Clemens Ladisch schrieb am 06.03.2015 um 08:34:
Alexandre Ratchov wrote:
On Thu, Mar 05, 2015 at 09:39:17AM -0800, Robin _ wrote:
does anyone know of a readymade open source program for 24/7/365
logging?
If JACK programs are an option: How about Rotter?
http://www.aelius.com/njh/rotter/
I
Hi Vincent,
Vincent Gulinao schrieb am 02.05.2013 19:19:
Hi everyone,
I have a task to setup a system that will continuously capture multiple
stereo signals using a MADI audio card (RME HDSPe MADI) and write them into
files (perhaps in 1 hour chunks). Few checks I've learned while googling
Hi Matt,
I didn't use a RME HDSP9632 for quite a long time (also I never used it
with the dmix plugin). However, the dmesg message sounds like a clock
source problem. All I can suggest is to check for the correct rate
settings, e.g. compare what hdspconf is showing to the output of
cat
Felix Pfeifer wrote:
Hi list,
is there a possibility to use 2 Hammerfall cards at the same time?
I would like to cascade them and use 48 adat channels for 1 application.
I heard that some people do that, but i guess they work on windows.
greets
Felix
Hi Felix,
this is definitely
Matt Snow wrote:
On Fri, Jul 10, 2009 at 2:20 AM, Clemens Ladischcladi...@googlemail.com
wrote:
Matt Snow wrote:
On Thu, Jul 9, 2009 at 12:05 AM, Clemens Ladischcladi...@googlemail.com
wrote:
Matt Snow wrote:
After upgrading to KnoppMyth/LinHES R6(ArchLinux), the analog out
plays audio
Tim Barker schrieb:
...
The card appears to be recognised by the system, the red light on the front
of the breakout box does extinguish on startup as expected (and even
extinguishes when inserting the card while the system is running, which
surprised me).
hotplug should work. this is ok.
Hi,
I've collected some additional information to my previous mail CMI9739
pcm not working: attached you find the output of
proc/asound/card0/pcm0p/sub0/status when running aplay. Before running
aplay, the status is 'closed'.
When running aplay, the only number that changes over time is
Hi,
I've set up a new PC using Mandrake 9.2. The onboard sound chip was
correctly detected and installed. The modules are loaded, the
sound-devices exist, /proc/asound/ looks ok, alsamixer is working fine,
but...
... there is no sound output from PCM. Playing back a CD through analoug
Hi,
see below...
Henri wrote:
sorry, but i'm still stuck.
mplayer doesn't work w/ -ao alsa9
aplay works w/ -D hw:0,0 and hw:0,2
alsa-init: soundcard set to spdif
alsa-init: unable to set periodsize: Invalid argument
Could not open/initialize audio device - no sound.
Audio: no sound
Video: no
Hmm, I also had some lockups with the nvidia drivers.
Download newest nvidia driver + GLX module.
Check for the correct driver and GLX version (must match you kernel).
I solved my problem by seeting the AGP mode to
Option NvAgp 0.
This works for my TNT2 M64 card, don't know if there are similar
Ede Wolf wrote:
Hello,
I have a problem with 0.9.1 and the hammerfall card (true for 0.9_rcx, too): I am only able to play back mono files. I have seen this issue has occurred before, but without any solution, so maybe there is some more experience outside there by now.
aplay -D foo.wav just
Hi,
IIRC, the phase reversal problem (causes lack of bass) was fixed with
the patch from 2002-12-09. On 2002-12-10 a patch was released with some
corrections that worked my DMX XFire, i.e. I would try
2002-12-09.tar.bz2 from cvs
plus the patch from 2002-12-10.
An additional patch (regarding
Nathan Allworth wrote:
2 things here, i finally got this thing working, using rpms.
my first problem is there is no bass like there is when im in windows,
it sounds good, just no bass, the only thing i've tested this in is
xmms, so maybe thats my problem. but i've tried fooling with the
Hi,
I don't know anything about the ibm 600e but perhaps there is no audio cable
connecting cdrom and soundcard in it. This is needed if your CDROM should do
the DA conversion (CDROM acting as audio CD player).
Some windows versions are grabbing the CD in real time via IDE and deliver
the PCM
Hi,
I am also using a cs46xx based card (Terratec XFire 1024) and haven't
seen such error messages with any alsa version I've installed so far.
I've installed the alsa rpms of several Mandrake distros as well as
several tarballs from the alsa project page. I've also never seen this
errors in
Hi,
that's a bug in rc6 version of cs46xx driver. The lack of bass is caused
by phase reversal of one channel, i.e. wrong sign of the values of one
channel. Therefore correlating sound (low frequency components are
stronger correlated because of the long wavelengths) in left and right
channel
Hi,
regarding the XFire initialization: do you reload the cs46xx module
automatically and -if so- how? I have to do
su -c '/etc/init.d/alsa restart'
manually after rebooting and would like to have this done automatically
by some change in an init script or my modules.conf but don't know how
to
Hi Pete,
start alsamixer in a shell and set the input source (hit space at the
correct item). You should also set the capture flag for the 'ADC' item
(which is the analog-digital converter) and for the 'capture' item. When
I'm recording I have the 'ADC' volume set to 100 and change the
Hi!
Hopefully, the following information is of some help.
Takashi Iwai wrote:
to be sure, could you elaborate the symptom again and the detail of
your system (kernel version, applied patches, ALSA version)?
kernel: 2.4.19-16mdkcustom, compiled with sound and without alsa. I
compiled from
Hi!
The new cs46xx driver can talk to (up to) 2 codecs on one card now .
You can see duplicate controls because your card uses 2 codecs (CS4297A
and CS4294) for multichannel playback/(recording?). Therefore, e.g. one
master controls your front output, one master controls your surround
outputs
Takashi Iwai wrote:
you can check this also via lspci command.
in the output of lspci in your last mail, the i/o ports and irq are
missing. please check again?
here we go...
00:0b.0 Multimedia audio controller: Fortemedia, Inc Xwave QS3000A
[FM801] (rev a0)
Subsystem: Fortemedia, Inc:
Takashi Iwai schrieb:
[...]
00:0b.0 Multimedia audio controller: Fortemedia, Inc Xwave QS3000A
[FM801] (rev a0)
Subsystem: Fortemedia, Inc: Unknown device 1319
Flags: bus master, medium devsel, latency 40, IRQ 10
I/O ports at ec00 [size=128]
Capabilities: [dc] Power
Adam Jones wrote:
On 12-Dec-02, Takashi Iwai wrote:
I've got a fm801 based sound card which locks my computer completely
some seconds after loading the alsa driver module.
it looks like a hardware problem rather than the driver itself.
as lspci shows, your fm801 card has no resources
Takashi Iwai wrote:
lspci -n output:
00:0b.0 Class 0401: 1319:0801 (rev a0)
00:0b.1 Class 0980: 1319:0801 (rev a0)
thanks, could you try the attached patch?
at least, the weird messages for allocation of invalid i/o ports
should disappear. not sure whether this cures the lock-up problem,
Hi!
I've got a fm801 based sound card which locks my computer completely
some seconds after loading the alsa driver module. I compiled drivers
with debug=detect -- please see my mail(s) from 12/09/2002 on alsa
userlist for more info. What else could I do? Meanwhile I found several
reports
find
some spare time).
TIA
fe
Takashi Iwai schrieb:
At Thu, 12 Dec 2002 15:45:29 +0100,
Friedrich Ewaldt wrote:
Hi!
I've got a fm801 based sound card which locks my computer completely
some seconds after loading the alsa driver module. I compiled drivers
with debug=detect -- please see my
Hi,
because the alsa drivers shipped with Mandrake 9.0 (RPMs, version
0.9.0rc2) locked my system hard when using my fm801 soundcard, I
recompiled the kernel with sound support and without alsa, compiled
alsa-dirvers, -lib and utils version 0.9.0rc6 (I have the correct kernel
sources of the
I forgot to mention the following information:
The system has an enmic/NMC board (KT133 chipset, via686a) with duron
processor. The onboard soundcard (vi82xx) is switched off in the bios.
(onboard sound works with oss driver without problems, if I switch it on)
If you need any further
Hi Benny,
I changed from Mandrake's AlsaRPMs(=0.9.0rc2) to tarballs
(0.9.0rc6) now. I noticed some (not-)changes:
* the DAC control slider now works for left right channel (it worked
only for one channel before). Thanks!
* the DSP hang problem when switching the SPDIF controls is gone! Great!
*
Hi Richard, hi Benny(perhaps you have some idea what's going on ;-))
Richard Stevens wrote:
[...]
The problem is massively distorted sound. I get the driver loaded and
there is
some sound coming out of the card. But it's distorted. If I set very low
levels for Master, PCM and DAC in
Hi,
I attached the outputs of
'cat /proc/asound/card0/*' and
'cat /proc/asound/card0/dsp/*'
before and after reloading the snd-cs46xx module. After reloading the
distortion and popping noises are gone. Does this help finding an error?
Richard Stevens wrote:
When they get loaded the second time
Hi!
Benny Sjostrand wrote:
same problem for me and my DMX XFire 1024 :-(.
I think the distortion sounds like a DC offset introduced by the PCM
channel. Do you also get popping noise when changing the PCM volume? It
just sounds like changing the supply voltage of a running amplifier for
me.
Hi!
You need to unmute the optical out. It shows up as a device with no
volume
controls or anything, all you can do is mute/unmute it.
How do I do that? It's been a while since I've fiddled around with the
settings...
Would IEC 958 Output be the SPDIF output? How would I unmute it, as
Hi!
Just out of interest... Is S/PDIF support for that card recent? (i.e.
in CVS) I inquired a couple of months ago, and was told that there
wasn't S/PDIF support... I have exactly the same card.
Yes, S/P-DIF ouput support is quite new. S/P-DIF input is even newer!
(There are some other
Hi all,
I asked this question several weeks before but did not succeed in solving my
problem :-(
is there a possibility to redirect audio output from one application
(that
outputs to some pcm device that I've defined in my .asoundrc) to a second
pcm
device so that I can record the sound
Hi,
I tried to get the copy plugin running using the following lines in my
.asoundrc:
pcm.cp_test {
type copy # Copy PCM
slave cp_slave # Slave name
}
pcm.pb {# playback device for copy plugin test
type hw
card 0
}
pcm_slave.cp_slave {
Hi all!
I've got some problems in writing a 5.1 audio stream from DVD (or a mpeg2
ripped from a DVD) to harddisk as one (or six) WAV file(s).
I tried to use a program called a52dec for this, but the audio files I get
seem to consist of 2 channels only (2ch downmix) when using the -owav option.
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