May 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda/realtek - ALC292 dock fix for Thinkpad L450
by Ansgar Hegerfeld 18 May '15

18 May '15

The Lenovo ThinkPad L450 requires the ALC292_FIXUP_TPT440_DOCK fix in order to get sound output on the docking stations audio port. This patch was tested using a ThinkPad L450 (20DSS00B00) using kernel 4.0.3 and a ThinkPad Pro Dock. Signed-off-by: Ansgar Hegerfeld <linux(a)hegerfeld.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e2afd53..a9f41a4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5148,6 +5148,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),

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[alsa-devel] [PATCH] ALSA: hda - Add Conexant codecs CX20721, CX20722, CX20723 and CX20724
by David Henningsson 18 May '15

18 May '15

This patch adds support for Conexant HD Audio codecs CX20721, CX20722, CX20723 and CX20724. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1454656 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f8f0dfb..78b719b 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -968,6 +968,14 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_conexant_auto }, { .id = 0x14f150b9, .name = "CX20665", .patch = patch_conexant_auto }, + { .id = 0x14f150f1, .name = "CX20721", + .patch = patch_conexant_auto }, + { .id = 0x14f150f2, .name = "CX20722", + .patch = patch_conexant_auto }, + { .id = 0x14f150f3, .name = "CX20723", + .patch = patch_conexant_auto }, + { .id = 0x14f150f4, .name = "CX20724", + .patch = patch_conexant_auto }, { .id = 0x14f1510f, .name = "CX20751/2", .patch = patch_conexant_auto }, { .id = 0x14f15110, .name = "CX20751/2", @@ -1002,6 +1010,10 @@ MODULE_ALIAS("snd-hda-codec-id:14f150ab"); MODULE_ALIAS("snd-hda-codec-id:14f150ac"); MODULE_ALIAS("snd-hda-codec-id:14f150b8"); MODULE_ALIAS("snd-hda-codec-id:14f150b9"); +MODULE_ALIAS("snd-hda-codec-id:14f150f1"); +MODULE_ALIAS("snd-hda-codec-id:14f150f2"); +MODULE_ALIAS("snd-hda-codec-id:14f150f3"); +MODULE_ALIAS("snd-hda-codec-id:14f150f4"); MODULE_ALIAS("snd-hda-codec-id:14f1510f"); MODULE_ALIAS("snd-hda-codec-id:14f15110"); MODULE_ALIAS("snd-hda-codec-id:14f15111"); -- 1.9.1

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[alsa-devel] [PATCH 0/4] Fixes and cleanups for the recent procfs changes
by Takashi Iwai 18 May '15

18 May '15

Hi, here is a series of patches to fix and clean up the recent changes for ALSA core procfs support code. Takashi === Takashi Iwai (4): ALSA: info: Fix leaks of child entries at snd_info_free_entry() ALSA: info: Register proc entries recursively, too ALSA: info: Move list addition to snd_info_create_entry() ALSA: info: Drop kerneldoc comment from snd_info_create_entry() sound/core/info.c | 62 +++++++++++++++++++++++++++++++++++++++++-------------- sound/core/init.c | 18 ++++++---------- 2 files changed, 52 insertions(+), 28 deletions(-) -- 2.3.6

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[alsa-devel] [PATCH 1/1] ALSA: core: Fix regression for ELD/codec information files
by han.lu＠intel.com 18 May '15

18 May '15

From: "Lu, Han" <han.lu(a)intel.com> The commit c560a6797e3b ("ALSA: core: Remove child proc file elements recursively") introduced a regression that ELD/codec information files can not be found on /proc/asound/cardX/. This patch corrects it. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/include/sound/info.h b/include/sound/info.h index 1626995..ae8d30a 100644 --- a/include/sound/info.h +++ b/include/sound/info.h @@ -149,6 +149,10 @@ static inline int snd_card_proc_new(struct snd_card *card, const char *name, struct snd_info_entry **entryp) { *entryp = snd_info_create_card_entry(card, name, card->proc_root); + if (snd_info_register(*entryp) < 0) { + snd_info_free_entry(*entryp); + *entryp = NULL; + } return *entryp ? 0 : -ENOMEM; } -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - Fix headset mic and mic-in for a Dell desktop
by David Henningsson 18 May '15

18 May '15

ALC662 does not need any special verbs to change the jack functionality, and enables mic in through the headphone jack mode by changing the direction of the headphone pin node. BugLink: https://bugs.launchpad.net/bugs/1454235 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 46 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 45 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ce41271..30ec30f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3674,6 +3674,10 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, alc_process_coef_fw(codec, coef0293); snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); break; + case 0x10ec0662: + snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); + snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); + break; case 0x10ec0668: alc_write_coef_idx(codec, 0x11, 0x0001); snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); @@ -4012,7 +4016,7 @@ static void alc_update_headset_mode(struct hda_codec *codec) if (new_headset_mode != ALC_HEADSET_MODE_MIC) { snd_hda_set_pin_ctl_cache(codec, hp_pin, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); - if (spec->headphone_mic_pin) + if (spec->headphone_mic_pin && spec->headphone_mic_pin != hp_pin) snd_hda_set_pin_ctl_cache(codec, spec->headphone_mic_pin, PIN_VREFHIZ); } @@ -4215,6 +4219,18 @@ static void alc_fixup_dell_xps13(struct hda_codec *codec, } } +static void alc_fixup_headset_mode_alc662(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; + spec->gen.hp_mic = 1; /* Mic-in is same pin as headphone */ + } else + alc_fixup_headset_mode(codec, fix, action); +} + static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6080,7 +6096,9 @@ enum { ALC662_FIXUP_NO_JACK_DETECT, ALC662_FIXUP_ZOTAC_Z68, ALC662_FIXUP_INV_DMIC, + ALC662_FIXUP_DELL_MIC_NO_PRESENCE, ALC668_FIXUP_DELL_MIC_NO_PRESENCE, + ALC662_FIXUP_HEADSET_MODE, ALC668_FIXUP_HEADSET_MODE, ALC662_FIXUP_BASS_MODE4_CHMAP, ALC662_FIXUP_BASS_16, @@ -6273,6 +6291,20 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE }, + [ALC662_FIXUP_DELL_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a1113c }, /* use as headset mic, without its own jack detect */ + /* headphone mic by setting pin control of 0x1b (headphone out) to in + vref_50 */ + { } + }, + .chained = true, + .chain_id = ALC662_FIXUP_HEADSET_MODE + }, + [ALC662_FIXUP_HEADSET_MODE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_headset_mode_alc662, + }, [ALC668_FIXUP_DELL_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -6432,6 +6464,18 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {0x1f, 0x411111f0} static const struct snd_hda_pin_quirk alc662_pin_fixup_tbl[] = { + SND_HDA_PIN_QUIRK(0x10ec0662, 0x1028, "Dell", ALC662_FIXUP_DELL_MIC_NO_PRESENCE, + {0x12, 0x4004c000}, + {0x14, 0x01014010}, + {0x15, 0x411111f0}, + {0x16, 0x411111f0}, + {0x18, 0x01a19020}, + {0x19, 0x411111f0}, + {0x1a, 0x0181302f}, + {0x1b, 0x0221401f}, + {0x1c, 0x411111f0}, + {0x1d, 0x4054c601}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0668, 0x1028, "Dell", ALC668_FIXUP_AUTO_MUTE, ALC668_STANDARD_PINS, {0x12, 0x99a30130}, -- 1.9.1

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[alsa-devel] [GIT PULL] ASoC fixes for v4.1
by Mark Brown 18 May '15

18 May '15

The following changes since commit 030bbdbf4c833bc69f502eae58498bc5572db736: Linux 4.1-rc3 (2015-05-10 15:12:29 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.1-rc3 for you to fetch changes up to 7730c0b55043f6ff1c27e9cb45e13679995b2361: Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/mc13783', 'asoc/fix/uda1380', 'asoc/fix/wm8960' and 'asoc/fix/wm8994' into asoc-linus (2015-05-12 20:31:33 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.1 A few more fixes for v4.1, some driver fixes plus one core fix which fixes registration of DAI links when adding prefixes to CODECs to deuplicate in multi-CODEC systems. ---------------------------------------------------------------- Axel Lin (2): ASoC: mc13783: Fix wrong mask value used in mc13xxx_reg_rmw() calls ASoC: rt5645: Fix mask for setting RT5645_DMIC_2_DP_GPIO12 bit Fang, Yang A (1): ASoC: rt5645: Add ACPI match ID Koro Chen (1): ASoC: dapm: Modify widget stream name according to prefix Mark Brown (3): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/mc13783', 'asoc/fix/uda1380', 'asoc/fix/wm8960' and 'asoc/fix/wm8994' into asoc-linus Peter Ujfalusi (1): ASoC: davinci-mcasp: Correct pm status check in suspend callback Vasily Khoruzhick (1): ASoC: uda1380: Avoid accessing i2c bus when codec is disabled Zidan Wang (2): ASoC: wm8960: fix "RINPUT3" audio route error ASoC: wm8994: correct BCLK DIV 348 to 384 sound/soc/codecs/mc13783.c | 4 ++-- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wm8960.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/soc-dapm.c | 11 ++++++++--- 6 files changed, 14 insertions(+), 9 deletions(-)

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Re: [alsa-devel] Fwd: Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 17 May '15

17 May '15

> > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/patch/sound/pci > > /hda/patch_realtek.c?id=c3e837bbcc03e39991da5dd1f791c06cde9f7a45 > > > > When the static model were used, the jack was configured as line out > > instead of headphone jack and there is no headphone playback volume > > That's rather "relevant change". Good job in finding it. I would like such > configuration (or better) back. > I don't see any side effect neither with your changes to hda_generic nor > with > vanilla kernel, but I don't know what problems should I expect. > Do headphone get same volume level when the jack is configured as Line Out (without enable Headphone Amplifier) ? The driver use different automute method when the jack is configured as HP and Line Out https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… if you look at speaker.conf , lineout.conf and headphone.conf, it seem pulseaudio is unlikely mute all internal speakers when one pair of the playback volume control need to be shared by internal speaker and external speakers you need dirver 's automute instead of pulseaudio 's automute ==> Best config: lo_type=2, wired=1, mio=1 multi_outs = 15/0/0/0 : 2/3/4/0 (type HP) out path: depth=3 '02:0c:15' multi_ios(2) = 1a/18 : 3/4 mio path: depth=3 '03:0d:1a' mio path: depth=3 '04:0e:18' spk_outs = 14/17/16/0 : 5/25/4/0 spk path: depth=3 '05:0f:14' spk path: depth=3 '25:26:17' spk path: depth=3 '04:0e:16' https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… when line out type == AUTO_PIN_HP_OUT should first do_mute() use speaker_path instead of out_path <https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p…> >>> I started MS Windows and without externals speakers plugged in I >> >> could test >> > 5.1 configuration and it seemed that sound was coming from >> different directions. >> >> > >> > What I managed to determine: >> > I've got 3 stacks: >> > 1 ST: Headphone, Line-out (FL and FR), SPDIF >> > 2 ST: Microphone, Line-out (CLFE) >> > 3 ST: Line-in, Line-out (RL and RR) >> > >> > I've got 5 responsive output pins >> > PIN 0x14: Internal FL FR >> > PIN 0x15: External FL FR >> > PIN 0x17: Internal LFE >> > PIN 0x18: External CLFE >> > PIN 0x1a: External RL RR >> > >> > How can we make it work with these information? speaker-test -Dhw:0,0 -c6 -t wav >> >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/plain/Document… >> [1] >> >> seem chnmap type _FIXED , _VAR and _PAIRED does not fit when LFE and >> FC can be swapped for internal speaker and external speaker > > > Right, you can't do it with a single map of one of those types. http://git.alsa-project.org/?p=alsa-utils.git&a=search&h=HEAD&st=commit&s=c… do chmap of internal speakers need {FL FR RL RR LFE NA} ? http://git.alsa-project.org/?p=alsa-lib.git&a=search&h=HEAD&st=commit&s=chm… as set_chmap() still not yet implemented in pcm_route.c do speaker-test need to specify channel map when using those odd number channels ? speaker-test -D surround21 -c3 -t wav speaker-test -D surround41 -c5 -t wav -mFL,FR,RL,RR,LFE speaker-test -D surround50 -c5 -t wav http://git.alsa-project.org/?p=alsa-utils.git;a=commitdiff;h=7b14c00b56784f… this example seem not working speaker-test -c2 -t wav -Dplug:front -mFR,FL > > >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >> [2] >> >> How do the driver implement channel map when codec has swap >> center/lfe switch ? >> >> Do chmap contain both {FL FR RL RR FC LFE} and {FL FR RL RR LFE FC} ? > > > If you want to provide support for userspace doing it both ways, then yes, you need those two _FIXED maps. Can chmap contain two _FIXED maps of same number of channels ?

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[alsa-devel] Microphone issues with Microsoft LifeCam Studio
by Dainius Masiliūnas 16 May '15

16 May '15

I'm having an issue with the Microsoft LifeCam Studio in that the camera does not work if the integrated microphone in the camera is activated at the same time. When that happens, the kernel log outputs a variety of warnings such as: cannot get freq at ep 0x82 ALSA lib pcm_dsnoop.c:614:(snd_pcm_dsnoop_open) unable to open slave uvcvideo: Failed to query (GET_DEF) UVC control 2 on unit 4: -32 (exp. 2). uvcvideo: Failed to query (GET_DEF) UVC control 2 on unit 4: 0 (exp. 2). ALSA lib pcm.c:7980:(snd_pcm_set_params) Channels count (1) not available for CAPTURE: Incorrect argument ALSA lib pcm.c:7956:(snd_pcm_set_params) Broken configuration for CAPTURE: no configurations available ALSA lib pcm_route.c:867:(find_matching_chmap) Found no matching channel map uvcvideo: Failed to set UVC probe control : -32 (exp. 26). There is a workaround in that disabling the built-in microphone by blacklisting snd_usb_audio does make the camera work, but obviously it's not ideal at all, as that way half the camera's functionality is not available and no other USB audio peripherals can be used. I'm assuming that this is some sort of a quirk in the camera itself; the UVC page ( http://www.ideasonboard.org/uvc/#footnote-13 ) does mention that there is an issue with the bandwidth requests, but setting the 0x80 quirk does not change the situation with the microphone in any way, unfortunately. I am running the latest mainline kernel (kernel-ml from elrepo) on CentOS 7.1. Can anything be done about this issue, and is there anything I could do to help debug it? Also, this issue has been noted a few years back, but it was left unsolved: https://bugs.launchpad.net/ubuntu/+source/guvcview/+bug/1033146 https://sourceforge.net/p/linux-uvc/mailman/message/29645043/ The original reporter said it was a Flash bug, but that is in no way the case because my machine does not have Flash installed to begin with (I'm testing this with WebRTC). -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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Re: [alsa-devel] [pulseaudio-discuss] ThinkPad T-510 audio output mute LED non-workingness
by Takashi Iwai 16 May '15

16 May '15

[Switched Cc to alsa-devel ML, as it's irrelevant from PA but rather a driver issue.] At Mon, 27 Apr 2015 07:58:59 -0600, Glenn Golden wrote: > > Takashi Iwai <tiwai(a)suse.de> [2015-04-27 08:50:34 +0200]: > > > > Gah, I hate reports without alsa-info.sh output (not a link to > > somewhere else). > > > > Glenn, please attach alsa-info.sh outputs (run it with --no-upload > > option) and the whole dmesg output after boot. 3.19 should already > > contain the hook to thinkpad_acpi.c. If anything is missing, you > > should have a warning indicating it. > > > > Apologies for the omissions. Both are attached. Thanks! > Fwiw -- perhaps nothing at this point -- a good deal of info has also been > posted to the kernel bugtracker ticket: > > https://bugzilla.kernel.org/show_bug.cgi?id=96171 > > Re: "...you should have a warning indicating it": I'm an unskilled laboror > here w.r.t. kernel affairs, and not even sure what sort of warnings to be > looking for, so may have missed something important without realizing it. > Just let me know what add'l info you need, I'll be happy to post it. I do > appreciate your attention to the issue. I see no such warning in the log you attached, so it's not the issue, as it seems. Below is a debug patch, and now you should see messages like "XXX ..." in kernel log at loading the driver. If there is no such a message, it means that the thinkpad hook isn't built in at all. Please check the kernel log again with the patch. Takashi --- diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 0a4ad5feb82e..48ce61e46599 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -34,6 +34,7 @@ static void update_tpacpi_mute_led(void *private_data, int enabled) if (old_vmaster_hook) old_vmaster_hook(private_data, enabled); + pr_info("XXX mute led updated %d\n", enabled); if (led_set_func) led_set_func(TPACPI_LED_MUTE, !enabled); } @@ -58,8 +59,10 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, bool removefunc = false; if (action == HDA_FIXUP_ACT_PROBE) { + codec_info(codec, "XXX hda_fixup_thinkpad_acpi init...\n"); if (!is_thinkpad(codec)) return; + codec_info(codec, "XXX ok, this is thinkpad\n"); if (!led_set_func) led_set_func = symbol_request(tpacpi_led_set); if (!led_set_func) { @@ -70,18 +73,24 @@ static void hda_fixup_thinkpad_acpi(struct hda_codec *codec, removefunc = true; if (led_set_func(TPACPI_LED_MUTE, false) >= 0) { + codec_info(codec, "XXX ok, mute LED is hooked\n"); old_vmaster_hook = spec->vmaster_mute.hook; spec->vmaster_mute.hook = update_tpacpi_mute_led; removefunc = false; + } else { + codec_err(codec, "XXX mute LED NOT WORKING!\n"); } if (led_set_func(TPACPI_LED_MICMUTE, false) >= 0) { if (spec->num_adc_nids > 1) codec_dbg(codec, "Skipping micmute LED control due to several ADCs"); else { + codec_info(codec, "XXX ok, mic LED is hooked\n"); spec->cap_sync_hook = update_tpacpi_micmute_led; removefunc = false; } + } else { + codec_err(codec, "XXX mic LED NOT WORKING!\n"); } }

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[alsa-devel] [PATCH] ASoC: wm8955: Fix setting wrong register for WM8955_K_8_0_MASK bits
by Axel Lin 16 May '15

16 May '15

WM8955_K_8_0_MASK bits is controlled by WM8955_PLL_CONTROL_3 rather than WM8955_PLL_CONTROL_2. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8955.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index f400d5c..08063a6 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -298,7 +298,7 @@ static int wm8955_configure_clocking(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8955_PLL_CONTROL_2, WM8955_K_17_9_MASK, (pll.k >> 9) & WM8955_K_17_9_MASK); - snd_soc_update_bits(codec, WM8955_PLL_CONTROL_2, + snd_soc_update_bits(codec, WM8955_PLL_CONTROL_3, WM8955_K_8_0_MASK, pll.k & WM8955_K_8_0_MASK); if (pll.k) -- 2.1.0

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