-audiometrics.com ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
[Flexradio] Questions about external reference oscillators
Hi all, I'm interested in doing some ionospheric research. Has anyone tried to use an external high-precision 10 MHz oscillator with a Flex-1500? I own and use a Flex-3000 right now. The F3K has an internal TCXO that is pretty good for most things, but it exhibits an irregular sawtooth variation in frequency over a period of about 45 minutes with a p-p amplitude of around 2-4 Hz. And that is almost as large as the kind of ionospheric doppler variations that I'm seeing on WWV. What I wonder about using an external reference with the F1.5K is whether or not it locks tightly to the reference -- i.e., shows no apparent variations beyond the reference drift (about 0.5 mHz for GPSDO Rubidium) -- or whether in fact an internal PLL inside the F1.5K will show the same kind of sawtooth wandering around the nominal reference frequency. In other words, is the sawtooth frequency variation due to the TCXO? or to the PLL used for DDO carrier generation? - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Questions about external reference oscillators
I have a long experience measuring various effects with differential techniques -- infrared measurements of stars (10-20 microns), dual- beam spectrophotometers, weak effects measured with synchronous detection systems... I'm sitting here trying to think of a way to measure frequencies with differential analysis, to remove the effects introduced by the receiver system. But so far, I always cycle back to needing a pristine reference signal for comparison. Does anyone have any ideas about how to do this without a secondary reference (e.g., Rubidium or Cesium clock)? Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Questions about external reference oscillators
OMG!! I found the cause of the oscillator wander! Just to be absolutely sure that I'm comparing apples with apples, I put all three radio audio outputs into an audio mixer and then fed the resulting mix into one soundcard, feeding SpectrumLab. The radios were all tuned to 9.985 MHz to give a 1500 Hz tone on the main carrier of WWV. The Flex was offset +5 Hz, and the QS1R was offset -5 Hz, with the Icom Pro3 centered. I was watching this slow drift unfold over time, and then a curious "coincidence" happened... The Icom stayed centered while the Flex and the QS1R nearly simultaneously began a rapid shift (in opposite directions). And after about 15 minutes, they both reached their maximum deviation in frequency and then began a slower drift back toward their originally assigned frequency offsets. Um yeah I watched this occur repeatedly. The QS1R would move a minute or so before the Flex 3K, and it would move farther in frequency, but both moved essentially together. What is happening at 45 minute intervals that would affect both radios at the same time??? I live in Tucson. The radios are on a desktop in a far corner of a rather large living room. Sure enough, whenever the central air conditioning cycled on, blowing cold air from a distant ceiling vent, that is when the radios began their rapid sawtooth rise. And when the central air finished its roughly 5 minute period of blowing, that's when both radios reached their peak departures and then began a slow drift back to their assigned offsets. Dang!!! I guess there is about a 5-10 degree variation. I see about 4 Hz variation in the Flex3K, and about 5-6 Hz in the QS1R. And that is what is causing my weird TCXO sawtooths. I see about 4 Hz variation in the Flex3K, and about 5-6 Hz in the QS1R. The IC756 is so well insulated that it hardly registers any change down inside the case. The QS1R is very small and has a lower thermal mass than the Flex3K, and that's why it begins showing about 1 minutes before the Flex3K. But the QS1R and the Flex3K are absolutely phase locked to the thermal cycling induced by the central air conditioner. Cold air was blowing directly across the radios... Thought you might want to know. - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Questions about external reference oscillators
The real question is, of course, how do you tell? I have used the same reference as input and watched it over a long period of time and it was rock solid within my ability to measure (about 0.1mHz). In other words, is the sawtooth frequency variation due to the TCXO? or to the PLL used for DDO carrier generation? Well, I haven't tested the 3000 but I see nothing like that with the 5000 or 1500. Well, this is very strange because I see it on the QS1R as well. A periodic, irregular sawtooth, with a period of about 45-50 minutes, p- p variation of about 2-4 Hz, imposed on a longer term diurnal drift. And both units utilize recent vintage TCXO's. I don't see it at all on my IC756ProIII -- it shows a very slow drift, amounting to less than about 1-2 Hz over 24 hours. Now, if I hadn't seen it in the QS1R, I might be tempted to believe that I have an errant F3K. But since I see essentially similar behavior in both the F3K and the QS1R, I'm tempted to believe that the problem is endemic to late design TCXO's. In both cases, the irregular sawtooth looks exactly like the kind of thing you'd see in a PLL tracking a reference with insufficient damping. The IC756 appears to have the damping just right, or perhaps a bit over-damped. Now someone has stated that recent TCXO's use microprocessors with lookup tables for the nonlinear corrections. And perhaps we are operating at the last bit in a limit cycle on the TCXO control? Who knows? But I'm happy to hear of your experience with an external reference. - de Dave, N7AIG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Questions about external reference oscillators
Hi Gerald, Yes, I am well aware that I'm getting superb performance from my F3K. It was spec'd at 1 ppm. It only looks bad when you use a huge magnifier on it... Heh! (...and so does my face!) Please, be assured that I meant no criticism of the Flex radio. I was simply trying to understand why one radio shows the (now known to be thermal) drift, while another, older radio does not. For all practical Ham uses, these variations are a nit and completely inconsequential. - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Oct 4, 2010, at 10:22, Gerald Youngblood wrote: David, That calculates to 0.4ppm drift on the FLEX-3000. Gerald Gerald Youngblood, K5SDR President and CEO FlexRadio Systems(TM) 13091 Pond Springs Road, #250 Austin, TX 78729 Phone: 512-535-4713 Ext. 202 Email: ger...@flex-radio.com Web: www.flex-radio.com Tune In Excitement (TM) PowerSDR(TM) is a trademark of FlexRadio Systems On Mon, Oct 4, 2010 at 6:51 AM, David McClain audiometrics.com> wrote: OMG!! I found the cause of the oscillator wander! Just to be absolutely sure that I'm comparing apples with apples, I put all three radio audio outputs into an audio mixer and then fed the resulting mix into one soundcard, feeding SpectrumLab. The radios were all tuned to 9.985 MHz to give a 1500 Hz tone on the main carrier of WWV. The Flex was offset +5 Hz, and the QS1R was offset -5 Hz, with the Icom Pro3 centered. I was watching this slow drift unfold over time, and then a curious "coincidence" happened... The Icom stayed centered while the Flex and the QS1R nearly simultaneously began a rapid shift (in opposite directions). And after about 15 minutes, they both reached their maximum deviation in frequency and then began a slower drift back toward their originally assigned frequency offsets. Um yeah I watched this occur repeatedly. The QS1R would move a minute or so before the Flex 3K, and it would move farther in frequency, but both moved essentially together. What is happening at 45 minute intervals that would affect both radios at the same time??? I live in Tucson. The radios are on a desktop in a far corner of a rather large living room. Sure enough, whenever the central air conditioning cycled on, blowing cold air from a distant ceiling vent, that is when the radios began their rapid sawtooth rise. And when the central air finished its roughly 5 minute period of blowing, that's when both radios reached their peak departures and then began a slow drift back to their assigned offsets. Dang!!! I guess there is about a 5-10 degree variation. I see about 4 Hz variation in the Flex3K, and about 5-6 Hz in the QS1R. And that is what is causing my weird TCXO sawtooths. I see about 4 Hz variation in the Flex3K, and about 5-6 Hz in the QS1R. The IC756 is so well insulated that it hardly registers any change down inside the case. The QS1R is very small and has a lower thermal mass than the Flex3K, and that's why it begins showing about 1 minutes before the Flex3K. But the QS1R and the Flex3K are absolutely phase locked to the thermal cycling induced by the central air conditioner. Cold air was blowing directly across the radios... Thought you might want to know. - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
[Flexradio] Interesting IMD?
I was listening to some WSPR up on 30m at around 10.140 MHz this afternoon. A bunch of DX'ers were booming in down around 10.115 MHz, and I was hearing them. They were registering around -60 dBm when I looked at them. Then I stood back and watched the waterfall for a while. Switching in the Attenuator helps a lot, but I noticed sidebands occurring about every 7 kHz both up and down the band. If I inject a strong carrier on my own, then I can make the situation much worse, with sidebands clearly visible from the DX crowd extending as far as 100 kHz above their frequencies. So having a strong signal somewhere in the A/D passband is a clue. In AM mode I also noticed that I was hearing a strong B'cast station down around 9.95 MHz, and sometimes WWV at 10 MHz too, all up at 10.140 MHz. Now the B'cast and WWV I can understand as due to pre-D/A self mixing in the receiver front-end, since they weren't even in the passband of the D/A. But the other stuff, and that curious spacing of sidebands at 7 kHz intervals really has me puzzled. What in the world would cause this sideband formation? I can think of no Post-A/D digital algorithms in the chain that would cause it. The signal levels going into the D/A were down around -40 dBFS (is that an overall peak level across the entire 96 kHz?), so there should be no clipping going on. Is this an artifact of Tayloe mixing? On a more conventional receiver these images do not appear. Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Interesting IMD?
Ach! The culprit was the NB. Disengaging the NB clears up the problem substantially. - de Dave, N7AIG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] 48000 v 96000 Sample Rate
You wouldn't want to just throw away every other sample unless you had also pre-filtered with a lowpass filter. Otherwise you could get severe aliasing, depending on what else is in the main passband. - de Dave, N7AIG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] 48000 v 96000 Sample Rate
Hi Joe, You have to change both PowerSDR on the Audio setup page -- sample rate and buffer size, *AND* change the Flex Driver to match. PowerSDR has to be turned off when you make the change. There may be some other buffer size settings that affect the filtering too, on the VAC page, but not sure about that. But the Driver and PowerSDR have to be on the same sheet of music... - de Dave, N7AIG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] 48000 v 96000 Sample Rate
Hi Joe, Not sure what "panel" you are talking about. The driver is the thing you use to set safe-mode and check for DPC's. Sounds like you have the correct thing. My understanding is that Normal mode is overly optimistic, and you should choose Safe Mode 1. In my case, with a Dell laptop running my F3K I have resorted to using Safe Mode 2 to avoid all the DPC problems. They still happen to me, but not as often. (People laugh at me when I state that I'm using a Dell laptop -- seems a better choice is a Mac Mini in BootCamp mode.) Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Oct 11, 2010, at 12:10, Joe Word wrote: Dave, Thanks for the answer. Is the Flex Driver in the panel labelled FlexRadio/Global Settings? If so, I would change Sample Rate and Buffer Size correct? Also should Operation Mode be Normal or Safe Mode Level 1? I am running v2.0.8 Joe N9VX On Mon, Oct 11, 2010 at 2:43 PM, David McClain wrote: Hi Joe, You have to change both PowerSDR on the Audio setup page -- sample rate and buffer size, *AND* change the Flex Driver to match. PowerSDR has to be turned off when you make the change. There may be some other buffer size settings that affect the filtering too, on the VAC page, but not sure about that. But the Driver and PowerSDR have to be on the same sheet of music... - de Dave, N7AIG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] 48000 v 96000 Sample Rate
This is not the case for PowerSDR v2.x. You do NOT change the sampling rate or buffer sizes in the Firewire Control Panel. Only change them in the PowerSDR Setup form - they will automatically sync the Firewire sampling rate and buffers. Changing it in the Firewire control panel may get them out of sync since the synchronization process is not bi-directional. ... interesting new feature... - de Dave, N7AIG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] My Rookie Performance in the Recent FMT
Brian, I found the Fldigi system interesting... and at first you had me thinking someone had found a way around the Heisenberg Uncertainty Principle, in terms of making rapid measurements of fine frequency differences... So I did some research on Fldigi, tried it out, and looked up AFC algorithms on Google... I found this really interesting survey article, with some real meat in it (not for the faint of heart): http://engnet.anu.edu.au/DEcourses/engn3214/notes/FDNatali.pdf At any rate, in there you find the DFT variant (discrete Fourier Transform) which Fldigi may, or may not, be using. At any rate, I'm right at home with FFT's and I could readily see that there is no speedup by using a tracking filter. It takes the same amount of time to approximate fine frequency deviations, no matter how you do it. That is very reassuring to me, having spent much of my career trying desperately to estimate things like precession periods of incoming nuclear warheads, (which are around 100 mHz), based on only a few seconds of observations... But what is great about Fldigi, compared to SpectrumLab, is that the tracking filter approach performs a kind of continuous averaging for you, in displaying the estimated tracked frequency. However, I tried using SpectrumLab last night, where I produced a strip-chart of the estimated FFT frequencies. SL uses FFT bin interpolation, on the assumption that (a) SNR is good and high, (b) no nearby interfering signals, and (c) they know the shape of the windowing function. It is a fancier form of DFT AFC than described in the paper (above). There is a huge advantage to having a strip chart recording because, instead of trying to estimate changing trends by eyeballing individual measurements as they come flying past, you can actually see the p-p frequency deviations and the chatter on top of the longer term cyclic trends. Fldigi seems to use hard-coded loop bandwidths and capture ranges. 2 Hz capture range, 5 sec integration time. Good, but I would sure like to be able to tweak those myself. SpectrumLab also has its limitations in that there aren't sufficient computational capabilities built into its tiny analysis programming language. (Oh, how I dream of just doing it all myself in Lisp, with Lisp fully present all the time... that way you can make up ad-hoc measurements that would never have been thought of until you need them. Closed systems can't possibly anticipate every need. I'm getting dangerously close to rolling up my sleeves and just doing it. If I do, then I'll share with everyone too...) - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Oct 10, 2010, at 16:34, Brian Lloyd wrote: On Sat, Oct 9, 2010 at 4:41 PM, Jerry Flanders wrote: An additional source of error is the fact that the 5000 does not tune continuously. It tunes in steps, and they are irregular, so an additional correction is required. When you are watching your WWV phase display, a part of the error you see may be due to this. This may not be important unless you are trying for sub -100 mHz accuracy. Well, I did pretty well and figure I should comment here. First thing I want to say is: I was lucky. I should not have done as well as I did. I will explain why as I go along. My setup: Flex 5000, LPRO-101 Rb reference, beta PowerSDR, Fldigi 3.21.0AM. Sources of error: 1. accuracy of the reference; 2. tuning accuracy of the Flex 5000 DDS; 3. ionospheric doppler error. The LPRO-101 Rb reference is pretty close to on-the-money. Error there should be less than 1mHz at 10MHz so this is not a significant source of error. The Flex 5000 uses an Analog Devices AD9959 DDS chip to generate the LO signals. This chip uses a control word which is really a fraction by which the 500MHz clock is multiplied. The result does not necessarily fall on an exact 1Hz boundary. It is correct to +/- 55mHz (an overall peak-to- peak frequency error of 110mHz). Analog Devices has a calculator that will tell you the actual error for any given input. It also appears that Flex does not use all of the bits in the tuning word further reducing accuracy. (The error is still well under +/-0.2Hz but that isn't really good enough for an FMT.) The last source of error is the ionosphere. I use fldigi's frequency measurement function to repeatedly sample the frequency. When measuring frequency fldigi phase-locks to the signal and accumulates phase error in order to calculate frequency rapidly. I get about 1800 frequency data points in a 2 minute sample period. That data goes into a spreadsheet where I plot the data and do statistical analysis. The plots for 40m and 80m (I couldn't hear the
Re: [Flexradio] My Rookie Performance in the Recent FMT
Okay Frank! Wow, I didn't expect that response! I mostly use LispWorks 6 at this time, but I also have Allegro 8.1 here, and try to keep my code mostly portable between them. The latest LW has some really awesome SMP capabilities. But I am a firm believer in Lisp for this kind of thing, and for embedded systems too! (Lisp is what Forth always wanted to be when it grew up...) Everytime I turn on the computer and start using other people's code, I just squirm in my seat, chomping at the bits to correct this or that, or to provide ad-hoc capabilities that the rest of the world apparently hasn't ever experienced. I may take you up on this offer. I already have a solid basis for the signal processing chains. See: http://www.spectrodynamics.com/id65.html - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Oct 11, 2010, at 13:02, Frank Goenninger wrote: Am 11.10.2010 um 21:55 schrieb David McClain: (Oh, how I dream of just doing it all myself in Lisp, with Lisp fully present all the time... that way you can make up ad-hoc measurements that would never have been thought of until you need them. Closed systems can't possibly anticipate every need. I'm getting dangerously close to rolling up my sleeves and just doing it. If I do, then I'll share with everyone too...) If you need support, a few more hands in coding Lisp, or anything else Lisp-related then please let me know. I'd be ready to jump in on such an effort ... AllegroCL Enterprise Edition ready to use here. Currently developing a Flexible UI as a PowerSDR replacement (à la QT-based SDR Console over at dttsp mailing list). I did have the same dream for a looong time now ;-) 73 Frank DG1SBG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] My Rookie Performance in the Recent FMT
yes, one of my points in my note, but not made clearly, was that with an integration time of only 5 seconds, by rights, you could lay claim to 200 mHz precision, or thereabouts. So you did very well, considering. Don't know what TANSTAAFL means? having it all there where you can come up with new manipulations on the fly. Yeah, that would be pretty cool. I don't think that is going to happen with the current crop of operating systems. And I don't understand your statement here. What has to OS to do with it? I normally live inside a totally interactive and incremental system, in which previously developed code can be molded to fit ad- hoc needs. That is provided by the Lisp systems, and they run the same code on OS X, Windows, and Linux, so you have almost instant portability. Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Oct 11, 2010, at 19:14, Brian Lloyd wrote: On Mon, Oct 11, 2010 at 12:55 PM, David McClain audiometrics.com> wrote: Brian, I found the Fldigi system interesting... and at first you had me thinking someone had found a way around the Heisenberg Uncertainty Principle, in terms of making rapid measurements of fine frequency differences... Nope. TANSTAAFL. So I did some research on Fldigi, tried it out, and looked up AFC algorithms on Google... I found this really interesting survey article, with some real meat in it (not for the faint of heart): http://engnet.anu.edu.au/DEcourses/engn3214/notes/FDNatali.pdf At any rate, in there you find the DFT variant (discrete Fourier Transform) which Fldigi may, or may not, be using. fldigi is an open source software project. You can download the code and poke at it if your are of a mind to do that. At any rate, I'm right at home with FFT's and I could readily see that there is no speedup by using a tracking filter. It takes the same amount of time to approximate fine frequency deviations, no matter how you do it. I wouldn't expect there to be any speedup. That is very reassuring to me, having spent much of my career trying desperately to estimate things like precession periods of incoming nuclear warheads, (which are around 100 mHz), based on only a few seconds of observations... But what is great about Fldigi, compared to SpectrumLab, is that the tracking filter approach performs a kind of continuous averaging for you, in displaying the estimated tracked frequency. Yes, it does. But it is still changing too rapidly and you just can't get a feel for where the mean or RMS value is. That is why I take the csv data that fldigi produces and then massage that. Plotting it (equivalent to your strip chart below) is necessary for me to get a feeling for the kinds of periodic errors present in the data. However, I tried using SpectrumLab last night, where I produced a strip-chart of the estimated FFT frequencies. SL uses FFT bin interpolation, on the assumption that (a) SNR is good and high, (b) no nearby interfering signals, and (c) they know the shape of the windowing function. It is a fancier form of DFT AFC than described in the paper (above). There is a huge advantage to having a strip chart recording because, instead of trying to estimate changing trends by eyeballing individual measurements as they come flying past, you can actually see the p-p frequency deviations and the chatter on top of the longer term cyclic trends. Fldigi seems to use hard-coded loop bandwidths and capture ranges. 2 Hz capture range, 5 sec integration time. Good, but I would sure like to be able to tweak those myself. Yup! But since fldigi is open source, you should be able to add in the knobs to let you tweak the capture range and integration time. I could also see changing the coefficients of the tracking filter on-the-fly depending on rate-of-change of frequency. SpectrumLab also has its limitations in that there aren't sufficient computational capabilities built into its tiny analysis programming language. I can imagine that. I was planning to develop some tools using Mathematica. It has a number of intrinsics that might make massaging that data useful. (Oh, how I dream of just doing it all myself in Lisp, with Lisp fully present all the time... that way you can make up ad-hoc measurements that would never have been thought of until you need them. Closed systems can't possibly anticipate every need. I'm getting dangerously close to rolling up my sleeves and just doing it. If I do, then I'll share with everyone too...) I agree with this. Have things running inside your programming environment where you can change anything you want on-the-fly seems very cool to me. We have gotten so used to static, unc
Re: [Flexradio] Digital mode 48000 v 96000 Sample Rates
But the other thing you have to consider is that your digital mode program also has its own DSP filters. It probably isn't necessary to have brick-wall filters in the radio. All I can say is that you should experiment. Yes, and all the digital mode programs I am aware of use FFT's. So the ultimate filter is the width of 1 FFT bin (longer FFT's imply narrower filters), and the kind of pre-windowing function used (Hann, Blackman, etc). So no matter what kind of IF and audio filtering you use, it makes no difference -- as long as those IF and audio filters aren't distorting the passband to any great degree. In fact, an argument could be made that you should strive for the widest reasonable bandwidth on the incoming audio and IF, so that you can avoid noise aliasing artifacts that might increase the noise level within each FFT bin. In other words, take the full bandwidth that your digital mode program can handle, and don't do any filtering, apart from the FFT itself. eh? - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Oct 12, 2010, at 14:20, Brian Lloyd wrote: On Tue, Oct 12, 2010 at 12:57 PM, Brian Lloyd wrote: On Tue, Oct 12, 2010 at 9:57 AM, Drax Felton wrote: So what sample rate / buffer sizes would be best for someone interested in weak signal digital mode operation? I'd assume matching the 48000 VAC sample rate because a. Latency isn't that big of a deal to the digi mode software. (with JT65 maybe not) b. Only need to see a small bandwidth on the panadapter at high resolution. Any others? Am I correct? Well, there is no real advantage to matching the 48kHz sampling rate out to VAC. At each point where you change sample rates you have to do sample-rate conversion. When the new sample rate is a power-of-two submultiple (divide by 2, 4, 8, 16, 32, etc.) then resampling is no issue. When the new rate is not related to the old rate then you can have problems. So going from 192kHz, 96kHz, or 48kHz to 48kHz is not a problem. But you have to remember that there is yet another sample rate conversion that is going to happen in your digital mode program as most of them use either 8kHz or 11.025kHz internally, depending on the modem. (11.025 kHz is the standard 44.1kHz used by CDs and most sound cards divided by 4.) Resampling to 8kHz is not a problem (power of 2 again) but resampling to 11.025kHz can produce distortion products depending on the quality of the interpolator code. You might want to try to match the digital modem's internal sample rate in VAC to eliminate one resampling. So if you don't care about pan/waterfall span, 48kHz sample rate is good. Do that with a 4096 sample buffer (in the DSP RX buffer settings) will net you some *seriously* sharp filters. But the other thing you have to consider is that your digital mode program also has its own DSP filters. It probably isn't necessary to have brick-wall filters in the radio. All I can say is that you should experiment. -- Brian Lloyd, WB6RQN/J79BPL 3191 Western Dr. Cameron Park, CA 95682 br...@lloyd.com +1.767.617.1365 (Dominica) +1.931.492.6776 (USA) (+1.931.4.WB6RQN) ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Digital mode 48000 v 96000 Sample Rates
... sorry, my wife always accuses me of starting in the middle of the story... Noise aliasing? That gets back to Brian's original point about sample rate conversion. You want to avoid that because it invariably involves some noise aliasing, above and beyond what has already happened to the passband by way of the Flex radio itself. Adding additional filtering does not, in itself, add noise aliasing. But then again, it doesn't do anything for you beyond adding additional ripple across the passband. Best approach would seem to be to accept whatever passband you get from the Flex, and try to match that passband width with your digital mode soundcard input. So if the soundcard selection limits you to a max of 48 kHz, then choose 48 kHz from Flex too. - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Oct 12, 2010, at 14:20, Brian Lloyd wrote: On Tue, Oct 12, 2010 at 12:57 PM, Brian Lloyd wrote: On Tue, Oct 12, 2010 at 9:57 AM, Drax Felton wrote: So what sample rate / buffer sizes would be best for someone interested in weak signal digital mode operation? I'd assume matching the 48000 VAC sample rate because a. Latency isn't that big of a deal to the digi mode software. (with JT65 maybe not) b. Only need to see a small bandwidth on the panadapter at high resolution. Any others? Am I correct? Well, there is no real advantage to matching the 48kHz sampling rate out to VAC. At each point where you change sample rates you have to do sample-rate conversion. When the new sample rate is a power-of-two submultiple (divide by 2, 4, 8, 16, 32, etc.) then resampling is no issue. When the new rate is not related to the old rate then you can have problems. So going from 192kHz, 96kHz, or 48kHz to 48kHz is not a problem. But you have to remember that there is yet another sample rate conversion that is going to happen in your digital mode program as most of them use either 8kHz or 11.025kHz internally, depending on the modem. (11.025 kHz is the standard 44.1kHz used by CDs and most sound cards divided by 4.) Resampling to 8kHz is not a problem (power of 2 again) but resampling to 11.025kHz can produce distortion products depending on the quality of the interpolator code. You might want to try to match the digital modem's internal sample rate in VAC to eliminate one resampling. So if you don't care about pan/waterfall span, 48kHz sample rate is good. Do that with a 4096 sample buffer (in the DSP RX buffer settings) will net you some *seriously* sharp filters. But the other thing you have to consider is that your digital mode program also has its own DSP filters. It probably isn't necessary to have brick-wall filters in the radio. All I can say is that you should experiment. -- Brian Lloyd, WB6RQN/J79BPL 3191 Western Dr. Cameron Park, CA 95682 br...@lloyd.com +1.767.617.1365 (Dominica) +1.931.492.6776 (USA) (+1.931.4.WB6RQN) ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Digital mode 48000 v 96000 Sample Rates
Generally speaking you are correct. The problem occurs when an undesired signal gets into the passband of the radio and actuates the AGC of the receiver. If the offending signal is much stronger than the desired signal the gain of the receiver may be Ahh.. very good point about that AGC action... Brian is, of course, quite correct. Also, some seriously strong QRM will bleed across the FFT bins, depending on the windowing function, of an unfiltered input passband. And that is probably far worse than any noise aliasing. So in that case, use the best possible filtering you can, and that happens at low Flex sample rates and large buffer sizes. Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Digital mode 48000 v 96000 Sample Rates
Also, a hint about what *really* matters in weak signal work... Your theoretical noise floor is going to be the noise contained in the entire incoming bandwidth divided by the FFT size. The single most important thing I have found for my weak signal work is -- ta da !! -- a good resonant antenna! I began with a junk random length dipole on the roof, and then constructed a 30m HW dipole out of some old RG58 network cable just for kicks. I found that, for the same QRN noisy location, the tuned dipole gave at least 6 dB better SNR coming into the receiver. To get that much improvement from purely digital processing you'd have to quadruple the FFT length. But remember, if the signal is already below the noise, no matter how much processing you do, you won't be able to dig it out. That 6 dB improvement applies across the board. So for the bin or two in the FFT where the signal resides, it was 6 dB stronger than before, allowing it to stand out from the previous bin noise floor. This improvement works wonderfully, until your signals are about as weak as the cosmic background noise (or atmospherics at lower frequencies). - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Oct 12, 2010, at 14:55, Jack Haverty wrote: Wow. Sample, sample, fft, dsp, resample and send as serial stream, FFT again, more dsp, etc. It seems like there's way too many superfluous steps in the path when you're using digital modes. I wonder if anyone's working on some kind of "native" digital SDR - e.g., PowerSDR or equivalent where the digital modes are embedded just like they are now with AM, SSB, etc. No more VAC and redundant filtering et al. /Jack On Tue, 2010-10-12 at 14:41 -0700, David McClain wrote: But the other thing you have to consider is that your digital mode program also has its own DSP filters. It probably isn't necessary to have brick-wall filters in the radio. All I can say is that you should experiment. Yes, and all the digital mode programs I am aware of use FFT's. So the ultimate filter is the width of 1 FFT bin (longer FFT's imply narrower filters), and the kind of pre-windowing function used (Hann, Blackman, etc). So no matter what kind of IF and audio filtering you use, it makes no difference -- as long as those IF and audio filters aren't distorting the passband to any great degree. In fact, an argument could be made that you should strive for the widest reasonable bandwidth on the incoming audio and IF, so that you can avoid noise aliasing artifacts that might increase the noise level within each FFT bin. In other words, take the full bandwidth that your digital mode program can handle, and don't do any filtering, apart from the FFT itself. eh? - de Dave, N7AIG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Digital mode 48000 v 96000 Sample Rates
you can!! As it happens, I use CWSkimmer directly off the (Microphone Flex) input source (as it shows up in my audio selection list). You don't need VAC to use CWSkimmer. You can do this for SpectrumLab too, but be aware of the 9 kHz IF shift that Flex uses, or else use the SPEC demodulation choise in PowerSDR (i.e., no demodulation). - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Oct 12, 2010, at 15:06, Drax Felton wrote: I have been wondering the same thing. Why do I have to pretend to have a sound card interface. With the bandwidth of the Flex I ought to be able to have a digi mode software that reads the direct I/Q stream of an entire RTTY/Data sub band, detects the modes, and gives me a huge panadapter decoding of all signals simultaneously regardless of mode like the HRD Super Browser does for PSK31. Simon! .. you up for it? -Original Message- From: flexradio-boun...@flex-radio.biz [mailto:flexradio-boun...@flex-radio.biz] On Behalf Of Jack Haverty Sent: Tuesday, October 12, 2010 5:56 PM To: David McClain Cc: Flexradio list Subject: Re: [Flexradio] Digital mode 48000 v 96000 Sample Rates Wow. Sample, sample, fft, dsp, resample and send as serial stream, FFT again, more dsp, etc. It seems like there's way too many superfluous steps in the path when you're using digital modes. I wonder if anyone's working on some kind of "native" digital SDR - e.g., PowerSDR or equivalent where the digital modes are embedded just like they are now with AM, SSB, etc. No more VAC and redundant filtering et al. /Jack ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Digital mode 48000 v 96000 Sample Rates
Hi Dan, That happens becuase (A) the signal is strong enough to see above the noise floor, and (B) that noise floor drops, inside each FFT bin, by the size of the FFT being used. So while noise divides, more or less, evenly across all the FFT bins, the signal itself does not. Hence a vast improvement in the in-bin SNR. - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Oct 12, 2010, at 15:10, Drax Felton wrote: So how do the Olivia and JT65 modes software report that they receive signals, that I can't see that, are -26db S/N ? -Original Message- But remember, if the signal is already below the noise, no matter how much processing you do, you won't be able to dig it out. ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Digital mode 48000 v 96000 Sample Rates
For JT65 - that reported -26 dB SNR is a fictitious number, created by showing you what SNR your signal would have had, if it had been measured in a 2.5 kHz bandwidth. In fact, the signal is very weak, but still above the ultimate noise floor. And you can get closer to that ultimate noise floor by using extremely narrow filters = FFT bin size. But if your location QRN is very high, or you have a lousy antenna, then the signal itself would fall below the actual ultimate noise floor, and you wouldn't ever be able to see it no matter how large your FFT size is. - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Oct 12, 2010, at 15:18, David McClain wrote: Hi Dan, That happens becuase (A) the signal is strong enough to see above the noise floor, and (B) that noise floor drops, inside each FFT bin, by the size of the FFT being used. So while noise divides, more or less, evenly across all the FFT bins, the signal itself does not. Hence a vast improvement in the in-bin SNR. - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Oct 12, 2010, at 15:10, Drax Felton wrote: So how do the Olivia and JT65 modes software report that they receive signals, that I can't see that, are -26db S/N ? -Original Message- But remember, if the signal is already below the noise, no matter how much processing you do, you won't be able to dig it out. ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Digital mode 48000 v 96000 Sample Rates
I've accidentally set the output of HRD/DM780 to be the "Microphone (FlexRadio Microphone)" device. The result was blasting loud audio bringing the Flex to its maximum RF wattage in the 110 watt range! Heh! I did that too the first night I had mine... and I was listening on another receiver and noticed I was actually producing DSB modulation... No, you just want to hook up to the Microphone Input Line from Flex, as in Windows vernacular. You never want to send anything directly to the Flex Output port. Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Oct 12, 2010, at 15:21, Drax Felton wrote: I've accidentally set the output of HRD/DM780 to be the "Microphone (FlexRadio Microphone)" device. The result was blasting loud audio bringing the Flex to its maximum RF wattage in the 110 watt range! I was thinking more of a program designed with Flex in mind though. -Original Message- From: flexradio-boun...@flex-radio.biz [mailto:flexradio-boun...@flex-radio.biz] On Behalf Of David McClain Sent: Tuesday, October 12, 2010 6:17 PM To: Flexradio list Subject: Re: [Flexradio] Digital mode 48000 v 96000 Sample Rates you can!! As it happens, I use CWSkimmer directly off the (Microphone Flex) input source (as it shows up in my audio selection list). You don't need VAC to use CWSkimmer. You can do this for SpectrumLab too, but be aware of the 9 kHz IF shift that Flex uses, or else use the SPEC demodulation choise in PowerSDR (i.e., no demodulation). ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
[Flexradio] Question about Flex CODEC Stability?
In preparation for the arrival of a Rb GPSDO here, I have been watching the 100 Hz sideband on 10 MHz WWV all day long, and recording its reported variations -- presumably due to CODEC clocking jitter and drift. With a sample rate from the Flex3K of 48 kHz, and an effective FFT resolution of around 11 mHz per cell, I have been watching this on SpectrumLab, and find frequency deviations of around 0.2 mHz RMS, and +/- 1 mHz p-p. The fine-scale resolution comes from SpectrumLabs interpolation routines for finding peak frequencies. Sounds pretty good -- but this implies an accuracy in the CODEC clock of around 0.004 ppm !! (10^-9). Furthermore, I see essentially zero long-term drift at this resolution. Now, how can this be? The Flex3K, if it used its internal 0.1 ppm clock as the clock for the CODEC, should give about 25x worse than this. And if it isn't using the internal TCXO for the CODEC, then why aren't we using the same superb CODEC clock for driving the whole rig? Just doesn't make any sense to me... - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Break-in Keying from either the keyboard or an external keyer.
Ahh! I get that kind of noise all the time from my ComCast DVR. It appears to be a "faulty" switching power supply. I hunted for a week to find all the sources of noise around here (very high QRN). I finally tracked it down to that DVR in the kitchen. It happens to be about 10 feet from my counterpoise, and I suspect that counterpoise is picking it up. My antennas are at least one wavelength distant from it. When I unplug the DVR from the wall outlet, the noise vanishes. - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Oct 13, 2010, at 20:03, Ron Feltman wrote: Can anyone tellme why I can't execute break-in keying with the flex-1500.The transmitter never switches fromm transmit to recedive! Ron Feltman "Congress shall make no law that applies to the citizens of the United States that does not apply equally to the Senators and/or Representatives; and, Congress shall make no law that applies to the Senators and/or Representatives that does not apply equally to the citizens of the United States ." The Godless Constitution: It is not true that the Founders designed a Christian commonwealth, which was then eroded by secular humanists and liberals; the reverse is true. The framers erected a god-less federal constitution structure, which was then undermined as God entered first, the U.S. currency in 1863, then the federal mail service in 1912, and finally the Pledge of Allegiance in 1954. Global Warming: Not man, but nature rules the climate. The Kyoto Protocol and the IPCC reports, tuned by Malthusian ideas, may surely make a lot of noise and cause enormous harm for the global economy and for the well-being of billions of people. But they can do nothing for the climate. This we shall learn in the near future. Post Script: It seems that that this has been confirmed by the recent "revelations " of the scientific community! ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Anyone want to take a crack at identifying the source of this noise?
Around here I also found the Comcast telephone / internet modems particularly bad. And they generate similar 20 dB over noise floor noise, with a repetition every 100 kHz or so. Keep your antenna as far away from it as possible. The Comcast tech made some measurements, and mostly found it "acceptable" (to whom?!). He did shorten some of the coax lines leading to the junction box, and that may have detuned the coax shields from resonance. When I researched the modems, I found that they use barely any shielding at all, lots of RF bleeds over the outer shield of the coax and over the RJ45 network connections. I made the network lines into chokes by coiling them. Could probably use some good ferrites over them too. Enclosing the whole thing in a NEMA box would help enormously. I used aluminum foil one night, but the modem will cook itself to death with prolonged use. But what I found most infuriating, apart from Comcasts cavalier attitude about polluting the RF space, is that the units are made in China (where else?), and they have a fake FCC certification sticker on them. Anything for a buck, eh? Final solution was to move the antennas as far away as possible from all the Comcast coax lines leading around the outside of the house. - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Oct 14, 2010, at 18:06, Steve Sterling wrote: This has been a good thread-- a couple more sources I lost hair over. -- those darn touch-on/ touch-off lamps. My wife had about 5 of those in the house. Down to one. 200khz up to about 60mhz -- a particular brand of multi-stage battery charger/float "converter" in the travel trailer or motor home. Its the OEM unit in 90% of the newer trailers and motor homes sold. It took a major effort to find, since it was s strong, it would break squelch on my 70cm FM mobile a block away. It just swamped my whole property. I couldn't find it with signal strength walk-around. Finally, I tripped all the breakers in the panel, then powered each circuit up until I came to the 30amp breaker feeding the trailer power. ah ha. Replaced with a name-brand unit-- silence was golden. -- various consumer wireless routers, port switches, et. al. Linksys are the worst, but I haven't found any that didn't create spurs somewhere in the ham bands, clear up through 2 meters. Replaced my main switch with a Cisco 24 port commercial unit off of ebay for $60, and a Cisco commercial wireless access port-- end of those problems. Not every noise solved, and like others reported, new ones keep showing up. On 10/13/2010 8:23 PM, David McKenzie wrote: Check out this video of the noise I get on 2 meters. It comes up well over 20 over on some days, and a few weeks ago I heard a football game on the carrier. Escalated it to Comcast who is very certain it isn't them but is being very helpful in helping me track it down. It's strongest horizontally, and it's definitely not coming from here, and is strongest in a certain direction as identified by watching it while spinning the beams. It also drifts up and down the band but repeats out about 200kHz. http://www.youtube.com/watch?v=ZKKhySrO5Kw ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
[Flexradio] New F1.5K - Initial Findings
Hi all, I just received a Flex-1500 today, and started immediately taking some frequency measurements, since it has a reference input. Very interesting findings, but feel free to correct me if I'm wrong... All tests were performed this late evening at 10 MHz WWV, with little evidence of ionospheric Doppler shift. Alongside this, I used a Rb GPSDO for comparisons. 1. Using external reference to phase lock the F1.5K DDS TCXO, calibrate to WWV 10 MHz. I get a -6 count DDS clock offset. At first I was surprised that I should get any offset at all. But then I reasoned that the on-board TCXO is being phase locked to the reference, and there could be some offset from perfect sync. Does this indicate that we are slightly overdamped in the PLL? I see approximately 46.6 mHz per DDS step. Uncertainties in the step size arise from the CODEC feeding SpectrumLab. The closest calibration to WWV 10 MHz puts me about 5.6 mHz below with a -6 DDS offset. I made the calibration first, thinking that it may affect the apparent clock rate of the CODEC. (it turns out to have little or no effect on the CODEC) 2. It appears that the CODEC is running on its own crystal in the F1.5K. I see no significant difference in behavior between using the external reference, or not. For this test, I tune to 10 MHz, AM detection, with passband filtering from -1550 Hz to +50 Hz, set a Function Generator to 9.998490 MHz (which is 1510 Hz below WWV) to produce a 1510 Hz beat with the WWV carrier. I also AM modulate the Function Generator with a 1512.5 Hz tone. This puts another line 2.5 Hz above the WWV carrier at 1512.5 Hz. Both the Function Generator carrier and modulation tone are phase locked to the 10 MHz reference. The 2.5 Hz spacing was chosen to avoid the 1 Hz tick sidebands and its harmonics. Now I can track both the WWV carrier at 1510 Hz and my Function Generator tone at 1512.5 Hz, as well as measure their frequency difference. And this measurement is wholly unaffected by F1.5K LO wander, so it works as well with, or without, locking the F1.5K to the external reference. I tune the sample rate for the F1.5K sound-card input, in SpectrumLab, until I see a "best behavior" at both carrier tones. Best behavior is defined as seeing approximately equal frequency deviation ripple about the zero deviation line. Both tones actually wander by about 1 mHz p-p, even though their frequency difference ranges +/-0.3 mHz. So I attribute the small frequency difference variations to (A) true offset of my reference from Cesium, although that should be very tiny (< 1e-11, or 0.1 mHz at 10 MHz), and (B) variations in the actual CODEC clock frequency. The frequency difference between the two tones is much smaller than the total wander in the tones themselves -- they wander pretty much together, with an approximately sinusoidal variation of period 4 minutes. So this result surprised me quite a bit. I had expected that the CODEC clock would be derived from the TCXO which drives the DDS. But that appears not to be the case, since nearly identical wander, in period and p-p frequency variation, occurs whether or not the F1.5K is locked to the external reference. Offhand, the Flex-3000 shows better behavior in its CODEC wander, which is mainly driven by thermal variations. So I get the distinct impression that the CODEC *is* being driven by the DDS TCXO in the F3K. Tone wander is on the order of 0.1 mHz p-p, or about 10x better than I see on the F1.5K. But on the plus side, I no longer live in constant fear of the dreaded DPC's that I get from the F3K, running on a late model Dell laptop computer. And at 5 Watts output, I'm unlikely to see any problems arising from RF in the shack. I have yet to try transmitting with the F1.5K, but it looks like a dandy WSPR and QRSS rig. Nice work Flex!! - de Dave, N7AIG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] New F1.5K - Initial Findings
BTW... just fired up the new F1.5K here on 30m WSPR. It does really nicely, even down to 100 mW output. This is a great little rig! - de Dave, N7AIG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
[Flexradio] Re: Anyone know how to reset the firmware in the F1.5K?
Hi, I want to step back to a previous version, after having updated the F1.5K firmware. The new beta version goes banana's at 85%+ CPU utilization (on 4 CPU's !!), nothing but 100% audio stutter, and I need to step backwards... (I know, you can never go back...) But unless I can reset the firmware, I'm stuck with a brand new F1.5K that is useless until / unless they find the driver problems... Anybody? - de Dave, N7AIG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
[Flexradio] DPC's ?? Why a problem?
Hi, I have an audio studio with 3 different MOTU interfaces, a couple of Iomega interfaces, a Digidesign interface, and a couple of SymbolSound DSP processing units, plus several digital hard disk recorders, and numerous digital effects boxes from various manufacturers. The computers are Mac and PC's running Sonar, Live, Digital Performer, Cubase, Audition, FruityLoops, and many other DAW's. None of these systems ever produces a single audio dropout, stutter, or lock up. Of course, they would do it a lot if I weren't careful to designate one of them as the master clock source for the entire system. All sampling clocks are locked to one master, for better or worse. So why are DPC's a problem for the Flex radios? A Pentium class CPU is the equivalent of dozens of older MC56309 DSP's. And PowerSDR is not doing any really heavy lifting in terms of DSP processing. So why?? - de Dave, N7AIG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] PwrSDR 2.0.16, Flex-1500 and the TX Delay
I was avoiding getting too deep into the Flex CAT protocol, until this afternoon. WOW! I can't believe how *easy* it is to get the Flex to respond to my queries and commands. Took all of about 10 minutes to get things humming under Lisp. New wish list item: A Flex CAT command to start / stop audio recording. - de Dave, N7AIG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] VAC Level Settings
Here here here!!! I have been pushing for some time now, to have each application provide an input gain/attenuator. The sound source's only responsibility should be to ensure that clipping is not occurring. Beyond that, it is asking too much from a shared source to provide specific levels to client programs. A quick look around the audio engineering world would have made this a no-brainer requirement on all client programs. Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Nov 16, 2010, at 11:25, Steve Sterling wrote: Anyone know if there is a way to store and reselect VAC level settings? re: pSDR v1.18.06 Let me explain my question. After having my F5K for a few months, I am starting to get a plethora of apps I want to attach to VAC at various times. DigiPan, N1MM (voice drone), Skype, IP-Sound, etc. I haven't hooked up CW-skimmer yet, but soon! Each one of these seem to need different level settings. In the digital modes, at least the "Mode Specific Controls" for RX and TX gain are on the main panel. There doesn't seem to be a way to set the TX Profile (profile save under setup>>transmit is grayed out) like you can with mic audio, although the pulldown is there. Thus when switching between digital apps, I must record the gain values on paper and readjust on a change. On voice-- Skype, IP-Sound, etc there isn't even main screen adjustments. To change requires going into setup>>audio>>VAC and making the RX and TX gain changes there. Big difference in the numbers used also. My uneducated guess is they use different numbers of bits-- obviously different codecs. Anyway, on voice, again I have no way to store values within the app-- it requires me keeping track and manually making the settings as I shift. Am I miss understanding a feature in the software? Is there some grand VAC traffic cop app like ddUtil for the com port traffic? Steve WA7DUH ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
[Flexradio] Wish List...
New wish list item: Have an internal oscillator in software generate a tone that is fed to the audio output stream, for CODEC calibration in frequency measuring tests... Right now you have an internal frequency tone generator, but there is no specific frequency for it, just a slider, and the audio output gets muted, so I can't see or hear it on external tools. Dr. David McClain, N7AIG Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
[Flexradio] Questions about antennas and spectral displays...
I think I know the answer, but I want to double check... That 43 foot antenna... is it an end-fed random wire off the back of the tuner? or a center-fed dipole, with total span 43 feet? Also, as a side question... Anyone happen to know why all the other higher-end modern rigs (Icom, Kenwood, Yaesu, Ten-Tec), use swept spectrum analyzers instead of using an FFT like Flex Radio uses? - de Dave, N7AIG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Questions about antennas and spectral displays...
Wow! I'm really glad I asked the "dumb question"... Thanks for all the illuminating answers. - de Dave, N7AIG On Dec 21, 2010, at 13:29, Tim Ellison wrote: A 43 foot vertical or end-fed or a 86' dipole. I created a spreadsheet several years back that found all of the non-even harmonic radiator lengths from 160-6m and then graphed the points so you could visually "find" gaps in the graph to pick your antenna length. It even had an option for dipoles and loops. -Tim -Original Message- From: flexradio-boun...@flex-radio.biz [mailto:flexradio- boun...@flex-radio.biz] On Behalf Of David McClain Sent: Tuesday, December 21, 2010 1:50 PM To: flex list Subject: [Flexradio] Questions about antennas and spectral displays... I think I know the answer, but I want to double check... That 43 foot antenna... is it an end-fed random wire off the back of the tuner? or a center-fed dipole, with total span 43 feet? Also, as a side question... Anyone happen to know why all the other higher-end modern rigs (Icom, Kenwood, Yaesu, Ten-Tec), use swept spectrum analyzers instead of using an FFT like Flex Radio uses? - de Dave, N7AIG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Questions about antennas and spectral displays...
Nice explanation... but these other high-end radios all sport DSP's too. So I suspect the answer has more to do with the memory required to hold N-samples for an N-point FFT. All DSP's are quite capable of doing FFT's, so there seems no other excuse. - de Dave, N7AIG On Dec 22, 2010, at 09:49, Ray - K9DUR wrote: Spectrum analyzers display signals in the frequency domain. There are 2 basic methods of doing this: hardware & software. Traditional swept-frequency spectrum analyzers take a brute-force hardware approach to doing the conversion from time domain to frequency domain. They "tune" a detector across a specified frequency range, time-sample the results, & display the resulting output on a standard time-domain oscilloscope-type display. A French mathematician named Joseph Fourier, working in the late 18th & early 19th centuries, developed a series of mathematical functions which would do the conversion mathematically. The modern-day Fast Fourier Transform (FFT) is the result. So, to answer the original question, the other high-end radios use swept spectrum analyzers because they are hardware-based radios. PowerSDR uses FFT because it is a software-based radio. 73, Ray, K9DUR http://k9dur.info ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Questions about antennas and spectral displays...
Hi Neal, You might be correct - at least for some of the radios. But my Icom 756P3 shows traits that lead me to believe that it really is a swept analysis too. I have a MFJ-212 pulsed noise bridge for tuning up the rigs, and the pattern seen on the 756 scope indicates that it is sweeping. Otherwise, I'd expect to see less fast-chop patterns and just a more broadened spectral display. But when I look at radios like the higher-end Icoms (7600 7700 7800 R-9500), the TenTec Omni and Orion, and others, I am quite dismayed. Flex has spoiled me in what to expect of a panadapter and waterfall display. [ I have a mental tug to play with a high-end Icom, but I know in the back of my mind that the sexy display will lose its charm rapidly, and I would sink into despair with a radio that isn't really much better or different than the 756. Flex has raised the bar too high. ] - de Dave, N7AIG On Dec 22, 2010, at 12:05, Neal Campbell wrote: Dave I always assumed they were internally using FFT but the 'peripheral' interface was a sweep because of the speed of the transport between the radio and the display. Or the power of the dsp chip itself. Neal Campbell Abroham Neal Software www.abrohamnealsoftware.com (540) 645 5394 On Wed, Dec 22, 2010 at 1:58 PM, David McClain audiometrics.com> wrote: Nice explanation... but these other high-end radios all sport DSP's too. So I suspect the answer has more to do with the memory required to hold N-samples for an N-point FFT. All DSP's are quite capable of doing FFT's, so there seems no other excuse. - de Dave, N7AIG On Dec 22, 2010, at 09:49, Ray - K9DUR wrote: Spectrum analyzers display signals in the frequency domain. There are 2 basic methods of doing this: hardware & software. Traditional swept-frequency spectrum analyzers take a brute-force hardware approach to doing the conversion from time domain to frequency domain. They "tune" a detector across a specified frequency range, time-sample the results, & display the resulting output on a standard time-domain oscilloscope-type display. A French mathematician named Joseph Fourier, working in the late 18th & early 19th centuries, developed a series of mathematical functions which would do the conversion mathematically. The modern-day Fast Fourier Transform (FFT) is the result. So, to answer the original question, the other high-end radios use swept spectrum analyzers because they are hardware-based radios. PowerSDR uses FFT because it is a software-based radio. 73, Ray, K9DUR http://k9dur.info ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Questions about antennas and spectral displays...
I appreciate everyone giving such thoughtful answers about the spectrum analyzers on modern high-end rigs. I have never built a swept analyzer, but I have built hundreds, perhaps thousands, of FFT- based analyzers. I thought perhaps I was missing some important information. But it doesn't appear to be the case. Right now only the various SDR's, the Flex Radios, and the K3 / P3, seem like truly modern rigs. - de Dave, N7AIG On Dec 22, 2010, at 12:27, Brian Lloyd wrote: On Wed, Dec 22, 2010 at 11:22 AM, David McClain audiometrics.com> wrote: Hi Neal, You might be correct - at least for some of the radios. But my Icom 756P3 shows traits that lead me to believe that it really is a swept analysis too. I have a MFJ-212 pulsed noise bridge for tuning up the rigs, and the pattern seen on the 756 scope indicates that it is sweeping. Otherwise, I'd expect to see less fast-chop patterns and just a more broadened spectral display. I agree with David. In my experience with Icom radios the spectrum display comes as a result of the microcontroller stepping the VFO, sampling the S-meter value, and plotting as a bar-graph on an LCD. It is easy to do even in all-analog radios. That is certainly how it is done in the IC-706 variants. But when I look at radios like the higher-end Icoms (7600 7700 7800 R-9500), the TenTec Omni and Orion, and others, I am quite dismayed. Flex has spoiled me in what to expect of a panadapter and waterfall display. Yes. I have complaints with features in the Flex radios but when it comes right down to receiving signals, it is far and away the radio to beat. -- Brian Lloyd, WB6RQN/J79BPL 3191 Western Dr. Cameron Park, CA 95682 br...@lloyd.com +1.767.617.1365 (Dominica) +1.931.492.6776 (USA) (+1.931.4.WB6RQN) ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
[Flexradio] Flex-1500 at full power?
Up to now I have been using the Flex-1500 to do WSPR at amazingly low power levels. But now I'd like to run some PK31, RTTY, Olivia, etc. Is there any problem running the 1500 at full power in these modes? Or should I restrict it to no more than half-power? It doesn't appear to have a temperature monitoring pane like the Flex-3000. - de Dave, N7AIG ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] Flex-1500 at full power?
Good point on overdriving the digital modes. But my question really pertained to whether or not running at near-max power levels would be too hard on the little guy...? - de Dave, N7AIG On Jan 4, 2011, at 00:48, Brian Lloyd wrote: On Mon, Jan 3, 2011 at 11:18 PM, David McClain audiometrics.com> wrote: Up to now I have been using the Flex-1500 to do WSPR at amazingly low power levels. But now I'd like to run some PK31, RTTY, Olivia, etc. Is there any problem running the 1500 at full power in these modes? Or should I restrict it to no more than half-power? It doesn't appear to have a temperature monitoring pane like the Flex-3000. Remember that PSK31 has an amplitude component. Do not run the 1500 at full power on PSK or you will end up with a broad signal. I normally set TX Gain in VAC to -2dB to ensure that there is headroom in the PA. -- Brian Lloyd, WB6RQN/J79BPL 3191 Western Dr. Cameron Park, CA 95682 br...@lloyd.com +1.767.617.1365 (Dominica) +1.931.492.6776 (USA) (+1.931.4.WB6RQN) ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] OT: Bigfoot NIC?
umm maybe your timing statements are off by a factor of 10^3 ?? I see performance on the BigFoot site listed in milliseconds, not microseconds... - de Dave, N7AIG On Jan 22, 2011, at 10:40, Neal Campbell wrote: Hi guys The one thing almost everyone who buys a computer from me is interested in is remote control of their new Flex radio! I honestly would guess 75% or more are planning to do this. Subscribing, as I do, to a lot of computer builder magazines, I read recently about an ethernet card called BigFoot that supposedly reduces the latency of lan-directed transmissions from 10us+ to 0.3us! Supposedly, the embedded NICs in motherboards are as problematic as the 1394 devices (usually cheap chipsets from Realtec, etc). Just wondered if any of our guys who are remote controlling their rig have tried one of these cards and whether they saw any difference! 73 Neal Campbell Abroham Neal Software www.abrohamnealsoftware.com (540) 645 5394 Skype: neal.ansoftware ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] OT: Bigfoot NIC?
Okay... well, I'm keeping an open mind on this subject. But offhand, I would expect that the stack latency is only a small part of the overall responsiveness problem. You still need the application program to wake up and respond. Task switching in Windows is not lightning fast. The process granularity is around 10 ms. Interrupt handlers can be much faster, but that only handles the I/O and buffers it up for the application to respond to. It will be interesting to hear some real end-end measurements. I don't trust marketing hype. - de Dave, N7AIG On Jan 22, 2011, at 17:20, Neal Campbell wrote: The latency its addressing is inside the computer! I read they are bypassing the Windows internet stack(?) and have the equivalent of a Xbox360, processing-wise, handling the I/O. 73 Neal Campbell Abroham Neal Software www.abrohamnealsoftware.com (540) 645 5394 Skype: neal.ansoftware On Sat, Jan 22, 2011 at 6:46 PM, Brian Lloyd wb6...@lloyd.com> wrote: On Sat, Jan 22, 2011 at 10:20 AM, Neal Campbell wrote: Okay, but still the question stands, is anyone doing this and have they seen any reduction, for instance, in skype latencies for audio? You wouldn't. The latency on the local LAN is far less than the latency and latency jitter in the Internet in general. -- Brian Lloyd, WB6RQN/J79BPL 3191 Western Dr. Cameron Park, CA 95682 br...@lloyd.com +1.767.617.1365 (Dominica) +1.931.492.6776 (USA) (+1.931.4.WB6RQN) ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] OT: Bigfoot NIC?
I would add... in situations where bulk throughput needs to be maximized, an entirely different OS is normally needed. See the Project X-Kernel by Prof. Larry Peterson. (this has nothing to do with X-Windows from MIT). The difficulty being addressed in the X-Kernel has to do with data packets, and sub-packets, being routed up and down the stack. X- Kernel maps a common segment of physical memory across multiple process address spaces so that data copying could be minimized. To the best of my knowledge, MS Windows would be wholly unsuitable for an X-Kernel. But perhaps I'm mistaken about this. - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Jan 22, 2011, at 18:09, David McClain wrote: Okay... well, I'm keeping an open mind on this subject. But offhand, I would expect that the stack latency is only a small part of the overall responsiveness problem. You still need the application program to wake up and respond. Task switching in Windows is not lightning fast. The process granularity is around 10 ms. Interrupt handlers can be much faster, but that only handles the I/O and buffers it up for the application to respond to. It will be interesting to hear some real end-end measurements. I don't trust marketing hype. - de Dave, N7AIG On Jan 22, 2011, at 17:20, Neal Campbell wrote: The latency its addressing is inside the computer! I read they are bypassing the Windows internet stack(?) and have the equivalent of a Xbox360, processing-wise, handling the I/O. 73 Neal Campbell Abroham Neal Software www.abrohamnealsoftware.com (540) 645 5394 Skype: neal.ansoftware On Sat, Jan 22, 2011 at 6:46 PM, Brian Lloyd wb6...@lloyd.com> wrote: On Sat, Jan 22, 2011 at 10:20 AM, Neal Campbell wrote: Okay, but still the question stands, is anyone doing this and have they seen any reduction, for instance, in skype latencies for audio? You wouldn't. The latency on the local LAN is far less than the latency and latency jitter in the Internet in general. -- Brian Lloyd, WB6RQN/J79BPL 3191 Western Dr. Cameron Park, CA 95682 br...@lloyd.com +1.767.617.1365 (Dominica) +1.931.492.6776 (USA) (+1.931.4.WB6RQN) ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] OT: Bigfoot NIC?
I'll also add several other thoughts on this topic: 1. The BigFoot marketeers have made a very careful re-definition of latency in their claims. To me, an end user, latency measures the delay for a response to a request. That includes the portion they call latency, which pertains only to the TCP/IP stack. But it also includes everything else it takes to formulate a response and send it back to me. 2. The BigFoot marketeers chose a particularly lame example -- ping -- in which, perhaps, most of the latency is in the TCP/IP stack, since the application ping is so thin. For all we know they implemented their own ping which is operating at high priority just sitting there waiting to respond at the drop of a hat. 3. Total speedup is highly dependent upon the time taken in all parts of the responder. If you decreased the stack processing time to zero, and the application portion still takes 1 second to respond, you have achieved essentially zero performance improvement. That's why it looks especially good for a thin client like ping -- which itself takes almost zero time for its processing. [See Ahmdal's Law http:// en.wikipedia.org/wiki/Amdahl's_law ] - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Jan 22, 2011, at 18:14, David McClain wrote: I would add... in situations where bulk throughput needs to be maximized, an entirely different OS is normally needed. See the Project X-Kernel by Prof. Larry Peterson. (this has nothing to do with X-Windows from MIT). The difficulty being addressed in the X-Kernel has to do with data packets, and sub-packets, being routed up and down the stack. X- Kernel maps a common segment of physical memory across multiple process address spaces so that data copying could be minimized. To the best of my knowledge, MS Windows would be wholly unsuitable for an X-Kernel. But perhaps I'm mistaken about this. - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Jan 22, 2011, at 18:09, David McClain wrote: Okay... well, I'm keeping an open mind on this subject. But offhand, I would expect that the stack latency is only a small part of the overall responsiveness problem. You still need the application program to wake up and respond. Task switching in Windows is not lightning fast. The process granularity is around 10 ms. Interrupt handlers can be much faster, but that only handles the I/O and buffers it up for the application to respond to. It will be interesting to hear some real end-end measurements. I don't trust marketing hype. - de Dave, N7AIG On Jan 22, 2011, at 17:20, Neal Campbell wrote: The latency its addressing is inside the computer! I read they are bypassing the Windows internet stack(?) and have the equivalent of a Xbox360, processing-wise, handling the I/O. 73 Neal Campbell Abroham Neal Software www.abrohamnealsoftware.com (540) 645 5394 Skype: neal.ansoftware On Sat, Jan 22, 2011 at 6:46 PM, Brian Lloyd wb6...@lloyd.com> wrote: On Sat, Jan 22, 2011 at 10:20 AM, Neal Campbell wrote: Okay, but still the question stands, is anyone doing this and have they seen any reduction, for instance, in skype latencies for audio? You wouldn't. The latency on the local LAN is far less than the latency and latency jitter in the Internet in general. -- Brian Lloyd, WB6RQN/J79BPL 3191 Western Dr. Cameron Park, CA 95682 br...@lloyd.com +1.767.617.1365 (Dominica) +1.931.492.6776 (USA) (+1.931.4.WB6RQN) ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http:// www.flexradio.com/ ___ FlexRadio Systems Mailing List FlexRadio@flex-radio.biz http://mail.flex-radio.biz/mailman/listinfo/flexradio_flex-radio.biz Archives: http://www.mail-archive.com/flexradio%40flex-radio.biz/ Knowledge Base: http://kc.flexradio.com/ Homepage: http://www.flexradio.com/
Re: [Flexradio] OT: Bigfoot NIC?
HI Neal, I'm not a gamer either. But if I were, and I were programming such a system, I would probably seek to implement a form of distributed shared-memory, which holds the game state at every participant's node. In that case, I would pound on the code to make it as thin as I possibly could, and then UDP stack latency would become my bottleneck. But this is a very specialized application, tuned to maximum performance for a particular audience. I suspect that most Ham's don't program in such highly tuned programming languages (e.g., Assembly, at interrupt or kernel level). From what I witness in typical Ham software, they range from VBasic to Object Pascal (Delphi) to C or C++ (sometimes). It sounds from your comments that VOIP is a main consideration. In that case, and with remote sites being connected by the Internet, isn't Internet routing the main factor here? Completely beyond your control? - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Jan 22, 2011, at 18:32, Neal Campbell wrote: Since the premise for this card is multi-player gaming (and not being a gamer at all), I wonder what the nature of gaming communication is? Is it just voice communications or is it more complex? Ping is a particularly poor test of anything except connectivity and its questionable why it would be any meaningful test of improvement! Thanks for your insight Dave! Neal Campbell Abroham Neal Software www.abrohamnealsoftware.com (540) 645 5394 Skype: neal.ansoftware On Sat, Jan 22, 2011 at 8:27 PM, David McClain audiometrics.com> wrote: I'll also add several other thoughts on this topic: 1. The BigFoot marketeers have made a very careful re-definition of latency in their claims. To me, an end user, latency measures the delay for a response to a request. That includes the portion they call latency, which pertains only to the TCP/IP stack. But it also includes everything else it takes to formulate a response and send it back to me. 2. The BigFoot marketeers chose a particularly lame example -- ping -- in which, perhaps, most of the latency is in the TCP/IP stack, since the application ping is so thin. For all we know they implemented their own ping which is operating at high priority just sitting there waiting to respond at the drop of a hat. 3. Total speedup is highly dependent upon the time taken in all parts of the responder. If you decreased the stack processing time to zero, and the application portion still takes 1 second to respond, you have achieved essentially zero performance improvement. That's why it looks especially good for a thin client like ping -- which itself takes almost zero time for its processing. [See Ahmdal's Law http://en.wikipedia.org/wiki/ Amdahl's_law ] - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Jan 22, 2011, at 18:14, David McClain wrote: I would add... in situations where bulk throughput needs to be maximized, an entirely different OS is normally needed. See the Project X-Kernel by Prof. Larry Peterson. (this has nothing to do with X-Windows from MIT). The difficulty being addressed in the X-Kernel has to do with data packets, and sub-packets, being routed up and down the stack. X- Kernel maps a common segment of physical memory across multiple process address spaces so that data copying could be minimized. To the best of my knowledge, MS Windows would be wholly unsuitable for an X-Kernel. But perhaps I'm mistaken about this. - de Dave, N7AIG Dr. David McClain Chief Technical Officer Refined Audiometrics Laboratory 4391 N. Camino Ferreo Tucson, AZ 85750 email: d...@refined-audiometrics.com phone: 1.520.390.3995 web: http://refined-audiometrics.com On Jan 22, 2011, at 18:09, David McClain wrote: Okay... well, I'm keeping an open mind on this subject. But offhand, I would expect that the stack latency is only a small part of the overall responsiveness problem. You still need the application program to wake up and respond. Task switching in Windows is not lightning fast. The process granularity is around 10 ms. Interrupt handlers can be much faster, but that only handles the I/O and buffers it up for the application to respond to. It will be interesting to hear some real end-end measurements. I don't trust marketing hype. - de Dave, N7AIG On Jan 22, 2011, at 17:20, Neal Campbell wrote: The latency its addressing is inside the computer! I read they are bypassing the Windows internet stack(?) and have the equivalent of a Xbox360, pr
