Le Sun, 30 Jan 2011 20:07:28 -0600,
James Shatto <wwwshadow7@xxxxxxxxx> a Ãcrit :
> If you have a sufficiently fast computer, you MIGHT get by without it.
> But if you record/use audio and use the network at the same time
> (SKYPE?), without it you can have pops an clicks and other annoyances
> in the outgoing / recorded audio track.
I agree that with a rt kernel, you can setup the hardware priority of
the different pieces of hardware in the box. But the result will depend
of what you do with your network. Server like applications (as example
p2p) need throughput, and with a rt kernel, the price to pay to get
low latency is the throughput.
> If all your machine does is
> play youtube vids, then perhaps you don't need it. But you probably
> wont know until you DO need it. And you'll only know that if you went
> without it. There are benefits beyond just audio with a low latency
> kernel. If your network is slow, and other annoyances that could give
> you windows like responsiveness (not a compliment) if you don't have a
> low latency kernel.
Again, network operation need throughput, audio pro need low latency,
and you cannot have both. Well, almost. By setting the hardware
priorities with a rt kernel, the rt kernel will give you low latency,
and the setting of the hardware priorities will give you a high
throughput on the piece of hardware that have the highest priority.
Associated with the setting of the software priorities, you can get
the lowest latency with the sound card. But the other pieces of
hardware will get higher latency and lower throughput when the
system will be under heavy audio processing load.
With an uniprocessor machine, if the audio card have the highest
hardware priority, your system is under heavy audio load, and you are
into the audio group, this can freeze your desktop a few seconds, but
the audio processing will be OK. That is realtime kernel :)
With such a machine, you can even freeze the whole system, and a reboot
will be necessary.
Hopefully, with the cgroups in the kernel and in JACK, all this will be
soon an old story. But even with cgroups, the price to pay to get low
latency will be the throughput. cgroups is only a better way to
manage the priorities, it will insure the integrity of the system
even under very heavy load on a low end machine.
Dominique
>
>
> On 1/30/11, Dominique Michel <dominique.michel@xxxxxxxxx> wrote:
> > Le Sat, 22 Jan 2011 20:33:10 +0100,
> > c0 <elcorreodelcoco@xxxxxxxxx> a Ãcrit :
> >
> >> Hi.
> >>
> >> I have not read/understand what you are trying to explain because
> >> my english level is not native language (sorry). But i can tell you
> >> something:
> >>
> >> For audio works, i recommend you using JACK (with a realtime
> >> kernel).
> >>
> >> Gracias.
> >> Chau!
> >>
> >> On Sat, 2011-01-22 at 11:07 +0000, Jim Lesurf wrote:
> >> > Hi,
> >> >
> >> > This is my first posting to the email list. I have looked at the
> >> > archives, but apologies if the following is too far OT or an old
> >> > chestnut.
> >> >
> >> > I'm experimenting with two different USB sound (playing) devices
> >> > that should be capable of asynchronous transfers up to and
> >> > including 96k/24bit. The following is when testing a Halide
> >> > Bridge. Using two systems, but the one relevant here is a
> >> > Shuttle running Ubuntu.
> >> >
> >> > I've successfully arranged for the device(s) I'm trying out to be
> >> > the 'default' alsa device for playing by including
> >> >
> >> > pcm.!default {
> >> > type plug
> >> > slave {
> >> > pcm "hw:2"
> >> > channels 2
> >> > format S24_3LE
> >> > }
> >> > }
> >> >
> >> > in my 'asoundrc' file. (NB there are two other 'soundcards'
> >> > present and the one I'm testing expect S24_3LE and is hw:2.)
> >> >
> > Did you try to fix the card order into /etc/modprobes.d/alsa.conf
> > instead of inside .asoundrc ?
> >
> > You can use the options 'index=n' and 'snd cards_limit=x' for that.
> >
> > Ciao,
> > Dominique
> >
> >> > The Halide Bridge converts USB into (coax) SPDIF.
> >> >
> >> > If I play a 96k/24 LPCM wave file with the aplay command I can
> >> > record its SPDIF output with an audio recorder and compare it
> >> > sample-by-sample with the file I played. Using aplay, the result
> >> > for the brief tests so far show bit-for-bit correct output
> >> > values. So that looks like the Bridge and Shuttle hardware are
> >> > working OK.
> >> >
> >> > However if I use Audacious to play a file, the result comes out
> >> > as 16 bit per sample. i.e. the least significant byte is being
> >> > ignored/zerod. I have selected the '24 bit' output option, and
> >> > specified not to use its volume control. I have also selected the
> >> > 'advanced' option to 'avoid processing' and have selected the
> >> > default ALSA output.
> >> >
> >> > Have I missed something with Audacious? Or is this a known
> >> > problem? I've found on the web mentions of making sure you
> >> > select the 24bit output and the 'avoid processing'. And apart
> >> > from the missing bottom 8 bits the values don't immediately look
> >> > 'altered' in other ways. Not looked yet in detail though as the
> >> > loss of the lowest bytes is the obvious symptom!
> >> >
> >> > Can give more details if asked, but at present not sure what to
> >> > add and wish to avoid taking up too much space if this is OT or a
> >> > chestnut.
> >> >
> >> > Thanks in advance for any help,
> >> >
> >> > Jim
> >> >
> >>
> >>
> >>
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