[Alsa-user] Audigy 2 NX configuration
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[Alsa-user] Audigy 2 NX configuration
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From: Chris <ch...@mo...> - 2005-05-30 17:12:44
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(warning: long/involved message)
I'm hoping some kind soul can help me get this device working. It's
really close - I went through the ML archives but didn't see anything
in there that had a good/valid asound.conf or other discussion. I
haven't been able to pluck out the relevant bits from the wiki to make
this beastie work either. I'm at the point where I've been trying to
make this work for too long and am not seeing the solution. :(
System is Gentoo Linux, kernel 2.6.11-gentoo-r9, alsa 1.0.8 built into
the kernel. I've got both the via82xx (onboard VIA EPIA mobo audio) and
snd-usb-audio (Audigy 2 NX) "working" and am planning on using both.
//
// cat /etc/make.conf | grep ALSA
//
ALSA_CARDS="via82xx,emu10k1,usb-audio"
//
// cat /etc/modules.d/alsa
//
# ALSA portion
alias char-major-116 snd
alias snd-card-0 snd-usb-audio
alias snd-card-1 snd-via82xx
options snd-usb-audio index=0 id="audigy"
options snd-via82xx index=1 dxs_support=3 id="via"
# OSS/Free portion
alias char-major-14 soundcore
alias sound-slot-0 snd-card-0
alias sound-slot-1 snd-card-1
# OSS/Free portion - card #1
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
# OSS/Free portion - card #2
alias sound-service-1-0 snd-mixer-oss
alias sound-service-1-1 snd-seq-oss
alias sound-service-1-3 snd-pcm-oss
alias sound-service-1-8 snd-seq-oss
alias sound-service-1-12 snd-pcm-oss
alias /dev/sound/mixer snd-mixer-oss
alias /dev/sound/dsp snd-pcm-oss
alias /dev/sound/midi snd-seq-oss
# Set this to the correct number of cards.
options snd cards_limit=2
//
// lsmod
//
Module Size Used by
snd_pcm_oss 54304 0
snd_mixer_oss 20512 1 snd_pcm_oss
snd_seq_oss 36064 0
snd_seq_midi_event 7904 1 snd_seq_oss
snd_seq 55760 4 snd_seq_oss,snd_seq_midi_event
snd_via82xx 27936 0
snd_ac97_codec 78840 1 snd_via82xx
snd_mpu401_uart 8192 1 snd_via82xx
snd_usb_audio 67904 0
snd_pcm 96872 4
snd_pcm_oss,snd_via82xx,snd_ac97_codec,snd_usb_audio
snd_timer 26596 2 snd_seq,snd_pcm
snd_page_alloc 10020 2 snd_via82xx,snd_pcm
snd_usb_lib 13440 1 snd_usb_audio
snd_rawmidi 25536 2 snd_mpu401_uart,snd_usb_lib
snd_seq_device 8652 3 snd_seq_oss,snd_seq,snd_rawmidi
snd 56068 12
snd_pcm_oss,snd_mixer_oss,snd_seq_oss,snd_seq,snd_via82xx,snd_ac97_codec
,snd_mpu401_uart,snd_usb_audio,snd_pcm,snd_timer,snd_rawmidi,snd_seq_dev
ice
soundcore 10528 1 snd
vt1211 23220 0
via686a 20184 0
i2c_sensor 3520 2 vt1211,via686a
//
// ls -l /dev/sound /dev/snd
//
/dev/snd:
total 0
crw-rw---- 1 root audio 116, 0 May 30 12:42 controlC0
crw-rw---- 1 root audio 116, 32 May 30 12:42 controlC1
crw-rw---- 1 root audio 116, 24 May 30 12:42 pcmC0D0c
crw-rw---- 1 root audio 116, 16 May 30 12:42 pcmC0D0p
crw-rw---- 1 root audio 116, 56 May 30 12:42 pcmC1D0c
crw-rw---- 1 root audio 116, 48 May 30 12:42 pcmC1D0p
crw-rw---- 1 root audio 116, 57 May 30 12:42 pcmC1D1c
crw-rw---- 1 root audio 116, 49 May 30 12:42 pcmC1D1p
crw-rw---- 1 root audio 116, 1 May 30 12:42 seq
crw-rw---- 1 root audio 116, 33 May 30 12:42 timer
/dev/sound:
total 0
crw-rw---- 1 root audio 14, 28 May 30 12:42 adsp1
crw-rw---- 1 root audio 14, 4 May 30 12:42 audio
crw-rw---- 1 root audio 14, 20 May 30 12:42 audio1
crw-rw---- 1 root audio 14, 3 May 30 12:42 dsp
crw-rw---- 1 root audio 14, 19 May 30 12:42 dsp1
crw-rw---- 1 root audio 14, 0 May 30 12:42 mixer
crw-rw---- 1 root audio 14, 16 May 30 12:42 mixer1
crw-rw---- 1 root audio 14, 1 May 30 12:42 sequencer
crw-rw---- 1 root audio 14, 8 May 30 12:42 sequencer2
How do I make the device work with 4.0 sound? This is going into a car
so I only need balance (left-right) and fader (front-back) and a master
volume/mute. I also have another sound card input to the NX2 but can't
figure out how to control that volume level properly.
If I put the second card into the NX Mic in port the volume from that
is really loud. If I put it in the NX's line in it is really quiet. Is
there a way to "tune" this? The input is line out from an onboard
chipset so it should really go into the line in on the NX.
Is there something I can stick in asound.conf to make this work? I've
been plugging away at the stuff at the alsa wiki. I've tried
making/tweaking examples from the wiki but nothing seems to have an
effect.
Here is my aplay -l output:
//
// aplay -l
//
**** List of PLAYBACK Hardware Devices ****
card 0: audigy [SB Audigy 2 NX], device 0: USB Audio [USB Audio]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: via [VIA 8235], device 0: VIA 8235 [VIA 8235]
Subdevices: 4/4
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
Subdevice #2: subdevice #2
Subdevice #3: subdevice #3
card 1: via [VIA 8235], device 1: VIA 8235 [VIA 8235]
Subdevices: 1/1
Subdevice #0: subdevice #0
//
// aplay -L
//
PCM list:
hw {
@args.0 CARD
@args.1 DEV
@args.2 SUBDEV
@args.CARD {
type string
default {
@func getenv
vars {
0 ALSA_PCM_CARD
1 ALSA_CARD
}
default {
@func refer
name 'defaults.pcm.card'
}
}
}
@args.DEV {
type integer
default {
@func igetenv
vars {
0 ALSA_PCM_DEVICE
}
default {
@func refer
name 'defaults.pcm.device'
}
}
}
@args.SUBDEV {
type integer
default {
@func refer
name 'defaults.pcm.subdevice'
}
}
type hw
card $CARD
device $DEV
subdevice $SUBDEV
}
plughw {
@args.0 CARD
@args.1 DEV
@args.2 SUBDEV
@args.CARD {
type string
default {
@func getenv
vars {
0 ALSA_PCM_CARD
1 ALSA_CARD
}
default {
@func refer
name 'defaults.pcm.card'
}
}
}
@args.DEV {
type integer
default {
@func igetenv
vars {
0 ALSA_PCM_DEVICE
}
default {
@func refer
name 'defaults.pcm.device'
}
}
}
@args.SUBDEV {
type integer
default {
@func refer
name 'defaults.pcm.subdevice'
}
}
type plug
slave.pcm {
type hw
card $CARD
device $DEV
subdevice $SUBDEV
}
}
plug {
@args.0 SLAVE
@args.SLAVE {
type string
}
type plug
slave.pcm $SLAVE
}
dmix {
@args.0 SLAVE
@args.1 FORMAT
@args.2 RATE
@args.SLAVE {
type string
default 'hw:0,0'
}
@args.FORMAT {
type string
default S16_LE
}
@args.RATE {
type integer
default 48000
}
type dmix
ipc_key 5678293
ipc_key_add_uid yes
slave {
pcm $SLAVE
format $FORMAT
rate $RATE
}
}
dsnoop {
@args.0 SLAVE
@args.1 FORMAT
@args.2 RATE
@args.SLAVE {
type string
default 'hw:0,0'
}
@args.FORMAT {
type string
default S16_LE
}
@args.RATE {
type integer
default 48000
}
type dsnoop
ipc_key 5778293
ipc_key_add_uid yes
slave {
pcm $SLAVE
format $FORMAT
rate $RATE
}
}
shm {
@args.0 SOCKET
@args.1 PCM
@args.SOCKET {
type string
}
@args.PCM {
type string
}
type shm
server $SOCKET
pcm $PCM
}
tee {
@args.0 SLAVE
@args.1 FILE
@args.2 FORMAT
@args.SLAVE {
type string
}
@args.FILE {
type string
}
@args.FORMAT {
type string
default raw
}
type file
slave.pcm $SLAVE
file $FILE
format $FORMAT
}
file {
@args.0 FILE
@args.1 FORMAT
@args.FILE {
type string
}
@args.FORMAT {
type string
default raw
}
type file
slave.pcm null
file $FILE
format $FORMAT
}
null {
type null
}
cards 'cards.pcm'
front 'cards.pcm.front'
rear 'cards.pcm.rear'
center_lfe 'cards.pcm.center_lfe'
side 'cards.pcm.side'
surround40 'cards.pcm.surround40'
surround41 'cards.pcm.surround41'
surround50 'cards.pcm.surround50'
surround51 'cards.pcm.surround51'
surround71 'cards.pcm.surround71'
iec958 'cards.pcm.iec958'
spdif 'cards.pcm.iec958'
modem 'cards.pcm.modem'
default 'cards.pcm.default'
(note: surround40 and friends don't do anything, invoke a null plugin)
I tried this in the asound.conf and am getting some useful results. But
I can't seem to be able to apply the device I created called S40 to
/dev/sound/dsp. If I do something like "aplayer -DS40
/home/MP3/Trainspotting/2/10 nightclubbing (iggy pop).mp3" I get output
over the front and rear channels. I can't figure out how to get it to
be used in gqmpeg though (gqmpeg needs a /dev/sound/device style
interface). Using aoss to play the mp3 bears this out also.
//
// /etc/asound.conf
//
dshare {
type dmix
ipc_key 2048
slave {
pcm hw:0
rate 48000
period_time 0
period_size 1024
buffer_size 8192
channels 4
}
bindings {
0 0
1 1
2 2
3 3
}
}
frontx {
type plug
slave {
pcm dshare
channels 4
}
ttable.0.0 1
ttable.1.1 1
}
rearx {
type plug
slave {
pcm dshare
channels 4
}
ttable.0.2 1
ttable.1.3 1
}
S40 {
type plug
slave {
pcm dshare
channels 4
}
ttable.0.0 1
ttable.0.2 1
ttable.1.1 1
ttable.1.3 1
}
Can some one give some pointers? I guess I need to map S40 to
/dev/sound/dsp somehow?
A pointer about the input and master volume would be appreciated as
well.
--
http://moose.ca
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