On Sun, Jun 14, 2009 at 3:53 PM, Lennart Poettering<mzynq@email-addr-hidden> wrote:
> On Mon, 15.06.09 00:01, Fons Adriaensen (fons@email-addr-hidden) wrote:
>
>>
>> On Sat, Jun 13, 2009 at 12:16:22PM +0200, Jörn Nettingsmeier wrote:
>>
>> > that said, lennart, any chance you could make it to a LAC anytime soon?
>> > pa has got some heat lately, some deserved imho, some not, but it
>> > definitely serves a very palpable demand.
>> > i believe that after some time it can be a real blessing for casual
>> > users without necessary having to become a PITA for pro audio users -
>> > but for that, it would be nice to discuss the mutual needs and wishes
>> > over pizza. [2]
>>
>> I agree. I'm not a PA user - I dislike 'desktop sounds' and any
>> serious app I expect to use Jack. But if PA could provide an
>> interface that would allow Jack (or other apps) to connect to it
>> *without any loss of performamce* compared to direct access of
>> an Alsa hw: device, [1] then I wouldn't mind having it installed
>> and being used.
>
> The part about "without any loss of perfomance" is the big problem. Of
> course the latency increases each time you add an indirection. There
> is no way around this. A corollary of that is that PA and JACK need to
> have the same "distance" from the low-level device. The only way I see
> how that could work in an intregrated solution is if JACK and PA would
> merge in some way. While that might be a nice idea (which I certainly
> don't oppose) I don't see that coming in any way. It's simply that the
> playback models of PA and JACK are simply too different. PA uses very
> large hw buffers (2s), rewinding, timer-based audio scheduling (which
> allows us to adjust the wake up intervals dynamically to what the
> clients request), zero-copy IO, non-float sample types, and so
> on. That's all very useful for normal desktop use and for minimizing
> wakeups/power usage/CPU load. OTOH JACK's primary objective is low
> latencies, and power usage or non-floats, ... don't matter at
> all. Which effectively means sound card IRQ based scheduling and short
> buffers make much more sense. A hybrid engine that could do all of
> this and switch between timer-based scheduling and sound card IRQ
> based scheduling in one would probably be exceptionally complex, and I
> see noone who would be willing and have the time to work on that. [1]
>
> The only other way I see to give PA and JACK the same "distance" from
> the device is if PA goes out of the way when JACK runs. Which is
> basically what JACK2 and PA do now. It's a however only a pragmatic
> compromise, not more.
>
> Not many people work full-time on audio for Linux. Given that this is
> how it is pragmatic comprimises are probably what we have to work
> with.
>
>> Pizza ? In Holland ? More likely a 'broodje kroket'.
>
> So I read from that that LAC 2010 happens in .nl?
>
> Lennart
>
> [1] Canonical appears to be looking for someone to hack on PA full
> time. So if someone wants to attack this issue, he might
> even get some funding for it if he does so under the PA umbrella.
>
> http://webapps.ubuntu.com/employment/canonical_DASE/
>
> --
> Lennart Poettering Red Hat, Inc.
> lennart [at] poettering [dot] net
> http://0pointer.net/lennart/ GnuPG 0x1A015CC4
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>
What about piggybacking pulseaudio on jack as default behavior if jack
is running (without autoconnect to ports, preferably). You can figure
that if someone has jack running they can handle doing a manual port
connect, and with the kind of buffer sizes pulse likes, it would
hardly be hampered by a jack backend, right?
For a while I was using the pulseaudio jack-sink module as the default
pulse output, and it worked nicely, then it just stopped working (this
is debian lenny, PA version 0.9.10-3 with the goto10 and
debian-multimedia.org lenny repositories). Pulse just gives a "Module
load failed." when I try to load module-jack-sink in pacmd).
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Received on Mon Jun 15 12:15:02 2009
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