Home › Forums › Breakaway Professional Products – [discontinued] › The single most challenging tech feat a RS can deploy: SFN!
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September 12, 2009 at 10:49 pm #475AnonymousGuest
Greece has some issues regarding radio stations operating within the country’s territory.
First, they are not licensed. True, no one is! All licenses have either expired and never renewed since 1990, or issued on 2001, meant to last until 2005, never newed since.
"But then, isn’t it illegal to operate a broadcast radio station in Greece?" one would ask. Nope! not if you were operaating illegaly at a specific date! the November 11th, 1999, to be precise(!).
That is because no political party ever came to be goverment dared to bear the political cost to actually issue a proper number of licenses or auction them under terms like financial prognosis or technical and content completeness.
The long prologue is to show that here in Greece, owning a radio station isn’t exactly the revenue creating business one could expect it to be, and investments on the area won’t ever be funded either by any bank loan (as banks properly require the business to be licensed to operate on whatever field they would, ie a food processing business needs to have a health and foods dept. license and HACCP european food processing standard certificate) or by any serious investor who targets at profit.
The cherry atop the pie, is that radio stations are obliged by law to use a single frequency (the one used back to the date mentioned earlier) , and to cover just their home prefecture.
But Greece has a unique terrain morphology that includes several thousand feet high mountains, coastal cities, towns, villages and tourist resorts.
Atop the highest of the mountains, which a prefecture could be easily be covered with one transmitter,
-either the presence of ancient civilisation ruins makes the installation of antennas an act punished like a capital crime(!) (thank god CP is forbidden by constitution in GR lol)
-or, in the rare case that no ruins exist, power companies won’t install PL where a paved road doesn’t exist.So, radio stations need to use cliffs, rather than mountains, to cover the prefecture, but this is impossible due to the mountains between the cliffs(!). That leaves enormous gaps in coverage, but the only way to use a repeater is to use the same frequency on both sites.
In most cases, a severe percent of the coverage of the two sites overlap. This calls for extreme measures, like the ones used in digital television for reuse of the frequency. No matter how extreme this seems to look like, it is the only way a radio station here could cover the whole prefecture without using a pair or triad of frequencies.
Well, the technical issues regarding the problem are two-dimensional.
-First, you need to sync all transmitters down to cycle. This is easily achieved with unmodulated carrier, by using external PLL references synced to a reference bound to a GPS clock. Many new exciters have the relevant input, so a pair of GPS 10MHz reference clock generators is all that is needed. They are cheap, prices starting about 130$. The antennas are the standard car-type GPS antennas (they need just 3 satellites to extract the clock, anyway.)
-Second and hardest, is the aural sync. As I said before, the cost needs to be minimal. That by default excludes all AES-EBU digital STL’s and exciters, because their price range simply cannot be afforded by radio stations with maximum investment target 300.000$. It is half the whole invested value on STL’s and exciters. (What about offices, building and such?!?). We need to work with what we already have.
Of course, I understand that calibrating of spectral response curves should be incredibly accurate, precorrection would be needed in both sites, and extreme accuracy of analog levels should be achieved. But let’s skip this part for now (far easier compared to what follows)
I made some thoughts regarding WHAT should be used to achieve the delay needed (around 166 uS for a trip from main transmitter to repeater and half the way back, assuming a delay of a few uS from STL antenna to FM antenna (actually several phase shifts between the amplification circuits), and given a distance of 50km (~30miles) between the main and the secondary site. (75km the trip of secondary signal, 25km the primary signal which is already under way, 50km target delay [(1/300000)X50]plus the transmitter’s delay. The amount of the delay equals half a period of a sine wave near 6 KHz, which explains why bass usually escapes the delay and the phase difference but mid and highs sound muffled and thudded. When this comes to multiplex, you can not imagine the mess, you should hear it with your own ears.
Well, for the academic part, I made some thoughts of how to eliminate this part of the problem. An adjustable IF delay line could be made, and installed on a second identical discriminator stage inside the STL receiver. The delayed signal would then feed the primary transmitter, when the originally-timed signal would carry on to the secondary transmitter. But it can’t be done – SAW delay lines should be made application – specific, as noone builds delay lines @ 10,7MHz. SAW delay lines on order are usually fixed timed, and come in packs of 1000’s – bounces the cost sky-high.
Variable delay lines exist, but they are also frequency – specific, and maximum delay obtainable is in the range of 1000’s of nS – too little delay. We need to make a whole cluster of those to achieve the req’d delay.
Can we do it in software? If the software isn’t the “windows kernel mixer” of DX9, sure. But we need an enormous sampling rate and sample – accurate adjustable FIFO buffering to do that. A regular PC with the interrupts cannot do it – or CAN IT?
This is today’s question for you, Leif, the lord of audio DSP. Can one by any means set up a PC with whatever needed OS in it to run one card in loop or two cards in 384 or higher sampling rate with sample-wise adjustable FIFO buffering lenght?
I know that I can connect an ADC-DAC 768KHz pair through a serial memory and adjust the pick-up point for throughput for $20 parts and make big $$$ with it, but the question is, can it be done in software and save those poor guys of the development of PCB, the minimum order requirements, the P&P and the MTBF of such a device?
September 13, 2009 at 8:01 am #8304LeifKeymasterInteresting story!
Regular BBP will not work. Due to the buffering and the asynchronous SRCs used, latency will never be predictable enough.
BBP ASIO in low latency mode could work, though. In this mode it has no extraneous buffering, and all processing is done in the ASIO thread, so it’s possible that the audio from two PCs with identical ASIO sound cards, fed from the exact same audio source, would come out at exactly the same time. I haven’t tried this though! That’d be the first thing to test.
Sub-sample adjustable delay is easy! If that’s the only thing standing in the way, I could definitely implement it. Let’s make sure it’s possible to even get predictable delay through a PC first, though.
///Leif
September 13, 2009 at 9:06 am #8305AnonymousGuestWell, I’ll start experimenting on a few setups that I have, and see how the on-card buffers work. All the latency times I have ever read about asio are an order of magnitude higher, but my o’scope might have a different story to tell. I guess I’ll be back 😉
maybe some homework on the actual onboard chipsets (kinda rare to find, sadly, when it comes to sound cards) and on ASIO as well, would pop up with interesting things. I ‘ll go reading.
September 13, 2009 at 9:20 am #8306LeifKeymasterYes, the latency times are much longer than the delay times, but we don’t care how long the latency is as long as it’s the same at all transmitters (as long as it’s reasonable).
What we care about is consistency.
For example, if a box sometimes has 2ms of delay, and sometimes 4ms of delay (randomly selected on boot-up probably), then that would be really bad.
On the other hand, if a box has 17.000 ms of delay every time it boots up, all the time, then it would be possible to use this for SFN.
///Leif
September 13, 2009 at 9:42 am #8307rocco123456MemberHello… very interesing story….
In Italy…we have "ISORADIO"… 103.3(mono)…that is government radio..for traffic information in highway road….
All along highway is there an optical fiber to syncronize all transmitters……A lot a years ago..there was somebody that make this:
(example)
tx = 50mhz
tx1(at 25 km from tx)>>>>rx on 50mhz>>> duplicator freq>>> amplifier agc on 100mhz
tx2(at 25 km from tx)>>>>rx on 50mhz>>> duplicator freq>>> amplifier agc on 100mhzSo it is all in phase (more or less)…….but all this is illegal now…and honever it is empiric.. but it goes….
Now in Digital TElevision to make sfn is used gps reference… but it is important having MIP inserter at headhend and respect max distances between transmitter….
What low cost gps 10 mhz reference.. do you use?
thank you ..
Rocco..September 13, 2009 at 3:35 pm #8308camcloneMemberHey ..SV9OFO …are you from Crete Island ??? Giannis is your name? i..guess..
I made a much cheaper …project for having a clean audio when switching from one antenna to another from heraklion city to moires city or tympaki area..
The second transmitter on Axendria mountain… broadcasts with modulation 45 , 50 Khz , and the main transmitter at Vassilikos mountain broadcasts at 75 Khz.
with that difference i achieve a much much affortable distortion when the car driver goes to a place where there is equal signal from both transmitters!of course..this is very very shitty way…
i wish you could make a low cost project for clean audio on areas where there is a signal from both transmitters..This is ry@thm@oS 1o4,5 radio station from heraklion city – crete island ,
stavros christopoulos (steve )p.s.1 : the radio stations in athens are much tech freaks than..us ..here in crete island, but they don’t share information on forums.. unfortunately.
p.s.2
Leif, i am waiting for your hardware version of breakaway platform..with mpx clipping.. !! come on come on!!September 13, 2009 at 4:50 pm #8309AnonymousGuestRocco, the mod on 50MHz for one ensures "some" frequency phase lock, but the aural part remains delayed and muffed.
also, the 50mhz band is allocated to amateurs and such an option isn’t availabe by any means.even feeding with IF the STL links in-phase and the exciters on with upconversion, rf phase accuracy won’t be preserved, and of course the aural part is lost.
Target is to achieve aural accuracy to the geo point where the two fields overlap with the SAME field strenght. To keep the aural phase throughout the whole area is impossible due to multipath and different signal travel times (but a very narrow zone with a width smaller than 1km, in which only rf phase correlations occur leading to bumping signal strenght every 3 meters(!) with bumps up-down near 20db, but the audio remains clear in that zone and IF AGC does the rest in the receiver (given it is fast enough) for pedestrian reception. For mobile reception, this would be a pain, like constant short – distance multipath "vibrato", but stationary receiver operators would skip the frustration of having to strive to find a spot with noiseless reception – a small tilt of the antenna could lead it from the null to the peak.
The rest overlapping region will have "tolerable" reception, but not the huge interference. Loss of RDS lock and stereo separation would occur, noise levels would climb up but it would be far better than the current state of the network.
Checking on ebay popped up several new GPS reference 10mhz units, starting from $150. Haven’t bought one yet.
looking for solution on aural delay problem before investing on phase lock in the RF part.here is the link
http://cgi.ebay.it/ws/eBayISAPI.dll?Vie … 0297293656
___________________________________________________________________________Well, Yes dear friend Stayro, it’s me, your RF engineer.(!) 😆
and it is your radio station I was thinking about, along with another "corporate asset" radio station on 97,2 FM.
As I have explained in vivo to you, mismatcing the sound levels end up having excessive noise demodulating the undermodulated carrier due to sidebands slipping under and above the +20db local carrier, and leaving some noise underneath, as receivers filters woould still pass the whole spectrum to the slope discriminator. Meanwhile, on the domain of the normally modulated carrier, demodulation ends up having a "chirp" under the useful modulation content, that tends to "cancel" the modulation depth especially when (depending on distance) aural phases are opposite. The correlation creates new frequencies that sound like "chirping" due to harmonic distortion of the bass, the maximum energy modulation frequency.
- Regarding the design of the hardware device of BBP, I need to say to you that it is not "build it and sell it". Many potential customers do not have your expertise on uploading firmware to devices. ANY device that implements analog and digital domain components, as the BBP with stereo implementation and MPX clipping, requires prototyping, modifications of the prototype, software and hardware debugging, testing and then more testing, in harsh situations like low-Z STL inputs (like 100 ohm or less), contacting factories for estimating price tagging, quality of components vs pricing balancing, and several other aspects. It is NOT that easy. It takes time for good things to come.
for example, you need to find an interface capable of loading easily firmware to device. Could be RS232 (kinda rare nowadays, on new desktop PC’s, not an option at all for a laptop sigh),SD, XD, Memory stick, or USB flash. That is because in case something goes wrong, you need to have a bootloader to upload firmware. USB B connector wouldn’t work because it requires a higher software layer driver, and JTAG is not always the best option customer-wise. That bootloader should include loading and transferring from external flash memory, and which means FAT reading capability on the bootloader. This is NOT an easy task to accomplish and implement. And it is just 1/100 of the whole project.
You need to port the software to the hardware platform. This means a great amount of reading on how the hardware works. some deep reading on FPGA’s and DSP chips already available. Re-sort your code to fit in those, drop the math to the DSP,meet several problems on the way when instructions available under CISC i386 are unavailable in RISC,circumvent those by completely rewriting whole parts of your code, get a decent CPU and manage to make them cooperate.you need to build from scratch human interface, get to how your LCD screen works and refreshes, drive it in real-time but still minimizing CPU usage for the task as the major part goes to processing (and limiting(!) lol) then debug, debug, rewrite part, debug, debug, hey, interrupts collide, fix, hey, this releases CPU time from LCD driver, more agressive processing from spare cycles!, rewrite driver (less cpu for the screen) debug, debug. It NEVER IS "plug&play", as you are so keen on saying,dear friend.
__________________________________________________________________________________Leif, as you have already guessed, I do have some experience, the first ADC project I ever built was a 8bit sampler for my amiga500, back in 1993(!). (and Yes, it included the ADC0804, and yes, it worked 😳 😆 )
- Regarding the design of the hardware device of BBP, I need to say to you that it is not "build it and sell it". Many potential customers do not have your expertise on uploading firmware to devices. ANY device that implements analog and digital domain components, as the BBP with stereo implementation and MPX clipping, requires prototyping, modifications of the prototype, software and hardware debugging, testing and then more testing, in harsh situations like low-Z STL inputs (like 100 ohm or less), contacting factories for estimating price tagging, quality of components vs pricing balancing, and several other aspects. It is NOT that easy. It takes time for good things to come.
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