From f3b827e0b1841f4cfc18436e09f4f269f3be908e Mon Sep 17 00:00:00 2001
From: Bhumika Goyal <bhumirks@gmail.com>
Date: Mon, 20 Feb 2017 00:18:09 +0530
Subject: ALSA: pci: constify snd_kcontrol_new structures

Declare snd_kcontrol_new structures as const as they are only passed as
an argument to the function snd_ctl_new1. This argument is of type
const, so snd_kcontrol_new structures having the same property can be
made const too.
Done using Coccinelle:

@r1 disable optional_qualifier @
identifier i;
position p;
@@
static struct snd_kcontrol_new i@p = {...};

@ok1@
identifier r1.i;
position p;
expression e1;
@@
snd_ctl_new1(&i@p,e1)

@bad@
position p!={r1.p,ok1.p};
identifier r1.i;
@@
i@p

@depends on !bad disable optional_qualifier@
identifier r1.i;
@@
+const
struct snd_kcontrol_new i;

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/au88x0/au88x0_a3d.c    |  2 +-
 sound/pci/au88x0/au88x0_eq.c     |  6 +++---
 sound/pci/au88x0/au88x0_pcm.c    |  2 +-
 sound/pci/aw2/aw2-alsa.c         |  2 +-
 sound/pci/bt87x.c                |  6 +++---
 sound/pci/ca0106/ca0106_mixer.c  |  4 ++--
 sound/pci/cmipci.c               |  6 +++---
 sound/pci/cs4281.c               |  4 ++--
 sound/pci/echoaudio/echoaudio.c  | 26 +++++++++++++-------------
 sound/pci/emu10k1/emu10k1x.c     |  6 +++---
 sound/pci/emu10k1/emumixer.c     | 30 +++++++++++++++---------------
 sound/pci/emu10k1/emupcm.c       |  2 +-
 sound/pci/ens1370.c              |  4 ++--
 sound/pci/hda/hda_codec.c        |  4 ++--
 sound/pci/hda/patch_hdmi.c       |  2 +-
 sound/pci/ice1712/delta.c        |  2 +-
 sound/pci/ice1712/ews.c          |  4 ++--
 sound/pci/ice1712/ice1712.c      | 30 +++++++++++++++---------------
 sound/pci/ice1712/ice1724.c      | 20 ++++++++++----------
 sound/pci/lola/lola_mixer.c      |  2 +-
 sound/pci/lx6464es/lx6464es.c    |  2 +-
 sound/pci/mixart/mixart_mixer.c  |  6 +++---
 sound/pci/pcxhr/pcxhr_mix22.c    |  6 +++---
 sound/pci/pcxhr/pcxhr_mixer.c    | 22 +++++++++++-----------
 sound/pci/trident/trident_main.c | 22 +++++++++++-----------
 sound/pci/via82xx.c              |  6 +++---
 sound/pci/vx222/vx222_ops.c      |  4 ++--
 sound/pci/ymfpci/ymfpci_main.c   | 14 +++++++-------
 28 files changed, 123 insertions(+), 123 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c
index ab0f87312911..7a4558a70fb9 100644
--- a/sound/pci/au88x0/au88x0_a3d.c
+++ b/sound/pci/au88x0/au88x0_a3d.c
@@ -846,7 +846,7 @@ snd_vortex_a3d_filter_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new vortex_a3d_kcontrol = {
+static const struct snd_kcontrol_new vortex_a3d_kcontrol = {
 	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
 	.name = "Playback PCM advanced processing",
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
diff --git a/sound/pci/au88x0/au88x0_eq.c b/sound/pci/au88x0/au88x0_eq.c
index 9585c5c63b96..b566b44e4da7 100644
--- a/sound/pci/au88x0/au88x0_eq.c
+++ b/sound/pci/au88x0/au88x0_eq.c
@@ -757,7 +757,7 @@ snd_vortex_eqtoggle_put(struct snd_kcontrol *kcontrol,
 	return 1;		/* Allways changes */
 }
 
-static struct snd_kcontrol_new vortex_eqtoggle_kcontrol = {
+static const struct snd_kcontrol_new vortex_eqtoggle_kcontrol = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "EQ Enable",
 	.index = 0,
@@ -815,7 +815,7 @@ snd_vortex_eq_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucon
 	return changed;
 }
 
-static struct snd_kcontrol_new vortex_eq_kcontrol = {
+static const struct snd_kcontrol_new vortex_eq_kcontrol = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "                        .",
 	.index = 0,
@@ -855,7 +855,7 @@ snd_vortex_peaks_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *u
 	return 0;
 }
 
-static struct snd_kcontrol_new vortex_levels_kcontrol = {
+static const struct snd_kcontrol_new vortex_levels_kcontrol = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "EQ Peaks",
 	.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index df5741a78fd2..335979a753fe 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -601,7 +601,7 @@ static int snd_vortex_pcm_vol_put(struct snd_kcontrol *kcontrol,
 
 static const DECLARE_TLV_DB_MINMAX(vortex_pcm_vol_db_scale, -9600, 2400);
 
-static struct snd_kcontrol_new snd_vortex_pcm_vol = {
+static const struct snd_kcontrol_new snd_vortex_pcm_vol = {
 	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
 	.name = "PCM Playback Volume",
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 57bbb87d0c62..8356180bfe0e 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -202,7 +202,7 @@ static const struct snd_pcm_ops snd_aw2_capture_ops = {
 	.pointer = snd_aw2_pcm_pointer_capture,
 };
 
-static struct snd_kcontrol_new aw2_control = {
+static const struct snd_kcontrol_new aw2_control = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "PCM Capture Route",
 	.index = 0,
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index f2c0709d7441..099efb046b1c 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -598,7 +598,7 @@ static int snd_bt87x_capture_volume_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new snd_bt87x_capture_volume = {
+static const struct snd_kcontrol_new snd_bt87x_capture_volume = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Capture Volume",
 	.info = snd_bt87x_capture_volume_info,
@@ -634,7 +634,7 @@ static int snd_bt87x_capture_boost_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new snd_bt87x_capture_boost = {
+static const struct snd_kcontrol_new snd_bt87x_capture_boost = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Capture Boost",
 	.info = snd_bt87x_capture_boost_info,
@@ -676,7 +676,7 @@ static int snd_bt87x_capture_source_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new snd_bt87x_capture_source = {
+static const struct snd_kcontrol_new snd_bt87x_capture_source = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Capture Source",
 	.info = snd_bt87x_capture_source_info,
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index 025805cba779..b4d3415331f6 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -301,7 +301,7 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol,
         return change;
 }
 
-static struct snd_kcontrol_new snd_ca0106_capture_mic_line_in =
+static const struct snd_kcontrol_new snd_ca0106_capture_mic_line_in =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"Shared Mic/Line in Capture Switch",
@@ -310,7 +310,7 @@ static struct snd_kcontrol_new snd_ca0106_capture_mic_line_in =
 	.put =		snd_ca0106_capture_mic_line_in_put
 };
 
-static struct snd_kcontrol_new snd_ca0106_capture_line_in_side_out =
+static const struct snd_kcontrol_new snd_ca0106_capture_line_in_side_out =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"Shared Line in/Side out Capture Switch",
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index aeedc270ed9b..227c9d3802b8 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -1045,7 +1045,7 @@ static int snd_cmipci_spdif_default_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_cmipci_spdif_default =
+static const struct snd_kcontrol_new snd_cmipci_spdif_default =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
 	.name =		SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
@@ -1072,7 +1072,7 @@ static int snd_cmipci_spdif_mask_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_cmipci_spdif_mask =
+static const struct snd_kcontrol_new snd_cmipci_spdif_mask =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
@@ -1119,7 +1119,7 @@ static int snd_cmipci_spdif_stream_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_cmipci_spdif_stream =
+static const struct snd_kcontrol_new snd_cmipci_spdif_stream =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index fa7c51684dd2..f870697aca67 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -1055,7 +1055,7 @@ static int snd_cs4281_put_volume(struct snd_kcontrol *kcontrol,
 
 static const DECLARE_TLV_DB_SCALE(db_scale_dsp, -4650, 150, 0);
 
-static struct snd_kcontrol_new snd_cs4281_fm_vol = 
+static const struct snd_kcontrol_new snd_cs4281_fm_vol =
 {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Synth Playback Volume",
@@ -1066,7 +1066,7 @@ static struct snd_kcontrol_new snd_cs4281_fm_vol =
 	.tlv = { .p = db_scale_dsp },
 };
 
-static struct snd_kcontrol_new snd_cs4281_pcm_vol = 
+static const struct snd_kcontrol_new snd_cs4281_pcm_vol =
 {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "PCM Stream Playback Volume",
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 937071760bc4..d15ecf9febbf 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -1039,7 +1039,7 @@ static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol,
 
 #ifdef ECHOCARD_HAS_LINE_OUT_GAIN
 /* On the Mia this one controls the line-out volume */
-static struct snd_kcontrol_new snd_echo_line_output_gain = {
+static const struct snd_kcontrol_new snd_echo_line_output_gain = {
 	.name = "Line Playback Volume",
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
@@ -1050,7 +1050,7 @@ static struct snd_kcontrol_new snd_echo_line_output_gain = {
 	.tlv = {.p = db_scale_output_gain},
 };
 #else
-static struct snd_kcontrol_new snd_echo_pcm_output_gain = {
+static const struct snd_kcontrol_new snd_echo_pcm_output_gain = {
 	.name = "PCM Playback Volume",
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,
@@ -1120,7 +1120,7 @@ static int snd_echo_input_gain_put(struct snd_kcontrol *kcontrol,
 
 static const DECLARE_TLV_DB_SCALE(db_scale_input_gain, -2500, 50, 0);
 
-static struct snd_kcontrol_new snd_echo_line_input_gain = {
+static const struct snd_kcontrol_new snd_echo_line_input_gain = {
 	.name = "Line Capture Volume",
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,
@@ -1184,7 +1184,7 @@ static int snd_echo_output_nominal_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new snd_echo_output_nominal_level = {
+static const struct snd_kcontrol_new snd_echo_output_nominal_level = {
 	.name = "Line Playback Switch (-10dBV)",
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.info = snd_echo_output_nominal_info,
@@ -1250,7 +1250,7 @@ static int snd_echo_input_nominal_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new snd_echo_intput_nominal_level = {
+static const struct snd_kcontrol_new snd_echo_intput_nominal_level = {
 	.name = "Line Capture Switch (-10dBV)",
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.info = snd_echo_input_nominal_info,
@@ -1477,7 +1477,7 @@ static int snd_echo_digital_mode_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new snd_echo_digital_mode_switch = {
+static const struct snd_kcontrol_new snd_echo_digital_mode_switch = {
 	.name = "Digital mode Switch",
 	.iface = SNDRV_CTL_ELEM_IFACE_CARD,
 	.info = snd_echo_digital_mode_info,
@@ -1527,7 +1527,7 @@ static int snd_echo_spdif_mode_put(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_echo_spdif_mode_switch = {
+static const struct snd_kcontrol_new snd_echo_spdif_mode_switch = {
 	.name = "S/PDIF mode Switch",
 	.iface = SNDRV_CTL_ELEM_IFACE_CARD,
 	.info = snd_echo_spdif_mode_info,
@@ -1600,7 +1600,7 @@ static int snd_echo_clock_source_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new snd_echo_clock_source_switch = {
+static const struct snd_kcontrol_new snd_echo_clock_source_switch = {
 	.name = "Sample Clock Source",
 	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
 	.info = snd_echo_clock_source_info,
@@ -1643,7 +1643,7 @@ static int snd_echo_phantom_power_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new snd_echo_phantom_power_switch = {
+static const struct snd_kcontrol_new snd_echo_phantom_power_switch = {
 	.name = "Phantom power Switch",
 	.iface = SNDRV_CTL_ELEM_IFACE_CARD,
 	.info = snd_echo_phantom_power_info,
@@ -1686,7 +1686,7 @@ static int snd_echo_automute_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new snd_echo_automute_switch = {
+static const struct snd_kcontrol_new snd_echo_automute_switch = {
 	.name = "Digital Capture Switch (automute)",
 	.iface = SNDRV_CTL_ELEM_IFACE_CARD,
 	.info = snd_echo_automute_info,
@@ -1713,7 +1713,7 @@ static int snd_echo_vumeters_switch_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static struct snd_kcontrol_new snd_echo_vumeters_switch = {
+static const struct snd_kcontrol_new snd_echo_vumeters_switch = {
 	.name = "VU-meters Switch",
 	.iface = SNDRV_CTL_ELEM_IFACE_CARD,
 	.access = SNDRV_CTL_ELEM_ACCESS_WRITE,
@@ -1751,7 +1751,7 @@ static int snd_echo_vumeters_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_echo_vumeters = {
+static const struct snd_kcontrol_new snd_echo_vumeters = {
 	.name = "VU-meters",
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access = SNDRV_CTL_ELEM_ACCESS_READ |
@@ -1804,7 +1804,7 @@ static int snd_echo_channels_info_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_echo_channels_info = {
+static const struct snd_kcontrol_new snd_echo_channels_info = {
 	.name = "Channels info",
 	.iface = SNDRV_CTL_ELEM_IFACE_HWDEP,
 	.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 32842734ada6..77a4413f4564 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -1112,7 +1112,7 @@ static int snd_emu10k1x_shared_spdif_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_emu10k1x_shared_spdif =
+static const struct snd_kcontrol_new snd_emu10k1x_shared_spdif =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"Analog/Digital Output Jack",
@@ -1171,7 +1171,7 @@ static int snd_emu10k1x_spdif_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_emu10k1x_spdif_mask_control =
+static const struct snd_kcontrol_new snd_emu10k1x_spdif_mask_control =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
 	.iface =        SNDRV_CTL_ELEM_IFACE_PCM,
@@ -1181,7 +1181,7 @@ static struct snd_kcontrol_new snd_emu10k1x_spdif_mask_control =
 	.get =          snd_emu10k1x_spdif_get_mask
 };
 
-static struct snd_kcontrol_new snd_emu10k1x_spdif_control =
+static const struct snd_kcontrol_new snd_emu10k1x_spdif_control =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
 	.name =         SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c
index 076b117009c5..b2219a73c17c 100644
--- a/sound/pci/emu10k1/emumixer.c
+++ b/sound/pci/emu10k1/emumixer.c
@@ -795,7 +795,7 @@ static int snd_emu1010_internal_clock_put(struct snd_kcontrol *kcontrol,
         return change;
 }
 
-static struct snd_kcontrol_new snd_emu1010_internal_clock =
+static const struct snd_kcontrol_new snd_emu1010_internal_clock =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READWRITE,
 	.iface =        SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -847,7 +847,7 @@ static int snd_emu1010_optical_out_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_emu1010_optical_out = {
+static const struct snd_kcontrol_new snd_emu1010_optical_out = {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READWRITE,
 	.iface =        SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         "Optical Output Mode",
@@ -898,7 +898,7 @@ static int snd_emu1010_optical_in_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_emu1010_optical_in = {
+static const struct snd_kcontrol_new snd_emu1010_optical_in = {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READWRITE,
 	.iface =        SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         "Optical Input Mode",
@@ -978,7 +978,7 @@ static int snd_audigy_i2c_capture_source_put(struct snd_kcontrol *kcontrol,
         return change;
 }
 
-static struct snd_kcontrol_new snd_audigy_i2c_capture_source =
+static const struct snd_kcontrol_new snd_audigy_i2c_capture_source =
 {
 		.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name =		"Capture Source",
@@ -1177,7 +1177,7 @@ static int snd_emu10k1_spdif_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_emu10k1_spdif_mask_control =
+static const struct snd_kcontrol_new snd_emu10k1_spdif_mask_control =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
 	.iface =        SNDRV_CTL_ELEM_IFACE_PCM,
@@ -1187,7 +1187,7 @@ static struct snd_kcontrol_new snd_emu10k1_spdif_mask_control =
 	.get =          snd_emu10k1_spdif_get_mask
 };
 
-static struct snd_kcontrol_new snd_emu10k1_spdif_control =
+static const struct snd_kcontrol_new snd_emu10k1_spdif_control =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
 	.name =         SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
@@ -1293,7 +1293,7 @@ static int snd_emu10k1_send_routing_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_emu10k1_send_routing_control =
+static const struct snd_kcontrol_new snd_emu10k1_send_routing_control =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
 	.iface =        SNDRV_CTL_ELEM_IFACE_PCM,
@@ -1364,7 +1364,7 @@ static int snd_emu10k1_send_volume_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_emu10k1_send_volume_control =
+static const struct snd_kcontrol_new snd_emu10k1_send_volume_control =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
 	.iface =        SNDRV_CTL_ELEM_IFACE_PCM,
@@ -1429,7 +1429,7 @@ static int snd_emu10k1_attn_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_emu10k1_attn_control =
+static const struct snd_kcontrol_new snd_emu10k1_attn_control =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
 	.iface =        SNDRV_CTL_ELEM_IFACE_PCM,
@@ -1501,7 +1501,7 @@ static int snd_emu10k1_efx_send_routing_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_emu10k1_efx_send_routing_control =
+static const struct snd_kcontrol_new snd_emu10k1_efx_send_routing_control =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
 	.iface =        SNDRV_CTL_ELEM_IFACE_PCM,
@@ -1568,7 +1568,7 @@ static int snd_emu10k1_efx_send_volume_put(struct snd_kcontrol *kcontrol,
 }
 
 
-static struct snd_kcontrol_new snd_emu10k1_efx_send_volume_control =
+static const struct snd_kcontrol_new snd_emu10k1_efx_send_volume_control =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
 	.iface =        SNDRV_CTL_ELEM_IFACE_PCM,
@@ -1626,7 +1626,7 @@ static int snd_emu10k1_efx_attn_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_emu10k1_efx_attn_control =
+static const struct snd_kcontrol_new snd_emu10k1_efx_attn_control =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
 	.iface =        SNDRV_CTL_ELEM_IFACE_PCM,
@@ -1691,7 +1691,7 @@ static int snd_emu10k1_shared_spdif_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_emu10k1_shared_spdif =
+static const struct snd_kcontrol_new snd_emu10k1_shared_spdif =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"SB Live Analog/Digital Output Jack",
@@ -1700,7 +1700,7 @@ static struct snd_kcontrol_new snd_emu10k1_shared_spdif =
 	.put =		snd_emu10k1_shared_spdif_put
 };
 
-static struct snd_kcontrol_new snd_audigy_shared_spdif =
+static const struct snd_kcontrol_new snd_audigy_shared_spdif =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"Audigy Analog/Digital Output Jack",
@@ -1738,7 +1738,7 @@ static int snd_audigy_capture_boost_put(struct snd_kcontrol *kcontrol,
 	return snd_ac97_update(emu->ac97, AC97_REC_GAIN, val);
 }
 
-static struct snd_kcontrol_new snd_audigy_capture_boost =
+static const struct snd_kcontrol_new snd_audigy_capture_boost =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"Mic Extra Boost",
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index 37be1e14d756..ef1cf530c929 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -1542,7 +1542,7 @@ static int snd_emu10k1_pcm_efx_voices_mask_put(struct snd_kcontrol *kcontrol, st
 	return change;
 }
 
-static struct snd_kcontrol_new snd_emu10k1_pcm_efx_voices_mask = {
+static const struct snd_kcontrol_new snd_emu10k1_pcm_efx_voices_mask = {
 	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
 	.name = "Captured FX8010 Outputs",
 	.info = snd_emu10k1_pcm_efx_voices_mask_info,
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 164adad91650..5d10349d11ce 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -1530,7 +1530,7 @@ static int snd_es1373_rear_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_ens1373_rear =
+static const struct snd_kcontrol_new snd_ens1373_rear =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"AC97 2ch->4ch Copy Switch",
@@ -1575,7 +1575,7 @@ static int snd_es1373_line_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new snd_ens1373_line =
+static const struct snd_kcontrol_new snd_ens1373_line =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"Line In->Rear Out Switch",
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 9913be8532ab..bf491eb58b24 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1897,7 +1897,7 @@ static int vmaster_mute_mode_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-static struct snd_kcontrol_new vmaster_mute_mode = {
+static const struct snd_kcontrol_new vmaster_mute_mode = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Mute-LED Mode",
 	.info = vmaster_mute_mode_info,
@@ -2637,7 +2637,7 @@ static int spdif_share_sw_put(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new spdif_share_sw = {
+static const struct snd_kcontrol_new spdif_share_sw = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "IEC958 Default PCM Playback Switch",
 	.info = snd_ctl_boolean_mono_info,
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 3fc201c3b95a..3ce872c26436 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -354,7 +354,7 @@ static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new eld_bytes_ctl = {
+static const struct snd_kcontrol_new eld_bytes_ctl = {
 	.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
 	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
 	.name = "ELD",
diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c
index 3bfdc78cbc5f..da5f37b7fdd0 100644
--- a/sound/pci/ice1712/delta.c
+++ b/sound/pci/ice1712/delta.c
@@ -432,7 +432,7 @@ static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kco
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_ice1712_delta1010lt_wordclock_status =
+static const struct snd_kcontrol_new snd_ice1712_delta1010lt_wordclock_status =
 {
 	.access =	(SNDRV_CTL_ELEM_ACCESS_READ),
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c
index 5cb587cf360e..ec07136fc288 100644
--- a/sound/pci/ice1712/ews.c
+++ b/sound/pci/ice1712/ews.c
@@ -719,7 +719,7 @@ static int snd_ice1712_ews88mt_input_sense_put(struct snd_kcontrol *kcontrol, st
 	return ndata != data;
 }
 
-static struct snd_kcontrol_new snd_ice1712_ews88mt_input_sense = {
+static const struct snd_kcontrol_new snd_ice1712_ews88mt_input_sense = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Input Sensitivity Switch",
 	.info = snd_ice1712_ewx_io_sense_info,
@@ -728,7 +728,7 @@ static struct snd_kcontrol_new snd_ice1712_ews88mt_input_sense = {
 	.count = 8,
 };
 
-static struct snd_kcontrol_new snd_ice1712_ews88mt_output_sense = {
+static const struct snd_kcontrol_new snd_ice1712_ews88mt_output_sense = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Output Sensitivity Switch",
 	.info = snd_ice1712_ewx_io_sense_info,
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index b4aa4c1370a8..1d8612cabb9e 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -279,7 +279,7 @@ static int snd_ice1712_digmix_route_ac97_put(struct snd_kcontrol *kcontrol, stru
 	return val != nval;
 }
 
-static struct snd_kcontrol_new snd_ice1712_mixer_digmix_route_ac97 = {
+static const struct snd_kcontrol_new snd_ice1712_mixer_digmix_route_ac97 = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Digital Mixer To AC97",
 	.info = snd_ice1712_digmix_route_ac97_info,
@@ -1410,7 +1410,7 @@ static struct snd_kcontrol_new snd_ice1712_multi_capture_analog_switch = {
 	.private_value = 10,
 };
 
-static struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_switch = {
+static const struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_switch = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = SNDRV_CTL_NAME_IEC958("Multi ", CAPTURE, SWITCH),
 	.info = snd_ice1712_pro_mixer_switch_info,
@@ -1432,7 +1432,7 @@ static struct snd_kcontrol_new snd_ice1712_multi_capture_analog_volume = {
 	.tlv = { .p = db_scale_playback }
 };
 
-static struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_volume = {
+static const struct snd_kcontrol_new snd_ice1712_multi_capture_spdif_volume = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = SNDRV_CTL_NAME_IEC958("Multi ", CAPTURE, VOLUME),
 	.info = snd_ice1712_pro_mixer_volume_info,
@@ -1630,7 +1630,7 @@ static int snd_ice1712_eeprom_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_ice1712_eeprom = {
+static const struct snd_kcontrol_new snd_ice1712_eeprom = {
 	.iface = SNDRV_CTL_ELEM_IFACE_CARD,
 	.name = "ICE1712 EEPROM",
 	.access = SNDRV_CTL_ELEM_ACCESS_READ,
@@ -1666,7 +1666,7 @@ static int snd_ice1712_spdif_default_put(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_ice1712_spdif_default =
+static const struct snd_kcontrol_new snd_ice1712_spdif_default =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
 	.name =         SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
@@ -1717,7 +1717,7 @@ static int snd_ice1712_spdif_maskp_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_ice1712_spdif_maskc =
+static const struct snd_kcontrol_new snd_ice1712_spdif_maskc =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
@@ -1726,7 +1726,7 @@ static struct snd_kcontrol_new snd_ice1712_spdif_maskc =
 	.get =		snd_ice1712_spdif_maskc_get,
 };
 
-static struct snd_kcontrol_new snd_ice1712_spdif_maskp =
+static const struct snd_kcontrol_new snd_ice1712_spdif_maskp =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
@@ -1753,7 +1753,7 @@ static int snd_ice1712_spdif_stream_put(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_ice1712_spdif_stream =
+static const struct snd_kcontrol_new snd_ice1712_spdif_stream =
 {
 	.access =	(SNDRV_CTL_ELEM_ACCESS_READWRITE |
 			 SNDRV_CTL_ELEM_ACCESS_INACTIVE),
@@ -1878,7 +1878,7 @@ static int snd_ice1712_pro_internal_clock_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_ice1712_pro_internal_clock = {
+static const struct snd_kcontrol_new snd_ice1712_pro_internal_clock = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Multi Track Internal Clock",
 	.info = snd_ice1712_pro_internal_clock_info,
@@ -1943,7 +1943,7 @@ static int snd_ice1712_pro_internal_clock_default_put(struct snd_kcontrol *kcont
 	return change;
 }
 
-static struct snd_kcontrol_new snd_ice1712_pro_internal_clock_default = {
+static const struct snd_kcontrol_new snd_ice1712_pro_internal_clock_default = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Multi Track Internal Clock Default",
 	.info = snd_ice1712_pro_internal_clock_default_info,
@@ -1974,7 +1974,7 @@ static int snd_ice1712_pro_rate_locking_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_ice1712_pro_rate_locking = {
+static const struct snd_kcontrol_new snd_ice1712_pro_rate_locking = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Multi Track Rate Locking",
 	.info = snd_ice1712_pro_rate_locking_info,
@@ -2005,7 +2005,7 @@ static int snd_ice1712_pro_rate_reset_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_ice1712_pro_rate_reset = {
+static const struct snd_kcontrol_new snd_ice1712_pro_rate_reset = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Multi Track Rate Reset",
 	.info = snd_ice1712_pro_rate_reset_info,
@@ -2173,7 +2173,7 @@ static struct snd_kcontrol_new snd_ice1712_mixer_pro_analog_route = {
 	.put = snd_ice1712_pro_route_analog_put,
 };
 
-static struct snd_kcontrol_new snd_ice1712_mixer_pro_spdif_route = {
+static const struct snd_kcontrol_new snd_ice1712_mixer_pro_spdif_route = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, NONE) "Route",
 	.info = snd_ice1712_pro_route_info,
@@ -2215,7 +2215,7 @@ static int snd_ice1712_pro_volume_rate_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_ice1712_mixer_pro_volume_rate = {
+static const struct snd_kcontrol_new snd_ice1712_mixer_pro_volume_rate = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Multi Track Volume Rate",
 	.info = snd_ice1712_pro_volume_rate_info,
@@ -2248,7 +2248,7 @@ static int snd_ice1712_pro_peak_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_ice1712_mixer_pro_peak = {
+static const struct snd_kcontrol_new snd_ice1712_mixer_pro_peak = {
 	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
 	.name = "Multi Track Peak",
 	.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index 842744e7a139..9cd6e55c0642 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -1598,7 +1598,7 @@ static int snd_vt1724_eeprom_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_vt1724_eeprom = {
+static const struct snd_kcontrol_new snd_vt1724_eeprom = {
 	.iface = SNDRV_CTL_ELEM_IFACE_CARD,
 	.name = "ICE1724 EEPROM",
 	.access = SNDRV_CTL_ELEM_ACCESS_READ,
@@ -1711,7 +1711,7 @@ static int snd_vt1724_spdif_default_put(struct snd_kcontrol *kcontrol,
 	return val != old;
 }
 
-static struct snd_kcontrol_new snd_vt1724_spdif_default =
+static const struct snd_kcontrol_new snd_vt1724_spdif_default =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
 	.name =         SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
@@ -1743,7 +1743,7 @@ static int snd_vt1724_spdif_maskp_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_vt1724_spdif_maskc =
+static const struct snd_kcontrol_new snd_vt1724_spdif_maskc =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
@@ -1752,7 +1752,7 @@ static struct snd_kcontrol_new snd_vt1724_spdif_maskc =
 	.get =		snd_vt1724_spdif_maskc_get,
 };
 
-static struct snd_kcontrol_new snd_vt1724_spdif_maskp =
+static const struct snd_kcontrol_new snd_vt1724_spdif_maskp =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
@@ -1789,7 +1789,7 @@ static int snd_vt1724_spdif_sw_put(struct snd_kcontrol *kcontrol,
 	return old != val;
 }
 
-static struct snd_kcontrol_new snd_vt1724_spdif_switch =
+static const struct snd_kcontrol_new snd_vt1724_spdif_switch =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	/* FIXME: the following conflict with IEC958 Playback Route */
@@ -1964,7 +1964,7 @@ static int snd_vt1724_pro_internal_clock_put(struct snd_kcontrol *kcontrol,
 	return old_rate != new_rate;
 }
 
-static struct snd_kcontrol_new snd_vt1724_pro_internal_clock = {
+static const struct snd_kcontrol_new snd_vt1724_pro_internal_clock = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Multi Track Internal Clock",
 	.info = snd_vt1724_pro_internal_clock_info,
@@ -1995,7 +1995,7 @@ static int snd_vt1724_pro_rate_locking_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_vt1724_pro_rate_locking = {
+static const struct snd_kcontrol_new snd_vt1724_pro_rate_locking = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Multi Track Rate Locking",
 	.info = snd_vt1724_pro_rate_locking_info,
@@ -2026,7 +2026,7 @@ static int snd_vt1724_pro_rate_reset_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_vt1724_pro_rate_reset = {
+static const struct snd_kcontrol_new snd_vt1724_pro_rate_reset = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Multi Track Rate Reset",
 	.info = snd_vt1724_pro_rate_reset_info,
@@ -2151,7 +2151,7 @@ static struct snd_kcontrol_new snd_vt1724_mixer_pro_analog_route =
 	.put = snd_vt1724_pro_route_analog_put,
 };
 
-static struct snd_kcontrol_new snd_vt1724_mixer_pro_spdif_route = {
+static const struct snd_kcontrol_new snd_vt1724_mixer_pro_spdif_route = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, NONE) "Route",
 	.info = snd_vt1724_pro_route_info,
@@ -2187,7 +2187,7 @@ static int snd_vt1724_pro_peak_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_vt1724_mixer_pro_peak = {
+static const struct snd_kcontrol_new snd_vt1724_mixer_pro_peak = {
 	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
 	.name = "Multi Track Peak",
 	.access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
diff --git a/sound/pci/lola/lola_mixer.c b/sound/pci/lola/lola_mixer.c
index e7fe15dd5a90..cb25acf7bc49 100644
--- a/sound/pci/lola/lola_mixer.c
+++ b/sound/pci/lola/lola_mixer.c
@@ -645,7 +645,7 @@ static int lola_input_src_put(struct snd_kcontrol *kcontrol,
 	return lola_set_src_config(chip, mask, true);
 }
 
-static struct snd_kcontrol_new lola_input_src_mixer = {
+static const struct snd_kcontrol_new lola_input_src_mixer = {
 	.name = "Digital SRC Capture Switch",
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.info = lola_input_src_info,
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index c0f0c349c3ec..f9c3e86d55d5 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -899,7 +899,7 @@ static int lx_control_playback_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new lx_control_playback_switch = {
+static const struct snd_kcontrol_new lx_control_playback_switch = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "PCM Playback Switch",
 	.index = 0,
diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c
index 51e53497f0ad..4a4616aac787 100644
--- a/sound/pci/mixart/mixart_mixer.c
+++ b/sound/pci/mixart/mixart_mixer.c
@@ -448,7 +448,7 @@ static int mixart_audio_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele
 	return changed;
 }
 
-static struct snd_kcontrol_new mixart_control_output_switch = {
+static const struct snd_kcontrol_new mixart_control_output_switch = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         "Master Playback Switch",
 	.info =         mixart_sw_info,		/* shared */
@@ -1024,7 +1024,7 @@ static int mixart_monitor_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
 	return changed;
 }
 
-static struct snd_kcontrol_new mixart_control_monitor_vol = {
+static const struct snd_kcontrol_new mixart_control_monitor_vol = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
 		   SNDRV_CTL_ELEM_ACCESS_TLV_READ),
@@ -1091,7 +1091,7 @@ static int mixart_monitor_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 	return (changed != 0);
 }
 
-static struct snd_kcontrol_new mixart_control_monitor_sw = {
+static const struct snd_kcontrol_new mixart_control_monitor_sw = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         "Monitoring Switch",
 	.info =         mixart_sw_info,		/* shared */
diff --git a/sound/pci/pcxhr/pcxhr_mix22.c b/sound/pci/pcxhr/pcxhr_mix22.c
index 6a56e5306a65..8b4d0282efb8 100644
--- a/sound/pci/pcxhr/pcxhr_mix22.c
+++ b/sound/pci/pcxhr/pcxhr_mix22.c
@@ -744,7 +744,7 @@ static int hr222_mic_vol_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new hr222_control_mic_level = {
+static const struct snd_kcontrol_new hr222_control_mic_level = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access =	(SNDRV_CTL_ELEM_ACCESS_READWRITE |
 			 SNDRV_CTL_ELEM_ACCESS_TLV_READ),
@@ -794,7 +794,7 @@ static int hr222_mic_boost_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new hr222_control_mic_boost = {
+static const struct snd_kcontrol_new hr222_control_mic_boost = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access =	(SNDRV_CTL_ELEM_ACCESS_READWRITE |
 			 SNDRV_CTL_ELEM_ACCESS_TLV_READ),
@@ -836,7 +836,7 @@ static int hr222_phantom_power_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new hr222_phantom_power_switch = {
+static const struct snd_kcontrol_new hr222_phantom_power_switch = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Phantom Power Switch",
 	.info = hr222_phantom_power_info,
diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c
index 63136c4f3f3d..36875df30dbf 100644
--- a/sound/pci/pcxhr/pcxhr_mixer.c
+++ b/sound/pci/pcxhr/pcxhr_mixer.c
@@ -235,7 +235,7 @@ static int pcxhr_audio_sw_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new pcxhr_control_output_switch = {
+static const struct snd_kcontrol_new pcxhr_control_output_switch = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"Master Playback Switch",
 	.info =		pcxhr_sw_info,		/* shared */
@@ -460,7 +460,7 @@ static int pcxhr_pcm_sw_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new pcxhr_control_pcm_switch = {
+static const struct snd_kcontrol_new pcxhr_control_pcm_switch = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"PCM Playback Switch",
 	.count =	PCXHR_PLAYBACK_STREAMS,
@@ -509,7 +509,7 @@ static int pcxhr_monitor_vol_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new pcxhr_control_monitor_vol = {
+static const struct snd_kcontrol_new pcxhr_control_monitor_vol = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access =	(SNDRV_CTL_ELEM_ACCESS_READWRITE |
 			 SNDRV_CTL_ELEM_ACCESS_TLV_READ),
@@ -562,7 +562,7 @@ static int pcxhr_monitor_sw_put(struct snd_kcontrol *kcontrol,
 	return (changed != 0);
 }
 
-static struct snd_kcontrol_new pcxhr_control_monitor_sw = {
+static const struct snd_kcontrol_new pcxhr_control_monitor_sw = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         "Monitoring Playback Switch",
 	.info =         pcxhr_sw_info,		/* shared */
@@ -697,7 +697,7 @@ static int pcxhr_audio_src_put(struct snd_kcontrol *kcontrol,
 	return ret;
 }
 
-static struct snd_kcontrol_new pcxhr_control_audio_src = {
+static const struct snd_kcontrol_new pcxhr_control_audio_src = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"Capture Source",
 	.info =		pcxhr_audio_src_info,
@@ -798,7 +798,7 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol,
 	return ret;
 }
 
-static struct snd_kcontrol_new pcxhr_control_clock_type = {
+static const struct snd_kcontrol_new pcxhr_control_clock_type = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"Clock Mode",
 	.info =		pcxhr_clock_type_info,
@@ -842,7 +842,7 @@ static int pcxhr_clock_rate_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new pcxhr_control_clock_rate = {
+static const struct snd_kcontrol_new pcxhr_control_clock_rate = {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
 	.iface =	SNDRV_CTL_ELEM_IFACE_CARD,
 	.name =		"Clock Rates",
@@ -1017,14 +1017,14 @@ static int pcxhr_iec958_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-static struct snd_kcontrol_new pcxhr_control_playback_iec958_mask = {
+static const struct snd_kcontrol_new pcxhr_control_playback_iec958_mask = {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
 	.name =		SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
 	.info =		pcxhr_iec958_info,
 	.get =		pcxhr_iec958_mask_get
 };
-static struct snd_kcontrol_new pcxhr_control_playback_iec958 = {
+static const struct snd_kcontrol_new pcxhr_control_playback_iec958 = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
 	.name =         SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
 	.info =         pcxhr_iec958_info,
@@ -1033,14 +1033,14 @@ static struct snd_kcontrol_new pcxhr_control_playback_iec958 = {
 	.private_value = 0 /* playback */
 };
 
-static struct snd_kcontrol_new pcxhr_control_capture_iec958_mask = {
+static const struct snd_kcontrol_new pcxhr_control_capture_iec958_mask = {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
 	.name =		SNDRV_CTL_NAME_IEC958("",CAPTURE,MASK),
 	.info =		pcxhr_iec958_info,
 	.get =		pcxhr_iec958_mask_get
 };
-static struct snd_kcontrol_new pcxhr_control_capture_iec958 = {
+static const struct snd_kcontrol_new pcxhr_control_capture_iec958 = {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
 	.name =         SNDRV_CTL_NAME_IEC958("",CAPTURE,DEFAULT),
diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
index 92ad2d7a6bf8..64d3b8eba4bb 100644
--- a/sound/pci/trident/trident_main.c
+++ b/sound/pci/trident/trident_main.c
@@ -2356,7 +2356,7 @@ static int snd_trident_spdif_control_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_trident_spdif_control =
+static const struct snd_kcontrol_new snd_trident_spdif_control =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH),
@@ -2419,7 +2419,7 @@ static int snd_trident_spdif_default_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_trident_spdif_default =
+static const struct snd_kcontrol_new snd_trident_spdif_default =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
 	.name =         SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
@@ -2452,7 +2452,7 @@ static int snd_trident_spdif_mask_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_trident_spdif_mask =
+static const struct snd_kcontrol_new snd_trident_spdif_mask =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
@@ -2514,7 +2514,7 @@ static int snd_trident_spdif_stream_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_trident_spdif_stream =
+static const struct snd_kcontrol_new snd_trident_spdif_stream =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
@@ -2564,7 +2564,7 @@ static int snd_trident_ac97_control_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_trident_ac97_rear_control =
+static const struct snd_kcontrol_new snd_trident_ac97_rear_control =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         "Rear Path",
@@ -2622,7 +2622,7 @@ static int snd_trident_vol_control_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_trident_vol_music_control =
+static const struct snd_kcontrol_new snd_trident_vol_music_control =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         "Music Playback Volume",
@@ -2633,7 +2633,7 @@ static struct snd_kcontrol_new snd_trident_vol_music_control =
 	.tlv = { .p = db_scale_gvol },
 };
 
-static struct snd_kcontrol_new snd_trident_vol_wave_control =
+static const struct snd_kcontrol_new snd_trident_vol_wave_control =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         "Wave Playback Volume",
@@ -2700,7 +2700,7 @@ static int snd_trident_pcm_vol_control_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_trident_pcm_vol_control =
+static const struct snd_kcontrol_new snd_trident_pcm_vol_control =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         "PCM Front Playback Volume",
@@ -2764,7 +2764,7 @@ static int snd_trident_pcm_pan_control_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_trident_pcm_pan_control =
+static const struct snd_kcontrol_new snd_trident_pcm_pan_control =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         "PCM Pan Playback Control",
@@ -2821,7 +2821,7 @@ static int snd_trident_pcm_rvol_control_put(struct snd_kcontrol *kcontrol,
 
 static const DECLARE_TLV_DB_SCALE(db_scale_crvol, -3175, 25, 1);
 
-static struct snd_kcontrol_new snd_trident_pcm_rvol_control =
+static const struct snd_kcontrol_new snd_trident_pcm_rvol_control =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         "PCM Reverb Playback Volume",
@@ -2877,7 +2877,7 @@ static int snd_trident_pcm_cvol_control_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_trident_pcm_cvol_control =
+static const struct snd_kcontrol_new snd_trident_pcm_cvol_control =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =         "PCM Chorus Playback Volume",
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 2d8c14e3f8d2..d078e86414c2 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -1683,7 +1683,7 @@ static int snd_via8233_dxs3_spdif_put(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_via8233_dxs3_spdif_control = {
+static const struct snd_kcontrol_new snd_via8233_dxs3_spdif_control = {
 	.name = SNDRV_CTL_NAME_IEC958("Output ",NONE,SWITCH),
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.info = snd_via8233_dxs3_spdif_info,
@@ -1772,7 +1772,7 @@ static int snd_via8233_pcmdxs_volume_put(struct snd_kcontrol *kcontrol,
 
 static const DECLARE_TLV_DB_SCALE(db_scale_dxs, -4650, 150, 1);
 
-static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control = {
+static const struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control = {
 	.name = "PCM Playback Volume",
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
@@ -1783,7 +1783,7 @@ static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control = {
 	.tlv = { .p = db_scale_dxs }
 };
 
-static struct snd_kcontrol_new snd_via8233_dxs_volume_control = {
+static const struct snd_kcontrol_new snd_via8233_dxs_volume_control = {
 	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
 	.device = 0,
 	/* .subdevice set later */
diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c
index 8e457ea27f89..7df1663ea510 100644
--- a/sound/pci/vx222/vx222_ops.c
+++ b/sound/pci/vx222/vx222_ops.c
@@ -945,7 +945,7 @@ static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
 	return 0;
 }
 
-static struct snd_kcontrol_new vx_control_input_level = {
+static const struct snd_kcontrol_new vx_control_input_level = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access =	(SNDRV_CTL_ELEM_ACCESS_READWRITE |
 			 SNDRV_CTL_ELEM_ACCESS_TLV_READ),
@@ -956,7 +956,7 @@ static struct snd_kcontrol_new vx_control_input_level = {
 	.tlv = { .p = db_scale_mic },
 };
 
-static struct snd_kcontrol_new vx_control_mic_level = {
+static const struct snd_kcontrol_new vx_control_mic_level = {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.access =	(SNDRV_CTL_ELEM_ACCESS_READWRITE |
 			 SNDRV_CTL_ELEM_ACCESS_TLV_READ),
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index ffee284898b3..fe4ba463b57c 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -1316,7 +1316,7 @@ static int snd_ymfpci_spdif_default_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_ymfpci_spdif_default =
+static const struct snd_kcontrol_new snd_ymfpci_spdif_default =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
 	.name =         SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
@@ -1344,7 +1344,7 @@ static int snd_ymfpci_spdif_mask_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_ymfpci_spdif_mask =
+static const struct snd_kcontrol_new snd_ymfpci_spdif_mask =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
@@ -1391,7 +1391,7 @@ static int snd_ymfpci_spdif_stream_put(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
-static struct snd_kcontrol_new snd_ymfpci_spdif_stream =
+static const struct snd_kcontrol_new snd_ymfpci_spdif_stream =
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_INACTIVE,
 	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
@@ -1439,7 +1439,7 @@ static int snd_ymfpci_drec_source_put(struct snd_kcontrol *kcontrol, struct snd_
 	return reg != old_reg;
 }
 
-static struct snd_kcontrol_new snd_ymfpci_drec_source = {
+static const struct snd_kcontrol_new snd_ymfpci_drec_source = {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READWRITE,
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name =		"Direct Recording Source",
@@ -1609,7 +1609,7 @@ static int snd_ymfpci_put_dup4ch(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 	return change;
 }
 
-static struct snd_kcontrol_new snd_ymfpci_dup4ch = {
+static const struct snd_kcontrol_new snd_ymfpci_dup4ch = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "4ch Duplication",
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
@@ -1712,7 +1712,7 @@ static int snd_ymfpci_gpio_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_ymfpci_rear_shared = {
+static const struct snd_kcontrol_new snd_ymfpci_rear_shared = {
 	.name = "Shared Rear/Line-In Switch",
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.info = snd_ymfpci_gpio_sw_info,
@@ -1776,7 +1776,7 @@ static int snd_ymfpci_pcm_vol_put(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static struct snd_kcontrol_new snd_ymfpci_pcm_volume = {
+static const struct snd_kcontrol_new snd_ymfpci_pcm_volume = {
 	.iface = SNDRV_CTL_ELEM_IFACE_PCM,
 	.name = "PCM Playback Volume",
 	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
-- 
cgit v1.2.3


From 5824ce8de7b1c0a75e5942e4df4652d04f3e263d Mon Sep 17 00:00:00 2001
From: Chris Chiu <chiu@endlessm.com>
Date: Tue, 28 Feb 2017 14:17:11 -0600
Subject: ALSA: hda/realtek - Add support for Acer Aspire E5-475 headset mic

The Acer laptop Aspire E5-475 with ALC255 can't detect the headset
microphone until we modify a pin definition.

Signed-off-by: Chris Chiu <chiu@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 14 ++++++++++++++
 1 file changed, 14 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 73a00460b5c1..a3fa4a8574df 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4833,6 +4833,7 @@ enum {
 	ALC290_FIXUP_SUBWOOFER_HSJACK,
 	ALC269_FIXUP_THINKPAD_ACPI,
 	ALC269_FIXUP_DMIC_THINKPAD_ACPI,
+	ALC255_FIXUP_ACER_MIC_NO_PRESENCE,
 	ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
 	ALC255_FIXUP_DELL2_MIC_NO_PRESENCE,
 	ALC255_FIXUP_HEADSET_MODE,
@@ -5287,6 +5288,15 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_THINKPAD_ACPI,
 	},
+	[ALC255_FIXUP_ACER_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC255_FIXUP_HEADSET_MODE
+	},
 	[ALC255_FIXUP_DELL1_MIC_NO_PRESENCE] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -5851,6 +5861,10 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{0x21, 0x03211020}
 
 static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
+	SND_HDA_PIN_QUIRK(0x10ec0255, 0x1025, "Acer", ALC255_FIXUP_ACER_MIC_NO_PRESENCE,
+		{0x12, 0x90a601c0},
+		{0x14, 0x90171120},
+		{0x21, 0x02211030}),
 	SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
 		ALC225_STANDARD_PINS,
 		{0x12, 0xb7a60130},
-- 
cgit v1.2.3


From 615966adc4b616dbc2f67ec2c44c25034d458cd9 Mon Sep 17 00:00:00 2001
From: Chris Chiu <chiu@endlessm.com>
Date: Tue, 28 Feb 2017 14:17:12 -0600
Subject: ALSA: hda/realtek - Fix headset mic on several Asus laptops with
 ALC255

Add pin quirks to enable use of the headset mic on Asus Z550MA,
X540LA, X540LJ, X556UR, Z450LA, and X441NC.

Signed-off-by: Chris Chiu <chiu@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 21 +++++++++++++++++++++
 1 file changed, 21 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a3fa4a8574df..990994077e97 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4834,6 +4834,7 @@ enum {
 	ALC269_FIXUP_THINKPAD_ACPI,
 	ALC269_FIXUP_DMIC_THINKPAD_ACPI,
 	ALC255_FIXUP_ACER_MIC_NO_PRESENCE,
+	ALC255_FIXUP_ASUS_MIC_NO_PRESENCE,
 	ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
 	ALC255_FIXUP_DELL2_MIC_NO_PRESENCE,
 	ALC255_FIXUP_HEADSET_MODE,
@@ -5297,6 +5298,15 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC255_FIXUP_HEADSET_MODE
 	},
+	[ALC255_FIXUP_ASUS_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC255_FIXUP_HEADSET_MODE
+	},
 	[ALC255_FIXUP_DELL1_MIC_NO_PRESENCE] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -5689,6 +5699,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
 	SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1043, 0x10d0, "ASUS X540LA/X540LJ", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
@@ -5865,6 +5878,14 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x12, 0x90a601c0},
 		{0x14, 0x90171120},
 		{0x21, 0x02211030}),
+	SND_HDA_PIN_QUIRK(0x10ec0255, 0x1043, "ASUS", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE,
+		{0x14, 0x90170110},
+		{0x1b, 0x90a70130},
+		{0x21, 0x03211020}),
+	SND_HDA_PIN_QUIRK(0x10ec0255, 0x1043, "ASUS", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE,
+		{0x1a, 0x90a70130},
+		{0x1b, 0x90170110},
+		{0x21, 0x03211020}),
 	SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
 		ALC225_STANDARD_PINS,
 		{0x12, 0xb7a60130},
-- 
cgit v1.2.3


From c1732ede5e8008324f908861077b50f4ca55701d Mon Sep 17 00:00:00 2001
From: Chris Chiu <chiu@endlessm.com>
Date: Tue, 28 Feb 2017 14:17:13 -0600
Subject: ALSA: hda/realtek - Fix headset and mic on several Asus laptops with
 ALC256
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

Enable jack sensing and fix internal mic and headset mic on
Asus X555UB and X540SA.

Fix internal mic and headset mic on Asus E402NA and E403NA.

Fix headset mic on Asus X541UV, X541SA and Z550SA.

Unfortunately jack sensing for the headset mic is still not working.
We believe this is a codec limitation.

Some of these quirks were authored by João Paulo Rechi Vita.

Signed-off-by: Chris Chiu <chiu@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 30 ++++++++++++++++++++++++++++++
 1 file changed, 30 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 990994077e97..a9f6b1d750d2 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4872,6 +4872,8 @@ enum {
 	ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER,
 	ALC269_FIXUP_ATIV_BOOK_8,
 	ALC221_FIXUP_HP_MIC_NO_PRESENCE,
+	ALC256_FIXUP_ASUS_HEADSET_MODE,
+	ALC256_FIXUP_ASUS_MIC,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -5578,6 +5580,20 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MODE
 	},
+	[ALC256_FIXUP_ASUS_HEADSET_MODE] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_headset_mode,
+	},
+	[ALC256_FIXUP_ASUS_MIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x13, 0x90a60160 }, /* use as internal mic */
+			{ 0x19, 0x04a11120 }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -5690,9 +5706,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC),
 	SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x1043, 0x10c0, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC),
+	SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC),
+	SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK),
 	SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A),
 	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
@@ -5702,6 +5723,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x10d0, "ASUS X540LA/X540LJ", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
@@ -6006,6 +6028,14 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x21, 0x02211020}),
 	SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
 		ALC256_STANDARD_PINS),
+	SND_HDA_PIN_QUIRK(0x10ec0256, 0x1043, "ASUS", ALC256_FIXUP_ASUS_MIC,
+		{0x14, 0x90170110},
+		{0x1b, 0x90a70130},
+		{0x21, 0x04211020}),
+	SND_HDA_PIN_QUIRK(0x10ec0256, 0x1043, "ASUS", ALC256_FIXUP_ASUS_MIC,
+		{0x14, 0x90170110},
+		{0x1b, 0x90a70130},
+		{0x21, 0x03211020}),
 	SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
 		{0x12, 0x90a60130},
 		{0x14, 0x90170110},
-- 
cgit v1.2.3


From 216d7aebbfbe1d3361e21c3a97d1607e1c1c48cd Mon Sep 17 00:00:00 2001
From: Chris Chiu <chiu@endlessm.com>
Date: Tue, 28 Feb 2017 14:17:14 -0600
Subject: ALSA: hda/realtek - Fix headset mic and speaker on Asus X441SA/X441UV

ASUS X441SA and X441UV laptops with ALC3236 (ALC223) codec require
the known fixup (ALC269_FIXUP_HEADSET_MIC) and a different pin value on
pin 0x19 to make the headset mic work.

To make the speaker work, it requires an EAPD verb fixup.

Signed-off-by: Chris Chiu <chiu@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 24 ++++++++++++++++++++++++
 1 file changed, 24 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a9f6b1d750d2..662ecf0abfa0 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4874,6 +4874,8 @@ enum {
 	ALC221_FIXUP_HP_MIC_NO_PRESENCE,
 	ALC256_FIXUP_ASUS_HEADSET_MODE,
 	ALC256_FIXUP_ASUS_MIC,
+	ALC233_FIXUP_ASUS_MIC_NO_PRESENCE,
+	ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -5594,6 +5596,26 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
 	},
+	[ALC233_FIXUP_ASUS_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MIC
+	},
+	[ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			/* Enables internal speaker */
+			{0x20, AC_VERB_SET_COEF_INDEX, 0x40},
+			{0x20, AC_VERB_SET_PROC_COEF, 0x8800},
+			{}
+		},
+		.chained = true,
+		.chain_id = ALC233_FIXUP_ASUS_MIC_NO_PRESENCE
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -5723,6 +5745,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x10d0, "ASUS X540LA/X540LJ", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
-- 
cgit v1.2.3


From eeed4cd15ae0a8d3f2751adae8b56a3571e8aeca Mon Sep 17 00:00:00 2001
From: Chris Chiu <chiu@endlessm.com>
Date: Tue, 28 Feb 2017 14:17:15 -0600
Subject: ALSA: hda/realtek - Fix speaker support for Asus AiO ZN270IE

Asus AiO ZN270IE with ALC256 has no audio ouput for internal speaker
and headphone. It requires GPIO 2 as an amp. This commit enables the
GPIO and pulls it high.

Signed-off-by: Chris Chiu <chiu@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 12 ++++++++++++
 1 file changed, 12 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 662ecf0abfa0..f3919bb45af6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4874,6 +4874,7 @@ enum {
 	ALC221_FIXUP_HP_MIC_NO_PRESENCE,
 	ALC256_FIXUP_ASUS_HEADSET_MODE,
 	ALC256_FIXUP_ASUS_MIC,
+	ALC256_FIXUP_ASUS_AIO_GPIO2,
 	ALC233_FIXUP_ASUS_MIC_NO_PRESENCE,
 	ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE,
 };
@@ -5596,6 +5597,16 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE
 	},
+	[ALC256_FIXUP_ASUS_AIO_GPIO2] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			/* Set up GPIO2 for the speaker amp */
+			{ 0x01, AC_VERB_SET_GPIO_MASK, 0x04 },
+			{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x04 },
+			{ 0x01, AC_VERB_SET_GPIO_DATA, 0x04 },
+			{}
+		},
+	},
 	[ALC233_FIXUP_ASUS_MIC_NO_PRESENCE] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -5748,6 +5759,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC),
+	SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC),
-- 
cgit v1.2.3


From a16fbb85c78a3ce56dc4515ffb8632b82cc969c7 Mon Sep 17 00:00:00 2001
From: Matthias Kaehlcke <mka@chromium.org>
Date: Wed, 15 Mar 2017 15:41:23 -0700
Subject: ALSA: hda/ca0132: Remove double parentheses

The extra pairs of parantheses are not needed and causes clang to
generate warnings like this:

sound/pci/hda/patch_ca0132.c:1171:14: error: equality comparison with extraneous parentheses [-Werror,-Wparentheses-equality]
        if ((buffer == NULL))
             ~~~~~~~^~~~~~~
sound/pci/hda/patch_ca0132.c:1171:14: note: remove extraneous parentheses around the comparison to silence this warning
        if ((buffer == NULL))
            ~       ^      ~
sound/pci/hda/patch_ca0132.c:1171:14: note: use '=' to turn this equality comparison into an assignment
        if ((buffer == NULL))

Signed-off-by: Matthias Kaehlcke <mka@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 07a9deb17477..fb2e242c2522 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1168,7 +1168,7 @@ static int dspio_write_multiple(struct hda_codec *codec,
 	int status = 0;
 	unsigned int count;
 
-	if ((buffer == NULL))
+	if (buffer == NULL)
 		return -EINVAL;
 
 	count = 0;
@@ -1210,7 +1210,7 @@ static int dspio_read_multiple(struct hda_codec *codec, unsigned int *buffer,
 	unsigned int skip_count;
 	unsigned int dummy;
 
-	if ((buffer == NULL))
+	if (buffer == NULL)
 		return -1;
 
 	count = 0;
-- 
cgit v1.2.3


From 13f99ebdd602ebdafb909e15ec6ffb1e34690167 Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Thu, 23 Mar 2017 16:15:55 +0100
Subject: ALSA: au88x0: avoid theoretical uninitialized access

The latest gcc-7.0.1 snapshot points out that we if nr_ch is zero, we never
initialize some variables:

sound/pci/au88x0/au88x0_core.c: In function 'vortex_adb_allocroute':
sound/pci/au88x0/au88x0_core.c:2304:68: error: 'mix[0]' may be used uninitialized in this function [-Werror=maybe-uninitialized]
sound/pci/au88x0/au88x0_core.c:2305:58: error: 'src[0]' may be used uninitialized in this function [-Werror=maybe-uninitialized]

I assume this can never happen in practice, but adding a check here doesn't
hurt either and avoids the warning. The code has been unchanged since
the start of git history.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/au88x0/au88x0_core.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index e1af24f87566..c308a4f70550 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -2279,6 +2279,9 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir,
 	} else {
 		int src[2], mix[2];
 
+		if (nr_ch < 1)
+			return -EINVAL;
+
 		/* Get SRC and MIXER hardware resources. */
 		for (i = 0; i < nr_ch; i++) {
 			if ((mix[i] =
-- 
cgit v1.2.3


From ac310dc9fafcffaddbde7ca2137edb59cd4f414f Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Wed, 29 Mar 2017 20:55:50 +0200
Subject: ALSA: oxygen: simply setting of the shortname for Xonar DG cards

We don't need to manually set the card name; with an entry in the
names[] array, this happens automatically.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/oxygen/oxygen.c | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index 74afb6b75976..e36ed8af55ad 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -767,6 +767,8 @@ static int get_oxygen_model(struct oxygen *chip,
 		[MODEL_FANTASIA]	= "TempoTec HiFier Fantasia",
 		[MODEL_SERENADE]	= "TempoTec HiFier Serenade",
 		[MODEL_HG2PCI]		= "CMI8787-HG2PCI",
+		[MODEL_XONAR_DG]        = "Xonar DG",
+		[MODEL_XONAR_DGX]       = "Xonar DGX",
 	};
 
 	chip->model = model_generic;
@@ -829,12 +831,8 @@ static int get_oxygen_model(struct oxygen *chip,
 		chip->model.dac_channels_mixer = 2;
 		break;
 	case MODEL_XONAR_DG:
-		chip->model = model_xonar_dg;
-		chip->model.shortname = "Xonar DG";
-		break;
 	case MODEL_XONAR_DGX:
 		chip->model = model_xonar_dg;
-		chip->model.shortname = "Xonar DGX";
 		break;
 	}
 	if (id->driver_data == MODEL_MERIDIAN ||
-- 
cgit v1.2.3


From d1600401faad4bc186bfdb291d8af644465e20bd Mon Sep 17 00:00:00 2001
From: Matthias Kaehlcke <mka@chromium.org>
Date: Fri, 31 Mar 2017 18:00:04 -0700
Subject: ALSA: hda/ca0132: Limit values for chip addresses to 32-bit

With the previous unsigned long value clang generates warnings like
this:

sound/pci/hda/patch_ca0132.c:860:37: error: implicit conversion from
'unsigned long' to 'u32' (aka 'unsigned int') changes value from
18446744073709551615 to 4294967295 [-Werror,-Wconstant-conversion]
        spec->curr_chip_addx = (res < 0) ? ~0UL : chip_addx;
                             ~             ^~~~

Signed-off-by: Matthias Kaehlcke <mka@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index fb2e242c2522..a148176c16a9 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -857,7 +857,7 @@ static int chipio_write_address(struct hda_codec *codec,
 				  chip_addx >> 16);
 	}
 
-	spec->curr_chip_addx = (res < 0) ? ~0UL : chip_addx;
+	spec->curr_chip_addx = (res < 0) ? ~0U : chip_addx;
 
 	return res;
 }
@@ -882,7 +882,7 @@ static int chipio_write_data(struct hda_codec *codec, unsigned int data)
 	/*If no error encountered, automatically increment the address
 	as per chip behaviour*/
 	spec->curr_chip_addx = (res != -EIO) ?
-					(spec->curr_chip_addx + 4) : ~0UL;
+					(spec->curr_chip_addx + 4) : ~0U;
 	return res;
 }
 
@@ -933,7 +933,7 @@ static int chipio_read_data(struct hda_codec *codec, unsigned int *data)
 	/*If no error encountered, automatically increment the address
 	as per chip behaviour*/
 	spec->curr_chip_addx = (res != -EIO) ?
-					(spec->curr_chip_addx + 4) : ~0UL;
+					(spec->curr_chip_addx + 4) : ~0U;
 	return res;
 }
 
-- 
cgit v1.2.3


From 70eafad849f8af3a83e139eda36712d3c1da5b6a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 29 Mar 2017 08:39:19 +0200
Subject: ALSA: hda - Move SKL+ vendor specific register definitions to
 hda_register.h

They may be used by both legacy and ASoC drivers.

Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c8256a89375a..a48330f4a1a9 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -534,9 +534,9 @@ static void bxt_reduce_dma_latency(struct azx *chip)
 {
 	u32 val;
 
-	val = azx_readl(chip, SKL_EM4L);
+	val = azx_readl(chip, VS_EM4L);
 	val &= (0x3 << 20);
-	azx_writel(chip, SKL_EM4L, val);
+	azx_writel(chip, VS_EM4L, val);
 }
 
 static void hda_intel_init_chip(struct azx *chip, bool full_reset)
-- 
cgit v1.2.3


From f87e7f25893d7db4da465c6b50882197e518d4af Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 29 Mar 2017 08:46:00 +0200
Subject: ALSA: hda - Improved position reporting on SKL+

Apply the same methods to obtain the current stream position as ASoC
Intel SKL driver uses.  It reads the position from DPIB for a playback
stream while it still reads from the position buffer for a capture
stream.  For a capture stream, some ugly workaround is needed to
settle down the inconsistent position.

Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 36 ++++++++++++++++++++++++++++++++++--
 1 file changed, 34 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index a48330f4a1a9..64db6698214c 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -77,6 +77,7 @@ enum {
 	POS_FIX_POSBUF,
 	POS_FIX_VIACOMBO,
 	POS_FIX_COMBO,
+	POS_FIX_SKL,
 };
 
 /* Defines for ATI HD Audio support in SB450 south bridge */
@@ -148,7 +149,7 @@ module_param_array(model, charp, NULL, 0444);
 MODULE_PARM_DESC(model, "Use the given board model.");
 module_param_array(position_fix, int, NULL, 0444);
 MODULE_PARM_DESC(position_fix, "DMA pointer read method."
-		 "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO).");
+		 "(-1 = system default, 0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO, 5 = SKL+).");
 module_param_array(bdl_pos_adj, int, NULL, 0644);
 MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset.");
 module_param_array(probe_mask, int, NULL, 0444);
@@ -815,6 +816,31 @@ static unsigned int azx_via_get_position(struct azx *chip,
 	return bound_pos + mod_dma_pos;
 }
 
+static unsigned int azx_skl_get_dpib_pos(struct azx *chip,
+					 struct azx_dev *azx_dev)
+{
+	return _snd_hdac_chip_readl(azx_bus(chip),
+				    AZX_REG_VS_SDXDPIB_XBASE +
+				    (AZX_REG_VS_SDXDPIB_XINTERVAL *
+				     azx_dev->core.index));
+}
+
+/* get the current DMA position with correction on SKL+ chips */
+static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev)
+{
+	/* DPIB register gives a more accurate position for playback */
+	if (azx_dev->core.substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		return azx_skl_get_dpib_pos(chip, azx_dev);
+
+	/* For capture, we need to read posbuf, but it requires a delay
+	 * for the possible boundary overlap; the read of DPIB fetches the
+	 * actual posbuf
+	 */
+	udelay(20);
+	azx_skl_get_dpib_pos(chip, azx_dev);
+	return azx_get_pos_posbuf(chip, azx_dev);
+}
+
 #ifdef CONFIG_PM
 static DEFINE_MUTEX(card_list_lock);
 static LIST_HEAD(card_list);
@@ -1351,6 +1377,7 @@ static int check_position_fix(struct azx *chip, int fix)
 	case POS_FIX_POSBUF:
 	case POS_FIX_VIACOMBO:
 	case POS_FIX_COMBO:
+	case POS_FIX_SKL:
 		return fix;
 	}
 
@@ -1371,6 +1398,10 @@ static int check_position_fix(struct azx *chip, int fix)
 		dev_dbg(chip->card->dev, "Using LPIB position fix\n");
 		return POS_FIX_LPIB;
 	}
+	if (IS_SKL_PLUS(chip->pci)) {
+		dev_dbg(chip->card->dev, "Using SKL position fix\n");
+		return POS_FIX_SKL;
+	}
 	return POS_FIX_AUTO;
 }
 
@@ -1382,6 +1413,7 @@ static void assign_position_fix(struct azx *chip, int fix)
 		[POS_FIX_POSBUF] = azx_get_pos_posbuf,
 		[POS_FIX_VIACOMBO] = azx_via_get_position,
 		[POS_FIX_COMBO] = azx_get_pos_lpib,
+		[POS_FIX_SKL] = azx_get_pos_skl,
 	};
 
 	chip->get_position[0] = chip->get_position[1] = callbacks[fix];
@@ -1390,7 +1422,7 @@ static void assign_position_fix(struct azx *chip, int fix)
 	if (fix == POS_FIX_COMBO)
 		chip->get_position[1] = NULL;
 
-	if (fix == POS_FIX_POSBUF &&
+	if ((fix == POS_FIX_POSBUF || fix == POS_FIX_SKL) &&
 	    (chip->driver_caps & AZX_DCAPS_COUNT_LPIB_DELAY)) {
 		chip->get_delay[0] = chip->get_delay[1] =
 			azx_get_delay_from_lpib;
-- 
cgit v1.2.3


From 1f9d3d98694b1cef93f99a54e6830e9717616ba6 Mon Sep 17 00:00:00 2001
From: Libin Yang <libin.yang@intel.com>
Date: Thu, 6 Apr 2017 19:18:21 +0800
Subject: ALSA: hda - set intel audio clock to a proper value

On some Intel platforms, the audio clock may not be set correctly
with initial setting. This will cause the audio playback/capture
rates wrong.

This patch checks the audio clock setting and will set it to a
proper value if it is set incorrectly.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=188411

Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 95 +++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 95 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 64db6698214c..59ab34fa0bc8 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -540,6 +540,98 @@ static void bxt_reduce_dma_latency(struct azx *chip)
 	azx_writel(chip, VS_EM4L, val);
 }
 
+/*
+ * ML_LCAP bits:
+ *  bit 0: 6 MHz Supported
+ *  bit 1: 12 MHz Supported
+ *  bit 2: 24 MHz Supported
+ *  bit 3: 48 MHz Supported
+ *  bit 4: 96 MHz Supported
+ *  bit 5: 192 MHz Supported
+ */
+static int intel_get_lctl_scf(struct azx *chip)
+{
+	struct hdac_bus *bus = azx_bus(chip);
+	static int preferred_bits[] = { 2, 3, 1, 4, 5 };
+	u32 val, t;
+	int i;
+
+	val = readl(bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCAP);
+
+	for (i = 0; i < ARRAY_SIZE(preferred_bits); i++) {
+		t = preferred_bits[i];
+		if (val & (1 << t))
+			return t;
+	}
+
+	dev_warn(chip->card->dev, "set audio clock frequency to 6MHz");
+	return 0;
+}
+
+static int intel_ml_lctl_set_power(struct azx *chip, int state)
+{
+	struct hdac_bus *bus = azx_bus(chip);
+	u32 val;
+	int timeout;
+
+	/*
+	 * the codecs are sharing the first link setting by default
+	 * If other links are enabled for stream, they need similar fix
+	 */
+	val = readl(bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL);
+	val &= ~AZX_MLCTL_SPA;
+	val |= state << AZX_MLCTL_SPA_SHIFT;
+	writel(val, bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL);
+	/* wait for CPA */
+	timeout = 50;
+	while (timeout) {
+		if (((readl(bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL)) &
+		    AZX_MLCTL_CPA) == (state << AZX_MLCTL_CPA_SHIFT))
+			return 0;
+		timeout--;
+		udelay(10);
+	}
+
+	return -1;
+}
+
+static void intel_init_lctl(struct azx *chip)
+{
+	struct hdac_bus *bus = azx_bus(chip);
+	u32 val;
+	int ret;
+
+	/* 0. check lctl register value is correct or not */
+	val = readl(bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL);
+	/* if SCF is already set, let's use it */
+	if ((val & ML_LCTL_SCF_MASK) != 0)
+		return;
+
+	/*
+	 * Before operating on SPA, CPA must match SPA.
+	 * Any deviation may result in undefined behavior.
+	 */
+	if (((val & AZX_MLCTL_SPA) >> AZX_MLCTL_SPA_SHIFT) !=
+		((val & AZX_MLCTL_CPA) >> AZX_MLCTL_CPA_SHIFT))
+		return;
+
+	/* 1. turn link down: set SPA to 0 and wait CPA to 0 */
+	ret = intel_ml_lctl_set_power(chip, 0);
+	udelay(100);
+	if (ret)
+		goto set_spa;
+
+	/* 2. update SCF to select a properly audio clock*/
+	val &= ~ML_LCTL_SCF_MASK;
+	val |= intel_get_lctl_scf(chip);
+	writel(val, bus->mlcap + AZX_ML_BASE + AZX_REG_ML_LCTL);
+
+set_spa:
+	/* 4. turn link up: set SPA to 1 and wait CPA to 1 */
+	intel_ml_lctl_set_power(chip, 1);
+	udelay(100);
+}
+
 static void hda_intel_init_chip(struct azx *chip, bool full_reset)
 {
 	struct hdac_bus *bus = azx_bus(chip);
@@ -565,6 +657,9 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset)
 	/* reduce dma latency to avoid noise */
 	if (IS_BXT(pci))
 		bxt_reduce_dma_latency(chip);
+
+	if (bus->mlcap != NULL)
+		intel_init_lctl(chip);
 }
 
 /* calculate runtime delay from LPIB */
-- 
cgit v1.2.3


From 9f3dadb156692e15933a4a97ef659f65151be4a1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 10 Apr 2017 17:12:33 +0200
Subject: ALSA: hda - A new flag to enforce prefix to each pin

This is a preliminary patch for a smooth multi-codec support, and it
introduces a new flag, force_pin_prefix, to struct hda_codec.
This flag is used to force to add the pin location prefix to each
input pin.  For example, when there is only one microphone pin,
usually the auto-parser assigns the string "Mic".  With this flag on,
it'll be like "Front Mic".  Also, the creation of "Master" or "PCM"
playback volume for a single pin is suppressed, too.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195305
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_auto_parser.c | 1 +
 sound/pci/hda/hda_codec.h       | 1 +
 sound/pci/hda/hda_generic.c     | 2 ++
 3 files changed, 4 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index a03cf68d0bcd..d3ea73171a3d 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -580,6 +580,7 @@ const char *hda_get_autocfg_input_label(struct hda_codec *codec,
 		has_multiple_pins = 1;
 	if (has_multiple_pins && type == AUTO_PIN_MIC)
 		has_multiple_pins &= check_mic_location_need(codec, cfg, input);
+	has_multiple_pins |= codec->force_pin_prefix;
 	return hda_get_input_pin_label(codec, &cfg->inputs[input],
 				       cfg->inputs[input].pin,
 				       has_multiple_pins);
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index f17f25245e52..d6fb2d5d01a7 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -256,6 +256,7 @@ struct hda_codec {
 	unsigned int dump_coef:1; /* dump processing coefs in codec proc file */
 	unsigned int power_save_node:1; /* advanced PM for each widget */
 	unsigned int auto_runtime_pm:1; /* enable automatic codec runtime pm */
+	unsigned int force_pin_prefix:1; /* Add location prefix */
 #ifdef CONFIG_PM
 	unsigned long power_on_acct;
 	unsigned long power_off_acct;
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index e7c8f4f076d5..443832870a44 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -1125,6 +1125,7 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch,
 
 	*index = 0;
 	if (cfg->line_outs == 1 && !spec->multi_ios &&
+	    !codec->force_pin_prefix &&
 	    !cfg->hp_outs && !cfg->speaker_outs)
 		return spec->vmaster_mute.hook ? "PCM" : "Master";
 
@@ -1132,6 +1133,7 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch,
 	 * use it master (or "PCM" if a vmaster hook is present)
 	 */
 	if (spec->multiout.num_dacs == 1 && !spec->mixer_nid &&
+	    !codec->force_pin_prefix &&
 	    !spec->multiout.hp_out_nid[0] && !spec->multiout.extra_out_nid[0])
 		return spec->vmaster_mute.hook ? "PCM" : "Master";
 
-- 
cgit v1.2.3


From 7480316c265c9fcdbf73b1b8dec061b893b7e987 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 10 Apr 2017 17:37:34 +0200
Subject: ALSA: hda - Allow to enable/disable vmaster build explicitly

Another preliminary patch for the dual-codec support: since the
support of vmaster over multiple codecs is difficult, simply disable
it by a new flag to hda_codec struct.  A new user hint is added as
well for consistency.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c | 7 +++++--
 sound/pci/hda/hda_generic.h | 1 +
 2 files changed, 6 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 443832870a44..2842c82363c0 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -196,6 +196,9 @@ static void parse_user_hints(struct hda_codec *codec)
 	val = snd_hda_get_bool_hint(codec, "hp_mic_detect");
 	if (val >= 0)
 		spec->suppress_hp_mic_detect = !val;
+	val = snd_hda_get_bool_hint(codec, "vmaster");
+	if (val >= 0)
+		spec->suppress_vmaster = !val;
 
 	if (!snd_hda_get_int_hint(codec, "mixer_nid", &val))
 		spec->mixer_nid = val;
@@ -5033,7 +5036,7 @@ int snd_hda_gen_build_controls(struct hda_codec *codec)
 	}
 
 	/* if we have no master control, let's create it */
-	if (!spec->no_analog &&
+	if (!spec->no_analog && !spec->suppress_vmaster &&
 	    !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
 		err = snd_hda_add_vmaster(codec, "Master Playback Volume",
 					  spec->vmaster_tlv, slave_pfxs,
@@ -5041,7 +5044,7 @@ int snd_hda_gen_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 	}
-	if (!spec->no_analog &&
+	if (!spec->no_analog && !spec->suppress_vmaster &&
 	    !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
 		err = __snd_hda_add_vmaster(codec, "Master Playback Switch",
 					    NULL, slave_pfxs,
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index f66fc7e25e07..61772317de46 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -229,6 +229,7 @@ struct hda_gen_spec {
 	unsigned int add_jack_modes:1; /* add i/o jack mode enum ctls */
 	unsigned int power_down_unused:1; /* power down unused widgets */
 	unsigned int dac_min_mute:1; /* minimal = mute for DACs */
+	unsigned int suppress_vmaster:1; /* don't create vmaster kctls */
 
 	/* other internal flags */
 	unsigned int no_analog:1; /* digital I/O only */
-- 
cgit v1.2.3


From 12ee4022f67f8854061b46e5c0a7ad6258ab66c2 Mon Sep 17 00:00:00 2001
From: "Subhransu S. Prusty" <subhransu.s.prusty@intel.com>
Date: Wed, 12 Apr 2017 09:54:00 +0530
Subject: ALSA: hda: Add Geminilake id to SKL_PLUS

Geminilake is Skylake family platform. So add it's id to skl_plus check.

Fixes: 126cfa2f5e15 ("ALSA: hda: Add Geminilake HDMI codec ID")
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Cc: Senthilnathan Veppur <senthilnathanx.veppur@intel.com>
Cc: Vinod Koul <vinod.koul@intel.com>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 4 +++-
 1 file changed, 3 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 59ab34fa0bc8..b786fbab029f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -370,8 +370,10 @@ enum {
 #define IS_KBL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d71)
 #define IS_KBL_H(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa2f0)
 #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98)
+#define IS_GLK(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x3198)
 #define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) || \
-			IS_KBL(pci) || IS_KBL_LP(pci) || IS_KBL_H(pci)
+			IS_KBL(pci) || IS_KBL_LP(pci) || IS_KBL_H(pci)	|| \
+			IS_GLK(pci)
 
 static char *driver_short_names[] = {
 	[AZX_DRIVER_ICH] = "HDA Intel",
-- 
cgit v1.2.3


From a87a4d23e86b7ff9b3f1300774c942de139ba16a Mon Sep 17 00:00:00 2001
From: Ander Conselvan De Oliveira <ander.conselvan.de.oliveira@intel.com>
Date: Thu, 13 Apr 2017 13:05:35 +0530
Subject: ALSA: hda: Move common haswell init to a helper

Geminilake vendor nid is different from other Skylake variants, but rest
of the initialization code is same.

So a variable is added in hdmi_spec to store the platform specific vendor
nid and move the initialization code to a helper function to be used by
both platform specific init.

Fixes: 126cfa2f5e15 ("ALSA: hda: Add Geminilake HDMI codec ID")
Signed-off-by: Ander Conselvan De Oliveira <ander.conselvan.de.oliveira@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Jaikrishna Nemallapudi <jaikrishnax.nemallapudi@intel.com>
Cc: Senthilnathan Veppur <senthilnathanx.veppur@intel.com>
Cc: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 27 +++++++++++++++++++++------
 1 file changed, 21 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 37f11560186a..90e4ff87445e 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -177,6 +177,7 @@ struct hdmi_spec {
 	bool i915_bound; /* was i915 bound in this driver? */
 
 	struct hdac_chmap chmap;
+	hda_nid_t vendor_nid;
 };
 
 #ifdef CONFIG_SND_HDA_I915
@@ -2372,6 +2373,7 @@ static void intel_haswell_fixup_connect_list(struct hda_codec *codec,
 }
 
 #define INTEL_VENDOR_NID 0x08
+#define INTEL_GLK_VENDOR_NID 0x0B
 #define INTEL_GET_VENDOR_VERB 0xf81
 #define INTEL_SET_VENDOR_VERB 0x781
 #define INTEL_EN_DP12			0x02 /* enable DP 1.2 features */
@@ -2381,14 +2383,15 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec,
 					  bool update_tree)
 {
 	unsigned int vendor_param;
+	struct hdmi_spec *spec = codec->spec;
 
-	vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0,
+	vendor_param = snd_hda_codec_read(codec, spec->vendor_nid, 0,
 				INTEL_GET_VENDOR_VERB, 0);
 	if (vendor_param == -1 || vendor_param & INTEL_EN_ALL_PIN_CVTS)
 		return;
 
 	vendor_param |= INTEL_EN_ALL_PIN_CVTS;
-	vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0,
+	vendor_param = snd_hda_codec_read(codec, spec->vendor_nid, 0,
 				INTEL_SET_VENDOR_VERB, vendor_param);
 	if (vendor_param == -1)
 		return;
@@ -2400,8 +2403,9 @@ static void intel_haswell_enable_all_pins(struct hda_codec *codec,
 static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec)
 {
 	unsigned int vendor_param;
+	struct hdmi_spec *spec = codec->spec;
 
-	vendor_param = snd_hda_codec_read(codec, INTEL_VENDOR_NID, 0,
+	vendor_param = snd_hda_codec_read(codec, spec->vendor_nid, 0,
 				INTEL_GET_VENDOR_VERB, 0);
 	if (vendor_param == -1 || vendor_param & INTEL_EN_DP12)
 		return;
@@ -2409,7 +2413,7 @@ static void intel_haswell_fixup_enable_dp12(struct hda_codec *codec)
 	/* enable DP1.2 mode */
 	vendor_param |= INTEL_EN_DP12;
 	snd_hdac_regmap_add_vendor_verb(&codec->core, INTEL_SET_VENDOR_VERB);
-	snd_hda_codec_write_cache(codec, INTEL_VENDOR_NID, 0,
+	snd_hda_codec_write_cache(codec, spec->vendor_nid, 0,
 				INTEL_SET_VENDOR_VERB, vendor_param);
 }
 
@@ -2503,7 +2507,7 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec,
 }
 
 /* Intel Haswell and onwards; audio component with eld notifier */
-static int patch_i915_hsw_hdmi(struct hda_codec *codec)
+static int intel_hsw_common_init(struct hda_codec *codec, hda_nid_t vendor_nid)
 {
 	struct hdmi_spec *spec;
 	int err;
@@ -2520,6 +2524,7 @@ static int patch_i915_hsw_hdmi(struct hda_codec *codec)
 	spec = codec->spec;
 	codec->dp_mst = true;
 	spec->dyn_pcm_assign = true;
+	spec->vendor_nid = vendor_nid;
 
 	intel_haswell_enable_all_pins(codec, true);
 	intel_haswell_fixup_enable_dp12(codec);
@@ -2548,6 +2553,16 @@ static int patch_i915_hsw_hdmi(struct hda_codec *codec)
 	return 0;
 }
 
+static int patch_i915_hsw_hdmi(struct hda_codec *codec)
+{
+	return intel_hsw_common_init(codec, INTEL_VENDOR_NID);
+}
+
+static int patch_i915_glk_hdmi(struct hda_codec *codec)
+{
+	return intel_hsw_common_init(codec, INTEL_GLK_VENDOR_NID);
+}
+
 /* Intel Baytrail and Braswell; with eld notifier */
 static int patch_i915_byt_hdmi(struct hda_codec *codec)
 {
@@ -3800,7 +3815,7 @@ HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI",	patch_i915_hsw_hdmi),
 HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI",	patch_i915_hsw_hdmi),
 HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI",	patch_i915_hsw_hdmi),
 HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI",	patch_i915_hsw_hdmi),
-HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI",	patch_i915_hsw_hdmi),
+HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI",	patch_i915_glk_hdmi),
 HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI",	patch_generic_hdmi),
 HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI",	patch_i915_byt_hdmi),
 HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI",	patch_i915_byt_hdmi),
-- 
cgit v1.2.3


From 5cd5b1bdfb0137d0e814a51ff203d72c76b9f375 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Jer=C3=B3nimo=20Borque?= <jeronimo@borque.com.ar>
Date: Wed, 19 Apr 2017 12:09:50 -0300
Subject: ALSA: hda - Add HP ZBook 15u G3 Conexant CX20724 GPIO mute leds
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit

The HP ZBook 15u G3 has a Conexant CX20724 with mute led on GPIO1 and
mic mute led on GPIO2.
Adding CXT_FIXUP_MUTE_LED_GPIO inspired on patch_realtek's one.

Signed-off-by: Jerónimo Borque <jeronimo@borque.com.ar>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 81 ++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 81 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 69266b8ea2ad..e8253737c47a 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -52,6 +52,12 @@ struct conexant_spec {
 	bool dc_enable;
 	unsigned int dc_input_bias; /* offset into olpc_xo_dc_bias */
 	struct nid_path *dc_mode_path;
+
+	int mute_led_polarity;
+	unsigned int gpio_led;
+	unsigned int gpio_mute_led_mask;
+	unsigned int gpio_mic_led_mask;
+
 };
 
 
@@ -264,6 +270,7 @@ enum {
 	CXT_FIXUP_HP_DOCK,
 	CXT_FIXUP_HP_SPECTRE,
 	CXT_FIXUP_HP_GATE_MIC,
+	CXT_FIXUP_MUTE_LED_GPIO,
 };
 
 /* for hda_fixup_thinkpad_acpi() */
@@ -646,6 +653,74 @@ static void cxt_fixup_hp_gate_mic_jack(struct hda_codec *codec,
 		snd_hda_jack_set_gating_jack(codec, 0x19, 0x16);
 }
 
+/* update LED status via GPIO */
+static void cxt_update_gpio_led(struct hda_codec *codec, unsigned int mask,
+				bool enabled)
+{
+	struct conexant_spec *spec = codec->spec;
+	unsigned int oldval = spec->gpio_led;
+
+	if (spec->mute_led_polarity)
+		enabled = !enabled;
+
+	if (enabled)
+		spec->gpio_led &= ~mask;
+	else
+		spec->gpio_led |= mask;
+	if (spec->gpio_led != oldval)
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+				    spec->gpio_led);
+}
+
+/* turn on/off mute LED via GPIO per vmaster hook */
+static void cxt_fixup_gpio_mute_hook(void *private_data, int enabled)
+{
+	struct hda_codec *codec = private_data;
+	struct conexant_spec *spec = codec->spec;
+
+	cxt_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled);
+}
+
+/* turn on/off mic-mute LED via GPIO per capture hook */
+static void cxt_fixup_gpio_mic_mute_hook(struct hda_codec *codec,
+					 struct snd_kcontrol *kcontrol,
+					 struct snd_ctl_elem_value *ucontrol)
+{
+	struct conexant_spec *spec = codec->spec;
+
+	if (ucontrol)
+		cxt_update_gpio_led(codec, spec->gpio_mic_led_mask,
+				    ucontrol->value.integer.value[0] ||
+				    ucontrol->value.integer.value[1]);
+}
+
+
+static void cxt_fixup_mute_led_gpio(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	struct conexant_spec *spec = codec->spec;
+	static const struct hda_verb gpio_init[] = {
+		{ 0x01, AC_VERB_SET_GPIO_MASK, 0x03 },
+		{ 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03 },
+		{}
+	};
+	codec_info(codec, "action: %d gpio_led: %d\n", action, spec->gpio_led);
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+		spec->gen.vmaster_mute.hook = cxt_fixup_gpio_mute_hook;
+		spec->gen.cap_sync_hook = cxt_fixup_gpio_mic_mute_hook;
+		spec->gpio_led = 0;
+		spec->mute_led_polarity = 0;
+		spec->gpio_mute_led_mask = 0x01;
+		spec->gpio_mic_led_mask = 0x02;
+	}
+	snd_hda_add_verbs(codec, gpio_init);
+	if (spec->gpio_led)
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+				    spec->gpio_led);
+}
+
+
 /* ThinkPad X200 & co with cxt5051 */
 static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = {
 	{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
@@ -799,6 +874,10 @@ static const struct hda_fixup cxt_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cxt_fixup_hp_gate_mic_jack,
 	},
+	[CXT_FIXUP_MUTE_LED_GPIO] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = cxt_fixup_mute_led_gpio,
+	},
 };
 
 static const struct snd_pci_quirk cxt5045_fixups[] = {
@@ -851,6 +930,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE),
 	SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC),
+	SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO),
 	SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
 	SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
@@ -882,6 +962,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = {
 	{ .id = CXT_FIXUP_OLPC_XO, .name = "olpc-xo" },
 	{ .id = CXT_FIXUP_MUTE_LED_EAPD, .name = "mute-led-eapd" },
 	{ .id = CXT_FIXUP_HP_DOCK, .name = "hp-dock" },
+	{ .id = CXT_FIXUP_MUTE_LED_GPIO, .name = "mute-led-gpio" },
 	{}
 };
 
-- 
cgit v1.2.3


From fc7438b1eb12b6c93d7b7a62423779eb5dfc673c Mon Sep 17 00:00:00 2001
From: Mikhail Paulyshka <me@mixaill.tk>
Date: Fri, 21 Apr 2017 08:52:42 +0200
Subject: ALSA: hda - Fix headset microphone detection for ASUS N551 and N751

Headset microphone does not work out of the box on ASUS Nx51
laptops. This patch fixes it.

Patch tested on Asus N551 laptop. Asus N751 part is not tested, but
according to [1] this laptop uses the same audiosystem.

1. https://bugzilla.kernel.org/show_bug.cgi?id=117781

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195437
Signed-off-by: Mikhail Paulyshka <me@mixaill.tk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 12 ++++++++++--
 1 file changed, 10 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9371f1a95b33..05a26fba5ef2 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6815,6 +6815,7 @@ enum {
 	ALC668_FIXUP_DELL_DISABLE_AAMIX,
 	ALC668_FIXUP_DELL_XPS13,
 	ALC662_FIXUP_ASUS_Nx50,
+	ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE,
 	ALC668_FIXUP_ASUS_Nx51,
 	ALC891_FIXUP_HEADSET_MODE,
 	ALC891_FIXUP_DELL_MIC_NO_PRESENCE,
@@ -7068,14 +7069,21 @@ static const struct hda_fixup alc662_fixups[] = {
 		.chained = true,
 		.chain_id = ALC662_FIXUP_BASS_1A
 	},
+	[ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_headset_mode_alc668,
+		.chain_id = ALC662_FIXUP_BASS_CHMAP
+	},
 	[ALC668_FIXUP_ASUS_Nx51] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
-			{0x1a, 0x90170151}, /* bass speaker */
+			{ 0x19, 0x03a1913d }, /* use as headphone mic, without its own jack detect */
+			{ 0x1a, 0x90170151 }, /* bass speaker */
+			{ 0x1b, 0x03a1113c }, /* use as headset mic, without its own jack detect */
 			{}
 		},
 		.chained = true,
-		.chain_id = ALC662_FIXUP_BASS_CHMAP,
+		.chain_id = ALC668_FIXUP_ASUS_Nx51_HEADSET_MODE,
 	},
 	[ALC891_FIXUP_HEADSET_MODE] = {
 		.type = HDA_FIXUP_FUNC,
-- 
cgit v1.2.3


From 7beb3a6e9335d13872f24566b451f8371edd238f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 10 Apr 2017 18:05:52 +0200
Subject: ALSA: hda - Support Gigabyte Gaming board with dual Realtek codecs

This patch adds some workarounds to make Gigabyte GA-AX370 Gaming 5
board working without the conflicts of kctls, etc.  In general, the
dual codec configs result in the conflicts of the following stuff:
- Master controls
- Capture controls
- Analog loopback controls
In addition, the auto-mute and the auto-mic can't work well among
multiple codecs.

The current "solution" is to disable all these features, and use UCM
for a better PulseAudio management.  For a dedicated UCM profile, the
patch overrides the card longname so that the system an get a unique
profile path.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195305
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 61 +++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 61 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 9371f1a95b33..e81cf83d2afd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1800,6 +1800,7 @@ enum {
 	ALC882_FIXUP_NO_PRIMARY_HP,
 	ALC887_FIXUP_ASUS_BASS,
 	ALC887_FIXUP_BASS_CHMAP,
+	ALC1220_FIXUP_GB_DUAL_CODECS,
 };
 
 static void alc889_fixup_coef(struct hda_codec *codec,
@@ -1962,6 +1963,61 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec,
 static void alc_fixup_bass_chmap(struct hda_codec *codec,
 				 const struct hda_fixup *fix, int action);
 
+/* For dual-codec configuration, we need to disable some features to avoid
+ * conflicts of kctls and PCM streams
+ */
+static void alc_fixup_dual_codecs(struct hda_codec *codec,
+				  const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (action != HDA_FIXUP_ACT_PRE_PROBE)
+		return;
+	/* disable vmaster */
+	spec->gen.suppress_vmaster = 1;
+	/* auto-mute and auto-mic switch don't work with multiple codecs */
+	spec->gen.suppress_auto_mute = 1;
+	spec->gen.suppress_auto_mic = 1;
+	/* disable aamix as well */
+	spec->gen.mixer_nid = 0;
+	/* add location prefix to avoid conflicts */
+	codec->force_pin_prefix = 1;
+}
+
+static void rename_ctl(struct hda_codec *codec, const char *oldname,
+		       const char *newname)
+{
+	struct snd_kcontrol *kctl;
+
+	kctl = snd_hda_find_mixer_ctl(codec, oldname);
+	if (kctl)
+		strcpy(kctl->id.name, newname);
+}
+
+static void alc1220_fixup_gb_dual_codecs(struct hda_codec *codec,
+					 const struct hda_fixup *fix,
+					 int action)
+{
+	alc_fixup_dual_codecs(codec, fix, action);
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		/* override card longname to provide a unique UCM profile */
+		strcpy(codec->card->longname, "HDAudio-Gigabyte-ALC1220DualCodecs");
+		break;
+	case HDA_FIXUP_ACT_BUILD:
+		/* rename Capture controls depending on the codec */
+		rename_ctl(codec, "Capture Volume",
+			   codec->addr == 0 ?
+			   "Rear-Panel Capture Volume" :
+			   "Front-Panel Capture Volume");
+		rename_ctl(codec, "Capture Switch",
+			   codec->addr == 0 ?
+			   "Rear-Panel Capture Switch" :
+			   "Front-Panel Capture Switch");
+		break;
+	}
+}
+
 static const struct hda_fixup alc882_fixups[] = {
 	[ALC882_FIXUP_ABIT_AW9D_MAX] = {
 		.type = HDA_FIXUP_PINS,
@@ -2198,6 +2254,10 @@ static const struct hda_fixup alc882_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc_fixup_bass_chmap,
 	},
+	[ALC1220_FIXUP_GB_DUAL_CODECS] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc1220_fixup_gb_dual_codecs,
+	},
 };
 
 static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -2267,6 +2327,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD),
 	SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
 	SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD),
 	SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD),
-- 
cgit v1.2.3


From 56798e6b3a2264b58d50b93cfd66586ede77d661 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 11 Apr 2017 08:10:52 +0200
Subject: ALSA: hda - Use a helper function for renaming kctl names

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 11 ++++-------
 1 file changed, 4 insertions(+), 7 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e81cf83d2afd..57eeefd36f19 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4724,7 +4724,6 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec,
 		{ 0x1b, 0x21114000 }, /* dock speaker pin */
 		{}
 	};
-	struct snd_kcontrol *kctl;
 
 	switch (action) {
 	case HDA_FIXUP_ACT_PRE_PROBE:
@@ -4739,12 +4738,10 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec,
 		/* this is a bit tricky; give more sane names for the main
 		 * (tablet) speaker and the dock speaker, respectively
 		 */
-		kctl = snd_hda_find_mixer_ctl(codec, "Speaker Playback Switch");
-		if (kctl)
-			strcpy(kctl->id.name, "Dock Speaker Playback Switch");
-		kctl = snd_hda_find_mixer_ctl(codec, "Bass Speaker Playback Switch");
-		if (kctl)
-			strcpy(kctl->id.name, "Speaker Playback Switch");
+		rename_ctl(codec, "Speaker Playback Switch",
+			   "Dock Speaker Playback Switch");
+		rename_ctl(codec, "Bass Speaker Playback Switch",
+			   "Speaker Playback Switch");
 		break;
 	}
 }
-- 
cgit v1.2.3


From ca169cc2f9e1f8ed9c867b197a49d6dd05e5436d Mon Sep 17 00:00:00 2001
From: Kailang Yang <kailang@realtek.com>
Date: Tue, 25 Apr 2017 16:17:40 +0800
Subject: ALSA: hda/realtek - Add Dual Codecs support for Lenovo P520/420

Lenovo P520/420 build with two codecs.
ALC233 for front panel.
ALC662 for rear panel.
This patch will rename capture name for slove conflicts.
And create a card longname for UCM profile.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 36 ++++++++++++++++++++++++++++++++++++
 1 file changed, 36 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 93846bf65a23..58df440013c5 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4824,6 +4824,30 @@ static void alc280_fixup_hp_9480m(struct hda_codec *codec,
 	}
 }
 
+static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec,
+					 const struct hda_fixup *fix,
+					 int action)
+{
+	alc_fixup_dual_codecs(codec, fix, action);
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		/* override card longname to provide a unique UCM profile */
+		strcpy(codec->card->longname, "HDAudio-Lenovo-DualCodecs");
+		break;
+	case HDA_FIXUP_ACT_BUILD:
+		/* rename Capture controls depending on the codec */
+		rename_ctl(codec, "Capture Volume",
+			   codec->addr == 0 ?
+			   "Rear-Panel Capture Volume" :
+			   "Front-Panel Capture Volume");
+		rename_ctl(codec, "Capture Switch",
+			   codec->addr == 0 ?
+			   "Rear-Panel Capture Switch" :
+			   "Front-Panel Capture Switch");
+		break;
+	}
+}
+
 /* for hda_fixup_thinkpad_acpi() */
 #include "thinkpad_helper.c"
 
@@ -4937,6 +4961,7 @@ enum {
 	ALC256_FIXUP_ASUS_AIO_GPIO2,
 	ALC233_FIXUP_ASUS_MIC_NO_PRESENCE,
 	ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE,
+	ALC233_FIXUP_LENOVO_MULTI_CODECS,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -5706,6 +5731,10 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC233_FIXUP_ASUS_MIC_NO_PRESENCE
 	},
+	[ALC233_FIXUP_LENOVO_MULTI_CODECS] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc233_alc662_fixup_lenovo_dual_codecs,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -5860,6 +5889,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
 	SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC),
+	SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
@@ -6881,6 +6911,7 @@ enum {
 	ALC892_FIXUP_ASROCK_MOBO,
 	ALC662_FIXUP_USI_FUNC,
 	ALC662_FIXUP_USI_HEADSET_MODE,
+	ALC662_FIXUP_LENOVO_MULTI_CODECS,
 };
 
 static const struct hda_fixup alc662_fixups[] = {
@@ -7186,6 +7217,10 @@ static const struct hda_fixup alc662_fixups[] = {
 		.chained = true,
 		.chain_id = ALC662_FIXUP_USI_FUNC
 	},
+	[ALC662_FIXUP_LENOVO_MULTI_CODECS] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc233_alc662_fixup_lenovo_dual_codecs,
+	},
 };
 
 static const struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -7223,6 +7258,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2),
 	SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x14cd, 0x5003, "USI", ALC662_FIXUP_USI_HEADSET_MODE),
+	SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC662_FIXUP_LENOVO_MULTI_CODECS),
 	SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD),
 	SND_PCI_QUIRK(0x1849, 0x5892, "ASRock B150M", ALC892_FIXUP_ASROCK_MOBO),
-- 
cgit v1.2.3


From 0997e378be1d888a5d093eeee55f842cdfe55ebe Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.king@canonical.com>
Date: Sat, 29 Apr 2017 22:52:55 +0100
Subject: ALSA: ali5451: fix spelling mistake in "ali_capture_preapre"

trivial fix to spelling mistake in dev_warn message,
"ali_capture_preapre" should be "ali_capture_prepare"

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ali5451/ali5451.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 92b819e4f729..34bbc2e730a6 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -1339,7 +1339,7 @@ static int snd_ali_prepare(struct snd_pcm_substream *substream)
 		rate = snd_ali_get_spdif_in_rate(codec);
 		if (rate == 0) {
 			dev_warn(codec->card->dev,
-				 "ali_capture_preapre: spdif rate detect err!\n");
+				 "ali_capture_prepare: spdif rate detect err!\n");
 			rate = 48000;
 		}
 		spin_lock_irq(&codec->reg_lock);
-- 
cgit v1.2.3