hi,
I just bought one of these little boxes (Shuttle Spacewalker XPC SN41G2
nForce2 Aluminium Barebones System - AMD Athlon XP to use for an
installation and have installed redhat9 on it and soon will be
planetccrmaing it
- it is a very quiet machine as it comes and i believe you can get an
optional super quiet power supply for it - very easy to build/configure
- i'll let you know if i run into any audio worries.
Does anybody know if i can access more than 2 the 5.1 outputs on the
onboard audio chip with PD or should i go ahead and get an SBLive for 4
outs?
nVida Corporation nForce Multimedia audio controller [VIA VT82C686B]
(rev a2)
nVida Corporation nforce AC97 audio controller (MCP) (rev 1a)
rob
--
robcanning <rscanning(a)eircom.net>
www.robcanning.utvinternet.com
hi,
how can i simply protect my own prduced audio-cds?
which progs can i use? only commercial-ones?
any experiences??
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Hi,
A new laptop I got uses somethign called SoundMAX under Windows.
dyne:bolic boots and brings up Alsa running something called snd-atiixp,
and lspci showing a bunch of ATI chips, but looking on the Alsa page I'm
not finding ATI.
Can someone point me towards the correct configuration page on Alsa
for this sound chip?
Thanks,
Mark
Hello.
I've been out of luck trying to work with sound on Linux. My experience
with Audacity, Jack and Ardour haven't been successful at all.
I was trying to create a .asoundrc file as, from what I read, it would
be one of the first necessary steps.
My soundcard is
[cyb@kali cyb]$ cat /proc/asound/cards
0 [Live ]: EMU10K1 - Sound Blaster Live!
Sound Blaster Live! (rev.8) at 0xa800, irq 10
and it's working well and I'm running Mandrake Linux 9.2 with Gnome.
My attempt at editing a .asoundrc file is shown below
pcm.!sb {
type hw
card 0
}
ctl.!sb {
type hw
card 0
}
pcm.jackplug {
type plug
slave { pcm "jack" }
}
pcm.jack {
type jack
playback_ports {
0 alsa_pcm:playback_1
1 alsa_pcm:playback_2
}
capture_ports {
0 alsa_pcm:capture_1
1 alsa_pcm:capture_2
}
}
which I copied and pasted from the Alsa site. Apparently it is working
because when I issued some commands to play a file they worked.
[cyb@kali cyb]$ aplay -d sd docs/voz.wav
Playing WAVE 'docs/voz.wav' : Signed 16 bit Little Endian, Rate 44100
Hz, Stereo[cyb@kali cyb]$ aplay -djackplug docs/voz.wav
Playing WAVE 'docs/voz.wav' : Signed 16 bit Little Endian, Rate 44100
Hz, Stereo
However, Jack is not working properly and when I try to run Ardour all
my running apps are killed and disappear.
Jack shows the following messages:
JACK compiled with System V SHM support
loading driver ..
open
starting engine
cannot set thread to real-time priority (FIFO/10) (1: Operation not
permitted)
cannot set thread to real-time priority (FIFO/20) (1: Operation not
permitted)
**** alsa_pcm: xrun of at least 10.319 msecs
I'm very sorry if the message is bit lengthy, but I wanted to give all
the info that I think is needed. Any tip or hint appreciated.
TIA
--
josenildo marques
icq #289971493
homepage http://cyb.ezdir.net
registered linux user #341648
*********
"Mistakes are the portals of discovery." James Joyce
Hello:
Sorry by my english.
I'm request for people could help in coding of isisalsa driver (for
Guillemot Maxistudio Isis). I'm not a developer, only entusiast user,
and people developing in this project "know little of alsa coding rules"
and/or got little time.
The status is only playback and still have bugs for work with alsa
applications, this is the area for my request. The reversing is complete.
Maybe the card isn't a professional audio device, but offer the
possibility of several users to switch their environment to linux (for
me it does) and cause guillemot closed win2k/xp support for this card.
Feel free to forward (please).
http://uk.guillemot.com/brand/gamme/son/msisis/isis.htmlhttp://fr.guillemot.com/brand/gamme/son/msisis/isis.htmlhttp://sourceforge.net/projects/isisalsa/
irc://freenode/isisalsa
http://www.estudiohum.cl/Isis.htm
Thanks for hear us ;)
Felipe Sologuren
Sound Engineer
> -----Original Message-----
> From: Russell King [mailto:rmk@arm.linux.org.uk] On Behalf Of Russell King
> Sent: Tuesday, March 30, 2004 3:01 AM
> To: Ivica Ico Bukvic
> Cc: 'A list for linux audio users'; alsa-devel(a)lists.sourceforge.net;
> linux-kernel(a)vger.kernel.org; linux-pcmcia(a)lists.infradead.org
> Subject: Re: [linux-audio-user] snd-hdsp+cardbus=distortion -- the
> sagacontinues (cardbus driver=culprit?) UPDATE: 99.9% sure it is the
> cardbus driver yenta_socket
>
> On Tue, Mar 30, 2004 at 12:52:11AM -0500, Ivica Ico Bukvic wrote:
> > 6) Pester alsa-dev, lau, and kernel/pcmcia people to death begging for
> help
> > :-)
> >
> > IN-PROGRESS :-)
>
> What needs to happen is that the card driver author needs to investigate
> what is going on, and, if it seems related to the core PCMCIA core or
> the socket driver, we need to get involved.
>
> IOW, linux-pcmcia people don't debug card drivers.
>
To add to what Tim mentioned, I think that the driver is fine as it does
work on select notebooks and desktops (the card can be plugged into either
PCI card or PCMCIA cardbus). Yet, in these select instances it does not work
even though neither the cardbus driver nor the actual card driver do not
report any particular problems. Hence the only logical explanation is that
there is something wrong with the pcmcia controller driver.
This card does tax the throughput of the cardbus like no other card I can
think of, hence the problem may be more widespread, but exhibits itself just
in this case where the cardbus is being pushed to its limits. Yet, the
hardware is not the issue when the same notebook/soundcard combo works
flawlessly in WinXP.
Hope this helps!
Ico
hi russel
> The majority of PCMCIA is the same between the two kernels. There
> have been some cleanups and changes to the way card events (insertions
> and removals) occur, and some setup changes to the cardbus bridge to
> turn on some extra features.
>
> However, if you're saying that 2.4 and 2.6 behave the same way, then
> logically it isn't something that any of these changes have caused.
i tried to use the pcmcia-cs driver again (2.4.24-ck1 ... they didn't
work with the hdsp, either (afaik, the pcmcia-cs driver never worked
with the hdsp, but i'm not sure about that)
the dmesg output was:
PCI: Found IRQ 5 for device 02:00.0
PCI: Sharing IRQ 5 with 01:00.0
PCI: Found IRQ 11 for device 02:00.1
PCI: Sharing IRQ 11 with 00:1f.3
PCI: Sharing IRQ 11 with 00:1f.5
PCI: Sharing IRQ 11 with 00:1f.6
O2Micro OZ6933 rev 01 PCI-to-CardBus at slot 02:00, mem 0x20000000
host opts [0]: [pci/way] [pci irq 5] [lat 168/176] [bus 3/6]
host opts [1]: [pci/way] [pci irq 11] [lat 168/176] [bus 7/10]
ISA irqs (default) = 3,4,7,9,10 PCI status changes
cs: cb_alloc(bus 7): vendor 0x10ee, device 0x3fc5
cs: cb_free(bus 7)
cs: cb_alloc(bus 7): vendor 0x10ee, device 0x3fc5
PCI: No IRQ known for interrupt pin A of device .
ALSA ../../alsa-kernel/pci/rme9652/hdsp.c:5031: unable to grab memory
region 0x0-0x1bff
RME Hammerfall-DSP: no cards found
i have no idea of the differences between the cardbus drivers of the
kernel and of the pcmcia-cs project ...
anyway, the hdsp is working on a certain area of memory, that the
pcmcia-cs driver can't grab the kernel driver can ...
i disabled acpi and apic and used pci=biosirq as kernel flag
maybe this can give you a hint ... anyway, i hope thomas can comment on
that...
cheers and thanks for your help...
Tim mailto:TimBlechmann@gmx.de
ICQ: 96771783
--
The only people for me are the mad ones, the ones who are mad to live,
mad to talk, mad to be saved, desirous of everything at the same time,
the ones who never yawn or say a commonplace thing, but burn, burn,
burn, like fabulous yellow roman candles exploding like spiders across
the stars and in the middle you see the blue centerlight pop and
everybody goes "Awww!"
Jack Kerouac
> What needs to happen is that the card driver author needs to
> investigate what is going on, and, if it seems related to the core
> PCMCIA core or the socket driver, we need to get involved.
he pointed out that it's probably a kernel problem ...
we don't want you to solve problems of the alsa driver, but we suspect
there is a problem with the cardbus bridge...
basics:
1.) the card is a combination of 2 devices: a interface device, either a
pci or a cardbus card, and a dsp device, either digital or analog
audio ... no problems have ever been reported about the pci
interface, so it's likely to be a problem with the cardbus interface
... although i have to admit, that there are only 3 reported systems
that have the problems, we experience ... but these systems are
completely different ones (p4 / amd64, different cardbus bridges)
2.) the dsp device has an internal matrix mixer, that's independant from
the computer (this audio data isn't sent to the computer at all) ...
in fact it's possible to set the matrix mixer and unplug the dsp
device from the computer ...
there are three problems, that show the situation:
3.) if we start an audio application that produces interrupts on the
hdsp device, some of the audio data that's been transfered to the
computer will be copied or mapped to the buffer of the audio data
that's been transfered from the computer to the corresponding
output. this could either happen
- inside the device itself (i doubt that, since the same driver, the
same firmware, the same devices are working without and problems
on other linux machines or with windows on the same machine)
- inside the cardbus bridge
- inside the kernel
4.) if we start the audio playback, the audio playback seems to be
distorted:
one block of samples will be followed by one block of samples of 0
(zero) ... note that these block are not the blocks of data, that
are sent to the hdsp during one interrupt. it's usually 32 samples
of sound followed by 32 samples of silence, the driver sends blocks
of 64 to 8192 during one interrupt depending on user settings...
iirc one sample is 24 bit
on the other hand the blocks of silence aren't missing, but the alsa
layer will adapt, so that it takes twice as long as it should to
play back a soundfile (i'm not sure if it's because of the alsa
soundfile player or of the driver)
5.) the reason why we (or at least i) think, that it's a problem with
the cardbus interface, is because of jack's output ...
jack is both an audio connection interface for different softwares
and a very good alsa implementation... using jack as super user
in realtime mode, it complains about latency problems:
delay of xxx usecs exceeds estimated spare time of yyy; restart ...
at the default sample rate of 48000 xxx only about 0.06 % bigger
than yyy ... on the other hand, i only get this error on sample
rates of more than 32000 ... although the sound at 32000 is
distorted, as well...
other thoughts i had:
6.) the distortion isn't affected by the block size / the number of
interrupts ...
it is affected by the samplerate ... that's why i think, there are
always 32 samples (768 bit) followed by 32 samples of silence (768
bit zero ???)
7.) the problems occured on different cardbus bridges:
ENE C1410 (ico / mandrake)
o2micro6933 (myself / gentoo)
Texas Instruments PCI1250 (timothy / red hat)
i don't know what's the reason for the problems, the latency or the
mapping problem, or if there is something else that results this problem
... but since it's a software and not a hardware issue (works fine with
windows), i'd like to get into it, and at least try to solve it ... but
since i'm neiter a kernel hacker nor i have any idea about the hardware
internals, i'd need some help from some people, who are able to help
me... i don't want some people to solve my problems for me, but i'd
appreciate any help of people, who know what's going on inside the
computer...
cheers ...
Tim mailto:TimBlechmann@gmx.de
ICQ: 96771783
--
The only people for me are the mad ones, the ones who are mad to live,
mad to talk, mad to be saved, desirous of everything at the same time,
the ones who never yawn or say a commonplace thing, but burn, burn,
burn, like fabulous yellow roman candles exploding like spiders across
the stars and in the middle you see the blue centerlight pop and
everybody goes "Awww!"
Jack Kerouac
Hello!
Has anyone experienced stuttering and very slow playback via RME
multiface, cardbus, and Alsa on a notebook, or other computer? I am
using the CCRMA low latency kernel on Fedora.
How can I fix it?
My machine has is 2.8 Pentium 4, plenty of horsepower, and two pcmcia
slots (only one works with hdsploader for some reason).
I know that I have alsa installed correctly because when I modify the
modules.conf file to use the internal sound card and speakers,
everything is very fine playing audio back via xmms, audacity,
csound(from the command line), and pd.
However, once I switch over to the hdsp in the modules.conf file,
everything runs but extremely slow... Audacity will take a minute to
launch and then will play back a 2 minute file over 5 minutes, the sound
out of the multiface is stuttering and distorted. (Very beautiful in a
twisted sort of way!)
My guess is that the problem is the pcmcia bus is somehow 2000 times
slower than it should be.. csound seems to confirm this hypothesis from
the command line when attempting realtime synthesis:
$ csound -d -o devaudio toot01.orc toot01.sco
Using default language
0dBFS level = 32767.0
Csound Version 4.23f03 (Nov 18 2003)
orchname: toot01.orc
scorename: toot01.sco
orch compiler:
32 lines read
sorting score ...
... done
Csound Version 4.23f03 (Nov 18 2003)
displays suppressed
0dBFS level = 32767.0
orch now loaded
audio buffered in 1024 sample-frame blocks
hardware buffers set to 2048 bytes
writing 2048-byte blks of shorts to devaudio
SECTION 1:
ftable 1:
new alloc for instr 1:
/dev/dsp: could not write all bytes requested
/dev/dsp: could not write all bytes requested
/dev/dsp: could not write all bytes requested
/dev/dsp: could not write all bytes requested
/dev/dsp: could not write all bytes requested
/dev/dsp: could not write all bytes requested
[1]+ Stopped csound -d -o devaudio toot01.orc toot01.sco
Thanks for the help!
-Tim
hi,
where can i download very good (acoustic and others) drum-samples?
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