Has anyone managed to successfully get jackd talking to dmix?
My dmix setup works fix (i run VoIP stuff, artsd, and other assorted junk
through it daily).
jackd works great when connect to hw:0,0 or to plughw:0,0 (with the warning),
but if I point it at my dmix'd "default" it sits there with 100% CPU usage,
and doesn't otherwise operate.
Does anyone have this combination of jack and dmix working?
(i'm running alsa 1.0.11rc3 on 2.6.15.1, i've tried alsa 1.0.10 and jack
0.100.0 too)
>jackd -v -d alsa -d default -r 48000 -S -n 8 -p 1024 -P
jackd 0.100.7
Copyright 2001-2005 Paul Davis and others.
jackd comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
JACK compiled with System V SHM support.
server `default' registered
loading driver ..
registered builtin port type 32 bit float mono audio
new client: alsa_pcm, id = 1 type 1 @ 0x8056a48 fd = -1
apparent rate = 48000
creating alsa driver ... default|-|1024|8|48000|0|0|nomon|swmeter|-|16bit
configuring for 48000Hz, period = 1024 frames, buffer = 8 periods
You appear to be using the ALSA software "plug" layer, probably
a result of using the "default" ALSA device. This is less
efficient than it could be. Consider using a hardware device
instead rather than using the plug layer. Usually the name of the
hardware device that corresponds to the first soun
nperiods = 8 for playback
new buffer size 1024
registered port alsa_pcm:playback_1, offset = 0
registered port alsa_pcm:playback_2, offset = 0
++ jack_rechain_graph():
client alsa_pcm: internal client, execution_order=0.
-- jack_rechain_graph()
7129 waiting for signals
load = 1.1672 max usecs: 498.000, spare = 20835.000
load = 1.3102 max usecs: 310.000, spare = 21023.000
load = 1.3793 max usecs: 309.000, spare = 21024.000
load = 1.4162 max usecs: 310.000, spare = 21023.000
....
etc
....
Hi all !
I have installed a RME DIGI9652 on a Debian Sarge and it seems to work great
with Ardour ! However my two expansion boards AEB8-I [1] and AEB8-O [2] plugged
respectively on the "CD IN" input and "ADAT 1" output of the main board - as it
is mentionned in the documentation - don't get any signal, even when I change
existing interrupts in kmix, gamix, etc.. I would then ask to anybody who knows
the chipset/driver if there are special routing interrupts to be switched with
alsactl or something...
Thanks a lot for your help and useful work !
p--g
`
Parisson, Paris
(sorry for the wrong 1st Re: mail)
Question for the DSP gurus:
How hard would it be to implement an AC3 encoder as a LADSPA plugin that
could be used in an ALSA config to encode stereo and 5.1 sources on the
fly? Many Windows drivers seem to contain one, and it would be a nice
response to the naysayers on LKML who doubt the power of ALSA.
It appears that a simple AC3 encoder is ~1500 lines of code:
http://www.koders.com/c/fid04210C5E2BC83FC0BA5E0A2A1C37D52503E31EFD.aspxhttp://www.telos.de/Surround_Sound_Formats.360.0.html#1051
FWIW, when emu10k1/2 based cards first came out Creative claimed the DSP
was powerful enough to do AC3 encoding on the fly, but, they never
shipped an implementation - their drivers did it in software. One of
their engineers said the DSP wasn't suited for the frequency domain
processing required to encode AC3.
Lee
Hi,
new to this list and knowing absolutely nothing about C++ audio
programming I would like to ask for a little bit of help.
I want to create a console keyboard to MIDI application and for this I
would like to read the physical status of keyboard keys. Thank you.
Carlo Capocasa
Reuben Martin:
>> (Why hasn't anyone made a ladspa plugin with a GUI by the way? Its
>> really simple just spawning of a gui process program.)
>>
>
>Because you have no way of knowing if the platform you are running it
>
No no, you misunderstand. I said "spawning of a gui process" (I should
rather have said "spawning off a gui process", but I didn't. :-) ).
Well, I guess the question was more retorical. I personally think reason
is that linux programmers aren't that much into bells and whistles as
windows programmers.
>on will have support for the toolkit needed by the GUI. It would be
>nice to append the LADSPA spec to allow for a simple markup language
>to describe the GUI, and then depend on the host to render that markup
>language into a GUI. It would make it toolkit independent.
No, that would not be nice at all. Far too complicated for the hosts, and
guis would be different from host to host, and limited by the markup
language
What would be nice was if we used a common gui-designer like qdesigner
or glade, so that someone could make an automatic gui-spawner library
that used the xml-files from qdesigner or glade to make guis. That way,
anyone could make/edit gui's quite easely. The idea was proposed some
years ago, but no one has picked it up. Its not much work to do, but I
guess no one has got the time. It must be a community project as well,
because it would be useless if no one bothered to make guis for the
various plug-ins or the hosts didn't support it. (well, the host-problem
could be fixed automatically by making a wrapper ladspa plugin, but its
not the ideal solution)
Lee Revell:
> > Won't help if the code is to be part of a GPL'd
> > application.
>
> The Linux kernel is a GPL'ed application yet Nvidia
> gets away with linking into it.
Quite different. Anyone can distribute the kernel
without caring about the existence of the nVidia
drivers. But if an application includes the VSTSDK, it
presumably isn't complete without it.
Chris
Screenshot:
http://www.kvraudio.com/forum/viewtopic.php?
t=114488&postdays=0&postorder=asc&highlight=linux&start=105
i will be interested to see just how good its audio handling
capabilities are.
--p
Lee Revell writes:
> On Fri, 2006-01-27 at 15:57 +0100, Michael Bohle wrote:
> > But anyway, VST on Linux is dead now, beause
> > most of the user are not
> > able to compile it for themself.
>
> Wrong. You just need to write a wrapper that
> handles the compiling.
Won't help if the code is to be part of a GPL'd
application.
Also, I think (?) you have to register to download the
VSTSDK.
Chris
One Harold Chu on LKML is insisting that POSIX requires
pthread_mutex_unlock to reschedule if other threads are waiting on the
mutex, and that even if the calling thread immediately tries to lock the
mutex again another thread must get it. I contend that both of these
assertions are wrong - first, I just don't read the standard that way,
and second, it would lead to obviously incorrect behavior - unlocking a
mutex would no longer be an RT safe operation. What would be the point
of trylock() in RT code if unlocking is going to cause a reschedule
anyway?
Can anyone back me up on this?
Lee
Hello
I started a session in Ardour - drag and dropped a .wav file, then recorded (sucessfully) from a Jack input.
Checking the session folder, Ardour appears to record to disc as it's going along (on-the-fly).
I presume Ardour can also route the incoming sound to its outputs, for monitoring.
Does anyone know what mechanism Ardour uses to do this?
(I'll walk blindly into speculation. Ardour uses a jack ringbuffer on its Jack input port, in the approved way - but that only leaves one read on the ringbuffer output...)
Robert