January 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: mxs-saif: Use a signed integer for error value
by Fabio Estevam 08 Jan '13

08 Jan '13

saif->id and saif->master_id are unsigned, so they can not be negative. Fix the following warning when building with W=1 option: sound/soc/mxs/mxs-saif.c: In function 'mxs_saif_probe': sound/soc/mxs/mxs-saif.c:676:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:688:3: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] sound/soc/mxs/mxs-saif.c:692:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits] Use a signed variable 'ret' to handle the error values. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 752675d..a920206 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -672,9 +672,9 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!saif) return -ENOMEM; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; + ret = of_alias_get_id(np, "saif"); + if (ret < 0) + return ret; /* * If there is no "fsl,saif-master" phandle, it's a saif * master. Otherwise, it's a slave and its phandle points @@ -684,12 +684,12 @@ static int mxs_saif_probe(struct platform_device *pdev) if (!master) { saif->master_id = saif->id; } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; + ret = of_alias_get_id(master, "saif"); + if (ret < 0) + return ret; } - if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { + if (saif->master_id >= ARRAY_SIZE(mxs_saif)) { dev_err(&pdev->dev, "get wrong master id\n"); return -EINVAL; } -- 1.7.9.5

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[alsa-devel] [PATCH RESEND v5 2/2] ASoC: mxs-saif: Remove platform data
by Fabio Estevam 08 Jan '13

08 Jan '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> All MXS users have been converted to device tree and the board files have been removed. No need to keep platform data in the driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> Acked-by: Dong Aisheng <dong.aisheng(a)linaro.org> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> --- Changes since v4: - Rebased against fix/mxs branch and fix file removal Changes since v3: - Remove pdev->id check Changes since v2: - Also remove 'if (np)' as dt is the only probing mechanism. Changes since v1: - Also remove mxs_saif_platform_data and sound/saif.h, which was used only to include mxs_saif_platform_data definition. include/sound/saif.h | 16 ---------------- sound/soc/mxs/mxs-saif.c | 44 ++++++++++++++++---------------------------- 2 files changed, 16 insertions(+), 44 deletions(-) delete mode 100644 include/sound/saif.h diff --git a/include/sound/saif.h b/include/sound/saif.h deleted file mode 100644 index f22f3e1..0000000 --- a/include/sound/saif.h +++ /dev/null @@ -1,16 +0,0 @@ -/* - * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SAIF_H__ -#define __SOUND_SAIF_H__ - -struct mxs_saif_platform_data { - bool master_mode; /* if true use master mode */ - int master_id; /* id of the master if in slave mode */ -}; -#endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index c294fbb..05dbadb 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -32,7 +32,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <sound/saif.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/mxs.h> @@ -660,43 +659,32 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct mxs_saif_platform_data *pdata; struct pinctrl *pinctrl; int ret = 0; + struct device_node *master; - - if (!np && pdev->id >= ARRAY_SIZE(mxs_saif)) + if (!np) return -EINVAL; saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; - if (np) { - struct device_node *master; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; - /* - * If there is no "fsl,saif-master" phandle, it's a saif - * master. Otherwise, it's a slave and its phandle points - * to the master. - */ - master = of_parse_phandle(np, "fsl,saif-master", 0); - if (!master) { - saif->master_id = saif->id; - } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; - } + saif->id = of_alias_get_id(np, "saif"); + if (saif->id < 0) + return saif->id; + /* + * If there is no "fsl,saif-master" phandle, it's a saif + * master. Otherwise, it's a slave and its phandle points + * to the master. + */ + master = of_parse_phandle(np, "fsl,saif-master", 0); + if (!master) { + saif->master_id = saif->id; } else { - saif->id = pdev->id; - pdata = pdev->dev.platform_data; - if (pdata && !pdata->master_mode) - saif->master_id = pdata->master_id; - else - saif->master_id = saif->id; + saif->master_id = of_alias_get_id(master, "saif"); + if (saif->master_id < 0) + return saif->master_id; } if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { -- 1.7.9.5

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[alsa-devel] [PATCH v5 2/2] ASoC: mxs-saif: Remove platform data
by Fabio Estevam 08 Jan '13

08 Jan '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> All MXS users have been converted to device tree and the board files have been removed. No need to keep platform data in the driver. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> Acked-by: Dong Aisheng <dong.aisheng(a)linaro.org> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> --- Changes since v4: - Rebased against fix/mxs branch and fix file removal Changes since v3: - Remove pdev->id check Changes since v2: - Also remove 'if (np)' as dt is the only probing mechanism. Changes since v1: - Also remove mxs_saif_platform_data and sound/saif.h, which was used only to include mxs_saif_platform_data definition. include/sound/saif.h | 16 ---------------- sound/soc/mxs/mxs-saif.c | 44 ++++++++++++++++---------------------------- 2 files changed, 16 insertions(+), 44 deletions(-) delete mode 100644 include/sound/saif.h diff --git a/include/sound/saif.h b/include/sound/saif.h deleted file mode 100644 index f22f3e1..0000000 --- a/include/sound/saif.h +++ /dev/null @@ -1,16 +0,0 @@ -/* - * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __SOUND_SAIF_H__ -#define __SOUND_SAIF_H__ - -struct mxs_saif_platform_data { - bool master_mode; /* if true use master mode */ - int master_id; /* id of the master if in slave mode */ -}; -#endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index c294fbb..05dbadb 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -32,7 +32,6 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <sound/saif.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/mxs.h> @@ -660,43 +659,32 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct resource *iores, *dmares; struct mxs_saif *saif; - struct mxs_saif_platform_data *pdata; struct pinctrl *pinctrl; int ret = 0; + struct device_node *master; - - if (!np && pdev->id >= ARRAY_SIZE(mxs_saif)) + if (!np) return -EINVAL; saif = devm_kzalloc(&pdev->dev, sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; - if (np) { - struct device_node *master; - saif->id = of_alias_get_id(np, "saif"); - if (saif->id < 0) - return saif->id; - /* - * If there is no "fsl,saif-master" phandle, it's a saif - * master. Otherwise, it's a slave and its phandle points - * to the master. - */ - master = of_parse_phandle(np, "fsl,saif-master", 0); - if (!master) { - saif->master_id = saif->id; - } else { - saif->master_id = of_alias_get_id(master, "saif"); - if (saif->master_id < 0) - return saif->master_id; - } + saif->id = of_alias_get_id(np, "saif"); + if (saif->id < 0) + return saif->id; + /* + * If there is no "fsl,saif-master" phandle, it's a saif + * master. Otherwise, it's a slave and its phandle points + * to the master. + */ + master = of_parse_phandle(np, "fsl,saif-master", 0); + if (!master) { + saif->master_id = saif->id; } else { - saif->id = pdev->id; - pdata = pdev->dev.platform_data; - if (pdata && !pdata->master_mode) - saif->master_id = pdata->master_id; - else - saif->master_id = saif->id; + saif->master_id = of_alias_get_id(master, "saif"); + if (saif->master_id < 0) + return saif->master_id; } if (saif->master_id < 0 || saif->master_id >= ARRAY_SIZE(mxs_saif)) { -- 1.7.9.5

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Re: [alsa-devel] About Alsa arecord
by Raymond Yau 08 Jan '13

08 Jan '13

> Good morning! > Are you saying I should modify the source code of alsa-driver-1.0.25? >> >> > Hello! >> > There have 2 stereo ADCs support one stereo microphone and one legacy mixer recording simultaneously, said in the document of ALC888 Datasheet .I inputed 1KHz sine wave into every pins (Line1-R,Line1-L,MIC1-R,MIC1_L,MIC2-R,MIC2-L,LINE2-R,LINE2-L) of ALC888 by signal genrator,and used command 'arecord -c 4 -f S16_LE -r 44100 >test.wav' recording in fedora14.but ALL audio data are recorded in channel 1,2.No any data in channel 3,4.Why?How can I record every pins audio data in each channel? >> >> In alc_auto_create_input_ctls , add the connection from node 0x0b of mixer ADC as input source similar to those pin complex >> >> Those Mic/line/CD playback volume and switches of node 0x0b affect the signal of the new input source ”mixer” > > Yes, also need a switch which mute the input from 0x0b for all output jacks to prevent feedback if using mic the current hda driver only support stereo capture and you have to implement multi channel capturing similar to multi channel playback in hda_codec.c snd_hda_multi_in_analog_prepare for (i = 1; i < num_adcs; i++) { snd_hda_codec_setup_stream(codec, adc_nids[i], stream_tag, i * 2, format); or try multi plugin of two stereo capture devices

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[alsa-devel] [PATCH] ASoC: wm_adsp: Ensure that block writes are from DMA aligned addresses
by Mark Brown 07 Jan '13

07 Jan '13

Otherwise we won't run correctly on systems that require this for larger data transfers. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5d8bb1d..a765458 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -169,6 +169,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) const struct wm_adsp_region *mem; const char *region_name; char *file, *text; + void *buf; unsigned int reg; int regions = 0; int ret, offset, type, sizes; @@ -322,8 +323,18 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - ret = regmap_raw_write(regmap, reg, region->data, + buf = kmemdup(region->data, le32_to_cpu(region->len), + GFP_KERNEL); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + return -ENOMEM; + } + + ret = regmap_raw_write(regmap, reg, buf, le32_to_cpu(region->len)); + + kfree(buf); + if (ret != 0) { adsp_err(dsp, "%s.%d: Failed to write %d bytes at %d in %s: %d\n", -- 1.7.10.4

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[alsa-devel] [PATCH 0/4] ALSA: pxa27x: ac97 reset fixes
by Mike Dunn 07 Jan '13

07 Jan '13

AC97 reset is broken on the pxa27x. These patches fix it by addressing multiple problems: - cold reset fails because the GCR register is not set correctly - warm reset fails because some code used to work around a hardware bug is broken - the intent of the work-around code is obfuscated by unfortunate function naming This patch set replaces some earlier individual patches that had some problems. Thanks Robert and Igor. Mike Dunn (4): ALSA: pxa27x: fix ac97 cold reset ALSA: pxa27x: fix ac97 warm reset bug work-around code ALSA: pxa27x: ac97 controller driver requests gpio ALSA: pxa27x: rename pxa27x_assert_ac97() arch/arm/mach-pxa/include/mach/mfp-pxa27x.h | 3 ++ arch/arm/mach-pxa/pxa27x.c | 38 +++++++++++++++++++------- sound/arm/pxa2xx-ac97-lib.c | 26 +++++++++++++----- 3 files changed, 50 insertions(+), 17 deletions(-) -- 1.7.8.6

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[alsa-devel] [PATCH v2] ALSA: hda - add mute LED for HP Pavilion 17 (Realtek codec)
by David Henningsson 07 Jan '13

07 Jan '13

The mute LED is in this case connected to the Mic1 VREF. The machine also exposes the following string in BIOS: "HP_Mute_LED_0_A", so if more machines are coming, it probably makes sense to try to do something more generic, like for the IDT codec. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1096789 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6ee3459..71ae23d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5992,6 +5992,30 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec, spec->automute_hook = alc269_quanta_automute; } +/* update mute-LED according to the speaker mute state via mic1 VREF pin */ +static void alc269_fixup_mic1_mute_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + unsigned int pinval = AC_PINCTL_IN_EN + (enabled ? + AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80); + snd_hda_set_pin_ctl_cache(codec, 0x18, pinval); +} + +static void alc269_fixup_mic1_mute(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + switch (action) { + case ALC_FIXUP_ACT_BUILD: + spec->vmaster_mute.hook = alc269_fixup_mic1_mute_hook; + snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true); + /* fallthru */ + case ALC_FIXUP_ACT_INIT: + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + break; + } +} + /* update mute-LED according to the speaker mute state via mic2 VREF pin */ static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled) { @@ -6043,6 +6067,7 @@ enum { ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, + ALC269_FIXUP_MIC1_MUTE_LED, ALC269_FIXUP_MIC2_MUTE_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, @@ -6171,6 +6196,10 @@ static const struct alc_fixup alc269_fixups[] = { { } }, }, + [ALC269_FIXUP_MIC1_MUTE_LED] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_mic1_mute, + }, [ALC269_FIXUP_MIC2_MUTE_LED] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_mic2_mute, @@ -6215,6 +6244,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), -- 1.7.9.5

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[alsa-devel] Alsa snapshot directory layout changed
by David Henningsson 07 Jan '13

07 Jan '13

I have a script that picks daily snapshots from ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-snapshot.tar.gz ...and I just noticed that the directory layout has changed, so what was "alsa-driver" is now "alsa-driver/alsa". Is this expected, i e, should I update my script to follow this new layout, or was it a mistake? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] [PATCH] ALSA: pxa2xx: fix ac97 cold reset for pxa27x
by Mike Dunn 07 Jan '13

07 Jan '13

Currently, ac97 reset is broken on PXA27x. Through trial-and-error (the pxa270 developer's manual is mostly incoherent on the topic of ac97 reset), I fixed it by setting the WARM_RST bit in the GCR register at the end of the cold reset, and then skipping the warm reset if the link is already up and running. It appears that setting the WARM_RST bit is a necessary final step during cold reset. I think that the PXA25x and PXA3xx may currently be working correctly because WARM_RST is set within the warm reset routine, and the codec drivers always follow a cold reset with a warm reset during their initialization. so this combination effectively completes a cold reset. This doesn't work on the PXA27x because its warm reset routine contains additional code for working around a hardware bug in the warm reset sequence, which causes the reset sequence to fail. I only have a PXA27x platform for testing, so this patch only affects the PXA27x. Signed-off-by: Mike Dunn <mikedunn(a)newsguy.com> --- As an aside... I don't understand how always following a cold reset with a warm reset (as the codec drivers do) makes sense. Unless I'm mistaken, a warm reset can only occur on a link that's currently down. All the timing diagrams I've seen for reset (warm and cold) show the link down (BITCLK stopped) at the start of the reset sequence. The SYNC line is used to effect the warm reset, and SYNC is already pulsing every frame on a link that's up and running, so it seems physically impossible to do a warm reset on a link that's currently up. But I'm no ac97 expert, so I may just be embarassing myself. Any insight gratefully received. Thanks! sound/arm/pxa2xx-ac97-lib.c | 15 +++++++++++++++ 1 files changed, 15 insertions(+), 0 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 6fc0ae9..aa242ff 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -138,6 +138,10 @@ static inline void pxa_ac97_warm_pxa27x(void) { gsr_bits = 0; + /* don't attempt warm reset if link is up */ + if (GSR & (GSR_SCR | GSR_PCR)) + return; + /* warm reset broken on Bulverde, so manually keep AC97 reset high */ pxa27x_assert_ac97reset(reset_gpio, 1); udelay(10); @@ -148,6 +152,8 @@ static inline void pxa_ac97_warm_pxa27x(void) static inline void pxa_ac97_cold_pxa27x(void) { + unsigned int timeout; + GCR &= GCR_COLD_RST; /* clear everything but nCRST */ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ @@ -159,6 +165,15 @@ static inline void pxa_ac97_cold_pxa27x(void) clk_disable(ac97conf_clk); GCR = GCR_COLD_RST; udelay(50); + + /* + * Setting the WARM_RST bit is a necessary part of a cold reset, at + * least on the buggy and poorly documented PXA27x ac97 controller. + */ + GCR |= GCR_WARM_RST; + timeout = 100; /* wait for the codec-ready bit to be set */ + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); } #endif -- 1.7.8.6

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[alsa-devel] [PATCH 1/2] ALSA: hda - add mute LED for HP Pavilion 17 (Realtek codec)
by David Henningsson 07 Jan '13

07 Jan '13

The mute LED is in this case connected to the Mic1 VREF. The machine also exposes the following string in BIOS: "HP_Mute_LED_0_A", so if more machines are coming, it probably makes sense to try to do something more generic, like for the IDT codec. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1096789 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 31 ++++++++++++++++++++++++++++++- 1 file changed, 30 insertions(+), 1 deletion(-) I know this is a copy-paste, but I didn't really find a better and more condensed way to do it. Perhaps if we get a third variation it would make sense to write a more generic version. Also, Alsa-info is available at https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1096789/+attachment/34… diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6ee3459..2e6803e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5992,7 +5992,30 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec, spec->automute_hook = alc269_quanta_automute; } -/* update mute-LED according to the speaker mute state via mic2 VREF pin */ +/* update mute-LED according to the speaker mute state via mic VREF pin */ +static void alc269_fixup_mic1_mute_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + unsigned int pinval = AC_PINCTL_IN_EN + (enabled ? + AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80); + snd_hda_set_pin_ctl_cache(codec, 0x18, pinval); +} + +static void alc269_fixup_mic1_mute(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + switch (action) { + case ALC_FIXUP_ACT_BUILD: + spec->vmaster_mute.hook = alc269_fixup_mic1_mute_hook; + snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true); + /* fallthru */ + case ALC_FIXUP_ACT_INIT: + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + break; + } +} + static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled) { struct hda_codec *codec = private_data; @@ -6043,6 +6066,7 @@ enum { ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, + ALC269_FIXUP_MIC1_MUTE_LED, ALC269_FIXUP_MIC2_MUTE_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, @@ -6171,6 +6195,10 @@ static const struct alc_fixup alc269_fixups[] = { { } }, }, + [ALC269_FIXUP_MIC1_MUTE_LED] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_mic1_mute, + }, [ALC269_FIXUP_MIC2_MUTE_LED] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_mic2_mute, @@ -6215,6 +6243,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), -- 1.7.9.5

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