From 828fa8ce5a8d75169f16740c28c8a1b7c13dd96b Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Wed, 15 Apr 2015 13:29:05 +0200
Subject: [PATCH 01/13] ALSA: hda - simplify azx_has_pm_runtime
Because AZX_DCAPS_PM_RUNTIME is always defined as non-zero, the
initial part of the expression can be skipped.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_controller.h | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index be1b7ded8d82..0efdb094d21c 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -404,7 +404,7 @@ struct azx {
((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg))
#define azx_has_pm_runtime(chip) \
- (!AZX_DCAPS_PM_RUNTIME || ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME))
+ ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME)
/* PCM setup */
static inline struct azx_dev *get_azx_dev(struct snd_pcm_substream *substream)
From 3047755588e71b67c3f60409686fabf8506357e9 Mon Sep 17 00:00:00 2001
From: Scott Wood <scottwood@freescale.com>
Date: Wed, 15 Apr 2015 18:16:47 -0500
Subject: [PATCH 02/13] ALSA: intel8x0: Check pci_iomap() success for
DEVICE_ALI
DEVICE_ALI previously would jump to port_inited after calling
pci_iomap(), bypassing the check for bmaddr being NULL.
Signed-off-by: Scott Wood <scottwood@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/intel8x0.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 749069aa6997..b120925223ae 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -3101,13 +3101,13 @@ static int snd_intel8x0_create(struct snd_card *card,
chip->bmaddr = pci_iomap(pci, 3, 0);
else
chip->bmaddr = pci_iomap(pci, 1, 0);
+
+ port_inited:
if (!chip->bmaddr) {
dev_err(card->dev, "Controller space ioremap problem\n");
snd_intel8x0_free(chip);
return -EIO;
}
-
- port_inited:
chip->bdbars_count = bdbars[device_type];
/* initialize offsets */
From 7d4b5e978ad350916b5c3995490b09c4e59cec4a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 16 Apr 2015 15:14:53 +0200
Subject: [PATCH 03/13] ALSA: hda - Fix regression for slave SPDIF setups
The commit [a551d91473e5: ALSA: hda - Use regmap for command verb
caches, too] introduced a regression due to a typo in the conversion;
the IEC958 status bits of slave digital devices aren't updated
correctly. This patch corrects it.
Fixes: a551d91473e5 ('ALSA: hda - Use regmap for command verb caches, too')
Reported-and-tested-by: Markus Trippelsdorf <markus@trippelsdorf.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_codec.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index e70a7fb393dd..873ed1bce12b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2529,7 +2529,7 @@ static void set_dig_out(struct hda_codec *codec, hda_nid_t nid,
if (!d)
return;
for (; *d; d++)
- snd_hdac_regmap_update(&codec->core, nid,
+ snd_hdac_regmap_update(&codec->core, *d,
AC_VERB_SET_DIGI_CONVERT_1, mask, val);
}
From f4d770317997f89bb6997ee3e8dd495cb8356ae9 Mon Sep 17 00:00:00 2001
From: Dan Carpenter <dan.carpenter@oracle.com>
Date: Fri, 17 Apr 2015 16:19:46 +0300
Subject: [PATCH 04/13] ALSA: hda - potential (but unlikely) uninitialized
variable
This function is a bit unusual because it accepts negative values as
"conn_len". It's theoretically possible for both "cache_len" and
"conn_len" to be -ENOSPC and in that case we would oops trying to run
memcmp() on the uninitialized "list" pointer.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_proc.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index ee6230767c64..baaf7ed06875 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -582,8 +582,8 @@ static void print_conn_list(struct snd_info_buffer *buffer,
/* Get Cache connections info */
cache_len = snd_hda_get_conn_list(codec, nid, &list);
- if (cache_len != conn_len
- || memcmp(list, conn, conn_len)) {
+ if (cache_len >= 0 && (cache_len != conn_len ||
+ memcmp(list, conn, conn_len) != 0)) {
snd_iprintf(buffer, " In-driver Connection: %d\n", cache_len);
if (cache_len > 0) {
snd_iprintf(buffer, " ");
From 43c499dc2778b96e21ed14e1a71e7b117a0b406f Mon Sep 17 00:00:00 2001
From: Vinod Koul <vinod.koul@intel.com>
Date: Fri, 17 Apr 2015 22:53:32 +0530
Subject: [PATCH 05/13] ALSA: asound.h - use SNDRV_CTL_ELEM_ID_NAME_MAXLEN
we have defined SNDRV_CTL_ELEM_ID_NAME_MAXLEN as size of name array so use
this define instead of numeric value
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
include/uapi/sound/asound.h | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index 46145a5277fe..a45be6bdcf5b 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -864,7 +864,7 @@ struct snd_ctl_elem_id {
snd_ctl_elem_iface_t iface; /* interface identifier */
unsigned int device; /* device/client number */
unsigned int subdevice; /* subdevice (substream) number */
- unsigned char name[44]; /* ASCII name of item */
+ unsigned char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* ASCII name of item */
unsigned int index; /* index of item */
};
From bc26d4d06e337ade069f33d3f4377593b24e6e36 Mon Sep 17 00:00:00 2001
From: Alexey Khoroshilov <khoroshilov@ispras.ru>
Date: Sat, 18 Apr 2015 02:53:25 +0300
Subject: [PATCH 06/13] sound/oss: fix deadlock in
sequencer_ioctl(SNDCTL_SEQ_OUTOFBAND)
A deadlock can be initiated by userspace via ioctl(SNDCTL_SEQ_OUTOFBAND)
on /dev/sequencer with TMR_ECHO midi event.
In this case the control flow is:
sound_ioctl()
-> case SND_DEV_SEQ:
case SND_DEV_SEQ2:
sequencer_ioctl()
-> case SNDCTL_SEQ_OUTOFBAND:
spin_lock_irqsave(&lock,flags);
play_event();
-> case EV_TIMING:
seq_timing_event()
-> case TMR_ECHO:
seq_copy_to_input()
-> spin_lock_irqsave(&lock,flags);
It seems that spin_lock_irqsave() around play_event() is not necessary,
because the only other call location in seq_startplay() makes the call
without acquiring spinlock.
So, the patch just removes spinlocks around play_event().
By the way, it removes unreachable code in seq_timing_event(),
since (seq_mode == SEQ_2) case is handled in the beginning.
Compile tested only.
Found by Linux Driver Verification project (linuxtesting.org).
Signed-off-by: Alexey Khoroshilov <khoroshilov@ispras.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/oss/sequencer.c | 12 ++----------
1 file changed, 2 insertions(+), 10 deletions(-)
diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c
index c0eea1dfe90f..f19da4b47c1d 100644
--- a/sound/oss/sequencer.c
+++ b/sound/oss/sequencer.c
@@ -681,13 +681,8 @@ static int seq_timing_event(unsigned char *event_rec)
break;
case TMR_ECHO:
- if (seq_mode == SEQ_2)
- seq_copy_to_input(event_rec, 8);
- else
- {
- parm = (parm << 8 | SEQ_ECHO);
- seq_copy_to_input((unsigned char *) &parm, 4);
- }
+ parm = (parm << 8 | SEQ_ECHO);
+ seq_copy_to_input((unsigned char *) &parm, 4);
break;
default:;
@@ -1324,7 +1319,6 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a
int mode = translate_mode(file);
struct synth_info inf;
struct seq_event_rec event_rec;
- unsigned long flags;
int __user *p = arg;
orig_dev = dev = dev >> 4;
@@ -1479,9 +1473,7 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a
case SNDCTL_SEQ_OUTOFBAND:
if (copy_from_user(&event_rec, arg, sizeof(event_rec)))
return -EFAULT;
- spin_lock_irqsave(&lock,flags);
play_event(event_rec.arr);
- spin_unlock_irqrestore(&lock,flags);
return 0;
case SNDCTL_MIDI_INFO:
From 9476d369d7b39348945c297da5f2935904229813 Mon Sep 17 00:00:00 2001
From: Gabriele Mazzotta <gabriele.mzt@gmail.com>
Date: Sun, 19 Apr 2015 19:00:40 +0200
Subject: [PATCH 07/13] ALSA: hda - Mute headphone pin on suspend on XPS13 9333
Muting the headphone output pin right before the codec suspension
prevents pop noises when headphones are plugged in (except for a
barely audible click noise).
This solution allows to truly save some power when headphones are
plugged in unlike the previous solution (033b0a7ca9c:
"ALSA: hda - Pop noises fix for XPS13 9333")
Signed-off-by: Gabriele Mazzotta <gabriele.mzt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 17 +++++++----------
1 file changed, 7 insertions(+), 10 deletions(-)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b18b9c67b262..231d0e4b9a95 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4176,17 +4176,15 @@ static void alc_fixup_disable_aamix(struct hda_codec *codec,
}
}
-static unsigned int alc_power_filter_xps13(struct hda_codec *codec,
- hda_nid_t nid,
- unsigned int power_state)
+static void alc_shutup_dell_xps13(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
+ int hp_pin = spec->gen.autocfg.hp_pins[0];
- /* Avoid pop noises when headphones are plugged in */
- if (spec->gen.hp_jack_present)
- if (nid == codec->core.afg || nid == 0x02 || nid == 0x15)
- return AC_PWRST_D0;
- return snd_hda_gen_path_power_filter(codec, nid, power_state);
+ /* Prevent pop noises when headphones are plugged in */
+ snd_hda_codec_write(codec, hp_pin, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+ msleep(20);
}
static void alc_fixup_dell_xps13(struct hda_codec *codec,
@@ -4197,8 +4195,7 @@ static void alc_fixup_dell_xps13(struct hda_codec *codec,
struct hda_input_mux *imux = &spec->gen.input_mux;
int i;
- spec->shutup = alc_no_shutup;
- codec->power_filter = alc_power_filter_xps13;
+ spec->shutup = alc_shutup_dell_xps13;
/* Make the internal mic the default input source. */
for (i = 0; i < imux->num_items; i++) {
From f4c1a311d8dc55c90c39e9cf7b003254a769574d Mon Sep 17 00:00:00 2001
From: Mengdong Lin <mengdong.lin@intel.com>
Date: Mon, 20 Apr 2015 17:33:57 +0800
Subject: [PATCH 08/13] ALSA: hda - only sync BCLK to the display clock for
Haswell & Broadwell
Only Intel Haswell and Broadwell have a separate HD-A controller (PCI device 3)
for display audio, which needs to get 24MHz HD-A link BCLK from the variable
display core clock through vendor specific registers EM4 & EM5. Other platforms
(Baytrail, Braswell and Skylake) don't have this feature.
So this patch checks the PCI device ID of the controller in haswell_set_bclk()
and only sync BCLK for HSW and BDW.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_i915.c | 6 ++++++
1 file changed, 6 insertions(+)
diff --git a/sound/pci/hda/hda_i915.c b/sound/pci/hda/hda_i915.c
index 52a85d87c23c..3052a2b095f7 100644
--- a/sound/pci/hda/hda_i915.c
+++ b/sound/pci/hda/hda_i915.c
@@ -55,6 +55,12 @@ void haswell_set_bclk(struct hda_intel *hda)
int cdclk_freq;
unsigned int bclk_m, bclk_n;
struct i915_audio_component *acomp = &hda->audio_component;
+ struct pci_dev *pci = hda->chip.pci;
+
+ /* Only Haswell/Broadwell need set BCLK */
+ if (pci->device != 0x0a0c && pci->device != 0x0c0c
+ && pci->device != 0x0d0c && pci->device != 0x160c)
+ return;
if (!acomp->ops)
return;
From 40cc2392f4b144197d05eec73c1560f42fc25def Mon Sep 17 00:00:00 2001
From: Mengdong Lin <mengdong.lin@intel.com>
Date: Tue, 21 Apr 2015 13:12:23 +0800
Subject: [PATCH 09/13] ALSA: hda - add AZX_DCAPS_I915_POWERWELL to Baytrail
This patch addes AZX_DCAPS_I915_POWERWELL to BYT (Baytrail).
Like Braswell and Skylake, the HDMI codec on Bytrail is also in the shared
power well with GPU. This power well must be turned on before we reset link
to probe the codec, to avoid communication failure with the codec.
The side effect is that this power is always ON in S0 because the BYT HDMI
codec does not support EPSS or D3ClkStop and so the controller doesn't enter
D3 at runtime, and the HDMI codec and analog codec share a single physical
HD-A link and so we cannot reset the HD-A link freely when we re-enable the
power to use the HDMI codec.
Next step is to test if an AGP reset or double AGP reset on BYT HDMI codec is
okay to bring the HDMI codec back to a functional state after restoring the
power. If okay, we can bind the power on/off with the HDMI codec PM without
interrupting the analog audio.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/hda_intel.c | 5 ++++-
1 file changed, 4 insertions(+), 1 deletion(-)
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index e1c210515581..34040d26c94f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -297,6 +297,9 @@ enum {
AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\
AZX_DCAPS_SNOOP_TYPE(SCH))
+#define AZX_DCAPS_INTEL_BAYTRAIL \
+ (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_I915_POWERWELL)
+
#define AZX_DCAPS_INTEL_BRASWELL \
(AZX_DCAPS_INTEL_PCH | AZX_DCAPS_I915_POWERWELL)
@@ -1992,7 +1995,7 @@ static const struct pci_device_id azx_ids[] = {
.driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM },
/* BayTrail */
{ PCI_DEVICE(0x8086, 0x0f04),
- .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BAYTRAIL },
/* Braswell */
{ PCI_DEVICE(0x8086, 0x2284),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BRASWELL },
From 6d1f2f605601ec701b561eca143c03e2a22d6489 Mon Sep 17 00:00:00 2001
From: Takamichi Horikawa <takamichiho@gmail.com>
Date: Tue, 21 Apr 2015 11:23:57 +0900
Subject: [PATCH 10/13] ALSA: usb-audio: Fix audio output on Roland SC-D70
sound module
Roland SC-D70 reports its device class as vendor specific class and
the quirk QUIRK_AUDIO_FIXED_ENDPOINT was used for audio output.
In the quirks table the sampling rate was hard-coded to 44100 Hz
and therefore not worked when the sound module was in 48000 Hz mode.
In this change the quirk is changed to QUIRK_AUDIO_STANDARD_INTERFACE
but as the sound module reports incorrect bSubframeSize in its
descriptors, additional change is made in format.c to detect it and
to override it (which uses the existing code for Edirol SD-90).
Tested both when the sound module was in 44100 Hz mode and 48000 Hz
mode and both audio input and output. MIDI related part of the driver
is not touched.
Signed-off-by: Takamichi Horikawa <takamichiho@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/usb/format.c | 5 ++++-
sound/usb/quirks-table.h | 30 ++----------------------------
2 files changed, 6 insertions(+), 29 deletions(-)
diff --git a/sound/usb/format.c b/sound/usb/format.c
index 8bcc87cf5667..789d19ec035d 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -79,7 +79,10 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip,
format = 1 << UAC_FORMAT_TYPE_I_PCM;
}
if (format & (1 << UAC_FORMAT_TYPE_I_PCM)) {
- if (chip->usb_id == USB_ID(0x0582, 0x0016) /* Edirol SD-90 */ &&
+ if (((chip->usb_id == USB_ID(0x0582, 0x0016)) ||
+ /* Edirol SD-90 */
+ (chip->usb_id == USB_ID(0x0582, 0x000c))) &&
+ /* Roland SC-D70 */
sample_width == 24 && sample_bytes == 2)
sample_bytes = 3;
else if (sample_width > sample_bytes * 8) {
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 07f984d5f516..2f6d3e9a1bcd 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -816,37 +816,11 @@ YAMAHA_DEVICE(0x7010, "UB99"),
.data = (const struct snd_usb_audio_quirk[]) {
{
.ifnum = 0,
- .type = QUIRK_AUDIO_FIXED_ENDPOINT,
- .data = & (const struct audioformat) {
- .formats = SNDRV_PCM_FMTBIT_S24_3LE,
- .channels = 2,
- .iface = 0,
- .altsetting = 1,
- .altset_idx = 1,
- .attributes = 0,
- .endpoint = 0x01,
- .ep_attr = 0x01,
- .rates = SNDRV_PCM_RATE_CONTINUOUS,
- .rate_min = 44100,
- .rate_max = 44100,
- }
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
},
{
.ifnum = 1,
- .type = QUIRK_AUDIO_FIXED_ENDPOINT,
- .data = & (const struct audioformat) {
- .formats = SNDRV_PCM_FMTBIT_S24_3LE,
- .channels = 2,
- .iface = 1,
- .altsetting = 1,
- .altset_idx = 1,
- .attributes = 0,
- .endpoint = 0x81,
- .ep_attr = 0x01,
- .rates = SNDRV_PCM_RATE_CONTINUOUS,
- .rate_min = 44100,
- .rate_max = 44100,
- }
+ .type = QUIRK_AUDIO_STANDARD_INTERFACE
},
{
.ifnum = 2,
From 7d1b6e29327428993ba568bdd8c66734070f45e0 Mon Sep 17 00:00:00 2001
From: David Henningsson <david.henningsson@canonical.com>
Date: Tue, 21 Apr 2015 10:48:46 +0200
Subject: [PATCH 11/13] ALSA: hda - fix "num_steps = 0" error on ALC256
The ALC256 does not have a mixer nid at 0x0b, and there's no
loopback path (the output pins are directly connected to the DACs).
This commit fixes an "num_steps = 0 for NID=0xb (ctl = Beep Playback Volume)"
error (and as a result, problems with amixer/alsamixer).
If there's pcbeep functionality, it certainly isn't controlled by setting an
amp on 0x0b, so disable beep functionality (at least for now).
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1446517
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 5 +++--
1 file changed, 3 insertions(+), 2 deletions(-)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 231d0e4b9a95..03975d03b264 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5664,6 +5664,7 @@ static int patch_alc269(struct hda_codec *codec)
break;
case 0x10ec0256:
spec->codec_variant = ALC269_TYPE_ALC256;
+ spec->gen.mixer_nid = 0; /* ALC256 does not have any loopback mixer path */
break;
}
@@ -5677,8 +5678,8 @@ static int patch_alc269(struct hda_codec *codec)
if (err < 0)
goto error;
- if (!spec->gen.no_analog && spec->gen.beep_nid)
- set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
+ if (!spec->gen.no_analog && spec->gen.beep_nid && spec->gen.mixer_nid)
+ set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT);
codec->patch_ops = alc_patch_ops;
codec->patch_ops.stream_pm = snd_hda_gen_stream_pm;
From d32b66668c702aed0e330dc5ca186afbadcdacf8 Mon Sep 17 00:00:00 2001
From: Kailang Yang <kailang@realtek.com>
Date: Thu, 23 Apr 2015 15:10:53 +0800
Subject: [PATCH 12/13] ALSA: hda/realtek - Fix Headphone Mic doesn't recording
for ALC256
Switch default pcbeep path to Line in path.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Tested-by: Hui Wang <hui.wang@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
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 03975d03b264..4b10cde12831 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5665,6 +5665,7 @@ static int patch_alc269(struct hda_codec *codec)
case 0x10ec0256:
spec->codec_variant = ALC269_TYPE_ALC256;
spec->gen.mixer_nid = 0; /* ALC256 does not have any loopback mixer path */
+ alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/
break;
}
From e8191a8e475551b277d85cd76c3f0f52fdf42e86 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Fri, 24 Apr 2015 13:39:59 +0800
Subject: [PATCH 13/13] ALSA: hda - fix headset mic detection problem for one
more machine
We have two machines with alc256 codec in the pin quirk table, so
moving the common pins to ALC256_STANDARD_PINS.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1447909
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
sound/pci/hda/patch_realtek.c | 24 +++++++++++++++---------
1 file changed, 15 insertions(+), 9 deletions(-)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 4b10cde12831..06199e4e930f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5228,6 +5228,16 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
{0x1b, 0x411111f0}, \
{0x1e, 0x411111f0}
+#define ALC256_STANDARD_PINS \
+ {0x12, 0x90a60140}, \
+ {0x14, 0x90170110}, \
+ {0x19, 0x411111f0}, \
+ {0x1a, 0x411111f0}, \
+ {0x1b, 0x411111f0}, \
+ {0x1d, 0x40700001}, \
+ {0x1e, 0x411111f0}, \
+ {0x21, 0x02211020}
+
#define ALC282_STANDARD_PINS \
{0x14, 0x90170110}, \
{0x18, 0x411111f0}, \
@@ -5328,15 +5338,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
{0x1d, 0x40700001},
{0x21, 0x02211050}),
SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
- {0x12, 0x90a60140},
- {0x13, 0x40000000},
- {0x14, 0x90170110},
- {0x19, 0x411111f0},
- {0x1a, 0x411111f0},
- {0x1b, 0x411111f0},
- {0x1d, 0x40700001},
- {0x1e, 0x411111f0},
- {0x21, 0x02211020}),
+ ALC256_STANDARD_PINS,
+ {0x13, 0x40000000}),
+ SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
+ ALC256_STANDARD_PINS,
+ {0x13, 0x411111f0}),
SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
{0x12, 0x90a60130},
{0x13, 0x40000000},