Peter wrote:
>On Tuesday 22 Oct 2002 17:42, Paul Winkler wrote:
>> On Tue, Oct 22, 2002 at 11:14:52AM +0100, Steve Harris wrote:
>> > I can't answer this properly, but there is some issue to with mmap
>mode I
>> > believe. It is a very small number of cards that dont work.
>>
>>> We should compile a list of them, and maybe put it in the JACK FAQ.
>>
>> --PW
>I'd much rather have a list of current cards that work _well_ than a
>lists of cards that don't.
>
>The current ALSA card list just says whether or not a card has a
driver >(and the state of that driver). It doesn't indicate how well the
card >is going to work.
>
>I don't want to have to learn about DSPs and stuff to be able to
>identify a _good_ sound card. I've currently got a shortlist for my
>next machine:
> * MidiMan Delta Audiophile 2496 (Envy24)
> * VideoLogic SonicFury OEM (CS4360)
> * Creative SB PCI 128 (ES1371)
>
>I've read the specs and looked around various review sites. Nothing
>anywhere tells me how well they're going to work. Or does it? Am I
>failing to find some secret source of information?
>
>All these things just make life _easier_. I want to get on with
>developing code, not wondering why my hardware isn't performing. I
>don't _want_ to have to learn _that_ part of the system. Because I'll
>only need to do it once:
>when I spec the next machine. (The next time I spec a machine,
>everything I found out last time will be out of date.)
This is a good idea to have this support and you've picked the right
thread for asking for it. Currently there are only a handful of people
who have worked on the official ALSA docs. I have done 90% of the work
myself. It's great I'm learning heaps in the process and will definitely
think about how to make it possible to add the info you require.
Unfortunately we don't often get success stories on the ALSA lists so we
don't really know how well things work until someone says that it's
broken :(
As it is I have tried to provide a feedback inteface via the notes
additions in the docs. So far it is being used but not as much as it
could be. I guess it will just take time. You can lead a horse to water...
Do you have a specific idea envisioned for how we could present the info
more effectively or a way to make more people contribute their results?
I'm listening.
--
Patrick Shirkey - Boost Hardware Ltd.
For the discerning hardware connoisseur
Http://www.boosthardware.comHttp://www.djcj.org - The Linux Audio Users guide
========================================
"Um...symbol_get and symbol_put... They're
kindof like does anyone remember like get_symbol
and put_symbol I think we used to have..."
- Rusty Russell in his talk on the module subsystem
>From: Steve Harris <S.W.Harris(a)ecs.soton.ac.uk>
>On Tue, Oct 22, 2002 at 09:38:26 +0200, Tim Goetze wrote:
> > the problem with converging IIR is that either the IIR or the
> > converging algorithm or both don't like impulses that are too
> > lively. the converger will simply fall dead when it sees an
> > exponentially decaying white noise impulse. it makes sense,
>
>I don't understand why. Is it just beacuse the kernle is a bit too long to
>be stable, or is there someting more to it than that?
>
When the converger tries to find the best coefficients by using a recursive
error minimation i dont wonder that i may fail here. A IIR is nothing else
as a differential equation solver. As a result you can only approximate kind
of periodic signals i guess. Correct me if i am wrong, its quite a long time
ago
since i learned about linear system theorie and IIR stuff...
- Stefan
_________________________________________________________________
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> Lea Anthony <stonekeeper(a)stonekeeper.freeserve.co.uk> writes
>
> Wishing for people to write native apps for a system with no market is
> like wishing Windows would die. It might happen, but it's not bloody
> likely.
However, if you port a novel Linux application to Windows or OS X,
the users on those platforms are quite happy to add the free tool
to their workflow if it helps them do their work better. This is
how the GNU project got its start, after all -- free, usable software
that ran on popular commercial UNIX platforms. Sfront has taken
this route -- most of my users are non-Linux users now.
I think there's a migration path to Linux that could be based on
this strategy -- if the free software community comes up with a
set of audio content-creation tools that Windows or OS X users
are willing to use as a complete workflow, the case for switching
over to Linux to run the workflow more efficiently (or to avoid
OS license upgrade fees, etc) is easier to make. Certainly on the
CLI side, many people started out at Cygwin users to run emacs
and gcc and TeX under Windows, and then decided to add a dual-boot
option for Linux to get "the real thing."
-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------
> -----Original Message-----
> From: Joshua Haberman [mailto:joshua@haberman.com]
>
> "Paul Davis" <paul(a)linuxaudiosystems.com> wrote:
> > >So why, having studied the docs, am I completely stumped
> with jack? It
> > >refuses to play. I don't consider any solution based on a
> piece of software
> > >_I_ can't operate suitable for general use.
> >
> > JACK *isn't* intended for general use, and i get tired of
> suggestions
> > that it should be. there are lots of people working on solutions for
> > "general use". JACK is intended for people who are serious about
> > audio.
>
> I don't understand this attitude. I think it hinders
> acceptance of JACK
> (even for serious audio) and it contributes to further
> fragmentation of
> the Linux audio world.
>
> If you would rather everyday applications not use JACK, what
> should they
> be using instead? OSS is simple but limited, and spotty
> drivers/implementations make it difficult to write robust
> code. It will
> soon be available only through emulation. It forces use of
> the blocking
> model.
>
> ALSA is very powerful and complete, but very complex. This will make
<rant>
it is also pretty much useless for general users. I mean if I can't listen
to mp3 and browse the web at the same time ... (without sound servers which
were discussed recently and as far as I can tell the general consensus is
that they are bad and not to be used). Is there any rationale for the
blocking behaviour? I mean I can't imagine situation where I would want
playback to wait for another playback to finish and play sound then...
either report error (device busy) or just ignore the sound... (or do the
mixing) as far as I know this behaviour is not configurable - am I mistaken?
(rarely I wish to be proven wrong so much:-)
maybe this is not the best place for a rant but since we are discussing
the acceptance of linux in audio world, here's my own experience.
except of the blocking of sounds (the problem mentioned above) I am quite
dissapointed by functionality of linux drivers I have tried, I have sb live
platinum and (last time I checked):
- front panel midi connectors do not work
- game port midi functionality - sort of works, works OK with one keyboard
but does not work with another midi controller (yamaha drum toy) (both work
fine with front panel midi under windows)
- front panel RCA connectors do not work
- infra red sensor probably does not work (at least I wasn't able to make
it work and I got no responses on lasa mailing list)
</rant>
erik
> I don't know if the 3 points I've made have already been done, I'm new
> to LAD, but they are important. Especially 3 as it gives a "migration
> path" to existing windoze users. I know, I know, DirectX plugins are
> written against Microsoft libraries, but how come mplayer on Linux can
> use windows binaries (codecs) in it's system? Could the same not be
>done
>for audio? Surely it's the same principle? (Media data input, process,
>media data ouput).
Yes this should be possibile since both DirectX and VST are documented
APIs.
Form a latency point of view, I am not sure how this works out, but I
believe the process() calls do not involve windows specific stuff so the
plugs should perform quite well on a low latency kernel.
Of course almost any DX, VST plugin do have a GUI.
The ideal would be letting wine to display the GUI part while executing
the process() stuff directly.
I am not a wine expert so I cannot say how difficult this can be, but
given that wine is able to execute beasts like Word and Excel I guess
it has not problems with plugin GUIs.
Indeed the ability to run DX and VST on Linux would be a good selling
point for pro audio on linux but there is still lot to do in the fields
of basic audio platform infrastructure and integration of the different
components.
cheers,
Benno
Hello Joern,
Are you using ALSA right ?
Perhaps an OSS emulation problem of ALSA ?
I recall that I got grabled sound (even lockups) on the SBLive
when using 128byte fragments (seemed like a driver or hardware problem).
What kind of audio card do you have ?
Anyway latencytest is completely outdated (does not compile cleanly
on newer distros due to wrong includes etc).
Now that I have some spare time again (seems unbelievable but I were
able to finish my university degree in CS just a month before turning
30 ... better late than newer as we use to say :-) )
I'll try to rewrite some of the useful tools including latencytest,
adding alsa support and new stress test methods and like one guy of the
list asked , the possibility to chose the disks (or the path) on which
perform the disk i/o tests.
Anyway low latency came useful in my thesis since a part of the project
consisted in a real time laser scanner that tracks a laser spot that
you move over the object you like to scan that is filmed by two cameras
that permit a 3D reconstruction of the surface.
(at 25 FPS we have processing cycles of 40msec so it is quite easy for
the low lat patch to keep up since there is basically no disk i/o
present during the scanning activity).
cheers,
Benno.
---
You wrote:
i ran latencytest on kernel 2.5.44-mm2 yesterday, and the audio was
totally garbled, just barely recognizable.
strangely, i could aplay somesound.wav simultaneously, and it sounded
ok. (except for dropouts here and there.)
might it be an oss problem ?
i'm stumped.
i habe never seen this before with the 2.5 kernels.
regards,
jörn
> -----Original Message-----
> From: Kai Vehmanen [mailto:kai.vehmanen@wakkanet.fi]
>
> On Tue, 22 Oct 2002, Conrad Parker wrote:
>
> > it might save you some hassles if you changed the intro to
> jack's web
> > pages, which currently read:
> >
> > JACK is a low-latency audio server, written primarily for the
> > GNU/Linux operating system. It can connect a number of different
> > applications to an audio device, as well as allowing
> them to share
> > audio between themselves.
> >
> > that, by itself, sounds to the average user an awful lot
> like a general
> > purpose audio server. Perhaps what you wrote in the email
> below, comparing
> > JACK to ASIO, would be more appropriate.
>
> But the second paragraph of the intro basicly already mentions
> the focus:
>
> --cut--
> JACK is different from other audio server efforts in that it has been
> designed from the ground up to be suitable for professional
> audio work.
> This means that it focuses on two key areas: synchronous
> execution of all
> clients, and low latency operation.
> --cut--
'suitable' does not mean 'exclusively suitable'. and 'focus' does not
(usually) mean 'completely ignore anything else'. all in all if it can
handle tough tasks I would expect it to handle easy tasks as well (which I
think was the point of original post)
erik
If I remember correctly, the guy that wrote the VQF (an mp3-like codec)
plugin for xmms
(home page here:http://www.csn.ul.ie/~mel/projects/linux/vqfplugin/ ) used
wine to run the windows version of the audio codec under linux. Reading form
his page he has now switched to a native version of the codec but perhaps he
could share his experiences with us on this list. I'll try to concact him (if
he isn't already on the LAD list) and send him the URL of this thread.
cheers,
Benno
On 10/22/2002 - 04:46:47, Richard Bown said:
> On Monday 21 October 2002 20:21, Patrick Shirkey wrote:
> > But am I just wasting my breath because the Agnula crew are going to
> > do all the work for us?
>
> Oh well _now_ you come on to my pet subject.
>
> > Anyone from the Agnula project have a position on this?
>
> A while ago I got involved in a flamespat with the head honcho of AGNULA
> when I suggested that they should improve their communication with the
> rest of the LAD community and specifically with the development teams.
> I think we'd pretty much all like to know what's happening with AGNULA
> (apart from the minutes of their meetings) and naively I saw it as
> their job to tell us and be nice to us.
>
> While they do have selective communication with those that they consider
> to be key people I think my fundamental mistake was to actually think
> of AGNULA as an organisation. While it appears to be an organisation
> it's really just a tech project - it's run by geeks and it will aim to
> deliver some packaged apps and code in the time allowed. Once it's
> packaged it'll ship and LAD type projects will get exposure to the rest
> of the world through their distro. AFAICT that's the deal. While
> that's not a bad deal of course I think AGNULA should have more of a
> personality and a PR role with and in the LAD community. I've said in
> the past that I think it'd benefit "them" and "us".
>
> But hey. I'm not going to get on my hobby horse again over that one.
> No chance. No sir.
>
Too late :)
So we are not wasting our time to debate this. There is a real problem
that the professional side of this community is not really taking into
account.
I was sceptical that the Agnula team would be doing massive promotional
campaigns.
Peter you said that if we intend to make money out of Linux Audio we
*should* be paying for it. Well doesn't that alsao apply to studios and
professionals who intend solely to use the finished products to make money?
Patrick.
--
hello benno !
i ran latencytest on kernel 2.5.44-mm2 yesterday, and the audio was
totally garbled, just barely recognizable.
strangely, i could aplay somesound.wav simultaneously, and it sounded
ok. (except for dropouts here and there.)
might it be an oss problem ?
i'm stumped.
i habe never seen this before with the 2.5 kernels.
regards,
jörn
--
Jörn Nettingsmeier
Kurfürstenstr 49, 45138 Essen, Germany
http://spunk.dnsalias.org (my server)
http://www.linuxdj.com/audio/lad/ (Linux Audio Developers)