Russell King wrote:
But unfortunately I don't have the driver code myself to be able to
comment, so its probably been fscked.
If the code was posted publically, the author of the code would get a
lot more useful help from more eyes.
---
This SF.
Roc Wu wrote:
Unable to find an usable access for 'default'
aplay: set_params:832: Sample format non
available
Yes. Thanks for your replay. Maybe I should send the
mail to arm-linux mailist.
PS. Could you recommend some docs about the ALSA
internals and Low level drivers? There are too many
docs i
William wrote:
James Courtier-Dutton wrote:
William wrote:
James Courtier-Dutton wrote:
Audigy LS driver is now ready for inclusion into alsa-driver.
Get it from http://www.superbug.demon.co.uk/alsa
Are you going to work on any other Audigy-series drivers, e.g. Audigy 2 ZS?
I'm findin
William wrote:
James Courtier-Dutton wrote:
Audigy LS driver is now ready for inclusion into alsa-driver.
Get it from http://www.superbug.demon.co.uk/alsa
Are you going to work on any other Audigy-series drivers, e.g. Audigy 2 ZS?
I'm finding the emu10k1 driver in alsa-driver-1.0.5a has se
Audigy LS driver is now ready for inclusion into alsa-driver.
Get it from http://www.superbug.demon.co.uk/alsa
The "install.txt" tells one how to patch the current alsa-driver 1.0.5a
with it, and also explains where some other files should be put.
The indentation might need correcting before incl
Zack Borschuk wrote:
I was wondering if a petition asking Creative Labs to release the needed
information to create efficient support for the Audigy LS would be
plausible or not. Please let me know, so that if it is plausible, I
could start working on getting the petition signed by enough peop
Hi,
I have an Audigy2 el cheepo edition, and an Audigy LS.
The Audigy2 luckily has a separate digital output jack, so I can easily
connect a mono or stereo jack into the Audigy2, have an RCA plug on the
other end, and plug it into an external AC3 decoder, and AC3 passthru works.
The Audigy LS ha
ALSA driver available from:
http://www.superbug.demon.co.uk/alsa/
* FEATURES currently supported:
*Front, Rear and Center/LFE.
*Surround40 and Surround51.
*Capture from MIC input.
*
* BUGS:
*--
*
* TODO:
*Need to add a way to select capture source.
*4 Capture channels, on
Hi,
I am trying to do work on the Audigy LS driver.
I have now discovered that I can send sound to the Front, Rear and
Center/LFE.
I have not found out how to set the amount of interleaved channels that
the sound card can do, so it is fixed at 2 channels per stream.
The sound card has 4 voices f
A linux user, Greg Turpin (from Colorado, USA), kindly donated an Audigy
LS to me, so that I could try to provide an Audigy LS driver to the ALSA
project.
As you all know, I have already been relatively successful and now have
sound coming from the Front Speakers, and hope to improve support soo
The Audigy2 has lots of registers for the emu10k2 chip.
These are accessed by programming
#define PTR 0x00 /* Indexed register set pointer register*/
/* NOTE: The CHANNELNUM and ADDRESS words can */
/* be modified independently of each other. */
#d
Takashi Iwai wrote:
At Thu, 27 May 2004 20:17:17 +0100,
James Courtier-Dutton wrote:
Here is my first go at Audigy LS support.
It can play sound to the front speakers.
great!
/* hardware definition */
static snd_pcm_hardware_t snd_audigyls_playback_hw = {
.info = (SNDRV_PCM_INFO_MMAP
nd-emu10k1x-objs := emu10k1x.o
+snd-audigyls-objs := audigyls.o
obj-$(CONFIG_SND_EMU10K1X) += snd-emu10k1x.o
+obj-$(CONFIG_SND_AUDIGYLS) += snd-audigyls.o
export-objs := emu10k1_main.o
/*
* Copyright (c) by James Courtier-Dutton <[EMAIL PROTECTED]>
* Driver AUDIGYLS chi
Here is a status update.
I have received an Audigy LS sound card which was kindly donated by Greg
Turpin.
So, far, I have managed to get some sound out of the Front speakers. My
speaker-test program outputs a nice constant tone.
When playing in xine, it is not perfect sound, it stops and starts
Nicolas Hüppelshäuser wrote:
Hi!
I'm using alsa 1.0.4. How can I get a larger alsa buffer size? The
maximum size returned by snd_pcm_hw_params_get_buffer_size_max is too
small. I found no module option and nothing apropriate within the driver
code. Mainly I'm using the snd-usb-audio module but
Peter Zubaj wrote:
Hi,
If you think Line In on back side of card then:
try to unmute and set volume for
Line In
Analog Mix (best 100 %)
Front (best 100 %)
Master (best 100 %)
I am sure that it works on audigy 1.
Peter Zubaj
It needs the "Line In" playback slider turned up.
It also needs the "Analo
i have found out that on the Audigy2. and someone else has found the
same with the Audigy1, the line in jack does not work.
I expect this is just a matter of finding the correct FX bus for it, but
wanted to check here, to see if anyone else has got it working first,
before I take a look.
Cheer
Thomas Witzel wrote:
What needs to be done in order to improve the Audigy 2 support ? If there is
anything that can be done by just implementing things that are not
implemented because of lack in human resources then I wouldn't mind
volunteering with some of my time.
Getting some datasheets on
Would it be a good idea to keep this page more up to date.
e.g.
SB Audigy 2, but mention that it only works in 16bit 48Khz mode.
Possible add the SB Audigy 2 ZS.
Dell SB Live! Value.
I keep getting people asking me if the Audigy 2 is supported, because it
is not on the list.
Also enter in the lis
I cannot find any documentation on any of the following functions in mixer.h
I want to create a function that takes the elem, and comes back and
tells me if it is used for playback, or capture, and thus allow me to
filter the display of mixer elements based on whether they are used for
capture o
You sent me an email off list, but when I reply to it, the reply fails
with the messages below.
So, I cannot email you direct with any responses to your off list emails.
Due to standard etiquette, I cannot respond to off list emails on list.
Can you please fix the problems your end.
Cheers
James
Takashi Iwai wrote:
At Tue, 18 May 2004 16:47:01 +0100,
James Courtier-Dutton wrote:
What would I need to change in the emu10k1 driver, to get alsa-lib to
send it 32bit audio samples.
I tried just adding the SNDRV_PCM_FMTBIT_S32_LE to the playback options,
but that did not work.
When I did that
What would I need to change in the emu10k1 driver, to get alsa-lib to
send it 32bit audio samples.
I tried just adding the SNDRV_PCM_FMTBIT_S32_LE to the playback options,
but that did not work.
When I did that, everything just played at half speed.
Can anyone give me any pointers as to where el
Manuel Jander wrote:
As the main author of the Aureal Vortex driver, its very stupid having
to handle arbitrary period sizes, introducing a lot of overhead and
complexity in the driver, while the hardware just is not designed to
handle period sizes that are not powers of two, due to page boundary
o
I understand the the Audigy 2 DSP (emu10k2) handles 24bits in the DSP at
a max rate of 48Khz. It uses an as yet undocumented extra chip for
96/192 Khz.
Does anyone know how we could get the alsa emu10k1 live/audigy driver to
accept 24bit audio from the application.
At the moment, when I send al
James Courtier-Dutton wrote:
I have been doing some tests with 24bit audio.
It seems that any 24bit stream is muted if one tries to send it to
alsa-lib.
I am going to do further tests, but I just wanted to see if anyone else
has ever tried 24bit sound with the Audigy2?
Cheers
James
Please
Ronald S. Bultje wrote:
Hi,
for both my ALI 5451 and my Audigy 2 NX, snd_pcm_delay() sometimes
returns (in the second argument pointer) ridiculously high values in the
range of 2^32/bytes_per_sample (in my case, 16bitLE/stereo, that comes
down to roughly 1,1E9). I'm guessing there's some kind of
si
I attach a patch that updates the speaker-test program.
It corrects a few minor bugs as well as add a new -s option.
Diff is with the alsa-utils cvs.
Cheers
James
Index: alsa-utils/speaker-test/readme.txt
===
RCS file: /cvsroot/alsa/al
I have been doing some tests with 24bit audio.
It seems that any 24bit stream is muted if one tries to send it to alsa-lib.
I am going to do further tests, but I just wanted to see if anyone else
has ever tried 24bit sound with the Audigy2?
Cheers
James
---
Takashi Iwai wrote:
At Fri, 14 May 2004 18:14:47 +0100,
James Courtier-Dutton wrote:
I would like to add some information that might help people modifying
this for the Audigy LS.
The outputs for the card work in 2 modes.
1) Probably analogue on the output jacks.
snd_emu10k1x_ptr_write(chip, 0x41
Takashi Iwai wrote:
At Wed, 12 May 2004 03:16:14 +0100,
James Courtier-Dutton wrote:
[EMAIL PROTECTED] wrote:
Here's the first pass at the driver. I've tested it mainly with XMMS with the ALSA output plugin.
alsaplayer didn't work, not sure why. I've also tested with the
Giuliano Pochini wrote:
On Wed, 12 May 2004, James Courtier-Dutton wrote:
Just for general info.
pci10b5,1142 NOT-HANDLED "lynxone"
Add the LynxTWO, Lynx L22 and Lynx AES16. LynxStudio provides programming
info only under a NDA that doesn't allow the licencee to release any
Hello,
Does anyone have an Audigy LS lying around that they are not using, and
would like to donate it to me, so I can add support for it in ALSA ?
I just need a card to try out some Kung Fu on it.
Cheers
James
---
This SF.Net email is sponsor
Just for general info.
This list is a lot longer than I expected.
This list only includes PCI devices.
This list only includes devices that ALSA does NOT support currently.
PCI AUDIO DEVICES handled by OSS but not ALSA.
pci-id, comment, oss binary module name.
pci4005,308 NOT-HANDLED "als300"
pci10
[EMAIL PROTECTED] wrote:
Here's the first pass at the driver. I've tested it mainly with XMMS with the ALSA output plugin.
alsaplayer didn't work, not sure why. I've also tested with the pcm test in alsa-lib which seems to be jumping, so that's another problem.
I've removed the joystick support f
I have been helping some people with writing alsa drivers.
One thing that they did not totally understand from the alsa
documentation was the concept of frames.
To help with this, could we add a html link between in the following
document: -
http://www.alsa-project.org/~iwai/writing-an-alsa-drive
Patrick Shirkey wrote:
Tim Blechmann wrote:
Just wondering if anyone has found time to look into fixing the input
quality of the usb quattro?
I'm not sure it has ever functioned perfectly for input quality.
Currently I get a better sound quality from my $10 cmipci than my $400
usb-quattro ;-P
Paul Davis wrote:
Kernel people: is poll() less effective than using SND_PCM_ASYNC and a
signal handler for low-latency sound? I'm guessing it is, but there
at the risk of endlessly repeating myself, SIGIO is basically
useless. your handler executes in signal-handling context, and can do
very, v
Måns Rullgård wrote:
James Courtier-Dutton <[EMAIL PROTECTED]> writes:
The OP was recording.
Oops. My explanation only covers playback, not capture.
But if you use the snd_pcm_avail_update() just before you capture some
samples from the buffer, then do a gettimeofday(), you will
Juan Carlos Granda wrote:
That's my app does:
1.- Open the device for capture
2.- Set the access mode SND_PCM_ACCESS_RW_INTERLEAVED
3.- Set format 16 bits (SND_PCM_FORMAT_S16_LE)
4.- Set channels 2 (stereo)
5.- Set buffer time near 1 second
6.- Set period time near 0.1 seconds
7.- Copy the hardwar
Hi,
I have been updating the wine alsa driver to work better with alsa.
So far, all I have done is update it to use the new alsa api.
Windows uses an api called Direct Sound.
Direct Sound uses direct hardware buffer access.
A Win32 program can quiry the sound driver and ask for the currently
play
I just received an email from sigmatel.com, so I thought I would pass on
relevent information.
The STAC9758 datasheet has been made public and can be attained at:
http://www.sigmatel.com/products/technical_docs.htm#9758
Cheers
James
---
This SF.
Just for everyone's information.
I now have all the info I need to get the Rear channel working.
I thank Voluspa for giving me ssh root access so that I could find out
the details via trial and error.
But the summary is: -
This tells the emu10k1 chip which slots to fill on the AC-LINK between
th
Which section of code in alsa-lib is doing these conversions.
I would like to see what code you use for the task of converting samples
in float format to samples in int format.
James
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Get certifi
Christian Ege wrote:
Hi anyone working on something simalar to this?
http://www.dcs.gla.ac.uk/~jp/snd-bt-sco/
The author of this driver won't continue his work because he switched to MAC OS
;-(
I think the way he tried to integrate this driver into alsa isn't the best way.
I am not that kind of
Clemens Ladisch wrote:
ns wrote:
I have WDM driver for our lab's own propertiary sound
card(ADAT+SPDIF+DB...) and wanna write ALSA drv.
I have linux w/kernel 2.4.20 (original).
I know that I must patch kernel for ALSA and write ALSA driver.
HOw to do it fastest?
look into alsa-kernel/Documentati
Bug #232 has been incorrectly closed, but I can't reopen it because I
cannot [ Add Bugnote ]. I click on the [ Add Bugnote ] link, but nothing
happens.
Summary: That sound card still does not work correctly even after using
the latest cvs which was supposed to fix the bug.
Cheers
James
---
[EMAIL PROTECTED] wrote:
Hi,
i need some clarifying about how the defines ALSA_PCM_OLD_HW_PARAMS_API
etc. behaves.
for alsa-1.x its clear that if the app uses the new api i have to define
ALSA_PCM_NEW_HW_PARAMS_API, but what is for the case if i have an app with
the new api and the user uses onl
Brian Furey wrote:
Hi James,
Im asking you becuae looking thru the archives you asked a similar
question a long time ago.
i'm using an open source VoIP application with the
alsa driver. My card is the onboard intel8x0.
My problem is figuring out the patterns I am
getting with the alsa driver
Does anyone have a list of sound cards that alsa does not support.
It would be nice to have a list of sound cards that alsa does not
support, and next to it, details of anyone currently working on writing
a driver.
We could start a list on the wiki at http://alsa.opensrc.org/
My initial questio
bash-2.05b#arecord
RIFF$WAVEfmt @dataRecording WAVE 'stdout' : Unsigned 8 bit, Rate 8000
Hz, Mono
bash-2.05b#
Thats it. No error message, nothing recorded, just immeadiately exits.
also: -
bash-2.05b# arecord -Dhw:0
RIFF$WAVEfmt @dataRecording WAVE 'stdout' : Unsigned 8 bit, Rate 8000
Hz, Mono
a
Glenn Maynard wrote:
On Tue, Apr 06, 2004 at 05:04:34PM -0400, Paul Davis wrote:
I can't find any way to detect the running ALSA version, for diagnostic
cat /proc/asound/version
That's the driver version, which I'm already logging. Like I said, I
want the alsa-lib version that's being linked in
Pavana Sharma wrote:
Hi,
I am running a test application on my alsa driver for arm platform.Driver
is statically built.
The test application calls the snd_pcm_open.
With device hw:0,0
At the target I have created sound driver files with snddevices.sh
In /pro
alsa-lib contains a test program pcm.c that tests playback using all the
different modes alsa-lib can do.
Can we have a similar application for capture.
When developing an alsa driver, it works fine with arecord, but fails
with jackd, and the only difference is that arecord just uses
snd_pcm_rea
Jaroslav Kysela wrote:
On Wed, 31 Mar 2004, James Courtier-Dutton wrote:
Is there any reason why this patch was not added to the alsa-lib cvs ?
It's better to put this information to configure.in?
No, because it selects which version of aclocal etc. that are used,
which is before configu
Is there any reason why this patch was not added to the alsa-lib cvs ?
James Courtier-Dutton wrote:
James Courtier-Dutton wrote:
I attach the output I see on the screen when running ./cvscompile.
Cheers
James
I attach a patch to fix the problem for me
Russell King wrote:
On Wed, Mar 31, 2004 at 11:22:56AM +0200, Jaroslav Kysela wrote:
On Wed, 31 Mar 2004, Russell King wrote:
I suggest we add a load of preprocessor junk into the ALSA core and
comment exactly _why_ its needed, thereby laying the reason completely
at the door of these ill-define
Brian Furey wrote:
Hi all,
I have an intel810 onboard soundcard.I am using the
alsa driver with a VoIP session.
The intel8x0.c file has a minimum period byte size
of 32 bytes with the minimum no. of periods being
1.The min and max rate is set to 48k.
How can I find out what actual(runtime) si
guan yim wrote:
FYI too, the Audigy 2001 I have been talking about is using the
following chips:
SIGMATEL
STAC9721T
LC5A01E
0201
CREATIVE
Audigy(tm)
CA0100-IDF
(C) CREATIVE TECH'01
2BA70KW
I can have 6 channels output with this card. I think if creative uses
the SIGMATEL chip on that SBLive 2003
I want a sound card to work in full duplex.
I also want the in and out directions in sample sync.
I.E. The playback period size is 160. I have my code polling, so that it
is executed once every 160 samples.
How do I ensure that each time my code executed, there will be at least
160 samples ready
Extract from ./sound/pci/emu10k1/emupcm.c
static unsigned int capture_period_sizes[31] = {
384,448,512,640,
384*2, 448*2, 512*2, 640*2,
384*4, 448*4, 512*4, 640*4,
384*8, 448*8, 512*8, 640*8,
384*16, 448*16, 512*16, 640*16,
384*3
Jaroslav Kysela wrote:
On Thu, 25 Mar 2004, James Courtier-Dutton wrote:
bash-2.05b# arecord -fcd -Dplug:duplex | aplay -Dplug:duplex
Recording WAVE 'stdout' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
Playing WAVE 'stdin' : Signed 16 bit Little Endian, Rate 4410
See attached patch for suggested new pcm device.
It works for: -
arecord -fdat -Dplug:duplex | aplay -Dplug:duplex
bash-2.05b# arecord -fdat -Dplug:duplex | aplay -Dplug:duplex
Recording WAVE 'stdout' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo
Playing WAVE 'stdin' : Signed 16 bit Little
Jaroslav Kysela wrote:
On Wed, 24 Mar 2004, Paul Davis wrote:
open pcm, and get a handle.
snd_pcm_poll_descriptors(handle, &pfd, err);
Get a poll file scriptor in pfd.
select(nfds, rfds, wfds, efds, tvp);
Is it possible to use this call with alsa ?
select is generally deprecated in linux (lin
open pcm, and get a handle.
snd_pcm_poll_descriptors(handle, &pfd, err);
Get a poll file scriptor in pfd.
select(nfds, rfds, wfds, efds, tvp);
Is it possible to use this call with alsa ?
It seems that the select functions as expected with the descriptor so
that we can do a snd_pcm_writei().
I
James Courtier-Dutton wrote:
I attach the output I see on the screen when running ./cvscompile.
Cheers
James
I attach a patch to fix the problem for me.
--- cvscompile 2002-10-24 13:09:30.0 +0100
+++ cvscompile.new 2004-03-22 16:58:07.072241360 +
@@ -1,4 +1,9 @@
#!/bin/bash
Jaroslav Kysela wrote:
So, when is a PCM ready?
If a PCM is already in SND_PCM_STATE_RUNNING, when is snd_pcm_wait()
supposed to return ?
When avail >= avail_min.
1) Does this depend on period size in any way?
For example, if period size is 6000 frames, and I set avail_min to 2000
frames, will
I attach the output I see on the screen when running ./cvscompile.
Cheers
James
Script started on Mon Mar 22 16:50:52 2004
sh-2.05b# ./cvscompile
automake-1.5: configure.in: installing `./install-sh'
automake-1.5: configure.in: installing `./mkinstalldirs'
automake-1.5: configure.in: installing `.
Pavana Sharma wrote:
Hello,
I am trying to export the controls to user space. I want to know the
complete list of controls
which ALSA expects the codec to support. For example, few are as below,
Master Playback Volume
Master Playback Switch
Tone Control - Bass
Tone Control - Treble
Line Capture V
David Lloyd wrote:
On Thu, 18 Mar 2004, James Courtier-Dutton wrote:
What is the conclusion regarding fopen/fclose/fwrite/fread. Can it be
done?
I thought that one rather pie-in-the-sky idea might be to use a kernel
module that made /dev/dsp, /dev/mixer, etc., and reflects back to a
userspace
I need more details on exactly what snd_pcm_wait() is supposed to do.
The documentation on the www.alsa-project.org gives: -
Wait for a PCM to become ready.
Parameters:
pcm PCM handle
timeout maximum time in milliseconds to wait
Returns:
a positive value on success otherwise
Joerg Mayer wrote:
On Fri, Mar 19, 2004 at 08:30:13PM +, James Courtier-Dutton wrote:
The guys there just don't understand what you told them. If they took
the time to actually read what you said, they could easily fix their
problem.
Summary: -
No bug in alsa.
Summary 2:
There seem
Lars Heineken wrote:
I used the latest alsa-tarballs to upgrade alsa-lib, alsa-driver and
alsa-oss.
To make it short: The problem is even worse now !
This is the output from strace (relevant part):
.
.
0.000531 ioctl(4, SNDCTL_DSP_RESET, 0) = 0
0.000232 ioctl(4, SNDCTL_DSP_GE
Jaroslav Kysela wrote:
On Fri, 19 Mar 2004, James Courtier-Dutton wrote:
--- You are receiving this mail because: ---
You are a voter for the bug, or are watching someone who is.
http://bugs.kde.org/show_bug.cgi?id=76413
CCMAIL: 76413 bugs kde org
CCMAIL: 70802 bugs kde org
Florian Schmidt wrote:
He sounds pissed.. :) If you get pissed by these forwards, please tell
me.. I figured they might be of interest..
Begin forwarded message:
Date: 19 Mar 2004 15:05:31 -
From: Allan Sandfeld <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: [Bug 76413] arts does not foll
Jaroslav Kysela wrote:
Hi all,
I would like to point all developers to the OSS API redirector
which is in our alsa-oss package. It's universal piece of code which can
redirect all OSS API calls (actually mixer & PCM API only) to any shared
library.
Pros:
- no more LD_PRELOAD hacks
- any libra
Takashi Iwai wrote:
there are test codes in alsa-lib/tests directory.
also James Courtier-Dutton wrote some neat test programs (i'd like to
include them too).
Takashi
Although you have not asked me directly, feel free to include my test
programs. I assume you are talking about the ones
Jaroslav Kysela wrote:
On Thu, 11 Mar 2004, James Courtier-Dutton wrote:
Which devices can a mono stream be played on?
It seems that the only device a mono stream can be played on is the
"default" pcm device.
Devices like "front" and "rear" are stereo only devices
Which devices can a mono stream be played on?
It seems that the only device a mono stream can be played on is the
"default" pcm device.
Devices like "front" and "rear" are stereo only devices.
It would be nice if alsa-lib would open one of these devices, and if the
application tries to set 1 cha
Ove Kaaven wrote:
Well, the requirements that raised this thread should be fairly clear.
For example,
ALTERNATIVE 1
snd_pcm_set_volume(snd_pcm_t* pcm, int volume)
and
snd_pcm_set_pan(snd_pcm_t* pcm, int pan)
using whatever value range makes the most sense, and perhaps some query
on whether th
I would like to add that this patch has been tested by someone on #alsa
in irc.freenode.net and confirmed as a fix.
Please add this to the alsa cvs.
Cheers
James
Tommy Schultz Lassen wrote:
Hi
I have been having som problems with mixer settings on my Sound Blaster
Extigy.
I am using the alsa fr
Takashi Iwai wrote:
Hi,
it seems that some mobo with ALC650 uses GPIO 0 as the mic bias +5V.
in ac97_patch.c, the GPIO 0 is turned on/off in conjunction with
the mic/center sharing switch, but this handling appears only for the
old ALC650 revision (D or older).
interestingly, there is a report th
William wrote:
James Courtier-Dutton <[EMAIL PROTECTED]> wrote:
Use directions at: -
http://alsa.opensrc.org/index.php?page=AlsaBuild2.6
Thanks, James. Like I said in my previous reply to you, the section
describing option 1 is confusing because it seems to contradict itself
firstly by
William wrote:
Jaroslav wrote on alsa-project.org:
"simply copy files from the ALSA's alsa-kernel CVS module to relevant
locations in the 2.6 kernel tree."
This is one method of upgrading to CVS ALSA.
However, using this method with Linux 2.6.3 and current CVS alsa-kernel
gives errors:
Use d
Frank Barknecht wrote:
Hallo,
James Courtier-Dutton hat gesagt: // James Courtier-Dutton wrote:
You will be lucky to find any sound card complying with the USB Audio
spec, as the spec is written so badly.
This is the USB spec I'm referring, not the USB AUDIO spec. Those
M-Audio devices a
Frank Barknecht wrote:
Hallo,
I'd like to propose changing the status of three M-Audio USB
devices in the soundcard matrix to "unsupported on 2.6". The Quattro,
Audiophile USB and reportedly the Duo all don't comply to the USB
specification, as has been discussed here and on linux-usb-dev in the
r
Looks like a Sound Blaster compatible.
Try
modprobe snd-sb16 enable=1 isapnp=0 port=0x0220 mpu_port=0x330 irq=5
dma8=1 dma16=5
Cheers
James
Tom Watson wrote:
Ah, a new card to try...
This time it is an Ensonic ESS-1869 that is built into an older Compaq
laptop (Armada 3500). When I put in the "s
Accidentally install alsa-lib 0.9.6 before alsa-utils, instead of
alsa-lib-1.0.3rc2.
Compiles and installs fine now.
Sorry
James
James Courtier-Dutton wrote:
Jaroslav Kysela wrote:
Hello all,
I released 1.0.3rc2 packages. The full changelog from 1.0.2 will
came with the final release
Jaroslav Kysela wrote:
Hello all,
I released 1.0.3rc2 packages. The full changelog from 1.0.2 will
came with the final release, but it would be nice to do some tests with
this code with smaller number of testers to not follow the 1.0.2 situation
when we have to quickly release several versions
James Courtier-Dutton wrote:
00:0e.0 Multimedia audio controller: Aureal Semiconductor Vortex 1 (rev 02)
00:0e.0 Class 0401: 12eb:0001 (rev 02)
Cheers
James
It is not in alsa-kernel, only in alsa-driver!
Cheers
James
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00:0e.0 Multimedia audio controller: Aureal Semiconductor Vortex 1 (rev 02)
00:0e.0 Class 0401: 12eb:0001 (rev 02)
Cheers
James
---
SF.Net is sponsored by: Speed Start Your Linux Apps Now.
Build and deploy apps & Web services for Linux with
a free
Paul Davis wrote:
The ideal scheduler for realtime apps would be one that has an api that
allows for a call like "schedule me at exactly 10ms intervals+-1ms".
no, thats not true.
the system clock does not run in sync with the sample clock. the drift
in this would become noticeable in a few minu
Adam Tla/lka wrote:
sigh. of course! because the kernel has no idea that your audio
application needs to run with real-time priority, and is instead
treating all apps as if they are normal interactive programs. if you
tell the kernel that your app needs to run with RT priority (there are
So why I
Will wrote:
Clemens Ladisch <[EMAIL PROTECTED]> wrote:
Did you change asequencer.h in both the kernel and alsa-lib?
No, I didn't see my script had actually failed to patch
include/sound/asequencer.h
BTW I know there are several correct ways of updating the ALSA in Linux 2.6.x
according to the do
Adam Tla/lka wrote:
nice but many people just haven't this hardware and want to use
normal PCI sound cards or even matherboard build in codecs
and mix many applications PCM sound together, use MIDI (software
emulated or not) without need of special configuring of aplications.
VirtualMixer, InputMul
Adds some info to /usr/src/linux/include/sound/emu10k1.h
Cheers
James
--- emu10k1.h.org 2004-02-25 01:08:35.129501584 +
+++ emu10k1.h 2004-02-25 01:26:34.617394480 +
@@ -644,9 +644,13 @@
#define SOLEH 0x5d /* Stop on loop enable high register */
#define SPBYPASS 0x5e /* SPDIF BYP
Jaroslav Kysela wrote:
On Tue, 24 Feb 2004, James Courtier-Dutton wrote:
http://www.alsa-project.org/black.html
How can this list be NONE!
We need to add Creative for their SB Live Dell edition (emu10k1x) and
the SB Audigy LS.
Well, have we a response from Creative that they're not wi
http://www.alsa-project.org/black.html
How can this list be NONE!
We need to add Creative for their SB Live Dell edition (emu10k1x) and
the SB Audigy LS.
Cheers
James
---
SF.Net is sponsored by: Speed Start Your Linux Apps Now.
Build and deplo
Jaroslav Kysela wrote:
On Tue, 24 Feb 2004, James Courtier-Dutton wrote:
Jaroslav Kysela wrote:
http://www.opensound.com/cuckoo.html
:-) No comment, except that some comments are really wrong.
Jaroslav
Hehe! ;-)
Is there a list anywhere listing the differences between OSS and ALSA
Jaroslav Kysela wrote:
http://www.opensound.com/cuckoo.html
:-) No comment, except that some comments are really wrong.
Jaroslav
Hehe! ;-)
Is there a list anywhere listing the differences between OSS and ALSA
with regard to sound card hardware.
It would be nice to have a nice small list
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