Thank you

Posted Sep 19, 2008 18:43 UTC (Fri) by dmarti (subscriber, #11625)
In reply to: LPC: Linux audio: it's a mess by jwoithe
Parent article: LPC: Linux audio: it's a mess

Thanks for the corrections -- does netjack have a low enough latency to work for recording?

Thank you

Posted Sep 19, 2008 21:43 UTC (Fri) by jebba (guest, #4439) [Link] (7 responses)

"Thanks for the corrections -- does netjack have a low enough latency to work for recording?"

Yes. You can also do things like have a remote machine just do the LADSPA filtering to distribute the load, for example. It works and is solid. I've done similar things with pulseaudio and it felt like I had just duct taped a boat together. Jack rules.

Also, netjack2 is under development with the next generation of jack, which is based on jackdmp.
http://trac.jackaudio.org/wiki/WalkThrough/User/NetJack2

I wish the netgods would all settle on jackd and figure out how to solve its power consumption issues, if at all possible. Use the best! If some of the features are overkill for the desktop, just don't expose them in the simpler GUIs...

-Jeff

Thank you

Posted Sep 20, 2008 7:24 UTC (Sat) by jwoithe (subscriber, #10521) [Link] (4 responses)

"I wish the netgods would all settle on jackd and figure out how to solve its power consumption issues, if at all possible."

Although completely untested, I wonder if simply specifying a very large period size for JACK would go some way to addressing this. At least some of the reason for the high resource usage of JACK is that currently most of its users require low latency and so run with very small buffers. Desktop users don't require low latency, so running large buffers wouldn't matter there. Larger buffers mean fewer wakeups and thus lower power consumption.

JACK is certainly not a replacement for anything else at present, but I do wonder whether with some work it could be used as the basic building block for a universal audio system.

The other thing worth reiterating here is that JACK does much more than just provide access to audio hardware. It is also a signal router allowing audio to be routed arbitarily between applications as well as to/from a physical soundcard. This, along with its low latency performance, is what makes it extremely useful to pro audio and music production users.

Thank you

Posted Sep 20, 2008 7:52 UTC (Sat) by jebba (guest, #4439) [Link] (2 responses)

"I do wonder whether with some work it could be used as the basic building block for a universal audio system."

As an example, I use it the base of http://freeeee.org a distro for eee pc.

Thank you

Posted Sep 21, 2008 23:55 UTC (Sun) by mezcalero (subscriber, #45103) [Link] (1 responses)

Hehe, I like that. Is that a distro optimized for minimal battery lifetime of your Eeepc? Sounds like a great selling point, makes a lot of sense to me ;-)

Lennart

Thank you

Posted Sep 23, 2008 0:46 UTC (Tue) by jebba (guest, #4439) [Link]

"Is that a distro optimized for minimal battery lifetime of your Eeepc?"

There are two FREEEEE modes[1]: full-on media and low power ebook.

With jack, multiple mplayers, freej, firefox, chat, xournal, an offline copy of wikipedia, an image viewer, wifi, camera, etc all running it runs hot, yes. :) But it also runs all that fine without dropouts or crashing. jack performs reliably *now* and is never the weak link. Occasionally I have used pulseaudio as a jack client, which is convenient, but it is no where near as reliable as jack.

jack is clean, "feels" like a nice unix utility (it does its "one thing" extremely well), and has a committed upstream community.

-Jeff

[1]Second mode yet to be completed :)

Thank you

Posted Sep 21, 2008 23:54 UTC (Sun) by mezcalero (subscriber, #45103) [Link]

It's not just about using 2s large buffers. If you use large buffers like that you also need to allow rewriting of them (which means "rewinding" of all filters in the system and so on). Because if you couldn't rewrite them pressing stop or start in your media playback would actually only result in what you hear 2s later.

Allowing rewrite of audio buffers is what JACK doesn't do. And adding this makes things very complicated. I did it for PA, but I simply see no reason for JACK to do it as well. Also, since JACK allows much more complex routing than PA, the rewriting buffer issue becomes equally more complex. i.e. every part that becomes part of a JACK pipeline would need to be able to "rewind" its state. That means every JACK client application would have to be patched non-trivially.

Lennart

Thank you

Posted Sep 21, 2008 23:48 UTC (Sun) by mezcalero (subscriber, #45103) [Link] (1 responses)

PA now uses a "glitch-free" playback model which allows dynamic adjusting of the latency (and thus CPU load) to what the currently connected clients require without any discontinuities in audio. That's why PA is (or "can be" in a good setup) more power friendly than JACK is.

JACK is optimized for lowest latencies, CPU load comes second, and dynamic interruption-free changing of these parameters is not required. For PA however, we want dynamic latencies, generally low CPU load and the parameters need to be flexible.

But even if you would add "glitch-free" to JACK (which makes no real sense), it still is not designed for desktop use. PA carefully makes sure not to convert (or not even touch) audio data if not necessary. OTOH JACK only knows FLOAT samples, and thus audio type conversion is necessary usually at output and often enough on input too. That makes a lot sense for them. It's a very good choice, but it's not the right choice for the desktop.

The requirements of desktop/embedded use, and pro audio use are different. This results in different products. Trying to marry them is a great idea, but also very difficult to achieve. Of course, we should be trying to do this, but we have much bigger problems to fix before we start with that.

Lennart

Thank you

Posted Sep 22, 2008 7:24 UTC (Mon) by alankila (guest, #47141) [Link]

In fact with my brief tussle with JACK a year or two back, it seemed to me that the JACK itself might not actually mandate any particular sampling format. It's just that everybody talks float by default, because it is an excellent choice for professional audio work. (If there are any intents to support any other format, they aren't readily apparent.)

The buffers passed between applications were still void * pointers and the only clue for their contents was a simple string describing what kind of format you had requested from JACK, a #define which expanded to a string such as "32-bit pcm floating point audio"...