[Alsa-user] My audigy progress

2003-11-15 Thread Ingo Reimann
Hi list,

as mentioned before, i can`t get ac52 through the digital/analog output
Jack of my audigy2.

Maenwhile i have some progress: When i do

aplay -D spdif SOMETHING.WAV

i get an output, so the device works in principle. When i say xine and
xmms to send their pcm to spdif, i get nice sound. Nearly.

xine plays the pcm-track or downmix
xmms does not play my cds - cdrom also connected via spdif to audigy

So all digital sources - xine with passthrough enabled and xmms cd -
don`t produce neither any sound nor any error.

Maybe i should try to find some .asoundrc with the ultimate solution...

best wishes,

ingo




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Re: [Alsa-user] sndconfig

2003-11-15 Thread Frank Barknecht
Hallo,
Alberto Monteiro hat gesagt: // Alberto Monteiro wrote:

 Frank Barknecht wrote:
 
  sndconfig returns some stupid error related to via82cxxx_audio -
  this is not a module from alsa, or is it?
 
  No, it's not. As far as I know sndconfig only configures OSS modules.
  Use alsaconf for ALSA.
 
 But alsaconf is not configuring ALSA after I install ALSA. I have
 no idea why. I have no idea how I can see what's going on.
 
 Everything seems fine during the ./configure, make, make install,
 and alsaconf phases, but then when I want to play something,
 either nothing happens, or I get back some crazy error messages.

Well, but this isn't fixed by using sndconfig instead. Some common
possible causes: Did you turn up the mixer settings with alsamixer?
What errors do you get? Are the alsa-modules loaded and the OSS
modules not loaded? Could you post the relevant part of your module
configuration? You can get it with: grep 'snd\|sound' /etc/modules.conf

ciao
-- 
 Frank Barknecht   _ __footils.org__


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[Alsa-user] mandrake 9.2 how to upgrade to alsa 0.9.8?

2003-11-15 Thread Chera Bekker
Hi all,

I am running mandrake 9.2. I am trying to upgrade to the 0.9.8 version of the 
cs46xx driver. After following the instructions I end up with unresolved 
symbols in sound/acore/snd.o (snd_verbose_printk) which prevent me from 
loading the cs46xx driver. 

How can I resolve this.

the reason that I would like to upgrade is that at the moment (using the 
Mandrake supplied 0.9.6 cs46xx)  my Hercules Fortissimo II digttal edition 
only produces sound from the front and back speakers output but the dedicated 
headphone output is dead as a dodo. Microphone is working o.k. though.

in /dev/snd I have three playback devices:

pcmC0D0p
pcmC0D1p
pcmC0D2p

C0D0 is front speakers
C0D1 is back speakers
C0D2 is unknown 

I hope this issue is solved in 0.9.8. The sound level for the front and back 
speakers is too low for use with the headphone. 

regards

Chera Bekker




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Re: [Alsa-user] mandrake 9.2 how to upgrade to alsa 0.9.8?

2003-11-15 Thread Frans Ketelaars
On Saturday 15 November 2003 13:22, Chera Bekker wrote:
 Hi all,

 I am running mandrake 9.2. I am trying to upgrade to the 0.9.8
 version of the cs46xx driver. After following the instructions I end
 up with unresolved symbols in sound/acore/snd.o (snd_verbose_printk)
 which prevent me from loading the cs46xx driver.

 How can I resolve this.

On the ./configure line for the drivers use the --disable-verbose-printk 
option.

 the reason that I would like to upgrade is that at the moment (using
 the Mandrake supplied 0.9.6 cs46xx)  my Hercules Fortissimo II
 digttal edition only produces sound from the front and back speakers
 output but the dedicated headphone output is dead as a dodo.
 Microphone is working o.k. though.

 in /dev/snd I have three playback devices:

 pcmC0D0p
 pcmC0D1p
 pcmC0D2p

 C0D0 is front speakers
 C0D1 is back speakers
 C0D2 is unknown

 I hope this issue is solved in 0.9.8. The sound level for the front
 and back speakers is too low for use with the headphone.

 regards

 Chera Bekker

HTH,

-Frans



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Re: [Alsa-user] sndconfig

2003-11-15 Thread Alberto Monteiro
Frank Barknecht wrote:
 
 Everything seems fine during the ./configure, make, make install,
 and alsaconf phases, but then when I want to play something,
 either nothing happens, or I get back some crazy error messages.

 Well, but this isn't fixed by using sndconfig instead. Some common
 possible causes: Did you turn up the mixer settings with alsamixer?

Yes

 What errors do you get? 

No errors

 Are the alsa-modules loaded and the OSS
 modules not loaded? 

No, all modules are loaded, without error

 Could you post the relevant part of your module
 configuration? 

Ok:

# --- BEGIN: Generated by ALSACONF, do not edit. ---
# --- ALSACONF verion 0.9.0 ---
alias char-major-116 snd
alias snd-card-0 snd-via82xx
alias char-major-14 soundcore
alias sound-slot-0 snd-card-0
alias sound-service-0-0 snd-mixer-oss
alias sound-service-0-1 snd-seq-oss
alias sound-service-0-3 snd-pcm-oss
alias sound-service-0-8 snd-seq-oss
alias sound-service-0-12 snd-pcm-oss
options snd major=116 cards_limit=1
options snd-via82xx index=0 dxs_support=1
# --- END: Generated by ALSACONF, do not edit. ---

[BTW, the second-to-last line, options snd-via82xx, was
tested without dxs_support, and with dxs_support set to
other numbers - all without success]

Alberto Monteiro



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Re: [Alsa-user] sndconfig

2003-11-15 Thread Frank Barknecht
Hallo,
Alberto Monteiro hat gesagt: // Alberto Monteiro wrote:

 Frank Barknecht wrote:
  
  Everything seems fine during the ./configure, make, make install,
  and alsaconf phases, but then when I want to play something,
  either nothing happens, or I get back some crazy error messages.
 
  Well, but this isn't fixed by using sndconfig instead. Some common
  possible causes: Did you turn up the mixer settings with alsamixer?
 
 Yes
 
  What errors do you get? 
 
 No errors

I meant those crazy error messages from above. How are you trying to
play something? You should at first try it with aplay. 

  Are the alsa-modules loaded and the OSS
  modules not loaded? 
 
 No, all modules are loaded, without error

This is good. So at least it looks as if that part is fine. You can
also check if your card is recognized by cat /proc/asound/cards.

  Could you post the relevant part of your module
  configuration? 
 
 Ok:
 
 # --- BEGIN: Generated by ALSACONF, do not edit. ---
 # --- ALSACONF verion 0.9.0 ---
 alias char-major-116 snd
 alias snd-card-0 snd-via82xx
 alias char-major-14 soundcore
 alias sound-slot-0 snd-card-0
 alias sound-service-0-0 snd-mixer-oss
 alias sound-service-0-1 snd-seq-oss
 alias sound-service-0-3 snd-pcm-oss
 alias sound-service-0-8 snd-seq-oss
 alias sound-service-0-12 snd-pcm-oss
 options snd major=116 cards_limit=1
 options snd-via82xx index=0 dxs_support=1
 # --- END: Generated by ALSACONF, do not edit. ---
 
 [BTW, the second-to-last line, options snd-via82xx, was
 tested without dxs_support, and with dxs_support set to
 other numbers - all without success]

Looks fine to my eyes. Hmmm...

ciao
-- 
 Frank Barknecht   _ __footils.org__


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[Alsa-user] Spdif output

2003-11-15 Thread Gianmarco Di Loreto
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 the analog? Currently I have installed the 
alsa plug-in but still nothing...PLEASE HELP! I´ m getting demotivated...



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Re: [Alsa-user] Spdif output

2003-11-15 Thread Florian Schmidt
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 
 applications in Linux! In particular, how can I tell xmms to use the 
 digital output instead of the analog? Currently I have installed the 
 alsa plug-in but still nothing...PLEASE HELP! I´ m getting
 demotivated...

Well, in xmms, you can configure the alsa-driver to use a pcm other than
the default. check pcm names like front, rear, hw:0,0, hw:0,1,
etc..

You can also override the default pcm and ctl devices by editing your
.asoundrc and adding entries like:

pcm.!default {
type hw
card 0
device n
}

where n is the device number.. i suppose you won't need to override the
ctl default..

-- 
music: http://www.soundclick.com/bands/9/florianschmidt.htm




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Re: [Alsa-user] uhm...

2003-11-15 Thread Florian Schmidt
On Sat, 15 Nov 2003 11:44:36 -0700
Gianmarco Di Loreto [EMAIL PROTECTED] wrote:

 Thanks Florian,  but whatever and wherever i write h¨hw:0,0¨ xmms
 gives me an error (mixer error)...and second,  I don have a .asoundrc
 file!! Only asound.state...

Ok, create a file called .asoundrc in your home dir. put this into it:


pcm.!default {
type hw
card 0
device n
}

ctl.!default {
type hw
card 0
}

Where n is the number of the device you want the sound to be sent to..
Check /proc/asound/devices for the numbers... Then set the xmms-alsa
plugin to the device default [without the '' :)]

Does that work?

Flo

-- 
music: http://www.soundclick.com/bands/9/florianschmidt.htm




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[Alsa-user] uhm...

2003-11-15 Thread Gianmarco Di Loreto
Thanks Florian,  but whatever and wherever i write h¨hw:0,0¨ xmms gives 
me an error (mixer error)...and second,  I don have a .asoundrc file!! 
Only asound.state...



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[Alsa-user] spdif

2003-11-15 Thread Gianmarco Di Loreto
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 selected: alsa - 
configure and checked ¨user defined¨, then wrote ¨default¨
and PCM as Mixer device, but when I play something...xmms hangs and I 
have to kill it...how can I check that the .asoundrc thing is actually 
working and setting my digital output as a default? What's wrong with 
xmms? Thnaks again, Marco



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Re: [Alsa-user] spdif

2003-11-15 Thread Florian Schmidt
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:   : sequencer
   33:   : timer

Ah, ok, this means you have only one playback device, so all of my
previous tips do not apply.. 

How did you make xine use the spdif then?

 so instead of n I set 16 (is this right?). In xmms I selected: alsa - 
 configure and checked ¨user defined¨, then wrote ¨default¨
 and PCM as Mixer device, but when I play something...xmms hangs and I 
 have to kill it...how can I check that the .asoundrc thing is actually
 
 working and setting my digital output as a default? What's wrong with 
 xmms? Thnaks again, Marco
 

you can always try to use 

aplay some_sound_file.wav

to test the different pcm devices use:

aplay -D pcm_name some_sound_file.wav

for example:

aplay -D hw:0,0 Der\ Microsoft\ Sound.wav  

;)

Flo


-- 
music: http://www.soundclick.com/bands/9/florianschmidt.htm




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Re: [Alsa-user] uhm...

2003-11-15 Thread Florian Schmidt
On Sat, 15 Nov 2003 18:58:28 +0100
Florian Schmidt [EMAIL PROTECTED] wrote:

 On Sat, 15 Nov 2003 11:44:36 -0700
 Gianmarco Di Loreto [EMAIL PROTECTED] wrote:
 
  Thanks Florian,  but whatever and wherever i write h¨hw:0,0¨ xmms
  gives me an error (mixer error)...and second,  I don have a
  .asoundrc file!! Only asound.state...
 
 Ok, create a file called .asoundrc in your home dir. put this into it:

also try this:

Have alook at the output of

$ cat /proc/asound/devices
1: : sequencer
0: [0- 0]: ctl
8: [0- 0]: raw midi
18: [0- 2]: digital audio playback
17: [0- 1]: digital audio playback
16: [0- 0]: digital audio playback
24: [0- 0]: digital audio capture
33: : timer

On my system there's 3 playback devices. If i want to make the third the
default i just add this line to my ~/.asoundrc [counting starts at 0]:

defaults.pcm.device 2


Hmmm, you could also create a symbolic name for the device you want like
this:

pcm.foo {
type hw
card 0
device n
}
#replace n by the device number 

ctl.foo {
type hw
card 0
}

and then select foo in the alsa-xmms plugin...



Flo

-- 
music: http://www.soundclick.com/bands/9/florianschmidt.htm




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[Alsa-user] spdif

2003-11-15 Thread Gianmarco Di Loreto
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 working again but xmms (with foo in the alsa config) always 
plays from the analog output!! Im' getting depressed here... To get xine 
to work I just selected the expert mode and enabled all the ac52 and 
dolby options I could find!! :-)) One more thing...when I type this:

 aplay -D hw:0,0 /osshare/mp3/Old.wav

I get this message:

Playing WAVE '/osshare/mp3/Old.wav' : Signed 16 bit Little Endian, Rate 
44100 Hz, Mono
aplay: set_params:829: Channels count non available

Thanks again, Marco



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[Alsa-user] RH9 still progressing

2003-11-15 Thread Sujan Swearingen
My sound still works but causes X-window lag.  It has
not caused and lock ups recently.  I found that my
sound is sharring interrupts with acpi and usb.  I
need to tune the hard disks and enable low latency but
I am not sure if the two can be done concurrently
without wrecking the system.

=
Sujan Swearingen B.S.
[EMAIL PROTECTED]

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Re: [Alsa-user] sndconfig

2003-11-15 Thread Alberto Monteiro
Frank Barknecht wrote:

 What errors do you get?

 No errors

 I meant those crazy error messages from above. How are you
 trying to play something? 

play, aplay, xmms - nothing works.

 You should at first try it with aplay.

aplay: main:503: audio open error: Permission denied

BTW, the permissions for /usr/dsp are:

crw-rw-rw-1 albmont  audio 14,   3 Nov  7 03:05 /dev/dsp


 No, all modules are loaded, without error

 This is good. So at least it looks as if that part is fine. You can
 also check if your card is recognized by cat /proc/asound/cards.

0 [V8235  ]: VIA8233 - VIA 8235
 VIA 8235 at 0xe800, irq 22

[whatever that means :-)]


  Could you post the relevant part of your module
  configuration?

 Ok:

 # --- BEGIN: Generated by ALSACONF, do not edit. ---
 # --- ALSACONF verion 0.9.0 ---
 alias char-major-116 snd
 alias snd-card-0 snd-via82xx
 alias char-major-14 soundcore
 alias sound-slot-0 snd-card-0
 alias sound-service-0-0 snd-mixer-oss
 alias sound-service-0-1 snd-seq-oss
 alias sound-service-0-3 snd-pcm-oss
 alias sound-service-0-8 snd-seq-oss
 alias sound-service-0-12 snd-pcm-oss
 options snd major=116 cards_limit=1
 options snd-via82xx index=0 dxs_support=1
 # --- END: Generated by ALSACONF, do not edit. ---

 [BTW, the second-to-last line, options snd-via82xx, was
 tested without dxs_support, and with dxs_support set to
 other numbers - all without success]

 Looks fine to my eyes. Hmmm...

Now you can see that I have a _big_ problem :-)

Also, when trying to install ALSA, I did something that
crashed the gcc compilation: now it's including wrong files :-(

Alberto Monteiro



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[Alsa-user] Module Option Parameters

2003-11-15 Thread Brett I. Holcomb
The docs for my sound card (TB Santa Cruz) list several options (see below).  

1.  When I put them in the options snd-cs46xx statement how do I know what 
string to use.  The card document tells me id is for the ID string and 
enable_midi is to enable midi support.  What about the rest.

2.  Also the docs for the Santa cruz list an enable_midi which is NOT in the 
list of paramters.  So how do I know what options ARE available for my card.

Thanks.


# parm: index int array (min = 1, max = 8), description Index value for the 
CS46xx soundcard.
# parm: id string array (min = 1, max = 8), description ID string for the 
CS46xx soundcard.
# parm: enable int array (min = 1, max = 8), description Enable CS46xx 
soundcard.
# parm: external_amp int array (min = 1, max = 8), description Force to 
enable external amplifer.
# parm: thinkpad int array (min = 1, max = 8), description Force to enable 
Thinkpad's CLKRUN control.
# parm: mmap_valid int array (min = 1, max = 8), description Support OSS 
mmap.



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[Alsa-user] CVS, Realtek ALC655 and 'Line-In As Surround' not working

2003-11-15 Thread Richard Chan
I've tried CVS which has now got support for the Realtek ALC655.
I tried setting 'Line-In As Surround' and playing through the rear
speakers (connected to line-in jack BTW) but I'm not getting any sound.
Can anyone see any flaws here? Am I right in assuming that

1. hw:0,1 is a 6-channel device
2.  0,1 -front
   2,3-rear
   4,5-center/LFE
.asoundrc
pcm.test {
   type plug
   slave {
   pcm hw:0,1
   rate 48000
   }
   ttable.0.2  1
   ttable.1.3  1
}
Simple mixer control 'Master',0
 Capabilities: pvolume pswitch pswitch-joined
 Playback channels: Front Left - Front Right
 Limits: Playback 0 - 31
 Front Left: Playback 25 [81%] [on]
 Front Right: Playback 25 [81%] [on]
Simple mixer control 'Master Mono',0
 Capabilities: pvolume pvolume-joined pswitch pswitch-joined
 Playback channels: Mono
 Limits: Playback 0 - 31
 Mono: Playback 0 [0%] [off]
Simple mixer control 'PCM',0
 Capabilities: pvolume pswitch pswitch-joined
 Playback channels: Front Left - Front Right
 Limits: Playback 0 - 31
 Front Left: Playback 26 [84%] [on]
 Front Right: Playback 26 [84%] [on]
Simple mixer control 'Surround',0
 Capabilities: pvolume pswitch
 Playback channels: Front Left - Front Right
 Limits: Playback 0 - 31
 Front Left: Playback 31 [100%] [on]
 Front Right: Playback 31 [100%] [on]
Simple mixer control 'Surround Down Mix',0
 Capabilities: pswitch pswitch-joined
 Playback channels: Mono
 Mono: Playback [off]
Simple mixer control 'Center',0
 Capabilities: pvolume pvolume-joined pswitch pswitch-joined
 Playback channels: Mono
 Limits: Playback 0 - 31
 Mono: Playback 31 [100%] [on]
Simple mixer control 'Center/LFE Down Mix',0
 Capabilities: pswitch pswitch-joined
 Playback channels: Mono
 Mono: Playback [off]
Simple mixer control 'LFE',0
 Capabilities: pvolume pvolume-joined pswitch pswitch-joined
 Playback channels: Mono
 Limits: Playback 0 - 31
 Mono: Playback 31 [100%] [on]
Simple mixer control 'Line',0
 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-joined 
cswitch-exclusive
 Capture exclusive group: 0
 Playback channels: Front Left - Front Right
 Capture channels: Front Left - Front Right
 Limits: Playback 0 - 31
 Front Left: Playback 26 [84%] [on] Capture [off]
 Front Right: Playback 26 [84%] [on] Capture [off]
Simple mixer control 'Line-In As Surround',0
 Capabilities: pswitch pswitch-joined
 Playback channels: Mono
 Mono: Playback [on]
Simple mixer control 'CD',0
 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-joined 
cswitch-exclusive
 Capture exclusive group: 0
 Playback channels: Front Left - Front Right
 Capture channels: Front Left - Front Right
 Limits: Playback 0 - 31
 Front Left: Playback 25 [81%] [on] Capture [off]
 Front Right: Playback 25 [81%] [on] Capture [off]
Simple mixer control 'Mic',0
 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch 
cswitch-joined cswitch-exclusive
 Capture exclusive group: 0
 Playback channels: Mono
 Capture channels: Front Left - Front Right
 Limits: Playback 0 - 31
 Mono: Playback 0 [0%] [off]
 Front Left: Capture [on]
 Front Right: Capture [on]
Simple mixer control 'Mic As Center/LFE',0
 Capabilities: pswitch pswitch-joined
 Playback channels: Mono
 Mono: Playback [on]
Simple mixer control 'Mic Boost (+20dB)',0
 Capabilities: pswitch pswitch-joined
 Playback channels: Mono
 Mono: Playback [off]
Simple mixer control 'Mic Select',0
 Capabilities:
 Mono:
Simple mixer control 'Video',0
 Capabilities: cswitch cswitch-joined cswitch-exclusive
 Capture exclusive group: 0
 Capture channels: Front Left - Front Right
 Front Left: Capture [off]
 Front Right: Capture [off]
Simple mixer control 'Phone',0
 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch 
cswitch-joined cswitch-exclusive
 Capture exclusive group: 0
 Playback channels: Mono
 Capture channels: Front Left - Front Right
 Limits: Playback 0 - 31
 Mono: Playback 0 [0%] [off]
 Front Left: Capture [off]
 Front Right: Capture [off]
Simple mixer control 'IEC958',0
 Capabilities: pswitch pswitch-joined cswitch cswitch-joined
 Playback channels: Mono
 Capture channels: Mono
 Mono: Playback [on] Capture [off]
Simple mixer control 'IEC958 Input Monitor',0
 Capabilities: pswitch pswitch-joined
 Playback channels: Mono
 Mono: Playback [off]
Simple mixer control 'IEC958 Output',0
 Capabilities: pswitch pswitch-joined
 Playback channels: Mono
 Mono: Playback [on]
Simple mixer control 'IEC958 Playback AC97-SPSA',0
 Capabilities: volume volume-joined
 Playback channels: Mono
 Limits: 0 - 3
 Mono: 3 [100%]
Simple mixer control 'PC Speaker',0
 Capabilities: pvolume pvolume-joined pswitch pswitch-joined
 Playback channels: Mono
 Limits: Playback 0 - 15
 Mono: Playback 0 [0%] [on]
Simple mixer control 'Aux',0
 Capabilities: pvolume pswitch pswitch-joined cswitch cswitch-joined 
cswitch-exclusive
 Capture exclusive group: 0
 Playback channels: Front Left - Front Right
 Capture channels: Front Left - Front Right
 Limits: Playback 0 - 31
 Front 

[Alsa-user] AWE 64 FM port problem (new kernel or new ALSA?)

2003-11-15 Thread Joe Morris (NTM)
Though it isn't affecting my work, I would like to get the OPL3 loading 
on my computer.  It works with my 2.4.20 kernel, which has ALSA 
.9.0rc7.  I am testing 2.6.0-test9, and have it mostly working except 
for the fm port (OPL3).  The kernel came with ALSA .9.7.  I have updated 
it to .9.8, then to the latest CVS in hopes of getting this fixed, but 
no go.  My error is:
Nov 16 11:04:30 jmorris kernel: ALSA sound/drivers/opl3/opl3_lib.c:133: 
OPL3: stat1 = 0xff
Nov 16 11:04:30 jmorris kernel: ALSA sound/drivers/opl3/opl3_lib.c:444: 
OPL2/3 chip not detected at 0x388/0x38a
Nov 16 11:04:30 jmorris kernel: ALSA sound/isa/sb/sb16.c:489: sbawe: no 
OPL device at 0x388-0x38a
My question, to help me figure out which to troubleshoot, is, Does 
anyone have an AWE64 with ALSA .9.7 or .9.8 and have the FM port 
working, with any kernel.  It may be a 2.6 kernel problem.  I have even 
tried almost matching the code in the OPL3 directory with the code from 
.9.0rc7, but still no go.  Thanks.  BTW, this is on a SuSE 8.2 machine.

--
Joe Morris
New Tribes Mission
Email Address: [EMAIL PROTECTED]
Web Address: http://www.mydestiny.net/~joe_morris
Registered Linux user 231871
God said, I AM that I AM.  I say, by the grace of
God, I am what I am.






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[Alsa-user] via82xx multichannel help

2003-11-15 Thread matt hatfield
I am using a MSI KT4V Motherboard with onboard sound (ALC 650)

I compiled installed alsa 0.9.3 and the onboard sound output works great,
but the two outputs on the S-bracket for multichannel sound do not work. I
have tried adjusting volumes and muting and unmuting and I can't get those
two outputs to work. Any ideas?? Thanks in advance


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