[draconius]

Quote:

Originally Posted by Weedy /img/forum/go_quote.gif i want your babies

seconded. respect.

I got this running under (X)ubuntu 6.10 "Edgy"

Code:

[left]draconius@incognito pts/0 09:49 AM Fri Dec 08 (~) $ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.12rc1 (Thu Jun 22 13:55:50 2006 UTC).[/left]

The only problem I seem to encounter is occasional "popping" noises coming from my system...not sure what it is, If I figure out a way to fix it, I'll post here.

 

[darklegion]

Quote:

Originally Posted by draconius /img/forum/go_quote.gif The only problem I seem to encounter is occasional "popping" noises coming from my system...not sure what it is, If I figure out a way to fix it, I'll post here.

I haven't had any problems with popping.I do have an issue where I get loud noise on bootup, occasionally (which can be fixed by restarting ALSA).I think it might have just been the kernel I was using though, as I haven't got the issue at all (yet) with 2.6.19-rc6.

 

[predder]

Hi, I'm new to configuring alsa.. I tried about 4 different asound.state files from around the place (including this thread) and the only one to give me sound out of the Wolfson DAC jack is the most recent one posted by darklegion..

I get none of the popping you descibed but problem is all the music I play sounds kind've warped.. and it sounds like it's outputting at 90% speed?

I'm using Debian 4.0 (etch) and 2.6.20. Here's some more info:
Code:

[left]nathan@predgame:~$ sudo alsactl restore -f /etc/asound.state alsactl: set_control:991: warning: name mismatch (Mic Boost (+20dB) Switch/Mic Boost (+20dB)) for control #19 alsactl: set_control:993: warning: index mismatch (0/0) for control #19 alsactl: set_control:991: warning: name mismatch (Mic Select Capture Switch/Mic Select) for control #35 alsactl: set_control:993: warning: index mismatch (0/0) for control #35 nathan@predgame:~$ lspci | grep audio 01:0a.0 Multimedia audio controller: VIA Technologies Inc. VT1720/24 [Envy24PT/HT] PCI Multi-Channel Audio Controller (rev 01) nathan@predgame:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.14rc1 (Tue Jan 09 09:56:17 2007 UTC). nathan@predgame:~$ lsmod | grep snd snd_ice1724 62836 2 snd_ice17xx_ak4xxx 4416 1 snd_ice1724 snd_ac97_codec 88160 1 snd_ice1724 ac97_bus 2752 1 snd_ac97_codec snd_ak4114 9664 1 snd_ice1724 snd_pcm_oss 39328 0 snd_mixer_oss 15808 1 snd_pcm_oss snd_pcm 71556 4 snd_ice1724,snd_ac97_codec,snd_ak4114,snd_pcm_oss snd_timer 20932 1 snd_pcm snd_page_alloc 10056 1 snd_pcm snd_ak4xxx_adda 7104 2 snd_ice1724,snd_ice17xx_ak4xxx snd_mpu401_uart 8320 1 snd_ice1724 snd_rawmidi 22624 1 snd_mpu401_uart snd_seq_device 8076 1 snd_rawmidi snd 48292 15 snd_ice1724,snd_ac97_codec,snd_ak4114,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer,snd_ak4xxx_adda,snd_mpu401_uart,snd_rawmidi,snd_seq_device soundcore 7904 1 snd[/left]

Any help would be greatly appreciated

Edit:
After a bit of messing around I realised that if I change the sound to use OSS rather than ALSA it sounds fine and at normal speed? That's a bit weird because I'm fairly sure it's just a wrapper for ALSA? I checked out your bug --> https://bugtrack.alsa-project.org/al...ew.php?id=2348 and downloaded that asound.state. Trying to load it produces:
nathan@predgame:~$ sudo alsactl restore -f /etc/asound.state
ALSA lib conf.c:1587

snd_config_load1) _toplevel_:428:20:Unexpected char
alsactl: load_state:1312: snd_config_load error: Invalid argument

So I don't know if it loaded or not... sounds better but I might be imagining things, having no music for half the day is playing tricks on me

 

[levgelb]

Hello all -

I'm having only mixed luck with an AV710 on Fedora Core 6 -

I can get sound through the regular line out (green) without a problem, but nothing is present on the black (rear channel, Wolfson) outputs.

I've tried several asound.state files from around the forum, but ALSA gives name-recognition errors from all of them. I've also tried changing the installer-generated asound.state file to 'match' the ones posted here, but I'm afraid that I don't really understand which are the important settings in those files or why.

FC6 uses the latest ALSA, version 1.0.13 - does anyone have a working asound.state file for this version that gives high-quality stereo output on the
rear channel?

Cheers,

Lev

 

[darklegion]

predder: You probably had the samplerate set to a value different to what your audio player uses.Make sure "Multi Track Rate Locking" is set to off, in alsamixer.Could be related to dmix too, which is what allows software mixing on linux.BTW I recommend that everyone disable dmix completely as the resample algorithm is really bad.You can disable it by creating asound.conf with these contents:
/etc/asound.conf:
Code:

[left]pcm.!default { type plug slave.pcm hw }[/left]

levgelb :
I've attached another asound.state that I am currently using which should work with alsa-1.0.13.If it doesn't, post the error output and I'll try and figure out what's going on.

 

[levgelb]

Darklegion - Thanks for the new asound.state file.

Unfortunately, it doesn't work - I get the following errors upon 'alsactl restore' -
alsactl: set_control:991: warning: name mismatch (Headphone Playback Switch/PC Speaker Playback Switch) for control #9
alsactl: set_control:993: warning: index mismatch (0/0) for control #9
alsactl: set_control:995: failed to obtain info for control #9 (Operation not permitted)

The problem seems to be that the number and ordering of controls is detected differently on my system and yours. I was able to renumber things until your asound.state file was usable (gave no errors), and did obtain sound on the usual headphone jack (green output) but still nothing coming from the rear outputs (checked with both xmms and Aqualung).

The great majority of the controls seem to have nothing to do with the rear outputs - can you tell me what particularly needs to be set to enable duplicate routing to the rear channels?

Oh, and here are some typical debugging output:

>lspci:
09:03.0 Multimedia audio controller: VIA Technologies Inc. VT1720/24 [Envy24PT/HT] PCI Multi-Channel Audio Controller (rev 01)

>lsmod | grep snd
snd_intel8x0 46825 0
snd_ice1724 95081 3
snd_ice17xx_ak4xxx 13249 1 snd_ice1724
snd_ac97_codec 125849 2 snd_intel8x0,snd_ice1724
snd_ac97_bus 11585 1 snd_ac97_codec
snd_ak4114 20673 1 snd_ice1724
snd_seq_dummy 12869 0
snd_seq_oss 45377 0
snd_seq_midi_event 17473 1 snd_seq_oss
snd_seq 71905 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event
snd_pcm_oss 57185 0
snd_mixer_oss 26945 1 snd_pcm_oss
snd_pcm 101449 6 snd_intel8x0,snd_ice1724,snd_ac97_codec,snd_ak4114 ,snd_pcm_oss
snd_ak4xxx_adda 17089 2 snd_ice1724,snd_ice17xx_ak4xxx
snd_timer 35017 2 snd_seq,snd_pcm
snd_mpu401_uart 19009 1 snd_ice1724
snd_rawmidi 37697 1 snd_mpu401_uart
snd_seq_device 17877 4 snd_seq_dummy,snd_seq_oss,snd_seq,snd_rawmidi
snd 78697 19 snd_intel8x0,snd_ice1724,snd_ac97_codec,snd_ak4114 ,snd_seq_oss,snd_seq,snd_pcm_oss,snd_mixer_oss,snd _pcm,snd_ak4xxx_adda,snd_timer,snd_mpu401_uart,snd _rawmidi,snd_seq_device
soundcore 18145 1 snd
snd_page_alloc 19665 2 snd_intel8x0,snd_pcm


Cheers, Lev

 

[AdamWill]

just dropping by to say I can't help any more with this topic as my setup moved on a while ago: I now run an external DAC (Firestone Spitfire). I did this from the optical output of the AV710 for a while, now I use a Turtle Beach Audio Advantage USB (because I switched to a laptop), which works out of the box.

so good luck all

 

[darklegion]

Post the output of 'amixer' and 'cat /proc/asound/AV710/ice1724' into a text file.

 

[levgelb]

OK, amixer and /proc/asound/AV710/ice1724 contents are
in the attached file.

(Note that the machine also has onboard Intel sound, ESB2 type)

Lev

 

[Anfibolita]

well, i just installed ubuntu and i get the Wolfson DAC working with the last asound.state but i have some problems with the volume? if i raise the PCM volume bar beyond 80% the sound begins to distort, is this normal??? also the volume is far high from my windows installation with ASIO & Foobar? one more thing. how about the alsa tool gui for envy24. is this usefull, i installed but i can't get it to work.

 

[drizek]

Thanks for the help on this thread. My sound card does not work in windows and it used to have problems in linux(lineout only does the right channel, it would play music too quickly from surround out). Now I have it hooked up to the Center out(is that the right one?) and it seems to be working well. I have never heard a higher end DAC so I cant be sure that it is using the wolfson, but it sounds very good IMO.

 

[brodiepearce]

I'm not getting any sound out of my AV710. I've tried the asound.state in the original Graphical guide, and a few of the newer ones in this thread.

The first time I did alsa spat out "No state is present for card AV710", however, every attempt since then has given me "No state is present for card CK804". CK804 being the NVidia chipset.

I'm running v1.0.14rc1 of ALSA at the moment, I have a feeling there are no asound.state files around for this version yet?

*edit*
Based on other users' issues, I figure some outputs might be necessary:

cat /proc/asound/AV710/ice1724
lsmod | grep snd
lspci | grep VIA

*edit*
I've finally got sound coming out the Front channel on the card.... somehow, not really game to restart it in case alsa switches back to the CK804 either! >.< I think I'll wait a bit before trying to get the Wolfson DAC going, as my headphone amp won't arrive for at least another week.

 

[MCC]

Quote:

Originally Posted by brodiepearce /img/forum/go_quote.gif The first time I did alsa spat out "No state is present for card AV710", however, every attempt since then has given me "No state is present for card CK804". CK804 being the NVidia chipset.

Try disabling the NVidia audio in your BIOS. This should hopefully resolve the conflicts you're having.

I have a question myself. Everything outputted through the AV-710 in Linux is automatically resampled to 48kHz. Is there any way to get bitperfect (44.1kHz) with this card in Linux? I'd rather not build a dedicated Windows machine just for kernel streaming when I get a DAC next year-ish.

If not, is there any other card/device that supports bitperfect audio out of Linux?

Edit: I just found the Diamond XS71 with a CMI 8768 chipset that supports 44.1kHz w/o resampling in windows. Anyone know if the same goes for Linux?

Edit 2: I did it!!

First, create ~/.asoundrc and add this this (or add it on if it already exists):

Code:

[left]defaults.pcm.dmix.rate 44100[/left]

Then, replace /usr/share/alsa/cards/ICE1724.conf with this:

Code:

[left]# # Configuration for the ICE1724 (Envy24HT) chip # # default with dmix & dsnoop ICE1724.pcm.44.1khq { type rate slave { pcm "hw:0,0" rate 44100 } converter "samplerate_best" } pcm.44.1k { type rate slave { pcm "hw:0,0" rate 44100 } converter "samplerate" } <confdir:pcm/front.conf> ICE1724.pcm.front.0 { @args [ CARD ] @args.CARD { type string } type hw card $CARD } <confdir:pcm/rear.conf> ICE1724.pcm.rear.0 { @args [ CARD ] @args.CARD { type string } type hw card $CARD device 2 subdevice 1 } <confdir:pcm/center_lfe.conf> ICE1724.pcm.center_lfe.0 { @args [ CARD ] @args.CARD { type string } type hw card $CARD device 2 } <confdir:pcm/side.conf> ICE1724.pcm.side.0 { @args [ CARD ] @args.CARD { type string } type hw card $CARD device 2 subdevice 2 } <confdir:pcm/surround40.conf> ICE1724.pcm.surround40.0 { @args [ CARD ] @args.CARD { type string } type route ttable.0.0 1 ttable.1.1 1 ttable.2.4 1 ttable.3.5 1 slave { channels 6 pcm { type hw card $CARD } } } <confdir:pcm/surround41.conf> <confdir:pcm/surround50.conf> <confdir:pcm/surround51.conf> ICE1724.pcm.surround51.0 { @args [ CARD ] @args.CARD { type string } type route ttable.0.0 1 ttable.1.1 1 ttable.2.4 1 ttable.3.5 1 ttable.4.2 1 ttable.5.3 1 slave { channels 6 pcm { type hw card $CARD } } } <confdir:pcm/surround71.conf> ICE1724.pcm.surround71.0 { @args [ CARD ] @args.CARD { type string } type route ttable.0.0 1 ttable.1.1 1 ttable.2.4 1 ttable.3.5 1 ttable.4.2 1 ttable.5.3 1 ttable.6.6 1 ttable.7.7 1 slave { channels 8 pcm { type hw card $CARD } } } <confdir:pcm/iec958.conf> ICE1724.pcm.iec958.0 { @args [ CARD AES0 AES1 AES2 AES3 ] @args.CARD { type string } @args.AES0 { type integer } @args.AES1 { type integer } @args.AES2 { type integer } @args.AES3 { type integer } type asym playback.pcm { type hooks slave.pcm { type hw card $CARD device 1 } hooks.0 { type ctl_elems hook_args [ { interface MIXER name "IEC958 Output Switch" lock true preserve true value true } { interface PCM name "IEC958 Playback Default" device 1 lock true preserve true value [ $AES0 $AES1 $AES2 $AES3 ] } ] } } capture.pcm { type hooks slave.pcm { type hw card $CARD device 1 } hooks.0 { type ctl_elems hook_args [ { interface MIXER name "IEC958 Capture Switch" lock true preserve true value true } ] } } }[/left]

Once done, reboot. (simply restarting alsa gives me issues)

With these modifications I'm able to unmute Multi Track Rate Locking and keep a 44.1kHz sample rate through Amarok without slowed-down output. I perceive a noticeable increase in SQ but it might just be placebo.

I got inspiration for this from hydrogen audio.

The disadvantage is that 48kHz audio will play back too fast unless resampled.

Edit 3: Games that previously had messed audio (e.g., neverball) now work fine. Frozen Bubble doesn't have sound anymore though.

Fix one thing, break another...

 

[brodiepearce]

I found the problem, the NVidia chipset was taking the place of default sound device every time I restarted (except last night... for some reason), so I added it to the 'blacklist' in /etc/modprobe.d/alsa-base:

Code:

[left]. . . # Prevent abnormal drivers from grabbing index 0 options snd-bt87x index=-2 options cx88-alsa index=-2 options saa7134-alsa index=-2 options snd-atiixp-modem index=-2 options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-usb-audio index=-2 options snd-usb-usx2y index=-2 [b]options snd-intel8x0 index=1[/b] # Ubuntu #62691, enable MPU for snd-cmipci options snd-cmipci mpu_port=0x330 fm_port=0x388[/left]

By assigning it index=1 this basically prevents it from using index=0.

*edit*
Figured I might try to get the Wolfson functioning before the amp arrives, I get this bunch of errors with darklegion's asound.state for ALSA 1.0.13rc1:

Code:

[left]brodiepearce@ubuntu:~$ alsactl restore alsactl: set_control:991: warning: name mismatch (Headphone Playback Switch/PC Speaker Playback Switch) for control #9 alsactl: set_control:993: warning: index mismatch (0/0) for control #9 alsactl: set_control:995: failed to obtain info for control #9 (Operation not permitted)[/left]

 

[MCC]

This asound.state works great for me with Alsa 1.0.14. You might need to change

name 'Multi Track Internal Clock'
value '44100'

to

name 'Multi Track Internal Clock'
value '48000'

unless you do the fixes I detailed above.