From bf2aa9ccc8e598d788132ded9c7e94c6af3a9d1e Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:20 -0400
Subject: ALSA: hda/ca0132 - Cleanup ca0132_mmio_init function.

Cleanup the ca0132_mmio_init function, separating into two separate
functions, one for Sound Blaster Z/ZxR/Recon3D, and another for the
AE-5.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-2-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 158 +++++++++++++++++++++++++++----------------
 1 file changed, 99 insertions(+), 59 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index b7dbf2e7f77a..7491e2044638 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -8108,78 +8108,118 @@ static void r3di_pre_dsp_setup(struct hda_codec *codec)
  * what they do, or if they're necessary. Could possibly
  * be removed. Figure they're better to leave in.
  */
-static void ca0132_mmio_init(struct hda_codec *codec)
+static const unsigned int ca0113_mmio_init_address_sbz[] = {
+	0x400, 0x408, 0x40c, 0x01c, 0xc0c, 0xc00, 0xc04, 0xc0c, 0xc0c, 0xc0c,
+	0xc0c, 0xc08, 0xc08, 0xc08, 0xc08, 0xc08, 0xc04
+};
+
+static const unsigned int ca0113_mmio_init_data_sbz[] = {
+	0x00000030, 0x00000000, 0x00000003, 0x00000003, 0x00000003,
+	0x00000003, 0x000000c1, 0x000000f1, 0x00000001, 0x000000c7,
+	0x000000c1, 0x00000080
+};
+
+static const unsigned int ca0113_mmio_init_data_zxr[] = {
+	0x00000030, 0x00000000, 0x00000000, 0x00000003, 0x00000003,
+	0x00000003, 0x00000001, 0x000000f1, 0x00000001, 0x000000c7,
+	0x000000c1, 0x00000080
+};
+
+static const unsigned int ca0113_mmio_init_address_ae5[] = {
+	0x400, 0x42c, 0x46c, 0x4ac, 0x4ec, 0x43c, 0x47c, 0x4bc, 0x4fc, 0x408,
+	0x100, 0x410, 0x40c, 0x100, 0x100, 0x830, 0x86c, 0x800, 0x86c, 0x800,
+	0x804, 0x20c, 0x01c, 0xc0c, 0xc00, 0xc04, 0xc0c, 0xc0c, 0xc0c, 0xc0c,
+	0xc08, 0xc08, 0xc08, 0xc08, 0xc08, 0xc04, 0x01c
+};
+
+static const unsigned int ca0113_mmio_init_data_ae5[] = {
+	0x00000001, 0x00000000, 0x00000000, 0x00000000, 0x00000000,
+	0x00000000, 0x00000000, 0x00000000, 0x00000000, 0x00000001,
+	0x00000600, 0x00000014, 0x00000001, 0x0000060f, 0x0000070f,
+	0x00000aff, 0x00000000, 0x0000006b, 0x00000001, 0x0000006b,
+	0x00000057, 0x00800000, 0x00880680, 0x00000080, 0x00000030,
+	0x00000000, 0x00000000, 0x00000003, 0x00000003, 0x00000003,
+	0x00000001, 0x000000f1, 0x00000001, 0x000000c7, 0x000000c1,
+	0x00000080, 0x00880680
+};
+
+static void ca0132_mmio_init_sbz(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp[2], i, count, cur_addr;
+	const unsigned int *addr, *data;
 
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000001, spec->mem_base + 0x400);
-	else
-		writel(0x00000000, spec->mem_base + 0x400);
+	addr = ca0113_mmio_init_address_sbz;
+	for (i = 0; i < 3; i++)
+		writel(0x00000000, spec->mem_base + addr[i]);
 
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000001, spec->mem_base + 0x408);
-	else
-		writel(0x00000000, spec->mem_base + 0x408);
+	cur_addr = i;
+	switch (ca0132_quirk(spec)) {
+	case QUIRK_ZXR:
+		tmp[0] = 0x00880480;
+		tmp[1] = 0x00000080;
+		break;
+	case QUIRK_SBZ:
+		tmp[0] = 0x00820680;
+		tmp[1] = 0x00000083;
+		break;
+	case QUIRK_R3D:
+		tmp[0] = 0x00880680;
+		tmp[1] = 0x00000083;
+		break;
+	default:
+		tmp[0] = 0x00000000;
+		tmp[1] = 0x00000000;
+		break;
+	}
 
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000001, spec->mem_base + 0x40c);
-	else
-		writel(0x00000000, spec->mem_base + 0x40C);
+	for (i = 0; i < 2; i++)
+		writel(tmp[i], spec->mem_base + addr[cur_addr + i]);
 
-	if (ca0132_quirk(spec) == QUIRK_ZXR)
-		writel(0x00880640, spec->mem_base + 0x01C);
-	else
-		writel(0x00880680, spec->mem_base + 0x01C);
+	cur_addr += i;
 
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000080, spec->mem_base + 0xC0C);
-	else
-		writel(0x00000083, spec->mem_base + 0xC0C);
+	switch (ca0132_quirk(spec)) {
+	case QUIRK_ZXR:
+		count = ARRAY_SIZE(ca0113_mmio_init_data_zxr);
+		data = ca0113_mmio_init_data_zxr;
+		break;
+	default:
+		count = ARRAY_SIZE(ca0113_mmio_init_data_sbz);
+		data = ca0113_mmio_init_data_sbz;
+		break;
+	}
 
-	writel(0x00000030, spec->mem_base + 0xC00);
-	writel(0x00000000, spec->mem_base + 0xC04);
+	for (i = 0; i < count; i++)
+		writel(data[i], spec->mem_base + addr[cur_addr + i]);
+}
 
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000000, spec->mem_base + 0xC0C);
-	else
-		writel(0x00000003, spec->mem_base + 0xC0C);
+static void ca0132_mmio_init_ae5(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	const unsigned int *addr, *data;
+	unsigned int i, count;
 
-	writel(0x00000003, spec->mem_base + 0xC0C);
-	writel(0x00000003, spec->mem_base + 0xC0C);
-	writel(0x00000003, spec->mem_base + 0xC0C);
+	addr = ca0113_mmio_init_address_ae5;
+	data = ca0113_mmio_init_data_ae5;
+	count = ARRAY_SIZE(ca0113_mmio_init_data_ae5);
 
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000001, spec->mem_base + 0xC08);
-	else
-		writel(0x000000C1, spec->mem_base + 0xC08);
+	for (i = 0; i < count; i++)
+		writel(data[i], spec->mem_base + addr[i]);
+}
 
-	writel(0x000000F1, spec->mem_base + 0xC08);
-	writel(0x00000001, spec->mem_base + 0xC08);
-	writel(0x000000C7, spec->mem_base + 0xC08);
-	writel(0x000000C1, spec->mem_base + 0xC08);
-	writel(0x00000080, spec->mem_base + 0xC04);
+static void ca0132_mmio_init(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
 
-	if (ca0132_quirk(spec) == QUIRK_AE5) {
-		writel(0x00000000, spec->mem_base + 0x42c);
-		writel(0x00000000, spec->mem_base + 0x46c);
-		writel(0x00000000, spec->mem_base + 0x4ac);
-		writel(0x00000000, spec->mem_base + 0x4ec);
-		writel(0x00000000, spec->mem_base + 0x43c);
-		writel(0x00000000, spec->mem_base + 0x47c);
-		writel(0x00000000, spec->mem_base + 0x4bc);
-		writel(0x00000000, spec->mem_base + 0x4fc);
-		writel(0x00000600, spec->mem_base + 0x100);
-		writel(0x00000014, spec->mem_base + 0x410);
-		writel(0x0000060f, spec->mem_base + 0x100);
-		writel(0x0000070f, spec->mem_base + 0x100);
-		writel(0x00000aff, spec->mem_base + 0x830);
-		writel(0x00000000, spec->mem_base + 0x86c);
-		writel(0x0000006b, spec->mem_base + 0x800);
-		writel(0x00000001, spec->mem_base + 0x86c);
-		writel(0x0000006b, spec->mem_base + 0x800);
-		writel(0x00000057, spec->mem_base + 0x804);
-		writel(0x00800000, spec->mem_base + 0x20c);
+	switch (ca0132_quirk(spec)) {
+	case QUIRK_R3D:
+	case QUIRK_SBZ:
+	case QUIRK_ZXR:
+		ca0132_mmio_init_sbz(codec);
+		break;
+	case QUIRK_AE5:
+		ca0132_mmio_init_ae5(codec);
+		break;
 	}
 }
 
-- 
cgit v1.2.3


From 896e361e82423aed4490f485dc25de1958c724ed Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:21 -0400
Subject: ALSA: hda/ca0132 - Add speaker tuning initialization commands.

Add speaker tuning initialization DSP commands, and also define
previously unknown DSP command values.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-3-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 119 +++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 119 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 7491e2044638..2e664aeee1c4 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -589,6 +589,60 @@ static const struct ct_eq_preset ca0132_alt_eq_presets[] = {
 	}
 };
 
+/*
+ * Definitions for the DSP req's to handle speaker tuning. These all belong to
+ * module ID 0x96, the output effects module.
+ */
+enum speaker_tuning_reqs {
+	/*
+	 * Currently, this value is always set to 0.0f. However, on Windows,
+	 * when selecting certain headphone profiles on the new Sound Blaster
+	 * connect software, the QUERY_SPEAKER_EQ_ADDRESS req on mid 0x80 is
+	 * sent. This gets the speaker EQ address area, which is then used to
+	 * send over (presumably) an equalizer profile for the specific
+	 * headphone setup. It is sent using the same method the DSP
+	 * firmware is uploaded with, which I believe is why the 'ctspeq.bin'
+	 * file exists in linux firmware tree but goes unused. It would also
+	 * explain why the QUERY_SPEAKER_EQ_ADDRESS req is defined but unused.
+	 * Once this profile is sent over, SPEAKER_TUNING_USE_SPEAKER_EQ is
+	 * set to 1.0f.
+	 */
+	SPEAKER_TUNING_USE_SPEAKER_EQ           = 0x1f,
+	SPEAKER_TUNING_ENABLE_CENTER_EQ         = 0x20,
+	SPEAKER_TUNING_FRONT_LEFT_VOL_LEVEL     = 0x21,
+	SPEAKER_TUNING_FRONT_RIGHT_VOL_LEVEL    = 0x22,
+	SPEAKER_TUNING_CENTER_VOL_LEVEL         = 0x23,
+	SPEAKER_TUNING_LFE_VOL_LEVEL            = 0x24,
+	SPEAKER_TUNING_REAR_LEFT_VOL_LEVEL      = 0x25,
+	SPEAKER_TUNING_REAR_RIGHT_VOL_LEVEL     = 0x26,
+	SPEAKER_TUNING_SURROUND_LEFT_VOL_LEVEL  = 0x27,
+	SPEAKER_TUNING_SURROUND_RIGHT_VOL_LEVEL = 0x28,
+	/*
+	 * Inversion is used when setting headphone virtualization to line
+	 * out. Not sure why this is, but it's the only place it's ever used.
+	 */
+	SPEAKER_TUNING_FRONT_LEFT_INVERT        = 0x29,
+	SPEAKER_TUNING_FRONT_RIGHT_INVERT       = 0x2a,
+	SPEAKER_TUNING_CENTER_INVERT            = 0x2b,
+	SPEAKER_TUNING_LFE_INVERT               = 0x2c,
+	SPEAKER_TUNING_REAR_LEFT_INVERT         = 0x2d,
+	SPEAKER_TUNING_REAR_RIGHT_INVERT        = 0x2e,
+	SPEAKER_TUNING_SURROUND_LEFT_INVERT     = 0x2f,
+	SPEAKER_TUNING_SURROUND_RIGHT_INVERT    = 0x30,
+	/* Delay is used when setting surround speaker distance in Windows. */
+	SPEAKER_TUNING_FRONT_LEFT_DELAY         = 0x31,
+	SPEAKER_TUNING_FRONT_RIGHT_DELAY        = 0x32,
+	SPEAKER_TUNING_CENTER_DELAY             = 0x33,
+	SPEAKER_TUNING_LFE_DELAY                = 0x34,
+	SPEAKER_TUNING_REAR_LEFT_DELAY          = 0x35,
+	SPEAKER_TUNING_REAR_RIGHT_DELAY         = 0x36,
+	SPEAKER_TUNING_SURROUND_LEFT_DELAY      = 0x37,
+	SPEAKER_TUNING_SURROUND_RIGHT_DELAY     = 0x38,
+	/* Of these two, only mute seems to ever be used. */
+	SPEAKER_TUNING_MAIN_VOLUME              = 0x39,
+	SPEAKER_TUNING_MUTE                     = 0x3a,
+};
+
 /* DSP command sequences for ca0132_alt_select_out */
 #define ALT_OUT_SET_MAX_COMMANDS 9 /* Max number of commands in sequence */
 struct ca0132_alt_out_set {
@@ -6874,6 +6928,67 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec)
 	}
 }
 
+/*
+ * Default speaker tuning values setup for alternative codecs.
+ */
+static const unsigned int sbz_default_delay_values[] = {
+	/* Non-zero values are floating point 0.000198. */
+	0x394f9e38, 0x394f9e38, 0x00000000, 0x00000000, 0x00000000, 0x00000000
+};
+
+static const unsigned int zxr_default_delay_values[] = {
+	/* Non-zero values are floating point 0.000220. */
+	0x00000000, 0x00000000, 0x3966afcd, 0x3966afcd, 0x3966afcd, 0x3966afcd
+};
+
+static const unsigned int ae5_default_delay_values[] = {
+	/* Non-zero values are floating point 0.000100. */
+	0x00000000, 0x00000000, 0x38d1b717, 0x38d1b717, 0x38d1b717, 0x38d1b717
+};
+
+/*
+ * If we never change these, probably only need them on initialization.
+ */
+static void ca0132_alt_init_speaker_tuning(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int i, tmp, start_req, end_req;
+	const unsigned int *values;
+
+	switch (ca0132_quirk(spec)) {
+	case QUIRK_SBZ:
+		values = sbz_default_delay_values;
+		break;
+	case QUIRK_ZXR:
+		values = zxr_default_delay_values;
+		break;
+	case QUIRK_AE5:
+		values = ae5_default_delay_values;
+		break;
+	default:
+		values = sbz_default_delay_values;
+		break;
+	}
+
+	tmp = FLOAT_ZERO;
+	dspio_set_uint_param(codec, 0x96, SPEAKER_TUNING_ENABLE_CENTER_EQ, tmp);
+
+	start_req = SPEAKER_TUNING_FRONT_LEFT_VOL_LEVEL;
+	end_req = SPEAKER_TUNING_REAR_RIGHT_VOL_LEVEL;
+	for (i = start_req; i < end_req + 1; i++)
+		dspio_set_uint_param(codec, 0x96, i, tmp);
+
+	start_req = SPEAKER_TUNING_FRONT_LEFT_INVERT;
+	end_req = SPEAKER_TUNING_REAR_RIGHT_INVERT;
+	for (i = start_req; i < end_req + 1; i++)
+		dspio_set_uint_param(codec, 0x96, i, tmp);
+
+
+	for (i = 0; i < 6; i++)
+		dspio_set_uint_param(codec, 0x96,
+				SPEAKER_TUNING_FRONT_LEFT_DELAY + i, values[i]);
+}
+
 /*
  * Creates a dummy stream to bind the output to. This seems to have to be done
  * after changing the main outputs source and destination streams.
@@ -7373,6 +7488,8 @@ static void sbz_setup_defaults(struct hda_codec *codec)
 		}
 	}
 
+	ca0132_alt_init_speaker_tuning(codec);
+
 	ca0132_alt_create_dummy_stream(codec);
 }
 
@@ -7440,6 +7557,8 @@ static void ae5_setup_defaults(struct hda_codec *codec)
 		}
 	}
 
+	ca0132_alt_init_speaker_tuning(codec);
+
 	ca0132_alt_create_dummy_stream(codec);
 }
 
-- 
cgit v1.2.3


From 01464a566eed527dbf977e2430afa249c257c008 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:22 -0400
Subject: ALSA: hda/ca0132 - Add surround channel config control.

Add a surround channel configuration enumeration control. Setting up
different channel configurations allows the DSP to upmix stereo audio
into multi-channel audio, and allows for redirection of bass to a
subwoofer.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-4-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 110 ++++++++++++++++++++++++++++++++++++++++++-
 1 file changed, 108 insertions(+), 2 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 2e664aeee1c4..1a5fb30b69ab 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -38,6 +38,8 @@
 #define FLOAT_ONE	0x3f800000
 #define FLOAT_TWO	0x40000000
 #define FLOAT_THREE     0x40400000
+#define FLOAT_FIVE	0x40a00000
+#define FLOAT_SIX       0x40c00000
 #define FLOAT_EIGHT     0x41000000
 #define FLOAT_MINUS_5	0xc0a00000
 
@@ -143,7 +145,8 @@ enum {
 	MIC_BOOST_ENUM,
 	AE5_HEADPHONE_GAIN_ENUM,
 	AE5_SOUND_FILTER_ENUM,
-	ZXR_HEADPHONE_GAIN
+	ZXR_HEADPHONE_GAIN,
+	SPEAKER_CHANNEL_CFG_ENUM,
 #define EFFECTS_COUNT  (EFFECT_END_NID - EFFECT_START_NID)
 };
 
@@ -686,6 +689,39 @@ static const struct ca0132_alt_out_set alt_out_presets[] = {
 	}
 };
 
+/* Surround output channel count configuration structures. */
+#define SPEAKER_CHANNEL_CFG_COUNT 5
+enum {
+	SPEAKER_CHANNELS_2_0,
+	SPEAKER_CHANNELS_2_1,
+	SPEAKER_CHANNELS_4_0,
+	SPEAKER_CHANNELS_4_1,
+	SPEAKER_CHANNELS_5_1,
+};
+
+struct ca0132_alt_speaker_channel_cfg {
+	char *name;
+	unsigned int val;
+};
+
+static const struct ca0132_alt_speaker_channel_cfg speaker_channel_cfgs[] = {
+	{ .name = "2.0",
+	  .val = FLOAT_ONE
+	},
+	{ .name = "2.1",
+	  .val = FLOAT_TWO
+	},
+	{ .name = "4.0",
+	  .val = FLOAT_FIVE
+	},
+	{ .name = "4.1",
+	  .val = FLOAT_SIX
+	},
+	{ .name = "5.1",
+	  .val = FLOAT_EIGHT
+	}
+};
+
 /*
  * DSP volume setting structs. Req 1 is left volume, req 2 is right volume,
  * and I don't know what the third req is, but it's always zero. I assume it's
@@ -1063,6 +1099,7 @@ struct ca0132_spec {
 	/* ca0132_alt control related values */
 	unsigned char in_enum_val;
 	unsigned char out_enum_val;
+	unsigned char channel_cfg_val;
 	unsigned char mic_boost_enum_val;
 	unsigned char smart_volume_setting;
 	long fx_ctl_val[EFFECT_LEVEL_SLIDERS];
@@ -4476,7 +4513,8 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		snd_hda_set_pin_ctl(codec, spec->out_pins[3],
 				    pin_ctl | PIN_OUT);
 
-		dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
+		tmp = speaker_channel_cfgs[spec->channel_cfg_val].val;
+		dspio_set_uint_param(codec, 0x80, 0x04, tmp);
 		break;
 	}
 	/*
@@ -5582,6 +5620,54 @@ static int ca0132_alt_output_select_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
+/* Select surround output type: 2.1, 4.0, 4.1, or 5.1. */
+static int ca0132_alt_speaker_channel_cfg_get_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	unsigned int items = SPEAKER_CHANNEL_CFG_COUNT;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = items;
+	if (uinfo->value.enumerated.item >= items)
+		uinfo->value.enumerated.item = items - 1;
+	strcpy(uinfo->value.enumerated.name,
+			speaker_channel_cfgs[uinfo->value.enumerated.item].name);
+	return 0;
+}
+
+static int ca0132_alt_speaker_channel_cfg_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+
+	ucontrol->value.enumerated.item[0] = spec->channel_cfg_val;
+	return 0;
+}
+
+static int ca0132_alt_speaker_channel_cfg_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+	int sel = ucontrol->value.enumerated.item[0];
+	unsigned int items = SPEAKER_CHANNEL_CFG_COUNT;
+
+	if (sel >= items)
+		return 0;
+
+	codec_dbg(codec, "ca0132_alt_speaker_channels: sel=%d, channels=%s\n",
+		    sel, speaker_channel_cfgs[sel].name);
+
+	spec->channel_cfg_val = sel;
+
+	if (spec->out_enum_val == SURROUND_OUT)
+		ca0132_alt_select_out(codec);
+
+	return 1;
+}
+
 /*
  * Smart Volume output setting control. Three different settings, Normal,
  * which takes the value from the smart volume slider. The two others, loud
@@ -6226,6 +6312,23 @@ static int ca0132_alt_add_output_enum(struct hda_codec *codec)
 				snd_ctl_new1(&knew, codec));
 }
 
+/*
+ * Add a control for selecting channel count on speaker output. Setting this
+ * allows the DSP to do bass redirection and channel upmixing on surround
+ * configurations.
+ */
+static int ca0132_alt_add_speaker_channel_cfg_enum(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_MUTE_MONO("Surround Channel Config",
+				    SPEAKER_CHANNEL_CFG_ENUM, 1, 0, HDA_OUTPUT);
+	knew.info = ca0132_alt_speaker_channel_cfg_get_info;
+	knew.get = ca0132_alt_speaker_channel_cfg_get;
+	knew.put = ca0132_alt_speaker_channel_cfg_put;
+	return snd_hda_ctl_add(codec, SPEAKER_CHANNEL_CFG_ENUM,
+				snd_ctl_new1(&knew, codec));
+}
+
 /*
  * Create an Input Source enumerated control for the alternate ca0132 codecs
  * because the front microphone has no auto-detect, and Line-in has to be set
@@ -6530,6 +6633,9 @@ static int ca0132_build_controls(struct hda_codec *codec)
 	 */
 	if (ca0132_use_alt_functions(spec)) {
 		err = ca0132_alt_add_output_enum(codec);
+		if (err < 0)
+			return err;
+		err = ca0132_alt_add_speaker_channel_cfg_enum(codec);
 		if (err < 0)
 			return err;
 		err = ca0132_alt_add_mic_boost_enum(codec);
-- 
cgit v1.2.3


From 670c5f484a4403e5c218442d5a0e337a2224741f Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:23 -0400
Subject: ALSA: hda/ca0132 - Add full-range speaker selection controls.

Add functions for setting full-range speakers and controls to
enable/disable the setting. Setting a speaker to full-range means that
the channels won't have their bass redirected to the LFE channel.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-5-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 117 +++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 117 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 1a5fb30b69ab..469cefe9a51a 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -147,6 +147,8 @@ enum {
 	AE5_SOUND_FILTER_ENUM,
 	ZXR_HEADPHONE_GAIN,
 	SPEAKER_CHANNEL_CFG_ENUM,
+	SPEAKER_FULL_RANGE_FRONT,
+	SPEAKER_FULL_RANGE_REAR,
 #define EFFECTS_COUNT  (EFFECT_END_NID - EFFECT_START_NID)
 };
 
@@ -592,6 +594,24 @@ static const struct ct_eq_preset ca0132_alt_eq_presets[] = {
 	}
 };
 
+/*
+ * DSP reqs for handling full-range speakers/bass redirection. If a speaker is
+ * set as not being full range, and bass redirection is enabled, all
+ * frequencies below the crossover frequency are redirected to the LFE
+ * channel. If the surround configuration has no LFE channel, this can't be
+ * enabled. X-Bass must be disabled when using these.
+ */
+enum speaker_range_reqs {
+	SPEAKER_BASS_REDIRECT            = 0x15,
+	SPEAKER_BASS_REDIRECT_XOVER_FREQ = 0x16,
+	/* Between 0x16-0x1a are the X-Bass reqs. */
+	SPEAKER_FULL_RANGE_FRONT_L_R     = 0x1a,
+	SPEAKER_FULL_RANGE_CENTER_LFE    = 0x1b,
+	SPEAKER_FULL_RANGE_REAR_L_R      = 0x1c,
+	SPEAKER_FULL_RANGE_SURROUND_L_R  = 0x1d,
+	SPEAKER_BASS_REDIRECT_SUB_GAIN   = 0x1e,
+};
+
 /*
  * Definitions for the DSP req's to handle speaker tuning. These all belong to
  * module ID 0x96, the output effects module.
@@ -1100,6 +1120,7 @@ struct ca0132_spec {
 	unsigned char in_enum_val;
 	unsigned char out_enum_val;
 	unsigned char channel_cfg_val;
+	unsigned char speaker_range_val[2];
 	unsigned char mic_boost_enum_val;
 	unsigned char smart_volume_setting;
 	long fx_ctl_val[EFFECT_LEVEL_SLIDERS];
@@ -4259,6 +4280,50 @@ static void ae5_mmio_select_out(struct hda_codec *codec)
 			ae5_ca0113_output_presets[spec->cur_out_type].vals[i]);
 }
 
+static int ca0132_alt_set_full_range_speaker(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp;
+	int err;
+
+	/* 2.0/4.0 setup has no LFE channel, so setting full-range does nothing. */
+	if (spec->channel_cfg_val == SPEAKER_CHANNELS_4_0
+			|| spec->channel_cfg_val == SPEAKER_CHANNELS_2_0)
+		return 0;
+
+	/* Set front L/R full range. Zero for full-range, one for redirection. */
+	tmp = spec->speaker_range_val[0] ? FLOAT_ZERO : FLOAT_ONE;
+	err = dspio_set_uint_param(codec, 0x96,
+			SPEAKER_FULL_RANGE_FRONT_L_R, tmp);
+	if (err < 0)
+		return err;
+
+	/* When setting full-range rear, both rear and center/lfe are set. */
+	tmp = spec->speaker_range_val[1] ? FLOAT_ZERO : FLOAT_ONE;
+	err = dspio_set_uint_param(codec, 0x96,
+			SPEAKER_FULL_RANGE_CENTER_LFE, tmp);
+	if (err < 0)
+		return err;
+
+	err = dspio_set_uint_param(codec, 0x96,
+			SPEAKER_FULL_RANGE_REAR_L_R, tmp);
+	if (err < 0)
+		return err;
+
+	/*
+	 * Only the AE series cards set this value when setting full-range,
+	 * and it's always 1.0f.
+	 */
+	if (ca0132_quirk(spec) == QUIRK_AE5) {
+		err = dspio_set_uint_param(codec, 0x96,
+				SPEAKER_FULL_RANGE_SURROUND_L_R, FLOAT_ONE);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
 /*
  * These are the commands needed to setup output on each of the different card
  * types.
@@ -4539,6 +4604,9 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 			goto exit;
 	}
 
+	if (spec->cur_out_type == SURROUND_OUT)
+		err = ca0132_alt_set_full_range_speaker(codec);
+
 exit:
 	snd_hda_power_down_pm(codec);
 
@@ -5269,6 +5337,7 @@ static int ca0132_alt_xbass_xover_slider_ctl_get(struct snd_kcontrol *kcontrol,
 	long *valp = ucontrol->value.integer.value;
 
 	*valp = spec->xbass_xover_freq;
+
 	return 0;
 }
 
@@ -5894,6 +5963,11 @@ static int ca0132_switch_get(struct snd_kcontrol *kcontrol,
 		return 0;
 	}
 
+	if (nid == SPEAKER_FULL_RANGE_FRONT || nid == SPEAKER_FULL_RANGE_REAR) {
+		*valp = spec->speaker_range_val[nid - SPEAKER_FULL_RANGE_FRONT];
+		return 0;
+	}
+
 	return 0;
 }
 
@@ -5972,6 +6046,14 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
 		goto exit;
 	}
 
+	if (nid == SPEAKER_FULL_RANGE_FRONT || nid == SPEAKER_FULL_RANGE_REAR) {
+		spec->speaker_range_val[nid - SPEAKER_FULL_RANGE_FRONT] = *valp;
+		if (spec->cur_out_type == SURROUND_OUT)
+			ca0132_alt_set_full_range_speaker(codec);
+
+		changed = 0;
+	}
+
 exit:
 	snd_hda_power_down(codec);
 	return changed;
@@ -6329,6 +6411,31 @@ static int ca0132_alt_add_speaker_channel_cfg_enum(struct hda_codec *codec)
 				snd_ctl_new1(&knew, codec));
 }
 
+/*
+ * Full range front stereo and rear surround switches. When these are set to
+ * full range, the lower frequencies from these channels are no longer
+ * redirected to the LFE channel.
+ */
+static int ca0132_alt_add_front_full_range_switch(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		CA0132_CODEC_MUTE_MONO("Full-Range Front Speakers",
+				    SPEAKER_FULL_RANGE_FRONT, 1, HDA_OUTPUT);
+
+	return snd_hda_ctl_add(codec, SPEAKER_FULL_RANGE_FRONT,
+				snd_ctl_new1(&knew, codec));
+}
+
+static int ca0132_alt_add_rear_full_range_switch(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		CA0132_CODEC_MUTE_MONO("Full-Range Rear Speakers",
+				    SPEAKER_FULL_RANGE_REAR, 1, HDA_OUTPUT);
+
+	return snd_hda_ctl_add(codec, SPEAKER_FULL_RANGE_REAR,
+				snd_ctl_new1(&knew, codec));
+}
+
 /*
  * Create an Input Source enumerated control for the alternate ca0132 codecs
  * because the front microphone has no auto-detect, and Line-in has to be set
@@ -6636,6 +6743,12 @@ static int ca0132_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 		err = ca0132_alt_add_speaker_channel_cfg_enum(codec);
+		if (err < 0)
+			return err;
+		err = ca0132_alt_add_front_full_range_switch(codec);
+		if (err < 0)
+			return err;
+		err = ca0132_alt_add_rear_full_range_switch(codec);
 		if (err < 0)
 			return err;
 		err = ca0132_alt_add_mic_boost_enum(codec);
@@ -7982,6 +8095,10 @@ static void ca0132_init_chip(struct hda_codec *codec)
 	 * ca0132 codecs. Also sets x-bass crossover frequency to 80hz.
 	 */
 	if (ca0132_use_alt_controls(spec)) {
+		/* Set speakers to default to full range. */
+		spec->speaker_range_val[0] = 1;
+		spec->speaker_range_val[1] = 1;
+
 		spec->xbass_xover_freq = 8;
 		for (i = 0; i < EFFECT_LEVEL_SLIDERS; i++)
 			spec->fx_ctl_val[i] = effect_slider_defaults[i];
-- 
cgit v1.2.3


From f49b3063ad0d8638da2e7eb994549863f2c74923 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:24 -0400
Subject: ALSA: hda/ca0132 - Add bass redirection controls.

Add bass redirection controls for surround outputs. This uses the DSP to
redirect audio below the bass redirection crossover frequency to the LFE
channel from the front/rear L/R speakers. This only goes into effect if
the speakers aren't set as full range, and only if the surround
configuration has an LFE channel.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-6-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 130 +++++++++++++++++++++++++++++++++++++++++--
 1 file changed, 126 insertions(+), 4 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 469cefe9a51a..8ad2fc5ab30b 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -149,6 +149,8 @@ enum {
 	SPEAKER_CHANNEL_CFG_ENUM,
 	SPEAKER_FULL_RANGE_FRONT,
 	SPEAKER_FULL_RANGE_REAR,
+	BASS_REDIRECTION,
+	BASS_REDIRECTION_XOVER,
 #define EFFECTS_COUNT  (EFFECT_END_NID - EFFECT_START_NID)
 };
 
@@ -1123,6 +1125,8 @@ struct ca0132_spec {
 	unsigned char speaker_range_val[2];
 	unsigned char mic_boost_enum_val;
 	unsigned char smart_volume_setting;
+	unsigned char bass_redirection_val;
+	long bass_redirect_xover_freq;
 	long fx_ctl_val[EFFECT_LEVEL_SLIDERS];
 	long xbass_xover_freq;
 	long eq_preset_val;
@@ -4324,6 +4328,35 @@ static int ca0132_alt_set_full_range_speaker(struct hda_codec *codec)
 	return 0;
 }
 
+static int ca0132_alt_surround_set_bass_redirection(struct hda_codec *codec,
+		bool val)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp;
+	int err;
+
+	if (val && spec->channel_cfg_val != SPEAKER_CHANNELS_4_0 &&
+			spec->channel_cfg_val != SPEAKER_CHANNELS_2_0)
+		tmp = FLOAT_ONE;
+	else
+		tmp = FLOAT_ZERO;
+
+	err = dspio_set_uint_param(codec, 0x96, SPEAKER_BASS_REDIRECT, tmp);
+	if (err < 0)
+		return err;
+
+	/* If it is enabled, make sure to set the crossover frequency. */
+	if (tmp) {
+		tmp = float_xbass_xover_lookup[spec->xbass_xover_freq];
+		err = dspio_set_uint_param(codec, 0x96,
+				SPEAKER_BASS_REDIRECT_XOVER_FREQ, tmp);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
 /*
  * These are the commands needed to setup output on each of the different card
  * types.
@@ -4593,6 +4626,15 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		ca0132_effects_set(codec, X_BASS,
 			spec->effects_switch[X_BASS - EFFECT_START_NID]);
 
+	if (spec->cur_out_type == SURROUND_OUT)
+		err = ca0132_alt_surround_set_bass_redirection(codec,
+				spec->bass_redirection_val);
+	else
+		err = ca0132_alt_surround_set_bass_redirection(codec, 0);
+
+	if (err < 0)
+		goto exit;
+
 	/* run through the output dsp commands for the selected output. */
 	for (i = 0; i < alt_out_presets[spec->cur_out_type].commands; i++) {
 		err = dspio_set_uint_param(codec,
@@ -5282,6 +5324,18 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
 	return ret;
 }
 /* End of control change helpers. */
+
+static void ca0132_alt_bass_redirection_xover_set(struct hda_codec *codec,
+		long idx)
+{
+	snd_hda_power_up(codec);
+
+	dspio_set_param(codec, 0x96, 0x20, SPEAKER_BASS_REDIRECT_XOVER_FREQ,
+			&(float_xbass_xover_lookup[idx]), sizeof(unsigned int));
+
+	snd_hda_power_down(codec);
+}
+
 /*
  * Below I've added controls to mess with the effect levels, I've only enabled
  * them on the Sound Blaster Z, but they would probably also work on the
@@ -5290,6 +5344,7 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
  */
 
 /* Sets DSP effect level from the sliders above the controls */
+
 static int ca0132_alt_slider_ctl_set(struct hda_codec *codec, hda_nid_t nid,
 			  const unsigned int *lookup, int idx)
 {
@@ -5335,8 +5390,12 @@ static int ca0132_alt_xbass_xover_slider_ctl_get(struct snd_kcontrol *kcontrol,
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct ca0132_spec *spec = codec->spec;
 	long *valp = ucontrol->value.integer.value;
+	hda_nid_t nid = get_amp_nid(kcontrol);
 
-	*valp = spec->xbass_xover_freq;
+	if (nid == BASS_REDIRECTION_XOVER)
+		*valp = spec->bass_redirect_xover_freq;
+	else
+		*valp = spec->xbass_xover_freq;
 
 	return 0;
 }
@@ -5391,16 +5450,25 @@ static int ca0132_alt_xbass_xover_slider_put(struct snd_kcontrol *kcontrol,
 	struct ca0132_spec *spec = codec->spec;
 	hda_nid_t nid = get_amp_nid(kcontrol);
 	long *valp = ucontrol->value.integer.value;
+	long *cur_val;
 	int idx;
 
+	if (nid == BASS_REDIRECTION_XOVER)
+		cur_val = &spec->bass_redirect_xover_freq;
+	else
+		cur_val = &spec->xbass_xover_freq;
+
 	/* any change? */
-	if (spec->xbass_xover_freq == *valp)
+	if (*cur_val == *valp)
 		return 0;
 
-	spec->xbass_xover_freq = *valp;
+	*cur_val = *valp;
 
 	idx = *valp;
-	ca0132_alt_slider_ctl_set(codec, nid, float_xbass_xover_lookup, idx);
+	if (nid == BASS_REDIRECTION_XOVER)
+		ca0132_alt_bass_redirection_xover_set(codec, *cur_val);
+	else
+		ca0132_alt_slider_ctl_set(codec, nid, float_xbass_xover_lookup, idx);
 
 	return 0;
 }
@@ -5968,6 +6036,11 @@ static int ca0132_switch_get(struct snd_kcontrol *kcontrol,
 		return 0;
 	}
 
+	if (nid == BASS_REDIRECTION) {
+		*valp = spec->bass_redirection_val;
+		return 0;
+	}
+
 	return 0;
 }
 
@@ -6054,6 +6127,14 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
 		changed = 0;
 	}
 
+	if (nid == BASS_REDIRECTION) {
+		spec->bass_redirection_val = *valp;
+		if (spec->cur_out_type == SURROUND_OUT)
+			ca0132_alt_surround_set_bass_redirection(codec, *valp);
+
+		changed = 0;
+	}
+
 exit:
 	snd_hda_power_down(codec);
 	return changed;
@@ -6436,6 +6517,39 @@ static int ca0132_alt_add_rear_full_range_switch(struct hda_codec *codec)
 				snd_ctl_new1(&knew, codec));
 }
 
+/*
+ * Bass redirection redirects audio below the crossover frequency to the LFE
+ * channel on speakers that are set as not being full-range. On configurations
+ * without an LFE channel, it does nothing. Bass redirection seems to be the
+ * replacement for X-Bass on configurations with an LFE channel.
+ */
+static int ca0132_alt_add_bass_redirection_crossover(struct hda_codec *codec)
+{
+	const char *namestr = "Bass Redirection Crossover";
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_VOLUME_MONO(namestr, BASS_REDIRECTION_XOVER, 1, 0,
+				HDA_OUTPUT);
+
+	knew.tlv.c = NULL;
+	knew.info = ca0132_alt_xbass_xover_slider_info;
+	knew.get = ca0132_alt_xbass_xover_slider_ctl_get;
+	knew.put = ca0132_alt_xbass_xover_slider_put;
+
+	return snd_hda_ctl_add(codec, BASS_REDIRECTION_XOVER,
+			snd_ctl_new1(&knew, codec));
+}
+
+static int ca0132_alt_add_bass_redirection_switch(struct hda_codec *codec)
+{
+	const char *namestr = "Bass Redirection";
+	struct snd_kcontrol_new knew =
+		CA0132_CODEC_MUTE_MONO(namestr, BASS_REDIRECTION, 1,
+				HDA_OUTPUT);
+
+	return snd_hda_ctl_add(codec, BASS_REDIRECTION,
+			snd_ctl_new1(&knew, codec));
+}
+
 /*
  * Create an Input Source enumerated control for the alternate ca0132 codecs
  * because the front microphone has no auto-detect, and Line-in has to be set
@@ -6749,6 +6863,12 @@ static int ca0132_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 		err = ca0132_alt_add_rear_full_range_switch(codec);
+		if (err < 0)
+			return err;
+		err = ca0132_alt_add_bass_redirection_crossover(codec);
+		if (err < 0)
+			return err;
+		err = ca0132_alt_add_bass_redirection_switch(codec);
 		if (err < 0)
 			return err;
 		err = ca0132_alt_add_mic_boost_enum(codec);
@@ -8102,6 +8222,8 @@ static void ca0132_init_chip(struct hda_codec *codec)
 		spec->xbass_xover_freq = 8;
 		for (i = 0; i < EFFECT_LEVEL_SLIDERS; i++)
 			spec->fx_ctl_val[i] = effect_slider_defaults[i];
+
+		spec->bass_redirect_xover_freq = 8;
 	}
 
 	spec->voicefx_val = 0;
-- 
cgit v1.2.3


From ed8156c86f958744ba2088d57b1959b86fb32344 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:25 -0400
Subject: ALSA: hda/ca0132 - Remove surround output selection.

Remove the surround output selection and merge it with the speaker
output selection. Now that the extra commands that were being run on
surround output setting are known, there's no need to have it be
separate.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-7-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 151 +++++++++++++------------------------------
 1 file changed, 46 insertions(+), 105 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 8ad2fc5ab30b..5743bdd7cc88 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -82,11 +82,10 @@ MODULE_FIRMWARE(R3DI_EFX_FILE);
 
 static const char *const dirstr[2] = { "Playback", "Capture" };
 
-#define NUM_OF_OUTPUTS 3
+#define NUM_OF_OUTPUTS 2
 enum {
 	SPEAKER_OUT,
 	HEADPHONE_OUT,
-	SURROUND_OUT
 };
 
 enum {
@@ -699,16 +698,6 @@ static const struct ca0132_alt_out_set alt_out_presets[] = {
 		    0x00000000, 0x00000000, 0x00000000,
 		    0x00000000 }
 	},
-	{ .name = "Surround",
-	  .commands = 8,
-	  .mids = { 0x96, 0x8F, 0x96, 0x96,
-		    0x96, 0x96, 0x96, 0x96 },
-	  .reqs = { 0x18, 0x01, 0x1F, 0x15,
-		    0x3A, 0x1A, 0x1B, 0x1C },
-	  .vals = { 0x00000000, 0x00000000, 0x00000000,
-		    0x00000000, 0x00000000, 0x00000000,
-		    0x00000000, 0x00000000 }
-	}
 };
 
 /* Surround output channel count configuration structures. */
@@ -785,10 +774,6 @@ static const struct ae5_ca0113_output_set ae5_ca0113_output_presets[] = {
 	{ .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
 	  .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
 	  .vals =   { 0x3f, 0x3f, 0x00, 0x00, 0x00, 0x00 }
-	},
-	{ .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
-	  .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
-	  .vals =   { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f }
 	}
 };
 
@@ -4440,42 +4425,6 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec)
 			break;
 		}
 		break;
-	case SURROUND_OUT:
-		switch (ca0132_quirk(spec)) {
-		case QUIRK_SBZ:
-			ca0113_mmio_gpio_set(codec, 7, false);
-			ca0113_mmio_gpio_set(codec, 4, true);
-			ca0113_mmio_gpio_set(codec, 1, true);
-			chipio_set_control_param(codec, 0x0d, 0x18);
-			break;
-		case QUIRK_ZXR:
-			ca0113_mmio_gpio_set(codec, 2, true);
-			ca0113_mmio_gpio_set(codec, 3, true);
-			ca0113_mmio_gpio_set(codec, 5, false);
-			zxr_headphone_gain_set(codec, 0);
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			break;
-		case QUIRK_R3DI:
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			r3di_gpio_out_set(codec, R3DI_LINE_OUT);
-			break;
-		case QUIRK_R3D:
-			ca0113_mmio_gpio_set(codec, 1, true);
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			break;
-		case QUIRK_AE5:
-			ae5_mmio_select_out(codec);
-			ae5_headphone_gain_set(codec, 2);
-			tmp = FLOAT_ZERO;
-			dspio_set_uint_param(codec, 0x96, 0x29, tmp);
-			dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
-			chipio_set_control_param(codec, 0x0d, 0xa4);
-			chipio_write(codec, 0x18b03c, 0x00000012);
-			break;
-		default:
-			break;
-		}
-		break;
 	}
 }
 
@@ -4492,11 +4441,10 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec)
 static int ca0132_alt_select_out(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp, outfx_set, i;
 	unsigned int pin_ctl;
 	int jack_present;
 	int auto_jack;
-	unsigned int i;
-	unsigned int tmp;
 	int err;
 	/* Default Headphone is rear headphone */
 	hda_nid_t headphone_nid = spec->out_pins[1];
@@ -4523,6 +4471,8 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 	} else
 		spec->cur_out_type = spec->out_enum_val;
 
+	outfx_set = spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID];
+
 	/* Begin DSP output switch */
 	tmp = FLOAT_ONE;
 	err = dspio_set_uint_param(codec, 0x96, 0x3A, tmp);
@@ -4536,6 +4486,7 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		codec_dbg(codec, "%s speaker\n", __func__);
 
 		/* disable headphone node */
+
 		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0,
 					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
 		snd_hda_set_pin_ctl(codec, spec->out_pins[1],
@@ -4549,15 +4500,34 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		snd_hda_codec_write(codec, spec->out_pins[0], 0,
 			AC_VERB_SET_EAPD_BTLENABLE, 0x01);
 
-		/* If PlayEnhancement is enabled, set different source */
-		if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
-			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
+		/* enable center/lfe out node */
+		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[2], 0,
+				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+		snd_hda_set_pin_ctl(codec, spec->out_pins[2],
+				pin_ctl | PIN_OUT);
+		/* Now set rear surround node as out. */
+		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[3], 0,
+				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+		snd_hda_set_pin_ctl(codec, spec->out_pins[3],
+				pin_ctl | PIN_OUT);
+
+		/*
+		 * Without PlayEnhancement being enabled, if we've got a 2.0
+		 * setup, set it to floating point eight to disable any DSP
+		 * processing effects.
+		 */
+		if (!outfx_set && spec->channel_cfg_val == SPEAKER_CHANNELS_2_0)
+			tmp = FLOAT_EIGHT;
 		else
-			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
+			tmp = speaker_channel_cfgs[spec->channel_cfg_val].val;
+
+		err = dspio_set_uint_param(codec, 0x80, 0x04, tmp);
+		if (err < 0)
+			goto exit;
+
 		break;
 	case HEADPHONE_OUT:
 		codec_dbg(codec, "%s hp\n", __func__);
-
 		snd_hda_codec_write(codec, spec->out_pins[0], 0,
 			AC_VERB_SET_EAPD_BTLENABLE, 0x00);
 
@@ -4568,7 +4538,6 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 				pin_ctl & ~PIN_HP);
 
 		/* enable headphone, either front or rear */
-
 		if (snd_hda_jack_detect(codec, spec->unsol_tag_front_hp))
 			headphone_nid = spec->out_pins[2];
 		else if (snd_hda_jack_detect(codec, spec->unsol_tag_hp))
@@ -4579,54 +4548,22 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		snd_hda_set_pin_ctl(codec, headphone_nid,
 				    pin_ctl | PIN_HP);
 
-		if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
+		if (outfx_set)
 			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
 		else
 			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ZERO);
 		break;
-	case SURROUND_OUT:
-		codec_dbg(codec, "%s surround\n", __func__);
-
-		/* enable line out node */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0,
-				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[0],
-						pin_ctl | PIN_OUT);
-		/* Disable headphone out */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[1],
-				    pin_ctl & ~PIN_HP);
-		/* Enable EAPD on line out */
-		snd_hda_codec_write(codec, spec->out_pins[0], 0,
-			AC_VERB_SET_EAPD_BTLENABLE, 0x01);
-		/* enable center/lfe out node */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[2], 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[2],
-				    pin_ctl | PIN_OUT);
-		/* Now set rear surround node as out. */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[3], 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[3],
-				    pin_ctl | PIN_OUT);
-
-		tmp = speaker_channel_cfgs[spec->channel_cfg_val].val;
-		dspio_set_uint_param(codec, 0x80, 0x04, tmp);
-		break;
 	}
 	/*
-	 * Surround always sets it's scp command to req 0x04 to FLOAT_EIGHT.
-	 * With this set though, X_BASS cannot be enabled. So, if we have OutFX
-	 * enabled, we need to make sure X_BASS is off, otherwise everything
-	 * sounds all muffled. Running ca0132_effects_set with X_BASS as the
-	 * effect should sort this out.
+	 * If output effects are enabled, set the X-Bass effect value again to
+	 * make sure that it's properly enabled/disabled for speaker
+	 * configurations with an LFE channel.
 	 */
-	if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
+	if (outfx_set)
 		ca0132_effects_set(codec, X_BASS,
 			spec->effects_switch[X_BASS - EFFECT_START_NID]);
 
-	if (spec->cur_out_type == SURROUND_OUT)
+	if (spec->cur_out_type == SPEAKER_OUT)
 		err = ca0132_alt_surround_set_bass_redirection(codec,
 				spec->bass_redirection_val);
 	else
@@ -4646,7 +4583,7 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 			goto exit;
 	}
 
-	if (spec->cur_out_type == SURROUND_OUT)
+	if (spec->cur_out_type == SPEAKER_OUT)
 		err = ca0132_alt_set_full_range_speaker(codec);
 
 exit:
@@ -5054,7 +4991,7 @@ static int ca0132_voicefx_set(struct hda_codec *codec, int enable)
 static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
 {
 	struct ca0132_spec *spec = codec->spec;
-	unsigned int on, tmp;
+	unsigned int on, tmp, channel_cfg;
 	int num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT;
 	int err = 0;
 	int idx = nid - EFFECT_START_NID;
@@ -5067,8 +5004,12 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
 		/* if PE if off, turn off out effects. */
 		if (!spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
 			val = 0;
-		if (spec->cur_out_type == SURROUND_OUT && nid == X_BASS)
-			val = 0;
+		if (spec->cur_out_type == SPEAKER_OUT && nid == X_BASS) {
+			channel_cfg = spec->channel_cfg_val;
+			if (channel_cfg != SPEAKER_CHANNELS_2_0 &&
+					channel_cfg != SPEAKER_CHANNELS_4_0)
+				val = 0;
+		}
 	}
 
 	/* for in effect, qualify with CrystalVoice */
@@ -5799,7 +5740,7 @@ static int ca0132_alt_speaker_channel_cfg_put(struct snd_kcontrol *kcontrol,
 
 	spec->channel_cfg_val = sel;
 
-	if (spec->out_enum_val == SURROUND_OUT)
+	if (spec->out_enum_val == SPEAKER_OUT)
 		ca0132_alt_select_out(codec);
 
 	return 1;
@@ -6121,7 +6062,7 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
 
 	if (nid == SPEAKER_FULL_RANGE_FRONT || nid == SPEAKER_FULL_RANGE_REAR) {
 		spec->speaker_range_val[nid - SPEAKER_FULL_RANGE_FRONT] = *valp;
-		if (spec->cur_out_type == SURROUND_OUT)
+		if (spec->cur_out_type == SPEAKER_OUT)
 			ca0132_alt_set_full_range_speaker(codec);
 
 		changed = 0;
@@ -6129,7 +6070,7 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
 
 	if (nid == BASS_REDIRECTION) {
 		spec->bass_redirection_val = *valp;
-		if (spec->cur_out_type == SURROUND_OUT)
+		if (spec->cur_out_type == SPEAKER_OUT)
 			ca0132_alt_surround_set_bass_redirection(codec, *valp);
 
 		changed = 0;
-- 
cgit v1.2.3


From 8e00dc7cedb37c43a2eeb2147d67409ae2e8abf5 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:26 -0400
Subject: ALSA: hda/ca0132 - Clean up ca0132_alt_out_select.

Remove the output structures that were in use before and instead set the
DSP commands line by line. Now that the commands use is known, it makes
the functionality more clear this way.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-8-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 142 +++++++++++++++++--------------------------
 1 file changed, 57 insertions(+), 85 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 5743bdd7cc88..39e333866be3 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -83,6 +83,7 @@ MODULE_FIRMWARE(R3DI_EFX_FILE);
 static const char *const dirstr[2] = { "Playback", "Capture" };
 
 #define NUM_OF_OUTPUTS 2
+static const char *const out_type_str[2] = { "Speakers", "Headphone" };
 enum {
 	SPEAKER_OUT,
 	HEADPHONE_OUT,
@@ -667,39 +668,6 @@ enum speaker_tuning_reqs {
 	SPEAKER_TUNING_MUTE                     = 0x3a,
 };
 
-/* DSP command sequences for ca0132_alt_select_out */
-#define ALT_OUT_SET_MAX_COMMANDS 9 /* Max number of commands in sequence */
-struct ca0132_alt_out_set {
-	char *name; /*preset name*/
-	unsigned char commands;
-	unsigned int mids[ALT_OUT_SET_MAX_COMMANDS];
-	unsigned int reqs[ALT_OUT_SET_MAX_COMMANDS];
-	unsigned int vals[ALT_OUT_SET_MAX_COMMANDS];
-};
-
-static const struct ca0132_alt_out_set alt_out_presets[] = {
-	{ .name = "Line Out",
-	  .commands = 7,
-	  .mids = { 0x96, 0x96, 0x96, 0x8F,
-		    0x96, 0x96, 0x96 },
-	  .reqs = { 0x19, 0x17, 0x18, 0x01,
-		    0x1F, 0x15, 0x3A },
-	  .vals = { 0x3F000000, 0x42A00000, 0x00000000,
-		    0x00000000, 0x00000000, 0x00000000,
-		    0x00000000 }
-	},
-	{ .name = "Headphone",
-	  .commands = 7,
-	  .mids = { 0x96, 0x96, 0x96, 0x8F,
-		    0x96, 0x96, 0x96 },
-	  .reqs = { 0x19, 0x17, 0x18, 0x01,
-		    0x1F, 0x15, 0x3A },
-	  .vals = { 0x3F000000, 0x42A00000, 0x00000000,
-		    0x00000000, 0x00000000, 0x00000000,
-		    0x00000000 }
-	},
-};
-
 /* Surround output channel count configuration structures. */
 #define SPEAKER_CHANNEL_CFG_COUNT 5
 enum {
@@ -4428,21 +4396,31 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec)
 	}
 }
 
+static void ca0132_set_out_node_pincfg(struct hda_codec *codec, hda_nid_t nid,
+		bool out_enable, bool hp_enable)
+{
+	unsigned int pin_ctl;
+
+	pin_ctl = snd_hda_codec_read(codec, nid, 0,
+			AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+
+	pin_ctl = hp_enable ? pin_ctl | PIN_HP_AMP : pin_ctl & ~PIN_HP_AMP;
+	pin_ctl = out_enable ? pin_ctl | PIN_OUT : pin_ctl & ~PIN_OUT;
+	snd_hda_set_pin_ctl(codec, nid, pin_ctl);
+}
+
 /*
  * This function behaves similarly to the ca0132_select_out funciton above,
  * except with a few differences. It adds the ability to select the current
  * output with an enumerated control "output source" if the auto detect
  * mute switch is set to off. If the auto detect mute switch is enabled, it
  * will detect either headphone or lineout(SPEAKER_OUT) from jack detection.
- * It also adds the ability to auto-detect the front headphone port. The only
- * way to select surround is to disable auto detect, and set Surround with the
- * enumerated control.
+ * It also adds the ability to auto-detect the front headphone port.
  */
 static int ca0132_alt_select_out(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
-	unsigned int tmp, outfx_set, i;
-	unsigned int pin_ctl;
+	unsigned int tmp, outfx_set;
 	int jack_present;
 	int auto_jack;
 	int err;
@@ -4473,9 +4451,8 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 
 	outfx_set = spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID];
 
-	/* Begin DSP output switch */
-	tmp = FLOAT_ONE;
-	err = dspio_set_uint_param(codec, 0x96, 0x3A, tmp);
+	/* Begin DSP output switch, mute DSP volume. */
+	err = dspio_set_uint_param(codec, 0x96, SPEAKER_TUNING_MUTE, FLOAT_ONE);
 	if (err < 0)
 		goto exit;
 
@@ -4485,31 +4462,18 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 	case SPEAKER_OUT:
 		codec_dbg(codec, "%s speaker\n", __func__);
 
-		/* disable headphone node */
-
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[1],
-				    pin_ctl & ~PIN_HP);
-		/* enable line-out node */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0,
-				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[0],
-				    pin_ctl | PIN_OUT);
 		/* Enable EAPD */
 		snd_hda_codec_write(codec, spec->out_pins[0], 0,
 			AC_VERB_SET_EAPD_BTLENABLE, 0x01);
 
-		/* enable center/lfe out node */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[2], 0,
-				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[2],
-				pin_ctl | PIN_OUT);
-		/* Now set rear surround node as out. */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[3], 0,
-				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[3],
-				pin_ctl | PIN_OUT);
+		/* Disable headphone node. */
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[1], 0, 0);
+		/* Set front L-R to output. */
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[0], 1, 0);
+		/* Set Center/LFE to output. */
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[2], 1, 0);
+		/* Set rear surround to output. */
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[3], 1, 0);
 
 		/*
 		 * Without PlayEnhancement being enabled, if we've got a 2.0
@@ -4531,11 +4495,10 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		snd_hda_codec_write(codec, spec->out_pins[0], 0,
 			AC_VERB_SET_EAPD_BTLENABLE, 0x00);
 
-		/* disable speaker*/
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[0],
-				pin_ctl & ~PIN_HP);
+		/* Disable all speaker nodes. */
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[0], 0, 0);
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[2], 0, 0);
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[3], 0, 0);
 
 		/* enable headphone, either front or rear */
 		if (snd_hda_jack_detect(codec, spec->unsol_tag_front_hp))
@@ -4543,15 +4506,15 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		else if (snd_hda_jack_detect(codec, spec->unsol_tag_hp))
 			headphone_nid = spec->out_pins[1];
 
-		pin_ctl = snd_hda_codec_read(codec, headphone_nid, 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, headphone_nid,
-				    pin_ctl | PIN_HP);
+		ca0132_set_out_node_pincfg(codec, headphone_nid, 1, 1);
 
 		if (outfx_set)
-			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
+			err = dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
 		else
-			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ZERO);
+			err = dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ZERO);
+
+		if (err < 0)
+			goto exit;
 		break;
 	}
 	/*
@@ -4563,29 +4526,38 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 		ca0132_effects_set(codec, X_BASS,
 			spec->effects_switch[X_BASS - EFFECT_START_NID]);
 
+	/* Set speaker EQ bypass attenuation to 0. */
+	err = dspio_set_uint_param(codec, 0x8f, 0x01, FLOAT_ZERO);
+	if (err < 0)
+		goto exit;
+
+	/*
+	 * Although unused on all cards but the AE series, this is always set
+	 * to zero when setting the output.
+	 */
+	err = dspio_set_uint_param(codec, 0x96,
+			SPEAKER_TUNING_USE_SPEAKER_EQ, FLOAT_ZERO);
+	if (err < 0)
+		goto exit;
+
 	if (spec->cur_out_type == SPEAKER_OUT)
 		err = ca0132_alt_surround_set_bass_redirection(codec,
 				spec->bass_redirection_val);
 	else
 		err = ca0132_alt_surround_set_bass_redirection(codec, 0);
 
+	/* Unmute DSP now that we're done with output selection. */
+	err = dspio_set_uint_param(codec, 0x96,
+			SPEAKER_TUNING_MUTE, FLOAT_ZERO);
 	if (err < 0)
 		goto exit;
 
-	/* run through the output dsp commands for the selected output. */
-	for (i = 0; i < alt_out_presets[spec->cur_out_type].commands; i++) {
-		err = dspio_set_uint_param(codec,
-		alt_out_presets[spec->cur_out_type].mids[i],
-		alt_out_presets[spec->cur_out_type].reqs[i],
-		alt_out_presets[spec->cur_out_type].vals[i]);
-
+	if (spec->cur_out_type == SPEAKER_OUT) {
+		err = ca0132_alt_set_full_range_speaker(codec);
 		if (err < 0)
 			goto exit;
 	}
 
-	if (spec->cur_out_type == SPEAKER_OUT)
-		err = ca0132_alt_set_full_range_speaker(codec);
-
 exit:
 	snd_hda_power_down_pm(codec);
 
@@ -5659,7 +5631,7 @@ static int ca0132_alt_output_select_get_info(struct snd_kcontrol *kcontrol,
 	if (uinfo->value.enumerated.item >= NUM_OF_OUTPUTS)
 		uinfo->value.enumerated.item = NUM_OF_OUTPUTS - 1;
 	strcpy(uinfo->value.enumerated.name,
-			alt_out_presets[uinfo->value.enumerated.item].name);
+			out_type_str[uinfo->value.enumerated.item]);
 	return 0;
 }
 
@@ -5686,7 +5658,7 @@ static int ca0132_alt_output_select_put(struct snd_kcontrol *kcontrol,
 		return 0;
 
 	codec_dbg(codec, "ca0132_alt_output_select: sel=%d, preset=%s\n",
-		    sel, alt_out_presets[sel].name);
+		    sel, out_type_str[sel]);
 
 	spec->out_enum_val = sel;
 
-- 
cgit v1.2.3


From def3f0a5c7007c899a9a4d4034130e9d175d952d Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:27 -0400
Subject: ALSA: hda/ca0132 - Add quirk output selection structures.

Add structures containing the changes that need to happen on output
selection for each quirk. This should streamline the addition of new
quirks.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-9-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 336 +++++++++++++++++++++++++++++++------------
 1 file changed, 241 insertions(+), 95 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 39e333866be3..ab84ea397552 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1256,6 +1256,164 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
 	{}
 };
 
+/* Output selection quirk info structures. */
+#define MAX_QUIRK_MMIO_GPIO_SET_VALS 3
+#define MAX_QUIRK_SCP_SET_VALS 2
+struct ca0132_alt_out_set_info {
+	unsigned int dac2port; /* ParamID 0x0d value. */
+
+	bool has_hda_gpio;
+	char hda_gpio_pin;
+	char hda_gpio_set;
+
+	unsigned int mmio_gpio_count;
+	char mmio_gpio_pin[MAX_QUIRK_MMIO_GPIO_SET_VALS];
+	char mmio_gpio_set[MAX_QUIRK_MMIO_GPIO_SET_VALS];
+
+	unsigned int scp_cmds_count;
+	unsigned int scp_cmd_mid[MAX_QUIRK_SCP_SET_VALS];
+	unsigned int scp_cmd_req[MAX_QUIRK_SCP_SET_VALS];
+	unsigned int scp_cmd_val[MAX_QUIRK_SCP_SET_VALS];
+
+	bool has_chipio_write;
+	unsigned int chipio_write_addr;
+	unsigned int chipio_write_data;
+};
+
+struct ca0132_alt_out_set_quirk_data {
+	int quirk_id;
+
+	bool has_headphone_gain;
+	bool is_ae_series;
+
+	struct ca0132_alt_out_set_info out_set_info[NUM_OF_OUTPUTS];
+};
+
+static const struct ca0132_alt_out_set_quirk_data quirk_out_set_data[] = {
+	{ .quirk_id = QUIRK_R3DI,
+	  .has_headphone_gain = false,
+	  .is_ae_series       = false,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port         = 0x24,
+		  .has_hda_gpio     = true,
+		  .hda_gpio_pin     = 2,
+		  .hda_gpio_set     = 1,
+		  .mmio_gpio_count  = 0,
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		},
+		/* Headphones. */
+		{ .dac2port         = 0x21,
+		  .has_hda_gpio     = true,
+		  .hda_gpio_pin     = 2,
+		  .hda_gpio_set     = 0,
+		  .mmio_gpio_count  = 0,
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		} },
+	},
+	{ .quirk_id = QUIRK_R3D,
+	  .has_headphone_gain = false,
+	  .is_ae_series       = false,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port         = 0x24,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 1,
+		  .mmio_gpio_pin    = { 1 },
+		  .mmio_gpio_set    = { 1 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		},
+		/* Headphones. */
+		{ .dac2port         = 0x21,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 1,
+		  .mmio_gpio_pin    = { 1 },
+		  .mmio_gpio_set    = { 0 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		} },
+	},
+	{ .quirk_id = QUIRK_SBZ,
+	  .has_headphone_gain = false,
+	  .is_ae_series       = false,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port         = 0x18,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 3,
+		  .mmio_gpio_pin    = { 7, 4, 1 },
+		  .mmio_gpio_set    = { 0, 1, 1 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false, },
+		/* Headphones. */
+		{ .dac2port         = 0x12,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 3,
+		  .mmio_gpio_pin    = { 7, 4, 1 },
+		  .mmio_gpio_set    = { 1, 1, 0 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		} },
+	},
+	{ .quirk_id = QUIRK_ZXR,
+	  .has_headphone_gain = true,
+	  .is_ae_series       = false,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port         = 0x24,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 3,
+		  .mmio_gpio_pin    = { 2, 3, 5 },
+		  .mmio_gpio_set    = { 1, 1, 0 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		},
+		/* Headphones. */
+		{ .dac2port         = 0x21,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 3,
+		  .mmio_gpio_pin    = { 2, 3, 5 },
+		  .mmio_gpio_set    = { 0, 1, 1 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		} },
+	},
+	{ .quirk_id = QUIRK_AE5,
+	  .has_headphone_gain = true,
+	  .is_ae_series       = true,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port          = 0xa4,
+		  .has_hda_gpio      = false,
+		  .mmio_gpio_count   = 0,
+		  .scp_cmds_count    = 2,
+		  .scp_cmd_mid       = { 0x96, 0x96 },
+		  .scp_cmd_req       = { SPEAKER_TUNING_FRONT_LEFT_INVERT,
+					 SPEAKER_TUNING_FRONT_RIGHT_INVERT },
+		  .scp_cmd_val       = { FLOAT_ZERO, FLOAT_ZERO },
+		  .has_chipio_write  = true,
+		  .chipio_write_addr = 0x0018b03c,
+		  .chipio_write_data = 0x00000012
+		},
+		/* Headphones. */
+		{ .dac2port          = 0xa1,
+		  .has_hda_gpio      = false,
+		  .mmio_gpio_count   = 0,
+		  .scp_cmds_count    = 2,
+		  .scp_cmd_mid       = { 0x96, 0x96 },
+		  .scp_cmd_req       = { SPEAKER_TUNING_FRONT_LEFT_INVERT,
+					 SPEAKER_TUNING_FRONT_RIGHT_INVERT },
+		  .scp_cmd_val       = { FLOAT_ONE, FLOAT_ONE },
+		  .has_chipio_write  = true,
+		  .chipio_write_addr = 0x0018b03c,
+		  .chipio_write_data = 0x00000012
+		} },
+	}
+};
+
 /*
  * CA0132 codec access
  */
@@ -3513,26 +3671,6 @@ static void r3di_gpio_mic_set(struct hda_codec *codec,
 			    AC_VERB_SET_GPIO_DATA, cur_gpio);
 }
 
-static void r3di_gpio_out_set(struct hda_codec *codec,
-		enum r3di_out_select cur_out)
-{
-	unsigned int cur_gpio;
-
-	/* Get the current GPIO Data setup */
-	cur_gpio = snd_hda_codec_read(codec, 0x01, 0, AC_VERB_GET_GPIO_DATA, 0);
-
-	switch (cur_out) {
-	case R3DI_HEADPHONE_OUT:
-		cur_gpio &= ~(1 << R3DI_OUT_SELECT_BIT);
-		break;
-	case R3DI_LINE_OUT:
-		cur_gpio |= (1 << R3DI_OUT_SELECT_BIT);
-		break;
-	}
-	snd_hda_codec_write(codec, codec->core.afg, 0,
-			    AC_VERB_SET_GPIO_DATA, cur_gpio);
-}
-
 static void r3di_gpio_dsp_status_set(struct hda_codec *codec,
 		enum r3di_dsp_status dsp_status)
 {
@@ -4314,86 +4452,93 @@ static int ca0132_alt_surround_set_bass_redirection(struct hda_codec *codec,
  * These are the commands needed to setup output on each of the different card
  * types.
  */
-static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec)
+static void ca0132_alt_select_out_get_quirk_data(struct hda_codec *codec,
+		const struct ca0132_alt_out_set_quirk_data **quirk_data)
 {
 	struct ca0132_spec *spec = codec->spec;
-	unsigned int tmp;
+	int quirk = ca0132_quirk(spec);
+	unsigned int i;
 
-	switch (spec->cur_out_type) {
-	case SPEAKER_OUT:
-		switch (ca0132_quirk(spec)) {
-		case QUIRK_SBZ:
-			ca0113_mmio_gpio_set(codec, 7, false);
-			ca0113_mmio_gpio_set(codec, 4, true);
-			ca0113_mmio_gpio_set(codec, 1, true);
-			chipio_set_control_param(codec, 0x0d, 0x18);
-			break;
-		case QUIRK_ZXR:
-			ca0113_mmio_gpio_set(codec, 2, true);
-			ca0113_mmio_gpio_set(codec, 3, true);
-			ca0113_mmio_gpio_set(codec, 5, false);
-			zxr_headphone_gain_set(codec, 0);
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			break;
-		case QUIRK_R3DI:
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			r3di_gpio_out_set(codec, R3DI_LINE_OUT);
-			break;
-		case QUIRK_R3D:
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			ca0113_mmio_gpio_set(codec, 1, true);
-			break;
-		case QUIRK_AE5:
-			ae5_mmio_select_out(codec);
-			ae5_headphone_gain_set(codec, 2);
-			tmp = FLOAT_ZERO;
-			dspio_set_uint_param(codec, 0x96, 0x29, tmp);
-			dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
-			chipio_set_control_param(codec, 0x0d, 0xa4);
-			chipio_write(codec, 0x18b03c, 0x00000012);
-			break;
-		default:
-			break;
+	*quirk_data = NULL;
+	for (i = 0; i < ARRAY_SIZE(quirk_out_set_data); i++) {
+		if (quirk_out_set_data[i].quirk_id == quirk) {
+			*quirk_data = &quirk_out_set_data[i];
+			return;
 		}
-		break;
-	case HEADPHONE_OUT:
-		switch (ca0132_quirk(spec)) {
-		case QUIRK_SBZ:
-			ca0113_mmio_gpio_set(codec, 7, true);
-			ca0113_mmio_gpio_set(codec, 4, true);
-			ca0113_mmio_gpio_set(codec, 1, false);
-			chipio_set_control_param(codec, 0x0d, 0x12);
-			break;
-		case QUIRK_ZXR:
-			ca0113_mmio_gpio_set(codec, 2, false);
-			ca0113_mmio_gpio_set(codec, 3, false);
-			ca0113_mmio_gpio_set(codec, 5, true);
-			zxr_headphone_gain_set(codec, spec->zxr_gain_set);
-			chipio_set_control_param(codec, 0x0d, 0x21);
-			break;
-		case QUIRK_R3DI:
-			chipio_set_control_param(codec, 0x0d, 0x21);
-			r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT);
-			break;
-		case QUIRK_R3D:
-			chipio_set_control_param(codec, 0x0d, 0x21);
-			ca0113_mmio_gpio_set(codec, 0x1, false);
-			break;
-		case QUIRK_AE5:
-			ae5_mmio_select_out(codec);
-			ae5_headphone_gain_set(codec,
-					spec->ae5_headphone_gain_val);
-			tmp = FLOAT_ONE;
-			dspio_set_uint_param(codec, 0x96, 0x29, tmp);
-			dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
-			chipio_set_control_param(codec, 0x0d, 0xa1);
-			chipio_write(codec, 0x18b03c, 0x00000012);
-			break;
-		default:
-			break;
+	}
+}
+
+static int ca0132_alt_select_out_quirk_set(struct hda_codec *codec)
+{
+	const struct ca0132_alt_out_set_quirk_data *quirk_data;
+	const struct ca0132_alt_out_set_info *out_info;
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int i, gpio_data;
+	int err;
+
+	ca0132_alt_select_out_get_quirk_data(codec, &quirk_data);
+	if (!quirk_data)
+		return 0;
+
+	out_info = &quirk_data->out_set_info[spec->cur_out_type];
+	if (quirk_data->is_ae_series)
+		ae5_mmio_select_out(codec);
+
+	if (out_info->has_hda_gpio) {
+		gpio_data = snd_hda_codec_read(codec, codec->core.afg, 0,
+				AC_VERB_GET_GPIO_DATA, 0);
+
+		if (out_info->hda_gpio_set)
+			gpio_data |= (1 << out_info->hda_gpio_pin);
+		else
+			gpio_data &= ~(1 << out_info->hda_gpio_pin);
+
+		snd_hda_codec_write(codec, codec->core.afg, 0,
+				    AC_VERB_SET_GPIO_DATA, gpio_data);
+	}
+
+	if (out_info->mmio_gpio_count) {
+		for (i = 0; i < out_info->mmio_gpio_count; i++) {
+			ca0113_mmio_gpio_set(codec, out_info->mmio_gpio_pin[i],
+					out_info->mmio_gpio_set[i]);
 		}
-		break;
 	}
+
+	if (out_info->scp_cmds_count) {
+		for (i = 0; i < out_info->scp_cmds_count; i++) {
+			err = dspio_set_uint_param(codec,
+					out_info->scp_cmd_mid[i],
+					out_info->scp_cmd_req[i],
+					out_info->scp_cmd_val[i]);
+			if (err < 0)
+				return err;
+		}
+	}
+
+	chipio_set_control_param(codec, 0x0d, out_info->dac2port);
+
+	if (out_info->has_chipio_write) {
+		chipio_write(codec, out_info->chipio_write_addr,
+				out_info->chipio_write_data);
+	}
+
+	if (quirk_data->has_headphone_gain) {
+		if (spec->cur_out_type != HEADPHONE_OUT) {
+			if (quirk_data->is_ae_series)
+				ae5_headphone_gain_set(codec, 2);
+			else
+				zxr_headphone_gain_set(codec, 0);
+		} else {
+			if (quirk_data->is_ae_series)
+				ae5_headphone_gain_set(codec,
+						spec->ae5_headphone_gain_val);
+			else
+				zxr_headphone_gain_set(codec,
+						spec->zxr_gain_set);
+		}
+	}
+
+	return 0;
 }
 
 static void ca0132_set_out_node_pincfg(struct hda_codec *codec, hda_nid_t nid,
@@ -4456,7 +4601,8 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 	if (err < 0)
 		goto exit;
 
-	ca0132_alt_select_out_quirk_handler(codec);
+	if (ca0132_alt_select_out_quirk_set(codec) < 0)
+		goto exit;
 
 	switch (spec->cur_out_type) {
 	case SPEAKER_OUT:
-- 
cgit v1.2.3


From b7a8b9e8e797236a62716118c513e13a2670fd07 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:28 -0400
Subject: ALSA: hda/ca0132 - Fix Recon3D Center/LFE output.

Properly set the GPIO pin to un-mute the Center/LFE channel on the
Recon3D.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-10-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 6 ++++++
 1 file changed, 6 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index ab84ea397552..138403fd1639 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -7819,6 +7819,12 @@ static void r3d_setup_defaults(struct hda_codec *codec)
 	if (ca0132_quirk(spec) == QUIRK_R3DI)
 		r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED);
 
+	/* Disable mute on Center/LFE. */
+	if (ca0132_quirk(spec) == QUIRK_R3D) {
+		ca0113_mmio_gpio_set(codec, 2, false);
+		ca0113_mmio_gpio_set(codec, 4, true);
+	}
+
 	/* Setup effect defaults */
 	num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1;
 	for (idx = 0; idx < num_fx; idx++) {
-- 
cgit v1.2.3


From 620f08eea6d6961b789af3fa3ea86725c8c93ece Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:29 -0400
Subject: ALSA: hda/ca0132 - Add new quirk ID for SoundBlaster AE-7.

Add a new PCI subsystem ID for the SoundBlaster AE-7 card.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-11-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 138403fd1639..284f63dc2749 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1134,6 +1134,7 @@ enum {
 	QUIRK_R3DI,
 	QUIRK_R3D,
 	QUIRK_AE5,
+	QUIRK_AE7,
 };
 
 #ifdef CONFIG_PCI
@@ -1253,6 +1254,7 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
 	SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D),
 	SND_PCI_QUIRK(0x1102, 0x0018, "Recon3D", QUIRK_R3D),
 	SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5),
+	SND_PCI_QUIRK(0x1102, 0x0081, "Sound Blaster AE-7", QUIRK_AE7),
 	{}
 };
 
-- 
cgit v1.2.3


From a35e37a3a2c5fc9a6dcf1e0445e8472cffac37c1 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:30 -0400
Subject: ALSA: hda/ca0132 - Add SoundBlaster AE-7 pincfg.

Add AE-7 pincfg, based on the values set within Windows.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-12-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 19 +++++++++++++++++++
 1 file changed, 19 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 284f63dc2749..6791aaf18e63 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1238,6 +1238,20 @@ static const struct hda_pintbl r3di_pincfgs[] = {
 	{}
 };
 
+static const struct hda_pintbl ae7_pincfgs[] = {
+	{ 0x0b, 0x01017010 },
+	{ 0x0c, 0x014510f0 },
+	{ 0x0d, 0x414510f0 },
+	{ 0x0e, 0x01c520f0 },
+	{ 0x0f, 0x01017114 },
+	{ 0x10, 0x01017011 },
+	{ 0x11, 0x018170ff },
+	{ 0x12, 0x01a170f0 },
+	{ 0x13, 0x908700f0 },
+	{ 0x18, 0x500000f0 },
+	{}
+};
+
 static const struct snd_pci_quirk ca0132_quirks[] = {
 	SND_PCI_QUIRK(0x1028, 0x057b, "Alienware M17x R4", QUIRK_ALIENWARE_M17XR4),
 	SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE),
@@ -9105,6 +9119,10 @@ static void ca0132_config(struct hda_codec *codec)
 		codec_dbg(codec, "%s: QUIRK_AE5 applied.\n", __func__);
 		snd_hda_apply_pincfgs(codec, ae5_pincfgs);
 		break;
+	case QUIRK_AE7:
+		codec_dbg(codec, "%s: QUIRK_AE7 applied.\n", __func__);
+		snd_hda_apply_pincfgs(codec, ae7_pincfgs);
+		break;
 	default:
 		break;
 	}
@@ -9186,6 +9204,7 @@ static void ca0132_config(struct hda_codec *codec)
 		spec->dig_in = 0x09;
 		break;
 	case QUIRK_AE5:
+	case QUIRK_AE7:
 		spec->num_outputs = 2;
 		spec->out_pins[0] = 0x0B; /* Line out */
 		spec->out_pins[1] = 0x11; /* Rear headphone out */
-- 
cgit v1.2.3


From 76d257d67f4104bd3089be1a486047e3f3b40c24 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:31 -0400
Subject: ALSA: hda/ca0132 - Set AE-7 bools and select mixer.

Set the boolean values used for desktop cards, and select the desktop
mixer.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-13-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 6791aaf18e63..bd5d4f0bd6f5 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -9403,6 +9403,10 @@ static int patch_ca0132(struct hda_codec *codec)
 		spec->mixers[0] = desktop_mixer;
 		snd_hda_codec_set_name(codec, "Sound BlasterX AE-5");
 		break;
+	case QUIRK_AE7:
+		spec->mixers[0] = desktop_mixer;
+		snd_hda_codec_set_name(codec, "Sound Blaster AE-7");
+		break;
 	default:
 		spec->mixers[0] = ca0132_mixer;
 		break;
@@ -9413,6 +9417,7 @@ static int patch_ca0132(struct hda_codec *codec)
 	case QUIRK_SBZ:
 	case QUIRK_R3D:
 	case QUIRK_AE5:
+	case QUIRK_AE7:
 	case QUIRK_ZXR:
 		spec->use_alt_controls = true;
 		spec->use_alt_functions = true;
-- 
cgit v1.2.3


From 4e356d56df9dbd4631ce2e40a5fb491829f4e8ce Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:32 -0400
Subject: ALSA: hda/ca0132 - Add ca0132_mmio_init data for SoundBlaster AE-7.

Modify the AE-5 ca0132_mmio_init function to add AE-7 specific writes.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-14-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 20 +++++++++++++++++++-
 1 file changed, 19 insertions(+), 1 deletion(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index bd5d4f0bd6f5..10aaa4806946 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -8748,8 +8748,26 @@ static void ca0132_mmio_init_ae5(struct hda_codec *codec)
 	data = ca0113_mmio_init_data_ae5;
 	count = ARRAY_SIZE(ca0113_mmio_init_data_ae5);
 
-	for (i = 0; i < count; i++)
+	if (ca0132_quirk(spec) == QUIRK_AE7) {
+		writel(0x00000680, spec->mem_base + 0x1c);
+		writel(0x00880680, spec->mem_base + 0x1c);
+	}
+
+	for (i = 0; i < count; i++) {
+		/*
+		 * AE-7 shares all writes with the AE-5, except that it writes
+		 * a different value to 0x20c.
+		 */
+		if (i == 21 && ca0132_quirk(spec) == QUIRK_AE7) {
+			writel(0x00800001, spec->mem_base + addr[i]);
+			continue;
+		}
+
 		writel(data[i], spec->mem_base + addr[i]);
+	}
+
+	if (ca0132_quirk(spec) == QUIRK_AE5)
+		writel(0x00880680, spec->mem_base + 0x1c);
 }
 
 static void ca0132_mmio_init(struct hda_codec *codec)
-- 
cgit v1.2.3


From 77bdbae90445771f76510e8c6a4ac0b722a19472 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:33 -0400
Subject: ALSA: hda/ca0132 - Add pre-init function for SoundBlaster AE-7.

Add pre DSP initialization function for the AE-7.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-15-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 70 ++++++++++++++++++++++++++++++++++----------
 1 file changed, 55 insertions(+), 15 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 10aaa4806946..cd46112c827e 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -8786,6 +8786,16 @@ static void ca0132_mmio_init(struct hda_codec *codec)
 	}
 }
 
+static const unsigned int ca0132_ae5_register_set_addresses[] = {
+	0x304, 0x304, 0x304, 0x304, 0x100, 0x304, 0x100, 0x304, 0x100, 0x304,
+	0x100, 0x304, 0x86c, 0x800, 0x86c, 0x800, 0x804
+};
+
+static const unsigned char ca0132_ae5_register_set_data[] = {
+	0x0f, 0x0e, 0x1f, 0x0c, 0x3f, 0x08, 0x7f, 0x00, 0xff, 0x00, 0x6b,
+	0x01, 0x6b, 0x57
+};
+
 /*
  * This function writes to some SFR's, does some region2 writes, and then
  * eventually resets the codec with the 0x7ff verb. Not quite sure why it does
@@ -8794,6 +8804,18 @@ static void ca0132_mmio_init(struct hda_codec *codec)
 static void ae5_register_set(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
+	unsigned int count = ARRAY_SIZE(ca0132_ae5_register_set_addresses);
+	const unsigned int *addr = ca0132_ae5_register_set_addresses;
+	const unsigned char *data = ca0132_ae5_register_set_data;
+	unsigned int i, cur_addr;
+	unsigned char tmp[3];
+
+	if (ca0132_quirk(spec) == QUIRK_AE7) {
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x41);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc8);
+	}
 
 	chipio_8051_write_direct(codec, 0x93, 0x10);
 	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
@@ -8801,25 +8823,43 @@ static void ae5_register_set(struct hda_codec *codec)
 	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
 			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2);
 
-	writeb(0x0f, spec->mem_base + 0x304);
-	writeb(0x0f, spec->mem_base + 0x304);
-	writeb(0x0f, spec->mem_base + 0x304);
-	writeb(0x0f, spec->mem_base + 0x304);
-	writeb(0x0e, spec->mem_base + 0x100);
-	writeb(0x1f, spec->mem_base + 0x304);
-	writeb(0x0c, spec->mem_base + 0x100);
-	writeb(0x3f, spec->mem_base + 0x304);
-	writeb(0x08, spec->mem_base + 0x100);
-	writeb(0x7f, spec->mem_base + 0x304);
-	writeb(0x00, spec->mem_base + 0x100);
-	writeb(0xff, spec->mem_base + 0x304);
+	if (ca0132_quirk(spec) == QUIRK_AE7) {
+		tmp[0] = 0x03;
+		tmp[1] = 0x03;
+		tmp[2] = 0x07;
+	} else {
+		tmp[0] = 0x0f;
+		tmp[1] = 0x0f;
+		tmp[2] = 0x0f;
+	}
 
-	ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f);
+	for (i = cur_addr = 0; i < 3; i++, cur_addr++)
+		writeb(tmp[i], spec->mem_base + addr[cur_addr]);
+
+	/*
+	 * First writes are in single bytes, final are in 4 bytes. So, we use
+	 * writeb, then writel.
+	 */
+	for (i = 0; cur_addr < 12; i++, cur_addr++)
+		writeb(data[i], spec->mem_base + addr[cur_addr]);
+
+	for (; cur_addr < count; i++, cur_addr++)
+		writel(data[i], spec->mem_base + addr[cur_addr]);
+
+	writel(0x00800001, spec->mem_base + 0x20c);
+
+	if (ca0132_quirk(spec) == QUIRK_AE7) {
+		ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83);
+		ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f);
+	} else {
+		ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f);
+	}
 
 	chipio_8051_write_direct(codec, 0x90, 0x00);
 	chipio_8051_write_direct(codec, 0x90, 0x10);
 
-	ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+	if (ca0132_quirk(spec) == QUIRK_AE5)
+		ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
 
 	chipio_write(codec, 0x18b0a4, 0x000000c2);
 
@@ -8918,7 +8958,7 @@ static int ca0132_init(struct hda_codec *codec)
 
 	snd_hda_power_up_pm(codec);
 
-	if (ca0132_quirk(spec) == QUIRK_AE5)
+	if (ca0132_quirk(spec) == QUIRK_AE5 || ca0132_quirk(spec) == QUIRK_AE7)
 		ae5_register_set(codec);
 
 	ca0132_init_unsol(codec);
-- 
cgit v1.2.3


From cfa736e2f02d768dad1b611abdb85fe6c10d8000 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:34 -0400
Subject: ALSA: hda/ca0132 - Add init data for SoundBlaster AE-7.

Add initialization data for the SoundBlaster AE-7 card.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-16-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 14 ++++++++++++++
 1 file changed, 14 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index cd46112c827e..dc1eb9bfcc5e 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -3582,6 +3582,7 @@ static void ca0132_gpio_init(struct hda_codec *codec)
 	switch (ca0132_quirk(spec)) {
 	case QUIRK_SBZ:
 	case QUIRK_AE5:
+	case QUIRK_AE7:
 		snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
 		snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
 		snd_hda_codec_write(codec, 0x01, 0, 0x790, 0x23);
@@ -8911,6 +8912,19 @@ static void ca0132_alt_init(struct hda_codec *codec)
 		snd_hda_sequence_write(codec, spec->desktop_init_verbs);
 		ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
 		break;
+	case QUIRK_AE7:
+		ca0132_gpio_init(codec);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x49);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				VENDOR_CHIPIO_PLL_PMU_WRITE, 0x88);
+		snd_hda_sequence_write(codec, spec->chip_init_verbs);
+		snd_hda_sequence_write(codec, spec->desktop_init_verbs);
+		chipio_write(codec, 0x18b008, 0x000000f8);
+		chipio_write(codec, 0x18b008, 0x000000f0);
+		chipio_write(codec, 0x18b030, 0x00000020);
+		ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
+		break;
 	case QUIRK_ZXR:
 		snd_hda_sequence_write(codec, spec->chip_init_verbs);
 		snd_hda_sequence_write(codec, spec->desktop_init_verbs);
-- 
cgit v1.2.3


From e5b21888882bb340bf0fbcbcea909b09eef2933f Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:35 -0400
Subject: ALSA: hda/ca0132 - Add DSP setup functions for AE-7.

Add DSP setup functions for the Sound Blaster AE-7 post DSP download.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-17-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 290 +++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 290 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index dc1eb9bfcc5e..8519119ef7a6 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -7378,6 +7378,7 @@ static void ca0132_alt_init_speaker_tuning(struct hda_codec *codec)
 		values = zxr_default_delay_values;
 		break;
 	case QUIRK_AE5:
+	case QUIRK_AE7:
 		values = ae5_default_delay_values;
 		break;
 	default:
@@ -7551,6 +7552,7 @@ static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec)
 		switch (ca0132_quirk(spec)) {
 		case QUIRK_SBZ:
 		case QUIRK_AE5:
+		case QUIRK_AE7:
 			tmp = 0x00000003;
 			dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
 			tmp = 0x00000000;
@@ -7760,6 +7762,206 @@ static void ae5_post_dsp_startup_data(struct hda_codec *codec)
 	mutex_unlock(&spec->chipio_mutex);
 }
 
+static const unsigned int ae7_port_set_data[] = {
+	0x0001e0c0, 0x0001e1c1, 0x0001e4c2, 0x0001e5c3, 0x0001e2c4, 0x0001e3c5,
+	0x0001e8c6, 0x0001e9c7, 0x0001ecc8, 0x0001edc9, 0x0001eaca, 0x0001ebcb
+};
+
+static void ae7_post_dsp_setup_ports(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int i, count, addr;
+
+	mutex_lock(&spec->chipio_mutex);
+
+	chipio_set_stream_channels(codec, 0x0c, 6);
+	chipio_set_stream_control(codec, 0x0c, 1);
+
+	count = ARRAY_SIZE(ae7_port_set_data);
+	addr = 0x190030;
+	for (i = 0; i < count; i++) {
+		chipio_write_no_mutex(codec, addr, ae7_port_set_data[i]);
+
+		/* Addresses are incremented by 4-bytes. */
+		addr += 0x04;
+	}
+
+	/*
+	 * Port setting always ends with a write of 0x1 to address 0x19042c.
+	 */
+	chipio_write_no_mutex(codec, 0x19042c, 0x00000001);
+
+	ca0113_mmio_command_set(codec, 0x30, 0x30, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x0d, 0x40);
+	ca0113_mmio_command_set(codec, 0x48, 0x17, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x19, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x11, 0xff);
+	ca0113_mmio_command_set(codec, 0x48, 0x12, 0xff);
+	ca0113_mmio_command_set(codec, 0x48, 0x13, 0xff);
+	ca0113_mmio_command_set(codec, 0x48, 0x14, 0x7f);
+
+	mutex_unlock(&spec->chipio_mutex);
+}
+
+static void ae7_post_dsp_asi_stream_setup(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	mutex_lock(&spec->chipio_mutex);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x725, 0x81);
+	ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x00);
+
+	chipio_set_conn_rate_no_mutex(codec, 0x70, SR_96_000);
+	chipio_set_stream_channels(codec, 0x0c, 6);
+	chipio_set_stream_control(codec, 0x0c, 1);
+
+	chipio_set_stream_source_dest(codec, 0x05, 0x43, 0x00);
+	chipio_set_stream_source_dest(codec, 0x18, 0x09, 0xd0);
+
+	chipio_set_conn_rate_no_mutex(codec, 0xd0, SR_96_000);
+	chipio_set_stream_channels(codec, 0x18, 6);
+	chipio_set_stream_control(codec, 0x18, 1);
+
+	chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 4);
+
+	mutex_unlock(&spec->chipio_mutex);
+}
+
+static void ae7_post_dsp_pll_setup(struct hda_codec *codec)
+{
+	const unsigned int addr[] = { 0x41, 0x45, 0x40, 0x43, 0x51 };
+	const unsigned int data[] = { 0xc8, 0xcc, 0xcb, 0xc7, 0x8d };
+	unsigned int i;
+
+	for (i = 0; i < ARRAY_SIZE(addr); i++) {
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				    VENDOR_CHIPIO_8051_ADDRESS_LOW, addr[i]);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				    VENDOR_CHIPIO_PLL_PMU_WRITE, data[i]);
+	}
+}
+
+static void ae7_post_dsp_asi_setup_ports(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	const unsigned int target[] = { 0x0b, 0x04, 0x06, 0x0a, 0x0c, 0x11,
+					0x12, 0x13, 0x14 };
+	const unsigned int data[]   = { 0x12, 0x00, 0x48, 0x05, 0x5f, 0xff,
+					0xff, 0xff, 0x7f };
+	unsigned int i;
+
+	mutex_lock(&spec->chipio_mutex);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7);
+
+	chipio_write_no_mutex(codec, 0x189000, 0x0001f101);
+	chipio_write_no_mutex(codec, 0x189004, 0x0001f101);
+	chipio_write_no_mutex(codec, 0x189024, 0x00014004);
+	chipio_write_no_mutex(codec, 0x189028, 0x0002000f);
+
+	ae7_post_dsp_pll_setup(codec);
+	chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 7);
+
+	for (i = 0; i < ARRAY_SIZE(target); i++)
+		ca0113_mmio_command_set(codec, 0x48, target[i], data[i]);
+
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83);
+	ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00);
+
+	chipio_set_stream_source_dest(codec, 0x21, 0x64, 0x56);
+	chipio_set_stream_channels(codec, 0x21, 2);
+	chipio_set_conn_rate_no_mutex(codec, 0x56, SR_8_000);
+
+	chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_NODE_ID, 0x09);
+	/*
+	 * In the 8051's memory, this param is referred to as 'n2sid', which I
+	 * believe is 'node to streamID'. It seems to be a way to assign a
+	 * stream to a given HDA node.
+	 */
+	chipio_set_control_param_no_mutex(codec, 0x20, 0x21);
+
+	chipio_write_no_mutex(codec, 0x18b038, 0x00000088);
+
+	/*
+	 * Now, at this point on Windows, an actual stream is setup and
+	 * seemingly sends data to the HDA node 0x09, which is the digital
+	 * audio input node. This is left out here, because obviously I don't
+	 * know what data is being sent. Interestingly, the AE-5 seems to go
+	 * through the motions of getting here and never actually takes this
+	 * step, but the AE-7 does.
+	 */
+
+	ca0113_mmio_gpio_set(codec, 0, 1);
+	ca0113_mmio_gpio_set(codec, 1, 1);
+
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83);
+	chipio_write_no_mutex(codec, 0x18b03c, 0x00000000);
+	ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00);
+
+	chipio_set_stream_source_dest(codec, 0x05, 0x43, 0x00);
+	chipio_set_stream_source_dest(codec, 0x18, 0x09, 0xd0);
+
+	chipio_set_conn_rate_no_mutex(codec, 0xd0, SR_96_000);
+	chipio_set_stream_channels(codec, 0x18, 6);
+
+	/*
+	 * Runs again, this has been repeated a few times, but I'm just
+	 * following what the Windows driver does.
+	 */
+	ae7_post_dsp_pll_setup(codec);
+	chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 7);
+
+	mutex_unlock(&spec->chipio_mutex);
+}
+
+/*
+ * The Windows driver has commands that seem to setup ASI, which I believe to
+ * be some sort of audio serial interface. My current speculation is that it's
+ * related to communicating with the new DAC.
+ */
+static void ae7_post_dsp_asi_setup(struct hda_codec *codec)
+{
+	chipio_8051_write_direct(codec, 0x93, 0x10);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2);
+
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83);
+	ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f);
+
+	chipio_set_control_param(codec, 3, 3);
+	chipio_set_control_flag(codec, CONTROL_FLAG_ASI_96KHZ, 1);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x724, 0x83);
+	chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0);
+	snd_hda_codec_write(codec, 0x17, 0, 0x794, 0x00);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x92);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0xfa);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_DATA_WRITE, 0x22);
+
+	ae7_post_dsp_pll_setup(codec);
+	ae7_post_dsp_asi_stream_setup(codec);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7);
+
+	ae7_post_dsp_asi_setup_ports(codec);
+}
+
 /*
  * Setup default parameters for DSP
  */
@@ -7983,6 +8185,91 @@ static void ae5_setup_defaults(struct hda_codec *codec)
 	ca0132_alt_create_dummy_stream(codec);
 }
 
+/*
+ * Setup default parameters for the Sound Blaster AE-7 DSP.
+ */
+static void ae7_setup_defaults(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp;
+	int num_fx;
+	int idx, i;
+
+	if (spec->dsp_state != DSP_DOWNLOADED)
+		return;
+
+	ca0132_alt_dsp_scp_startup(codec);
+	ca0132_alt_init_analog_mics(codec);
+	ae7_post_dsp_setup_ports(codec);
+
+	tmp = FLOAT_ZERO;
+	dspio_set_uint_param(codec, 0x96,
+			SPEAKER_TUNING_FRONT_LEFT_INVERT, tmp);
+	dspio_set_uint_param(codec, 0x96,
+			SPEAKER_TUNING_FRONT_RIGHT_INVERT, tmp);
+
+	ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f);
+
+	/* New, unknown SCP req's */
+	dspio_set_uint_param(codec, 0x80, 0x0d, tmp);
+	dspio_set_uint_param(codec, 0x80, 0x0e, tmp);
+
+	ca0113_mmio_gpio_set(codec, 0, false);
+
+	/* Internal loopback off */
+	tmp = FLOAT_ONE;
+	dspio_set_uint_param(codec, 0x37, 0x08, tmp);
+	dspio_set_uint_param(codec, 0x37, 0x10, tmp);
+
+	/*remove DSP headroom*/
+	tmp = FLOAT_ZERO;
+	dspio_set_uint_param(codec, 0x96, 0x3C, tmp);
+
+	/* set WUH source */
+	tmp = FLOAT_TWO;
+	dspio_set_uint_param(codec, 0x31, 0x00, tmp);
+	chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000);
+
+	/* Set speaker source? */
+	dspio_set_uint_param(codec, 0x32, 0x00, tmp);
+	ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
+
+	/*
+	 * This is the second time we've called this, but this is seemingly
+	 * what Windows does.
+	 */
+	ca0132_alt_init_analog_mics(codec);
+
+	ae7_post_dsp_asi_setup(codec);
+
+	/*
+	 * Not sure why, but these are both set to 1. They're only set to 0
+	 * upon shutdown.
+	 */
+	ca0113_mmio_gpio_set(codec, 0, true);
+	ca0113_mmio_gpio_set(codec, 1, true);
+
+	/* Volume control related. */
+	ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x04);
+	ca0113_mmio_command_set(codec, 0x48, 0x10, 0x04);
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x80);
+
+	/* out, in effects + voicefx */
+	num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1;
+	for (idx = 0; idx < num_fx; idx++) {
+		for (i = 0; i <= ca0132_effects[idx].params; i++) {
+			dspio_set_uint_param(codec,
+					ca0132_effects[idx].mid,
+					ca0132_effects[idx].reqs[i],
+					ca0132_effects[idx].def_vals[i]);
+		}
+	}
+
+	ca0132_alt_init_speaker_tuning(codec);
+
+	ca0132_alt_create_dummy_stream(codec);
+}
+
 /*
  * Initialization of flags in chip
  */
@@ -9000,6 +9287,9 @@ static int ca0132_init(struct hda_codec *codec)
 	case QUIRK_AE5:
 		ae5_setup_defaults(codec);
 		break;
+	case QUIRK_AE7:
+		ae7_setup_defaults(codec);
+		break;
 	default:
 		ca0132_setup_defaults(codec);
 		ca0132_init_analog_mic2(codec);
-- 
cgit v1.2.3


From 91b94a933f28b0b819224bc638866b0844317776 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:36 -0400
Subject: ALSA: hda/ca0132 - Add output selection for SoundBlaster AE-7.

Add output selection quirk table information for SoundBlaster AE-7, and
slightly modify the AE-5's ca0113 command table to accommodate the AE-7.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-18-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 88 ++++++++++++++++++++++++++++++++++----------
 1 file changed, 68 insertions(+), 20 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 8519119ef7a6..dcc8d29d934c 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -727,22 +727,29 @@ static const struct ct_dsp_volume_ctl ca0132_alt_vol_ctls[] = {
 };
 
 /* Values for ca0113_mmio_command_set for selecting output. */
-#define AE5_CA0113_OUT_SET_COMMANDS 6
-struct ae5_ca0113_output_set {
-	unsigned int group[AE5_CA0113_OUT_SET_COMMANDS];
-	unsigned int target[AE5_CA0113_OUT_SET_COMMANDS];
-	unsigned int vals[AE5_CA0113_OUT_SET_COMMANDS];
+#define AE_CA0113_OUT_SET_COMMANDS 6
+struct ae_ca0113_output_set {
+	unsigned int group[AE_CA0113_OUT_SET_COMMANDS];
+	unsigned int target[AE_CA0113_OUT_SET_COMMANDS];
+	unsigned int vals[NUM_OF_OUTPUTS][AE_CA0113_OUT_SET_COMMANDS];
 };
 
-static const struct ae5_ca0113_output_set ae5_ca0113_output_presets[] = {
-	{ .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
-	  .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
-	  .vals =   { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f }
-	},
-	{ .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
-	  .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
-	  .vals =   { 0x3f, 0x3f, 0x00, 0x00, 0x00, 0x00 }
-	}
+static const struct ae_ca0113_output_set ae5_ca0113_output_presets = {
+	.group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
+	.target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
+		    /* Speakers. */
+	.vals =   { { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f },
+		    /* Headphones. */
+		    { 0x3f, 0x3f, 0x00, 0x00, 0x00, 0x00 } },
+};
+
+static const struct ae_ca0113_output_set ae7_ca0113_output_presets = {
+	.group  = { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
+	.target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
+		    /* Speakers. */
+	.vals   = { { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f },
+		    /* Headphones. */
+		    { 0x3f, 0x3f, 0x00, 0x00, 0x02, 0x00 } },
 };
 
 /* ae5 ca0113 command sequences to set headphone gain levels. */
@@ -1427,6 +1434,41 @@ static const struct ca0132_alt_out_set_quirk_data quirk_out_set_data[] = {
 		  .chipio_write_addr = 0x0018b03c,
 		  .chipio_write_data = 0x00000012
 		} },
+	},
+	{ .quirk_id = QUIRK_AE7,
+	  .has_headphone_gain = true,
+	  .is_ae_series       = true,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port          = 0x58,
+		  .has_hda_gpio      = false,
+		  .mmio_gpio_count   = 1,
+		  .mmio_gpio_pin     = { 0 },
+		  .mmio_gpio_set     = { 1 },
+		  .scp_cmds_count    = 2,
+		  .scp_cmd_mid       = { 0x96, 0x96 },
+		  .scp_cmd_req       = { SPEAKER_TUNING_FRONT_LEFT_INVERT,
+					 SPEAKER_TUNING_FRONT_RIGHT_INVERT },
+		  .scp_cmd_val       = { FLOAT_ZERO, FLOAT_ZERO },
+		  .has_chipio_write  = true,
+		  .chipio_write_addr = 0x0018b03c,
+		  .chipio_write_data = 0x00000000
+		},
+		/* Headphones. */
+		{ .dac2port          = 0x58,
+		  .has_hda_gpio      = false,
+		  .mmio_gpio_count   = 1,
+		  .mmio_gpio_pin     = { 0 },
+		  .mmio_gpio_set     = { 1 },
+		  .scp_cmds_count    = 2,
+		  .scp_cmd_mid       = { 0x96, 0x96 },
+		  .scp_cmd_req       = { SPEAKER_TUNING_FRONT_LEFT_INVERT,
+					 SPEAKER_TUNING_FRONT_RIGHT_INVERT },
+		  .scp_cmd_val       = { FLOAT_ONE, FLOAT_ONE },
+		  .has_chipio_write  = true,
+		  .chipio_write_addr = 0x0018b03c,
+		  .chipio_write_data = 0x00000010
+		} },
 	}
 };
 
@@ -4383,18 +4425,24 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val);
 static void ae5_mmio_select_out(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
+	const struct ae_ca0113_output_set *out_cmds;
 	unsigned int i;
 
-	for (i = 0; i < AE5_CA0113_OUT_SET_COMMANDS; i++)
-		ca0113_mmio_command_set(codec,
-			ae5_ca0113_output_presets[spec->cur_out_type].group[i],
-			ae5_ca0113_output_presets[spec->cur_out_type].target[i],
-			ae5_ca0113_output_presets[spec->cur_out_type].vals[i]);
+	if (ca0132_quirk(spec) == QUIRK_AE5)
+		out_cmds = &ae5_ca0113_output_presets;
+	else
+		out_cmds = &ae7_ca0113_output_presets;
+
+	for (i = 0; i < AE_CA0113_OUT_SET_COMMANDS; i++)
+		ca0113_mmio_command_set(codec, out_cmds->group[i],
+				out_cmds->target[i],
+				out_cmds->vals[spec->cur_out_type][i]);
 }
 
 static int ca0132_alt_set_full_range_speaker(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
+	int quirk = ca0132_quirk(spec);
 	unsigned int tmp;
 	int err;
 
@@ -4426,7 +4474,7 @@ static int ca0132_alt_set_full_range_speaker(struct hda_codec *codec)
 	 * Only the AE series cards set this value when setting full-range,
 	 * and it's always 1.0f.
 	 */
-	if (ca0132_quirk(spec) == QUIRK_AE5) {
+	if (quirk == QUIRK_AE5 || quirk == QUIRK_AE7) {
 		err = dspio_set_uint_param(codec, 0x96,
 				SPEAKER_FULL_RANGE_SURROUND_L_R, FLOAT_ONE);
 		if (err < 0)
-- 
cgit v1.2.3


From ed93f9750c6c2ed371347d0aac3dcd31cb9cf256 Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:37 -0400
Subject: ALSA: hda/ca0132 - Add AE-7 microphone selection commands.

Add AE-7 quirk data for setting of microphone. The AE-7 has no front
panel connector, so only rear-mic/line-in have new commands.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-19-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 22 +++++++++++++++++++++-
 1 file changed, 21 insertions(+), 1 deletion(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index dcc8d29d934c..8c6e38734489 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -4997,6 +4997,15 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
 			tmp = FLOAT_THREE;
 			break;
+		case QUIRK_AE7:
+			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
+			tmp = FLOAT_THREE;
+			chipio_set_conn_rate(codec, MEM_CONNID_MICIN2,
+					SR_96_000);
+			chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2,
+					SR_96_000);
+			dspio_set_uint_param(codec, 0x80, 0x01, FLOAT_ZERO);
+			break;
 		default:
 			tmp = FLOAT_ONE;
 			break;
@@ -5042,6 +5051,14 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 		case QUIRK_AE5:
 			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
 			break;
+		case QUIRK_AE7:
+			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x3f);
+			chipio_set_conn_rate(codec, MEM_CONNID_MICIN2,
+					SR_96_000);
+			chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2,
+					SR_96_000);
+			dspio_set_uint_param(codec, 0x80, 0x01, FLOAT_ZERO);
+			break;
 		default:
 			break;
 		}
@@ -5051,7 +5068,10 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 		if (ca0132_quirk(spec) == QUIRK_R3DI)
 			chipio_set_conn_rate(codec, 0x0F, SR_96_000);
 
-		tmp = FLOAT_ZERO;
+		if (ca0132_quirk(spec) == QUIRK_AE7)
+			tmp = FLOAT_THREE;
+		else
+			tmp = FLOAT_ZERO;
 		dspio_set_uint_param(codec, 0x80, 0x00, tmp);
 
 		switch (ca0132_quirk(spec)) {
-- 
cgit v1.2.3


From 24a28eaeb23b2ff7791350bb44875ca4dc3bef3f Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:38 -0400
Subject: ALSA: hda/ca0132 - Add AE-7 custom controls.

Add headphone gain and DAC filter controls, which use the same commands
as the AE-5. Also, change input source enumerated control item count to
exclude front microphone.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-20-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 20 ++++++++++++++++----
 1 file changed, 16 insertions(+), 4 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 8c6e38734489..52f6d3740e0a 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -5839,6 +5839,13 @@ static int ca0132_alt_input_source_put(struct snd_kcontrol *kcontrol,
 	int sel = ucontrol->value.enumerated.item[0];
 	unsigned int items = IN_SRC_NUM_OF_INPUTS;
 
+	/*
+	 * The AE-7 has no front microphone, so limit items to 2: rear mic and
+	 * line-in.
+	 */
+	if (ca0132_quirk(spec) == QUIRK_AE7)
+		items = 2;
+
 	if (sel >= items)
 		return 0;
 
@@ -7029,20 +7036,25 @@ static int ca0132_build_controls(struct hda_codec *codec)
 		}
 	}
 
-	if (ca0132_quirk(spec) == QUIRK_AE5) {
+	switch (ca0132_quirk(spec)) {
+	case QUIRK_AE5:
+	case QUIRK_AE7:
 		err = ae5_add_headphone_gain_enum(codec);
 		if (err < 0)
 			return err;
 		err = ae5_add_sound_filter_enum(codec);
 		if (err < 0)
 			return err;
-	}
-
-	if (ca0132_quirk(spec) == QUIRK_ZXR) {
+		break;
+	case QUIRK_ZXR:
 		err = zxr_add_headphone_gain_switch(codec);
 		if (err < 0)
 			return err;
+		break;
+	default:
+		break;
 	}
+
 #ifdef ENABLE_TUNING_CONTROLS
 	add_tuning_ctls(codec);
 #endif
-- 
cgit v1.2.3


From 685a04a53747d07d22f5139c04ae35745200a91d Mon Sep 17 00:00:00 2001
From: Connor McAdams <conmanx360@gmail.com>
Date: Tue, 25 Aug 2020 16:10:39 -0400
Subject: ALSA: hda/ca0132 - Add AE-7 exit commands.

Add exit commands for the AE-7.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-21-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 29 +++++++++++++++++++++++++++++
 1 file changed, 29 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 52f6d3740e0a..9779978e4bc7 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -8838,6 +8838,32 @@ static void ae5_exit_chip(struct hda_codec *codec)
 	snd_hda_codec_write(codec, 0x01, 0, 0x724, 0x83);
 }
 
+static void ae7_exit_chip(struct hda_codec *codec)
+{
+	chipio_set_stream_control(codec, 0x18, 0);
+	chipio_set_stream_source_dest(codec, 0x21, 0xc8, 0xc8);
+	chipio_set_stream_channels(codec, 0x21, 0);
+	chipio_set_control_param(codec, CONTROL_PARAM_NODE_ID, 0x09);
+	chipio_set_control_param(codec, 0x20, 0x01);
+
+	chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0);
+
+	chipio_set_stream_control(codec, 0x18, 0);
+	chipio_set_stream_control(codec, 0x0c, 0);
+
+	ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x00);
+	snd_hda_codec_write(codec, 0x15, 0, 0x724, 0x83);
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83);
+	ca0113_mmio_command_set(codec, 0x30, 0x30, 0x00);
+	ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x00);
+	ca0113_mmio_gpio_set(codec, 0, false);
+	ca0113_mmio_gpio_set(codec, 1, false);
+	ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
+
+	snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
+	snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
+}
+
 static void zxr_exit_chip(struct hda_codec *codec)
 {
 	chipio_set_stream_control(codec, 0x03, 0);
@@ -9457,6 +9483,9 @@ static void ca0132_free(struct hda_codec *codec)
 	case QUIRK_AE5:
 		ae5_exit_chip(codec);
 		break;
+	case QUIRK_AE7:
+		ae7_exit_chip(codec);
+		break;
 	case QUIRK_R3DI:
 		r3di_gpio_shutdown(codec);
 		break;
-- 
cgit v1.2.3


From e9bd25885c7f799134534989b9f13c3db5ce6a16 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:25 -0500
Subject: ALSA: hda: auto_parser: remove shadowed variable declaration

Fix cppcheck warning:

sound/pci/hda/hda_auto_parser.c:353:7: style: Local variable 'i'
shadows outer variable [shadowVariable]
  int i = 0;
      ^
sound/pci/hda/hda_auto_parser.c:182:6: note: Shadowed declaration
 int i;
     ^
sound/pci/hda/hda_auto_parser.c:353:7: note: Shadow variable
  int i = 0;
      ^

It's not clear why a new declaration was added, remove and reuse
variable declared with larger scope.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_auto_parser.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 824f4ac1a8ce..4dc01647753c 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -350,7 +350,7 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
 	 */
 	if (!cfg->line_outs && cfg->hp_outs > 1 &&
 	    !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) {
-		int i = 0;
+		i = 0;
 		while (i < cfg->hp_outs) {
 			/* The real HPs should have the sequence 0x0f */
 			if ((hp_out[i].seq & 0x0f) == 0x0f) {
-- 
cgit v1.2.3


From 74610eaf310ab9b9b00a015adb54afcae038bc06 Mon Sep 17 00:00:00 2001
From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Date: Wed, 2 Sep 2020 16:21:26 -0500
Subject: ALSA: hda: (cosmetic) align function parameters

Fix cppcheck warnings and use same names in headers and C code.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200902212133.30964-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_jack.h  | 2 +-
 sound/pci/hda/hda_local.h | 8 ++++----
 2 files changed, 5 insertions(+), 5 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index 727b6d3ba454..8ceaf0ef5df1 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -77,7 +77,7 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid,
 
 struct hda_jack_callback *
 snd_hda_jack_detect_enable_callback_mst(struct hda_codec *codec, hda_nid_t nid,
-					int dev_id, hda_jack_callback_fn cb);
+					int dev_id, hda_jack_callback_fn func);
 
 /**
  * snd_hda_jack_detect_enable - enable the jack-detection
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 8c28b1022f49..5beb8aa44ecd 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -100,7 +100,7 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
 int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol);
 int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
-			  unsigned int size, unsigned int __user *tlv);
+			  unsigned int size, unsigned int __user *_tlv);
 int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_info *uinfo);
 int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol,
@@ -119,7 +119,7 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
 int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid,
 			     int ch, int dir, int idx, int mask, int val);
 int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
-			     int dir, int idx, int mask, int val);
+			     int direction, int idx, int mask, int val);
 int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch,
 			   int direction, int idx, int mask, int val);
 int snd_hda_codec_amp_init_stereo(struct hda_codec *codec, hda_nid_t nid,
@@ -198,7 +198,7 @@ int snd_hda_input_mux_put(struct hda_codec *codec,
 			  unsigned int *cur_val);
 int snd_hda_add_imux_item(struct hda_codec *codec,
 			  struct hda_input_mux *imux, const char *label,
-			  int index, int *type_index_ret);
+			  int index, int *type_idx);
 
 /*
  * Multi-channel / digital-out PCM helper
@@ -642,7 +642,7 @@ unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec,
  */
 int snd_hda_enum_helper_info(struct snd_kcontrol *kcontrol,
 			     struct snd_ctl_elem_info *uinfo,
-			     int num_entries, const char * const *texts);
+			     int num_items, const char * const *texts);
 #define snd_hda_enum_bool_helper_info(kcontrol, uinfo) \
 	snd_hda_enum_helper_info(kcontrol, uinfo, 0, NULL)
 
-- 
cgit v1.2.3


From c3cdf189276c2a63da62ee250615bd55e3fb680d Mon Sep 17 00:00:00 2001
From: Luke D Jones <luke@ljones.dev>
Date: Mon, 7 Sep 2020 20:19:59 +1200
Subject: ALSA: hda: fixup headset for ASUS GX502 laptop

The GX502 requires a few steps to enable the headset i/o: pincfg,
verbs to enable and unmute the amp used for headpone out, and
a jacksense callback to toggle output via internal or jack using
a verb.

Signed-off-by: Luke D Jones <luke@ljones.dev>
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208005
Link: https://lore.kernel.org/r/20200907081959.56186-1-luke@ljones.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 65 +++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 65 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c521a1f17096..abfc602c3b92 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -5993,6 +5993,40 @@ static void alc_fixup_disable_mic_vref(struct hda_codec *codec,
 		snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ);
 }
 
+
+static void alc294_gx502_toggle_output(struct hda_codec *codec,
+					struct hda_jack_callback *cb)
+{
+	/* The Windows driver sets the codec up in a very different way where
+	 * it appears to leave 0x10 = 0x8a20 set. For Linux we need to toggle it
+	 */
+	if (snd_hda_jack_detect_state(codec, 0x21) == HDA_JACK_PRESENT)
+		alc_write_coef_idx(codec, 0x10, 0x8a20);
+	else
+		alc_write_coef_idx(codec, 0x10, 0x0a20);
+}
+
+static void alc294_fixup_gx502_hp(struct hda_codec *codec,
+					const struct hda_fixup *fix, int action)
+{
+	/* Pin 0x21: headphones/headset mic */
+	if (!is_jack_detectable(codec, 0x21))
+		return;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		snd_hda_jack_detect_enable_callback(codec, 0x21,
+				alc294_gx502_toggle_output);
+		break;
+	case HDA_FIXUP_ACT_INIT:
+		/* Make sure to start in a correct state, i.e. if
+		 * headphones have been plugged in before powering up the system
+		 */
+		alc294_gx502_toggle_output(codec, NULL);
+		break;
+	}
+}
+
 static void  alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec,
 			      const struct hda_fixup *fix, int action)
 {
@@ -6173,6 +6207,9 @@ enum {
 	ALC285_FIXUP_THINKPAD_HEADSET_JACK,
 	ALC294_FIXUP_ASUS_HPE,
 	ALC294_FIXUP_ASUS_COEF_1B,
+	ALC294_FIXUP_ASUS_GX502_HP,
+	ALC294_FIXUP_ASUS_GX502_PINS,
+	ALC294_FIXUP_ASUS_GX502_VERBS,
 	ALC285_FIXUP_HP_GPIO_LED,
 	ALC285_FIXUP_HP_MUTE_LED,
 	ALC236_FIXUP_HP_MUTE_LED,
@@ -7338,6 +7375,33 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC
 	},
+	[ALC294_FIXUP_ASUS_GX502_PINS] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x03a11050 }, /* front HP mic */
+			{ 0x1a, 0x01a11830 }, /* rear external mic */
+			{ 0x21, 0x03211020 }, /* front HP out */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC294_FIXUP_ASUS_GX502_VERBS
+	},
+	[ALC294_FIXUP_ASUS_GX502_VERBS] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			/* set 0x15 to HP-OUT ctrl */
+			{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+			/* unmute the 0x15 amp */
+			{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC294_FIXUP_ASUS_GX502_HP
+	},
+	[ALC294_FIXUP_ASUS_GX502_HP] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc294_fixup_gx502_hp,
+	},
 	[ALC294_FIXUP_ASUS_COEF_1B] = {
 		.type = HDA_FIXUP_VERBS,
 		.v.verbs = (const struct hda_verb[]) {
@@ -7711,6 +7775,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502),
 	SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401),
+	SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS),
 	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),
-- 
cgit v1.2.3


From fc19d559b0d31b5b831fd468b10d7dadafc0d0ec Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Wed, 9 Sep 2020 10:00:41 +0800
Subject: ALSA: hda/realtek - The Mic on a RedmiBook doesn't work

The Mic connects to the Nid 0x19, but the configuration of Nid 0x19
is not defined to Mic, and also need to set the coeff to enable the
auto detection on the Nid 0x19. After this change, the Mic plugging
in or plugging out could be detected and could record the sound from
the Mic.

And the coeff value is suggested by Kailang of Realtek.

Cc: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200909020041.8967-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 13 +++++++++++++
 1 file changed, 13 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index abfc602c3b92..85e207173f5d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6228,6 +6228,7 @@ enum {
 	ALC269_FIXUP_LEMOTE_A1802,
 	ALC269_FIXUP_LEMOTE_A190X,
 	ALC256_FIXUP_INTEL_NUC8_RUGGED,
+	ALC255_FIXUP_XIAOMI_HEADSET_MIC,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -7591,6 +7592,16 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MODE
 	},
+	[ALC255_FIXUP_XIAOMI_HEADSET_MIC] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x45 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x5089 },
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC289_FIXUP_ASUS_GA401
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7888,6 +7899,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI),
 	SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101),
 	SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
+	SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802),
@@ -8065,6 +8077,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"},
 	{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
 	{.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"},
+	{.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
-- 
cgit v1.2.3


From 3f74249057827c5f6676c41c18f6be12ce1469ce Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Mon, 14 Sep 2020 14:51:18 +0800
Subject: ALSA: hda/realtek - Couldn't detect Mic if booting with headset
 plugged

We found a Mic detection issue on many Lenovo laptops, those laptops
belong to differnt models and they have different audio design like
internal mic connects to the codec or PCH, they all have this problem,
the problem is if plugging a headset before powerup/reboot the
machine, after booting up, the headphone could be detected but Mic
couldn't. If we plug out and plug in the headset, both headphone and
Mic could be detected then.

Through debugging we found the codec on those laptops are same, it is
alc257, and if we don't disable the 3k pulldown in alc256_shutup(),
the issue will be fixed. So far there is no pop noise or power
consumption regression on those laptops after this change.

Cc: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200914065118.19238-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 6 +++++-
 1 file changed, 5 insertions(+), 1 deletion(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 85e207173f5d..b6dc47da1d7b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3428,7 +3428,11 @@ static void alc256_shutup(struct hda_codec *codec)
 
 	/* 3k pull low control for Headset jack. */
 	/* NOTE: call this before clearing the pin, otherwise codec stalls */
-	alc_update_coef_idx(codec, 0x46, 0, 3 << 12);
+	/* If disable 3k pulldown control for alc257, the Mic detection will not work correctly
+	 * when booting with headset plugged. So skip setting it for the codec alc257
+	 */
+	if (codec->core.vendor_id != 0x10ec0257)
+		alc_update_coef_idx(codec, 0x46, 0, 3 << 12);
 
 	if (!spec->no_shutup_pins)
 		snd_hda_codec_write(codec, hp_pin, 0,
-- 
cgit v1.2.3


From f73bbf639b32acb6b409e188fdde5644b301978f Mon Sep 17 00:00:00 2001
From: Kai-Heng Feng <kai.heng.feng@canonical.com>
Date: Mon, 14 Sep 2020 15:02:29 +0800
Subject: ALSA: hda/realtek: Enable front panel headset LED on Lenovo
 ThinkStation P520

On Lenovo P520, the front panel headset LED isn't lit up right now.

Realtek states that the LED needs to be enabled by ALC233's GPIO2, so
let's do it accordingly to light the LED up.

Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Acked-by: Hui Wang <hui.wang@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200914070231.13192-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 7 +++++++
 1 file changed, 7 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b6dc47da1d7b..c4d615cf8de1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6055,6 +6055,7 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec,
 #include "hp_x360_helper.c"
 
 enum {
+	ALC269_FIXUP_GPIO2,
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
 	ALC269_FIXUP_DELL_M101Z,
@@ -6236,6 +6237,10 @@ enum {
 };
 
 static const struct hda_fixup alc269_fixups[] = {
+	[ALC269_FIXUP_GPIO2] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_gpio2,
+	},
 	[ALC269_FIXUP_SONY_VAIO] = {
 		.type = HDA_FIXUP_PINCTLS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -7055,6 +7060,8 @@ static const struct hda_fixup alc269_fixups[] = {
 	[ALC233_FIXUP_LENOVO_MULTI_CODECS] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc233_alc662_fixup_lenovo_dual_codecs,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_GPIO2
 	},
 	[ALC233_FIXUP_ACER_HEADSET_MIC] = {
 		.type = HDA_FIXUP_VERBS,
-- 
cgit v1.2.3


From 8f8bf00b1c00b37ef5c32a642e5c3e0a364182b6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 21 Sep 2020 12:26:32 +0200
Subject: Revert "ALSA: hda - Fix silent audio output and corrupted input on
 MSI X570-A PRO"

This reverts commit 15cbff3fbbc6 ("ALSA: hda - Fix silent audio output
and corrupted input on MSI X570-A PRO").

A regression reported by a Fedora user for MSI X570-A PRO mobo.
Until the correct solution is found out, let's revert the quirk as a
quick workaround.

Fixes: 15cbff3fbbc6 ("ALSA: hda - Fix silent audio output and corrupted input on MSI X570-A PRO")
Reported-by: Hans de Goede <hdegoede@redhat.com>
Cc: <stable@vger.kernel.org>
Cc: Dan Crawford <dnlcrwfrd@gmail.com>
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1879277
Link: https://lore.kernel.org/r/7efd2fe5-bf38-7f85-891a-eee3845d1493@redhat.com
Link: https://lore.kernel.org/r/20200921102632.31139-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 -
 1 file changed, 1 deletion(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c4d615cf8de1..d4f17b465892 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2475,7 +2475,6 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD),
-	SND_PCI_QUIRK(0x1462, 0x9c37, "MSI X570-A PRO", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS),
 	SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
 	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
-- 
cgit v1.2.3


From 69b08bdfa8181bc7babd7d81c93dc60142c4bfd3 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Mon, 21 Sep 2020 17:17:38 +0300
Subject: ALSA: hda - add Intel DG1 PCI and HDMI ids

Add Intel DG1 PCI id to list of supported HDA controllers and
add its HDMI id as well.

Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200921141741.2983072-2-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c  | 3 +++
 sound/pci/hda/patch_hdmi.c | 1 +
 2 files changed, 4 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 36a9dbc33aa0..904d3af5b99e 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2493,6 +2493,9 @@ static const struct pci_device_id azx_ids[] = {
 	/* Tigerlake-H */
 	{ PCI_DEVICE(0x8086, 0x43c8),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+	/* DG1 */
+	{ PCI_DEVICE(0x8086, 0x490d),
+	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
 	/* Elkhart Lake */
 	{ PCI_DEVICE(0x8086, 0x4b55),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 402050088090..055440740184 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -4269,6 +4269,7 @@ HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI",	patch_i915_glk_hdmi),
 HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI",	patch_i915_glk_hdmi),
 HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI",	patch_i915_icl_hdmi),
 HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI",	patch_i915_tgl_hdmi),
+HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI",	patch_i915_tgl_hdmi),
 HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI",	patch_i915_tgl_hdmi),
 HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI",	patch_i915_icl_hdmi),
 HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI",	patch_i915_icl_hdmi),
-- 
cgit v1.2.3


From 1bee263dfda57e45ad39c59a663c123a357ce38b Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Mon, 21 Sep 2020 17:17:39 +0300
Subject: ALSA: hda - controller is in GPU on the DG1

Add Intel DG1 to the CONTROLLER_IN_GPU list to ensure audio power is
requested whenever programming the controller.

Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200921141741.2983072-3-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 904d3af5b99e..61e495187b1a 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -368,7 +368,8 @@ enum {
 #define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \
 					((pci)->device == 0x0c0c) || \
 					((pci)->device == 0x0d0c) || \
-					((pci)->device == 0x160c))
+					((pci)->device == 0x160c) || \
+					((pci)->device == 0x490d))
 
 #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98)
 
-- 
cgit v1.2.3


From 13468bfa8c58731dc9ecda1cd9b22a191114f944 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Mon, 28 Sep 2020 16:01:17 +0800
Subject: ALSA: hda/realtek - set mic to auto detect on a HP AIO machine

Recently we enabled a HP AIO machine, we found the mic on the machine
couldn't record any sound and it couldn't detect plugging and
unplugging as well.

Through debugging we found the mic is set to manual detect mode, after
setting it to auto detect mode, it could detect plugging and
unplugging and could record sound.

Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200928080117.12435-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 11 +++++++++++
 1 file changed, 11 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d4f17b465892..4e5809648df1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6233,6 +6233,7 @@ enum {
 	ALC269_FIXUP_LEMOTE_A190X,
 	ALC256_FIXUP_INTEL_NUC8_RUGGED,
 	ALC255_FIXUP_XIAOMI_HEADSET_MIC,
+	ALC274_FIXUP_HP_MIC,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -7612,6 +7613,14 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC289_FIXUP_ASUS_GA401
 	},
+	[ALC274_FIXUP_HP_MIC] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x45 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x5089 },
+			{ }
+		},
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -7763,6 +7772,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT),
+	SND_PCI_QUIRK(0x103c, 0x874e, "HP", ALC274_FIXUP_HP_MIC),
 	SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
@@ -8088,6 +8098,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
 	{.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"},
 	{.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"},
 	{.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"},
+	{.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"},
 	{}
 };
 #define ALC225_STANDARD_PINS \
-- 
cgit v1.2.3


From f4794c6064a83d2c57b264bd299c367d172d1044 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Wed, 30 Sep 2020 13:51:46 +0800
Subject: ALSA: hda - Don't register a cb func if it is registered already

If the caller of enable_callback_mst() passes a cb func, the callee
function will malloc memory and link this cb func to the list
unconditionally. This will introduce problem if caller is in the
hda_codec_ops.init() since the init() will be repeatedly called in the
codec rt_resume().

So far, the patch_hdmi.c and patch_ca0132.c call enable_callback_mst()
in the hda_codec_ops.init().

Signed-off-by: Hui Wang <hui.wang@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200930055146.5665-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_jack.c | 14 +++++++++++++-
 1 file changed, 13 insertions(+), 1 deletion(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index 02cc682caa55..ded4813f8b54 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -275,6 +275,18 @@ int snd_hda_jack_detect_state_mst(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_hda_jack_detect_state_mst);
 
+static bool func_is_already_in_callback_list(struct hda_jack_tbl *jack,
+					     hda_jack_callback_fn func)
+{
+	struct hda_jack_callback *cb;
+
+	for (cb = jack->callback; cb; cb = cb->next) {
+		if (cb->func == func)
+			return true;
+	}
+	return false;
+}
+
 /**
  * snd_hda_jack_detect_enable_mst - enable the jack-detection
  * @codec: the HDA codec
@@ -297,7 +309,7 @@ snd_hda_jack_detect_enable_callback_mst(struct hda_codec *codec, hda_nid_t nid,
 	jack = snd_hda_jack_tbl_new(codec, nid, dev_id);
 	if (!jack)
 		return ERR_PTR(-ENOMEM);
-	if (func) {
+	if (func && !func_is_already_in_callback_list(jack, func)) {
 		callback = kzalloc(sizeof(*callback), GFP_KERNEL);
 		if (!callback)
 			return ERR_PTR(-ENOMEM);
-- 
cgit v1.2.3


From a0645daf16101bb9a6d87598c17e9a8b7bd60ea7 Mon Sep 17 00:00:00 2001
From: Harsha Priya <harshapriya.n@intel.com>
Date: Wed, 30 Sep 2020 14:41:39 +0300
Subject: ALSA: HDA: Early Forbid of runtime PM

For certain codecs (like Realtek), pm_runtime_forbid() is invoked
in the probe function after build_controls(). In a stress test,
its observed occasionally that runtime PM calls are invoked
before controls are built. This causes the codec to be
runtime suspended before probe completes. Because of this, not all
controls are enumerated correctly, and audio does not work until
system is rebooted.

This issue being common across all codecs, pm_runtime_forbid() is
called when the codec object is created to fix this issue.
A codec enables or disables runtime pm in its own probe function.

Multiple stress tests of 2000+ cycles has been done to test the fix.

Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Signed-off-by: Emmanuel Jillela <emmanuel.jillela@intel.com>
Reviewed-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200930114140.3839617-2-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index e96a87f1b611..a356c21edb90 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1000,6 +1000,9 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card,
 	if (err < 0)
 		goto error;
 
+	/* PM runtime needs to be enabled later after binding codec */
+	pm_runtime_forbid(&codec->core.dev);
+
 	return 0;
 
  error:
-- 
cgit v1.2.3


From 08befca40026136c14c3cd84f9e36c4cd20a358e Mon Sep 17 00:00:00 2001
From: Qiu Wenbo <qiuwenbo@kylinos.com.cn>
Date: Fri, 2 Oct 2020 20:44:54 +0800
Subject: ALSA: hda/realtek - Add mute Led support for HP Elitebook 845 G7

After installing archlinux, the mute led and micmute led are not working
at all. This patch fix this issue by applying a fixup from similar
model. These mute leds are confirmed working on HP Elitebook 845 G7.

Signed-off-by: Qiu Wenbo <qiuwenbo@kylinos.com.cn>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201002124454.7240-1-qiuwenbo@kylinos.com.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 4e5809648df1..cf896f398ae7 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7773,6 +7773,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x874e, "HP", ALC274_FIXUP_HP_MIC),
+	SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
-- 
cgit v1.2.3


From ca184355db8e60290fa34bf61c13308e6f4f50d3 Mon Sep 17 00:00:00 2001
From: Jian-Hong Pan <jhp@endlessos.org>
Date: Wed, 7 Oct 2020 13:22:25 +0800
Subject: ALSA: hda/realtek: Enable audio jacks of ASUS D700SA with ALC887

The ASUS D700SA desktop's audio (1043:2390) with ALC887 cannot detect
the headset microphone and another headphone jack until
ALC887_FIXUP_ASUS_HMIC and ALC887_FIXUP_ASUS_AUDIO quirks are applied.
The NID 0x15 maps as the headset microphone and NID 0x19 maps as another
headphone jack. Also need the function like alc887_fixup_asus_jack to
enable the audio jacks.

Signed-off-by: Jian-Hong Pan <jhp@endlessos.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201007052224.22611-1-jhp@endlessos.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 42 ++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 42 insertions(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index cf896f398ae7..c18077b39449 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1929,6 +1929,8 @@ enum {
 	ALC1220_FIXUP_CLEVO_P950,
 	ALC1220_FIXUP_CLEVO_PB51ED,
 	ALC1220_FIXUP_CLEVO_PB51ED_PINS,
+	ALC887_FIXUP_ASUS_AUDIO,
+	ALC887_FIXUP_ASUS_HMIC,
 };
 
 static void alc889_fixup_coef(struct hda_codec *codec,
@@ -2141,6 +2143,31 @@ static void alc1220_fixup_clevo_pb51ed(struct hda_codec *codec,
 	alc_fixup_headset_mode_no_hp_mic(codec, fix, action);
 }
 
+static void alc887_asus_hp_automute_hook(struct hda_codec *codec,
+					 struct hda_jack_callback *jack)
+{
+	struct alc_spec *spec = codec->spec;
+	unsigned int vref;
+
+	snd_hda_gen_hp_automute(codec, jack);
+
+	if (spec->gen.hp_jack_present)
+		vref = AC_PINCTL_VREF_80;
+	else
+		vref = AC_PINCTL_VREF_HIZ;
+	snd_hda_set_pin_ctl(codec, 0x19, PIN_HP | vref);
+}
+
+static void alc887_fixup_asus_jack(struct hda_codec *codec,
+				     const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+	if (action != HDA_FIXUP_ACT_PROBE)
+		return;
+	snd_hda_set_pin_ctl_cache(codec, 0x1b, PIN_HP);
+	spec->gen.hp_automute_hook = alc887_asus_hp_automute_hook;
+}
+
 static const struct hda_fixup alc882_fixups[] = {
 	[ALC882_FIXUP_ABIT_AW9D_MAX] = {
 		.type = HDA_FIXUP_PINS,
@@ -2398,6 +2425,20 @@ static const struct hda_fixup alc882_fixups[] = {
 		.chained = true,
 		.chain_id = ALC1220_FIXUP_CLEVO_PB51ED,
 	},
+	[ALC887_FIXUP_ASUS_AUDIO] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x15, 0x02a14150 }, /* use as headset mic, without its own jack detect */
+			{ 0x19, 0x22219420 },
+			{}
+		},
+	},
+	[ALC887_FIXUP_ASUS_HMIC] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc887_fixup_asus_jack,
+		.chained = true,
+		.chain_id = ALC887_FIXUP_ASUS_AUDIO,
+	},
 };
 
 static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -2431,6 +2472,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD),
 	SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V),
 	SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC),
+	SND_PCI_QUIRK(0x1043, 0x2390, "Asus D700SA", ALC887_FIXUP_ASUS_HMIC),
 	SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601),
 	SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS),
 	SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3),
-- 
cgit v1.2.3


From 148ebf548a1af366fc797fcc7d03f0bb92b12a79 Mon Sep 17 00:00:00 2001
From: Jeremy Szu <jeremy.szu@canonical.com>
Date: Thu, 8 Oct 2020 18:56:44 +0800
Subject: ALSA: hda/realtek - The front Mic on a HP machine doesn't work

On a HP ZCentral, the front Mic could not be detected.

The codec of the HP ZCentrol is alc671 and it needs to override the pin
configuration to enable the headset mic.

Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201008105645.65505-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index c18077b39449..f89506dffd5b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -9676,6 +9676,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
+	SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2),
 	SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE),
 	SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50),
 	SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A),
-- 
cgit v1.2.3


From 46394db4410bb640cccb0985dfb03ed4312cad2b Mon Sep 17 00:00:00 2001
From: Julia Lawall <Julia.Lawall@inria.fr>
Date: Sun, 11 Oct 2020 11:19:33 +0200
Subject: ALSA: hda: use semicolons rather than commas to separate statements

Replace commas with semicolons.  What is done is essentially described by
the following Coccinelle semantic patch (http://coccinelle.lip6.fr/):

// <smpl>
@@ expression e1,e2; @@
e1
-,
+;
e2
... when any
// </smpl>

Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr>
Link: https://lore.kernel.org/r/1602407979-29038-3-git-send-email-Julia.Lawall@inria.fr
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 2 +-
 sound/pci/hda/patch_hdmi.c   | 2 +-
 2 files changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 9779978e4bc7..2b38b2a716a1 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -3114,7 +3114,7 @@ static int dspxfr_one_seg(struct hda_codec *codec,
 	}
 
 	data = fls->data;
-	chip_addx = fls->chip_addr,
+	chip_addx = fls->chip_addr;
 	words_to_write = fls->count;
 
 	if (!words_to_write)
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 055440740184..0ffbfcb91256 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2451,7 +2451,7 @@ static int alloc_generic_hdmi(struct hda_codec *codec)
 	spec->chmap.ops.get_chmap = hdmi_get_chmap;
 	spec->chmap.ops.set_chmap = hdmi_set_chmap;
 	spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached;
-	spec->chmap.ops.get_spk_alloc = hdmi_get_spk_alloc,
+	spec->chmap.ops.get_spk_alloc = hdmi_get_spk_alloc;
 
 	codec->spec = spec;
 	hdmi_array_init(spec, 4);
-- 
cgit v1.2.3


From a6e7d0a4bdb02a7a3ffe0b44aaa8842b7efdd056 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Mon, 12 Oct 2020 13:27:04 +0300
Subject: ALSA: hda: fix jack detection with Realtek codecs when in D3

In case HDA controller becomes active, but codec is runtime suspended,
jack detection is not successful and no interrupt is raised. This has
been observed with multiple Realtek codecs and HDA controllers from
different vendors. Bug does not occur if both codec and controller are
active, or both are in suspend. Bug can be easily hit on desktop systems
with no built-in speaker.

The problem can be fixed by powering up the codec once after every
controller runtime resume. Even if codec goes back to suspend later, the
jack detection will continue to work. Add a flag to 'hda_codec' to
describe codecs that require this flow from the controller driver.
Modify __azx_runtime_resume() to use pm_request_resume() to make the
intent clearer.

Mark all Realtek codecs with the new forced_resume flag.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=209379
Cc: Kailang Yang <kailang@realtek.com>
Co-developed-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201012102704.794423-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/hda_codec.h     |  1 +
 sound/pci/hda/hda_intel.c     | 14 ++++++++------
 sound/pci/hda/patch_realtek.c |  1 +
 3 files changed, 10 insertions(+), 6 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h
index 0fea49bfc5e8..73827b7d17e0 100644
--- a/include/sound/hda_codec.h
+++ b/include/sound/hda_codec.h
@@ -253,6 +253,7 @@ struct hda_codec {
 	unsigned int force_pin_prefix:1; /* Add location prefix */
 	unsigned int link_down_at_suspend:1; /* link down at runtime suspend */
 	unsigned int relaxed_resume:1;	/* don't resume forcibly for jack */
+	unsigned int forced_resume:1; /* forced resume for jack */
 	unsigned int mst_no_extra_pcms:1; /* no backup PCMs for DP-MST */
 
 #ifdef CONFIG_PM
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 61e495187b1a..749b88090970 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1002,12 +1002,14 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt)
 	azx_init_pci(chip);
 	hda_intel_init_chip(chip, true);
 
-	if (status && from_rt) {
-		list_for_each_codec(codec, &chip->bus)
-			if (!codec->relaxed_resume &&
-			    (status & (1 << codec->addr)))
-				schedule_delayed_work(&codec->jackpoll_work,
-						      codec->jackpoll_interval);
+	if (from_rt) {
+		list_for_each_codec(codec, &chip->bus) {
+			if (codec->relaxed_resume)
+				continue;
+
+			if (codec->forced_resume || (status & (1 << codec->addr)))
+				pm_request_resume(hda_codec_dev(codec));
+		}
 	}
 
 	/* power down again for link-controlled chips */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f89506dffd5b..f2398721ac1e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1150,6 +1150,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid)
 	codec->single_adc_amp = 1;
 	/* FIXME: do we need this for all Realtek codec models? */
 	codec->spdif_status_reset = 1;
+	codec->forced_resume = 1;
 	codec->patch_ops = alc_patch_ops;
 
 	err = alc_codec_rename_from_preset(codec);
-- 
cgit v1.2.3


From ce1558c285f9ad04c03b46833a028230771cc0a7 Mon Sep 17 00:00:00 2001
From: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Date: Tue, 13 Oct 2020 18:26:28 +0300
Subject: ALSA: hda/hdmi: fix incorrect locking in hdmi_pcm_close

A race exists between closing a PCM and update of ELD data. In
hdmi_pcm_close(), hinfo->nid value is modified without taking
spec->pcm_lock. If this happens concurrently while processing an ELD
update in hdmi_pcm_setup_pin(), converter assignment may be done
incorrectly.

This bug was found by hitting a WARN_ON in snd_hda_spdif_ctls_assign()
in a HDMI receiver connection stress test:

[2739.684569] WARNING: CPU: 5 PID: 2090 at sound/pci/hda/patch_hdmi.c:1898 check_non_pcm_per_cvt+0x41/0x50 [snd_hda_codec_hdmi]
...
[2739.684707] Call Trace:
[2739.684720]  update_eld+0x121/0x5a0 [snd_hda_codec_hdmi]
[2739.684736]  hdmi_present_sense+0x21e/0x3b0 [snd_hda_codec_hdmi]
[2739.684750]  check_presence_and_report+0x81/0xd0 [snd_hda_codec_hdmi]
[2739.684842]  intel_audio_codec_enable+0x122/0x190 [i915]

Fixes: 42b2987079ec ("ALSA: hda - hdmi playback without monitor in dynamic pcm bind mode")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201013152628.920764-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_hdmi.c | 20 ++++++++++++--------
 1 file changed, 12 insertions(+), 8 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 0ffbfcb91256..ccd1df059654 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2046,22 +2046,25 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
 	int pinctl;
 	int err = 0;
 
+	mutex_lock(&spec->pcm_lock);
 	if (hinfo->nid) {
 		pcm_idx = hinfo_to_pcm_index(codec, hinfo);
-		if (snd_BUG_ON(pcm_idx < 0))
-			return -EINVAL;
+		if (snd_BUG_ON(pcm_idx < 0)) {
+			err = -EINVAL;
+			goto unlock;
+		}
 		cvt_idx = cvt_nid_to_cvt_index(codec, hinfo->nid);
-		if (snd_BUG_ON(cvt_idx < 0))
-			return -EINVAL;
+		if (snd_BUG_ON(cvt_idx < 0)) {
+			err = -EINVAL;
+			goto unlock;
+		}
 		per_cvt = get_cvt(spec, cvt_idx);
-
 		snd_BUG_ON(!per_cvt->assigned);
 		per_cvt->assigned = 0;
 		hinfo->nid = 0;
 
 		azx_stream(get_azx_dev(substream))->stripe = 0;
 
-		mutex_lock(&spec->pcm_lock);
 		snd_hda_spdif_ctls_unassign(codec, pcm_idx);
 		clear_bit(pcm_idx, &spec->pcm_in_use);
 		pin_idx = hinfo_to_pin_index(codec, hinfo);
@@ -2091,10 +2094,11 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
 		per_pin->setup = false;
 		per_pin->channels = 0;
 		mutex_unlock(&per_pin->lock);
-	unlock:
-		mutex_unlock(&spec->pcm_lock);
 	}
 
+unlock:
+	mutex_unlock(&spec->pcm_lock);
+
 	return err;
 }
 
-- 
cgit v1.2.3


From 9ce88a13b3016436441fec0b8b00ce8116f91269 Mon Sep 17 00:00:00 2001
From: Colin Ian King <colin.king@canonical.com>
Date: Fri, 16 Oct 2020 23:49:13 +0100
Subject: ALSA: hda/ca0132: make some const arrays static, makes object smaller

Don't populate const arrays on the stack but instead make them
static. Makes the object code smaller by 57 bytes.

Before:
   text	   data	    bss	    dec	    hex	filename
 173256	  38016	    192	 211464	  33a08	sound/pci/hda/patch_ca0132.o

After:
   text	   data	    bss	    dec	    hex	filename
 172879	  38336	    192	 211407	  339cf	sound/pci/hda/patch_ca0132.o

(gcc version 10.2.0)

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Link: https://lore.kernel.org/r/20201016224913.687724-1-colin.king@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0132.c | 18 ++++++++++++------
 1 file changed, 12 insertions(+), 6 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 2b38b2a716a1..e0c38f2735c6 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -7910,8 +7910,12 @@ static void ae7_post_dsp_asi_stream_setup(struct hda_codec *codec)
 
 static void ae7_post_dsp_pll_setup(struct hda_codec *codec)
 {
-	const unsigned int addr[] = { 0x41, 0x45, 0x40, 0x43, 0x51 };
-	const unsigned int data[] = { 0xc8, 0xcc, 0xcb, 0xc7, 0x8d };
+	static const unsigned int addr[] = {
+		0x41, 0x45, 0x40, 0x43, 0x51
+	};
+	static const unsigned int data[] = {
+		0xc8, 0xcc, 0xcb, 0xc7, 0x8d
+	};
 	unsigned int i;
 
 	for (i = 0; i < ARRAY_SIZE(addr); i++) {
@@ -7925,10 +7929,12 @@ static void ae7_post_dsp_pll_setup(struct hda_codec *codec)
 static void ae7_post_dsp_asi_setup_ports(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
-	const unsigned int target[] = { 0x0b, 0x04, 0x06, 0x0a, 0x0c, 0x11,
-					0x12, 0x13, 0x14 };
-	const unsigned int data[]   = { 0x12, 0x00, 0x48, 0x05, 0x5f, 0xff,
-					0xff, 0xff, 0x7f };
+	static const unsigned int target[] = {
+		0x0b, 0x04, 0x06, 0x0a, 0x0c, 0x11, 0x12, 0x13, 0x14
+	};
+	static const unsigned int data[] = {
+		0x12, 0x00, 0x48, 0x05, 0x5f, 0xff, 0xff, 0xff, 0x7f
+	};
 	unsigned int i;
 
 	mutex_lock(&spec->chipio_mutex);
-- 
cgit v1.2.3


From 033e4040d453f1f7111e5957a54f3019eb089cc6 Mon Sep 17 00:00:00 2001
From: Hui Wang <hui.wang@canonical.com>
Date: Thu, 22 Oct 2020 11:02:21 +0800
Subject: ALSA: hda - Fix the return value if cb func is already registered

If the cb function is already registered, should return the pointer
of the structure hda_jack_callback which contains this cb func, but
instead it returns the NULL.

Now fix it by replacing func_is_already_in_callback_list() with
find_callback_from_list().

Fixes: f4794c6064a8 ("ALSA: hda - Don't register a cb func if it is registered already")
Reported-and-suggested-by: Dan Carpenter <dan.carpenter@oracle.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20201022030221.22393-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_jack.c | 18 +++++++++++++-----
 1 file changed, 13 insertions(+), 5 deletions(-)

(limited to 'sound/pci/hda')

diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index ded4813f8b54..588059428d8f 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -275,16 +275,21 @@ int snd_hda_jack_detect_state_mst(struct hda_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_hda_jack_detect_state_mst);
 
-static bool func_is_already_in_callback_list(struct hda_jack_tbl *jack,
-					     hda_jack_callback_fn func)
+static struct hda_jack_callback *
+find_callback_from_list(struct hda_jack_tbl *jack,
+			hda_jack_callback_fn func)
 {
 	struct hda_jack_callback *cb;
 
+	if (!func)
+		return NULL;
+
 	for (cb = jack->callback; cb; cb = cb->next) {
 		if (cb->func == func)
-			return true;
+			return cb;
 	}
-	return false;
+
+	return NULL;
 }
 
 /**
@@ -309,7 +314,10 @@ snd_hda_jack_detect_enable_callback_mst(struct hda_codec *codec, hda_nid_t nid,
 	jack = snd_hda_jack_tbl_new(codec, nid, dev_id);
 	if (!jack)
 		return ERR_PTR(-ENOMEM);
-	if (func && !func_is_already_in_callback_list(jack, func)) {
+
+	callback = find_callback_from_list(jack, func);
+
+	if (func && !callback) {
 		callback = kzalloc(sizeof(*callback), GFP_KERNEL);
 		if (!callback)
 			return ERR_PTR(-ENOMEM);
-- 
cgit v1.2.3