From 3f343f8512c7882a3637d9aea4ec6b3801cbcdc5 Mon Sep 17 00:00:00 2001 From: Dmitry Artamonow Date: Wed, 8 Dec 2010 23:36:17 +0300 Subject: [PATCH 01/11] ASoC: fix deemphasis control in wm8904/55/60 codecs Deemphasis control's .get callback should update control's value instead of returning it - return value of callback function is used for indicating error or success of operation. Signed-off-by: Dmitry Artamonow Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8904.c | 3 ++- sound/soc/codecs/wm8955.c | 3 ++- sound/soc/codecs/wm8960.c | 3 ++- 3 files changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index fca60a0b57b8..9001cc48ba13 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -818,7 +818,8 @@ static int wm8904_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - return wm8904->deemph; + ucontrol->value.enumerated.item[0] = wm8904->deemph; + return 0; } static int wm8904_put_deemph(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index f89ad6c9a80b..9cbab8e1de01 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -380,7 +380,8 @@ static int wm8955_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8955_priv *wm8955 = snd_soc_codec_get_drvdata(codec); - return wm8955->deemph; + ucontrol->value.enumerated.item[0] = wm8955->deemph; + return 0; } static int wm8955_put_deemph(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 8d5efb333c33..21986c42272f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -138,7 +138,8 @@ static int wm8960_get_deemph(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - return wm8960->deemph; + ucontrol->value.enumerated.item[0] = wm8960->deemph; + return 0; } static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, From a0968628097380be52db8b4664da98fc425546a5 Mon Sep 17 00:00:00 2001 From: Seungwhan Youn Date: Thu, 9 Dec 2010 18:07:52 +0900 Subject: [PATCH 02/11] ASoC: WM8580: Fix R8 initial value Acc to WM8580 manual, the default value for R8 is 0x10, not 0x1c. Signed-off-by: Seungwhan Youn Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8580.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 879dff2714dd..8725d4e75431 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -161,7 +161,7 @@ static const u16 wm8580_reg[] = { 0x0121, 0x017e, 0x007d, 0x0014, /*R3*/ 0x0121, 0x017e, 0x007d, 0x0194, /*R7*/ - 0x001c, 0x0002, 0x0002, 0x00c2, /*R11*/ + 0x0010, 0x0002, 0x0002, 0x00c2, /*R11*/ 0x0182, 0x0082, 0x000a, 0x0024, /*R15*/ 0x0009, 0x0000, 0x00ff, 0x0000, /*R19*/ 0x00ff, 0x00ff, 0x00ff, 0x00ff, /*R23*/ From 862af8adbe6b9ccb7c00c13717b1f92465f79aa2 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 10 Dec 2010 20:53:55 +0200 Subject: [PATCH 03/11] ASoC: Fix bias power down of non-DAPM codec Currently bias of non-DAPM codec will be powered down (standby/off) whenever there is a stream stop. This is wrong in simultaneous playback/capture since the bias is put down immediately after stopping the first stream. Fix this by using the codec->active count when figuring out the needed bias level after stream stop. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 75ed6491222d..c721502833bc 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -944,6 +944,9 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event) case SND_SOC_DAPM_STREAM_RESUME: sys_power = 1; break; + case SND_SOC_DAPM_STREAM_STOP: + sys_power = !!codec->active; + break; case SND_SOC_DAPM_STREAM_SUSPEND: sys_power = 0; break; From 87a1c8aaa0bced8acf4cd64672362492460c31ae Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Tue, 21 Dec 2010 00:03:17 +0100 Subject: [PATCH 04/11] ALSA: pcm: remember to always call va_end() on stuff that we va_start() The Coverity checker spotted that we do not always remember to call va_end() on 'args' in failure paths in snd_pcm_hw_rule_add(). Here's a patch to fix that up (compile tested only) - it also removes some annoying trailing whitespace that caught my eye while I was in the area.. Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index b75db8e9cc0f..11446a1506da 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1070,8 +1070,10 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, struct snd_pcm_hw_rule *new; unsigned int new_rules = constrs->rules_all + 16; new = kcalloc(new_rules, sizeof(*c), GFP_KERNEL); - if (!new) + if (!new) { + va_end(args); return -ENOMEM; + } if (constrs->rules) { memcpy(new, constrs->rules, constrs->rules_num * sizeof(*c)); @@ -1087,8 +1089,10 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, c->private = private; k = 0; while (1) { - if (snd_BUG_ON(k >= ARRAY_SIZE(c->deps))) + if (snd_BUG_ON(k >= ARRAY_SIZE(c->deps))) { + va_end(args); return -EINVAL; + } c->deps[k++] = dep; if (dep < 0) break; @@ -1097,7 +1101,7 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, constrs->rules_num++; va_end(args); return 0; -} +} EXPORT_SYMBOL(snd_pcm_hw_rule_add); From 2785591a9760c677a7ee6f541e751c23086f5bfd Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 21 Dec 2010 09:09:53 +0100 Subject: [PATCH 05/11] ALSA: hda - Add fix-up for Sony VAIO with ALC275 codecs Set GPIO2 for some Sony VAIO with ALC275 to fix speaker output. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dd56d8833ad2..c9af538323ea 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14806,6 +14806,7 @@ static int alc269_resume(struct hda_codec *codec) enum { ALC269_FIXUP_SONY_VAIO, + ALC275_FIX_SONY_VAIO_GPIO2, ALC269_FIXUP_DELL_M101Z, ALC269_FIXUP_SKU_IGNORE, ALC269_FIXUP_ASUS_G73JW, @@ -14818,6 +14819,14 @@ static const struct alc_fixup alc269_fixups[] = { {} } }, + [ALC275_FIX_SONY_VAIO_GPIO2] = { + .verbs = (const struct hda_verb[]) { + {0x01, AC_VERB_SET_GPIO_MASK, 0x04}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, + { } + } + }, [ALC269_FIXUP_DELL_M101Z] = { .verbs = (const struct hda_verb[]) { /* Enables internal speaker */ @@ -14839,6 +14848,9 @@ static const struct alc_fixup alc269_fixups[] = { static struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), + SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIX_SONY_VAIO_GPIO2), + SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIX_SONY_VAIO_GPIO2), + SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIX_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), From c793bec550c68a1da1034090b43a886e8fee5eb0 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Tue, 21 Dec 2010 09:14:13 +0100 Subject: [PATCH 06/11] ALSA: hda - Don't apply ALC269-specific initialization to ALC275 ALC275 doesn't require the ALC269 (and its variants) specific init sequences. Add the check of codec id. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 45 ++++++++++++++++++----------------- 1 file changed, 23 insertions(+), 22 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c9af538323ea..69aa62eb9548 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -15104,28 +15104,29 @@ static int patch_alc269(struct hda_codec *codec) alc_auto_parse_customize_define(codec); - coef = alc_read_coef_idx(codec, 0); - if ((coef & 0x00f0) == 0x0010) { - if (codec->bus->pci->subsystem_vendor == 0x1025 && - spec->cdefine.platform_type == 1) { - alc_codec_rename(codec, "ALC271X"); - spec->codec_variant = ALC269_TYPE_ALC271X; - } else if ((coef & 0xf000) == 0x1000) { - spec->codec_variant = ALC269_TYPE_ALC270; - } else if ((coef & 0xf000) == 0x2000) { - alc_codec_rename(codec, "ALC259"); - spec->codec_variant = ALC269_TYPE_ALC259; - } else if ((coef & 0xf000) == 0x3000) { - alc_codec_rename(codec, "ALC258"); - spec->codec_variant = ALC269_TYPE_ALC258; - } else { - alc_codec_rename(codec, "ALC269VB"); - spec->codec_variant = ALC269_TYPE_ALC269VB; - } - } else - alc_fix_pll_init(codec, 0x20, 0x04, 15); - - alc269_fill_coef(codec); + if (codec->vendor_id == 0x10ec0269) { + coef = alc_read_coef_idx(codec, 0); + if ((coef & 0x00f0) == 0x0010) { + if (codec->bus->pci->subsystem_vendor == 0x1025 && + spec->cdefine.platform_type == 1) { + alc_codec_rename(codec, "ALC271X"); + spec->codec_variant = ALC269_TYPE_ALC271X; + } else if ((coef & 0xf000) == 0x1000) { + spec->codec_variant = ALC269_TYPE_ALC270; + } else if ((coef & 0xf000) == 0x2000) { + alc_codec_rename(codec, "ALC259"); + spec->codec_variant = ALC269_TYPE_ALC259; + } else if ((coef & 0xf000) == 0x3000) { + alc_codec_rename(codec, "ALC258"); + spec->codec_variant = ALC269_TYPE_ALC258; + } else { + alc_codec_rename(codec, "ALC269VB"); + spec->codec_variant = ALC269_TYPE_ALC269VB; + } + } else + alc_fix_pll_init(codec, 0x20, 0x04, 15); + alc269_fill_coef(codec); + } board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST, alc269_models, From 2d7ec12b902ae00920cee50d98757376b2fa9467 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 23 Dec 2010 10:16:05 +0100 Subject: [PATCH 07/11] ALSA: hda - Fix conflict of d-mic capture volume controls When the d-mics are assigned to the same purpose of another analog mic pins, the driver doesn't compute the index properly, resulting in an error with "existing control". This patch fixes it. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index efa4225f5fd6..f03b2ff90496 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3481,6 +3481,8 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, label = hda_get_input_pin_label(codec, nid, 1); snd_hda_add_imux_item(dimux, label, index, &type_idx); + if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) + snd_hda_add_imux_item(imux, label, index, &type_idx); err = create_elem_capture_vol(codec, nid, label, type_idx, HDA_INPUT); @@ -3492,9 +3494,6 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, if (err < 0) return err; } - - if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) - snd_hda_add_imux_item(imux, label, index, NULL); } return 0; From 1afe206ab6998ecd5f5485e02006b0578720a691 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 23 Dec 2010 10:17:52 +0100 Subject: [PATCH 08/11] ALSA: hda - Try to find an empty control index when it's occupied When a mixer control element was already created with the given name, try to find another index for avoiding conflicts, instead of breaking with an error. This makes the driver more robust. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 57 +++++++++++++++++++++++---------------- 1 file changed, 34 insertions(+), 23 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 644e3f14f8ca..98b6d02a36c9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1919,6 +1919,16 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_find_mixer_ctl); +static int find_empty_mixer_ctl_idx(struct hda_codec *codec, const char *name) +{ + int idx; + for (idx = 0; idx < 16; idx++) { /* 16 ctlrs should be large enough */ + if (!_snd_hda_find_mixer_ctl(codec, name, idx)) + return idx; + } + return -EBUSY; +} + /** * snd_hda_ctl_add - Add a control element and assign to the codec * @codec: HD-audio codec @@ -2654,8 +2664,6 @@ static struct snd_kcontrol_new dig_mixes[] = { { } /* end */ }; -#define SPDIF_MAX_IDX 4 /* 4 instances should be enough to probe */ - /** * snd_hda_create_spdif_out_ctls - create Output SPDIF-related controls * @codec: the HDA codec @@ -2673,12 +2681,8 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) struct snd_kcontrol_new *dig_mix; int idx; - for (idx = 0; idx < SPDIF_MAX_IDX; idx++) { - if (!_snd_hda_find_mixer_ctl(codec, "IEC958 Playback Switch", - idx)) - break; - } - if (idx >= SPDIF_MAX_IDX) { + idx = find_empty_mixer_ctl_idx(codec, "IEC958 Playback Switch"); + if (idx < 0) { printk(KERN_ERR "hda_codec: too many IEC958 outputs\n"); return -EBUSY; } @@ -2829,12 +2833,8 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) struct snd_kcontrol_new *dig_mix; int idx; - for (idx = 0; idx < SPDIF_MAX_IDX; idx++) { - if (!_snd_hda_find_mixer_ctl(codec, "IEC958 Capture Switch", - idx)) - break; - } - if (idx >= SPDIF_MAX_IDX) { + idx = find_empty_mixer_ctl_idx(codec, "IEC958 Capture Switch"); + if (idx < 0) { printk(KERN_ERR "hda_codec: too many IEC958 inputs\n"); return -EBUSY; } @@ -3808,21 +3808,32 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) for (; knew->name; knew++) { struct snd_kcontrol *kctl; + int addr = 0, idx = 0; if (knew->iface == -1) /* skip this codec private value */ continue; - kctl = snd_ctl_new1(knew, codec); - if (!kctl) - return -ENOMEM; - err = snd_hda_ctl_add(codec, 0, kctl); - if (err < 0) { - if (!codec->addr) - return err; + for (;;) { kctl = snd_ctl_new1(knew, codec); if (!kctl) return -ENOMEM; - kctl->id.device = codec->addr; + if (addr > 0) + kctl->id.device = addr; + if (idx > 0) + kctl->id.index = idx; err = snd_hda_ctl_add(codec, 0, kctl); - if (err < 0) + if (!err) + break; + /* try first with another device index corresponding to + * the codec addr; if it still fails (or it's the + * primary codec), then try another control index + */ + if (!addr && codec->addr) + addr = codec->addr; + else if (!idx && !knew->index) { + idx = find_empty_mixer_ctl_idx(codec, + knew->name); + if (idx <= 0) + return err; + } else return err; } } From 7039c74cb54652ba6d726ad4d2a42dbac95a97be Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 23 Dec 2010 16:35:34 +0100 Subject: [PATCH 09/11] ALSA: hda - Fix GPIO2-fixup for Sony laptops The fix-up entries by the commit 2785591a9760c677a7ee6f541e751c23086f5bfd ALSA: hda - Add fix-up for Sony VAIO with ALC275 codecs weren't applied in the right position. They had to be before the quirk entry matching to all Sony devices. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 69aa62eb9548..552a09e9211f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14847,10 +14847,10 @@ static const struct alc_fixup alc269_fixups[] = { }; static struct snd_pci_quirk alc269_fixup_tbl[] = { - SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIX_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIX_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIX_SONY_VAIO_GPIO2), + SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), From d81a12bc29ae4038770e05dce4ab7f26fd5880fb Mon Sep 17 00:00:00 2001 From: Dan Rosenberg Date: Sat, 25 Dec 2010 16:23:40 -0500 Subject: [PATCH 10/11] sound: Prevent buffer overflow in OSS load_mixer_volumes The load_mixer_volumes() function, which can be triggered by unprivileged users via the SOUND_MIXER_SETLEVELS ioctl, is vulnerable to a buffer overflow. Because the provided "name" argument isn't guaranteed to be NULL terminated at the expected 32 bytes, it's possible to overflow past the end of the last element in the mixer_vols array. Further exploitation can result in an arbitrary kernel write (via subsequent calls to load_mixer_volumes()) leading to privilege escalation, or arbitrary kernel reads via get_mixer_levels(). In addition, the strcmp() may leak bytes beyond the mixer_vols array. Signed-off-by: Dan Rosenberg Cc: stable Signed-off-by: Takashi Iwai --- sound/oss/soundcard.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index 46c0d03dbecc..fcb14a099822 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -87,7 +87,7 @@ int *load_mixer_volumes(char *name, int *levels, int present) int i, n; for (i = 0; i < num_mixer_volumes; i++) { - if (strcmp(name, mixer_vols[i].name) == 0) { + if (strncmp(name, mixer_vols[i].name, 32) == 0) { if (present) mixer_vols[i].num = i; return mixer_vols[i].levels; @@ -99,7 +99,7 @@ int *load_mixer_volumes(char *name, int *levels, int present) } n = num_mixer_volumes++; - strcpy(mixer_vols[n].name, name); + strncpy(mixer_vols[n].name, name, 32); if (present) mixer_vols[n].num = n; From e03fa055bc126e536c7f65862e08a9b143138ea9 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Tue, 28 Dec 2010 17:20:02 -0500 Subject: [PATCH 11/11] ALSA: hda: Use LPIB quirk for Dell Inspiron m101z/1120 Sjoerd Simons reports that, without using position_fix=1, recording experiences overruns. Work around that by applying the LPIB quirk for his hardware. Reported-and-tested-by: Sjoerd Simons Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b030c8eba21f..a1c4008af891 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2300,6 +2300,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1028, 0x0470, "Dell Inspiron 1120", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),