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 4410
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
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
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
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
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
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
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
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
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
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
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
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
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
[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
[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
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
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
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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
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
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
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
---
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
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
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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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
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.
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
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
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
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
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
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
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: /cvsroot/alsa/al
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
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 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
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
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
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
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 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
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
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
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
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
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
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
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
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. */
#d
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
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
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 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
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
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
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
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
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
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.
Hello
I have an application that has many different threads.
The sound card's PCM buffer is full during playback, and one thread is
currently in snd_pcm_wait() waiting for enough space to appear in the
buffer before doing the next snd_pcm_write().
A different thread wants to flush the buffer. Th
Hello
I was wondering how easy it would be to add a classification to each
control element. (switches, volume, capture on/off etc.)
The classification would be as follows: -
1) Used during capture. I.E. Switches and volume controls for anything
one can record.
2) Used during playback. I.E. Switc
Jaroslav Kysela wrote:
>On Sat, 12 Oct 2002, James Courtier-Dutton wrote:
>
>
>
>>Currently, the state of play is that "snd_pcm_hw_params_can_pause ()"
>>should not be called until one has first done the first
>>"snd_pcm_hw_params()"
>&g
Has anyone run a diagnosis tool like "memprof" on an application that
uses alsa for audio out ?
I recently did this, and have found that alsa is a little leaky. It is
mainly the mixer part of alsa.
Here is an example backtrace for the call to the malloc that is never
freed even if the applicati
Takashi Iwai wrote:
At Fri, 11 Oct 2002 01:25:19 +1000,
James Courtier-Dutton wrote:
Hello
I have an application that has many different threads.
The sound card's PCM buffer is full during playback, and one thread is
currently in snd_pcm_wait() waiting for enough space to appear i
Thankyou, I will use snd_pcm_drop(), but as a side note, what actually
does "snd_pcm_reset()" do.
Just resetting delay to 0 does not make much sense to me.
It drops all samples in the ring buffer (thus reseting delay to 0). Note
that everybody are welcome to improve the current do
During setup of the sound card, I adjust the HW params and then save
them to the card during the open() stage.
Is there any alsa api call that can retrieve hw settings later. For
example something like: -
snd_pcm_get_buffersize()
snd_pcm_get_periodsize()
Another problem is that snd_pcm_can_pau
Currently, the state of play is that "snd_pcm_hw_params_can_pause ()"
should not be called until one has first done the first
"snd_pcm_hw_params()"
I start with
snd_pcm_hw_params_any(this->audio_fd, params);
then go about setting params, e.g.
snd_pcm_hw_params_set_access()
snd_pcm_hw_params_set_
Scott Parkerson wrote:
Has anyone else seen a kernel panic when unplugging a busy (i.e.
actively playing music) USB audio device using snd-usb-audio with Linux
2.4.19 and ALSA 0.90-rc5?
I'll send my ksymoops output in a bit... Just wondering if this is just
seen on my box.
Thanks,
Scott Par
Just as a general point to note. We are not putting the audio card into AC3 mode.
We might be doing "passthru mode" or "spdif non-audio", but never just AC3 mode.
I help with the "xine" (a free media player http://xine.sf.net) that can play DVDs.
DVDs have AC3 audio tracks and DTS audio tracks.
xin
Robert Spier wrote:
Paul,
Thank you for your clear explanation. I've submitted a small
documentation patch to the sf.net project which might prevent the
next person who comes along from falling into the same trap.
the basic problem is that you are going about this in the wrong
way
Paul Davis wrote:
I am currently taking the following approach: -
Always prepare 2 audio hardware periods of sample frames in advance
inside the user app.
1) snd_pcm_wait()
2) write()
3) prepare new sample frames, then go back to (1).
for lower latency, you'd do:
1) snd_pcm_wait()
2) pre
tomasz motylewski wrote:
Please stop the complication of "available/delay" etc. Just the raw pointer.
Each application knows where its application pointer is, so it can easily
calculate delay/available and decide for itself whether there was an overrun or
not.
I use the delay() function.
I he
the "plug" interface does make the user app easier to write, but
is using the "plug" interface adding too much overhead so as to increase
the risk of xruns too much ?
Cheers
James
Jaroslav Kysela wrote:
On Wed, 27 Nov 2002, James Courtier-Dutton wrote:
Paul Davis wrote
Bruce Paterson wrote:
Am I sending these queries about the operation of alsa through the API
to the right place ? I'm trying to use alsa for a real scientific
application and I'm starting to worry it simply isn't ready yet.
I don't pretend to be a developer of alsa drivers themselves, and I'd
Paul Davis wrote:
the APIs that are used to write almost all audio software code in
production these days all use a callback model.
Sorry for questioning this statement. Of course we all don't have any statisti
cal data but
you miss what I see as the majority of applications that use
[EMAIL PROTECTED] wrote:
>Hello,
>
>I allways have the same bug when trying to modprobe the cs4232 or cs4236
>sounddriver for my Terratec EWS654XL soundcard.
>
>The bug is called unresolved symbol, this bug exists in ALSA 0.9 since the
>existing of ALSA 0.9.
>
>I tried nearly every ALSA 0.9x versi
Takashi Iwai wrote:
thanks. looking at the codes, it seems that no special handling for
the chip. it simply sets up the ac97 registers.
the patch below is a quick hack to set the spdif rate on the first
playback pcm device. in addition, you'll need to set up the following
mixer controls:
- 'I
Takashi Iwai wrote:
oops, it seems like my mistake in the last change.
could you try the attached patch?
Takashi
That seems to have helped. We now get sliders.
We now get ac3 and dts sound with device name
iec958:AES0=0x6,AES1=0x82,AES2=0x0,AES3=0x2
and pcm with device name
iec958:AES0=0x4,AES1
Jaroslav Kysela wrote:
It is limit of current alsa-lib configuration. We cannot distinct playback
and capture. But I am not sure, if returning an error helps you
(surround40 configuration is NOT valid for emu10k1). I suggest to fix jack
to allow different names for playback and capture with diffe
Hello,
I have a problem. At some point my PC just halts, no panic message,
nothing, except that the following: -
Number lock - off
Caps lock - flashing
Scroll lock - flashing.
The only way out if the power cycle the pc.
Can anyone tell me where to start looking to track this down ?
I can't find
Orm Finnendahl wrote:
Am Freitag, den 30. Mai 2003 um 07:00:02 Uhr (-0700) schrieb Mark Knecht:
Orm,
I must say that I think this is the biggest bunch of crap I've seen
on a Linux list in a long time. This list is no place for this sort of
discussion and it's really a low act on your part to pub
Is it possible to get the OSS emulation to use the dmix alsa plugin ?
Basically, so one can get multi-open /dev/dsp on a single-open sound
hardware.
Cheers
James
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Is the SPDIF output of the SIS7012 (i810 with ac97) supported in alsa.
If so, how does one enable the SPDIF out.
The current kernel OSS module supports it.
Cheers
James
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James Courtier-Dutton wrote:
See subject.
It does not compile here.
Cheers
James
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See subject.
It does not compile here.
Cheers
James
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Alsa-devel
Takashi Iwai wrote:
At Mon, 02 Jun 2003 17:01:42 +0100,
James Courtier-Dutton wrote:
Is the SPDIF output of the SIS7012 (i810 with ac97) supported in alsa.
If so, how does one enable the SPDIF out.
The current kernel OSS module supports it.
which OSS (kernel) version supports spdif out on this
Hi,
If a user has a 5.1 sound card, but only has 2 speakers connected, it
would be nice if the user could tell alsa this fact.
Then the amount of sliders in the mixer could be reduced, and also, an
application could detect what the user has set up.
It would be nice for the user to only have to t
Hi,
I am about to start writing an alsa driver for a bluetooth headset.
Is anyone else working on this, or shall I start from scratch.
I will be using information from the kernel bluetooth modules together
with the affix project. (affix.sf.net)
Once finished I will post the source code to this li
mahendra sp wrote:
hi, alsa people,
I have wriiten the application which realises full duplex.But there is
playback buffer underrun proble. The buffer tries to read the value but
-ve value. After writink the -ve value for some timesay 3 minutes, it
becomes alright. again after some time buff
Ray Heasman wrote:
Hi,
The ALSA YMF-754 SPDIF support is incomplete in a way that ensures that
AC-3 streams can not be decoded by standards compliant receivers. Very
forgiving receivers will render the AC-3, but they are being kind.
According to the standard, they should mute.
FYI, I am using 0.9.
Jaroslav Kysela wrote:
On Sat, 5 Jul 2003, James Courtier-Dutton wrote:
Hi,
The current snd-usb-audio driver assumes that the audio device is
attached to this computer, so it only talks to this endpoint.
If I have an audio usb device with USB_CLASS = 0xe0, it will be a
wireless device. I.E. A
Hi,
The current snd-usb-audio driver assumes that the audio device is
attached to this computer, so it only talks to this endpoint.
If I have an audio usb device with USB_CLASS = 0xe0, it will be a
wireless device. I.E. A bluetooth headset.
It uses isoc's just like the current snd-usb-audio driv
I have a question regarding the callbacks in the usbaudio.c driver.
The callback is defined as: -
/*
* complete callback from data urb
*/
static void snd_complete_urb(struct urb *urb, struct pt_regs *regs)
The callback is set up with: -
u->urb->complete = snd_usb_complete_callback(snd_complete_ur
p z wrote:
Hi,
I wanted to add support for TRAM on Audigy to emu10k1 driver.
I look (tryed) at OSS driver and found that TRAM is not working
too. :-(
Then I use trial and error method and found how to setup TRAM on
Audigy. I know how to read from and write to TRAM in EMU10k2 DSP
program b
Carlo Wood wrote:
I pinpointed the problem and wrote a small program
to illustrate the problem.
When compiling the program given below with:
gcc -DUSE_DSP_SETFMT=0 troep.c
it runs fine.
But, when compiling it with:
gcc -DUSE_DSP_SETFMT=1 troep.c
It stops rather quickly with reading data.
Hi,
When an application reads the "avail" or "delay" pcm values: -
1) how accurate are they?
2) does the accuracy depend on the sound card driver being used.
The reason I ask this, is that I am going to be writing an alsa driver
for bluetooth headsets.
With bluetooth headsets, sound is sent to th
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