5 Feb
2005
5 Feb
'05
11 a.m.
Hi!
Wanted to test the DSP-parts of ld10k1 (0.1.6) against my sblive, but
I'm not very successful. When trying to start ld10k1 I only get:
Error: unable to peek code
Error: error in main loop
Using alsa 1.0.5, Mandrake, Thac's 2.4.21 lowlatency-kernel.
I understand people have managed to get this up and running. Before
upgrading alsa (to be able to use ld10k1 > 0.1.6) what versions of alsa
and ld10k1 are you using?
/Mathias
5 Feb
5 Feb
7:50 p.m.
On Saturday 05 February 2005 09:57 am, Mathias Lundgren wrote:
Hi!
Wanted to test the DSP-parts of ld10k1 (0.1.6) against my sblive, but
I'm not very successful. When trying to start ld10k1 I only get:
Error: unable to peek code
Error: error in main loop
I got the first initially. Check the INSTALL file for the startup sequence.
Did you start the server with root permissions?
Using alsa 1.0.5, Mandrake, Thac's 2.4.21 lowlatency-kernel.
I understand people have managed to get this up and running. Before
upgrading alsa (to be able to use ld10k1 > 0.1.6) what versions of alsa
and ld10k1 are you using?
/Mathias
ALSA 1.0.8 (debian sources), 2.4.27 with low latency and preempt patches
(debian sources).
ld10k1 and qlo10k1) from
http://ld10k1.sourceforge.net/sources.html
2004/12/25 qlo10k1 version 0.1.2
qlo10k1-0.1.2.tar.gz
2004/12/25 ld10k1 version 0.1.8
ld10k1-0.1.8.tar.gz
Still getting a handle on making FX come out, but this is a beautiful thing.
9:07 p.m.
Hi
On Sábado 05 Febrero 2005 14:57, Mathias Lundgren wrote:
Wanted to test the DSP-parts of ld10k1 (0.1.6) against
my sblive, but
I'm not very successful. When trying to start ld10k1 I only get:
Error: unable to peek code
Error: error in main loop
I was the same error in some versions ... try to compile with different gcc
-Ox number ... i was solve this errors doing that.
Josep
10:40 p.m.
On Sat, 2005-02-05 at 15:57 +0100, Mathias Lundgren wrote:
Hi!
Wanted to test the DSP-parts of ld10k1 (0.1.6) against my sblive, but
I'm not very successful. When trying to start ld10k1 I only get:
Error: unable to peek code
Error: error in main loop
Using alsa 1.0.5, Mandrake, Thac's 2.4.21 lowlatency-kernel.
I understand people have managed to get this up and running. Before
upgrading alsa (to be able to use ld10k1 > 0.1.6) what versions of alsa
and ld10k1 are you using?
First, 0.1.6 is old, you should use ld10k1 0.1.8 which requires ALSA
1.0.8. Generally ld10k1 tracks the ALSA versioning. If you use ALSA
CVS, use ld10k1 CVS.
Lee
6 Feb
6 Feb
3:02 p.m.
sön 2005-02-06 klockan 03.40 skrev Lee Revell:
First, 0.1.6 is old, you should use ld10k1 0.1.8 which
requires ALSA
1.0.8. Generally ld10k1 tracks the ALSA versioning. If you use ALSA
CVS, use ld10k1 CVS.
Lee
First, thanks everyone! alsa 1.0.8 and ld10k1 0.1.8 worked.
Now for the big question... How do I do anything useful with it? I
understand it as I (finally) actually have to try to understand what's
happening behind the scenes in the hardware. Does anyone have any
pointers to information on what's going on in "the matrix" visualized
in qlo10k1?
/Mathias
7 Feb
7 Feb
6:32 a.m.
Mathias Lundgren wrote:
sön 2005-02-06 klockan 03.40 skrev Lee Revell:
... yes, the tough part :)
First, 0.1.6 is old, you should use ld10k1 0.1.8
which requires ALSA
1.0.8. Generally ld10k1 tracks the ALSA versioning. If you use ALSA
CVS, use ld10k1 CVS.
Lee
First, thanks everyone! alsa 1.0.8 and ld10k1 0.1.8 worked.
Now for the big question... How do I do anything useful with it?
I
understand it as I (finally) actually have to try to understand what's
happening behind the scenes in the hardware.
... the harder part ... the epiphany will come (well, i hope it comes to
me soon!) ...
i shied away from ld10k1 after i installed it, because i just knew it
would take me *ages* to get the qlo10k1 interface working for me, and
uncluttered enough not to give me headaches :(
... it's probably possible to achieve most of the good things i
associated with the kx audio drivers under windows (comprehensive saving
of dsp configurations and routing - and, consequently, changing default
values - , flexible routings and placements of dsp effects, integration
with mixer etc), but i'm not sure this would be easy or intuitive at the
moment.
Does anyone have any
pointers to information on what's going on in "the matrix" visualized
in qlo10k1?
i'm also not quite sure what the deal with the dsp effects and filters
are though - i couldn't find any way to load them in the dsp routing
interface (like the ones included with the kxaudio drivers - maybe these
have to be installed seperately ... i should read more documentation
...)? all i saw were the chorus and reverb fx busses ...
shayne
11:19 a.m.
On Mon, 2005-02-07 at 21:43 +1100, Shayne O'Connor wrote:
... it's probably possible to achieve most of the
good things i
associated with the kx audio drivers under windows (comprehensive saving
of dsp configurations and routing - and, consequently, changing default
values - , flexible routings and placements of dsp effects, integration
with mixer etc), but i'm not sure this would be easy or intuitive at the
moment.
ld10k1 (and the emu10k1 ALSA driver) still need a bit of work before
they will be as full featured as the kX driver.
The ld10k1 API provides a very thin layer between the DSP microcode and
the ALSA mixer API, creating standard ALSA mixer objects on the fly for
the control GPRs. ld10k1 currently just loads the DSP patches. The kX
driver OTOH provides a pretty thin C++ wrapper around the DSP patches,
which lets them have custom GUIs, load, unload, and generate microcode
on the fly and control everything with midi. It can query the kernel
part of the driver what hardware is available and load specific patches
accordingly.
i'm also not quite sure what the deal with the dsp
effects and filters
are though - i couldn't find any way to load them in the dsp routing
interface (like the ones included with the kxaudio drivers - maybe these
have to be installed seperately ... i should read more documentation
...)? all i saw were the chorus and reverb fx busses ...
Many of those plugins are free (some speech, some beer). There are also
many free plugins (well patches anyway) available with the OSS driver,
and floating around on the net from miscellaneous soures. It would be
good if someone had time to assemble them all and figure out what works
and what we can redistribute.
Lee
2:34 p.m.
mån 2005-02-07 klockan 16.19 skrev Lee Revell:
Would be fun, if I can get any clues to how this works, and if I have
the time.
/Mathias
On Mon, 2005-02-07 at 21:43 +1100, Shayne O'Connor
wrote:
If I can manage to use a reverb and a chorus for the midi I'm happy (at
least for now).
... it's probably possible to achieve most of
the good things i
associated with the kx audio drivers under windows (comprehensive saving
of dsp configurations and routing - and, consequently, changing default
values - , flexible routings and placements of dsp effects, integration
with mixer etc), but i'm not sure this would be easy or intuitive at the
moment.
ld10k1 (and the emu10k1 ALSA driver) still need a bit of work before
they will be as full featured as the kX driver.
The ld10k1 API provides a very thin layer between the DSP microcode and
the ALSA mixer API, creating standard ALSA mixer objects on the fly for
the control GPRs. ld10k1 currently just loads the DSP patches. The kX
driver OTOH provides a pretty thin C++ wrapper around the DSP patches,
which lets them have custom GUIs, load, unload, and generate microcode
on the fly and control everything with midi. It can query the kernel
part of the driver what hardware is available and load specific patches
accordingly.
i'm also
not quite sure what the deal with the dsp effects and filters
are though - i couldn't find any way to load them in the dsp routing
interface (like the ones included with the kxaudio drivers - maybe these
have to be installed seperately ... i should read more documentation
...)? all i saw were the chorus and reverb fx busses ...
Many of those plugins are free (some speech, some beer). There are also
many free plugins (well patches anyway) available with the OSS driver,
and floating around on the net from miscellaneous soures. It would be
good if someone had time to assemble them all and figure out what works
and what we can redistribute.
Lee
2:28 p.m.
mån 2005-02-07 klockan 11.43 skrev Shayne O'Connor:
I'm not sure I saw a lot of documentation. I saw some doco for audigy in
ld10k1, I guess a lot is applicable on an sblive, but by reading that I
can't really relate it to the gui. I've managed to load effects, but I
need to get an idea of how things actually make their way to the output,
and which things they pass on their way there, to connect them in a sane
way. I also need to get an idea of how the different things in qlo10k1
relate to the controls in alsamixer (I really hope there is _some_
relation between them :-). I'll dig along...
/Mathias
Mathias Lundgren wrote:
I've tried kx, but I'm not sure the DSP-routing did look much better. I
never used the DSP parts of it, just went with the default, so it's a
different story for me this time. :-)
sön 2005-02-06 klockan 03.40 skrev Lee Revell:
... yes, the tough part :)
First, 0.1.6 is old, you should use ld10k1 0.1.8
which requires ALSA
1.0.8. Generally ld10k1 tracks the ALSA versioning. If you use ALSA
CVS, use ld10k1 CVS.
Lee
First, thanks everyone! alsa 1.0.8 and ld10k1 0.1.8 worked.
Now for the big question... How do I do anything useful with it?
I
understand it as I (finally) actually have to try to understand what's
happening behind the scenes in the hardware.
... the harder part ... the epiphany will come (well, i hope it comes to
me soon!) ...
i shied away from ld10k1 after i installed it, because i just knew it
would take me *ages* to get the qlo10k1 interface working for me, and
uncluttered enough not to give me headaches :(
... it's probably possible to achieve most of the good things i
associated with the kx audio drivers under windows (comprehensive saving
of dsp configurations and routing - and, consequently, changing default
values - , flexible routings and placements of dsp effects, integration
with mixer etc), but i'm not sure this would be easy or intuitive at the
moment.
Does anyone
have any
pointers to information on what's going on in "the matrix" visualized
in qlo10k1?
i'm also not quite sure what the deal with the dsp effects and filters
are though - i couldn't find any way to load them in the dsp routing
interface (like the ones included with the kxaudio drivers - maybe these
have to be installed seperately ... i should read more documentation
...)? all i saw were the chorus and reverb fx busses ...
shayne
8 Feb
8 Feb
12:54 a.m.
On Monday 07 February 2005 01:26 pm, Mathias Lundgren wrote:
mån 2005-02-07 klockan 11.43 skrev Shayne
O'Connor:
I'm not sure I saw a lot of documentation. I saw some doco for audigy in
ld10k1, I guess a lot is applicable on an sblive, but by reading that I
The init_live script appears to have some naming issues FWIW.
Mathias Lundgren wrote:
I've tried kx, but I'm not sure the DSP-routing did look much better. I
never used the DSP parts of it, just went with the default, so it's a
different story for me this time. :-)
sön 2005-02-06 klockan 03.40 skrev Lee Revell:
... yes, the tough part :)
First, 0.1.6 is old, you should use ld10k1 0.1.8
which requires ALSA
1.0.8. Generally ld10k1 tracks the ALSA versioning. If you use ALSA
CVS, use ld10k1 CVS.
Lee
First, thanks everyone! alsa 1.0.8 and ld10k1 0.1.8 worked.
Now for the big question... How do I do anything useful with it? I
understand it as I (finally) actually have to try to understand what's
happening behind the scenes in the hardware.
... the harder part ... the epiphany will come (well, i hope it comes to
me soon!) ...
i shied away from ld10k1 after i installed it, because i just knew it
would take me *ages* to get the qlo10k1 interface working for me, and
uncluttered enough not to give me headaches :(
... it's probably possible to achieve most of the good things i
associated with the kx audio drivers under windows (comprehensive saving
of dsp configurations and routing - and, consequently, changing default
values - , flexible routings and placements of dsp effects, integration
with mixer etc), but i'm not sure this would be easy or intuitive at the
moment. Does
anyone have any
pointers to information on what's going on in "the matrix" visualized
in qlo10k1?
i'm also not quite sure what the deal with the dsp effects and filters
are though - i couldn't find any way to load them in the dsp routing
interface (like the ones included with the kxaudio drivers - maybe these
have to be installed seperately ... i should read more documentation
...)? all i saw were the chorus and reverb fx busses ...
shayne can't really relate it to the gui. I've
managed to load effects, but I
I keep getting "not enough tram" or similar when I try to load FX.
Can you clue me on what you did?
need to get an idea of how things actually make their
way to the output,
and which things they pass on their way there, to connect them in a sane
way. I also need to get an idea of how the different things in qlo10k1
relate to the controls in alsamixer (I really hope there is _some_
relation between them :-). I'll dig along...
AFAICT, you can just drop in the blocks you need and connect 'em up
(context menus for the connections are available if the cursor is hovering
over the line - took me a while to find that), then the controls show up in
the mixer. It crashes kmix when things change but IIRC you get the AC97
section + whatever you define in alsamixer (or aumix, what have you)
IOW you can apparently define the whole signal path
/Mathias
3:34 p.m.
I'm not
sure I saw a lot of documentation. I saw some doco for audigy in
ld10k1, I guess a lot is applicable on an sblive, but by reading that I
The init_live script appears to have some naming issues FWIW.
can't
really relate it to the gui. I've managed to load effects, but I
I keep getting "not enough tram" or similar when I try to load FX.
Can you clue me on what you did?
need to get an
idea of how things actually make their way to the output,
and which things they pass on their way there, to connect them in a sane
way. I also need to get an idea of how the different things in qlo10k1
relate to the controls in alsamixer (I really hope there is _some_
relation between them :-). I'll dig along...
AFAICT, you can just drop in the blocks you need and connect 'em up
(context menus for the connections are available if the cursor is hovering
over the line - took me a while to find that), then the controls show up in
the mixer. It crashes kmix when things change but IIRC you get the AC97
section + whatever you define in alsamixer (or aumix, what have you)
IOW you can apparently define the whole signal path
4:21 p.m.
Mathias Lundgren wrote:
Yes, it seems so. I located it to the line that results in the following
output:
/usr/local/bin/lo10k1 -n --ctrl "c-Left,Right:Wave Playback
Volume,t-Wave Playback Volume:1,c-LeftSurr,RightSurr:Wave Surro
und Playback Volume,t-Wave Surround Playback Volume:1,c-Center:Wave
Center Playback Volume,t-Wave Center Playback Volume:1,c
-LFE:Wave LFE Playback Volume,t-LFE Playback Volume:1" --patch_name
"Wave Multi Volume" -a simple.emu10k1
Apparently c- has to do with "concatenating controls", not sure what it
means... simple.emu10k1 was loaded when I used this instead:
lo10k1 -n --ctrl "" --patch_name "Wave Multi Volume" -a
simple.emu10k1
I guess it breaks things...
I launched ld10k1 with -t 7, it says it initializes tram with 1024k.
Whatever that means...
as far as i can remember, tram is the onboard memory for the emu10k1
chip - different dsp effects take up different amounts of the cards
memory, so you can't load up an unlimited number of effects ... i think
the loading of soundfonts would also reduce the available tram size?
shayne
I'm not sure I saw a lot of documentation. I saw
some doco for audigy in
ld10k1, I guess a lot is applicable on an sblive, but by reading that I
The init_live script appears to have some naming issues FWIW.
can't
really relate it to the gui. I've managed to load effects, but I
I keep getting "not enough tram" or similar when I try to load FX.
Can you clue me on what you did?
9 Feb
9 Feb
2:02 a.m.
On Tuesday 08 February 2005 03:21 pm, Shayne O'Connor wrote:
Mathias Lundgren wrote:
Yes, it seems so. I located it to the line that results in the following
output:
/usr/local/bin/lo10k1 -n --ctrl "c-Left,Right:Wave Playback
Volume,t-Wave Playback Volume:1,c-LeftSurr,RightSurr:Wave Surro
und Playback Volume,t-Wave Surround Playback Volume:1,c-Center:Wave
Center Playback Volume,t-Wave Center Playback Volume:1,c
-LFE:Wave LFE Playback Volume,t-LFE Playback Volume:1" --patch_name
"Wave Multi Volume" -a simple.emu10k1
Apparently c- has to do with "concatenating controls", not sure what it
means... simple.emu10k1 was loaded when I used this instead:
lo10k1 -n --ctrl "" --patch_name "Wave Multi Volume" -a
simple.emu10k1
I guess it breaks things...
I launched ld10k1 with -t 7, it says it initializes tram with 1024k.
Whatever that means...
as far as i can remember, tram is the onboard memory for the emu10k1
chip - different dsp effects take up different amounts of the cards
memory, so you can't load up an unlimited number of effects ... i think
the loading of soundfonts would also reduce the available tram size?
shayne
j
j4strngs@araka:~$ asfxload -M
DRAM memory left = 131068 kB
I'll have to check once I get some FX loaded
I'm not sure I saw a lot of documentation. I saw
some doco for audigy in
ld10k1, I guess a lot is applicable on an sblive, but by reading that I
The init_live script appears to have some naming issues FWIW. can't
really relate it to the gui. I've managed to load effects, but I
I keep getting "not enough tram" or similar when I try to load FX.
Can you clue me on what you did?
2:19 a.m.
On Wednesday 09 February 2005 01:02 am, John Check wrote:
On Tuesday 08 February 2005 03:21 pm, Shayne
O'Connor wrote:
The TRAM and DRAM are separate apparently..
j4strngs@araka:~$ asfxload -M
DRAM memory left = 131068 kB
Mathias Lundgren wrote:
as far as i can remember, tram is the onboard memory for the emu10k1
chip - different dsp effects take up different amounts of the cards
memory, so you can't load up an unlimited number of effects ... i think
the loading of soundfonts would also reduce the available tram size?
shayne
j
j4strngs@araka:~$ asfxload -M
DRAM memory left = 131068 kB
I'll have to check once I get some FX loaded
>I'm not sure I saw a lot of documentation. I saw some doco for audigy
> in ld10k1, I guess a lot is applicable on an sblive, but by reading
> that I
The init_live script appears to have some naming issues FWIW.
Yes, it seems so. I located it to the line that results in the following
output:
/usr/local/bin/lo10k1 -n --ctrl "c-Left,Right:Wave Playback
Volume,t-Wave Playback Volume:1,c-LeftSurr,RightSurr:Wave Surro
und Playback Volume,t-Wave Surround Playback Volume:1,c-Center:Wave
Center Playback Volume,t-Wave Center Playback Volume:1,c
-LFE:Wave LFE Playback Volume,t-LFE Playback Volume:1" --patch_name
"Wave Multi Volume" -a simple.emu10k1
Apparently c- has to do with "concatenating controls", not sure what it
means... simple.emu10k1 was loaded when I used this instead:
lo10k1 -n --ctrl "" --patch_name "Wave Multi Volume" -a
simple.emu10k1
I guess it breaks things...
>can't really relate it to the gui.
I've managed to load effects, but I
I keep getting "not enough tram" or similar when I try to load FX.
Can you clue me on what you did?
I launched ld10k1 with -t 7, it says it initializes tram with 1024k.
Whatever that means...
7:35 a.m.
Hi
Well i have a success stories, Reverb and Chorus for my sblive hardware
soundfonts, i was modified a OSS implementation efect for freeverb, and
stereo chorus, not the best reverb and chorus in the world, but runs.
Anyway the TRAM is not unlimited, i dont know why, but in ld10k1 1.0.8 version
i was need to reduce the chorus delay line because ld10k1 returns no free
TRAM available, that occurs also in kxproject drivers if you wanna load the
Creative Reverb and Chorus effects, almost on my sb live card. In ld10k1
1.0.1 was run fine with a bigger delay line, maybe was a bug, I dont know.
But, ld10k1 is so beatiful, reverb and chorus controls appears in alsamixer
and you can modify their parameters.
Josep
11 Feb
11 Feb
9:48 a.m.
New subject: [linux-audio-user] What to exprect from a dial p3 800 @ ardour
Hmm,
I think i'm running into a system limitation...
here are the details.
dual p3 800 with 1256 mb ram
delta 44 using envy mixer..
fresh planet ccrma install..
while using ardour 9.24 with 4 tracks and 1 bus (the bus has gverb)
each track has a compressor and 2 have eq.
so..
@24/96...
Automation on 2 tracks..
4 tracks 1 bus
7 ladspa plugins: 4 compressors (dynasone) 2 eq's (tap eq) 1 reverb (
gverbon a bus)
Should my system handle this?
It seems to be getting beat-up... (
10:41 a.m.
New subject: [linux-audio-user] What to exprect from a dial p3 800 @ ardour
Its asking a lot from it. Only one processor can be used for DSP work
currently.
Did you really mean 1256 MB of ram (I dont see how you could get that
ammount) or some other value?
- Steve
On Fri, Feb 11, 2005 at 08:48:22 -0500, Timothy Alan Gorman wrote:
Hmm,
I think i'm running into a system limitation...
here are the details.
dual p3 800 with 1256 mb ram
delta 44 using envy mixer..
fresh planet ccrma install..
while using ardour 9.24 with 4 tracks and 1 bus (the bus has gverb)
each track has a compressor and 2 have eq.
so..
@24/96...
Automation on 2 tracks..
4 tracks 1 bus
7 ladspa plugins: 4 compressors (dynasone) 2 eq's (tap eq) 1 reverb (
gverbon a bus)
Should my system handle this?
It seems to be getting beat-up... (
11:01 a.m.
New subject: [linux-audio-user] What to exprect from a dial p3 800 @ ardour
Tha ram is
2 512mb strips and
1 256mb..
..
DSP's use 1 processor?
interesting.
what kind of,performance can I exprect from a p4 1.7 (my current
winblows box) ;)
so that's a bit much for a dual p3 800 ...
Hmmm,
Steve Harris wrote:
Its asking a lot from it. Only one processor can be
used for DSP work
currently.
Did you really mean 1256 MB of ram (I dont see how you could get that
ammount) or some other value?
- Steve
On Fri, Feb 11, 2005 at 08:48:22 -0500, Timothy Alan Gorman wrote:
Hmm,
I think i'm running into a system limitation...
here are the details.
dual p3 800 with 1256 mb ram
delta 44 using envy mixer..
fresh planet ccrma install..
while using ardour 9.24 with 4 tracks and 1 bus (the bus has gverb)
each track has a compressor and 2 have eq.
so..
@24/96...
Automation on 2 tracks..
4 tracks 1 bus
7 ladspa plugins: 4 compressors (dynasone) 2 eq's (tap eq) 1 reverb (
gverbon a bus)
Should my system handle this?
It seems to be getting beat-up... (
11:33 a.m.
New subject: [linux-audio-user] What to exprect from a dial p3 800 @ ardour
On Fri, Feb 11, 2005 at 10:01:58 -0500, Timothy Alan Gorman wrote:
Tha ram is
2 512mb strips and
1 256mb..
..
DSP's use 1 processor?
Currently yes, though I think someones made a version of JACK to use
both on OSX.
interesting.
what kind of,performance can I exprect from a p4 1.7 (my current
winblows box) ;)
Probably better overall, but you will suffer from denormals more, and the
UIs wont update as quickly (thats what the other processor was doing).
You might find the realtime stuff easier going on a single processor
machine too, there have been problems with realtime work on dual cpu
machines in the past.
so that's a bit much for a dual p3 800 ...
It's certainly borderline. I have a similarly specced machine, but its
mostly acting as a webserver, so I can't tell you for sure.
- Steve
>Its asking a lot from it. Only one processor can
be used for DSP work
>currently.
>
>Did you really mean 1256 MB of ram (I dont see how you could get that
>ammount) or some other value?
>
>- Steve
>
>On Fri, Feb 11, 2005 at 08:48:22 -0500, Timothy Alan Gorman wrote:
>
>
>>Hmm,
>>I think i'm running into a system limitation...
>>here are the details.
>>dual p3 800 with 1256 mb ram
>>delta 44 using envy mixer..
>>fresh planet ccrma install..
>>
>>while using ardour 9.24 with 4 tracks and 1 bus (the bus has gverb)
>>each track has a compressor and 2 have eq.
>>
>>so..
>>@24/96...
>>Automation on 2 tracks..
>>4 tracks 1 bus
>>7 ladspa plugins: 4 compressors (dynasone) 2 eq's (tap eq) 1 reverb (
>>gverbon a bus)
>>
>>Should my system handle this?
>>It seems to be getting beat-up... (
>>
>>
>
>
>
12 Feb
12 Feb
3:27 a.m.
New subject: [linux-audio-user] What to exprect from a dial p3 800 @ ardour
On Friday 11 February 2005 08:48 am, Timothy Alan Gorman wrote:
Hmm,
I think i'm running into a system limitation...
here are the details.
dual p3 800 with 1256 mb ram
delta 44 using envy mixer..
fresh planet ccrma install..
while using ardour 9.24 with 4 tracks and 1 bus (the bus has gverb)
each track has a compressor and 2 have eq.
so..
@24/96...
Automation on 2 tracks..
4 tracks 1 bus
7 ladspa plugins: 4 compressors (dynasone) 2 eq's (tap eq) 1 reverb (
gverbon a bus)
Should my system handle this?
It seems to be getting beat-up... (
You might consider bouncing some or all of the compressed tracks (you can
always go back right?)
What are you disks like? I've done 8 tracks (not all stereo) on 2 busses with
a similar number of plugins on a Celeron 733 @ 16/44.1 before things got
dicey.
4:51 a.m.
New subject: [linux-audio-user] What to exprect from a dial p3 800 @ ardour
On Fri, 2005-02-11 at 08:48 -0500, Timothy Alan Gorman wrote:
Hmm,
I think i'm running into a system limitation...
here are the details.
dual p3 800 with 1256 mb ram
delta 44 using envy mixer..
fresh planet ccrma install..
while using ardour 9.24 with 4 tracks and 1 bus (the bus has gverb)
each track has a compressor and 2 have eq.
so..
@24/96...
Automation on 2 tracks..
4 tracks 1 bus
7 ladspa plugins: 4 compressors (dynasone) 2 eq's (tap eq) 1 reverb (
gverbon a bus)
Should my system handle this?
It seems to be getting beat-up... (
What do you mean? What are the symptoms? Do you get xruns, or
something else? What period size are you using?
You can try Rui's jack_test4 suite to get an objective benchmark of your
systems audio performance. You should be able to increase the number of
clients until the DSP load or CPU load get near 100%. How many clients
can you run jack_test4 with before your system falls over?
I have a 600Mhz C3 that can run about 7 jack_test4 clients before it
croaks.
I think jack_test4 is included in JACK CVS, or there is a plan to
include it. See the jackit-devel archives for details.
Lee
9 Feb
9 Feb
2 a.m.
On Tuesday 08 February 2005 02:31 pm, Mathias Lundgren wrote:
Yes, it seems so. I located it to the line that results in the following
output:
/usr/local/bin/lo10k1 -n --ctrl "c-Left,Right:Wave Playback
Volume,t-Wave Playback Volume:1,c-LeftSurr,RightSurr:Wave Surro
und Playback Volume,t-Wave Surround Playback Volume:1,c-Center:Wave
Center Playback Volume,t-Wave Center Playback Volume:1,c
-LFE:Wave LFE Playback Volume,t-LFE Playback Volume:1" --patch_name
"Wave Multi Volume" -a simple.emu10k1
Apparently c- has to do with "concatenating controls", not sure what it
means... simple.emu10k1 was loaded when I used this instead:
lo10k1 -n --ctrl "" --patch_name "Wave Multi Volume" -a
simple.emu10k1
I guess it breaks things...
Mmm.. looks like concatenating means binding left and right into a single
entity.. I have to dump
I launched ld10k1 with -t 7, it says it initializes tram with 1024k.
Whatever that means...
I'll give it a shot.. looks like 8 (base 0) 64k blocks
I remember the "t" stands for tank... clear as mud, eh?
Alsamixer. I understand you can define the whole signal path, and I can
connect/disconnect things but it's not much of a use if I don't know
what's going on. I still need to know how it relates to the hardware...
Digging along...
/Mathias
Indeed. I've just about managed to layout the GUI where it's possible to
follow what's going on.
I'm not sure I saw a lot of documentation. I saw
some doco for audigy
in ld10k1, I guess a lot is applicable on an sblive, but by reading
that I
The init_live script appears to have some naming issues FWIW. can't really relate it to the gui. I've
managed to load effects, but I
I keep getting "not enough tram" or similar when I try to load FX.
Can you clue me on what you did? need to get an idea of how things actually make their
way to the
output, and which things they pass on their way there, to connect them
in a sane way. I also need to get an idea of how the different things
in qlo10k1 relate to the controls in alsamixer (I really hope there is
_some_ relation between them :-). I'll dig along...
AFAICT, you can just drop in the blocks you need and connect 'em up
(context menus for the connections are available if the cursor is
hovering over the line - took me a while to find that), then the controls
show up in the mixer. It crashes kmix when things change but IIRC you get
the AC97 section + whatever you define in alsamixer (or aumix, what have
you) IOW you can apparently define the whole signal path
8 Feb
8 Feb
6:23 p.m.
On Mon, 2005-02-07 at 23:54 -0500, John Check wrote:
It crashes kmix when things change but IIRC you get
the AC97
section + whatever you define in alsamixer (or aumix, what have you)
IOW you can apparently define the whole signal path
I would like to develop a hardware specific mixer for the emu10k1 using
ld10k1 as the backend. The generic OS provided mixer facilities will
never work well for this card because the control space is flat. Please
mail me if you are interested this project, especially if you know
anything about GUI programming (I don't).
Lee
7:13 p.m.
Lee Revell wrote:
On Mon, 2005-02-07 at 23:54 -0500, John Check wrote:
this is *desperately* needed ...
in my travels, i found the gemu mixer (http://www.roadfeldt.com) which
is described as thus:
It crashes kmix when things change but IIRC you
get the AC97
section + whatever you define in alsamixer (or aumix, what have you)
IOW you can apparently define the whole signal path
I would like to develop a hardware specific mixer for the emu10k1 using
ld10k1 as the backend. The generic OS provided mixer facilities will
never work well for this card because the control space is flat. Please
mail me if you are interested this project, especially if you know
anything about GUI programming (I don't).
Lee
Gemu is a gnome application designed to control sound
cards based on
the EMU10K1 platform by Creative Labs <http://www.creative.com>. I am
not trying to develop an application that gives an overly aggresive
ability to change every little aspect of the sound card. Rather I am
trying to make an app that gives you control what you hear and where
it comes from, in a simply manor. I thought a more simplified approach
was best. Honestly, how often do you want the left channel of the
digital output of the cd-rom to go to the right channel of the
headphone jack? I didn't think so. This is why gemu gives you control
over both left and right channels when you select an source or a
destination.
hmmm ... not sure about the author's assumptions on what one would want
to do in regards to routing, though :\
i haven't been able to get this working, though - just get a segfault
when trying to launch it - so i don't know how far it goes to making
emu10k1 card options clear.
i would be interested in helping with some of the graphic design needs
of any gui, and anything else i can get my head around ...
if anyone on the list has some links to info on how to build User
Interfaces/GUI's, it would be much appreciated!
shayne
9 Feb
9 Feb
3:54 a.m.
On Tuesday 08 February 2005 23:23, Lee Revell wrote:
On Mon, 2005-02-07 at 23:54 -0500, John Check wrote:
One of the things I am planning for JackMix is more backends so it isn't only
a Jack mixer but could also get a backend for emu10k1 or artsd or alsa or
gstreamer or or or...
But the day only has 24 hours and thus development is slow...
Arnold
PS: Check http://roederberg.dyndns.org/~arnold/jackmix for more details,
stable versions, daily snapshots(!) and repository-access...
--
There is a theory which states that if ever anyone discovers exactly what the
Universe is for and why it is here, it will instantly disappear and be
replaced by something even more bizarre and inexplicable.
There is another theory which states that this has already happened.
-- Douglas Adams, The Restaurant at the End of the Universe
It crashes kmix when things change but IIRC you
get the AC97
section + whatever you define in alsamixer (or aumix, what have you)
IOW you can apparently define the whole signal path
I would like to develop a
hardware specific mixer for the emu10k1 using
ld10k1 as the backend. The generic OS provided mixer facilities will
never work well for this card because the control space is flat. Please
mail me if you are interested this project, especially if you know
anything about GUI programming (I don't).
7317
days inactive
7324
days old
linux-audio-user@lists.linuxaudio.org
24 comments
8 participants
participants (8)
-
Arnold Krille -
holborn -
John Check -
Lee Revell -
Mathias Lundgren -
Shayne O'Connor -
Steve Harris -
Timothy Alan Gorman