On 2014-04-28, Rob <nom...@example.com> wrote: > Jochen Bern <jochen.b...@linworks.de> wrote: .... >> However, that *does* leave me wondering where the 12 us figure comes >> into play. With the typical distances between 2m and even 70cm >> repeaters, the mobile transceivers will see shifts *far* beyond that >> between different repeaters' signals. > > The figure comes from two different experts in the field. They have > built such systems for the emergency services (police, fire brigade, > ambulance) that have been operating in the country for a few decades > and have been replaced by a digital (TETRA) system in the meantime. > But their knowledge and experience remains valid. > In those days we could not afford such an experiment, they used fixed > analog leased lines to transfer audio with a fixed and known delay, > but today we have internet and GPS and we can achieve the same thing > much easier. > >> FWIW, will you have the audio >> cards output AM (that will then get modulated onto an otherwise >> unsynchronized HF), or do you plan to have the card generate HF directly >> into the PA? > > The existing system mentioned above uses SDR techniques to synthesize > the FM signal directly from the digital samples, but we like to use > existing repeater hardware that already has FM modulation, so we want > to use soundcards to produce analog audio that is fed into the repeaters. > > The HF is not unsynchronized, it is locked to a GPSDO. But I agree > that it would be much more predictable to do it the digital way. Now > we have extra variables like the exact deviation setting and the > characteristics of the modulator. Fortunately all repeaters run the > same hardware (Tait TB8100). > > I am still investigating how to output digital samples to a soundcard > in Linux at exactly determined time (versus just writing sample blocks > to the soundcard driver at a predermined moment. they will still be > buffered after that) > > All in all it is funny to read all the "that cannot be done"-like comments > by several persons on a ntp newsgroup while systems like this have been in > use since the seventies, and in fact have already been build by amateurs > and are in operation today. So I prefer to go by the experience of > the people who built those networks, rather than the armchair experts.
Not, "it cannnot be done", but "it is silly to try". I simply have a really really hard time figuring out why you want to do that. For voice, the max frequency is something like 4KHz (eg telephone quality) which is 250us (125us sampling) . It is really really hard for me to figure out what bad things would happen if that were the resolution you used instead of that 12us. It just seems wildly over-specified. And then you make statements like it is the transmission time that needs to be 12us not the reception time, which also do not make sense. And a phase jump even of 120usec I doubt would be audible, maybe 12 ms would be. (are you sure you did not misunderstand your "experts"?) If it really is 12us I would be really interested in knowing why. I might even learn something. _______________________________________________ questions mailing list questions@lists.ntp.org http://lists.ntp.org/listinfo/questions
