Messages in this thread | | | From | Bill Huey <> | Subject | Re: a joint letter on low latency and Linux | Date | Sun, 2 Jul 2000 04:44:26 -0700 (PDT) |
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> Musicians are perfectly accustomed to working with a "raw" mix in > their cans, and hearing the far-superior results in the mixdown, and > this is a natural extension of that process. It also tallies better > with the real process in the studio, with temporary effects used in > the cans, but the final effects chosen during the lengthy mixing > process after all sund has been recorded.
What about interactive playback when a key is responding to velocity ?
That's the core of the issue and certainly isn't the 5ms delay BS you said earlier.
> When it came to playback after the original recording was done, an > asynchronous process or thread could be used to read ahead of the > playback and "render" the raw audio into processed sound buffers which > would play back just as the "actual" playback off disk happened. This > would mean that you could provide nice big buffers for your > third-party plugins, and wouldn't need realtime for them at all.
Audio programmers aren't *dipshits*, of course they know this stuff and neural network noise reduction, adaptive signal processing, complex math, etc...
> The case of making a instrument (eg a syth) or an effects processor > for live use (eg a guitar effects unit) is pretty different from this, > and clearly (in my mind anyway) is a hard realtime project.
The above that you've outlined is the core of the issue. This can be done either using soft or hard realtime factilities. It's realtime [soft/hard] DSP hell and fudge around the problem using typical Unix time sharing concepts misses the point[s] completely.
All the previous stuff about buffering is largely useless to this discussion. Everybody including their little sister[s] know about data buffering N seconds before the audio hardware output.
> Sean Hunter
bill
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