Hi guys,
I have an GA-890FXA-UD5 mobo with an intel-hda sound card. I've
purchased an additional SPDIF In adapter (bracket) to be able to play
digital content on my 5.1 speakers. I have the speakers connected to the
jacks (ie. not to the SPDIF Out). I've checked the Windows drivers, but
these
On Wed, Aug 18, 2010 at 4:25 AM, Gustaf Johansson gusta...@gmail.com wrote:
I get no audio when using digital out from my Nvidia nf4 CK804 ALC850 chip.
The card is listed as 2 devices, analog using the first and digital the
second.
However i never get audio when playing using the second
On Thu, Aug 19, 2010 at 8:31 AM, Niels Mayer nielsma...@gmail.com wrote:
On Wed, Aug 18, 2010 at 4:25 AM, Gustaf Johansson gusta...@gmail.com wrote:
I get no audio when using digital out from my Nvidia nf4 CK804 ALC850 chip.
The card is listed as 2 devices, analog using the first and digital
Hi,
I get no audio when using digital out from my Nvidia nf4 CK804 ALC850 chip.
The card is listed as 2 devices, analog using the first and digital the second.
However i never get audio when playing using the second device.
Ports spdif and iec958 is using the 2nd device (Hardware PCM card 0
BTW, here is the related bug:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4904, which
includes the alsa-info output with the laptop docked.
Thanks!
On Thu, 2010-02-04 at 12:12 +0100, Javier Ruiz wrote:
Hi!
I have a Dell Precision M90 laptop with Gentoo Linux installed and with
Hi!
I have a Dell Precision M90 laptop with Gentoo Linux installed and with
a Dell Dport Docking station (this:
http://www.laptoppartsshop.com/dell-precision-m90-laptop-parts/dell-d-port-port-replicator-with-130w-ac-adapter-p50201.html).
The laptop has the souncard “Audio device: Intel
Hi All,
I've installed ALSA 1.0.19 on a fresh install of Mythbuntu 8.10 on two
seperate machines, and I am trying to get digital audio pass through to
work properly. One machine with an ALC888 soundcard with SPDIF output, and
the other with a VT1708B soundcard with HDMI output. I have digital
SPDIF output is 2-channel PCM only. To get 5.1 output you need to
send your receiver a passthrough AC3 or DTS stream, or enable Dolby
ProLogic II on your receiver to simulate the surround effect.
You won't get the system to send 6 channels of raw PCM via S/PDIF,
it's only capable of
On Fri Feb 20 10:51 , Steve Steffler sent:
SPDIF output is 2-channel PCM only. To get 5.1 output you need to
send your receiver a passthrough AC3 or DTS stream, or enable Dolby
ProLogic II on your receiver to simulate the surround effect.
You won't get the system to send 6 channels of
Hi
I have a strange problem and coudn't find any help so far:
Every time I start an audio stream, alsa is muting spdif (only spdif)
on my soundblaster card. In Amarok I can play the whole playlist
without muting. But if I click on a song in the playlist, spdif gets
muted. If I run alsactl restore
Jason Tackaberry wrote:
I am testing with mplayer, and ensuring the digital device (as shown by
aplay -l) is being used. In my case it's device 1, and I am specifying
-ao alsa:device=hw=0.1 on the mplayer command line.
Try spdif instead of hw:0,1.
HTH
Clemens
On Thu, 2008-09-25 at 13:51 +0200, Clemens Ladisch wrote:
Try spdif instead of hw:0,1.
Indeed, yes, this has helped. Thanks. I was somehow under the
impression that spdif was merely an alias for the digital device, but I
now see I was mistaken.
I've been running into some intermittent
Hi,
I recently built a system with an Asus P5Q-E motherboard, whose product
specifications indicate the audio chipset is ADI AD2000B, and which ALSA
says is AD1989B. The driver is snd_hda_intel. (I am not passing any
module options; passing model=6stack-dig seems to have no effect.)
The analog
At Tue, 09 Sep 2008 22:53:50 +0200,
Ruediger Dohmhardt wrote:
Vedran Miletić schrieb:
Why don't you just pull the entire Takashi's tree with git and compile
it? That would probably require the least effort, because you won't
need to do any patching at all.
Ok! I did it. *The good
Vedran Miletić schrieb:
Why don't you just pull the entire Takashi's tree with git and compile
it? That would probably require the least effort, because you won't
need to do any patching at all.
Ok! I did it. *The good news*:
Building sound-unstable-2.6.git with kernel 2.6.26.3 made the
Sorry!
In my last e-mail there was a mistake in the
installation description of sound-unstable-2.6.git
and the formatting was terrible.
/*** Installation of sound-unstable-2.6.git ***/
cd linux-2.6.26.3
mv sound sound.org /* hide original sound
Why don't you just pull the entire Takashi's tree with git and compile
it? That would probably require the least effort, because you won't
need to do any patching at all.
2008/9/8 Ruediger Dohmhardt [EMAIL PROTECTED]:
Vedran Miletić schrieb:
It should work with hda-intel of Takashi's
Vedran Miletić schrieb:
There is another option, cheaper than Xonar DX. Creative has relaeased
PCI Express version of Xtreme Audio, which isn't based on X-Fi chip
but on CA0110. There is some support in Takashi's sound-unstable tree.
I'm not sure how well it works (let alone in regard to SPDIF
It should work with hda-intel of Takashi's sound-unstable-2.6 tree (it
has the patch_ca0110.c file there). Have you tried compiling that?
2008/9/7 Ruediger Dohmhardt [EMAIL PROTECTED]:
Vedran Miletić schrieb:
There is another option, cheaper than Xonar DX. Creative has relaeased
PCI Express
There is another option, cheaper than Xonar DX. Creative has relaeased
PCI Express version of Xtreme Audio, which isn't based on X-Fi chip
but on CA0110. There is some support in Takashi's sound-unstable tree.
I'm not sure how well it works (let alone in regard to SPDIF support),
hopefully Takashi
Jason Gauthier wrote:
I can't find many cards except for the Creative Labs X-Fi that seems to
fit both.. and from what I read SPDIF passthrough doesn't work.I'm
not 100% sure what that means. (Pretty new to this level of audio)
SPDIF was designed to transport two channels of 16-bit
Jason Gauthier wrote:
I can't find many cards except for the Creative Labs X-Fi that seems
to
fit both.. and from what I read SPDIF passthrough doesn't work.I'm
not 100% sure what that means. (Pretty new to this level of audio)
SPDIF was designed to transport two channels of 16-bit
Lee Revell wrote:
On Fri, May 2, 2008 at 5:20 AM, Bart de Boer [EMAIL PROTECTED] wrote:
Hi all,
I got it to work! :D I was telling MythTV to send everything to my digital
output directly. This was wrong. I needed to send everything to my analog
output and let the chip do the work. I'm
Dominique Dumont wrote:
Bart de Boer [EMAIL PROTECTED] writes:
That sounds plausible. :) I use the via82xx driver. Should I post my
case in the alsa-devel list?
I think so. Try to prove your case by playing with alsamixer.
If anyone knows any workarounds I'd like to know about
Matt Garman wrote:
On Fri, May 02, 2008 at 11:20:28AM +0200, Bart de Boer wrote:
I got it to work! :D I was telling MythTV to send everything to my
digital output directly. This was wrong. I needed to send
everything to my analog output and let the chip do the work. I'm
now playing music
Dominique Dumont wrote:
Bart de Boer [EMAIL PROTECTED] writes:
It seems I may have cheered too soon. Yes, I am able to get
bitperfect 44.1 kHz sound and bitperfect 48 kHz sound (with and
without AC3). Mplayer is somehow able to trigger the system into 48
kHz after listening to 44.1 kHz
On Fri, May 2, 2008 at 5:20 AM, Bart de Boer [EMAIL PROTECTED] wrote:
Hi all,
I got it to work! :D I was telling MythTV to send everything to my digital
output directly. This was wrong. I needed to send everything to my analog
output and let the chip do the work. I'm now playing music at
Hi all,
I got it to work! :D I was telling MythTV to send everything to my
digital output directly. This was wrong. I needed to send everything to
my analog output and let the chip do the work. I'm now playing music at
44.1 kHz and movies at 48 kHz. :)
Thanx anyway,
Bart
Bart de Boer
On Fri, May 02, 2008 at 11:20:28AM +0200, Bart de Boer wrote:
I got it to work! :D I was telling MythTV to send everything to my
digital output directly. This was wrong. I needed to send
everything to my analog output and let the chip do the work. I'm
now playing music at 44.1 kHz and movies
Dear list,
I'm having trouble passing pure 44.1 kHz audio through SPDIF properly.
I'm able to pass it through. But the sound gets sped up to 48 kHz making
it sound as if the pitch is turned up. I know alsa enables me to upmix
all sound to 48 kHz. But that causes too much quality loss. I want
On Thu, 1 May 2008, Bart de Boer wrote:
Dear list,
I'm having trouble passing pure 44.1 kHz audio through SPDIF properly.
I'm able to pass it through. But the sound gets sped up to 48 kHz making
it sound as if the pitch is turned up. I know alsa enables me to upmix
all sound to 48 kHz. But
On Thu, May 1, 2008 at 8:42 AM, Bart de Boer [EMAIL PROTECTED] wrote:
SNIP
Complete output of /proc/asound/card0/codec97#0/ac97#0-0:
0-0/0: Realtek ALC650F
PCI Subsys Vendor: 0x1509
PCI Subsys Device: 0x9202
Capabilities :
DAC resolution : 20-bit
ADC
Bill Unruh wrote:
According to Realtek that chip is capable of producing 32, 44.1 and 48
kHz sample rates through spdif. So I guess it should be possible? Is
ALSA
That may be lie. It may upmix internally-- just as badly as alsa does.
capable of switching those rates on the fly? Or
Mark Knecht wrote:
On Thu, May 1, 2008 at 8:42 AM, Bart de Boer [EMAIL PROTECTED] wrote:
SNIP
Complete output of /proc/asound/card0/codec97#0/ac97#0-0:
0-0/0: Realtek ALC650F
PCI Subsys Vendor: 0x1509
PCI Subsys Device: 0x9202
Capabilities :
DAC
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
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],
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 get AC3 passthrough to work on
James Courtier-Dutton wrote:
Arthur Yarwood wrote:
I'm having a nightmare getting ac3 pass through to work. I had this
problem originally with Fedora 4, then I upgraded (thinking it was down
to old drivers etc) to Fedora 7 and I'm still getting the very same
problem
Currently, AC3
I'm having a nightmare getting ac3 pass through to work. I had this
problem originally with Fedora 4, then I upgraded (thinking it was down
to old drivers etc) to Fedora 7 and I'm still getting the very same
problem.
I'm using xine, with pass through enabled. When I play a DVD, I just get
Arthur Yarwood wrote:
I'm having a nightmare getting ac3 pass through to work. I had this
problem originally with Fedora 4, then I upgraded (thinking it was down
to old drivers etc) to Fedora 7 and I'm still getting the very same
problem.
Currently, AC3 pass through is not supported on
I hadn't used sound for a couple months, but SPDIF has stopped
working. The optical out on the SPDIF module no longer lights up.
alsamixer no longer shows any IEC958 settings. iecset gives the
following error:
control IEC958 Playback Default not found
I'm currently using Fedora Core 6, with
Hello
Dominique Dumont [EMAIL PROTECTED] writes:
I'm trying to configure the ALC883 on my MSI K9A mobo to get
dolby digital on the spdif output.
Any news on this problem ? (which looks like bug 2622 [1])
Is the problem between keyboard and chair ? ;-)
Is this a bug ?
If yes, is anybody
On 3/15/07, Dominique Dumont [EMAIL PROTECTED] wrote:
Hello
Dominique Dumont [EMAIL PROTECTED] writes:
I'm trying to configure the ALC883 on my MSI K9A mobo to get
dolby digital on the spdif output.
Any news on this problem ? (which looks like bug 2622 [1])
You should take the
Lee Revell [EMAIL PROTECTED] writes:
You should take the initiative and try to figure out which release
broke it.
AFAIK, spdif passthrough never worked correctly.
At some point it did work on my machine, it was with the same kernel
and the same alsa release. Which is rather puzzling.
Dominique Dumont [EMAIL PROTECTED] writes:
$ ac3dec -C sound_129.ac3
Using PCM device 'plug:iec958:{AES0 0x2 AES1 0x82 AES2 0x0 AES3 0x2}'
AC3 Stream 48.0 KHz 448 kbps
One more detail: usually my Yamaha amp is able to display the sampling
frequency (either 44 or 48 KHz).
In this case, the
On 3/12/07, Dominique Dumont [EMAIL PROTECTED] wrote:
Dominique Dumont [EMAIL PROTECTED] writes:
$ ac3dec -C sound_129.ac3
Using PCM device 'plug:iec958:{AES0 0x2 AES1 0x82 AES2 0x0 AES3 0x2}'
AC3 Stream 48.0 KHz 448 kbps
One more detail: usually my Yamaha amp is able to display the
Lee Revell [EMAIL PROTECTED] writes:
I thought ac3dec just decodes an AC3 stream to PCM and plays it via
ALSA. I would not expect it to pass through a DD stream.
This is the behavior without the -C option.
AFAIR, with my former sound card, ac3dec -C play the raw ac3 stream.
Have you tried
Hi,
I am using iec958 spdif output on gigabyte ga-965-ds3 motherboard with
onboard hda-intel, kernel 2.6.18.6 alsa 1.0.13, I dont have a
asound.conf or asoundrc. Most of my sound is coming from mythtv (ac3
and dts passthru enabled), this works really well provided I run the
following commands
On 30/12/06, Andrew Lyon [EMAIL PROTECTED] wrote:
Hi,
I am using iec958 spdif output on gigabyte ga-965-ds3 motherboard with
onboard hda-intel, kernel 2.6.18.6 alsa 1.0.13, I dont have a
asound.conf or asoundrc. Most of my sound is coming from mythtv (ac3
and dts passthru enabled), this
Cross posted as this is a request for advice rather than a bug/issue report.
When playing media using mplayer or xine, and outputting using
ac3/dts passthru spdif, if I pause and resume playback, or seek, there
are brief, horrible clicking noises when playback resumes, I assume
this is because
Hi,
As a follow up to this, I just tried :
aplay -D hw:0,0 whatever.wav
and
aplay -D hw:0,1 whatever.wav
And the sound came out of my analogue output (headphones in this case). Does
anyone have a clue as to whats going on?
I'm getting kind of annoyed with this now, any ideas are welcome!
Hi,
Up to a recent kernel upgrade (I think it was Gentoo 2.6.16-r'something') the
following would result in sound being output to my amp via my spdif output :
aplay -D hw:0,1 whatever.wav
however, now it does not
aplay -D spdif whatever.wav
still does work. I've tried 1.0.11 and 1.0.12 with
Paul Lundin wrote:
Are there any cards that have both an spdif (or TOSlink) in and spdif out
that are known to work ?
Edirol UA-1D
cards with YMF574 chip (output at 48 kHz only)
HTH
Clemens
Using Tomcat but need to do more? Need to support web services, security?
Get stuff done quickly with
Clemens, Thanks. I was hoping for an internal card. I will see if I can track down something with that chipset.Regards.On 7/6/06, Clemens Ladisch
[EMAIL PROTECTED] wrote:Paul Lundin wrote:
Are there any cards that have both an spdif (or TOSlink)in and spdif out that are known to work ?Edirol
On 7/6/06, Paul Lundin [EMAIL PROTECTED] wrote:
Are there any cards that have both an spdif (or TOSlink) in and spdif out
that are known to work ? (I highly prefer optical) The matrix is woefully
out of date, and most people give conflicting reports at best. I would
prefer not to have to
Are there any cards that have both an spdif (or TOSlink) in and spdif out that are known to work ? (I highly prefer optical) The matrix is woefully out of date, and most people give conflicting reports at best. I would prefer not to have to perform voodoo, or spend more than $100 USD.
Thanks in
What you're pointing to looks like it has an ALC655 rather than a 665.
I've just recently got an ALC650 to work with spdif out:
http://mezzanines.blogspot.com/2006/02/digital-audio-out.html
[EMAIL PROTECTED] acpi]# cat /proc/asound/cards
0 [IXP]: ATIIXP-SPDMA - ATI IXP
Hello and a very happy new year to you all,
I posted this (or something similar) a while back and had no reply, so
I'm trying again in case someone new spots it.
Trying to route all sound to the spdif output of my onboard nForce4
(NVidia CK804 #9474;
#9474;
Hi,
I am trying to get mplayer to output sound through the SPDIF port on my
Toshiba Satellite 5100 with debian unstable + alsa + kernel 2.6.4.
What alsa configuration option should I use to activate the SPDIF port?
The audio card is an intel I820.
Thanks,
--
Should crematoriums give discounts
On Sun, 2004-03-21 at 09:02, Robert La Ferla wrote:
I have a Shuttle XPC (SN41G2 w/nVidia nForce) running Fedora Core 1 and
the latest ALSA 1.03. I cannot get digital audio out of the SPDIF port
but analog audio from at least the headphone jack works fine. I have
tried many things to no
Wayde Milas wrote:
On Sun, 2004-03-21 at 09:02, Robert La Ferla wrote:
I have a Shuttle XPC (SN41G2 w/nVidia nForce) running Fedora Core 1 and
the latest ALSA 1.03. I cannot get digital audio out of the SPDIF port
but analog audio from at least the headphone jack works fine. I have
tried
Here is some additional information:
% iecset
Mode: consumer
Data: audio
Rate: 44100 Hz
Copyright: permitted
Emphasis: none
Category: PCM coder
Original: original
Clock: 1000 ppm
% amixer
Simple mixer control 'Master',0
Capabilities: pvolume pswitch pswitch-joined
Playback channels: Front Left
I have a Shuttle XPC (SN41G2 w/nVidia nForce) running Fedora Core 1 and
the latest ALSA 1.03. I cannot get digital audio out of the SPDIF port
but analog audio from at least the headphone jack works fine. I have
tried many things to no avail. Please help.
ALSA Audio Debug v0.0.7 - Sun Mar
Hi,
I'm curious how to use spdif in/out under Alsa. Is it possible to use it in
parallel with analog sound ?
Regards,
Robert.
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a
On Tue, 24 Feb 2004, Robert Rozman wrote:
I'm curious how to use spdif in/out under Alsa. Is it possible to use it in
parallel with analog sound ?
Get ready for a really stupid question from somebody that just upgraded
from a SoundBlaster Pro ISA card to a M-Audio Delta 410.
I saw that my
I would like to record via SPDIF in and have no idea why it doesn't
work. Maybe it is something trivial. It could be that my
asound.state is not correct or I don't use the right syntax when
recording. I tried different apps like audacity, qarecord, ecasound
and others. Analogue playback works.
Am Dienstag, 10. Februar 2004 22:38 schrieb Al Bogner:
I would like to record via SPDIF in and have no idea why it
doesn't work. Maybe it is something trivial. It could be that my
asound.state is not correct or I don't use the right syntax when
recording. I tried different apps like audacity,
I've been previously using ALSA 0.98 with a SBlive 5.1. The SPDIF
worked by default, even did AC3/AC5 passtrhu. After upgrading to 1.0.1
the SPDIF seems to be off by default.
Can anyone give me a config line that will turn it on?
Thanks in advance
Hi,
I have a Terratec Aureon Space 7.1 and I'm able to play music through
my digital out without any problems. There is just one thing that I can't
seem to be able to do somehow. I would like to be able to duplicate
channels. Currently I can only send 2 channels (left + right). I have a
5.1 DTS
On Tue, 13 Jan 2004, Dennis van der Meer wrote:
Hi,
I have a Terratec Aureon Space 7.1 and I'm able to play music through
my digital out without any problems. There is just one thing that I can't
seem to be able to do somehow. I would like to be able to duplicate
channels. Currently I can
I have a Terratec Aureon Space 7.1 and I'm able to play music through
my digital out without any problems. There is just one thing that I can't
seem to be able to do somehow. I would like to be able to duplicate
channels. Currently I can only send 2 channels (left + right). I have a
5.1 DTS
Jaroslav Kysela wrote:
On Tue, 13 Jan 2004, Dennis van der Meer wrote:
Hi,
I have a Terratec Aureon Space 7.1 and I'm able to play music through
my digital out without any problems. There is just one thing that I can't
seem to be able to do somehow. I would like to be able to duplicate
I have a Terratec Aureon Space 7.1 and I'm able to play music through
my digital out without any problems. There is just one thing that I can't
seem to be able to do somehow. I would like to be able to duplicate
channels. Currently I can only send 2 channels (left + right). I have a
5.1 DTS
I have a Terratec Aureon Space 7.1 and I'm able to play music through
my digital out without any problems. There is just one thing that I can't
seem to be able to do somehow. I would like to be able to duplicate
channels. Currently I can only send 2 channels (left + right). I have a
5.1 DTS
hi list
i got this working once but now i stuck on it ... and when i got it working
the sound through spdif (dvd) was off by 1 sec. - so if anyone has an idea
how to get it working , plz help me ... thanks
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This SF.net email is sponsored
Hello,
Thanks to all those who made Alsa, its great and has helped me achieve one of my goals
- AC3 / DTS output via the digital SPDIF on my system! This means I can playback DVD's
using xine with full Dolby Digital 5.1 sound.
I've seen lots of people asking if this is possible with my
Hello *,
I got myself a Zoltrix Nightingale with optical module for optical
spdif input and output, which is cmi 8738 based.
Alsa-driver 0.9.8 compiled fine and was installed without problems.
Normal audio playback via speakers or via coaxial and optical spdif
works like a charm, too.
I
On Mon, 17 Nov 2003, Michael Hunold wrote:
In theory, I'd like to record the raw iec958 subframes, but using
arecord like this fails:
--schnipp
arecord -v -D hw:0,2 -f IEC958_SUBFRAME_LE
arecord: begin_wave:1608: Wave doesn't
Hi guys! I have this nightmare: I managed to get the spdif output to
work with xine, and now it works great! But I'mm' still going crazy
trying to enable the spdif output by default for all my sound
applications in Linux! In particular, how can I tell xmms to use the
digital output instead of
On Sat, 15 Nov 2003 11:22:35 -0700
Gianmarco Di Loreto [EMAIL PROTECTED] wrote:
Hi guys! I have this nightmare: I managed to get the spdif output to
work with xine, and now it works great! But I'mm' still going crazy
trying to enable the spdif output by default for all my sound
Ok...I tried what ou said...I created the file in /root (since Im' root),
the output for the devices is this
0: [0- 0]: ctl
16: [0- 0]: digital audio playback
24: [0- 0]: digital audio capture
1: : sequencer
33: : timer
so instead of n I set 16 (is this right?). In xmms I
On Sat, 15 Nov 2003 12:16:31 -0700
Gianmarco Di Loreto [EMAIL PROTECTED] wrote:
Ok...I tried what ou said...I created the file in /root (since Im'
root), the output for the devices is this
0: [0- 0]: ctl
16: [0- 0]: digital audio playback
24: [0- 0]: digital audio capture
1:
SO...when I changed the device in .asoundrc with number 16 instead of n,
I actually caused troubles to xine, since it started playing back
from the analog ouput too. I then deleted the file and rebuilt it with
device 0 (the only one I have) as a default and I called it foo :-) !!
XIne is
Hi there! I successfullt installed Alsa 0.9.8 packages and Im' using them
with xine and mi SPDIF output works great! I installed the plugin for
xmms too and it works, but no audio from the spdif output!! Please help me!!
---
This SF.Net
Hello,
In continuation of this :
http://www.mail-archive.com/[EMAIL PROTECTED]/msg09349.html
i too have the same problem with SPDiF Capture on an SIS7012 (kernel
linux 2.6 test4)
My directory looks like so :
flatmax# ls /proc/asound/card0/
ac97#0 ac97#1-1 idoss_mixer pcm0p
At Wed, 24 Sep 2003 22:43:17 +0200,
Matthias Hamel wrote:
alsa-init: soundcard set to spdif
ALSA lib pcm_hw.c:1055:(snd_pcm_hw_open) open /dev/snd/pcmC0D1p
^^^
here is the problem.
perhaps it's a bug of
Hi there,
I've got a Soundblaster Live soundcard, model name Live Player 5.1
here in europe. I'm wondering if it supports spdif input or more
correctly spdif passthrough. I'm trying to route spdif signal from a
dvb-c card through my sb live soundcard. There is a two-pin connector on
the soundcard
-Message d'origine-
De : Takashi Iwai [mailto:[EMAIL PROTECTED]
Envoyé : mardi 23 septembre 2003 15:51
À : HAMEL Matthias OCISI
Cc : Leandro Dardini; [EMAIL PROTECTED]
Objet : Re: RE : [Alsa-user] spdif on intel 8x0 (nforce2-MCT)
At Thu, 18 Sep 2003 23:42:11 +0200,
Matthias
At Wed, 24 Sep 2003 10:35:56 +0200,
HAMEL Matthias OCISI wrote:
-Message d'origine-
De : Takashi Iwai [mailto:[EMAIL PROTECTED]
Envoyé : mardi 23 septembre 2003 15:51
À : HAMEL Matthias OCISI
Cc : Leandro Dardini; [EMAIL PROTECTED]
Objet : Re: RE : [Alsa-user] spdif
alsa-init: soundcard set to spdif
ALSA lib pcm_hw.c:1055:(snd_pcm_hw_open) open /dev/snd/pcmC0D1p
^^^
here is the problem.
perhaps it's a bug of config file.
please try to change the line device 1 to device 2
At Thu, 18 Sep 2003 23:42:11 +0200,
Matthias Hamel wrote:
Hello,
I just bought a nforce2-MCT board, an asus a7n8x deluxe. It performs
well with the intel8x0 module, but I notice the lack of spdif output.
I read a post issued _before_ the latest release about spdif enabled
Hello,
I just bought a nforce2-MCT board, an asus a7n8x deluxe. It performs
well with the intel8x0 module, but I notice the lack of spdif output.
I read a post issued _before_ the latest release about spdif enabled
only in the cvs version, but lacking in the last release, I
I'm also really really interested ...
-Message d'origine-
De : Leandro Dardini [mailto:[EMAIL PROTECTED]
Envoyé : mercredi 17 septembre 2003 01:40
À : [EMAIL PROTECTED]
Objet : [Alsa-user] spdif on intel 8x0 (nforce2-MCT)
Hello,
I just bought a nforce2-MCT board, an asus a7n8x
At Wed, 17 Sep 2003 01:40:16 +0200,
Leandro Dardini wrote:
Hello,
I just bought a nforce2-MCT board, an asus a7n8x deluxe. It performs well
with the intel8x0 module, but I notice the lack of spdif output. I read a
post issued _before_ the latest release about spdif enabled only in the cvs
- Original Message -
From: Takashi Iwai [EMAIL PROTECTED]
To: Leandro Dardini [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Wednesday, September 17, 2003 12:28 PM
Subject: Re: [Alsa-user] spdif on intel 8x0 (nforce2-MCT)
At Wed, 17 Sep 2003 01:40:16 +0200,
Leandro Dardini wrote
At Wed, 17 Sep 2003 14:54:37 +0200,
Leandro Dardini wrote:
- Original Message -
From: Takashi Iwai [EMAIL PROTECTED]
To: Leandro Dardini [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Wednesday, September 17, 2003 12:28 PM
Subject: Re: [Alsa-user] spdif on intel 8x0 (nforce2-MCT
Hello,
I just bought a nforce2-MCT board, an asus a7n8x deluxe. It performs well
with the intel8x0 module, but I notice the lack of spdif output. I read a
post issued _before_ the latest release about spdif enabled only in the cvs
version, but lacking in the last release, I suppose there is some
I have seen some people have
running SPDIF with a VIA686A Chipset and VT1612A Audiochip. What is todo to
enable the SPDIF?
I have a STB from Allwell
with the same Chipset and a SPDIF in and out onboard, but when I run alsamixer
no SPDIF options are seen and when I call
cat
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
hello,
i'm using alsa 0.9.4 with my ICH5 chipset using the i820 module. is there a
way to enable spdif-out for this chipset?
keep on rockin'
sven
- --
..never argue with idiots. they drag you down to their level and beat you with
experience..
At Fri, 07 Mar 2003 01:18:14 +0800,
Anthony Magsino wrote:
Dear all,
First of all, I would like to thank all those responsible for ALSA. It's a
great project.
I have kernel 2.4.20, on-board via8235 and pci als4000, clean compile of
alsa-0.9.0rc8a from cvs. I can't get SPDIF to work
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