May 2007 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Jackd dies with snd-usb-usx2y (DRIVER NT error)
by Michael Bourgeous 15 May '07

15 May '07

Hi, I recently upgraded my system from 32-bit Ubuntu Edgy, with a custom hard-realtime-enabled 2.6.19 kernel, to 64-bit Ubuntu Feisty, with a hard-realtime-enabled 2.6.20 kernel. I also switched from an Athlon XP 2600 to an Athlon 64 X2 4600. In every way my new system should be faster, but jackd+usx2y worked on the last system, and doesn't work now. Here's what I'm doing to test the audio setup: ---------- sudo chrt -f 90 `pidof IRQ-23` # usb irq chrt -v -f 70 jackd -d alsa -dhw:1,2 -p 64 -n 2 # also tried defaults, and values up to -p 256 -n 8 chrt -v -f 25 zynaddsubfx ---------- After a short time (< 1 or 2 minutes) of switching instruments and playing notes with zynaddsubfx, all audio stops and jackd dies with the following error: ---------- ALSA: poll time out, polled for 7998858 usecs DRIVER NT: could not run driver cycle jack main caught signal 12 no message buffer overruns ---------- There is a corresponding message in the kernel log: ---------- [12546.132286] ALSA [...]/usx2yhwdeppcm.c:94: activ frame status -70. Most propably some hardware problem. ---------- I am loading the snd-usb-usx2y module with the nrpacks=1 option. I've gone through the Ubuntu HOWTOs I used to get things working on my last system (to some of which I have even contributed), and I am still having this problem. I already tried on alsa-user, so I'm taking this to the developer list (since I plan to help fix this if anybody will point me in the right direction). Additional note: I tried modifying the source code of the driver, commenting out the return statement after the "active status" error message. This began producing a new error message: "shouldn't be here with count=xx". So, I commented out the "return -EPIPE" after that error message, and now jackd doesn't crash. There are noticeable audio clicks whenever one of these errors occurs, but the driver seems to have no problem recovering, as audio continues. This error happens with the stock Ubuntu kernels as well. Thanks in advance, Mike Bourgeous

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[alsa-devel] [Alsa-devel] Crosstool for PDA iPAQ hx4700
by K 15 May '07

15 May '07

Hi, I want to port a audio program on my hx4700, So I will use a prebuilt toolchain. I have downloaded ftp://ftp.handhelds.org/projects/toolchain/arm-linux-gcc-3.4.1.tar.bz2 and I extract this file to /usr/local/arm I have downloaded alsa-lib from ftp://ftp.alsa-project.org/pub/lib/stable/alsa-lib-0.0.9.tar.gz I have compiled it : #./configure --prefix=/usr/local/arm-linux --host=arm-linux #make #make install after I compile my program : #arm-linux-gcc -Wa,-mfpu=fpa11 -mcpu=xscale record.c -lasound -I /usr/local/arm/3.4.1/include/ /usr/local/arm/3.4.1/lib/gcc/arm-linux/3.4.1/../../../../arm-linux/bin/ld: cannot find -lasound collect2: ld returned 1 exit status Do you have an idea ? Can you help me ? I want to port a Vumeter, Peakmeter and other audio tools on hx4700. Thank you -- K

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Re: [alsa-devel] 2.6.22-rc1 lost snd_4236 device
by Rene Herman 15 May '07

15 May '07

On 05/13/2007 06:23 PM, Pete Clements wrote: > Fyi: > With 2.6.22-rc1 get following when loading sound device module. Introduced > with 2.6.21-git16. > > FATAL: Error inserting snd_cs4236 (/lib/modules/2.6.22-rc1/kernel/sound/isa/cs423x/snd-cs4236.ko): No such device > FATAL: Error running install command for snd_cs4236 Yup; the conversion of the mixed legacy/pnp ALSA drivers to the isa_bus framework is not quite right. Wants the attached. Takashi, this needs to go to Linus. Can you make sure it gets there? This is against -linus HEAD. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> Rene. diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index 214d65d..f471f8a 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -109,6 +109,7 @@ module_param_array(wssdma, int, NULL, 0444); MODULE_PARM_DESC(wssdma, "DMA for CMI8330 WSS driver."); #ifdef CONFIG_PNP +static int isa_registered; static int pnp_registered; #endif @@ -686,14 +687,18 @@ static int __init alsa_card_cmi8330_init(void) int err; err = isa_register_driver(&snd_cmi8330_driver, SNDRV_CARDS); - if (err < 0) - return err; #ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + err = pnp_register_card_driver(&cmi8330_pnpc_driver); if (!err) pnp_registered = 1; + + if (isa_registered) + err = 0; #endif - return 0; + return err; } static void __exit alsa_card_cmi8330_exit(void) @@ -701,8 +706,10 @@ static void __exit alsa_card_cmi8330_exit(void) #ifdef CONFIG_PNP if (pnp_registered) pnp_unregister_card_driver(&cmi8330_pnpc_driver); + + if (isa_registered) #endif - isa_unregister_driver(&snd_cmi8330_driver); + isa_unregister_driver(&snd_cmi8330_driver); } module_init(alsa_card_cmi8330_init) diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 87f1392..1a14f33 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -127,6 +127,7 @@ module_param_array(dma2, int, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for " IDENT " driver."); #ifdef CONFIG_PNP +static int isa_registered; static int pnpc_registered; #ifdef CS4232 static int pnp_registered; @@ -770,9 +771,9 @@ static int __init alsa_card_cs423x_init(void) int err; err = isa_register_driver(&cs423x_isa_driver, SNDRV_CARDS); - if (err < 0) - return err; #ifdef CONFIG_PNP + if (!err) + isa_registered = 1; #ifdef CS4232 err = pnp_register_driver(&cs4232_pnp_driver); if (!err) @@ -781,8 +782,14 @@ static int __init alsa_card_cs423x_init(void) err = pnp_register_card_driver(&cs423x_pnpc_driver); if (!err) pnpc_registered = 1; -#endif /* CONFIG_PNP */ - return 0; +#ifdef CS4232 + if (pnp_registered) + err = 0; +#endif + if (isa_registered) + err = 0; +#endif + return err; } static void __exit alsa_card_cs423x_exit(void) @@ -794,8 +801,9 @@ static void __exit alsa_card_cs423x_exit(void) if (pnp_registered) pnp_unregister_driver(&cs4232_pnp_driver); #endif -#endif /* CONFIG_PNP */ - isa_unregister_driver(&cs423x_isa_driver); + if (isa_registered) +#endif + isa_unregister_driver(&cs423x_isa_driver); } module_init(alsa_card_cs423x_init) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index d2a9c7d..f7732bf 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -2036,7 +2036,9 @@ module_param_array(dma2, int, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA 2 # for ES18xx driver."); #ifdef CONFIG_PNP -static int pnp_registered, pnpc_registered; +static int isa_registered; +static int pnp_registered; +static int pnpc_registered; static struct pnp_device_id snd_audiodrive_pnpbiosids[] = { { .id = "ESS1869" }, @@ -2466,18 +2468,22 @@ static int __init alsa_card_es18xx_init(void) int err; err = isa_register_driver(&snd_es18xx_isa_driver, SNDRV_CARDS); - if (err < 0) - return err; - #ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + err = pnp_register_driver(&es18xx_pnp_driver); if (!err) pnp_registered = 1; + err = pnp_register_card_driver(&es18xx_pnpc_driver); if (!err) pnpc_registered = 1; + + if (isa_registered || pnp_registered) + err = 0; #endif - return 0; + return err; } static void __exit alsa_card_es18xx_exit(void) @@ -2487,8 +2493,9 @@ static void __exit alsa_card_es18xx_exit(void) pnp_unregister_card_driver(&es18xx_pnpc_driver); if (pnp_registered) pnp_unregister_driver(&es18xx_pnp_driver); + if (isa_registered) #endif - isa_unregister_driver(&snd_es18xx_isa_driver); + isa_unregister_driver(&snd_es18xx_isa_driver); } module_init(alsa_card_es18xx_init) diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 3e46572..0220cdb 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -135,6 +135,7 @@ struct snd_interwave { #ifdef CONFIG_PNP +static int isa_registered; static int pnp_registered; static struct pnp_card_device_id snd_interwave_pnpids[] = { @@ -934,15 +935,18 @@ static int __init alsa_card_interwave_init(void) int err; err = isa_register_driver(&snd_interwave_driver, SNDRV_CARDS); - if (err < 0) - return err; #ifdef CONFIG_PNP - /* ISA PnP cards */ + if (!err) + isa_registered = 1; + err = pnp_register_card_driver(&interwave_pnpc_driver); if (!err) pnp_registered = 1; + + if (isa_registered) + err = 0; #endif - return 0; + return err; } static void __exit alsa_card_interwave_exit(void) @@ -950,8 +954,9 @@ static void __exit alsa_card_interwave_exit(void) #ifdef CONFIG_PNP if (pnp_registered) pnp_unregister_card_driver(&interwave_pnpc_driver); + if (isa_registered) #endif - isa_unregister_driver(&snd_interwave_driver); + isa_unregister_driver(&snd_interwave_driver); } module_init(alsa_card_interwave_init) diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 48743eb..61a323c 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -92,6 +92,7 @@ module_param_array(opl3sa3_ymode, int, NULL, 0444); MODULE_PARM_DESC(opl3sa3_ymode, "Speaker size selection for 3D Enhancement mode: Desktop/Large Notebook/Small Notebook/HiFi."); #ifdef CONFIG_PNP +static int isa_registered; static int pnp_registered; static int pnpc_registered; #endif @@ -967,17 +968,22 @@ static int __init alsa_card_opl3sa2_init(void) int err; err = isa_register_driver(&snd_opl3sa2_isa_driver, SNDRV_CARDS); - if (err < 0) - return err; #ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + err = pnp_register_driver(&opl3sa2_pnp_driver); if (!err) pnp_registered = 1; + err = pnp_register_card_driver(&opl3sa2_pnpc_driver); if (!err) pnpc_registered = 1; + + if (isa_registered || pnp_registered) + err = 0; #endif - return 0; + return err; } static void __exit alsa_card_opl3sa2_exit(void) @@ -987,8 +993,9 @@ static void __exit alsa_card_opl3sa2_exit(void) pnp_unregister_card_driver(&opl3sa2_pnpc_driver); if (pnp_registered) pnp_unregister_driver(&opl3sa2_pnp_driver); + if (isa_registered) #endif - isa_unregister_driver(&snd_opl3sa2_isa_driver); + isa_unregister_driver(&snd_opl3sa2_isa_driver); } module_init(alsa_card_opl3sa2_init) diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 2a19b0a..c4ba24b 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -129,6 +129,7 @@ MODULE_PARM_DESC(seq_ports, "Number of sequencer ports for WaveTable synth."); #endif #ifdef CONFIG_PNP +static int isa_registered; static int pnp_registered; #endif @@ -702,15 +703,18 @@ static int __init alsa_card_sb16_init(void) int err; err = isa_register_driver(&snd_sb16_isa_driver, SNDRV_CARDS); - if (err < 0) - return err; #ifdef CONFIG_PNP - /* PnP cards at last */ + if (!err) + isa_registered = 1; + err = pnp_register_card_driver(&sb16_pnpc_driver); if (!err) pnp_registered = 1; + + if (isa_registered) + err = 0; #endif - return 0; + return err; } static void __exit alsa_card_sb16_exit(void) @@ -718,8 +722,9 @@ static void __exit alsa_card_sb16_exit(void) #ifdef CONFIG_PNP if (pnp_registered) pnp_unregister_card_driver(&sb16_pnpc_driver); + if (isa_registered) #endif - isa_unregister_driver(&snd_sb16_isa_driver); + isa_unregister_driver(&snd_sb16_isa_driver); } module_init(alsa_card_sb16_init) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 08c1497..9ea417b 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -69,7 +69,9 @@ module_param_array(dma, int, NULL, 0444); MODULE_PARM_DESC(dma, "DMA # for SoundScape driver."); #ifdef CONFIG_PNP +static int isa_registered; static int pnp_registered; + static struct pnp_card_device_id sscape_pnpids[] = { { .id = "ENS3081", .devs = { { "ENS0000" } } }, { .id = "" } /* end */ @@ -1405,22 +1407,21 @@ static struct pnp_card_driver sscape_pnpc_driver = { static int __init sscape_init(void) { - int ret; + int err; - /* - * First check whether we were passed any parameters. - * These MUST take precedence over ANY automatic way - * of allocating cards, because the operator is - * S-P-E-L-L-I-N-G it out for us... - */ - ret = isa_register_driver(&snd_sscape_driver, SNDRV_CARDS); - if (ret < 0) - return ret; + err = isa_register_driver(&snd_sscape_driver, SNDRV_CARDS); #ifdef CONFIG_PNP - if (pnp_register_card_driver(&sscape_pnpc_driver) == 0) + if (!err) + isa_registered = 1; + + err = pnp_register_card_driver(&sscape_pnpc_driver); + if (!err) pnp_registered = 1; + + if (isa_registered) + err = 0; #endif - return 0; + return err; } static void __exit sscape_exit(void) @@ -1428,8 +1429,9 @@ static void __exit sscape_exit(void) #ifdef CONFIG_PNP if (pnp_registered) pnp_unregister_card_driver(&sscape_pnpc_driver); + if (isa_registered) #endif - isa_unregister_driver(&snd_sscape_driver); + isa_unregister_driver(&snd_sscape_driver); } module_init(sscape_init); diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 75673f7..83c2fc4 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -86,6 +86,7 @@ module_param_array(use_cs4232_midi, bool, NULL, 0444); MODULE_PARM_DESC(use_cs4232_midi, "Use CS4232 MPU-401 interface (inaccessibly located inside your computer)"); #ifdef CONFIG_PNP +static int isa_registered; static int pnp_registered; static struct pnp_card_device_id snd_wavefront_pnpids[] = { @@ -706,14 +707,18 @@ static int __init alsa_card_wavefront_init(void) int err; err = isa_register_driver(&snd_wavefront_driver, SNDRV_CARDS); - if (err < 0) - return err; #ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + err = pnp_register_card_driver(&wavefront_pnpc_driver); if (!err) pnp_registered = 1; + + if (isa_registered) + err = 0; #endif - return 0; + return err; } static void __exit alsa_card_wavefront_exit(void) @@ -721,8 +726,9 @@ static void __exit alsa_card_wavefront_exit(void) #ifdef CONFIG_PNP if (pnp_registered) pnp_unregister_card_driver(&wavefront_pnpc_driver); + if (isa_registered) #endif - isa_unregister_driver(&snd_wavefront_driver); + isa_unregister_driver(&snd_wavefront_driver); } module_init(alsa_card_wavefront_init)

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[alsa-devel] Multichannel capture support in E-MU 1212
by dustin＠seznam.cz 15 May '07

15 May '07

Hello, I am working together with Ctirad on supporting more capture channels in E-MU1212. So far we have managed to increase the number of channels from the default of 8 to 16 (A_FXWC2 = 0xffffffff, added new snd_emu10k1_audigy_dsp_convert_32_to_2x16 calls in emufx.c), there is still a number of issues to solve. I would like to get your opinion on the form of adding this support to the module. The options are either hard-coding the 16 capture channels into the code, assuming that users would prefer as many channels as possible and do the routing in jack, or introducing a module parameter emu1010-capture-channels (range e.g. 1 to 16 - untested yet) and leaving the default number at the existing 8. Furthermore, the PCMs other than efx seem to be of no use for the EMU-1x1x models, probably fairly confusing for new users. Would you find it OK, if they were removed for this specific card (if (emu->card_capabilities->emu1010)), keeping only the playback and capture PCM id 2? Thanks a lot for your suggestions. Based on them, I will prepare the patch (with comments on what we found out about the card - it is a pitty that the driver has almost no big-picture comments on the supported cards' specifics, making new developers spend a number of long nights going through the nontrivial code and guessing). Pavel.

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[alsa-devel] intel8x0 / ALC655 no-capture solved
by Vegard Nossum 15 May '07

15 May '07

Hello, I've successfully debugged a problem I've had with being unable to record anything from microphones or line-in. Detailed information about the bug along with a temporary work-around can be found here: http://bugtrack.alsa-project.org/alsa-bug/view.php?id=3066 Now I'd just like to inform about the presence of the bug (at least I'm pretty sure it's a bug), and that it's solved by the means of a work-around. I'd be very happy to see a fix in the *code* for this, however. I'd even like to try it myself (heck, it's just a register write), but I'm not certain WHERE/WHEN to make the write, under which conditions, and for which range of cards (ie. is it specific to the ALC655 or ALC*, or perhaps Realtek chips, or should the fix be made for the Intel driver, or all AC'97 cards?). Guidance would be appreciated :) Maybe it's even as easy as a card config option, but I wouldn't know. Thanks. Kinds regards, Vegard Nossum PS: I'm not on the list, so please CC any replies.

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Re: [alsa-devel] [PATCH] Coping with short replies in usbmixer
by Thomas Reitmayr 15 May '07

15 May '07

Hi, here is it again and signed off. This patch makes sure that short USB replies are treated as an error when requesting the value of a certain mixer control. Signed-off-by: Thomas Reitmayr <thomas(a)devbase.at> --- sound/usb/usbmixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff -rup a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c --- a/sound/usb/usbmixer.c 2007-05-11 21:58:33.000000000 +0200 +++ b/sound/usb/usbmixer.c 2007-05-11 21:59:14.000000000 +0200 @@ -360,7 +360,7 @@ static int get_ctl_value(struct usb_mixe request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, cval->mixer->ctrlif | (cval->id << 8), - buf, val_len, 100) >= 0) { + buf, val_len, 100) >= val_len) { *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len)); return 0; }

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[alsa-devel] [PATCH] Include quirks from Ubuntu Dapper/Edgy/Feisty
by crimsun＠fungus.sh.nu 15 May '07

15 May '07

# HG changeset patch # User crimsun(a)Box.ncat.edu # Date 1179219408 14400 # Node ID f38d824e5df6fb7d9677cce13f91ad868e617fe4 # Parent 38fe50050f000c3a9660134a86f713dc23f7ec79 [PATCH] Include quirks from Ubuntu Dapper/Edgy/Feisty Included is a patch against current alsa-kernel hg tip adding quirks that Ubuntu Dapper/Edgy/Feisty contains. Signed-off-by: Daniel T Chen <crimsun(a)ubuntu.com> diff -r 38fe50050f00 -r f38d824e5df6 pci/ac97/ac97_patch.c --- a/pci/ac97/ac97_patch.c Mon May 14 18:40:07 2007 +0200 +++ b/pci/ac97/ac97_patch.c Tue May 15 04:56:48 2007 -0400 @@ -1782,6 +1782,11 @@ static unsigned int ad1981_jacks_blackli 0x10140534, /* Thinkpad X31 */ 0x10140537, /* Thinkpad T41p */ 0x10140554, /* Thinkpad T42p/R50p */ + 0x10140567, /* Thinkpad T43p 2668-G7U */ + 0x10140581, /* Thinkpad X41-2527 */ + 0x104380b0, /* Asus A7V8X-MX */ + 0x11790241, /* Toshiba Satellite A-15 S127 */ + 0x144dc01a, /* Samsung NP-X20C004/SEG */ 0 /* end */ }; diff -r 38fe50050f00 -r f38d824e5df6 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon May 14 18:40:07 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Tue May 15 04:56:48 2007 -0400 @@ -467,6 +467,8 @@ static struct snd_pci_quirk stac9200_cfg "Dell XPS M1710", STAC_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01cf, "Dell Precision M90", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d6, + "unknown Dell", STAC_REF), {} /* terminator */ }; diff -r 38fe50050f00 -r f38d824e5df6 pci/intel8x0.c --- a/pci/intel8x0.c Mon May 14 18:40:07 2007 +0200 +++ b/pci/intel8x0.c Tue May 15 04:56:48 2007 -0400 @@ -1799,6 +1799,18 @@ static struct ac97_quirk ac97_quirks[] _ }, { .subvendor = 0x1028, + .subdevice = 0x0186, + .name = "Dell Latitude D810", /* cf. Malone #41015 */ + .type = AC97_TUNE_HP_MUTE_LED + }, + { + .subvendor = 0x1028, + .subdevice = 0x0188, + .name = "Dell Inspiron 6000", + .type = AC97_TUNE_HP_MUTE_LED /* cf. Malone #41015 */ + }, + { + .subvendor = 0x1028, .subdevice = 0x0191, .name = "Dell Inspiron 8600", .type = AC97_TUNE_HP_ONLY @@ -1819,7 +1831,7 @@ static struct ac97_quirk ac97_quirks[] _ .subvendor = 0x103c, .subdevice = 0x088c, .name = "HP nc8000", - .type = AC97_TUNE_MUTE_LED + .type = AC97_TUNE_HP_MUTE_LED }, { .subvendor = 0x103c, @@ -1909,6 +1921,12 @@ static struct ac97_quirk ac97_quirks[] _ .subvendor = 0x10cf, .subdevice = 0x1253, .name = "Fujitsu S6210", /* STAC9750/51 */ + .type = AC97_TUNE_HP_ONLY + }, + { + .subvendor = 0x10cf, + .subdevice = 0x127e, + .name = "Fujitsu Lifebook C1211D", .type = AC97_TUNE_HP_ONLY }, { -- Daniel T. Chen crimsun(a)ubuntu.com GPG key: 0xC88ABDA3

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[alsa-devel] soft vol
by Michel Benoit 15 May '07

15 May '07

Hello Now that I have my pcm playback working with aplay with our at91 aSoc sound driver I would like to control the volume with softvol. I followed the instructions from http://alsa.opensrc.org/How_to_use_softvol_to_control_the_master_volume and I have the following asound.con file in /etc pcm.!default { type plug slave.pcm "softvol" } pcm.softvol { type softvol slave { pcm "plughw:0,0" } control { name "PCM volume" card 0 } } When I play a wav file directly to the device it works fine: # aplay -Dplughw:0,0 Front_Center.wav Playing WAVE 'Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono but when I try to use the default softvol device I get the following output (and no sound): # aplay Front_Center.wav Playing WAVE 'Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono ALSA sound/core/control.c:1244: unknown ioctl = 0xc2c45512 ALSA sound/core/control.c:1244: unknown ioctl = 0xc2c45512 ALSA sound/core/control.c:1244: unknown ioctl = 0xc2c45512 ALSA sound/core/control.c:1244: unknown ioctl = 0xc2c45512 ALSA sound/core/control.c:1244: unknown ioctl = 0xc2c45512 ... Does anyone have any ideas as to why this is happening? Michel

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[alsa-devel] Alsa 1.0.13 (and others) does not work on ALC880
by Eric 15 May '07

15 May '07

I installed Ubuntu Feisty on a D900t laptop : it contains a ALC880 audio chipset eric@gondor:~$ lspci |grep Audio 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller (rev 03) When modprobing the modules I get : eric@gondor:~$ sudo modprobe -v snd-hda-intel model=3stack-digout insmod /lib/modules/2.6.20-15-generic/kernel/sound/pci/hda/snd-hda-codec.ko insmod /lib/modules/2.6.20-15-generic/kernel/sound/pci/hda/snd-hda-intel.ko model=3stack-digout while /var/log/messages says : May 14 14:42:45 gondor kernel: [ 3699.070904] hda_intel: azx_get_response timeout, switching to polling mode... Feisty dist. is provided with alsa-1.0.13. I get no sound, here are the results of aplay : eric@gondor:~$ aplay ALSA lib pcm_direct.c:867:(snd_pcm_direct_initialize_slave) snd_pcm_hw_params_any failed ALSA lib pcm_dmix.c:876:(snd_pcm_dmix_open) unable to initialize slave aplay: main:550: Erreur d'ouverture audio: Argument invalide eric@gondor:~$ Here are some other infos : eric@gondor:~$ tail -2 /proc/asound/oss/sndstat 0: Realtek ALC880 1: SAA7134 Mixer amixer shows this : eric@gondor:~$ amixer Simple mixer control 'Headphone',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 228 [89%] Front Right: Playback 228 [89%] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 60 [47%] [-2144.00dB] [on] Front Right: Playback 60 [47%] [-2144.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 55 [43%] [-2304.00dB] [on] Front Right: Playback 55 [43%] [-2304.00dB] [on] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 64 [50%] [-2016.00dB] [on] Front Right: Playback 64 [50%] [-2016.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 56 [44%] [-2272.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 58 [46%] [-2208.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 60 [47%] [-2144.00dB] [on] Front Right: Playback 60 [47%] [-2144.00dB] [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 58 [46%] [-2208.00dB] [on] Front Right: Playback 58 [46%] [-2208.00dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 57 [45%] [-2240.00dB] [on] Front Right: Playback 57 [45%] [-2240.00dB] [on] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'PC Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 53 [42%] [-2368.00dB] [on] Front Right: Playback 53 [42%] [-2368.00dB] [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 31 [24%] [-3072.00dB] [on] Front Right: Capture 31 [24%] [-3072.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 27 [21%] [-3200.00dB] [on] Front Right: Capture 27 [21%] [-3200.00dB] [on] Simple mixer control 'Capture',2 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 24 [19%] [-3296.00dB] [on] Front Right: Capture 24 [19%] [-3296.00dB] [on] Simple mixer control 'Caller ID',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '6ch' Item0: '6ch' Simple mixer control 'Input Source',0 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',1 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',2 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Off-hook',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] aplay -l seems to work : eric@gondor:~$ aplay -l **** Liste des PLAYBACK périphériques **** carte 0: Intel [HDA Intel], périphérique 0 : ALC880 Analog [ALC880 Analog] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 carte 0: Intel [HDA Intel], périphérique 1 : ALC880 Digital [ALC880 Digital] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 carte 0: Intel [HDA Intel], périphérique 6 : Si3054 Modem [Si3054 Modem] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 eric@gondor:~$ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xb0000000 irq 17 1 [SAA7134 ]: SAA7134 - SAA7134 saa7133[0] at 0xb3006000 irq 22 if I try to do a cat /proc/asound/card0/codec#0 it hangs while /var/log/messages gives a lot of May 14 14:44:52 gondor kernel: [ 3825.457412] hda_codec: invalid dep_range_val 0:7fff May 14 14:44:52 gondor kernel: [ 3825.457696] hda_codec: invalid dep_range_val 0:7fff ........................ eric@gondor:~$ sudo lspci -nv 00:00.0 0600: 8086:2580 (rev 04) Subsystem: 1558:0900 Flags: bus master, fast devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:01.0 0604: 8086:2581 (rev 04) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 Memory behind bridge: b1000000-b2ffffff Prefetchable memory behind bridge: c0000000-cfffffff Capabilities: [88] Subsystem: 8086:2582 Capabilities: [80] Power Management version 2 Capabilities: [90] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Capabilities: [a0] Express Root Port (Slot+) IRQ 0 00:1b.0 0403: 8086:2668 (rev 03) Subsystem: 1558:0900 Flags: bus master, fast devsel, latency 0, IRQ 20 Memory at b0000000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 00:1d.0 0c03: 8086:2658 (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 18 I/O ports at 1800 [size=32] 00:1d.1 0c03: 8086:2659 (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 3000 [size=32] 00:1d.2 0c03: 8086:265a (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 17 I/O ports at 3020 [size=32] 00:1d.3 0c03: 8086:265b (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 20 I/O ports at 3040 [size=32] 00:1d.7 0c03: 8086:265c (rev 03) (prog-if 20 [EHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 18 Memory at b0004000 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port 00:1e.0 0604: 8086:244e (rev d3) (prog-if 01 [Subtractive decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=0a, subordinate=0e, sec-latency=32 I/O behind bridge: 00004000-00004fff Memory behind bridge: b3000000-b30fffff Prefetchable memory behind bridge: 0000000050000000-0000000055ffffff Capabilities: [50] Subsystem: 1558:0900 00:1f.0 0601: 8086:2640 (rev 03) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0 00:1f.1 0101: 8086:266f (rev 03) (prog-if 8a [Master SecP PriP]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 17 I/O ports at 01f0 [size=8] I/O ports at 03f4 [size=1] I/O ports at 0170 [size=8] I/O ports at 0374 [size=1] I/O ports at 3080 [size=16] 00:1f.3 0c05: 8086:266a (rev 03) Subsystem: 1558:0900 Flags: medium devsel, IRQ 10 I/O ports at 3060 [size=32] 01:00.0 0300: 10de:00c8 (rev a2) (prog-if 00 [VGA]) Subsystem: 1558:0900 Flags: bus master, fast devsel, latency 0, IRQ 11 Memory at b2000000 (32-bit, non-prefetchable) [size=16M] Memory at c0000000 (64-bit, prefetchable) [size=256M] Memory at b1000000 (64-bit, non-prefetchable) [size=16M] Capabilities: [60] Power Management version 2 Capabilities: [68] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [78] Express Endpoint IRQ 0 0a:00.0 0607: 104c:ac50 (rev 02) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 168, IRQ 17 Memory at b3007000 (32-bit, non-prefetchable) [size=4K] Bus: primary=0a, secondary=0b, subordinate=0e, sec-latency=176 Memory window 0: 50000000-53fff000 (prefetchable) Memory window 1: 58000000-5bfff000 I/O window 0: 00004800-000048ff I/O window 1: 00004c00-00004cff 16-bit legacy interface ports at 0001 0a:01.0 0c00: 104c:8023 (prog-if 10 [OHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 64, IRQ 21 Memory at b3004000 (32-bit, non-prefetchable) [size=2K] Memory at b3000000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 0a:02.0 0104: 105a:3373 (rev 02) Subsystem: 1558:0900 Flags: bus master, 66MHz, medium devsel, latency 96, IRQ 19 I/O ports at 4480 [size=64] I/O ports at 44c0 [size=16] I/O ports at 4400 [size=128] Memory at b3005000 (32-bit, non-prefetchable) [size=4K] Memory at b3020000 (32-bit, non-prefetchable) [size=128K] Capabilities: [60] Power Management version 2 0a:03.0 0200: 10ec:8169 (rev 10) Subsystem: 1558:0900 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 22 I/O ports at 4000 [size=256] Memory at b3004800 (32-bit, non-prefetchable) [size=256] [virtual] Expansion ROM at 54000000 [disabled] [size=128K] Capabilities: [dc] Power Management version 2 0a:04.0 0480: 1131:7133 (rev f0) Subsystem: 1461:f31e Flags: bus master, medium devsel, latency 64, IRQ 23 Memory at b3006000 (32-bit, non-prefetchable) [size=2K] Capabilities: [40] Power Management version 2 0a:05.0 0280: 1814:0201 (rev 01) Subsystem: 1462:6833 Flags: bus master, slow devsel, latency 64, IRQ 20 Memory at b3008000 (32-bit, non-prefetchable) [size=8K] Capabilities: [40] Power Management version 2 asoundconf output : eric@gondor:~$ asoundconf list Names of available sound cards: Intel SAA7134 I tried a lot of workarounds but none worked : boot options (irqpoll, pci=noacpi), single_cmd=1, position_fix=1, several models of hda-intel driver, I tried 1.0.13 and 1.0.14rc4 driver, with debug on... I must add that I already had this problem with previous distributions of Ubuntu. It worked with an old one (Breezy) then in Dapper and Edgy, it did worked ONLY when I self-compiled the driver with debug option enabled I never understood why. And now with Feisty/alsa 1.0.13/ubuntu kernel 2.6.20, or an ubuntu kernel with self-compiled alsa 1.0.14rc4, it does not work. I cannot have one sound :-(. Please help me .... Eric

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[alsa-devel] default PCM volume
by Michel Benoit 15 May '07

15 May '07

Hello again, I am running alsa on a custom AT91SAM9260 arm based SOC board together with a a simple external PCM1725 codec. When using alsa-lib and the at91 soc driver (pcm+i2s) I can playback PCM samples using both aplay and test/pcm. Using softvol I can also reduce the volume of the playback. The problem I have is that the playback volume (with softvol=100%) is not very loud. The volume is the same even without the softvol plugin. The PCM codec has no hw volume control. According to our hw team the sound level should be much higher. Is there a default PCM attenuation (volume reduction) built into alsa lib? If so, how can it be adjusted? Thanks, Michel

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