Takashi Iwai wrote:
The commit seems lacking Patch-level: ASAP although this fix is
definitely to go to 2.6.19. Could you add it to linus tree via ssh
mv, too?
Could you submit it to stable kernel tree, too?
Done.
Regards,
Clemens
Bob van der Poel wrote:
Clemens Ladisch wrote:
Bob van der Poel wrote:
How can I confirm that the line has actually taken effect?
Look into /proc/asound/timers when some MIDI is playing.
Timer for queue 0 : RTC timer
This says that the RTC timer is still used.
Regards,
Clemens
Bob van der Poel wrote:
options seq seq_default_timer_device=0
Oops, I tould you the wrong module name. This should be:
options snd-seq seq_default_timer_device=0
HTH
Clemens
-
Using Tomcat but need to do more? Need to
Jorma R wrote:
cannot submit datapipe for urb 0, error -28: not enough bandwidth
The bandwidth checking code in the kernel is known to be broken.
Please check whether CONFIG_USB_BANDWIDTH is set.
Regards,
Clemens
-
Using
Adelle Hartley wrote:
I have selected Line for capture. Using
arecord -f cd somefile.wav
produces a 2-channel wav file, but the second channel contains only silence.
Yet I can listen to the line-input (without recording) and it comes through
on both channels.
Probably the capture
Jacco Kramer wrote:
When I play a single note it is played by the synthesizer connected to it.
However, if I use a sequencer to play notes only the first one is played
(the lights on the M4U flash though, as if it was sending a midi event),
unless there's a 'silence' between two notes.
Please
Manuel Naranjo wrote:
T: Bus=04 Lev=01 Prnt=01 Port=00 Cnt=01 Dev#= 21 Spd=12 MxCh= 0
D: Ver= 1.10 Cls=00(ifc ) Sub=00 Prot=00 MxPS= 8 #Cfgs= 1
P: Vendor=0b7a ProdID=0100 Rev= 2.08
S: Manufacturer=Zeevo
S: Product=Bluetooth Audio Dongle
S: SerialNumber=00CC
C:* #Ifs= 3 Cfg#=
Berthold Höllmann wrote:
Most web pages state it does not need any special drivers,
The ion web page says that too.
but other report of driver problems.
This may be just bugs in the device's firmware.
HTH
Clemens
-
conrad berhörster wrote:
I want to capture my cdplayer output throught my onboard soundcard
line in. I have tested this with jaaa as a jack client and directly
through ALSA. The mic input is working perfect.
And what is the problem?
Did you select the Line input for recording?
Is the Line
Tom Frei wrote:
# --- ALSA configuration
alias char-major-116 snd
alias char-major-14 soundcore
alias snd-card-0 snd-cs4236
alias snd-card-1 snd-usb-audio
#--- OSS compatibility alias
alias sound-slot-0 snd-card-0
alias sound-service-0-0 snd-mixer-oss
alias sound-service-0-1 snd-seq-oss
conrad berhörster wrote:
Am Dienstag 14 November 2006 10:51 schrieben Sie:
conrad berhörster wrote:
I want to capture my cdplayer output throught my onboard soundcard
line in. I have tested this with jaaa as a jack client and directly
through ALSA. The mic input is working perfect.
Chuck Harrison wrote:
It looks to me that /dev/snd/controlC0 was opened successfully but
sound_ctl_nopdate disliked something about what it saw.
[EMAIL PROTECTED]:~# strace -eopen aplay -l
...
open(/dev/snd/controlC0, O_RDONLY)= 3
List of PLAYBACK Hardware Devices
ALSA lib
Jan Ries wrote:
I'd like to purchase a (good) usb soundcard for a laptop.
The technical specs the 'Creative Labs SoundBlaster Live 24-bit external
7.1 USB' fit my needs.
What are your needs?
I did not find a device with that name on Creative's web site.
The Live 24-bit external thingy has 5.1
Chuck Harrison wrote:
Contrary to what you expect, /dev/snd/controlC0 *does* get opened
before alsa.conf gets stat'ed or opened.
This is actually OK, that device file is also used to make sure that
the sound card driver is loaded, and this happens before the control
device is opened.
I added
Jacco Kramer wrote:
On 11/6/06, Clemens Ladisch [EMAIL PROTECTED] wrote:
It's possible that the M4U does not have an input buffer as big as it
claims it has. As a workaround, try to change line 921 or so of
usbmidi.c from
ep-max_transfer = usb_maxpacket(umidi-chip-dev, pipe, 1
Jan Ries wrote:
Am Mittwoch, 15. November 2006 13:20 schrieb Clemens Ladisch:
What are your needs?
Analog line-in/line-out (stereo)
Then you shouldn't need to get a 5.1 or 7.1 device.
However, it's very hard to find stereo USB devices because most USB
audio devices are designed
Chuck Harrison wrote:
I was being led astray by
http://www.alsa-project.org/alsa-doc/doc-php/asoundrc.php ,
which begins in no uncertain terms:
Neither of the .asoundrc or alsa.conf files are required for
ALSA to work properly.
Maybe someone should update that! ;-)
Done.
Clemens
Heinz Knoche wrote:
Without success I tried to set up my soundchip.
...
4. Is it at all possible that the soundchip's silence is related to the
sharing of an interrupt?
No.
What are the contents of /proc/asound/cards?
Regards,
Clemens
Victor Librado wrote:
I'm trying to connect a USB sound card to an ARM evaluation board. The ARM
is running a Linux 2.6.15 kernel compiled with the option of ALSA support
and USB device compiled as kernel options. Plugging the sound card to the
board causes next exception:
ALSA
(please don't top-post)
Victor Librado wrote:
I finally got the device to work. I have a look into the mailing list and
uncommented the #define IGNORE_CTL_ERROR.
Now if I start ARM with the sound card connected the card is recognised as
/dev/dsp and i can play audio. However i still got the
Tom Frei wrote:
bash-3.1# cat /proc/asound/cards
1 [tm ]: USB-Audio - Audiophile USB (tm)
M-Audio Audiophile USB (tm) at usb-:00:07.2-1,
full speed
Yea!! I cannot, however, get the M-Audio Audiophile to output any sound.
Does aplay -D plughw:1
Heinz Knoche wrote:
Clemens Ladisch schrieb:
What are the contents of /proc/asound/cards?
cat /proc/asound/cards
0 [rev60 ]: VIA8233 - VIA 823x rev60
VIA 823x rev60 with CMI9761 at 0xd000, irq 3
This codec should be supported even by your old ALSA version
Jason Martin wrote:
I've been attempting to get AC3 passthrough over iec958 to work with my abit
KN8 with NF4 chipset for some time now, and have had no success.
When using mplayer -ao alsa:device=hw=0.2 -ac hwac3 dvd:// there is nothing
but silence. Regular stereo playback over iec958 works
eli segal wrote:
I have 3 so called sound cards :
Onboard Intel ..IC .. something
Midisport 4x4 (usb midi device)
RME 9632 hdsp
every time I load the computer alsa recondnize all or some of the device
I can hear music on the onboard device and do restart and find out
the alsa deosn't
Phil Carter wrote:
dmesg tells me this every time I try to load snd-mpu401 with
modprobe:
isapnp: Scanning for PnP cards...
isapnp: No Plug Play device found
MPU-401 device not found or device busy
Apparently, your BIOS doesn't offer PP information for the MIDI port.
options snd-mpu401
sundru wrote:
Just finished a complete compile of alsa 1.0.13 but when i try to insert
into kernel it errors out with
FATAL: Error inserting snd_pcm_oss (/lib/modules/2.6.16.21-0.25-smp
/kernel/sound/acore/oss/snd-pcm-oss.ko): Unknown symbol in module, or
unknown parameter (see dmesg)
Allan Klinbail wrote:
I have only been getting a total lockup when trying to initialise the
mtpav driver for my MOTU midi timepiece (i.e. actually using it in a
sequencer), for a very long time.
Does this happen only with the snd-mtpav driver or also with other
drivers like snd-mpu401 or
Mark wrote:
I am looking for detailed information about how to use the snd_aloop loopback
driver. I guess it just creates a dummy interface that has inputs and outputs
linked together.
The code for the linking part has not yet been written.
Regards,
Clemens
Fabrice Renaud wrote:
I have an Asrock 939Dual-Vsta with C-Media CM6501 Audio chip, apparently
this chip is known to garbled 48 kHz stereo data by playing it at 96 kHz in
ALSA versions 1.0.13 or earlier
if this issue is only affecting stereo 48 Khz data would there be a
way to convert the
Robert Goodkin wrote:
For some reason alsa doesn't seem to work correctly with some apps or not
at all with others unless I run as root. I checked permissions in /dev/
and everything looks ok.
ALSA lib control.c:910:(snd_ctl_open_noupdate) Invalid CTL default:0
ALSA lib
Kiesel wrote:
Although ALSA seems to be set up fine here (playback is working
flawlessly, see below for setup details), recording does not work. arecord
does not record silence but simply blocks after writing the file header:
This usually happens when interrupts are not delivered.
Try
Tony Cantor wrote:
i have an embedded board that im trying to run alsa on, but unfortunately im
having trouble. i searched the list for similar problems and i found this:
http://sourceforge.net/mailarchive/message.php?msg_id=36898685
i have the exact same problems as this guy, however putting
Sebastian Schäfer wrote:
pcm.softvol {
type softvol
slave {
pcm hw:0,1
}
control {
name SoftMaster
}
}
But unfortunately absolutely nothing happens. In neiter alsamixer nor
amixer do I get a new control called SoftMaster.
IIRC you have to play
r10 kindsofpeople wrote:
FATAL: Module snd_serial_u16550 not found.
Do the other sound drivers exist?
Try asking the FC6 guys why they disabled this driver.
Is there a way to install and configure this driver on FC6 short of
downloading alsa-drivers and configure / make / make install ?
You
(please don't top-post)
Tony Cantor wrote:
let me just show you what my file structure looks like:
# ls
alsa.conf cards pcm sndo-mixer.alisp
# cd ../pcm
# ls
center_lfe.conf front.conf side.confsurround51.conf
does this look right?
Johnathan Bell wrote:
This just started happening. I fired up Sauerbraten (a SDL-based game), and
when it tried to access the sound card via SDL and OSS, SDL came back saying
that /dev/dsp1 was unavailable. I looked, and it didn't even exist. So I
symlink'd it to /dev/dsp and tried again. This
Johnathan Bell wrote:
On 8/8/07, Clemens Ladisch [EMAIL PROTECTED] wrote:
Please make sure that the device name configured for SDL's
ALSA output is default or plughw instead of hw.
Pardon my ignorance, but how would I do that?
Uh, not at all; it appears SDL doesn't allow to configure
Pascal wrote:
aplay can play with hw:1,0; plughw:1,0; hw:1,2 and plughw:1,2
mpd works with only plughw:1,0 all other devices are not working.
What exactly is not working?
Is there an error message when you try to use hw:1 or plughw:1,2?
Regards,
Clemens
Pascal wrote:
Clemens Ladisch [EMAIL PROTECTED] a écrit :
What exactly is not working?
mpd doesn't play the file. When I hit the play button, it stop by
itself in the same time.
Is there an error message when you try to use hw:1 or plughw:1,2?
No errors, neither in log files
[EMAIL PROTECTED] wrote:
I run alsaconf and the card is found, detected as sb16, then configured,
and levels adjusted (you may see a terminal window capture of this,
below asterisks line).
Lsmod shows the corresponding modules loaded, ok. But, the problem is,
when I reboot after
Daniel Porres wrote:
is anyone out there?
Just us chickens.
Im using alsa driver 1.0.14 on a Intel Corporation 82801G (ICH7 Family)
card, on Ubuntu Feisty.
Everything works fine but I cannot capture from a diferent source than the
microphone.
The ICH7 is a generic controller that can use
Daniel Porres wrote:
What codec are you using? (see /proc/asound/cards)
[EMAIL PROTECTED]:/proc/asound$ cat /proc/asound/cards
0 [Intel ]: HDA-Intel - HDA Intel
HDA Intel at 0xd240 irq 22
Oops, this driver doesn't show the codec name in this file.
Please
Johan De Groote wrote:
Title says it all. I got a laptop (Dell Vostro) and it has an Intel ICH8M
chipset with a Sigmatel codec and I don't get any sound out of it.
The module is either snd-intel8x0 or snd-hda-intel, depending on whether
you have an AC'97 or HDA codec.
HTH
Clemens
Maciej Łoziński wrote:
Recently I've bought a USB MIDI interface with 2 inputs and 2 outputs. I'd
like to use it in Linux, but something goes wrong with drivers. They load,
but with problems, and I can't see any MIDI device in my system.
usb1-2: new low speed usb device using ohci_hcd
Joris Huizer wrote:
I need help with alsa configuration. The problem is,
I'm not getting any sound (and I checked, on windows
sound plays just fine)
What happens when you try to run aplay something.wav?
Regards,
Clemens
Joris Huizer wrote:
--- Clemens Ladisch [EMAIL PROTECTED] wrote:
What happens when you try to run aplay something.wav?
Am just getting,
$ aplay sound.wav
Playing WAVE 'sound.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
Though no sound.
Probably some mixer control
Loris Caren wrote:
When I try 'amixer contents' (with sound modules loaded) I get error message
ALSA lib control.c:909 Invalid CTL default control default open error
Note that I'm running on a minimal embedded system
Probably the configuration files in /usr/share/alsa/ are missing.
Does
Zbigniew Baniewski wrote:
1. Does there exist an easy way to distinguish output devices from
capture devices? There's not always appropriate remark in the device's
name field.
Only the name determines whether the control is for playback or capture.
Some controls have an ambiguous
Loris Caren wrote:
On Wednesday 22 August 2007 16:07, you wrote:
Probably the configuration files in /usr/share/alsa/ are missing.
That sounds interesting, I definitely haven't put these on my embedded
system.
On my FC3 dev machine, theres quite a bit in this directory - can
anybody tell
If you have a card based on the C-Media CMI8788 chip (Asound A-8788,
Asus Xonar D2, Auzentech X-Meridian, Bgears b-Enspirer, Club3D
Theatron DTS, HTOmega Claro, Razer Barracuda AC-1, or Sondigo Inferno),
please help testing the new snd-cmi8788 driver.
To test it, get the source from the Hg
TheOneKEA wrote:
I have an Actiontec Internet Phone Wizard that I would like to use on
my Linux server, to turn it into a VoIP gateway. Unfortunately, the
snd-usb-audio driver's support of this device is incomplete and the
device does not work.
When I plug it in, the Ready light on the
paul blakeley wrote:
Can someone tell me how the alsa mixer controls map onto the audio
chipset?
This is different for every chipset (and, with AC'97 and HDA
controllers, for every mainboard).
Could you be a little more specific what your actual problem is?
Regards,
Clemens
Zbigniew Baniewski wrote:
On Wed, Aug 22, 2007 at 05:15:19PM +0200, Clemens Ladisch wrote:
TLV is additional data about the control, like dB scale information.
I'm not sure: what is mute variable for? Most controls have mute=1 -
although aren't muted
Some controls can be muted by setting
Jan-Benedict Glaw wrote:
I seem to not get correct sound output on the front and rear speakers.
Please show the output of amixer contents.
Regards,
Clemens
-
This SF.net email is sponsored by: Splunk Inc.
Still grepping
Jan-Benedict Glaw wrote:
On Mon, 2007-08-27 08:53:06 +0200, Clemens Ladisch [EMAIL PROTECTED] wrote:
Jan-Benedict Glaw wrote:
I seem to not get correct sound output on the front and rear speakers.
Please show the output of amixer contents.
numid=35,iface=MIXER,name='IEC958 In Monitor
Zbigniew Baniewski wrote:
On Mon, Aug 27, 2007 at 08:52:24AM +0200, Clemens Ladisch wrote:
Some controls can be muted by setting them to their lowest value.
Yes, I'm aware - when I set slider of playback volume to lowest value, it
will be muted.
Alsamixer shows all on/off controls
Jan-Benedict Glaw wrote:
On Mon, 2007-08-27 17:31:26 +0200, Clemens Ladisch [EMAIL PROTECTED] wrote:
Jan-Benedict Glaw wrote:
numid=35,iface=MIXER,name='IEC958 In Monitor'
: values=on
Try disabling this one. This setting causes the SPDIF input to be
copied to the analog output
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
paul blakeley wrote:
Is it possible to increase the buffer size of a ALSA driver?
Yes. No. Er, maybe.
Each driver has a certain limit for the buffer size. The ALSA framework
preallocates memory for the buffer, but usually only half the limit.
You can increase the amount by writing a new value
TheOneKEA wrote:
On 8/27/07, Clemens Ladisch [EMAIL PROTECTED] wrote:
I'd guess that the driver can send and receive audio, but that the
telephony-specific parts of the device need a separate driver.
That may be a possibility, but the way the device acts in Windows
doesn't suggest
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:
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
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
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
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:
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
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
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
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
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
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
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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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
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
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