John Sigler wrote:
To keep the level constant, it might need to add or remove a few
frames once in a while.
This won't work with Dolby data (except by dropping or adding a
complete Dolby frame).
Using a compressed format requires that the decoder is completely
synchronized to the source.
Bob van der Poel wrote:
But, I can't get a alsamixer to work:
bob$ alsamixer
No mixer elems found
alsamixer works; it just tells you that this device apparently has no
mixer controls.
Please show the output of lsusb -v for this device.
bob$ aplay mix.wav
aplay: main:550: audio
Sergei Steshenko wrote:
On the one hand, the original poster mentions onboard audio, on the other
it looks like a USB audio device is present.
Is this - two different soundcards - onboard + USB the case and should it
be the case ?
Or 6 channel onboard audio is USB ?
It's 8 channels, and
Thierry Bouchard wrote:
Your comment bring me another question though, why would enabling the input
monitoring on my sound card have fixed my problem and would have been a
better solution? What exactly is input monitoring?
Input monitoring means that the recorded data is played back
klondike wrote:
2008/1/8, Clemens Ladisch [EMAIL PROTECTED]:
Carlos Hernandez wrote:
I got a loopback device by doing modproble snd-aloop.
But now, I need a second loopback device. How can I up that second device?
do I need to modprobe with some parameter?
pcm_devs=2
Hi
Thierry Bouchard wrote:
I wrote a simple plugin for my microphone which is supposed to convert the
data into a 32 bps format. Here is how it looks like :
pcm.jcb-in-1 {
type hw
card 0
device 2
}
pcm.MicPlug
{
type plug
slave
{
pcm jcb-in-1
Thierry Bouchard wrote:
Im trying to capture the Microphone input and redirect it right away into
speakers. The problem is that whenever I use snd_pcm_readi
and snd_pcm_writei one after each other in my capture loop, the pipe
becomes broken on the output side, i.e. every call to snd_pcm_writei
David wrote:
I am running on a Linux kernel 2.4 (CentOS 3) and I was just trying to
compile the latest alsa drivers (1.0.16) for a Realtek ALC260 soundcard, but
when running ./configure --with-cards=hda-intel I get the following
error:
checking for which soundcards to compile driver for...
James Shatto wrote:
Is there any way to monitor arecord's progress?
In terms of time recorded, space used, and mixer levels?
Specify the -v option two or three times to see recorded levels.
Time and space aren't monitored.
HTH
Clemens
Amaury De Ganseman wrote:
I want to purchase a club3D theatron agrippa DTS 7.1.
On the website they say that the chip is CMI8770 but I don't find this
on the cmedia website and no info if it works on alsa or not... In
fact I'm confused because on the alsa website thay say it's CMI8788
and on
Janos Makadi wrote:
I have a system which has to run, with two soundcard. The old ones are
died, so I bought two Genius Soundmaker 5.1. My old kernel is 2.6.4,
and the alsa version, which cames with this kernel is 1.0.2. The two
card, was shown by the kernel, but just the first was adjustable
Stefan Thomas wrote:
I have a problem with the Edirol UM-1EX. It's a USB midi interface.
I connected it to my USB-port, but I could neither hear any sound from my
soundcard ... I use Aureon 5.1. Fun PCI as soundcard.
Your sound card does not have a MIDI synthesizer. You'll have to run
a
Justin wrote:
I'm trying to get a52 encoding to work on fedora 8.
...
What are the prereqs required to building this module?
http://www.alsa-project.org/main/index.php/A52_plugin
HTH
Clemens
-
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Martin Kemp wrote:
Does anyone know how to set up multiple instances of snd-virmidi or to
gain more than the standard 4 ports.
modprobe snd-virmidi enable=1,1,1,1
or put the options into /etc/modprobe.conf (or whatever module
configuration file your distribution uses):
Phil Carter wrote:
I'm trying to get MIDI output (through the game port) to work, but my
system locks up randomly when playing MIDI files using aplaymidi.
Everything works fine for a few seconds to a few minutes, and then my
system freezes and even the magic sysrq keys don't do anything, so I
not disclosed wrote:
I have linux install on a powerbook G4 titanium and the sound does not work.
At best I get a short
burst of sound 1-5seconds long and then nothing and the application hangs (in
this case aplay).
This bug has been fixed in ALSA 1.0.16rc2.
HTH
Clemens
* wrote:
When either my Master or PCM channel is set to zero, there is still
sound. Is there any way to calibrate the gain levels that alsa is using
for Master=0 and PCM=0?
No. These controls are implemented in hardware.
Regards,
Clemens
pramod gurav wrote:
But when i try to writei() to interface . i get the error
error : in writei : file descriptor is in bad state
I'd guess you didn't set the sample format with snd_pcm_hw_params().
HTH
Clemens
-
The Source wrote:
I'm Creative X-Fi Platinum Fatal1ty Champ1on owner and it works fine in
Vista, but alsa does not support this card so I don't have sound in
linux. I have found drivers on creative site but they are incompatible
with newer kernels (with SLUB for example). And it looks like
Nils Rennebarth wrote:
I have a creative sound blaster Audigy 2 NX USB sound card because the
internal sound card of my laptop isn't that great. The card is connected
to my Denon ADV 500SC amplifier by optical S/PDIF. This works very well
for stereo output. The amplifier however does support
Thomas Börkel wrote:
My Soundblaster 5.1 is sometimes recognized differently after boot.
Sometimes it has 223 controls, sometimes 224 controls.
...
I have saved the 2 different configurations that are being recognized
with alsactl:
http://www.boerkel.de/asound.state.1
W. Ryan Nestor wrote:
I have an Edirol UA-20 which works fine as an audio device in it's
'normal' mode, without MIDI functionality. In it's 'advanced' mode,
however, it doesn't seem to work at all.
kernel 2.6.23.11-rt14
Alsa1.0.14
proc/asound/cards: (advanced)
1 [UA20 ]:
immanuel litzroth wrote:
Quoting Clemens Ladisch [EMAIL PROTECTED]:
W. Ryan Nestor wrote:
I have an Edirol UA-20 which works fine as an audio device in it's
'normal' mode, without MIDI functionality. In it's 'advanced' mode,
however, it doesn't seem to work at all.
kernel 2.6.23.11
Quasi Steady State wrote:
Everytime the system is booting I run the command 'alsactl -F restore'.
Everything goes perfect. But if I *change the harddisk (flash-card) from the
device to another (same hardware) device I get an error during booting:*
Error if '*alsactl -F restore* ' is run I
Michelle Dupuis wrote:
My MB supports both toslink and SPDIF - from what I've read toslink has a
much higher bandwidth. Will the ALSA driver support moving more data
through toslink?
It's very likely that in your case, Toslink means optical as opposed
to the usual electrical coaxial SPDIF
Felipe Uderman wrote:
I have successfully ported the alsa library to the arm platform.
It shouldn't need porting. What did you have to change?
But I don't know what are the binaries of alsa, and where they are
located.
make install installs the binaries and the configuration files to the
Adrian McMenamin wrote:
On 10/01/2008, Clemens Ladisch [EMAIL PROTECTED] wrote:
Adrian McMenamin wrote:
$ aplay -D hw:0,0 /aine-email.wav
Playing WAVE '/aine-email.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono
aplay: set_params:895: Access type not available
This means that the device does
Adrian McMenamin wrote:
Lee Revell wrote:
You also need to find some way to confirm or deny that alsa-lib is
trying to open the right files.
Given this is nfs-root, the only realistic way is to hack the sources
and find out what the path is.
Done:
Carlos Hernandez wrote:
I got a loopback device by doing modproble snd-aloop.
But now, I need a second loopback device. How can I up that second device?
do I need to modprobe with some parameter?
pcm_devs=2
HTH
Clemens
Kim wrote:
$ cat sounds/alsa/Front_Center.wav /dev/dsp
This device doesn't support .wav files; the sample rate will be wrong.
Better try aplay. Anyway, you should get _some_ sound this way ...
== ./card0/oss_mixer ==
VOLUME Master 0
... if the volume is raised and unmuted.
HTH
Clemens
S.Çağlar Onur wrote:
I'm using ALSA 1.0.15 on 2.6.18. The MIDI device is detected and the needed
modules hotplugs itself as soon as i plug the cable, but I'm not getting any
data via the raw MIDI port or sequencer ports. Here is the output from dmesg
when plugging the device:
usb 2-1: new
Salatiel Filho wrote:
On 12/13/07, Clemens Ladisch [EMAIL PROTECTED] wrote:
Salatiel Filho wrote:
I am having a big trouble to make sound work on my ARM machine.
alsamixer: function snd_mixer_load failed: Inappropriate ioctl for device
Make sure that the kernel and the userspace tools
Salatiel Filho wrote:
I am having a big trouble to make sound work on my ARM machine.
alsamixer: function snd_mixer_load failed: Inappropriate ioctl for device
Make sure that the kernel and the userspace tools are compiled with
exactly the same architecture and other options that can affect
Giuliano Pochini wrote:
william estrada [EMAIL PROTECTED] wrote:
Is it possible to share an audio device for input? In other words, use
the same device for two running programs?
If audio device == ALSA card, yes you can if the card has more than one
ALSA device (e.g. line-in and mic).
AM wrote:
I am trying to use a LADSPA plugin (the multiband equalizer - 1197)
system-wide via the /etc/asound.conf file. I want to use my second
sound card for the output.
...
Then I tried the following which gives me an error with aplay;
defaults.pcm.rate_converter samplerate_best
Victor Hahn wrote:
I'm trying to set up ALSA to route all audio output to jackd using ALSA's JACK
plugin.
I set up /etc/asound.conf as described here:
http://alsa.opensrc.org/index.php/Jack_(plugin)
Unfortunately it doesn't work this way, I won't get any sound out of ALSA any
more. Trying
Paolo Saggese wrote:
On Thursday 22 November 2007 15:25, you wrote:
This is a valid wish, isn't it? And at least the
M-Audio Audiphile cards can sync themselves to the S/PDIF In clock.
They can, but the word clock is used only as a sample clock. It will
not reduce the amount of jitter of
Paolo Saggese wrote:
Of course I plan to use the PC only to provide a bitperfect
(exact copy of the original media, normally CD) digital stream
to an external DAC.
As you probably know better than me, the one major known problem
when you strive for the highest possible quality in digital
Rene Herman wrote:
On 22-11-07 11:10, Clemens Ladisch wrote:
An SPDIF input _always_ derives its clock from its signal.
Besides, the clock for the actual DAC has to be a multiple of the bit
clock anyway, so there must be a PLL to derive the DAC's clock from the
input signual, i.e
Rene Herman wrote:
On 21-11-07 20:33, Paolo Saggese wrote:
Thus, I would need a sound card which must be:
* cabable of bitperfect (pass through) operation at CD standard
16bit/44.1KHz (as well as, possibly, also at higher resolutions and
sample rates such as 16/48, 24/48, 24/96 and
Isaac wrote:
I'm having a lot of trouble with the Edirol UA-4FX USB/MIDI soundcard
in Ubuntu Gutsy. I saw the UA-3FX listed and hoped that the 4FX would
also work...
aplay -l
List of PLAYBACK Hardware Devices
card 1: UA4FX [UA-4FX], device 0: USB Audio [USB Audio]
It's there.
James Shatto wrote:
If I want the default alsa device to be something other than card 0,
is there a simple entry I can put in asoundrc to make that happen?
defaults.pcm.card 42
HTH
Clemens
-
This SF.net email is
Ramiro Barreiro wrote:
I have an Edirol UA-25 USB soundcard, which is my default and only
soundcard:
If there is no .asoundrc neither /etc/asound.conf aplay segfaults:
[EMAIL PROTECTED]:~$ aplay test.wav
Playing WAVE 'test.wav' : Unsigned 8 bit, Rate 22050 Hz, Mono
Segmentation fault
Huynh Phuoc Tai wrote:
Clemens Ladisch [EMAIL PROTECTED] wrote:
Does the file
/home/hptai/source/KZM/4.5/linux-2.6.16-alsa/include/linux/version.h
exist?
This file exists.
#define UTS_RELEASE 2.6.16.19-kzm-lttng-0.5.69byComputex-smp
#define LINUX_VERSION_CODE 132624
#define
Allen Kennedy wrote:
On 10/25/07, Clemens Ladisch [EMAIL PROTECTED] wrote:
Try this:
pcm.channel {
@args [ CHANNEL ]
@args.CHANNEL {
type integer
}
type plug
slave.pcm {
type dshare
ipc_key 220057
Huynh Phuoc Tai wrote:
...
checking for directory with kernel source...
/home/hptai/source/KZM/4.5/linux-2.6.16-alsa
checking for directory with kernel build...
/home/hptai/source/KZM/4.5/linux-2.6.16-alsa
checking for kernel version... 0.0.0
That directory contains only the source files
Allen Kennedy wrote:
I'm trying to get an asound.conf file set up that will allow me
to play 5 different wav files over separate channels
simultaneously, with a command like this:
#aplay -D channel1 1.wav aplay -D channel2 2.wav
aplay -D channel3 3.wav aplay -D channel4 4.wav
(please don't top-post)
Huynh Phuoc Tai wrote:
Clemens Ladisch [EMAIL PROTECTED] wrote:
Huynh Phuoc Tai wrote:
checking for directory with kernel source...
/home/hptai/source/KZM/4.5/linux-2.6.16-alsa
checking for kernel version... 0.0.0
That directory contains only the source
VoIP carrier wrote:
I need to setup a configuration that will allow me to route audio between
two virtual devices. For example, one application using ALSA will use
virtual device A's input and output, and another will use virtual device
B's. The output from A will route to the input of B,
Georg Mainik wrote:
The web site mentions a driver, so it's likely that the standard driver,
which assumes a USB audio class-compliant device, does not work.
I still could not find it. Are we talking about the same sites?
I meant Creative's web site.
Edmundo Carmona wrote:
I can get the line (Java Sound API) and send it the buffer of
information (one second) that I want to listen... however I'm not able
to listen to a single beep from the speakers.
...
This is the mixer java is using:
ICH5 [plughw:0,4]: Direct Audio Device: Intel ICH5,
Another Sillyname wrote:
I have three seperate machines outputting to a Sony STR-DB940 AC3 Amp
via SPDIF (two optical one RCA).
I can get all of them to output as standard ALSA devices from Fedora 7
setups and no specific asound.conf or asound.rc files however I
cannot for the life of me
Alexander Saydakov wrote:
This port shows garbage half of the time. Sometimes it shows right
note on/off events, but sometimes complete garbage events like channel
aftertouch (keyboard does not have any), resets or other events
unrelated to the keys I push.
Here I press the A note in the
Another Sillyname wrote:
Thanks for the response, so does that mean there is a way to get AC3
supported on the chipset using a different setup/configuration?
No.
card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 4: Intel ICH - IEC958
[Intel 82801DB-ICH4 - IEC958]
card 0: Intel [HDA Intel],
Georg Mainik wrote:
Hi,
I am trying to find out whether I would get the Soundblaster Connect
(external USB card) working.
The web site mentions a driver, so it's likely that the standard driver,
which assumes a USB audio class-compliant device, does not work.
It's possible that only small
[EMAIL PROTECTED] wrote:
I am using an m-audio Oxygen keyboard.
This has USB and a normal midi output.
The pitchbend does not work properly.
The pitchbend message for the default (middle position) should
should be data bytes 0 and 64.
Monitoring the pitch bend using the java
Alexander Saydakov wrote:
I am trying to hook up a MIDI keyboard. Here is my configuration:
Asus P4P800-E motherboard with on-board sound and MIDI/game port.
MIDI port is configured 0x330 irq 5 in BIOS setup
...
$ aseqdump -l
PortClient name Port name
...
20:0
David Bourgeois wrote:
Can you tell me which version of alsa-lib you're using? I tried the
stable/development/HG versions but your patch didn't apply straightaway as
the line numbers differ.
It should apply to the Hg version.
I patched it manually and had to add SND_PCM_FORMAT_U8 to
Per Andersson wrote:
when I try to use sound on my laptop it works fine
on the console, but when I start X11 and use programs
like sox or aplay the system hangs badly ( i need to
remove the battery to get going again ). Same thing if
sound is playing when i start X11.
The NM256 chip uses the
David Bourgeois wrote:
Or can I imagine patching dmix to add support for 8 bits
Please try this patch.
Index: alsa/alsa-lib/src/pcm/pcm_dmix.c
===
--- alsa.orig/alsa-lib/src/pcm/pcm_dmix.c 2007-07-07 11:07:05.0
David Bourgeois wrote:
I have a USB sound card that only accepts unsigned 8 bits format (a tux
droid).
That works fine with plug but I can't get dmix working as SND_PCM_FORMAT_U8 is
not in the supported formats of dmix.
Is it possible to do the mix in another format (S16_LE) then have plug
Daniel D Jones wrote:
Everything is turned up - nothing muted.
Probably some other control isn't set correctly. Please show the output
of amixer contents.
Regards,
Clemens
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Bernd Eggink wrote:
I can't get my AudioExel/MD-Mate to record from the spdif input.
How exactly are you trying to record?
Are you using the spdif device?
Even after editing /etc/asound.state and setting 'IEC958 In Select' to
'true' (and calling 'alsactl restore'), the corresponding control
michael norman wrote:
I have on of these running on OpenSUSE 10.2 using the snd-usb-audio modile as
provide by SUSE.
The only function available as far as Kmiix or Alsamixer seems to be PCM
volume.
Then that is probably the only mixer control implemented by the device.
Please show the
Aleksander Kamenik wrote:
Neither the E-MU 0404 or 0202 USB sound cards are supported in Linux.
I have samples, but work on a driver has not yet started.
They are not usb-audio standards compliant, so a new driver will have to
be written for them.
Any chance work on this driver been done?
Jan-Benedict Glaw wrote:
Fired up an alsaplayer to play some FLAC. Switched on Four Channel
Mode for a test, which nicely adds sound to the rear left/right
outputs.
After the FLAC ended, I fired up speaker-test again (and of course
switched off Four Channel Mode.) This time, it's different
Bernd Eggink wrote:
Clemens Ladisch schrieb:
Bernd Eggink wrote:
I can't get my AudioExel/MD-Mate to record from the spdif input.
How exactly are you trying to record?
Are you using the spdif device?
I had been using hw:0,2, but spdif doesn't work either. The command
arecord -f
Bill Unruh wrote:
I do not know if they have fixed the old problem in the make file. In 14.0
the script cleans out all of the snd-*.o files from
/lib/modules/version/kernel but not the .k0 files which are now more
usual in kernel module trees.
The current version removes all snd*.*o files.
leegold wrote:
[EMAIL PROTECTED]:~$ sudo modprobe snd-cmi8788
FATAL: Module snd_cmi8788 not found.
Here's the output form the install, maybe you'll see an error?
http://ljg.netfirms.com/s1.html
$ sudo ./configure --with-cards=hda-intel
This tells the configure script that no driver
James Pearson wrote:
Can anyone recommend a 'basic' PCI sound card that supports hardware mixing?
Any card for which the snd-emu10k1, snd-cs46xx or snd-ymfpci drivers are
used, i.e., most SB Live! models and cards based on CS46xx or YMF7x4
chips. None of these cards are manufactured anymore,
Jonathan Stowe wrote:
Changing the period to 2048 fixes the problem as you suggest, but I'm
not quite sure why and who if at all to report the bug to - does a
jack client need to take into account the buffer size?
Yes, jackd tells the clients what buffer size to use.
Regards,
Clemens
James Pearson wrote:
Do you happen known what card make/modules use the CS46xx or YMF7x4
chips?
CS46xx:
BlackGold II 5.1
Hercules Game Theater
Hercules Gamesurround Fortissimo III
Terratec DMX XFire 1024
Turtle Beach Santa Cruz
VideoLogic Sonic Fury
My YMF754 card is a Hoontech Soundtrack
leegold wrote:
On Wed, 12 Sep 2007 18:38:23 +0200, Clemens Ladisch
[EMAIL PROTECTED] said:
You didn't say that you're using the latest driver version,
so please download the alsa-driver-1.0.15rcX package from
http://www.alsa-project.org/, unpack it, go into the
alsa-driver-1.0.5rcX directory
Rene Herman wrote:
On 09/12/2007 05:39 PM, Clemens Ladisch wrote:
Rene Herman wrote:
Latency-wise, I expect softvol is a non-starter always but if this is not a
concern, I don't know.
Nothing needs to be scheduled separately; there shouldn't be any noticeable
latency.
Well, there _has_
Gene Heskett wrote:
On Wednesday 12 September 2007, Clemens Ladisch wrote:
The ADC is still as crappy as in earlier CMI chips, but the CMI8768's DACs
have been improved and are as good as other good 16-bit sound cards.
16 bit. Figures. at a 30 dollar bill, one doesn't get much
Jonathan Stowe wrote:
I'm getting some strange behaviour between LinuxSampler, JACK and the
USB audio driver and I'm not quite sure where to look to narrow it down
further.
What *appears* to be happening is that when LinuxSampler is being output
to JACK and JACK is using a usb audio device
Hal V. Engel wrote:
On Tuesday 11 September 2007 02:32:59 Clemens Ladisch wrote:
For example, to map /dev/dsp to hw:0,1, add the line
options snd-pcm-oss dsp_map=2
Well, there is a typo; this should have been dsp_map=1.
In /etc/modules.conf I changed
alias /dev/dsp snd-pcm-oss
Rene Herman wrote:
On 09/12/2007 03:12 PM, Gene Heskett wrote:
Grepping though the driver shows it doesn't support PCM volume, meaning
volume adjustment has to be in software (ALSA can do this) and that it
records only at 44.1/48 if that's important to you.
No wonder its quite affordable.
Denny Schierz wrote:
FATAL: Error inserting snd_serial_u16550
(/lib/modules/2.6.22-gentoo-r2/kernel/sound/drivers/snd-serial-u16550.ko):
Unknown symbol in module, or unknown parameter (see dmesg)
FATAL: Error running install command for snd_serial_u16550
snd_serial_u16550: Unknown parameter
leegold wrote:
$ sudo modprobe snd-CMI8788
FATAL: Module snd_CMI8788 not found.
The name is snd-cmi8788, not snd-CMI8788.
You didn't say that you're using the latest driver version,
so please download the alsa-driver-1.0.15rcX package from
http://www.alsa-project.org/, unpack it, go into the
Hal V. Engel wrote:
How do I get output from OSS based applications to use spdif by
default or at all?
The default mapping of OSS devices to ALSA devices is as follows:
/dev/dsp0 - hw:0,0 (first device on first card)
/dev/adsp0 - hw:0,1 (second device on first card)
/dev/dsp1 - hw:1,0
Rene Herman wrote:
On 09/08/2007 03:32 AM, James Roberts wrote:
I need to disable the creation of the ICEnsemble ICE1724 midi device on
my system. I am using a off-brand card (see below) and accessing the
midi device crashes the system hard.
Sounds like this may warrent a module
Lee Revell wrote:
On 9/6/07, Clemens Ladisch [EMAIL PROTECTED] wrote:
The snd-usb-audio has a pid parameter to specify that a device with a
certain product ID should go to a specifc index. For example,
options snd-usb-audio index=1,2 pid=0x1234,0x5678
would force the device
Rene Herman wrote:
If you load the modules with a index=N parameter, this will fix the card
at number N. Nornally, you do this by sticking
options snd-foo index=0
options snd-bar index=1
lines in /etc/modprobe.conf (or /etc/modprobe.d/whatever).
This alone doesn't work when
ATT TynTyn wrote:
Under Fedora 7 with the newest kernel, I get ~ 0.25 samples every ~
0.75 sec. on the main output.
That is, 0.25 s of sound at normal pitch, then 0.75 s of silence?
Is it different with 2/4/6 channels (the -c parameter)?
Regards,
Clemens
Jan-Benedict Glaw wrote:
On Thu, 2007-08-30 23:43:26 +0200, Jan-Benedict Glaw [EMAIL PROTECTED]
wrote:
Just out of interest, I ordered another, similar card (Ultron
Octosound 7.1, lspci: C-Media Electronics Inc CM8738 (rev 10),
/proc/asound/cards: C-Media PCI CMI8738-MC8 (model 68))
I have
James Shatto wrote:
Have a look at the Edirol UA-25. It is definitely supported.
Thanks, that looks like Exactly what I'm looking for. Out of
curiousity, since I've never used anything like these(yet), what else
does it do? Regular sound card stuff, or just a recording interface.
It does
Michael KAufmann wrote:
Is there any way to make dmix work with a52 as slave?
No. As it's currently designed, dmix requires a hw device as slave
because it needs access to the sound card's DMA buffer and interrupt for
timing purposes.
Regards,
Clemens
Jan-Benedict Glaw wrote:
Register 0x04
~
Before: 0x07 CM_ASFC_SHIFT | 0x00 CM_DSFC_SHIFT | CM_BREQ
After:0x07 CM_ASFC_SHIFT | 0x07 CM_DSFC_SHIFT | CM_BREQ
(DAC Sample Frequency changed)
Register 0x08
~
Before: CM_CHB3D5C |
Jan-Benedict Glaw wrote:
Just gave it a test in a Windows box. I did only test stereo output (I'm
not familiar with windows and didn't find a test program like
speaker-test) with this degenerated Media Player, but there was audible
output, which didn't work under Linux, probably due to the
paul blakeley wrote:
I have installed linux on a number of PCs over the last year or so.
Each PC having a different audio chipset. Recently I have installed
linux on PCs with the ALC888 and CS4299 chipset. I was able to work
out the CS4299 chipset basically downloading the chipset block
Daniel Porres wrote:
[EMAIL PROTECTED]:/proc/asound/card0$ cat codec#0
Codec: Conexant CX20551 (Waikiki)
...
Try loading the snd-hda-intel driver with the model=laptop-eapd option.
(This model should be automatically used for Toshiba P100 laptops, but
I guess you have a different one.)
HTH
TheOneKEA wrote:
On 8/30/07, Clemens Ladisch [EMAIL PROTECTED] wrote:
It does have an extra interface that is independent of the three audio
interfaces. I guess the driver uses that one to send vendor-specific
control messages.
So what else could be tried? I would really like to get
Mu Zike!! wrote:
I am new here. I upgraded my ALSA drivers about a month ago to get support
for my Edirol soundcard. I would think it is not so well supported so I am
offering for everything you could need to test it, or get information etc,
for improving the driver for this card.
The model
James Shatto wrote:
The few vendor:device numbers I've researched yielded no entries in
pciids.sf.net.
USB IDs are completely different from PCI IDs.
There is a USB device list at http://www.qbik.ch/usb/devices/.
HTH
Clemens
James Shatto wrote:
Is there a list of still produced USB based XLR3 microphone
interfaces that actually work with linux/alsa?
Tascam US-122L, US-144
I think the newer models are not supported.
Digidesign mBox2
The mBox doesn't work.
M-Audio Fast Track Pro
Has issues.
Lexicon Omega,
paul blakeley wrote:
I have successfully increased the amount of buffer by writing a new value into
/proc/asound/cardX/pcm*/sub*/prealloc.
For a mono .wav file I was able to successfully change this from 64 to
512. I was really happy with this. I presume this value is in the
units of
paul blakeley wrote:
I am writing an application that is generic enough to a number of
different drivers. If thats possible! I am trying to achieve a 2
second of buffer for a number of different sample rates. This means
that for a sample rate of 44.1KHz I require a buffer size of 88.2K
TheOneKEA wrote:
Was the lsusb output helpful in determining if the device does need a
separate telephony driver?
It does have an extra interface that is independent of the three audio
interfaces. I guess the driver uses that one to send vendor-specific
control messages.
Regards,
Clemens
Jan-Benedict Glaw wrote:
On Tue, 2007-08-28 08:28:39 +0200, Clemens Ladisch [EMAIL PROTECTED] wrote:
Jan-Benedict Glaw wrote:
[EMAIL PROTECTED]:~$ amixer cset numid=35 off
numid=35,iface=MIXER,name='IEC958 In Monitor'
; type=BOOLEAN,access=rw--,values=1
: values=on
Seems
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