Jonathan Richards writes: > In what units shall the time index be specified? The sampling rate sets a > resolution limit on the timing, so for 8kHz we only need 1/8000 sec = 125 > microseconds precision, but if we have an ambition for higher rates, we need > more. [1] > In reality, radio comms are not HiFi standard. Does anyone know what the > typical bandwidth is? Or should we simulate by taking a beautiful 22kHz > recording and filtering it to sound like the real thing? Perhaps as an > option, so one can do radio practice with bell-like clarity at first, and > graduate to crackly reception of foreign languages and accents later! > > Regards > Jonathan > > [1] It occurs to me that for chunked formats like WAV, there is a mathematical > relationship between the byte position and the time offset which could be > used for conversion, no? >
Fantastic, a post not talking about rotating planes on their noses ;-)) Time index should be specified in seconds I think. Yes there'll be a lot of decimal points, but it's SI, and it's typically what wavefile editors natively display. In practice it probably doesn't matter if the time resolution spans a few bytes, especially at higher rates, since there's generally a slight gap between words, and if there isn't it's a good candidate for a run-together phrase. I guess comms aren't always that clear in real life, but at least if recordings are made at good quality to start with its optional whether to degrade them or not. At present the 8,8 limitation forces some degradation, and I'm not sure if the sort of 'muffled' quality that it imparts is what we're after. Yes, there is a relationship between byte position and time for wave files, which is how I'm going to convert the current default.vce. I think time is the best way forward though ever since someone mentioned Ogg Vorbis at one point. Plus I think Audacity only displays time, not byte position. As I say, I'm not an audio guy, so I'm open to being persuaded that another way is better... Another thing, some folk running Linux (David Megginson and at least one other) have reported only being able to hear the ATIS extremely faintly in the past. I'm not sure why, but it might be best *not* to copy the volume characteristics of default.wav, but to do what you think is best and see how it goes. Anyone got any wavefile editor recommendations BTW? I used CoolEdit (Windows) for the ATIS, but the trial period is now long gone, and when I went to buy it I found the guy had sold it to Adobe and the price had tripled. No thanks! I'm using Audacity now, but it's not entirely polished, and the sound is not in sync with cursor movement when playing a small selection in recent versions. Crucial factors are the ability to extend or contract a selection by either edge, and easy copy and pasting into a new buffer, plus display of time or byte position. There's loads for Linux, but I haven't found a really likeable one yet, Audacity being the nearest so far. Cheers - Dave _______________________________________________ Flightgear-devel mailing list [EMAIL PROTECTED] http://mail.flightgear.org/mailman/listinfo/flightgear-devel
