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
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
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
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 serious
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 finding
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
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
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,
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
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
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
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. */
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
-$(CONFIG_SND_EMU10K1X) += snd-emu10k1x.o
+obj-$(CONFIG_SND_AUDIGYLS) += snd-audigyls.o
export-objs := emu10k1_main.o
/*
* Copyright (c) by James Courtier-Dutton [EMAIL PROTECTED]
* Driver AUDIGYLS chips
*
* BUGS:
*--
*
* TODO:
*Surround and Center/LFE playback.
*Capture
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
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.
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
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
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
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
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
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
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
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:
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
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 pcm test in alsa-lib
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
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
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 anything
in source form
[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
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
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: -
Måns Rullgård wrote:
James Courtier-Dutton [EMAIL PROTECTED] writes:
snip
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 get
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
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
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
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
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
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
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
[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
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
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
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
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
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
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 configure.in
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
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
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
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)
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,
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
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 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 44100 Hz, Stereo
arecord
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().
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
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
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
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
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
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
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
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
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
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 from
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 channel
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.
It would be nice if alsa-lib would
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
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
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
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 saying
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 aren't
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
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,
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
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
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
00:0e.0 Multimedia audio controller: Aureal Semiconductor Vortex 1 (rev 02)
00:0e.0 Class 0401: 12eb:0001 (rev 02)
Cheers
James
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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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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
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:
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
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
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
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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 willing
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
Francisco Moraes wrote:
From what I can see for the emu10k1x (aka. sb Live 5.1 from Dell) is
almost exactly the same as the Audigy LS.
It is a very simple device, and the driver should be based more like
the snd-intel8x0 driver than the snd-emu10k1 driver.
It has no dsp, and no hardware
Bastien Aracil wrote:
At Mon, 16 Feb 2004 18:39:46 -0500,
Francisco Moraes wrote:
I am trying to get the Emu10k1x chip supported in alsa. Anyone willing
to give me a hand, please email me. I've got a few register dumps
and I am trying to get it working.
I'm will be pleased to help you. I got
Takashi Iwai wrote:
At Thu, 05 Feb 2004 22:28:03 +,
James Courtier-Dutton wrote:
I checked out a new anonymouse cvs of alsa-driver/alsa-kernel, applied
your patch, and attach the output from doing modprobe snd-intel8x0.
I can only see printf's in the patch, so I don't see how it could have
Jaroslav Kysela wrote:
On Thu, 5 Feb 2004, Glenn Maynard wrote:
On Thu, Feb 05, 2004 at 12:01:28PM +0100, Jaroslav Kysela wrote:
My point is, I don't think setting start_threshold to buffer_size is
even wrong at all. Some people might want the buffer to be full before
it starts, and my patch
Takashi Iwai wrote:
At Wed, 04 Feb 2004 23:03:50 +,
James Courtier-Dutton wrote:
Takashi Iwai wrote:
At Mon, 02 Feb 2004 19:42:37 +,
James Courtier-Dutton wrote:
Once thing I have noticed, is that with the alc650, we used to have VRA
(alsa 0.9.8), but the 1.0.2 intel8x0 driver ignores
Did you some tests with xine and 1.0.2 libraries? I tried to fix all
problems related to xine there (including the last reported problem from
you).
The delay values might be affected by this issue: The dmix plugin can
start in the middle of period of the master (hw:x device). In this case,
Takashi Iwai wrote:
At Mon, 02 Feb 2004 19:42:37 +,
James Courtier-Dutton wrote:
Once thing I have noticed, is that with the alc650, we used to have VRA
(alsa 0.9.8), but the 1.0.2 intel8x0 driver ignores the VRA and fixes
itself at 48000.
yes, the detection of sample rate range seems
See attached file.
It contains a test program, and a readme.txt with details on how to
configure the dmix device in order to carry out the test.
Basically, the value of snd_pcm_delay() returned when using the dmix
device is wrong.
See readme.txt in delay-test-0.0.1.tar.bz2 for details.
Cheers
Jaroslav Kysela wrote:
On Mon, 2 Feb 2004, James Courtier-Dutton wrote:
James Courtier-Dutton wrote:
Once thing I have noticed, is that with the alc650, we used to have VRA
(alsa 0.9.8), but the 1.0.2 intel8x0 driver ignores the VRA and fixes
itself at 48000.
So, before I never needed
In alsa 0.9.8, the snd-intel8x0 worked find with my MB. An ICH5 with ALC650.
With alsa 1.0.2, I get: -
device: front - no-sound output
device: surround40: - Only sound on rear left, and rear right.
device: surround51: - Only sound on rear left, and rear right, center, lfe
So, in all cases there is
Takashi Iwai wrote:
At Mon, 02 Feb 2004 17:41:02 +,
James Courtier-Dutton wrote:
In alsa 0.9.8, the snd-intel8x0 worked find with my MB. An ICH5 with ALC650.
With alsa 1.0.2, I get: -
device: front - no-sound output
device: surround40: - Only sound on rear left, and rear right.
device
James Courtier-Dutton wrote:
Once thing I have noticed, is that with the alc650, we used to have VRA
(alsa 0.9.8), but the 1.0.2 intel8x0 driver ignores the VRA and fixes
itself at 48000.
So, before I never needed the sample rate converters, but as it now
fixes the rate at 48000, I need
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