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"Just around the corner" is already here (sort of) in terms of latency: the Layla card offers latencies well under 10ms, and a bunch of folks are doing development for it. On an SGI running MAX/FTS (or other stuff), latencies can get as low as 2ms (but who wants to carry around an SGI? Well, actually, I probably would if I had one). Even with the substantial latencies, I have found the uses of programs like LiSa to be really inspiring. It is also possible to learn how to "play" the latencies by slightly anticipating events. It's not ideal if you want really really tight timing, but given the flexibility, it may be worth it. Having an easily configurable sampling system that offers multiple-voice playback (as many as your CPU can handle: you can get lots more than 30 voices on some of the faster machines), transposition, looping, and signal processing is pretty exciting, and I've learned to work with the latency. But I also still farm stuff out to Jammen sometimes; I may not for long though... Some people of been exploring the musical possibilities of HUGE latencies through internet jams. Last month some folks in NYC jammed with some other folks in Japan where latencies where up around 10 seconds, at least! It's a feature, not a bug... Dan On Thu, 22 Jan 1998, Curtis Bahn wrote: > >At 1:38 PM -0800 1/21/98, Paolo Valladolid wrote: > >>> Curtis Bahn wrote: > >>> > >>> > Also check out MSP, David Zicarelli's signal processing extensions >to MAX > >>> > at http://www.cycling74.com. If you have a powerpc computer you >can > >>>easily > >>> > design custom looping and overdubbing algorithms (amoung many other > >>> > things). It's great ! Programs like MSP and LiSa are taking >looping > >>>away > >>> > from the restrictions of commercial hardware, to a whole new level >of > >>> > personal sound design and performance interaction. > >>> > crb > > >kim flint wrote > >One thing kind of troubling about the computer based systems for >real-time > >use is the latency problems. (meaning the time it would take for audio >to > >be sent in and sent back out again.) On the MSP site some typical >latencies > >are actually listed as: > > > >Audio input to audio output latency on a 9600/300: > > Using the Digidesign Audiomedia III: 46ms > > Using the Sound Manager: 294ms > > > >that's definitely in the range where you would notice it in some > >situations, especially with looping and trying to maintain precise >grooves. > >And this is on a very fast (and expensive) system! I know that PC's >suffer > >from the same problem. The audio has to go through a lot of operating > >system to get to where it's useable, and a lot more operating system to >get > >out again. > > > >So those of you using systems like these for real-time audio I/O, how do > >you deal with that? Are you able to operate it with any timing precision > >for real-time audio events? And I don't mean hard disk recording where >the > >system has opportunities to compensate for the latency. I mean audio >really > >going in and out, like you might have on a typical stand alone audio > >processor. Anyone? > > > > Latency can be a real problem sometimes. Especially on a powerbook, the > technology is not really there yet to have an inexpensive self-contained > system that can deal with all the audio processing and performance I/O in > real-time without some significant lag. > > This is part of the justification of KYMA where you have a dedicated > external audio engine to execute your looping and signal processing > programs. Systems like this and MARS have very powerful programming > environments for the musician to design their own performance algorithms. > More and more, development environments such as these combine complete > flexibility in terms of your sound designs, and easy, intuitive graphical > programming languages. (info on these systems can be found at > http://www.emf.org/sites/software.html.) > > It's just a matter of time until most all of our gadgets will offer some > level of this kind of interface. A significant issue for me is, "what >am I > gonna do with it once I have it." For that reason alone, I spend a lot >of > time on my powerbook hacking MAX/MSP. In performance I get more >realistic > about stuff like looping and either cart a couple of dedicated fast > computers along with me , or trim down the real-time signal processing on > my notebook computer and farm it out to my Echoplex, jam man, Boomerang >or > EH16. > > In some cases, like making rich ambient textures, the latency just >desn't > matter and it is a reasonable price to pay to be in control of more >aspects > of what your signal processing device is doing. We're certainly in a > transition period between hardware and software based models for our >toys. > It's pretty exciting. Especially in performance areas like looping >(which > isn't that "expensive" in terms of computer processing and won't incur as > much latency as more complicated signal processing) this technology is > offering new creative, idiosyncratic opportunities for musicians. It >also > frees us from our traditional relationship to signal processing device > makers where we can often grovel and beg for a simple feature that would > make a world of difference to our music, but won't ever be implemented > generally because it isn't important enough to the market as a whole. > > cheers, > crb > > > > > > > > > > > >-------------------------------------------------------------------------------- > Curtis Bahn > iEAR Studios, DCC 135 > Rensselaer Polytechnic Institute > Troy, New York, > 12180 > office (518) 276-4032 > fax (518) 276-4780 > email crb@rpi.edu > > >