[Alsa-user] Re: [linux-audio-user] quattro distortion under mandrake 9.0 - moredetails
iriXx wrote: any suggestions would be very gratefully received! Sorry. I forgot you weren't using the latest cvs. If you would like some help installing it again (after your last round) I could ssh in and do it for you. Let me know off list if you think that is a good idea. -- Patrick Shirkey - Boost Hardware Ltd. For the discerning hardware connoisseur Http://www.boosthardware.com Http://www.djcj.org - The Linux Audio Users guide Being on stage with the band in front of crowds shouting, "Get off! No! We want normal music!", I think that was more like acting than anything I've ever done. Goldie, 8 Nov, 2002 The Scotsman --- This SF.NET email is sponsored by: Thawte.com - A 128-bit supercerts will allow you to extend the highest allowed 128 bit encryption to all your clients even if they use browsers that are limited to 40 bit encryption. Get a guide here:http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0030en ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] /dev/sequencer no such device, but audio plays ok?
I've installed the alsa drivers and can play and record sound ok, but /dev/sequencer cannot be opened - I get the ENODEV no such device error. I have a Realtek ALC650 audio on motherboard and RedHat 8.0 (which misconfigures the audio in /etc/modules.conf as alias sound-slot-0 via82cxxx_audio, a known bug). From the Realtek website I was directed to use the alsa drivers and to set my /etc/modules.conf with alias char-major-116 snd options snd major=116 cards_limit=1 #--- Via8233 --- alias snd-card-0 snd-via82xx options snd-via82xx index=0 id="Via8233" #--- 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 #= looking at snddevices it will have created /dev/sequencer with these values mknod -m 666 /dev/sequencer c 14 1 but here I lose the trail - is there a fundamental problem here - am I talking to some missing hardware? I configured built and installed (the driver with --with-sequencer=yes) alsa-driver-0.9.0rc5 alsa-lib-0.9.0rc6 alsa-utils-0.9.0rc6 and ran snddevices from config.log of alsa-driver $ ./configure --with-sequencer=yes --with-debug=detect #define CONFIG_SND_VERSION "0.9.0rc5" $cat junk /dev/sequencer cat: /dev/sequencer: No such device #/sbin/lspci -v -v 00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233 AC97 Audio Controller (rev 50) Subsystem: Micro-star International Co Ltd: Unknown device 4720 Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- SERR- Interrupt: pin C routed to IRQ 11 Region 0: I/O ports at dc00 [size=256] Capabilities: [c0] Power Management version 2 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Any insights would be much appreciated, as Rosegarden refuses to run at all unless it can talk to /dev/sequencer. Richard Shann --- This SF.NET email is sponsored by: Thawte.com - A 128-bit supercerts will allow you to extend the highest allowed 128 bit encryption to all your clients even if they use browsers that are limited to 40 bit encryption. Get a guide here:http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0030en ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] How to get center/LFE working on SBLive 5.1 (SB0100)
I post this message some days ago, but nobody answer to me. I have SBLive 5.1 card: - card: EMU10K1 rev 7 model 0x8064 found, IO at 0xd000-0xd01f, IRQ 11; - card ac97 codec: AC97 Audio codec, id: 0x454d:0x4328 (Unknown) - id of card SB0100 - on other side with black ink "02DCER2420 AJW" - analog speakers; Problem is: center speaker don't work or work strange. 1) In ALSA OSS emulation I can't hear center (may be and LFE, I can't determine, because LFE may be mixed from other channels). Why? 2) With latest cvs mplayer with "-ao alsa9:surround51" I hear all 6 channels, but while seeking channels map changes to different 3 positions. 3) What means "Surround", "Surround Digital", "Center" in alsamixer? Why I must change "Surround" and "Center" to hear center in 2) case. 4) Why while initialization I hear loud click? What I should do with my card? p.s. emu10k1 OSS driver not help, it works like OSS emulation, but I found some intresting thing: if route center channel to left or right front or rear channel, I hear it in this speaker; but if I route sound for example from left front speaker to center, center speaker not produce any sound. p.p.s. Sorry for my bad English. -- Vladimir I. Umnov mailto:[EMAIL PROTECTED] --- This SF.NET email is sponsored by: Thawte.com - A 128-bit supercerts will allow you to extend the highest allowed 128 bit encryption to all your clients even if they use browsers that are limited to 40 bit encryption. Get a guide here:http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0030en ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Delta/66 - "multi-channel" ?
Hello - My Delta/66 has four outs - currently I'm using the first two as the usual left/right channels. Would it be possible to run another audio program and send the output to the other two channels, at the same time another program is using the first two? How would I do this using ALSA? How about mixing sound from two stereo audio sources down to the same two channels? Thanks for any suggestions --- This SF.NET email is sponsored by: Thawte.com - A 128-bit supercerts will allow you to extend the highest allowed 128 bit encryption to all your clients even if they use browsers that are limited to 40 bit encryption. Get a guide here:http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0030en ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] via82xx.o: init_module: No such device
On 17 Jan 2003 19:55:21 -0500 Mike Duncan <[EMAIL PROTECTED]> wrote: > Hello all, > I just got a Gigabyte-7ZXE mobo with a VIA82686B chipset. The old board > I had was a Chaintech, with a VIA82686A and I got alsa to work perfectly > with it, even with 2 sound cards (a SB16 too). > > I cannot seem to figure out why I am receiving this error after a > perfect build of the driver, libs, and tools v0.90rc6. I have tried > looking through the archives, but have come up with only one post that > seems like mine, but after trying what they said, no luck. > > * Here is the error... > snd-via82xx /lib/modules/2.4.20/kernel/sound/pci/snd-via82xx.o: > init_module: No such device > Hint: insmod errors can be caused by incorrect module parameters, > including invalid IO or IRQ parameters > /lib/modules/2.4.20/kernel/sound/pci/snd-via82xx.o: insmod > /lib/modules/2.4.20/kernel/sound/pci/snd-via82xx.o failed > /lib/modules/2.4.20/kernel/sound/pci/snd-via82xx.o: insmod snd-via82xx > failed Have you set 'PnP OS' or something like that no in the BIOS? > * So, I get the IRQ from lspci -v... > 00:07.5 Multimedia audio controller: VIA Technologies, Inc. AC97 Audio > Controller (rev 50) > Subsystem: Giga-byte Technology: Unknown device a002 > Flags: medium devsel, IRQ 10 > I/O ports at dc00 [size=256] > I/O ports at d800 [size=4] > I/O ports at d400 [size=4] > Capabilities: [c0] Power Management version 2 > > > * And try again with this in my /etc/modules.conf (added the IRQ=10)... > alias char-major-116 snd > alias char-major-14 soundcore > options snd major=116 cards_limit=1 > alias snd-card-0 snd-via82xx > options snd-via82xx index=0 id="AC97" irq=10 > 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 > > > * And I get this... > snd-via82xx /lib/modules/2.4.20/kernel/sound/pci/snd-via82xx.o: invalid > parameter parm_irq > /lib/modules/2.4.20/kernel/sound/pci/snd-via82xx.o: insmod > /lib/modules/2.4.20/kernel/sound/pci/snd-via82xx.o failed > /lib/modules/2.4.20/kernel/sound/pci/snd-via82xx.o: insmod snd-via82xx > failed > > > > Basically, I think it may be the IO settings, but I do not know the > correct format for the IRQ, nor the IO settings in the modules.conf file > (apparently). Some help would be appreciated. TIA. Your module doesn't support IO or IRQ parameters: see '/sbin/modinfo snd-via82xx' or the INSTALL file or http://www.alsa-project.org/alsa-doc/ for your card. The 'hint' just wasn't pertinent in this case :) HTH, -Frans --- This SF.NET email is sponsored by: Thawte.com - A 128-bit supercerts will allow you to extend the highest allowed 128 bit encryption to all your clients even if they use browsers that are limited to 40 bit encryption. Get a guide here:http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0030en ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] snd-emu10k1 + TeamSpeak + Quake3
Hi, im having trouble with alsa using quake3 together with teamspeak on my soundblaster live. With oss/free this works though. So, TeamSpeak is a voice-communication tool (www.teamspeak.org), which allows you to speak to a group of people while e.g. gaming. Since it is used so much with games, it is not unusuall to try and launch quake3 while TeamSpeak is running. The Problem is: quake cant load up if I try so, at "sound initialization" it hangs up. With the oss/kernel modules it workes fine (quake3 can open /dev/dsp although TeamSpeak has already done so)... The problem seems to be, that both try and open /dev/dsp for read AND write (TeamSpeak has to do so, of course, why quake does it this way i dont know). Somewhere I read about /dev/adsp being a playback only device, so I thought I could force quake3 to use that, and have /dev/dsp for TeamSpeak. The Problem is: /dev/adsp doesnt seem to work at all, all programms I tell to use it (xmms, zinf, xine...) tell me my sound setup is broken, or /dev/adsp no such device (yes /dev/adsp exists). Why does quake3 load fine with oss/free but not on alsa0.9 ? Whats the matter with /dev/adsp ? Is there no such thing with snd-emu10k1 ? How should I go about to fix this problem (without reverting to use oss/free) ? Thanks in advance Peter alsa-oss-0.9.0_rc1 alsa-lib-0.9.0_rc6 alsa-driver-0.9.0_rc6 alsa-utils-0.9.0_rc6 Soundblaster Live! Newest Versions of Quake3 + TeamSpeak -- Examinations are formidable even to the best prepared, for even the greatest fool may ask more the the wisest man can answer. -- C.C. Colton --- This SF.NET email is sponsored by: Thawte.com - A 128-bit supercerts will allow you to extend the highest allowed 128 bit encryption to all your clients even if they use browsers that are limited to 40 bit encryption. Get a guide here:http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0030en ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] snd-emu10k1 + TeamSpeak + Quake3
On Sat, 18 Jan 2003, Peter Kirk wrote: > Hi, > > im having trouble with alsa using quake3 together with teamspeak on my > soundblaster live. With oss/free this works though. > > So, TeamSpeak is a voice-communication tool (www.teamspeak.org), which allows > you to speak to a group of people while e.g. gaming. Since it is used so much > with games, it is not unusuall to try and launch quake3 while TeamSpeak is > running. The Problem is: quake cant load up if I try so, at "sound > initialization" it hangs up. With the oss/kernel modules it workes fine > (quake3 can open /dev/dsp although TeamSpeak has already done so)... The > problem seems to be, that both try and open /dev/dsp for read AND write > (TeamSpeak has to do so, of course, why quake does it this way i dont know). > Somewhere I read about /dev/adsp being a playback only device, so I thought I > could force quake3 to use that, and have /dev/dsp for TeamSpeak. The Problem > is: /dev/adsp doesnt seem to work at all, all programms I tell to use it > (xmms, zinf, xine...) tell me my sound setup is broken, or /dev/adsp no such > device (yes /dev/adsp exists). > > Why does quake3 load fine with oss/free but not on alsa0.9 ? > Whats the matter with /dev/adsp ? Is there no such thing with snd-emu10k1 ? > How should I go about to fix this problem (without reverting to use oss/free) > ? There's not an easy fix. This example shows exactly the broken API design as OSS. We don't know at open time, if application requests the device for read or write or duplex operation, so we assume that all directions are wanted. There is no /dev/adsp device for EMU10K1. Anyway, we have proc interface where you can tell to driver that only playback is wanted. Try this: % echo "quake 0 0 direct" > /proc/asound/card0/pcm0p/oss % echo "quake 0 0 disable" > /proc/asound/card0/pcm0c/oss Replace quake with real application name (use ps command to determine it at runtime). Jaroslav - Jaroslav Kysela <[EMAIL PROTECTED]> Linux Kernel Sound Maintainer ALSA Project, SuSE Labs --- This SF.NET email is sponsored by: Thawte.com - A 128-bit supercerts will allow you to extend the highest allowed 128 bit encryption to all your clients even if they use browsers that are limited to 40 bit encryption. Get a guide here:http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0030en ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Re: [linux-audio-user] quattro distortion under mandrake 9.0 - moredetails
Patrick Shirkey wrote: iriXx wrote: any suggestions would be very gratefully received! Sorry. I forgot you weren't using the latest cvs. If you would like some help installing it again (after your last round) I could ssh in and do it for you. Let me know off list if you think that is a good idea. The other option is that you could try going through the manual install again. This time run some checks before hand to make sure things are not going to conflict. First find out where exactly mandrake has put the alsa modules. Then remove them. Now make sure you have a /lib/modules/2.4.x/kernel If you do then install the alsa packages And do a rmmod alsa-modules modprobe snd-card; modprobe snd-mixer-oss,modprobe snd-pcm-oss; modprobe snd-seq-oss Now you definitely know that other modules can't be conflicting. If the sound is okay after doing this then next time you reboot you will know if it is boot scripts that are causing you alsa troubles or not. Then if they do just take out the offending lines. Although I've had no troubles for a while by just ignoring them until I find out what they mean :) -- Patrick Shirkey - Boost Hardware Ltd. For the discerning hardware connoisseur Http://www.boosthardware.com Http://www.djcj.org - The Linux Audio Users guide Being on stage with the band in front of crowds shouting, "Get off! No! We want normal music!", I think that was more like acting than anything I've ever done. Goldie, 8 Nov, 2002 The Scotsman --- This SF.NET email is sponsored by: Thawte.com - A 128-bit supercerts will allow you to extend the highest allowed 128 bit encryption to all your clients even if they use browsers that are limited to 40 bit encryption. Get a guide here:http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0030en ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
Re: [Alsa-user] snd-emu10k1 + TeamSpeak + Quake3
Am Samstag, 18. Januar 2003 19:42 schrieb Jaroslav Kysela: > On Sat, 18 Jan 2003, Peter Kirk wrote: > > Hi, > > > > im having trouble with alsa using quake3 together with teamspeak on my > > soundblaster live. With oss/free this works though. [...] > > There's not an easy fix. This example shows exactly the broken API design > as OSS. We don't know at open time, if application requests the device for > read or write or duplex operation, so we assume that all directions are > wanted. There is no /dev/adsp device for EMU10K1. > > Anyway, we have proc interface where you can tell to driver that only > playback is wanted. Try this: > > % echo "quake 0 0 direct" > /proc/asound/card0/pcm0p/oss > % echo "quake 0 0 disable" > /proc/asound/card0/pcm0c/oss > > Replace quake with real application name (use ps command to determine it > at runtime). > > Jaroslav Yeah =) you helped me to solve my problem, thanks a lot I even understood your explanation, one thing I dont understand though...Why does it work with oss/free, which should have the same problem as alsa/oss ?? thanks again Peter -- I don't mind arguing with myself. It's when I lose that it bothers me. -- Richard Powers --- This SF.NET email is sponsored by: Thawte.com - A 128-bit supercerts will allow you to extend the highest allowed 128 bit encryption to all your clients even if they use browsers that are limited to 40 bit encryption. Get a guide here:http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0030en ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
[Alsa-user] Cannot get SoundBlaster Platinum Live!Drive's line-in or headphone-out to work
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, About 3 years ago I bought the first version of the Creative SoundBlaster Platinum, the one that comes with the Live!Drive. I do not belive it is a 5.1, it does not say that anywhere on the box, enven though it has both "front" and "rear" speaker outs. The live drive has: SPIDIF in, SPIDIF out, 1/4 Headphone (out) with level control knob, 1/4 Line in with level control know, MIDI in, MIDI out. Nothing else, does not have an IR port. The PCI card is , "rear" out, "front" out (that's what I hook my speakers up too, no sound comes out of "rear" out), microphone in, line in (or line int /mic in) and digital out. (I think that's all, it's hard to see and that's how I remember it being. I know the mic in, line in and "front" out all work on the PCI card. Work as in, I can hear mic or line audio in comming out of the "Front" out. I really really want to be able to use the headphone out and line in on my Live!Drive, and every indication is tht this _can_ be done, but I can't find clear-cut instructions on this. Before we go any further I have actually wondered it possibly my card is bad. Don't rule it out. I don't have Windows so I cna't see if it's working right in Windows (I know it did when I first bought the card 3 years ago). First, I'm using SuSE 8.1 (I've tried this on many versions of SuSE though), alsa-0.9.0.cvs20020903-13, kernel 2.4.19-4GB. The relative entries for /etc/modules.conf: options snd-emu10k1 snd_enable=1 snd_extin=0x3fc3 snd_extout=0x1fcf snd_index=0 options snd snd_cards_limit=1 snd_major=116 # bSAa.6S9Jn9_7V7B:CT4760 SBLive! alias snd-card-0 snd-emu10k1 # YaST2: sound system dependent part # 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-0-11 snd-mixer-oss alias sound-service-0-12 snd-pcm-oss I belive those are correct, they match the document numbers for * Creative Card w/Digital out + LiveDrive [0x3fc3/0x1fcf] There is a CD digital in on the PCI card, I'm pretty sure, but I would think yast detects this correctly. Maybe not, what do you think? Anyway, let's start with the headphone. For this I am using the console version of alsamixer. Inside alsamixer (and this is the order they're shown in ): This info is shown at the top: Card: Sound Blaster Live! Chip: TriTech TR28602 Master: this changes the master volume Master Mono: this doesn't seem to have an effect on anythign, and is muted by default Headphone LFE 1: This is 0/0 and muted by default. unmuting it does nothing. It is not possible to increase it's volume with the up arrow key, as is the case with all known-good adjustments (that's how I raise master, for example) Headphone 1 unmuted by default. I can adjust it but it does not seem to effect anything. Headphone Center 1 0/0 by default, level cannot be adjust. Tone Bass Treble PCM Surround Digital 100/100 by default Surround Digital Capture unmuted 0/0 by default Cener unmuted 100/100 default LFE ditto Wave ditto Wave Capture unmuted 0/0 default Wave Center ditto Wave LFE ditto Wave Surround ditto Music 100/100 unmuted default Music Capture 00 unmuted default Line 0/0 muted Line LiveDrive changing this does not help (yes, I tried setting it as the "capture' source) Line LiveDrive Capture level adjustment does not help. Can't be set to "capture" Line LiveDrive 1 adjusting this doens't help CD Mic Mic Boost Video Phone There are a host of others: IEC958 C IEC958 C IEC958 O IEC958 O IEC958 O PC Speak AuxCaptureMix Mix Mono AC97 AC97 Cap EMU10K1 EMU10K1 EMU10K1 EMU10K1 EMU10K1 EMU10K1 EMU10K1 EMU10K1 EMU10K1 EMU10K1 External SB Live I have tried many of them, in many dirrent combinations of capture and level. I tried most of these things with the volume knobs on the Live Drive either full blasst 100% or 80% By the way, what is the meaning of the 6 dashes above certain channelsliek Surround Wave and Music? I have tried these instrcutions, among others: http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg04145.html They did not help either. Any help or ideas about what I could try would really, really be appreciated. Thank you! JW - -- - Jonathan Wilson Cedar Creek Software http://www.cedarcreeksoftware.com -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQE+Kf2OQ5u80xXOLBcRAsMjAJ4/UywEospGvP6E+D3Oq7icj1aicACghsS8 hYl7Qe5VLPDYvUni/VNKiME= =u2+1 -END PGP SIGNATURE- --- This SF.NET email is sponsored by: Thawte.com - A 128-bit supercerts will allow you to extend the highest allowed 128 bit encryption to all your clients even if they use browsers that are limited to 40 bit encryption. Get a guide here:http://ads.sourceforge.net/cgi-bin/r
Re: [Alsa-user] via82xx.o: init_module: No such device
> > * Here is the error... > > snd-via82xx /lib/modules/2.4.20/kernel/sound/pci/snd-via82xx.o: > > init_module: No such device > > Hint: insmod errors can be caused by incorrect module parameters, > > including invalid IO or IRQ parameters > > /lib/modules/2.4.20/kernel/sound/pci/snd-via82xx.o: insmod > > /lib/modules/2.4.20/kernel/sound/pci/snd-via82xx.o failed > > /lib/modules/2.4.20/kernel/sound/pci/snd-via82xx.o: insmod snd-via82xx > > failed > > Have you set 'PnP OS' or something like that no in the BIOS? After I read this, I checked the BIOS and found out that this was set to YES. After changing it to NO and rebooting, I came up empty handed. Same error has above. I did also notice in the BIOS, set by default to DISABLED, a setting for Sound blaster with relevant settings. Should this be enabled? > Your module doesn't support IO or IRQ parameters: see > '/sbin/modinfo snd-via82xx' or the INSTALL file or > http://www.alsa-project.org/alsa-doc/ for your card. The 'hint' just > wasn't pertinent in this case :) I think I see why, because the IRQ and other info would be given by the chip I guess. I am still at no end. I will try some more BIOS tweaking, but if anyone has any more info, it would be greatly appreciated. Thanks. -- Mike Duncan [EMAIL PROTECTED] http://www.randomtask.net --- This SF.NET email is sponsored by: Thawte.com - A 128-bit supercerts will allow you to extend the highest allowed 128 bit encryption to all your clients even if they use browsers that are limited to 40 bit encryption. Get a guide here:http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0030en ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user
