[PATCH 00/20] ALSA: hda/ca0132 - Add AE-7 support/Control cleanup.

[PATCH 00/20] ALSA: hda/ca0132 - Add AE-7 support/Control cleanup.

[Connor McAdams]

This patch series adds support for the Sound Blaster AE-7 along with
adding new controls and cleaning up the old ones. I have personally
tested the Recon3D, Recon3Di, Sound Blaster Z/ZxR, and AE-5 cards to
confirm they work with these patches. Another user has confirmed that
they work on the AE-7.

Connor McAdams (20):
  ALSA: hda/ca0132 - Cleanup ca0132_mmio_init function.
  ALSA: hda/ca0132 - Add speaker tuning initialization commands.
  ALSA: hda/ca0132 - Add surround channel config control.
  ALSA: hda/ca0132 - Add full-range speaker selection controls.
  ALSA: hda/ca0132 - Add bass redirection controls.
  ALSA: hda/ca0132 - Remove surround output selection.
  ALSA: hda/ca0132 - Clean up ca0132_alt_out_select.
  ALSA: hda/ca0132 - Add quirk output selection structures.
  ALSA: hda/ca0132 - Fix Recon3D Center/LFE output.
  ALSA: hda/ca0132 - Add new quirk ID for SoundBlaster AE-7.
  ALSA: hda/ca0132 - Add SoundBlaster AE-7 pincfg.
  ALSA: hda/ca0132 - Set AE-7 bools and select mixer.
  ALSA: hda/ca0132 - Add ca0132_mmio_init data for SoundBlaster AE-7.
  ALSA: hda/ca0132 - Add pre-init function for SoundBlaster AE-7.
  ALSA: hda/ca0132 - Add init data for SoundBlaster AE-7.
  ALSA: hda/ca0132 - Add DSP setup functions for AE-7.
  ALSA: hda/ca0132 - Add output selection for SoundBlaster AE-7.
  ALSA: hda/ca0132 - Add AE-7 microphone selection commands.
  ALSA: hda/ca0132 - Add AE-7 custom controls.
  ALSA: hda/ca0132 - Add AE-7 exit commands.

 sound/pci/hda/patch_ca0132.c | 1794 +++++++++++++++++++++++++++-------
 1 file changed, 1430 insertions(+), 364 deletions(-)

-- 
2.20.1

[PATCH 01/20] ALSA: hda/ca0132 - Cleanup ca0132_mmio_init function.

[Connor McAdams]

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>
---
 sound/pci/hda/patch_ca0132.c | 158 ++++++++++++++++++++++-------------
 1 file changed, 99 insertions(+), 59 deletions(-)

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;
 	}
 }
 
-- 
2.20.1

[PATCH 02/20] ALSA: hda/ca0132 - Add speaker tuning initialization commands.

[Connor McAdams]

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

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
---
 sound/pci/hda/patch_ca0132.c | 119 +++++++++++++++++++++++++++++++++++
 1 file changed, 119 insertions(+)

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);
 }
 
-- 
2.20.1

[PATCH 03/20] ALSA: hda/ca0132 - Add surround channel config control.

[Connor McAdams]

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>
---
 sound/pci/hda/patch_ca0132.c | 110 ++++++++++++++++++++++++++++++++++-
 1 file changed, 108 insertions(+), 2 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 2e664aeee1c4..dd580f4b741d 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("Output: Channels",
+				    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);
-- 
2.20.1

Re: [PATCH 03/20] ALSA: hda/ca0132 - Add surround channel config control.

[Takashi Iwai]

On Fri, 21 Aug 2020 20:52:20 +0200,
Connor McAdams wrote:
> 
> +static int ca0132_alt_add_speaker_channel_cfg_enum(struct hda_codec *codec)
> +{
> +	struct snd_kcontrol_new knew =
> +		HDA_CODEC_MUTE_MONO("Output: Channels",
> +				    SPEAKER_CHANNEL_CFG_ENUM, 1, 0, HDA_OUTPUT);

This isn't good as a control element name.  It's not only specific
about the purpose, also the use of a letter like ':' may confuse the
user-space tools.  Please choose a different name like "Surround
Channel Mode", "Surround Channel Config" or such one.


thanks,

Takashi

[PATCH 04/20] ALSA: hda/ca0132 - Add full-range speaker selection controls.

[Connor McAdams]

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>
---
 sound/pci/hda/patch_ca0132.c | 117 +++++++++++++++++++++++++++++++++++
 1 file changed, 117 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index dd580f4b741d..c9cd4fc218fe 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("Output: Full-Range Front Playback Switch",
+				    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("Output: Full-Range Rear Playback Switch",
+				    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];
-- 
2.20.1

[PATCH 05/20] ALSA: hda/ca0132 - Add bass redirection controls.

[Connor McAdams]

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>
---
 sound/pci/hda/patch_ca0132.c | 130 +++++++++++++++++++++++++++++++++--
 1 file changed, 126 insertions(+), 4 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index c9cd4fc218fe..d97358406401 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 = "Output: Bass Redirection Playback Volume";
+	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 = "Output: Bass Redirection Playback Switch";
+	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;
-- 
2.20.1

Re: [PATCH 05/20] ALSA: hda/ca0132 - Add bass redirection controls.

[Takashi Iwai]

On Fri, 21 Aug 2020 20:52:22 +0200,
Connor McAdams wrote:
> 
> +static int ca0132_alt_add_bass_redirection_crossover(struct hda_codec *codec)
> +{
> +	const char *namestr = "Output: Bass Redirection Playback Volume";
> +	struct snd_kcontrol_new knew =
> +		HDA_CODEC_VOLUME_MONO(namestr, BASS_REDIRECTION_XOVER, 1, 0,
> +				HDA_OUTPUT);

This is no ideal name string for a control element, too.
Better to omit "Output:" prefix at least.


thanks,

Takashi

[PATCH 06/20] ALSA: hda/ca0132 - Remove surround output selection.

[Connor McAdams]

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>
---
 sound/pci/hda/patch_ca0132.c | 151 +++++++++++------------------------
 1 file changed, 46 insertions(+), 105 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index d97358406401..f0fa1e8f41da 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;
-- 
2.20.1

[PATCH 07/20] ALSA: hda/ca0132 - Clean up ca0132_alt_out_select.

[Connor McAdams]

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>
---
 sound/pci/hda/patch_ca0132.c | 142 ++++++++++++++---------------------
 1 file changed, 57 insertions(+), 85 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index f0fa1e8f41da..ee25265ab600 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;
 
-- 
2.20.1

[PATCH 08/20] ALSA: hda/ca0132 - Add quirk output selection structures.

[Connor McAdams]

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>
---
 sound/pci/hda/patch_ca0132.c | 336 +++++++++++++++++++++++++----------
 1 file changed, 241 insertions(+), 95 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index ee25265ab600..ac9dcaf69a2e 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:
-- 
2.20.1

[PATCH 09/20] ALSA: hda/ca0132 - Fix Recon3D Center/LFE output.

[Connor McAdams]

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

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
---
 sound/pci/hda/patch_ca0132.c | 6 ++++++
 1 file changed, 6 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index ac9dcaf69a2e..9c61a10114aa 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++) {
-- 
2.20.1

[PATCH 10/20] ALSA: hda/ca0132 - Add new quirk ID for SoundBlaster AE-7.

[Connor McAdams]

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

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
---
 sound/pci/hda/patch_ca0132.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 9c61a10114aa..57cb63ea88e6 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),
 	{}
 };
 
-- 
2.20.1

[PATCH 11/20] ALSA: hda/ca0132 - Add SoundBlaster AE-7 pincfg.

[Connor McAdams]

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

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
---
 sound/pci/hda/patch_ca0132.c | 19 +++++++++++++++++++
 1 file changed, 19 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 57cb63ea88e6..5aad9d8ee5e4 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 */
-- 
2.20.1

[PATCH 12/20] ALSA: hda/ca0132 - Set AE-7 bools and select mixer.

[Connor McAdams]

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

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
---
 sound/pci/hda/patch_ca0132.c | 5 +++++
 1 file changed, 5 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 5aad9d8ee5e4..ffad29ac7290 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;
-- 
2.20.1

[PATCH 13/20] ALSA: hda/ca0132 - Add ca0132_mmio_init data for SoundBlaster AE-7.

[Connor McAdams]

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

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
---
 sound/pci/hda/patch_ca0132.c | 20 +++++++++++++++++++-
 1 file changed, 19 insertions(+), 1 deletion(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index ffad29ac7290..41838a5587b6 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)
-- 
2.20.1

[PATCH 14/20] ALSA: hda/ca0132 - Add pre-init function for SoundBlaster AE-7.

[Connor McAdams]

Add pre DSP initialization function for the AE-7.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
---
 sound/pci/hda/patch_ca0132.c | 70 ++++++++++++++++++++++++++++--------
 1 file changed, 55 insertions(+), 15 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 41838a5587b6..9a1491d33819 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);
-- 
2.20.1

[PATCH 15/20] ALSA: hda/ca0132 - Add init data for SoundBlaster AE-7.

[Connor McAdams]

Add initialization data for the SoundBlaster AE-7 card.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
---
 sound/pci/hda/patch_ca0132.c | 14 ++++++++++++++
 1 file changed, 14 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 9a1491d33819..4ec93a070d73 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);
-- 
2.20.1

[PATCH 16/20] ALSA: hda/ca0132 - Add DSP setup functions for AE-7.

[Connor McAdams]

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

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
---
 sound/pci/hda/patch_ca0132.c | 290 +++++++++++++++++++++++++++++++++++
 1 file changed, 290 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 4ec93a070d73..040233b33074 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);
-- 
2.20.1

[PATCH 17/20] ALSA: hda/ca0132 - Add output selection for SoundBlaster AE-7.

[Connor McAdams]

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>
---
 sound/pci/hda/patch_ca0132.c | 88 ++++++++++++++++++++++++++++--------
 1 file changed, 68 insertions(+), 20 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 040233b33074..ad594c2d47ec 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)
-- 
2.20.1

[PATCH 18/20] ALSA: hda/ca0132 - Add AE-7 microphone selection commands.

[Connor McAdams]

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>
---
 sound/pci/hda/patch_ca0132.c | 22 +++++++++++++++++++++-
 1 file changed, 21 insertions(+), 1 deletion(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index ad594c2d47ec..933f3b0be63e 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)) {
-- 
2.20.1

[PATCH 19/20] ALSA: hda/ca0132 - Add AE-7 custom controls.

[Connor McAdams]

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>
---
 sound/pci/hda/patch_ca0132.c | 20 ++++++++++++++++----
 1 file changed, 16 insertions(+), 4 deletions(-)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 933f3b0be63e..9c70d85b21e0 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
-- 
2.20.1

[PATCH 20/20] ALSA: hda/ca0132 - Add AE-7 exit commands.

[Connor McAdams]

Add exit commands for the AE-7.

Signed-off-by: Connor McAdams <conmanx360@gmail.com>
---
 sound/pci/hda/patch_ca0132.c | 29 +++++++++++++++++++++++++++++
 1 file changed, 29 insertions(+)

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 9c70d85b21e0..4d26440eceb0 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;
-- 
2.20.1