In follow up to this.
The problem seems to be the failure to properly expand:
snd_runtime_check macro in include/driver.h
Cheers
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of James
Courtier-Dutton
Sent: 18 November 2001 21:00
To: [EMAIL
Hello
I use ac3dec which comes with alsa as a very nice example on how to program
the alsa-lib for PCM out.
Is there a similar program showing how to use MMAP output ?
I would like to update the xine media player to use MMAP.
Cheers
James
--
Nothing in this world is exactly what it appears to
Will that mean that alsa-driver will be in the kernel.
Where will alsa-lib, alsa-utils and alsa-tools go ?
Cheers
James
___
Alsa-devel mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-devel
I don't know all about mmap, but why does one need to poll.
I would have thought that a callback with info on how many samples it wants
would be a better way.
Cheers
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Paul Davis
Sent: 28 November
Hello
With the following card: -
YMF744 - Yamaha DS-XG PCI (YMF744) (ymfpci driver)
Why does: -
tmp = snd_pcm_hw_params_set_period_size_near(this-audio_fd, params,
period_size, 0);
give tmp=257 no matter what value you give period_size ?
Is the period_size fixed on this card?
Cheers
James
If hardware manufactures wanted their products to have good support on
Linux, all they have to do is publish the hardware programming details, and
the linux community will do the actual driver development.
I still don't understand what manufactures are protecting by not releasing
the programming
What is wrong with using a PC ?
Use all 8 bits of the parallel port in DMA mode.
The make an external circuit which will buffer and then serial clock the 8
bits out.
If this is not quick enough, use more parallel ports.
It is probably a good idea to use a real time OS, and not linux.
Linux
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Jaroslav
Kysela
Sent: 29 November 2001 21:25
To: James Courtier-Dutton
Cc: [EMAIL PROTECTED]
Subject: Re: [Alsa-devel] YMF744 - Yamaha DS-XG PCI (YMF744) (ymfpci
driver) Bugs
This would probably be best done with the people at
mailto:[EMAIL PROTECTED]
(http://opensource.creative.com)
They already have sound coming out of the audigy, so it would be better to
only ask for stuff the emu10k1-devel people don't already know about.
Cheers
James
-Original
I think only few cards support the detection of non-audio data format
on hardware. And as mentioned before, on some chips, the stream is
resampled on other rates..
Takashi
Can anyone tell me which cards do NOT do resampling on SPDIF IN so Recording
AC3 works.
Cheers
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Takashi Iwai
Sent: 04 December 2001 09:31
To: Joshua Jacobs
Cc: Dan Hollis; [EMAIL PROTECTED]
Subject: Re: [Alsa-devel] linux sound card w/ digital output
Hi Josh,
At Mon, 03 Dec 2001 14:59:38
To: James Courtier-Dutton
Cc: [EMAIL PROTECTED]
Subject: Re: [Alsa-devel] linux sound card w/ digital output
At Tue, 4 Dec 2001 10:24:51 -,
James Courtier-Dutton wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of
Takashi Iwai
One thing I would like to know about the emu10k1 (SB Live).
I noticed a register in the emu10k1 which has
#define SPBYPASS0x5e/* SPDIF BYPASS mode register
*/
#define SPBYPASS_ENABLE 0x0001 /* Enable SPDIF bypass mode
Hello
I would like to use the sound card as a clock source for a DVD player.
I have a SB Live with alsa drivers 0.9.x.
Which API call can I make to the sound card which is equivalent to the
gettimeofday call?
Also, what will be the units of time received from the call?
Cheers
James
--
-Original Message-
From: Jaroslav Kysela [mailto:[EMAIL PROTECTED]]
Sent: 04 December 2001 13:07
To: James Courtier-Dutton
Cc: Ricardo Colon; [EMAIL PROTECTED]
Subject: RE: [Alsa-devel] linux sound card w/ digital output
On Mon, 3 Dec 2001, James Courtier-Dutton wrote:
Sort
When I see -22 returned, it normally means wrongly installed drivers.
Most users who report -22 errors, find the solution is a re-install of alsa
fixes it.
This is what I have found from users of xine (http://xine.sf.net) which has
alsa support.
Cheers
James
-Original Message-
From:
I personally think that PCM audio is not easy to deal with, when one needs
to do effects on it etc.
PCM is only a representation of the sound, why do we HAVE to use it?
It seems to me that a much better format for storing and manipulating sound
it is the in frequency density domain.
After all,
I originally was trying to talk about sound formats like AC3 and DTS which
store sound in small packets of FFT results, instead of the equivalent which
would be small groups of PCM samples. So people could do processing on FFT
sound sources without having to push them to PCM before they can be
To: James Courtier-Dutton
Cc: [EMAIL PROTECTED]
Subject: Re: [Alsa-devel] Alsa and the future of sound on Linux
I have looked at the Jack web page (http://jackit.sourceforge.net/)
It would help more if jack.h had more documentation for all api function,
and not just a few of them.
well, we're
Hello
I enclose a patch to alsa09, which fixes the problems I have been having
regarding no AC3 output from the SPDIF out.
I don't know if anyone remembers, but I have been having problems with my SB
Live card and AC3 out for a long time.
I could get AC3 passthru working with the
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Takashi Iwai
Sent: 01 February 2002 19:20
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: [Alsa-devel] Audigy support status
Hi,
snip
- digital non-audio i/o - not implemented
no TRAM
I would try to use the 0.9.x API if you can.
Although 0.9.x is beta, the api is mostly stable, and has better features.
Cheers
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Gerald
Grabner
Sent: 31 January 2002 22:19
To: [EMAIL PROTECTED]
Hello
I am thinking of writing a low level sound card driver for alsa.
The card in question is the dxr3 hardware DVD player, which has analogue and
SPDIF out.
It currently works fine with OSS drivers.
The card has no mixers, or recording function, just write and GETOPTR,
SETSPEED etc function in
Excellent news.
I am an opensource developer. My main project at the moment is xine,
http://xine.sf.net and dvdnav http://dvd.sf.net.
The main requirement from xine is to get audio and video in perfect sync
with each other. Alsa provides for that requirement very well.
I would like to say that
Another nice feature of getting alsa into the kernel is that the code will
then be checked with the Stanford checker.
This is an intelligent checker which checks for bad pointers and generally
tries to catch bugs in code even before they show themselves.
It has generally improved the quality of
As ALSA is now in the linux kernel, should the HOWTO be part of the linux
documents HOWTO pages ?
Cheers
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Patrick
Shirkey
Sent: 15 February 2002 07:35
To: Paul Davis
Cc: [EMAIL PROTECTED]
Could this be a devfs (device filesystem) problem ?
With devfs, devices appear as if by magic in the /dev directory.
The old system had all devices having entries in /dev whether present or
not.
Cheers
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On
I do this by counting how many frames have already been written(from the
app), minus the frames still waiting to be written (from alsa, i.e. How many
frames are still in the sound cards buffer waiting to be output.)
Cheers
James
-Original Message-
From: [EMAIL PROTECTED]
Any AC3 or DTS takes some time to decode even with an external decoder.
The nice thing about them is that the standard tell us that they are fixed
latency decoders.
So, AC3 decoder will take a fixed amount of time to take a digital SPDIF in
and convert it to 6 analogue out's for your speakers.
the cabling
as well.
Cheers
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Bob Ham
Sent: 18 February 2002 14:51
To: James Courtier-Dutton
Cc: [EMAIL PROTECTED]
Subject: RE: [Alsa-devel] dolby digital output
On Sun, 2002-02-17 at 18:44, James
The magic web site did not mention anything about an AMP.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Bob Ham
Sent: 18 February 2002 16:27
To: James Courtier-Dutton
Cc: ALSA Development Mailing List
Subject: RE: [Alsa-devel] dolby digital
I think it is very interesting hearing what peoples views are when comparing
current digital audio technologies like SPDIF and packet based technologies
like ethernet for digital audio.
Very few people have in depth knowledge of what is possible with both
technology.
Some people have in depth
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Scott Jonathan
Sent: 25 February 2002 02:39
To: [EMAIL PROTECTED]
Subject: [Alsa-devel] hello?
I have several times tried to erase my email address from this
list, but it
keeps telling me my
Hello
AC3 passthru on the SB Live does not seem to work on all platforms with the
latest alsa-cvs. (26-2-2002)
It works on my system, which is a SB Live Rev 6, which only started working
recently.
It does not work on a SB Live rev 7 (I don't have this, but others have
reported.)
Can anyone
a dma_mask (with pci_set_dma_mask) of 29 bits (512Mib).
Also: -
29 bit (512MiB), mostly speculation/trial-and-error though. See main.c and
the pci_set_dma_mask() calls.
Does this help alsa09 at all ?
Cheers
James
-Original Message-
From: James Courtier-Dutton [mailto:[EMAIL PROTECTED]]
Sent
-Original Message-
From: Paul Davis [mailto:[EMAIL PROTECTED]]
Sent: 06 March 2002 17:27
To: James Courtier-Dutton
Cc: Ulrich Zadow; [EMAIL PROTECTED]
Subject: Re: [Alsa-devel] Timer?
The delay value is calculated from getting the amount of samples still in
the sound buffer
As a programmer, alsa is just much easier to use.
alsa provides a consistent api interface for all sound cards thanks to the
alsa-lib. (shared user space code)
So, if I write an app for my sound card, I can be pretty sure the app will
work on all other sound cards.
With OSS/Free, hardly any of
Why not just change the app to support alsa 0.9.x.
alsa 0.9.x is now in kernel 2.5.x, so the app should really move to
supporting it anyway.
Cheers
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Xavier
ALLAMIGEON
Sent: 12 March 2002 21:11
try
alsaplayer -Dsurround40 foo.mp3
That works on a sb live.
Cheers
James
-Original Message-
From: Adam K Kirchhoff [mailto:[EMAIL PROTECTED]]
Sent: 13 March 2002 14:26
To: James Courtier-Dutton
Cc: Xavier ALLAMIGEON; [EMAIL PROTECTED]
Subject: RE: [Alsa-devel] ymfpci : four
It is a shame that this person is really missing the point of what the open
source community really wants.
Good technical documentation, so that the open source DSP audio engineers
around the world can write a good 46xx driver for Linux.
We really don't need cirrus audio engineers, we just need
Leif Lindholm wrote:
Hello
I'm working on a set-top-box based on the National Semiconductor SC1200
Geode integrated microprocessor.
National have been nice enough to write an Alsa-driver for the built-in
AC97-controller/audio thingy - unfortunetaly it is written for 0.5.10b.
So now I need to
Leif Lindholm wrote:
On Fri, 2002-04-05 at 04:00, James Courtier-Dutton wrote:
I would post the source code of your current driver somewhere (URL).
If you still think so after reading this mail (especially license stuff
at end of mail), I could post a diff somewehere.
If the set top box uses
Eric Dantan Rzewnicki wrote:
James Courtier-Dutton wrote:
Leif Lindholm wrote:
On Fri, 2002-04-05 at 04:00, James Courtier-Dutton wrote:
I guess you might be out of luck then, as there is currently no
documentation on how to write sound card drivers for alsa09.
I think the only way you can
Takashi Iwai wrote:
Hi James,
At Thu, 04 Apr 2002 21:55:50 +0100,
James Courtier-Dutton wrote:
Hello
I have just managed to get alsa09 sound to work on my portable after a
lot of difficulty with neither alsa or oss modules loading.
I would like to suggest that we start adding an alsa09 howto
Takashi Iwai wrote:
At 04 Apr 2002 15:41:36 +0200,
Leif Lindholm wrote:
Hello
I'm working on a set-top-box based on the National Semiconductor SC1200
Geode integrated microprocessor.
National have been nice enough to write an Alsa-driver for the built-in
AC97-controller/audio thingy -
Takashi Iwai wrote:
Update of /cvsroot/alsa/alsa-kernel/pci/emu10k1
In directory usw-pr-cvs1:/tmp/cvs-serv9192
Modified Files:
emu10k1_main.c
Log Message:
fixed default mask values
(these change won't affect the actual behavior, though)
Index: emu10k1_main.c
Patrick Shirkey wrote:
Forgot the crucial part :)
http://www.alsa-project.org/index-new.html
--
Patrick Shirkey - Boost Hardware Ltd
For the discerning hardware connoisseur
Http://www.boosthardware.com
Http://www.boosthardware.com/LAU/Linux_Audio_Users_Guide/
It would be even nicer if that
Hello
I just got the latest alsa 0.9.x cvs (as of about 5 mins ago)
Everything compiles fine, except alsa-conf
I enclose stdout and stderr output from the ./cvscompile command.
Cheers
James
`/usr/share/automake/install-sh' - `./install-sh'
`/usr/share/automake/ylwrap' - `./ylwrap'
James Courtier-Dutton wrote:
Hello
Are there any test applications which test the pause/resume
functionallity.
I have an app which is calling pause, then resume, but after resume,
the sound is very choppy.
I was hoping that there was an application out there which already
worked
Markus Plail wrote:
Hi Takashi!
* Takashi Iwai writes:
Markus, could you try the latest cvs version, which fixes the real
behavior, whether digital out works? if it doesn't work again, then
definitely GPOUT0 is the spot.
It doesn't work with latest CVS.
looking at OSS source
Takashi Iwai wrote:
There are several different versions of the SB Live.
My SB Live rev06 is not a SB Live 5.1, so it has 2 audio stereo output
jacks, one for front, one for rear, and then a spdif jack. The spdif
jack is not switchable between analogue and digital. It is fixed at digital.
Markus Plail wrote:
Hi James!
* James Courtier-Dutton writes:
I think this rev07 passthru issue might have been the user just not
knowing about the Digital/Analogue switch in alsamixer.
NO! I have been knowing about it all the time. And as I wrote to Takashi
this had/has nothing
Joern Nettingsmeier wrote:
James Courtier-Dutton wrote:
...
I think we should have a section on the web site for the different
applications which support different alsa versions.
N. could we please please please annihilate every last little
trace of alsa 0.5.x from that page
Maarten de Boer wrote:
Hello,
Updating my fltk alsamixer frontend, alsamixergui, I once again
found that alsamixer (and therefore alsamixergui) shows a lot
of sliders. With my Audigy, it's really too much. I think it
would be very nice to be able to configure alsamixer, to show
only a subset of
Paul Davis wrote:
Not very much:
a) if 32 bit integer could be enough expressive, what's the problem to
truncate it in your driver?
if i knew that this was true, i would do so. i don't know that. the
rms meters are the *only* registers on the h-dsp that are 64 bit, so i
assume there is
Has anyone managed to get alsa-conf to compile ?
I use the current CVS, but it will not compile.
Is alsa-conf of any use any more ?
Cheers
James
---
Sponsored by:
ThinkGeek at http://www.ThinkGeek.com/
Hello
It would be nice if there was a feature in alsa that would let the user
tell alsa which speakers are plugged into which sound card sockets.
E.g. On sound card 0, I have: -
Left speaker in Stereo Jack 1-Left
Right speaker in Stereo Jack 1-Right
Rear Left in Stereo Jack 2 - Left
Rear Right
Robert Robinson wrote:
Takashi,
I had communication from Kysela Jaroslav indicating that the Audiophile 2496
driver does support digital I/O.
The problem may be that I have not been using an audio interface program
that supports the digital option. This choice is never displayed as part of
the
There are other applications/tools you can use, one of which is JACK
that allows for multiple audio streams mixed in software and lots of
other cool stuff.
Cool, I'll see if I can find some information about that. I hadn't
heard of it before... is it GPL'd?
Rob
Hello
Can someone please explain to me the use of snd_pcm_status_get_delay()
If a buffer size is 16384, and avail=6384, should delay therefore equil
1 ? In all cases ?
I have found that in SND_PCM_STATE_RUNNING, delay = buffer_size - avail.
I have found that in SND_PCM_STATE_PREPARED,
Clemens Ladisch wrote:
The behaviour of polling during capture is just fine:
RUNNING: block until avail_min is available, then return POLLIN
DRAINING: return POLLIN until buffer is empty, then return POLLERR
(other states: POLLERR)
The current behaviour for playback is:
PREPARED: return
Anders Torger wrote:
On Wednesday 09 October 2002 15.28, you wrote:
Clemens Ladisch wrote:
The behaviour of polling during capture is just fine:
RUNNING: block until avail_min is available, then return POLLIN
DRAINING: return POLLIN until buffer is empty, then return POLLERR
(other
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.
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.
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()
I start with
snd_pcm_hw_params_any(this-audio_fd, params);
then go
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
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
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()
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
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.
xine
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
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)
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
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:
the APIs
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 version.
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:
-
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
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
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
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
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
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
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
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 =
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
Takashi Iwai wrote:
At Fri, 11 Jul 2003 17:28:08 +0200,
Giuliano Pochini wrote:
On Thu, 10 Jul 2003 16:31:28 +0100
James Courtier-Dutton [EMAIL PROTECTED] wrote:
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
Takashi Iwai wrote:
At Mon, 21 Jul 2003 00:40:32 +0100,
James Courtier-Dutton wrote:
Hi,
Is there any way to tell if a USB sound card has been disconnected or not?
IIRC, this is not notified to the application.
the app simply would get an error at the further access.
it sounds not bad to notify
Gregoire Favre wrote:
Hello list ;-)
I have a stupid 5.1 system from Hercules which don't work perfectly in
the sense, that when I connect it to a SBlive with a coaxial connexion
it works (I can hear lots of noise, but my cable is very bad), my
receiver goes in Dolby Digital mode with Dolby
pony wang wrote:
What I want to do?
I want to play a wave file (22050Hz, 2-ch) via SPDIF channel.
Whats my problem?
I think the ICH4 audio controller cant convert PCM data of 22050Hz to
48KHz.
And the Codec cant either (codec generally accept 3 sample rates: 32K,
44.1K,
Gregoire Favre wrote:
On Thu, Aug 14, 2003 at 06:05:48PM +0200, Takashi Iwai wrote:
The problem I was having was that the spdif non-audio bit was not
being set.
This patch fixes that problem.
thanks, now applied to cvs.
Is there any hope to do the same for the cs46xx?
Thank you very much,
I have a SB live audio card.
I can open it with device names like: -
surround40 (Front Left, Front Right, Rear Left, Rear Right)
surround51 (Front Left, Front Right, Rear Left, Rear Right, Center, LFE)
There is a requirement for: -
surround41 (Front Left, Front Right, Rear Left, Rear Right, LFE)
Takashi Iwai wrote:
At Tue, 12 Aug 2003 00:19:34 +0100,
James Courtier-Dutton wrote:
I have a SB live audio card.
I can open it with device names like: -
surround40 (Front Left, Front Right, Rear Left, Rear Right)
surround51 (Front Left, Front Right, Rear Left, Rear Right, Center, LFE
Takashi Iwai wrote:
At Wed, 21 May 2003 08:04:44 +0200,
p z wrote:
Hi,
I tested code and it doesn't work :-(.
When I deleted code to detect ALC650 rev.E, it worked perfect. :-)
May be:
1) MSI uses ALC650 rev.E and stil uses gpio0 to switch mic power
on/off
2) detection code for ALC650 rev.E
I have a Motherboard with the ICH5 and ALC650 so I use the intel8x0 alsa
kernel driver. (Currently the version from 2.6.0-test4.)
I have managed to get SPDIF passthru working. (I posted the patch some
time ago, and it is now in 2.6.0-test4)
I cannot get multichannel working.
The motherboard has
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