Subject: Re: [linux-audio-dev] [OT] Digital mixing console and soundcards
From: David Olofson (david_AT_gardena.net)
Date: Wed Mar 13 2002 - 08:39:49 EET
On Monday 11 March 2002 22.05, Remco Poelstra wrote:
[...]
> > Good news:
> > Good news 2:
>
> Ok, I'll use a multichannel card of course, might be more than one
> in the future.
> If I can afford 24 bit/96 kHz I'll buy it.
> But how do I know what cards uses a single dma stream for all
> channels?
Well, getting full specifications + programming docs helps a lot, but
is hard to get without finding the right persons, signing NDAs and so
an.
Either way, the first place I'd look is the ALSA drivers. Not only to
find out what kind of design we're dealing with, but also to make
*real* sure there actually is driver source code - because you might
need to do some minor hacks in it to make your RTL/RTAI "IRQ sync
PLL" hook work reliably.
> > As to latencies and not using an OS - indeed, there's no reason
> > not to use an OS. Taking it to the extrem, RTLinux and RTAI
> > commonly deliver worst case latencies below 10 µs on P-III and
> > Celeron systems, and even better on PPC systems. (PPC has much
> > better IRQ handling than x86 - mostly because it's not dragging
> > around that badly designed legacy garbage that is the "PC
> > chipset".)
> >
> > If you need to cut latency below that point, please tell me where
> > you found your AD/DA chips! ;-)
>
> If RTLinux is even faster than the converters, I'll stick to linux
> and use ALSA. One problem what of the many LL patches should I use?
The only version I've seen *really* good results with was the good
old 2.2.10-lowlatency - but I'm not up to date with the latest
developments.
I intend to try various new kernels on various hardware soon, as I'll
need two different solutions for work as well as personal projects.
I'll write an article for the RTLF about my findings, of course.
> > So get a real OS, and hack away - no hairy cross-compiling setups
> > need! :-)
>
> Of course if I'm going to use linux, I'll start development with my
> current PC and soundcard (SB Live). I suppose ecasound and ardour
> and such already try to get latencies down to the limit, so at the
> time I'm going to buy a multi channel card, I don't have to change
> the software to get the most out of the card?
Most probably not. Any properly written real time audio applications
should be able to handle "dedicated h/w class" latencies (ie below
this magic 3 ms number) with a good lowlatency kernel.
//David
.- M A I A -------------------------------------------------.
| Multimedia Application Integration Architecture |
| A Free/Open Source Plugin API for Professional Multimedia |
`----------------------> http://www.linuxaudiodev.com/maia -'
.- David Olofson -------------------------------------------.
| Audio Hacker - Open Source Advocate - Singer - Songwriter |
`-------------------------------------> http://olofson.net -'
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