February 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] hda: STAC927x analog mic
by Matthew Ranostay 21 Feb '08

21 Feb '08

Some laptops have a internal analog microphone that is not setup by the BIOS. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 6b53fa114faa pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Feb 19 15:03:57 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Wed Feb 20 17:11:51 2008 -0500 @@ -3629,6 +3629,8 @@ static int patch_stac927x(struct hda_cod spec->mixer = stac927x_mixer; break; case STAC_DELL_BIOS: + /* configure the analog microphone on some laptops */ + stac92xx_set_config_reg(codec, 0x0c, 0x90a79130); /* correct the front output jack as a hp out */ stac92xx_set_config_reg(codec, 0x0f, 0x02270110); /* correct the front input jack as a mic */

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Re: [alsa-devel] Ответы для Linux Format
by Serge A. Suchkov 21 Feb '08

21 Feb '08

В сообщении от Thursday 21 February 2008 12:33:26 Andrei Borovsky написал(а): > > Хмм... а можно урезать эту информацию до минимума ? > > Это ведь всего лишь хобби и я не склонен его афишировать настолько > > публично :) > > >Тем более мой вклад в GSPCA весьма скромен на общем фоне. > > Как скажете. В принципе, все это, конечно, необязательно. Даже фотографию > поместить не хотите? :) Нет, спасибо :) > > > С уважением, > Андрей Боровский. -- Regards, Serge A. Suchkov -- A baby is an alimentary canal with a loud voice at one end and no responsibility at the other.

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Re: [alsa-devel] [PATCH] Fix a compile warning under gcc-4.2.3.
by Clemens Ladisch 21 Feb '08

21 Feb '08

Joshua Roys wrote: > sound/core/init.c: In function ‘snd_card_disconnect’: > sound/core/init.c:307: warning: the address of ‘snd_shutdown_f_ops’ will always evaluate as ‘true’ > > Signed-off-by: Joshua Roys <roysjosh(a)gmail.com> > --- > sound/core/init.c | 1 - > 1 files changed, 0 insertions(+), 1 deletions(-) > > diff --git a/sound/core/init.c b/sound/core/init.c > index 2cb7099..1afc58b 100644 > --- a/sound/core/init.c > +++ b/sound/core/init.c > @@ -304,7 +304,6 @@ int snd_card_disconnect(struct snd_card *card) > list_add(&mfile->shutdown_list, &shutdown_files); > spin_unlock(&shutdown_lock); > > - fops_get(&snd_shutdown_f_ops); > mfile->file->f_op = &snd_shutdown_f_ops; > > mfile = mfile->next; This change may get rid of the warning, but is has the slight disadvantage that the kernel will blow up if you try to unload the sound modules before all device files have been closed. If you want to get rid of this warning, tell the compiler that fops_get knows what it does; or try making fops_get an inline function instead of a macro. Regards, Clemens

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[alsa-devel] [PATCH] hda: 92HDxxxx PCI Quirks
by Matthew Ranostay 21 Feb '08

21 Feb '08

Added PCI_QUIRKS for laptop that have the 92HDxxx family of codecs. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 6b53fa114faa pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Feb 19 15:03:57 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Wed Feb 20 17:18:15 2008 -0500 @@ -64,11 +64,14 @@ enum { enum { STAC_92HD73XX_REF, + STAC_DELL_M6_5ST, STAC_92HD73XX_MODELS }; enum { STAC_92HD71BXX_REF, + STAC_DELL_M4_5ST, + STAC_DELL_M4_6ST, STAC_92HD71BXX_MODELS }; @@ -334,10 +337,10 @@ static hda_nid_t stac922x_pin_nids[10] = 0x0f, 0x10, 0x11, 0x15, 0x1b, }; -static hda_nid_t stac92hd73xx_pin_nids[12] = { +static hda_nid_t stac92hd73xx_pin_nids[13] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, 0x13, - 0x14, 0x22 + 0x14, 0x1e, 0x22 }; static hda_nid_t stac92hd71bxx_pin_nids[10] = { @@ -1220,24 +1223,48 @@ static struct snd_pci_quirk stac925x_cfg {} /* terminator */ }; -static unsigned int ref92hd73xx_pin_configs[12] = { +static unsigned int ref92hd73xx_pin_configs[13] = { 0x02214030, 0x02a19040, 0x01a19020, 0x02214030, 0x0181302e, 0x01014010, 0x01014020, 0x01014030, 0x02319040, 0x90a000f0, 0x90a000f0, 0x01452050, + 0x01452050, +}; + +static unsigned int dell_m6_5st_pin_configs[13] = { + 0x0321101f, 0x4f00000f, 0x4f0000f0, 0x90170110, + 0x03a11020, 0x03011050, 0x4f0000f0, 0x4f0000f0, + 0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0, + 0x4f0000f0, }; static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { - [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, + [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, + [STAC_DELL_M6_5ST] = dell_m6_5st_pin_configs, }; static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { [STAC_92HD73XX_REF] = "ref", + [STAC_DELL_M6_5ST] = "dell-m6-5stack", }; static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, - "DFI LanParty", STAC_92HD73XX_REF), + "DFI LanParty", STAC_92HD73XX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254, + "unknown Dell", STAC_DELL_M6_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255, + "unknown Dell", STAC_DELL_M6_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0256, + "unknown Dell", STAC_DELL_M6_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0257, + "unknown Dell", STAC_DELL_M6_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x025e, + "unknown Dell", STAC_DELL_M6_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x025f, + "unknown Dell", STAC_DELL_M6_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0271, + "unknown Dell", STAC_DELL_M6_5ST), {} /* terminator */ }; @@ -1247,18 +1274,56 @@ static unsigned int ref92hd71bxx_pin_con 0x90a000f0, 0x01452050, }; +static unsigned int dell_m4_5st_pin_configs[13] = { + 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, + 0x23a1902e, 0x23014250, 0x40f000f0, 0x4f0000f0, + 0x40f000f0, 0x4f0000f0, +}; + +static unsigned int dell_m4_6st_pin_configs[13] = { + 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, + 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, + 0x40f000f0, 0x044413b0, +}; + static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs, + [STAC_DELL_M4_5ST] = dell_m4_5st_pin_configs, + [STAC_DELL_M4_6ST] = dell_m4_6st_pin_configs, }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = "ref", + [STAC_DELL_M4_5ST] = "dell-m4-5stack", + [STAC_DELL_M4_6ST] = "dell-m4-6stack", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, + "unknown Dell", STAC_DELL_M4_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234, + "unknown Dell", STAC_DELL_M4_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0250, + "unknown Dell", STAC_DELL_M4_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x024f, + "unknown Dell", STAC_DELL_M4_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x024d, + "unknown Dell", STAC_DELL_M4_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0251, + "unknown Dell", STAC_DELL_M4_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0277, + "unknown Dell", STAC_DELL_M4_5ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0263, + "unknown Dell", STAC_DELL_M4_6ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0265, + "unknown Dell", STAC_DELL_M4_6ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0262, + "unknown Dell", STAC_DELL_M4_6ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0264, + "unknown Dell", STAC_DELL_M4_6ST), {} /* terminator */ }; @@ -3355,12 +3420,38 @@ again: spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); - spec->num_dmics = STAC92HD73XX_NUM_DMICS; spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); spec->dinput_mux = &stac92hd73xx_dmux; /* GPIO0 High = Enable EAPD */ spec->gpio_mask = spec->gpio_dir = 0x1; spec->gpio_data = 0x01; + + switch (spec->board_config) { + case STAC_DELL_M6_5ST: + switch (codec->subsystem_id) { + case 0x1028025e: /* Analog Mics */ + case 0x1028025f: + stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); + spec->num_dmics = 0; + break; + case 0x10280254: /* Digital Mics */ + case 0x10280255: + case 0x10280271: + case 0x10280272: + stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + spec->num_dmics = 1; + break; + case 0x10280256: /* Both */ + case 0x10280057: + stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); + stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + spec->num_dmics = 1; + break; + } + break; + default: + spec->num_dmics = STAC92HD73XX_NUM_DMICS; + } spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids;

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[alsa-devel] [PATCH] hda: STAC927x power management
by Matthew Ranostay 21 Feb '08

21 Feb '08

Fix issue on STAC927x codecs that first DAC was getting powered down even if was being used. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 30701aaf404c pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Feb 20 12:13:29 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Wed Feb 20 14:16:59 2008 -0700 @@ -2893,7 +2893,8 @@ static void stac92xx_power_down(struct h /* power down inactive DACs */ hda_nid_t *dac; for (dac = spec->dac_list; *dac; dac++) - if (!is_in_dac_nids(spec, *dac)) + if (!is_in_dac_nids(spec, *dac) && + spec->multiout.hp_nid != *dac) snd_hda_codec_write_cache(codec, *dac, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); }

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Re: [alsa-devel] Alsa-devel Digest, Vol 12, Issue 70
by Anders Rønningen 21 Feb '08

21 Feb '08

I'll do the tests when I get home from work. Just wanted to check that I understood the instructions. The problem is like you said. The left HP and the internal speakers never give any sound, regardless of whether the right HP is connected. You want me to: 1. remove the mentioned line from alsa-kernel/pci/hda/patch_sigmatel.c. 2. configure and build the driver with full debug 3. reboot, run alsa-info.sh and upload the output Is that right? > No difference in the output but there must be a difference in the codec setting. You mean before and after I remove the line from patch_sigmatel.c? >What is the original configuration of pin 0xf? How can I know/find out this? - Anders On Feb 20, 2008 12:00 PM, <alsa-devel-request(a)alsa-project.org> wrote: > Send Alsa-devel mailing list submissions to > alsa-devel(a)alsa-project.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > or, via email, send a message with subject or body 'help' to > alsa-devel-request(a)alsa-project.org > > You can reach the person managing the list at > alsa-devel-owner(a)alsa-project.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Alsa-devel digest..." > > > Today's Topics: > > 1. Re: Problem with DELL XPS M1530 (Jiang zhe) > 2. Re: Problem with DELL XPS M1530 (Takashi Iwai) > 3. Re: Problem with DELL XPS M1530 (Takashi Iwai) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 20 Feb 2008 17:49:00 +0800 > From: Jiang zhe <zhe.jiang(a)intel.com> > Subject: Re: [alsa-devel] Problem with DELL XPS M1530 > To: Anders R?nningen <andersronningen(a)gmail.com> > Cc: Takashi Iwai <tiwai(a)suse.de>, alsa-devel(a)alsa-project.org > Message-ID: <1203500940.3006.4.camel(a)localhost.localdomain> > Content-Type: text/plain; charset=utf-8 > > On Wed, 2008-02-20 at 02:25 +0800, Anders R?nningen wrote: > > Hi Takashi. > > > > Some dmesg output: > > > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:2994: > > autoconfig: line_outs=1 (0xf/0x0/0x0/0x0/0x0) > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:2998: > > speaker_outs=1 (0xd/0x0/0x0/0x0/0x0) > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3002: > > hp_outs=1 (0xa/0xf/0x0/0x0/0x0) > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3003: > > mono: mono_out=0x0 > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3011: > > inputs: mic=0xe, fmic=0x0, line=0x0, fline=0x0, cd=0x0, aux=0x0 > > [ 14.196000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:2224: > > dac_nids=2 (0x5/0x4/0x0/0x0/0x0) > > > > Unmuted and adjusted Surround volume, but it made no difference. > > > > - Anders > > > It's weird that 0xf is in both the line_outs and hp_outs. > And the hp_outs=1 when there are two items. > > What is the original configuration of pin 0xf? > Please try to get rid of this line from the > alsa-kernel/pci/hda/patch_sigmatel.c. > > SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022e, "Dell ", STAC_DELL_BIOS), > > Rebuild and install the driver. > Please give the output of "cat /proc/asound/card0/codec#0". > > Thanks! > > > Takashi Iwai wrote: > > > At Tue, 19 Feb 2008 17:24:04 +0100, > > > Anders R?nningen wrote: > > > > > >> Hi. I am the mentioned user with this problem. Just thought I'd > > give you > > >> a little more info on this problem. > > >> > > >> What wishie described was correct, except it's the internal > > speakers and > > >> the /left/ hp-jack that stopped working, while the right one is > > still > > >> ok. Like he wrote, the reason we started this debugging was to get > > mic > > >> sound on my system. The thing I think is worth mentioning is that > > with > > >> alsa 1.0.14 on an out-of-the box Ubuntu Gutsy, both hp-jacks and > > the > > >> internal speakers worked for me. People in #alsa told me that the > > module > > >> was not loaded correctly, but whatever was loaded _almost_ worked > > well :-) > > >> > > > > > > OK, please build the driver with --with-debug=full option and show > > the > > > kernel messages such as "autoconfig: ..." and "dac_nids...", etc. > > > This contains the information that the pin configuration parser > > uses. > > > > > > Also, unmute and adjust "Surround" volume. > > > > > > > > > Takashi > > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > > > > > ------------------------------ > > Message: 2 > Date: Wed, 20 Feb 2008 11:50:44 +0100 > From: Takashi Iwai <tiwai(a)suse.de> > Subject: Re: [alsa-devel] Problem with DELL XPS M1530 > To: Jiang zhe <zhe.jiang(a)intel.com> > Cc: alsa-devel(a)alsa-project.org, Anders R?nningen > <andersronningen(a)gmail.com> > Message-ID: <s5hy79fsxyz.wl%tiwai(a)suse.de> > Content-Type: text/plain; charset=ISO-8859-1 > > At Wed, 20 Feb 2008 17:49:00 +0800, > Jiang zhe wrote: > > > > On Wed, 2008-02-20 at 02:25 +0800, Anders R?nningen wrote: > > > Hi Takashi. > > > > > > Some dmesg output: > > > > > > [ 14.192000] ALSA > > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:2994: > > > autoconfig: line_outs=1 (0xf/0x0/0x0/0x0/0x0) > > > [ 14.192000] ALSA > > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:2998: > > > speaker_outs=1 (0xd/0x0/0x0/0x0/0x0) > > > [ 14.192000] ALSA > > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3002: > > > hp_outs=1 (0xa/0xf/0x0/0x0/0x0) > > > [ 14.192000] ALSA > > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3003: > > > mono: mono_out=0x0 > > > [ 14.192000] ALSA > > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3011: > > > inputs: mic=0xe, fmic=0x0, line=0x0, fline=0x0, cd=0x0, aux=0x0 > > > [ 14.196000] ALSA > > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:2224: > > > dac_nids=2 (0x5/0x4/0x0/0x0/0x0) > > > > > > Unmuted and adjusted Surround volume, but it made no difference. > > > > > > - Anders > > > > > It's weird that 0xf is in both the line_outs and hp_outs. > > And the hp_outs=1 when there are two items. > > It's due to a fix for an ALC268 laptop that has also two HPs but one > is actually a line-out. Note that hp_outs=1 in the print above. The > 0x0f in hp_outs is simply a garbage. Maybe that patch broke the > double HPs with STAC9228... > > > Takashi > > > ------------------------------ > > Message: 3 > Date: Wed, 20 Feb 2008 11:52:43 +0100 > From: Takashi Iwai <tiwai(a)suse.de> > Subject: Re: [alsa-devel] Problem with DELL XPS M1530 > To: Anders R?nningen <andersronningen(a)gmail.com> > Cc: alsa-devel(a)alsa-project.org > Message-ID: <s5hwsozsxvo.wl%tiwai(a)suse.de> > Content-Type: text/plain; charset=ISO-8859-1 > > At Tue, 19 Feb 2008 19:25:24 +0100, > Anders R?nningen wrote: > > > > Hi Takashi. > > > > Some dmesg output: > > > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:2994: > > autoconfig: line_outs=1 (0xf/0x0/0x0/0x0/0x0) > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:2998: > > speaker_outs=1 (0xd/0x0/0x0/0x0/0x0) > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3002: > > hp_outs=1 (0xa/0xf/0x0/0x0/0x0) > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3003: > > mono: mono_out=0x0 > > [ 14.192000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/hda_codec.c:3011: > > inputs: mic=0xe, fmic=0x0, line=0x0, fline=0x0, cd=0x0, aux=0x0 > > [ 14.196000] ALSA > > > /mnt/storage/source/alsa/alsa-driver-hg20080217/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:2224: > > dac_nids=2 (0x5/0x4/0x0/0x0/0x0) > > > > Unmuted and adjusted Surround volume, but it made no difference. > > Please upload the alsa-info.sh output again. No difference in the > output but there must be a difference in the codec setting. > > Also, just to make the situation clear: > the problem is that the speaker and the rear HP don't give sound > output even if you *don't* plug in the front HP, right? > > > Takashi > > > ------------------------------ > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > End of Alsa-devel Digest, Vol 12, Issue 70 > ****************************************** > -- Mvh, Anders

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Re: [alsa-devel] Problem with DELL XPS M1530
by Anders Rønningen 21 Feb '08

21 Feb '08

Hi. I am the mentioned user with this problem. Just thought I'd give you a little more info on this problem. What wishie described was correct, except it's the internal speakers and the /left/ hp-jack that stopped working, while the right one is still ok. Like he wrote, the reason we started this debugging was to get mic sound on my system. The thing I think is worth mentioning is that with alsa 1.0.14 on an out-of-the box Ubuntu Gutsy, both hp-jacks and the internal speakers worked for me. People in #alsa told me that the module was not loaded correctly, but whatever was loaded _almost_ worked well :-) - Anders

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[alsa-devel] snd-hda-intel AD1981 - the microphone doesn't work
by Robert Gonciarz 20 Feb '08

20 Feb '08

Hi, I'm trying to activate a microphone (I have just external one) in my HP NX7300 (RU374ES) notebook (it has two jacks without internal mic) for six months. Alsamixer says that it's AD1981 chipset. The sound works great execept two things: - it's too silent when I set almost the maximum level (if you compare it with Windows XP driver) - but it's not so bad. It is said that a tothem can change some hidden settings to make it louder. - the microphone doesn't work even though I setted it in alsamiser (the settings: <capture tab> the "Mic" is red, "Mic boss" is setted with 67/67, "Capture" 80/80, Mix is blue and "-----", "Digital" 62/62, "Docking-" is blue and "----", "Internal" 67/67. The mic doesn't capture any voice at all. I was trying to load the module with different names: default, hp and toshiba works but the mic still doesn't work. lspci | grep Audio 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 01) lspci -nv 00:1b.0 0403: 8086:27d8 (rev 01) Subsystem: 103c:30a2 Flags: bus master, fast devsel, latency 0, IRQ 16 Memory at f4580000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel It was tested with snd-hda-intel module from the lastest sources from your website: alsa-driver-1.0.16rc2. I'm using 2.6.24.1 kernel. At the begining I tried to compile the kernel with a module which was included to it (it was working without a mic) and then I compiled the external module from your web. It was tested with many other kernels. Can you check it please? If you need more information just email me back. Robert

4 12

[alsa-devel] Regarding pointer callback
by Pharaoh . 20 Feb '08

20 Feb '08

Hi, In Writing an ALSA driver, it is written that pointer callback is called usually from the buffer-update routine in the pcm middle layer, which is invoked when snd_pcm_period_elapsed() is called in the interrupt routine. Then the pcm middle layer updates the position and calculates the available space, and wakes up the sleeping poll threads, etc. I know from a kernel space ALSA driver, the pointer callback is called after the driver calls snd_pcm_period_elapsed() from the interrupt handler or after a period is consumed. How pointer function of a user space plugin is called? I have a user space pcm external plugin which updates the buffer counter after every period is consumed, but when will it be called? How does the ALSA lib know when to call it since there is no snd_pcm_period_elapsed() in user space. -Phraoh.

1 0

[alsa-devel] Capture and playabck together
by Emmanuel Milou 20 Feb '08

20 Feb '08

Hi all, I am developping an open-source C++ softphone based on Alsa. We used to use Portaudio but we recently decided to switch to Alsa. I am quite unexperimented in Alsa dev and I got some difficulties to make it work fine. My capture and playback devices are plughw:0,0. I am using read/write transfers. When I disable the capture, playback works fine; when I disable the playback, capture works fine. The problems occurs when I used them together. If I set hardware parameters for both, the playback sound is awful and there is no capture, and finally core dumped. If I set hardware parameters for the playback stream only, the playback works fine, but no capture, I just get an echo with an input/ouput error. Where do you think the problem comes from? Isn't it possible to set hardware configuration for playback and capture? Is there a hardware ring buffer for each stream ? Thanks for your help Manu

1 0

[alsa-devel] [PATCH 0/2] TLV320AIC3X MCLK and rate updates
by Jarkko Nikula 20 Feb '08

20 Feb '08

Hi Two patches to TLV320AIC3X SoC codec driver. First one fixes typo (?) for 64 kHz sample rate and second one adds support to 12 MHz MCLK used e.g. in Nokia N810 Internet Tablet. Patches are generated from 2.6.25-rc2-omap1 tree but they apply to alsa git and alsa-kernel hg trees as well. -- Jarkko

2 3

[alsa-devel] Merging hg sources into kernel tree
by Andrew Boettcher 20 Feb '08

20 Feb '08

Is there a script to merge the hg sources into a kernel tree? If not do I need to worry about any changes other than the include/ directory? I have another module (cx88-alsa) that is in v4l but needs the newer alsa tree or I get "cx88_alsa: disagrees about version of symbol xxx", so I have to merge the sources for now. Andrew Boettcher

3 3

[alsa-devel] Moving X Window windows causes audio stuttering
by Timur Tabi 20 Feb '08

20 Feb '08

On our system, using mplayer to play a movie that has 44.1KHz audio (hardware only supports 48KHz) can cause audio stuttering if we're using OSS emulation, but not if we're using native ALSA. Is this a known problem? Does OSS emulation hog the CPU? -- Timur Tabi Linux kernel developer at Freescale

2 1

Re: [alsa-devel] [PATCH] Driver for Emagic Audiowerk 2
by Takashi Iwai 20 Feb '08

20 Feb '08

At Tue, 19 Feb 2008 00:38:54 +0100, Cédric Brégardis wrote: > > Signed-off-by: Cedric Bregardis <cedric.bregardis(a)free.fr> > > Signed-off-by: Jean-Christian Hassler <jhassler(a)free.fr> Could you avoid HTML mail? I cannot get the patch properly. If embedding text is difficult, rather use an attachment. thanks, Takashi

2 4

[alsa-devel] Ends up with write error
by Ning Zhang 20 Feb '08

20 Feb '08

Hi all, I am writing a ALSA driver for my ARM 926 platform. The ARM communicates with the DSP by accessing a 128-byte shared memory. I don't use IRQ mechanism.(In fact, I am not sure the platform has this mechanism) Here is some code slices: static struct snd_pcm_hardware snd_arch_playback_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER), .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000), .rate_min = 8000, .rate_max = 48000, .channels_min = 2, .channels_max = 2, .buffer_bytes_max = 128, .period_bytes_min = 2, .period_bytes_max = 128, .periods_min = 1, .periods_max = 64, }; static snd_pcm_uframes_t snd_arch_pcm_pointer(struct snd_pcm_substream *substream) { … /* * get the current hardware pointer */ current_ptr = arch_fw_read_dsp_reg(chip, AFE_RWADDR) & AFE_RD_ADDR; return bytes_to_frames(runtime, current_ptr); } static int snd_arch_pcm_playback_copy(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void __user *src, snd_pcm_uframes_t count) { …… count = frames_to_bytes(runtime, count); pos = frames_to_bytes(runtime, pos); chip->bufpos = pos; …. spin_lock(chip->reg_lock); if ( count > 0 && count <= MAX_BUF_SIZE ) { if ( copy_from_user(chip->bufptr, src, count) ) { snd_printk(KERN_ERR "can not copy user data!\n"); return -EFAULT; } if ( arch_fill_rx_buf(chip, (signed short*)(chip->bufptr), count) < 0 ) { snd_printk(KERN_ERR "fill rx buffer failed!\n"); return -1; } …………… } spin_unlock(chip->reg_lock); return 0; } The problem is my driver can make sounds, but ends up a write error. After much debugging, I think there might be some problem in the interworking between middle-level driver and the device driver. I found the runtime->control->avail_min=32, and the runtime->buffer_size and runtime->period_size are both 32. So at first middle-level driver call snd_pcm_lib_write1 transfers 32 frames(128 bytes) to the device driver's copy callback function and trigger function. Then when middle-level driver wants to transfer another 32 frames, it calls the pointer callback function and gets the position, because at this time the whole bytes has not been processed completely, so the pointer function's return value must less than 32, and the avail must less than runtime->control_->avail_min, and goes to this flow and ends up a write error. _______________________________________________________________________________ if (((avail < runtime->control->avail_min && size > avail) || (size >= runtime->xfer_align && avail < runtime->xfer_align))) { ……………... long tout; init_waitqueue_entry(&wait, current); add_wait_queue(&runtime->sleep, &wait); while (1) { if (signal_pending(current)) { state = SIGNALED; break; } set_current_state(TASK_INTERRUPTIBLE); snd_pcm_stream_unlock_irq(substream); tout = schedule_timeout(10 * HZ); snd_pcm_stream_lock_irq(substream); if (tout == 0) { if (runtime->status->state != SNDRV_PCM_STATE_PREPARED && runtime->status->state != SNDRV_PCM_STATE_PAUSED) { state = runtime->status->state == SNDRV_PCM_STATE_SUSPENDED ? SUSPENDED : EXPIRED; break; } } switch (runtime->status->state) { ………….. : state = ERROR; goto _end_loop; default: break; } avail = snd_pcm_playback_avail(runtime); if (avail >= runtime->control->avail_min) { state = READY; break; } } _end_loop: remove_wait_queue(&runtime->sleep, &wait); switch (state) { ……. case EXPIRED: snd_printd("playback write error (DMA or IRQ trouble?)\n"); err = -EIO; goto _end_unlock; ……… } } This code is a part of snd_pcm_lib_write1() in sound/core/pcm_lib.c I think that I misunderstood the mechanism of middle-level driver. Could you please correct me? What should the pointer callback function return? What's it for? How comes the runtime->control->avail_min? Because I don't use the IRQ, so should I use the snd_pcm_period_elapsed function? And where? ALSA Version is 1.0.14rc1 and I use alsa-lib-1.0.15. Can someone comment on this and give me some hints to solve this problem? Thanks in advance. BRs, David

1 0

[alsa-devel] ALSA MIDI interface - plugin
by Pharaoh . 20 Feb '08

20 Feb '08

Hi I am writing an user space ALSA i/o type PCM plugin for my board. I chose this rather unconventional approach for supporting ALSA apps since the the the driver which communicates with DSP using shared memory is in userspace and for some constraints the driver communication driver will always be in userspace. My audio plugin uses calls of the coomunication driver and talks to DSP. I already have a working plugin in place. I use it like following, #aplay -D my_plugin test.wav and it works. Now, I have to support rawmidi interface also. Is it possible to write a userspace plugin for making MIDI work? Where can I get sample code etc? -pharaoh.

4 5

Re: [alsa-devel] Problem with setting period time and buffer time.
by Clemens Ladisch 20 Feb '08

20 Feb '08

Ignacy Kasperowicz wrote: > But what about buffer_time, why can't I set this time ? The allowed range of values is hardware dependent, like the period time. Use snd_pcm_hw_params_set_buffer_time_near(). > Anyway is there a method to play my wave for a specified short time? Just write the data to the device. What is the problem? Regards, Clemens

2 4

[alsa-devel] Notebook Toshiba satellite x200-21g with realtek snd-hda-intel audio device still not supported (codec ALC268)
by Daniele (Mastro) 19 Feb '08

19 Feb '08

hi all i'm new to this mailing list i've bought a laptop and i would like to help you supporting my sound card what works: 2 speaker, low volume (it should be a 5+1 audio card) what doesn't -) low volume -) inoperating headphone jack -) not working mic -) no subwoofer -) may be some other stuff not tested ( SPDIF? hardware midi?) with speaker-test -t wav -c 6 only front left/right channel works my kernel is a debian kernel Linux Luffy 2.6.23-1-686-bigmem #1 SMP Wed Dec 5 02:42:05 UTC 2007 i686 GNU/Linux alsa 1.0.15 drivers (compiled with debian-way: module-assistant a-i alsa-source) i've already tried all the option i could think about auto (default) toshiba 3stacks 3stacks-6ch 3stacks-6ch-dig neither one has better/worst behavior I've checked the module has toke those option by cat /sys/module/snd_hda_intel/parameters/model anyway I've read the ALSA-Configuration.txt shipped with kernel and for Codec ALC268 it only has 4 options ALC268 3stack 3-stack model toshiba Toshiba A205 acer Acer laptops auto auto-config reading BIOS (default) I've googled a bit and i found out that the Toshiba A205 has completely different problem.. the sound doesn't work with it.. reading the changelog from 1.0.14 to 1.0.15 (alsa) it say: Added the proper model=toshiba for Toshiba A305 with ALC268 codec. so i think my audio card is an unknown model the laptop is brand new and i only found this model in Italy anyway i think that the same audio card is shipped with the Toshiba Satellite x205-S7483 witch is an american laptop some info about the audio card: lspci -n 00:1b.0 0403: 8086:284b (rev 03) lspci -vvv 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Toshiba America Info Systems Unknown device ff00 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at f6600000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) cat /proc/asound/card0/codec#0 Codec: Realtek ALC268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x1179ff0a Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x0221101f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x02a11830: [Jack] Mic at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x24: IN Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: Pincap 0x083724: IN Detect Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x24: IN Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x0281103e: [Jack] Line In at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4016852d: [N/A] Speaker at Ext N/A Conn = Digital, Color = Purple Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x0810: OUT Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x1f 0x1f] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x8a 0x8a] Connection: 7 0x18 0x19 0x1a* 0x1c 0x14 0x15 0x13 many of this think doesn't mean a thing to me but i will be glad to help you if you ask.. i don't mind compiling some other driver/kernel/tool for this just ask me one last note: searching the internet i've found this page: https://wiki.ubuntu.com/Gutsy_Intel_HD_Audio_Controller if you go down you find that the Toshiba Satellite x205-S7483 has the exact same problem here. I think they have the same card searching the Realtek web site i didn't find any 6 channel audio card with the ALC268 codec may be this mean the ALC268 is not the right codec for my audio card or this is a really new model/heavily Toshiba modified device i've already filed a bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3659 please ask me if you need any more info or want me to do some test thanks you all for your work ps: i've subscribed to vger mailing list, what's the right mailing list for alsa-devel? -- Daniele Segato

3 29

[alsa-devel] Maya44 revised patch
by Rainer Zimmermann 19 Feb '08

19 Feb '08

This is the revised version of the ESI Maya44 patch. changes are mainly coding style, some cleanups and using a patch file for ice1724.c . The added code now (mostly) passes checkpatch.pl, except for some "#if 0"'s which I'd leave in for the development phase. Also, I changed the rate setting logic, which (for now) allows all rates up to 192kHz even for capturing. As capturing actually only supports 96 kHz, while capturing & playback rates are linked, this is not really ok. I'd appreciate suggestions about how this could be handled properly, while still supporting playback up to 192 kHz... Again, feedback & testing is appreciated. For more information, see doc/README.maya44 . -Rainer -- Lightshed IT Services Löfflerstr. 27, 22765 Hamburg, Germany * mail(a)lightshed.de * www.lightshed.de

3 6

[alsa-devel] snd-hda-intel AD1981 - Sound volume low level
by Robert Gonciarz 19 Feb '08

19 Feb '08

Hi, I just want to ask you if you have heard about the problem with low sound volume level even though you set the alsamixer on the (almost) maximum volume level. The sound card: snd-hda-intel / AD1981 / ICH7 ) It's a big problem for me especially when I watch a movie with ac3 (DVD quality) decoded sound. The volume level is extrimly low in such situation (I mean the level you hear the sound). I have to connect my loudspeakers to a ampilifier and set it in very high level. I don't have such problem when I work with Windows XP. They are loud enough. That's alsa-info.sh result: http://pastebin.ca/908831 It is said that there are some preferences in the totem package (hidden in alsamixer) that can fix it and make alsa to remember higher sound level. I've noticed that problem with all kernels I tested (I've checked about 15-20 kernels from that range 2.6.13.x - 2.6.24.x) (I usually use vanillia kernel (from www.kernel.org) /with build alsa/). But I've tested the newest alsa-drivers: 1.0.16 with 2.6.24.2 kernel also and the result was the same. Could you chack that problem, please? Robert

2 1

Re: [alsa-devel] Problem with setting period time and buffer time.
by Ignacy Kasperowicz 19 Feb '08

19 Feb '08

Ok, so I misunderstood the idea of a period variable, many thanks for explanation Clemens. I've just read that buffer_time also isn't what I think it was ;) It's a size of the ring buffer of sound card and it's length depends on hardware, am I right? So the best way is to depend on default values of hardware? Anyway is there a method to play my wave for a specified short time? As I can see using timer interface is the only solution but still can't google anything useful :( Small example how to apply timers will be appreciated (I saw example timer.c from alsa-lib, but I'd prefer to read some explanation how the code works). Thanks from advance for any replays. -- ################################################ gry świetlicowe drogą do wyższej fazy przemian. gg:681548 jid: fik(a)jabbim.pl usecide.net ################################################

1 0

[alsa-devel] Problem with setting period time and buffer time.
by Ignacy Kasperowicz 19 Feb '08

19 Feb '08

Hello. I need to set proper period time for my application. What I need my code to do is to play sine wave with set frequency and for exact time of 1/1200 s (something about 834 us). When I try to set period time for 834 us I've got this error: Unable to set period time 834 for playback: Invalid argument . Honestly speaking I've got this error message no matter what value I try to set). The same situation take place when I try to set buffer time for something about 1000us (so one period could fit in my buffer). The purpose of this program is to generate sine wave for AFSK (audio frequency-shift keying). I want to play wave for period_time (834us) then if needed change frequency and play changed wave again for period_time with phase shifting. Wave generation with phase shifting for variable frequencies works, but not with period_time I need. Could anyone look at source and tell me why I can't set this values? I placed code with syntax highlighting at: http://pygments.org/demo/606/ . (Don't look at prepare_message function, it isn't used right now). Thanks from advice. Ignacy. -- ################################################ gry świetlicowe drogą do wyższej fazy przemian. gg:681548 jid: fik(a)jabbim.pl usecide.net ################################################

2 1

Re: [alsa-devel] installation steps for us-122l tascam driver
by belibem 18 Feb '08

18 Feb '08

Are there any news on the subject? What does it take to apply the patch? A packet for any major distribution would be greatly appreciated.

1 0

[alsa-devel] Cirrus Logic CS42L52 Low power codec
by Brian Austin 18 Feb '08

18 Feb '08

Hello, I have a codec I would like to submit. The Cirrus Logic CS42L52 Low Power codec. I have the codec code as well as the machine code for the Cirrus Logic EP9407A, but cannot submit that yet. Is it OK to just submit the codec code for now? This does not use the new API, so a hint as to which tree to make the diff from would help. Thanks, Brian Austin

4 10

[alsa-devel] Problem with DELL XPS M1530
by Travis Place 18 Feb '08

18 Feb '08

Hi all, Been helping a user on #alsa for a day or two now, with issues regarding recording from Mic on his Dell XPS M1530 laptop. He originally had alsa-driver 1.0.14, so i had him upgrade to 1.0.16. Now, his Mic works for recording, but he has lost playback on one of his HP jacks (the right one), and Internal laptop speakers. I had a brief look at the code in patch_sigmatel.c, to see what was going on.. But since im no code guru, i wasnt much help.. The output of alsa-info.sh is located at http://pastebin.ca/908579 I did notice some comments in the code about correcting HP and Mic jacks for certain models (atleast thats what I read it as). Any ideas would be appreciated. Travis (wishie). -- Happiness is like peeing your pants. Everyone can see it, but only you can feel its warmth

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