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

[alsa-devel] [PATCH] Add missing audio path between Mono Mixer and Mic PGAs.
by Mark Brown 18 Mar '08

18 Mar '08

From: Robert Jarzmik <rjarzmik(a)free.fr> Signed-off-by: Robert Jarzmik <rjarzmik(a)free.fr> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9713.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index e2949c5..172ca1a 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -500,6 +500,8 @@ static const char *audio_map[][3] = { {"Mono Mixer", "Aux Playback Switch", "Aux DAC"}, {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"}, {"Mono Mixer", "PCM Playback Switch", "AC97 Mixer"}, + {"Mono Mixer", "Mic 1 Sidetone Switch", "Mic A PGA"}, + {"Mono Mixer", "Mic 2 Sidetone Switch", "Mic B PGA"}, {"Mono Mixer", NULL, "Capture Mono Mux"}, /* DAC inv mux 1 */ -- 1.5.4.4

2 1

[alsa-devel] [PATCH] Map 3stack-6ch-dig ALC883 model for MSI 945GCM5 V2 (MSI-7267)
by Herton Ronaldo Krzesinski 18 Mar '08

18 Mar '08

Map 3stack-6ch-dig ALC883 model for MSI 945GCM5 V2 (MSI-7267). Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-18 00:18:52.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-18 00:48:18.000000000 -0300 @@ -7763,6 +7763,7 @@ static struct snd_pci_quirk alc883_cfg_t SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG), -- []'s Herton

2 1

[alsa-devel] [PATCH] Fix DAC assignment order in ALC883
by Herton Ronaldo Krzesinski 18 Mar '08

18 Mar '08

Actually clfe and surround DACs are inverted in alc883_dac_nids array (see ALC883 datasheet). I discovered this while testing multi-channel setup (using 3stack-6ch-dig model) on MSI 945GCM5 V2 motherboard that has an ALC883 codec. Simply Rear Left/Right and Center/LFE were swapped in 6 channel mode (also in 4 channel mode you didn't get rear left/right output). Other models also were affected by this bug, as can be seen by the mixer layouts that "workaround" this (the real bug was not noticed, and some other models simply played with mixer and initial verbs). Thus along with fixing the order of dac nids, also change the models that relied on previous dac ordering properly. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff -p -up alsa-kernel/pci/hda/patch_realtek.c.orig alsa-kernel/pci/hda/patch_realtek.c --- alsa-kernel/pci/hda/patch_realtek.c.orig 2008-03-18 00:16:12.000000000 -0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2008-03-18 00:18:52.000000000 -0300 @@ -6450,7 +6450,7 @@ static int patch_alc882(struct hda_codec static hda_nid_t alc883_dac_nids[4] = { /* front, rear, clfe, rear_surr */ - 0x02, 0x04, 0x03, 0x05 + 0x02, 0x03, 0x04, 0x05 }; static hda_nid_t alc883_adc_nids[2] = { @@ -6958,12 +6958,12 @@ static struct snd_kcontrol_new alc883_me static struct snd_kcontrol_new alc888_6st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -6998,12 +6998,12 @@ static struct snd_kcontrol_new alc888_6s static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), @@ -7036,12 +7036,12 @@ static struct snd_kcontrol_new alc888_3s static struct snd_kcontrol_new alc888_6st_dell_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -7332,26 +7332,14 @@ static struct hda_verb alc883_haier_w66_ { } /* end */ }; -static struct hda_verb alc888_6st_hp_verbs[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 2 (0x0e) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 1 (0x0d) */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ - { } -}; - static struct hda_verb alc888_3st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ + {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ { } }; static struct hda_verb alc888_6st_dell_verbs[] = { - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 1 (0x0e) */ - {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 2 (0x0d) */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, { } }; @@ -7985,7 +7973,7 @@ static struct alc_config_preset alc883_p }, [ALC888_6ST_HP] = { .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc888_6st_hp_verbs }, + .init_verbs = { alc883_init_verbs }, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, -- []'s Herton

2 1

[alsa-devel] query about snd_pcm_hw_refine
by Pharaoh . 18 Mar '08

18 Mar '08

Hi I get following error when trying to run aplay with my plugin: #aplay -D myplug test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, St ereo aplay: set_params:882: Broken configuration for this PCM: no configurations available. After looking at the code, I figured out that its snd_pcm_hw_refine which is causing problem. It looks like following in pcm_params.c in alsa lib 1.0.13, int snd_pcm_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) { int res; #ifdef REFINE_DEBUG snd_output_t *log; snd_output_stdio_attach(&log, stderr, 0); #endif printf("#####%s \n", __FUNCTION__); assert(pcm && params); #ifdef REFINE_DEBUG snd_output_printf(log, "REFINE called:\n"); snd_pcm_hw_params_dump(params, log); #endif printf("#####%s calling hw_refine\n", __FUNCTION__); res = pcm->ops->hw_refine(pcm->op_arg, params); printf("#####% return val of hw_refine = %d\n", res); #ifdef REFINE_DEBUG snd_output_printf(log, "refine done - result = %i\n", res); snd_pcm_hw_params_dump(params, log); snd_output_close(log); #endif return res; } I have modified the alsa.conf for my plugin by adding following in it at the end, pcm.myplug { type myplug card 0 } What could be the problem? -pharaoh.

1 3

[alsa-devel] DbAu1200/1210/1250 AC97 Driver
by Chris Gray 17 Mar '08

17 Mar '08

Hello, After looking for DbAu12xx AC97 ALSA support I ported the existing drivers to support Au1200 AC'97 PSC configuration. Being new to the development mailing list, is there a protocol on how to post patches for this sort of thing? Is this even the right forum? Thanks in advance! -Chris

1 0

[alsa-devel] no front speaker sound with ALC262
by Rene Dohmen 17 Mar '08

17 Mar '08

Dear List, I have a problem with an Intel HDA soundcard lspci output: ----- 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 01) The sound volume is very very low when i use the headphone output. The device has integrated speakers but i don't hear any sound coming out of them (maybe because the volume is so low...). I've put all the mixer channels to the maximum values. When booted in Win XP the soundcard works correctly. Is see one line in my logs that indicate that something is wrong: hda_codec: Unknown model for ALC262, trying auto-probe from BIOS... I didn't have it working with other versions of ALSA. Tested with 1.0.14(provided with ubunty gutsy), then started reading, lots of suggestions stated that the problem would be solved in 1.0.15, so i compiled 1.0.16 from src and replaced the default alsa modules with the new ones. De device i 'm using is supplied by tangent ( http://www.tangent.com/t_all_in_one_LCDPC/vita_8000s.htm) its an all in one solution with touchscreen and speakers integrated in one small cabinet. I runned the alsa-info.sh script output and uploaded it: http://pastebin.ca/913865 I don't know if this is the correct place for getting help but all help is appreciated, Best Regards, Rene Dohmen

5 14

[alsa-devel] Emvy24control s/pidf input issue
by Udo van den Heuvel 17 Mar '08

17 Mar '08

Hello, I just noticed that S/PDIF input on my Audiophile 2496 does not work. I see: [udo@wda ~]$ envy24control using --- input_channels: 0 --- output_channels: 4 --- pcm_output_channels: 8 --- spdif in/out channels: 2 Unable to read S/PDIF input switch: No such file or directory Changing the Digital Input from coax to optical or vice versa gives: Unable to write S/PDIF on/off switch: No such file or directory Unable to write S/PDIF input switch: No such file or directory This is on Fedora 8 x86_64 with alsa-tools 1.0.12-4 from Fedora Core 7, envycontrol 0.6.0. S/PDIF output works without a problem. Did not have S/PDIF input working before, as far as I remember. Using -v I do not see signal in the meters. In the source I see in spdif_input_update: if ((card_eeprom.subvendor != ICE1712_SUBDEVICE_DELTADIO2496) && ! card_is_dmx6fire) return; if ((err = snd_ctl_elem_read(ctl, spdif_input)) < 0) g_print("Unable to read S/PDIF input switch: %s\n", snd_strerror(err)); (etc) Why does snd_ctl_elem_read fail? How could I make S/PDIF input work again? Tips are very welcome! Kind regards, Udo

2 4

[alsa-devel] AD1988B: ALSA doesn't use dmix automatically
by Dâniel Fraga 17 Mar '08

17 Mar '08

ALSA 1.0.15 http://alsa.opensrc.org/index.php/Dmix NOTE: For ALSA 1.0.9rc2 and higher you don't need to setup dmix. Dmix is enabled as default for soundcards which don't support hw mixing. So why if I try to play 2 sounds at the same time, the second one complains the "resource is busy"? Shouldn't it detect that AD1988B can't do hardware mixing and use dmix automatically? Obs.: I don't have any ~/.asoundrc or /etc/asound.conf lying around. Thanks! -- Linux 2.6.24: Arr Matey! A Hairy Bilge Rat! http://u-br.net http://www.abusar.org/FELIZ_2008.html Napalm Death - "Circle Of Hypocrisy" (Harmony Corruption - 1990)

4 4

Re: [alsa-devel] ALSACTL wrong restore on ARM9
by Liam Girdwood 17 Mar '08

17 Mar '08

On Fri, 2008-03-07 at 16:28 +0100, Alexandre BOUIN wrote: > > Can you describe 'bad values' in a little more detail. > > You're right, some details are missing. > > For example, for first control, a playback volume control, value can go from > 0 to 63. > Before "store" command, I'd set it to 20. > I've check that in "/var/lib/alsa/asound/state", I had the same value. Here > is a part of this file : > > state.codec { > control.1 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 63' > iface MIXER > name 'HP Playback Volume' > value.0 20 > value.1 20 > } > > So saved value seems ok. > > After "restore" command, I get this : > > $amixer cget numid=1 > numid=1,iface=MIXER,name='HP Playback Volume' > ; type=INTEGER,access=rw------,values=2,min=0,max=63,step=0 > : values=45,45 > > That's why I say that values are not correct. > I'm currently using debian alsa-utils_1.0.16-1_arm package. > It's possible that your codec drivers register caching is borked. This can be easily confirmed with debug in your driver kcontrol code. If your driver looks ok, then I'd suggest looking into your alsa-lib cross build. There have been some problems with it being incorrectly cross compiled for ARM in the past (pls search the list for details). Fwiw, I use OpenEmbedded to build ALSA for ARM and I've never ran into this issue. Liam

2 2

[alsa-devel] PATCH - some fixes and cleanup for ICE1724 cards
by Pavel Hofman 17 Mar '08

17 Mar '08

Hi Takashi, here is a small patch with minor fixes and cleanup. * removing the hack with NON_AKM ak4xxx type * support for card-specific flags in ak4114_stats * definition of the flags for corresponding cards Signed-by: Pavel Hofman <dustin(a)seznam.cz> Thanks, Pavel

2 1