Google Git
Sign in
android-kvm / linux / df365887f83dda9386a02ea1aa48b3b566473c64 / . / sound / pci / hda / patch_ca0132.c
blob: 208933792787dd27d480db2b4b75fb050360c000 [file] [log] [blame]
// SPDX-License-Identifier: GPL-2.0-or-later
/*
* HD audio interface patch for Creative CA0132 chip
*
* Copyright (c) 2011, Creative Technology Ltd.
*
* Based on patch_ca0110.c
* Copyright (c) 2008 Takashi Iwai <tiwai@suse.de>
*/
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/mutex.h>
#include <linux/module.h>
#include <linux/firmware.h>
#include <linux/kernel.h>
#include <linux/types.h>
#include <linux/io.h>
#include <linux/pci.h>
#include <asm/io.h>
#include <sound/core.h>
#include <sound/hda_codec.h>
#include "hda_local.h"
#include "hda_auto_parser.h"
#include "hda_jack.h"
#include "ca0132_regs.h"
/* Enable this to see controls for tuning purpose. */
/*#define ENABLE_TUNING_CONTROLS*/
#ifdef ENABLE_TUNING_CONTROLS
#include <sound/tlv.h>
#endif
#define FLOAT_ZERO 0x00000000
#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
#define UNSOL_TAG_DSP 0x16
#define DSP_DMA_WRITE_BUFLEN_INIT (1UL<<18)
#define DSP_DMA_WRITE_BUFLEN_OVLY (1UL<<15)
#define DMA_TRANSFER_FRAME_SIZE_NWORDS 8
#define DMA_TRANSFER_MAX_FRAME_SIZE_NWORDS 32
#define DMA_OVERLAY_FRAME_SIZE_NWORDS 2
#define MASTERCONTROL 0x80
#define MASTERCONTROL_ALLOC_DMA_CHAN 10
#define MASTERCONTROL_QUERY_SPEAKER_EQ_ADDRESS 60
#define WIDGET_CHIP_CTRL 0x15
#define WIDGET_DSP_CTRL 0x16
#define MEM_CONNID_MICIN1 3
#define MEM_CONNID_MICIN2 5
#define MEM_CONNID_MICOUT1 12
#define MEM_CONNID_MICOUT2 14
#define MEM_CONNID_WUH 10
#define MEM_CONNID_DSP 16
#define MEM_CONNID_DMIC 100
#define SCP_SET 0
#define SCP_GET 1
#define EFX_FILE "ctefx.bin"
#define DESKTOP_EFX_FILE "ctefx-desktop.bin"
#define R3DI_EFX_FILE "ctefx-r3di.bin"
#ifdef CONFIG_SND_HDA_CODEC_CA0132_DSP
MODULE_FIRMWARE(EFX_FILE);
MODULE_FIRMWARE(DESKTOP_EFX_FILE);
MODULE_FIRMWARE(R3DI_EFX_FILE);
#endif
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,
};
enum {
DIGITAL_MIC,
LINE_MIC_IN
};
/* Strings for Input Source Enum Control */
static const char *const in_src_str[3] = { "Microphone", "Line In", "Front Microphone" };
#define IN_SRC_NUM_OF_INPUTS 3
enum {
REAR_MIC,
REAR_LINE_IN,
FRONT_MIC,
};
enum {
#define VNODE_START_NID 0x80
VNID_SPK = VNODE_START_NID, /* Speaker vnid */
VNID_MIC,
VNID_HP_SEL,
VNID_AMIC1_SEL,
VNID_HP_ASEL,
VNID_AMIC1_ASEL,
VNODE_END_NID,
#define VNODES_COUNT (VNODE_END_NID - VNODE_START_NID)
#define EFFECT_START_NID 0x90
#define OUT_EFFECT_START_NID EFFECT_START_NID
SURROUND = OUT_EFFECT_START_NID,
CRYSTALIZER,
DIALOG_PLUS,
SMART_VOLUME,
X_BASS,
EQUALIZER,
OUT_EFFECT_END_NID,
#define OUT_EFFECTS_COUNT (OUT_EFFECT_END_NID - OUT_EFFECT_START_NID)
#define IN_EFFECT_START_NID OUT_EFFECT_END_NID
ECHO_CANCELLATION = IN_EFFECT_START_NID,
VOICE_FOCUS,
MIC_SVM,
NOISE_REDUCTION,
IN_EFFECT_END_NID,
#define IN_EFFECTS_COUNT (IN_EFFECT_END_NID - IN_EFFECT_START_NID)
VOICEFX = IN_EFFECT_END_NID,
PLAY_ENHANCEMENT,
CRYSTAL_VOICE,
EFFECT_END_NID,
OUTPUT_SOURCE_ENUM,
INPUT_SOURCE_ENUM,
XBASS_XOVER,
EQ_PRESET_ENUM,
SMART_VOLUME_ENUM,
MIC_BOOST_ENUM,
AE5_HEADPHONE_GAIN_ENUM,
AE5_SOUND_FILTER_ENUM,
ZXR_HEADPHONE_GAIN,
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)
};
/* Effects values size*/
#define EFFECT_VALS_MAX_COUNT 12
/*
* Default values for the effect slider controls, they are in order of their
* effect NID's. Surround, Crystalizer, Dialog Plus, Smart Volume, and then
* X-bass.
*/
static const unsigned int effect_slider_defaults[] = {67, 65, 50, 74, 50};
/* Amount of effect level sliders for ca0132_alt controls. */
#define EFFECT_LEVEL_SLIDERS 5
/* Latency introduced by DSP blocks in milliseconds. */
#define DSP_CAPTURE_INIT_LATENCY 0
#define DSP_CRYSTAL_VOICE_LATENCY 124
#define DSP_PLAYBACK_INIT_LATENCY 13
#define DSP_PLAY_ENHANCEMENT_LATENCY 30
#define DSP_SPEAKER_OUT_LATENCY 7
struct ct_effect {
char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
hda_nid_t nid;
int mid; /*effect module ID*/
int reqs[EFFECT_VALS_MAX_COUNT]; /*effect module request*/
int direct; /* 0:output; 1:input*/
int params; /* number of default non-on/off params */
/*effect default values, 1st is on/off. */
unsigned int def_vals[EFFECT_VALS_MAX_COUNT];
};
#define EFX_DIR_OUT 0
#define EFX_DIR_IN 1
static const struct ct_effect ca0132_effects[EFFECTS_COUNT] = {
{ .name = "Surround",
.nid = SURROUND,
.mid = 0x96,
.reqs = {0, 1},
.direct = EFX_DIR_OUT,
.params = 1,
.def_vals = {0x3F800000, 0x3F2B851F}
},
{ .name = "Crystalizer",
.nid = CRYSTALIZER,
.mid = 0x96,
.reqs = {7, 8},
.direct = EFX_DIR_OUT,
.params = 1,
.def_vals = {0x3F800000, 0x3F266666}
},
{ .name = "Dialog Plus",
.nid = DIALOG_PLUS,
.mid = 0x96,
.reqs = {2, 3},
.direct = EFX_DIR_OUT,
.params = 1,
.def_vals = {0x00000000, 0x3F000000}
},
{ .name = "Smart Volume",
.nid = SMART_VOLUME,
.mid = 0x96,
.reqs = {4, 5, 6},
.direct = EFX_DIR_OUT,
.params = 2,
.def_vals = {0x3F800000, 0x3F3D70A4, 0x00000000}
},
{ .name = "X-Bass",
.nid = X_BASS,
.mid = 0x96,
.reqs = {24, 23, 25},
.direct = EFX_DIR_OUT,
.params = 2,
.def_vals = {0x3F800000, 0x42A00000, 0x3F000000}
},
{ .name = "Equalizer",
.nid = EQUALIZER,
.mid = 0x96,
.reqs = {9, 10, 11, 12, 13, 14,
15, 16, 17, 18, 19, 20},
.direct = EFX_DIR_OUT,
.params = 11,
.def_vals = {0x00000000, 0x00000000, 0x00000000, 0x00000000,
0x00000000, 0x00000000, 0x00000000, 0x00000000,
0x00000000, 0x00000000, 0x00000000, 0x00000000}
},
{ .name = "Echo Cancellation",
.nid = ECHO_CANCELLATION,
.mid = 0x95,
.reqs = {0, 1, 2, 3},
.direct = EFX_DIR_IN,
.params = 3,
.def_vals = {0x00000000, 0x3F3A9692, 0x00000000, 0x00000000}
},
{ .name = "Voice Focus",
.nid = VOICE_FOCUS,
.mid = 0x95,
.reqs = {6, 7, 8, 9},
.direct = EFX_DIR_IN,
.params = 3,
.def_vals = {0x3F800000, 0x3D7DF3B6, 0x41F00000, 0x41F00000}
},
{ .name = "Mic SVM",
.nid = MIC_SVM,
.mid = 0x95,
.reqs = {44, 45},
.direct = EFX_DIR_IN,
.params = 1,
.def_vals = {0x00000000, 0x3F3D70A4}
},
{ .name = "Noise Reduction",
.nid = NOISE_REDUCTION,
.mid = 0x95,
.reqs = {4, 5},
.direct = EFX_DIR_IN,
.params = 1,
.def_vals = {0x3F800000, 0x3F000000}
},
{ .name = "VoiceFX",
.nid = VOICEFX,
.mid = 0x95,
.reqs = {10, 11, 12, 13, 14, 15, 16, 17, 18},
.direct = EFX_DIR_IN,
.params = 8,
.def_vals = {0x00000000, 0x43C80000, 0x44AF0000, 0x44FA0000,
0x3F800000, 0x3F800000, 0x3F800000, 0x00000000,
0x00000000}
}
};
/* Tuning controls */
#ifdef ENABLE_TUNING_CONTROLS
enum {
#define TUNING_CTL_START_NID 0xC0
WEDGE_ANGLE = TUNING_CTL_START_NID,
SVM_LEVEL,
EQUALIZER_BAND_0,
EQUALIZER_BAND_1,
EQUALIZER_BAND_2,
EQUALIZER_BAND_3,
EQUALIZER_BAND_4,
EQUALIZER_BAND_5,
EQUALIZER_BAND_6,
EQUALIZER_BAND_7,
EQUALIZER_BAND_8,
EQUALIZER_BAND_9,
TUNING_CTL_END_NID
#define TUNING_CTLS_COUNT (TUNING_CTL_END_NID - TUNING_CTL_START_NID)
};
struct ct_tuning_ctl {
char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
hda_nid_t parent_nid;
hda_nid_t nid;
int mid; /*effect module ID*/
int req; /*effect module request*/
int direct; /* 0:output; 1:input*/
unsigned int def_val;/*effect default values*/
};
static const struct ct_tuning_ctl ca0132_tuning_ctls[] = {
{ .name = "Wedge Angle",
.parent_nid = VOICE_FOCUS,
.nid = WEDGE_ANGLE,
.mid = 0x95,
.req = 8,
.direct = EFX_DIR_IN,
.def_val = 0x41F00000
},
{ .name = "SVM Level",
.parent_nid = MIC_SVM,
.nid = SVM_LEVEL,
.mid = 0x95,
.req = 45,
.direct = EFX_DIR_IN,
.def_val = 0x3F3D70A4
},
{ .name = "EQ Band0",
.parent_nid = EQUALIZER,
.nid = EQUALIZER_BAND_0,
.mid = 0x96,
.req = 11,
.direct = EFX_DIR_OUT,
.def_val = 0x00000000
},
{ .name = "EQ Band1",
.parent_nid = EQUALIZER,
.nid = EQUALIZER_BAND_1,
.mid = 0x96,
.req = 12,
.direct = EFX_DIR_OUT,
.def_val = 0x00000000
},
{ .name = "EQ Band2",
.parent_nid = EQUALIZER,
.nid = EQUALIZER_BAND_2,
.mid = 0x96,
.req = 13,
.direct = EFX_DIR_OUT,
.def_val = 0x00000000
},
{ .name = "EQ Band3",
.parent_nid = EQUALIZER,
.nid = EQUALIZER_BAND_3,
.mid = 0x96,
.req = 14,
.direct = EFX_DIR_OUT,
.def_val = 0x00000000
},
{ .name = "EQ Band4",
.parent_nid = EQUALIZER,
.nid = EQUALIZER_BAND_4,
.mid = 0x96,
.req = 15,
.direct = EFX_DIR_OUT,
.def_val = 0x00000000
},
{ .name = "EQ Band5",
.parent_nid = EQUALIZER,
.nid = EQUALIZER_BAND_5,
.mid = 0x96,
.req = 16,
.direct = EFX_DIR_OUT,
.def_val = 0x00000000
},
{ .name = "EQ Band6",
.parent_nid = EQUALIZER,
.nid = EQUALIZER_BAND_6,
.mid = 0x96,
.req = 17,
.direct = EFX_DIR_OUT,
.def_val = 0x00000000
},
{ .name = "EQ Band7",
.parent_nid = EQUALIZER,
.nid = EQUALIZER_BAND_7,
.mid = 0x96,
.req = 18,
.direct = EFX_DIR_OUT,
.def_val = 0x00000000
},
{ .name = "EQ Band8",
.parent_nid = EQUALIZER,
.nid = EQUALIZER_BAND_8,
.mid = 0x96,
.req = 19,
.direct = EFX_DIR_OUT,
.def_val = 0x00000000
},
{ .name = "EQ Band9",
.parent_nid = EQUALIZER,
.nid = EQUALIZER_BAND_9,
.mid = 0x96,
.req = 20,
.direct = EFX_DIR_OUT,
.def_val = 0x00000000
}
};
#endif
/* Voice FX Presets */
#define VOICEFX_MAX_PARAM_COUNT 9
struct ct_voicefx {
char *name;
hda_nid_t nid;
int mid;
int reqs[VOICEFX_MAX_PARAM_COUNT]; /*effect module request*/
};
struct ct_voicefx_preset {
char *name; /*preset name*/
unsigned int vals[VOICEFX_MAX_PARAM_COUNT];
};
static const struct ct_voicefx ca0132_voicefx = {
.name = "VoiceFX Capture Switch",
.nid = VOICEFX,
.mid = 0x95,
.reqs = {10, 11, 12, 13, 14, 15, 16, 17, 18}
};
static const struct ct_voicefx_preset ca0132_voicefx_presets[] = {
{ .name = "Neutral",
.vals = { 0x00000000, 0x43C80000, 0x44AF0000,
0x44FA0000, 0x3F800000, 0x3F800000,
0x3F800000, 0x00000000, 0x00000000 }
},
{ .name = "Female2Male",
.vals = { 0x3F800000, 0x43C80000, 0x44AF0000,
0x44FA0000, 0x3F19999A, 0x3F866666,
0x3F800000, 0x00000000, 0x00000000 }
},
{ .name = "Male2Female",
.vals = { 0x3F800000, 0x43C80000, 0x44AF0000,
0x450AC000, 0x4017AE14, 0x3F6B851F,
0x3F800000, 0x00000000, 0x00000000 }
},
{ .name = "ScrappyKid",
.vals = { 0x3F800000, 0x43C80000, 0x44AF0000,
0x44FA0000, 0x40400000, 0x3F28F5C3,
0x3F800000, 0x00000000, 0x00000000 }
},
{ .name = "Elderly",
.vals = { 0x3F800000, 0x44324000, 0x44BB8000,
0x44E10000, 0x3FB33333, 0x3FB9999A,
0x3F800000, 0x3E3A2E43, 0x00000000 }
},
{ .name = "Orc",
.vals = { 0x3F800000, 0x43EA0000, 0x44A52000,
0x45098000, 0x3F266666, 0x3FC00000,
0x3F800000, 0x00000000, 0x00000000 }
},
{ .name = "Elf",
.vals = { 0x3F800000, 0x43C70000, 0x44AE6000,
0x45193000, 0x3F8E147B, 0x3F75C28F,
0x3F800000, 0x00000000, 0x00000000 }
},
{ .name = "Dwarf",
.vals = { 0x3F800000, 0x43930000, 0x44BEE000,
0x45007000, 0x3F451EB8, 0x3F7851EC,
0x3F800000, 0x00000000, 0x00000000 }
},
{ .name = "AlienBrute",
.vals = { 0x3F800000, 0x43BFC5AC, 0x44B28FDF,
0x451F6000, 0x3F266666, 0x3FA7D945,
0x3F800000, 0x3CF5C28F, 0x00000000 }
},
{ .name = "Robot",
.vals = { 0x3F800000, 0x43C80000, 0x44AF0000,
0x44FA0000, 0x3FB2718B, 0x3F800000,
0xBC07010E, 0x00000000, 0x00000000 }
},
{ .name = "Marine",
.vals = { 0x3F800000, 0x43C20000, 0x44906000,
0x44E70000, 0x3F4CCCCD, 0x3F8A3D71,
0x3F0A3D71, 0x00000000, 0x00000000 }
},
{ .name = "Emo",
.vals = { 0x3F800000, 0x43C80000, 0x44AF0000,
0x44FA0000, 0x3F800000, 0x3F800000,
0x3E4CCCCD, 0x00000000, 0x00000000 }
},
{ .name = "DeepVoice",
.vals = { 0x3F800000, 0x43A9C5AC, 0x44AA4FDF,
0x44FFC000, 0x3EDBB56F, 0x3F99C4CA,
0x3F800000, 0x00000000, 0x00000000 }
},
{ .name = "Munchkin",
.vals = { 0x3F800000, 0x43C80000, 0x44AF0000,
0x44FA0000, 0x3F800000, 0x3F1A043C,
0x3F800000, 0x00000000, 0x00000000 }
}
};
/* ca0132 EQ presets, taken from Windows Sound Blaster Z Driver */
#define EQ_PRESET_MAX_PARAM_COUNT 11
struct ct_eq {
char *name;
hda_nid_t nid;
int mid;
int reqs[EQ_PRESET_MAX_PARAM_COUNT]; /*effect module request*/
};
struct ct_eq_preset {
char *name; /*preset name*/
unsigned int vals[EQ_PRESET_MAX_PARAM_COUNT];
};
static const struct ct_eq ca0132_alt_eq_enum = {
.name = "FX: Equalizer Preset Switch",
.nid = EQ_PRESET_ENUM,
.mid = 0x96,
.reqs = {10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20}
};
static const struct ct_eq_preset ca0132_alt_eq_presets[] = {
{ .name = "Flat",
.vals = { 0x00000000, 0x00000000, 0x00000000,
0x00000000, 0x00000000, 0x00000000,
0x00000000, 0x00000000, 0x00000000,
0x00000000, 0x00000000 }
},
{ .name = "Acoustic",
.vals = { 0x00000000, 0x00000000, 0x3F8CCCCD,
0x40000000, 0x00000000, 0x00000000,
0x00000000, 0x00000000, 0x40000000,
0x40000000, 0x40000000 }
},
{ .name = "Classical",
.vals = { 0x00000000, 0x00000000, 0x40C00000,
0x40C00000, 0x40466666, 0x00000000,
0x00000000, 0x00000000, 0x00000000,
0x40466666, 0x40466666 }
},
{ .name = "Country",
.vals = { 0x00000000, 0xBF99999A, 0x00000000,
0x3FA66666, 0x3FA66666, 0x3F8CCCCD,
0x00000000, 0x00000000, 0x40000000,
0x40466666, 0x40800000 }
},
{ .name = "Dance",
.vals = { 0x00000000, 0xBF99999A, 0x40000000,
0x40466666, 0x40866666, 0xBF99999A,
0xBF99999A, 0x00000000, 0x00000000,
0x40800000, 0x40800000 }
},
{ .name = "Jazz",
.vals = { 0x00000000, 0x00000000, 0x00000000,
0x3F8CCCCD, 0x40800000, 0x40800000,
0x40800000, 0x00000000, 0x3F8CCCCD,
0x40466666, 0x40466666 }
},
{ .name = "New Age",
.vals = { 0x00000000, 0x00000000, 0x40000000,
0x40000000, 0x00000000, 0x00000000,
0x00000000, 0x3F8CCCCD, 0x40000000,
0x40000000, 0x40000000 }
},
{ .name = "Pop",
.vals = { 0x00000000, 0xBFCCCCCD, 0x00000000,
0x40000000, 0x40000000, 0x00000000,
0xBF99999A, 0xBF99999A, 0x00000000,
0x40466666, 0x40C00000 }
},
{ .name = "Rock",
.vals = { 0x00000000, 0xBF99999A, 0xBF99999A,
0x3F8CCCCD, 0x40000000, 0xBF99999A,
0xBF99999A, 0x00000000, 0x00000000,
0x40800000, 0x40800000 }
},
{ .name = "Vocal",
.vals = { 0x00000000, 0xC0000000, 0xBF99999A,
0xBF99999A, 0x00000000, 0x40466666,
0x40800000, 0x40466666, 0x00000000,
0x00000000, 0x3F8CCCCD }
}
};
/*
* 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.
*/
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,
};
/* 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
* some sort of update or set command to tell the DSP there's new volume info.
*/
#define DSP_VOL_OUT 0
#define DSP_VOL_IN 1
struct ct_dsp_volume_ctl {
hda_nid_t vnid;
int mid; /* module ID*/
unsigned int reqs[3]; /* scp req ID */
};
static const struct ct_dsp_volume_ctl ca0132_alt_vol_ctls[] = {
{ .vnid = VNID_SPK,
.mid = 0x32,
.reqs = {3, 4, 2}
},
{ .vnid = VNID_MIC,
.mid = 0x37,
.reqs = {2, 3, 1}
}
};
/* Values for ca0113_mmio_command_set for selecting output. */
#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 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. */
#define AE5_HEADPHONE_GAIN_PRESET_MAX_COMMANDS 4
struct ae5_headphone_gain_set {
char *name;
unsigned int vals[AE5_HEADPHONE_GAIN_PRESET_MAX_COMMANDS];
};
static const struct ae5_headphone_gain_set ae5_headphone_gain_presets[] = {
{ .name = "Low (16-31",
.vals = { 0xff, 0x2c, 0xf5, 0x32 }
},
{ .name = "Medium (32-149",
.vals = { 0x38, 0xa8, 0x3e, 0x4c }
},
{ .name = "High (150-600",
.vals = { 0xff, 0xff, 0xff, 0x7f }
}
};
struct ae5_filter_set {
char *name;
unsigned int val;
};
static const struct ae5_filter_set ae5_filter_presets[] = {
{ .name = "Slow Roll Off",
.val = 0xa0
},
{ .name = "Minimum Phase",
.val = 0xc0
},
{ .name = "Fast Roll Off",
.val = 0x80
}
};
/*
* Data structures for storing audio router remapping data. These are used to
* remap a currently active streams ports.
*/
struct chipio_stream_remap_data {
unsigned int stream_id;
unsigned int count;
unsigned int offset[16];
unsigned int value[16];
};
static const struct chipio_stream_remap_data stream_remap_data[] = {
{ .stream_id = 0x14,
.count = 0x04,
.offset = { 0x00, 0x04, 0x08, 0x0c },
.value = { 0x0001f8c0, 0x0001f9c1, 0x0001fac6, 0x0001fbc7 },
},
{ .stream_id = 0x0c,
.count = 0x0c,
.offset = { 0x00, 0x04, 0x08, 0x0c, 0x10, 0x14, 0x18, 0x1c,
0x20, 0x24, 0x28, 0x2c },
.value = { 0x0001e0c0, 0x0001e1c1, 0x0001e4c2, 0x0001e5c3,
0x0001e2c4, 0x0001e3c5, 0x0001e8c6, 0x0001e9c7,
0x0001ecc8, 0x0001edc9, 0x0001eaca, 0x0001ebcb },
},
{ .stream_id = 0x0c,
.count = 0x08,
.offset = { 0x08, 0x0c, 0x10, 0x14, 0x20, 0x24, 0x28, 0x2c },
.value = { 0x000140c2, 0x000141c3, 0x000150c4, 0x000151c5,
0x000142c8, 0x000143c9, 0x000152ca, 0x000153cb },
}
};
enum hda_cmd_vendor_io {
/* for DspIO node */
VENDOR_DSPIO_SCP_WRITE_DATA_LOW = 0x000,
VENDOR_DSPIO_SCP_WRITE_DATA_HIGH = 0x100,
VENDOR_DSPIO_STATUS = 0xF01,
VENDOR_DSPIO_SCP_POST_READ_DATA = 0x702,
VENDOR_DSPIO_SCP_READ_DATA = 0xF02,
VENDOR_DSPIO_DSP_INIT = 0x703,
VENDOR_DSPIO_SCP_POST_COUNT_QUERY = 0x704,
VENDOR_DSPIO_SCP_READ_COUNT = 0xF04,
/* for ChipIO node */
VENDOR_CHIPIO_ADDRESS_LOW = 0x000,
VENDOR_CHIPIO_ADDRESS_HIGH = 0x100,
VENDOR_CHIPIO_STREAM_FORMAT = 0x200,
VENDOR_CHIPIO_DATA_LOW = 0x300,
VENDOR_CHIPIO_DATA_HIGH = 0x400,
VENDOR_CHIPIO_8051_WRITE_DIRECT = 0x500,
VENDOR_CHIPIO_8051_READ_DIRECT = 0xD00,
VENDOR_CHIPIO_GET_PARAMETER = 0xF00,
VENDOR_CHIPIO_STATUS = 0xF01,
VENDOR_CHIPIO_HIC_POST_READ = 0x702,
VENDOR_CHIPIO_HIC_READ_DATA = 0xF03,
VENDOR_CHIPIO_8051_DATA_WRITE = 0x707,
VENDOR_CHIPIO_8051_DATA_READ = 0xF07,
VENDOR_CHIPIO_8051_PMEM_READ = 0xF08,
VENDOR_CHIPIO_8051_IRAM_WRITE = 0x709,
VENDOR_CHIPIO_8051_IRAM_READ = 0xF09,
VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE = 0x70A,
VENDOR_CHIPIO_CT_EXTENSIONS_GET = 0xF0A,
VENDOR_CHIPIO_PLL_PMU_WRITE = 0x70C,
VENDOR_CHIPIO_PLL_PMU_READ = 0xF0C,
VENDOR_CHIPIO_8051_ADDRESS_LOW = 0x70D,
VENDOR_CHIPIO_8051_ADDRESS_HIGH = 0x70E,
VENDOR_CHIPIO_FLAG_SET = 0x70F,
VENDOR_CHIPIO_FLAGS_GET = 0xF0F,
VENDOR_CHIPIO_PARAM_SET = 0x710,
VENDOR_CHIPIO_PARAM_GET = 0xF10,
VENDOR_CHIPIO_PORT_ALLOC_CONFIG_SET = 0x711,
VENDOR_CHIPIO_PORT_ALLOC_SET = 0x712,
VENDOR_CHIPIO_PORT_ALLOC_GET = 0xF12,
VENDOR_CHIPIO_PORT_FREE_SET = 0x713,
VENDOR_CHIPIO_PARAM_EX_ID_GET = 0xF17,
VENDOR_CHIPIO_PARAM_EX_ID_SET = 0x717,
VENDOR_CHIPIO_PARAM_EX_VALUE_GET = 0xF18,
VENDOR_CHIPIO_PARAM_EX_VALUE_SET = 0x718,
VENDOR_CHIPIO_DMIC_CTL_SET = 0x788,
VENDOR_CHIPIO_DMIC_CTL_GET = 0xF88,
VENDOR_CHIPIO_DMIC_PIN_SET = 0x789,
VENDOR_CHIPIO_DMIC_PIN_GET = 0xF89,
VENDOR_CHIPIO_DMIC_MCLK_SET = 0x78A,
VENDOR_CHIPIO_DMIC_MCLK_GET = 0xF8A,
VENDOR_CHIPIO_EAPD_SEL_SET = 0x78D
};
/*
* Control flag IDs
*/
enum control_flag_id {
/* Connection manager stream setup is bypassed/enabled */
CONTROL_FLAG_C_MGR = 0,
/* DSP DMA is bypassed/enabled */
CONTROL_FLAG_DMA = 1,
/* 8051 'idle' mode is disabled/enabled */
CONTROL_FLAG_IDLE_ENABLE = 2,
/* Tracker for the SPDIF-in path is bypassed/enabled */
CONTROL_FLAG_TRACKER = 3,
/* DigitalOut to Spdif2Out connection is disabled/enabled */
CONTROL_FLAG_SPDIF2OUT = 4,
/* Digital Microphone is disabled/enabled */
CONTROL_FLAG_DMIC = 5,
/* ADC_B rate is 48 kHz/96 kHz */
CONTROL_FLAG_ADC_B_96KHZ = 6,
/* ADC_C rate is 48 kHz/96 kHz */
CONTROL_FLAG_ADC_C_96KHZ = 7,
/* DAC rate is 48 kHz/96 kHz (affects all DACs) */
CONTROL_FLAG_DAC_96KHZ = 8,
/* DSP rate is 48 kHz/96 kHz */
CONTROL_FLAG_DSP_96KHZ = 9,
/* SRC clock is 98 MHz/196 MHz (196 MHz forces rate to 96 KHz) */
CONTROL_FLAG_SRC_CLOCK_196MHZ = 10,
/* SRC rate is 48 kHz/96 kHz (48 kHz disabled when clock is 196 MHz) */
CONTROL_FLAG_SRC_RATE_96KHZ = 11,
/* Decode Loop (DSP->SRC->DSP) is disabled/enabled */
CONTROL_FLAG_DECODE_LOOP = 12,
/* De-emphasis filter on DAC-1 disabled/enabled */
CONTROL_FLAG_DAC1_DEEMPHASIS = 13,
/* De-emphasis filter on DAC-2 disabled/enabled */
CONTROL_FLAG_DAC2_DEEMPHASIS = 14,
/* De-emphasis filter on DAC-3 disabled/enabled */
CONTROL_FLAG_DAC3_DEEMPHASIS = 15,
/* High-pass filter on ADC_B disabled/enabled */
CONTROL_FLAG_ADC_B_HIGH_PASS = 16,
/* High-pass filter on ADC_C disabled/enabled */
CONTROL_FLAG_ADC_C_HIGH_PASS = 17,
/* Common mode on Port_A disabled/enabled */
CONTROL_FLAG_PORT_A_COMMON_MODE = 18,
/* Common mode on Port_D disabled/enabled */
CONTROL_FLAG_PORT_D_COMMON_MODE = 19,
/* Impedance for ramp generator on Port_A 16 Ohm/10K Ohm */
CONTROL_FLAG_PORT_A_10KOHM_LOAD = 20,
/* Impedance for ramp generator on Port_D, 16 Ohm/10K Ohm */
CONTROL_FLAG_PORT_D_10KOHM_LOAD = 21,
/* ASI rate is 48kHz/96kHz */
CONTROL_FLAG_ASI_96KHZ = 22,
/* DAC power settings able to control attached ports no/yes */
CONTROL_FLAG_DACS_CONTROL_PORTS = 23,
/* Clock Stop OK reporting is disabled/enabled */
CONTROL_FLAG_CONTROL_STOP_OK_ENABLE = 24,
/* Number of control flags */
CONTROL_FLAGS_MAX = (CONTROL_FLAG_CONTROL_STOP_OK_ENABLE+1)
};
/*
* Control parameter IDs
*/
enum control_param_id {
/* 0: None, 1: Mic1In*/
CONTROL_PARAM_VIP_SOURCE = 1,
/* 0: force HDA, 1: allow DSP if HDA Spdif1Out stream is idle */
CONTROL_PARAM_SPDIF1_SOURCE = 2,
/* Port A output stage gain setting to use when 16 Ohm output
* impedance is selected*/
CONTROL_PARAM_PORTA_160OHM_GAIN = 8,
/* Port D output stage gain setting to use when 16 Ohm output
* impedance is selected*/
CONTROL_PARAM_PORTD_160OHM_GAIN = 10,
/*
* This control param name was found in the 8051 memory, and makes
* sense given the fact the AE-5 uses it and has the ASI flag set.
*/
CONTROL_PARAM_ASI = 23,
/* Stream Control */
/* Select stream with the given ID */
CONTROL_PARAM_STREAM_ID = 24,
/* Source connection point for the selected stream */
CONTROL_PARAM_STREAM_SOURCE_CONN_POINT = 25,
/* Destination connection point for the selected stream */
CONTROL_PARAM_STREAM_DEST_CONN_POINT = 26,
/* Number of audio channels in the selected stream */
CONTROL_PARAM_STREAMS_CHANNELS = 27,
/*Enable control for the selected stream */
CONTROL_PARAM_STREAM_CONTROL = 28,
/* Connection Point Control */
/* Select connection point with the given ID */
CONTROL_PARAM_CONN_POINT_ID = 29,
/* Connection point sample rate */
CONTROL_PARAM_CONN_POINT_SAMPLE_RATE = 30,
/* Node Control */
/* Select HDA node with the given ID */
CONTROL_PARAM_NODE_ID = 31
};
/*
* Dsp Io Status codes
*/
enum hda_vendor_status_dspio {
/* Success */
VENDOR_STATUS_DSPIO_OK = 0x00,
/* Busy, unable to accept new command, the host must retry */
VENDOR_STATUS_DSPIO_BUSY = 0x01,
/* SCP command queue is full */
VENDOR_STATUS_DSPIO_SCP_COMMAND_QUEUE_FULL = 0x02,
/* SCP response queue is empty */
VENDOR_STATUS_DSPIO_SCP_RESPONSE_QUEUE_EMPTY = 0x03
};
/*
* Chip Io Status codes
*/
enum hda_vendor_status_chipio {
/* Success */
VENDOR_STATUS_CHIPIO_OK = 0x00,
/* Busy, unable to accept new command, the host must retry */
VENDOR_STATUS_CHIPIO_BUSY = 0x01
};
/*
* CA0132 sample rate
*/
enum ca0132_sample_rate {
SR_6_000 = 0x00,
SR_8_000 = 0x01,
SR_9_600 = 0x02,
SR_11_025 = 0x03,
SR_16_000 = 0x04,
SR_22_050 = 0x05,
SR_24_000 = 0x06,
SR_32_000 = 0x07,
SR_44_100 = 0x08,
SR_48_000 = 0x09,
SR_88_200 = 0x0A,
SR_96_000 = 0x0B,
SR_144_000 = 0x0C,
SR_176_400 = 0x0D,
SR_192_000 = 0x0E,
SR_384_000 = 0x0F,
SR_COUNT = 0x10,
SR_RATE_UNKNOWN = 0x1F
};
enum dsp_download_state {
DSP_DOWNLOAD_FAILED = -1,
DSP_DOWNLOAD_INIT = 0,
DSP_DOWNLOADING = 1,
DSP_DOWNLOADED = 2
};
/* retrieve parameters from hda format */
#define get_hdafmt_chs(fmt) (fmt & 0xf)
#define get_hdafmt_bits(fmt) ((fmt >> 4) & 0x7)
#define get_hdafmt_rate(fmt) ((fmt >> 8) & 0x7f)
#define get_hdafmt_type(fmt) ((fmt >> 15) & 0x1)
/*
* CA0132 specific
*/
struct ca0132_spec {
const struct snd_kcontrol_new *mixers[5];
unsigned int num_mixers;
const struct hda_verb *base_init_verbs;
const struct hda_verb *base_exit_verbs;
const struct hda_verb *chip_init_verbs;
const struct hda_verb *desktop_init_verbs;
struct hda_verb *spec_init_verbs;
struct auto_pin_cfg autocfg;
/* Nodes configurations */
struct hda_multi_out multiout;
hda_nid_t out_pins[AUTO_CFG_MAX_OUTS];
hda_nid_t dacs[AUTO_CFG_MAX_OUTS];
unsigned int num_outputs;
hda_nid_t input_pins[AUTO_PIN_LAST];
hda_nid_t adcs[AUTO_PIN_LAST];
hda_nid_t dig_out;
hda_nid_t dig_in;
unsigned int num_inputs;
hda_nid_t shared_mic_nid;
hda_nid_t shared_out_nid;
hda_nid_t unsol_tag_hp;
hda_nid_t unsol_tag_front_hp; /* for desktop ca0132 codecs */
hda_nid_t unsol_tag_amic1;
/* chip access */
struct mutex chipio_mutex; /* chip access mutex */
u32 curr_chip_addx;
/* DSP download related */
enum dsp_download_state dsp_state;
unsigned int dsp_stream_id;
unsigned int wait_scp;
unsigned int wait_scp_header;
unsigned int wait_num_data;
unsigned int scp_resp_header;
unsigned int scp_resp_data[4];
unsigned int scp_resp_count;
bool startup_check_entered;
bool dsp_reload;
/* mixer and effects related */
unsigned char dmic_ctl;
int cur_out_type;
int cur_mic_type;
long vnode_lvol[VNODES_COUNT];
long vnode_rvol[VNODES_COUNT];
long vnode_lswitch[VNODES_COUNT];
long vnode_rswitch[VNODES_COUNT];
long effects_switch[EFFECTS_COUNT];
long voicefx_val;
long cur_mic_boost;
/* ca0132_alt control related values */
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;
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;
unsigned int tlv[4];
struct hda_vmaster_mute_hook vmaster_mute;
/* AE-5 Control values */
unsigned char ae5_headphone_gain_val;
unsigned char ae5_filter_val;
/* ZxR Control Values */
unsigned char zxr_gain_set;
struct hda_codec *codec;
struct delayed_work unsol_hp_work;
int quirk;
#ifdef ENABLE_TUNING_CONTROLS
long cur_ctl_vals[TUNING_CTLS_COUNT];
#endif
/*
* The Recon3D, Sound Blaster Z, Sound Blaster ZxR, and Sound Blaster
* AE-5 all use PCI region 2 to toggle GPIO and other currently unknown
* things.
*/
bool use_pci_mmio;
void __iomem *mem_base;
/*
* Whether or not to use the alt functions like alt_select_out,
* alt_select_in, etc. Only used on desktop codecs for now, because of
* surround sound support.
*/
bool use_alt_functions;
/*
* Whether or not to use alt controls: volume effect sliders, EQ
* presets, smart volume presets, and new control names with FX prefix.
* Renames PlayEnhancement and CrystalVoice too.
*/
bool use_alt_controls;
};
/*
* CA0132 quirks table
*/
enum {
QUIRK_NONE,
QUIRK_ALIENWARE,
QUIRK_ALIENWARE_M17XR4,
QUIRK_SBZ,
QUIRK_ZXR,
QUIRK_ZXR_DBPRO,
QUIRK_R3DI,
QUIRK_R3D,
QUIRK_AE5,
QUIRK_AE7,
};
#ifdef CONFIG_PCI
#define ca0132_quirk(spec) ((spec)->quirk)
#define ca0132_use_pci_mmio(spec) ((spec)->use_pci_mmio)
#define ca0132_use_alt_functions(spec) ((spec)->use_alt_functions)
#define ca0132_use_alt_controls(spec) ((spec)->use_alt_controls)
#else
#define ca0132_quirk(spec) ({ (void)(spec); QUIRK_NONE; })
#define ca0132_use_alt_functions(spec) ({ (void)(spec); false; })
#define ca0132_use_pci_mmio(spec) ({ (void)(spec); false; })
#define ca0132_use_alt_controls(spec) ({ (void)(spec); false; })
#endif
static const struct hda_pintbl alienware_pincfgs[] = {
{ 0x0b, 0x90170110 }, /* Builtin Speaker */
{ 0x0c, 0x411111f0 }, /* N/A */
{ 0x0d, 0x411111f0 }, /* N/A */
{ 0x0e, 0x411111f0 }, /* N/A */
{ 0x0f, 0x0321101f }, /* HP */
{ 0x10, 0x411111f0 }, /* Headset? disabled for now */
{ 0x11, 0x03a11021 }, /* Mic */
{ 0x12, 0xd5a30140 }, /* Builtin Mic */
{ 0x13, 0x411111f0 }, /* N/A */
{ 0x18, 0x411111f0 }, /* N/A */
{}
};
/* Sound Blaster Z pin configs taken from Windows Driver */
static const struct hda_pintbl sbz_pincfgs[] = {
{ 0x0b, 0x01017010 }, /* Port G -- Lineout FRONT L/R */
{ 0x0c, 0x014510f0 }, /* SPDIF Out 1 */
{ 0x0d, 0x014510f0 }, /* Digital Out */
{ 0x0e, 0x01c510f0 }, /* SPDIF In */
{ 0x0f, 0x0221701f }, /* Port A -- BackPanel HP */
{ 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */
{ 0x11, 0x01017014 }, /* Port B -- LineMicIn2 / Rear L/R */
{ 0x12, 0x01a170f0 }, /* Port C -- LineIn1 */
{ 0x13, 0x908700f0 }, /* What U Hear In*/
{ 0x18, 0x50d000f0 }, /* N/A */
{}
};
/* Sound Blaster ZxR pin configs taken from Windows Driver */
static const struct hda_pintbl zxr_pincfgs[] = {
{ 0x0b, 0x01047110 }, /* Port G -- Lineout FRONT L/R */
{ 0x0c, 0x414510f0 }, /* SPDIF Out 1 - Disabled*/
{ 0x0d, 0x014510f0 }, /* Digital Out */
{ 0x0e, 0x41c520f0 }, /* SPDIF In - Disabled*/
{ 0x0f, 0x0122711f }, /* Port A -- BackPanel HP */
{ 0x10, 0x01017111 }, /* Port D -- Center/LFE */
{ 0x11, 0x01017114 }, /* Port B -- LineMicIn2 / Rear L/R */
{ 0x12, 0x01a271f0 }, /* Port C -- LineIn1 */
{ 0x13, 0x908700f0 }, /* What U Hear In*/
{ 0x18, 0x50d000f0 }, /* N/A */
{}
};
/* Recon3D pin configs taken from Windows Driver */
static const struct hda_pintbl r3d_pincfgs[] = {
{ 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */
{ 0x0c, 0x014510f0 }, /* SPDIF Out 1 */
{ 0x0d, 0x014510f0 }, /* Digital Out */
{ 0x0e, 0x01c520f0 }, /* SPDIF In */
{ 0x0f, 0x0221401f }, /* Port A -- BackPanel HP */
{ 0x10, 0x01016011 }, /* Port D -- Center/LFE or FP Hp */
{ 0x11, 0x01011014 }, /* Port B -- LineMicIn2 / Rear L/R */
{ 0x12, 0x02a090f0 }, /* Port C -- LineIn1 */
{ 0x13, 0x908700f0 }, /* What U Hear In*/
{ 0x18, 0x50d000f0 }, /* N/A */
{}
};
/* Sound Blaster AE-5 pin configs taken from Windows Driver */
static const struct hda_pintbl ae5_pincfgs[] = {
{ 0x0b, 0x01017010 }, /* Port G -- Lineout FRONT L/R */
{ 0x0c, 0x014510f0 }, /* SPDIF Out 1 */
{ 0x0d, 0x014510f0 }, /* Digital Out */
{ 0x0e, 0x01c510f0 }, /* SPDIF In */
{ 0x0f, 0x01017114 }, /* Port A -- Rear L/R. */
{ 0x10, 0x01017012 }, /* Port D -- Center/LFE or FP Hp */
{ 0x11, 0x012170ff }, /* Port B -- LineMicIn2 / Rear Headphone */
{ 0x12, 0x01a170f0 }, /* Port C -- LineIn1 */
{ 0x13, 0x908700f0 }, /* What U Hear In*/
{ 0x18, 0x50d000f0 }, /* N/A */
{}
};
/* Recon3D integrated pin configs taken from Windows Driver */
static const struct hda_pintbl r3di_pincfgs[] = {
{ 0x0b, 0x01014110 }, /* Port G -- Lineout FRONT L/R */
{ 0x0c, 0x014510f0 }, /* SPDIF Out 1 */
{ 0x0d, 0x014510f0 }, /* Digital Out */
{ 0x0e, 0x41c520f0 }, /* SPDIF In */
{ 0x0f, 0x0221401f }, /* Port A -- BackPanel HP */
{ 0x10, 0x01016011 }, /* Port D -- Center/LFE or FP Hp */
{ 0x11, 0x01011014 }, /* Port B -- LineMicIn2 / Rear L/R */
{ 0x12, 0x02a090f0 }, /* Port C -- LineIn1 */
{ 0x13, 0x908700f0 }, /* What U Hear In*/
{ 0x18, 0x500000f0 }, /* N/A */
{}
};
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),
SND_PCI_QUIRK(0x1028, 0x0688, "Alienware 17 2015", QUIRK_ALIENWARE),
SND_PCI_QUIRK(0x1028, 0x0708, "Alienware 15 R2 2016", QUIRK_ALIENWARE),
SND_PCI_QUIRK(0x1102, 0x0010, "Sound Blaster Z", QUIRK_SBZ),
SND_PCI_QUIRK(0x1102, 0x0023, "Sound Blaster Z", QUIRK_SBZ),
SND_PCI_QUIRK(0x1102, 0x0027, "Sound Blaster Z", QUIRK_SBZ),
SND_PCI_QUIRK(0x1102, 0x0033, "Sound Blaster ZxR", QUIRK_SBZ),
SND_PCI_QUIRK(0x1458, 0xA016, "Recon3Di", QUIRK_R3DI),
SND_PCI_QUIRK(0x1458, 0xA026, "Gigabyte G1.Sniper Z97", QUIRK_R3DI),
SND_PCI_QUIRK(0x1458, 0xA036, "Gigabyte GA-Z170X-Gaming 7", QUIRK_R3DI),
SND_PCI_QUIRK(0x3842, 0x1038, "EVGA X99 Classified", QUIRK_R3DI),
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, 0x0191, "Sound Blaster AE-5 Plus", QUIRK_AE5),
SND_PCI_QUIRK(0x1102, 0x0081, "Sound Blaster AE-7", QUIRK_AE7),
{}
};
/* 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
} },
},
{ .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
} },
}
};
/*
* CA0132 codec access
*/
static unsigned int codec_send_command(struct hda_codec *codec, hda_nid_t nid,
unsigned int verb, unsigned int parm, unsigned int *res)
{
unsigned int response;
response = snd_hda_codec_read(codec, nid, 0, verb, parm);
*res = response;
return ((response == -1) ? -1 : 0);
}
static int codec_set_converter_format(struct hda_codec *codec, hda_nid_t nid,
unsigned short converter_format, unsigned int *res)
{
return codec_send_command(codec, nid, VENDOR_CHIPIO_STREAM_FORMAT,
converter_format & 0xffff, res);
}
static int codec_set_converter_stream_channel(struct hda_codec *codec,
hda_nid_t nid, unsigned char stream,
unsigned char channel, unsigned int *res)
{
unsigned char converter_stream_channel = 0;
converter_stream_channel = (stream << 4) | (channel & 0x0f);
return codec_send_command(codec, nid, AC_VERB_SET_CHANNEL_STREAMID,
converter_stream_channel, res);
}
/* Chip access helper function */
static int chipio_send(struct hda_codec *codec,
unsigned int reg,
unsigned int data)
{
unsigned int res;
unsigned long timeout = jiffies + msecs_to_jiffies(1000);
/* send bits of data specified by reg */
do {
res = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0,
reg, data);
if (res == VENDOR_STATUS_CHIPIO_OK)
return 0;
msleep(20);
} while (time_before(jiffies, timeout));
return -EIO;
}
/*
* Write chip address through the vendor widget -- NOT protected by the Mutex!
*/
static int chipio_write_address(struct hda_codec *codec,
unsigned int chip_addx)
{
struct ca0132_spec *spec = codec->spec;
int res;
if (spec->curr_chip_addx == chip_addx)
return 0;
/* send low 16 bits of the address */
res = chipio_send(codec, VENDOR_CHIPIO_ADDRESS_LOW,
chip_addx & 0xffff);
if (res != -EIO) {
/* send high 16 bits of the address */
res = chipio_send(codec, VENDOR_CHIPIO_ADDRESS_HIGH,
chip_addx >> 16);
}
spec->curr_chip_addx = (res < 0) ? ~0U : chip_addx;
return res;
}
/*
* Write data through the vendor widget -- NOT protected by the Mutex!
*/
static int chipio_write_data(struct hda_codec *codec, unsigned int data)
{
struct ca0132_spec *spec = codec->spec;
int res;
/* send low 16 bits of the data */
res = chipio_send(codec, VENDOR_CHIPIO_DATA_LOW, data & 0xffff);
if (res != -EIO) {
/* send high 16 bits of the data */
res = chipio_send(codec, VENDOR_CHIPIO_DATA_HIGH,
data >> 16);
}
/*If no error encountered, automatically increment the address
as per chip behaviour*/
spec->curr_chip_addx = (res != -EIO) ?
(spec->curr_chip_addx + 4) : ~0U;
return res;
}
/*
* Write multiple data through the vendor widget -- NOT protected by the Mutex!
*/
static int chipio_write_data_multiple(struct hda_codec *codec,
const u32 *data,
unsigned int count)
{
int status = 0;
if (data == NULL) {
codec_dbg(codec, "chipio_write_data null ptr\n");
return -EINVAL;
}
while ((count-- != 0) && (status == 0))
status = chipio_write_data(codec, *data++);
return status;
}
/*
* Read data through the vendor widget -- NOT protected by the Mutex!
*/
static int chipio_read_data(struct hda_codec *codec, unsigned int *data)
{
struct ca0132_spec *spec = codec->spec;
int res;
/* post read */
res = chipio_send(codec, VENDOR_CHIPIO_HIC_POST_READ, 0);
if (res != -EIO) {
/* read status */
res = chipio_send(codec, VENDOR_CHIPIO_STATUS, 0);
}
if (res != -EIO) {
/* read data */
*data = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_HIC_READ_DATA,
0);
}
/*If no error encountered, automatically increment the address
as per chip behaviour*/
spec->curr_chip_addx = (res != -EIO) ?
(spec->curr_chip_addx + 4) : ~0U;
return res;
}
/*
* Write given value to the given address through the chip I/O widget.
* protected by the Mutex
*/
static int chipio_write(struct hda_codec *codec,
unsigned int chip_addx, const unsigned int data)
{
struct ca0132_spec *spec = codec->spec;
int err;
mutex_lock(&spec->chipio_mutex);
/* write the address, and if successful proceed to write data */
err = chipio_write_address(codec, chip_addx);
if (err < 0)
goto exit;
err = chipio_write_data(codec, data);
if (err < 0)
goto exit;
exit:
mutex_unlock(&spec->chipio_mutex);
return err;
}
/*
* Write given value to the given address through the chip I/O widget.
* not protected by the Mutex
*/
static int chipio_write_no_mutex(struct hda_codec *codec,
unsigned int chip_addx, const unsigned int data)
{
int err;
/* write the address, and if successful proceed to write data */
err = chipio_write_address(codec, chip_addx);
if (err < 0)
goto exit;
err = chipio_write_data(codec, data);
if (err < 0)
goto exit;
exit:
return err;
}
/*
* Write multiple values to the given address through the chip I/O widget.
* protected by the Mutex
*/
static int chipio_write_multiple(struct hda_codec *codec,
u32 chip_addx,
const u32 *data,
unsigned int count)
{
struct ca0132_spec *spec = codec->spec;
int status;
mutex_lock(&spec->chipio_mutex);
status = chipio_write_address(codec, chip_addx);
if (status < 0)
goto error;
status = chipio_write_data_multiple(codec, data, count);
error:
mutex_unlock(&spec->chipio_mutex);
return status;
}
/*
* Read the given address through the chip I/O widget
* protected by the Mutex
*/
static int chipio_read(struct hda_codec *codec,
unsigned int chip_addx, unsigned int *data)
{
struct ca0132_spec *spec = codec->spec;
int err;
mutex_lock(&spec->chipio_mutex);
/* write the address, and if successful proceed to write data */
err = chipio_write_address(codec, chip_addx);
if (err < 0)
goto exit;
err = chipio_read_data(codec, data);
if (err < 0)
goto exit;
exit:
mutex_unlock(&spec->chipio_mutex);
return err;
}
/*
* Set chip control flags through the chip I/O widget.
*/
static void chipio_set_control_flag(struct hda_codec *codec,
enum control_flag_id flag_id,
bool flag_state)
{
unsigned int val;
unsigned int flag_bit;
flag_bit = (flag_state ? 1 : 0);
val = (flag_bit << 7) | (flag_id);
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_FLAG_SET, val);
}
/*
* Set chip parameters through the chip I/O widget.
*/
static void chipio_set_control_param(struct hda_codec *codec,
enum control_param_id param_id, int param_val)
{
struct ca0132_spec *spec = codec->spec;
int val;
if ((param_id < 32) && (param_val < 8)) {
val = (param_val << 5) | (param_id);
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_PARAM_SET, val);
} else {
mutex_lock(&spec->chipio_mutex);
if (chipio_send(codec, VENDOR_CHIPIO_STATUS, 0) == 0) {
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_PARAM_EX_ID_SET,
param_id);
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_PARAM_EX_VALUE_SET,
param_val);
}
mutex_unlock(&spec->chipio_mutex);
}
}
/*
* Set chip parameters through the chip I/O widget. NO MUTEX.
*/
static void chipio_set_control_param_no_mutex(struct hda_codec *codec,
enum control_param_id param_id, int param_val)
{
int val;
if ((param_id < 32) && (param_val < 8)) {
val = (param_val << 5) | (param_id);
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_PARAM_SET, val);
} else {
if (chipio_send(codec, VENDOR_CHIPIO_STATUS, 0) == 0) {
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_PARAM_EX_ID_SET,
param_id);
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_PARAM_EX_VALUE_SET,
param_val);
}
}
}
/*
* Connect stream to a source point, and then connect
* that source point to a destination point.
*/
static void chipio_set_stream_source_dest(struct hda_codec *codec,
int streamid, int source_point, int dest_point)
{
chipio_set_control_param_no_mutex(codec,
CONTROL_PARAM_STREAM_ID, streamid);
chipio_set_control_param_no_mutex(codec,
CONTROL_PARAM_STREAM_SOURCE_CONN_POINT, source_point);
chipio_set_control_param_no_mutex(codec,
CONTROL_PARAM_STREAM_DEST_CONN_POINT, dest_point);
}
/*
* Set number of channels in the selected stream.
*/
static void chipio_set_stream_channels(struct hda_codec *codec,
int streamid, unsigned int channels)
{
chipio_set_control_param_no_mutex(codec,
CONTROL_PARAM_STREAM_ID, streamid);
chipio_set_control_param_no_mutex(codec,
CONTROL_PARAM_STREAMS_CHANNELS, channels);
}
/*
* Enable/Disable audio stream.
*/
static void chipio_set_stream_control(struct hda_codec *codec,
int streamid, int enable)
{
chipio_set_control_param_no_mutex(codec,
CONTROL_PARAM_STREAM_ID, streamid);
chipio_set_control_param_no_mutex(codec,
CONTROL_PARAM_STREAM_CONTROL, enable);
}
/*
* Get ChipIO audio stream's status.
*/
static void chipio_get_stream_control(struct hda_codec *codec,
int streamid, unsigned int *enable)
{
chipio_set_control_param_no_mutex(codec,
CONTROL_PARAM_STREAM_ID, streamid);
*enable = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_PARAM_GET,
CONTROL_PARAM_STREAM_CONTROL);
}
/*
* Set sampling rate of the connection point. NO MUTEX.
*/
static void chipio_set_conn_rate_no_mutex(struct hda_codec *codec,
int connid, enum ca0132_sample_rate rate)
{
chipio_set_control_param_no_mutex(codec,
CONTROL_PARAM_CONN_POINT_ID, connid);
chipio_set_control_param_no_mutex(codec,
CONTROL_PARAM_CONN_POINT_SAMPLE_RATE, rate);
}
/*
* Set sampling rate of the connection point.
*/
static void chipio_set_conn_rate(struct hda_codec *codec,
int connid, enum ca0132_sample_rate rate)
{
chipio_set_control_param(codec, CONTROL_PARAM_CONN_POINT_ID, connid);
chipio_set_control_param(codec, CONTROL_PARAM_CONN_POINT_SAMPLE_RATE,
rate);
}
/*
* Writes to the 8051's internal address space directly instead of indirectly,
* giving access to the special function registers located at addresses
* 0x80-0xFF.
*/
static void chipio_8051_write_direct(struct hda_codec *codec,
unsigned int addr, unsigned int data)
{
unsigned int verb;
verb = VENDOR_CHIPIO_8051_WRITE_DIRECT | data;
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, verb, addr);
}
/*
* Writes to the 8051's exram, which has 16-bits of address space.
* Data at addresses 0x2000-0x7fff is mirrored to 0x8000-0xdfff.
* Data at 0x8000-0xdfff can also be used as program memory for the 8051 by
* setting the pmem bank selection SFR.
* 0xe000-0xffff is always mapped as program memory, with only 0xf000-0xffff
* being writable.
*/
static void chipio_8051_set_address(struct hda_codec *codec, unsigned int addr)
{
unsigned int tmp;
/* Lower 8-bits. */
tmp = addr & 0xff;
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_8051_ADDRESS_LOW, tmp);
/* Upper 8-bits. */
tmp = (addr >> 8) & 0xff;
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_8051_ADDRESS_HIGH, tmp);
}
static void chipio_8051_set_data(struct hda_codec *codec, unsigned int data)
{
/* 8-bits of data. */
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_8051_DATA_WRITE, data & 0xff);
}
static unsigned int chipio_8051_get_data(struct hda_codec *codec)
{
return snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_8051_DATA_READ, 0);
}
/* PLL_PMU writes share the lower address register of the 8051 exram writes. */
static void chipio_8051_set_data_pll(struct hda_codec *codec, unsigned int data)
{
/* 8-bits of data. */
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_PLL_PMU_WRITE, data & 0xff);
}
static void chipio_8051_write_exram(struct hda_codec *codec,
unsigned int addr, unsigned int data)
{
struct ca0132_spec *spec = codec->spec;
mutex_lock(&spec->chipio_mutex);
chipio_8051_set_address(codec, addr);
chipio_8051_set_data(codec, data);
mutex_unlock(&spec->chipio_mutex);
}
static void chipio_8051_write_exram_no_mutex(struct hda_codec *codec,
unsigned int addr, unsigned int data)
{
chipio_8051_set_address(codec, addr);
chipio_8051_set_data(codec, data);
}
/* Readback data from the 8051's exram. No mutex. */
static void chipio_8051_read_exram(struct hda_codec *codec,
unsigned int addr, unsigned int *data)
{
chipio_8051_set_address(codec, addr);
*data = chipio_8051_get_data(codec);
}
static void chipio_8051_write_pll_pmu(struct hda_codec *codec,
unsigned int addr, unsigned int data)
{
struct ca0132_spec *spec = codec->spec;
mutex_lock(&spec->chipio_mutex);
chipio_8051_set_address(codec, addr & 0xff);
chipio_8051_set_data_pll(codec, data);
mutex_unlock(&spec->chipio_mutex);
}
static void chipio_8051_write_pll_pmu_no_mutex(struct hda_codec *codec,
unsigned int addr, unsigned int data)
{
chipio_8051_set_address(codec, addr & 0xff);
chipio_8051_set_data_pll(codec, data);
}
/*
* Enable clocks.
*/
static void chipio_enable_clocks(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
mutex_lock(&spec->chipio_mutex);
chipio_8051_write_pll_pmu_no_mutex(codec, 0x00, 0xff);
chipio_8051_write_pll_pmu_no_mutex(codec, 0x05, 0x0b);
chipio_8051_write_pll_pmu_no_mutex(codec, 0x06, 0xff);
mutex_unlock(&spec->chipio_mutex);
}
/*
* CA0132 DSP IO stuffs
*/
static int dspio_send(struct hda_codec *codec, unsigned int reg,
unsigned int data)
{
int res;
unsigned long timeout = jiffies + msecs_to_jiffies(1000);
/* send bits of data specified by reg to dsp */
do {
res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, data);
if ((res >= 0) && (res != VENDOR_STATUS_DSPIO_BUSY))
return res;
msleep(20);
} while (time_before(jiffies, timeout));
return -EIO;
}
/*
* Wait for DSP to be ready for commands
*/
static void dspio_write_wait(struct hda_codec *codec)
{
int status;
unsigned long timeout = jiffies + msecs_to_jiffies(1000);
do {
status = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0,
VENDOR_DSPIO_STATUS, 0);
if ((status == VENDOR_STATUS_DSPIO_OK) ||
(status == VENDOR_STATUS_DSPIO_SCP_RESPONSE_QUEUE_EMPTY))
break;
msleep(1);
} while (time_before(jiffies, timeout));
}
/*
* Write SCP data to DSP
*/
static int dspio_write(struct hda_codec *codec, unsigned int scp_data)
{
struct ca0132_spec *spec = codec->spec;
int status;
dspio_write_wait(codec);
mutex_lock(&spec->chipio_mutex);
status = dspio_send(codec, VENDOR_DSPIO_SCP_WRITE_DATA_LOW,
scp_data & 0xffff);
if (status < 0)
goto error;
status = dspio_send(codec, VENDOR_DSPIO_SCP_WRITE_DATA_HIGH,
scp_data >> 16);
if (status < 0)
goto error;
/* OK, now check if the write itself has executed*/
status = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0,
VENDOR_DSPIO_STATUS, 0);
error:
mutex_unlock(&spec->chipio_mutex);
return (status == VENDOR_STATUS_DSPIO_SCP_COMMAND_QUEUE_FULL) ?
-EIO : 0;
}
/*
* Write multiple SCP data to DSP
*/
static int dspio_write_multiple(struct hda_codec *codec,
unsigned int *buffer, unsigned int size)
{
int status = 0;
unsigned int count;
if (buffer == NULL)
return -EINVAL;
count = 0;
while (count < size) {
status = dspio_write(codec, *buffer++);
if (status != 0)
break;
count++;
}
return status;
}
static int dspio_read(struct hda_codec *codec, unsigned int *data)
{
int status;
status = dspio_send(codec, VENDOR_DSPIO_SCP_POST_READ_DATA, 0);
if (status == -EIO)
return status;
status = dspio_send(codec, VENDOR_DSPIO_STATUS, 0);
if (status == -EIO ||
status == VENDOR_STATUS_DSPIO_SCP_RESPONSE_QUEUE_EMPTY)
return -EIO;
*data = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0,
VENDOR_DSPIO_SCP_READ_DATA, 0);
return 0;
}
static int dspio_read_multiple(struct hda_codec *codec, unsigned int *buffer,
unsigned int *buf_size, unsigned int size_count)
{
int status = 0;
unsigned int size = *buf_size;
unsigned int count;
unsigned int skip_count;
unsigned int dummy;
if (buffer == NULL)
return -1;
count = 0;
while (count < size && count < size_count) {
status = dspio_read(codec, buffer++);
if (status != 0)
break;
count++;
}
skip_count = count;
if (status == 0) {
while (skip_count < size) {
status = dspio_read(codec, &dummy);
if (status != 0)
break;
skip_count++;
}
}
*buf_size = count;
return status;
}
/*
* Construct the SCP header using corresponding fields
*/
static inline unsigned int
make_scp_header(unsigned int target_id, unsigned int source_id,
unsigned int get_flag, unsigned int req,
unsigned int device_flag, unsigned int resp_flag,
unsigned int error_flag, unsigned int data_size)
{
unsigned int header = 0;
header = (data_size & 0x1f) << 27;
header |= (error_flag & 0x01) << 26;
header |= (resp_flag & 0x01) << 25;
header |= (device_flag & 0x01) << 24;
header |= (req & 0x7f) << 17;
header |= (get_flag & 0x01) << 16;
header |= (source_id & 0xff) << 8;
header |= target_id & 0xff;
return header;
}
/*
* Extract corresponding fields from SCP header
*/
static inline void
extract_scp_header(unsigned int header,
unsigned int *target_id, unsigned int *source_id,
unsigned int *get_flag, unsigned int *req,
unsigned int *device_flag, unsigned int *resp_flag,
unsigned int *error_flag, unsigned int *data_size)
{
if (data_size)
*data_size = (header >> 27) & 0x1f;
if (error_flag)
*error_flag = (header >> 26) & 0x01;
if (resp_flag)
*resp_flag = (header >> 25) & 0x01;
if (device_flag)
*device_flag = (header >> 24) & 0x01;
if (req)
*req = (header >> 17) & 0x7f;
if (get_flag)
*get_flag = (header >> 16) & 0x01;
if (source_id)
*source_id = (header >> 8) & 0xff;
if (target_id)
*target_id = header & 0xff;
}
#define SCP_MAX_DATA_WORDS (16)
/* Structure to contain any SCP message */
struct scp_msg {
unsigned int hdr;
unsigned int data[SCP_MAX_DATA_WORDS];
};
static void dspio_clear_response_queue(struct hda_codec *codec)
{
unsigned long timeout = jiffies + msecs_to_jiffies(1000);
unsigned int dummy = 0;
int status;
/* clear all from the response queue */
do {
status = dspio_read(codec, &dummy);
} while (status == 0 && time_before(jiffies, timeout));
}
static int dspio_get_response_data(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
unsigned int data = 0;
unsigned int count;
if (dspio_read(codec, &data) < 0)
return -EIO;
if ((data & 0x00ffffff) == spec->wait_scp_header) {
spec->scp_resp_header = data;
spec->scp_resp_count = data >> 27;
count = spec->wait_num_data;
dspio_read_multiple(codec, spec->scp_resp_data,
&spec->scp_resp_count, count);
return 0;
}
return -EIO;
}
/*
* Send SCP message to DSP
*/
static int dspio_send_scp_message(struct hda_codec *codec,
unsigned char *send_buf,
unsigned int send_buf_size,
unsigned char *return_buf,
unsigned int return_buf_size,
unsigned int *bytes_returned)
{
struct ca0132_spec *spec = codec->spec;
int status;
unsigned int scp_send_size = 0;
unsigned int total_size;
bool waiting_for_resp = false;
unsigned int header;
struct scp_msg *ret_msg;
unsigned int resp_src_id, resp_target_id;
unsigned int data_size, src_id, target_id, get_flag, device_flag;
if (bytes_returned)
*bytes_returned = 0;
/* get scp header from buffer */
header = *((unsigned int *)send_buf);
extract_scp_header(header, &target_id, &src_id, &get_flag, NULL,
&device_flag, NULL, NULL, &data_size);
scp_send_size = data_size + 1;
total_size = (scp_send_size * 4);
if (send_buf_size < total_size)
return -EINVAL;
if (get_flag || device_flag) {
if (!return_buf || return_buf_size < 4 || !bytes_returned)
return -EINVAL;
spec->wait_scp_header = *((unsigned int *)send_buf);
/* swap source id with target id */
resp_target_id = src_id;
resp_src_id = target_id;
spec->wait_scp_header &= 0xffff0000;
spec->wait_scp_header |= (resp_src_id << 8) | (resp_target_id);
spec->wait_num_data = return_buf_size/sizeof(unsigned int) - 1;
spec->wait_scp = 1;
waiting_for_resp = true;
}
status = dspio_write_multiple(codec, (unsigned int *)send_buf,
scp_send_size);
if (status < 0) {
spec->wait_scp = 0;
return status;
}
if (waiting_for_resp) {
unsigned long timeout = jiffies + msecs_to_jiffies(1000);
memset(return_buf, 0, return_buf_size);
do {
msleep(20);
} while (spec->wait_scp && time_before(jiffies, timeout));
waiting_for_resp = false;
if (!spec->wait_scp) {
ret_msg = (struct scp_msg *)return_buf;
memcpy(&ret_msg->hdr, &spec->scp_resp_header, 4);
memcpy(&ret_msg->data, spec->scp_resp_data,
spec->wait_num_data);
*bytes_returned = (spec->scp_resp_count + 1) * 4;
status = 0;
} else {
status = -EIO;
}
spec->wait_scp = 0;
}
return status;
}
/**
* dspio_scp - Prepare and send the SCP message to DSP
* @codec: the HDA codec
* @mod_id: ID of the DSP module to send the command
* @src_id: ID of the source
* @req: ID of request to send to the DSP module
* @dir: SET or GET
* @data: pointer to the data to send with the request, request specific
* @len: length of the data, in bytes
* @reply: point to the buffer to hold data returned for a reply
* @reply_len: length of the reply buffer returned from GET
*
* Returns zero or a negative error code.
*/
static int dspio_scp(struct hda_codec *codec,
int mod_id, int src_id, int req, int dir, const void *data,
unsigned int len, void *reply, unsigned int *reply_len)
{
int status = 0;
struct scp_msg scp_send, scp_reply;
unsigned int ret_bytes, send_size, ret_size;
unsigned int send_get_flag, reply_resp_flag, reply_error_flag;
unsigned int reply_data_size;
memset(&scp_send, 0, sizeof(scp_send));
memset(&scp_reply, 0, sizeof(scp_reply));
if ((len != 0 && data == NULL) || (len > SCP_MAX_DATA_WORDS))
return -EINVAL;
if (dir == SCP_GET && reply == NULL) {
codec_dbg(codec, "dspio_scp get but has no buffer\n");
return -EINVAL;
}
if (reply != NULL && (reply_len == NULL || (*reply_len == 0))) {
codec_dbg(codec, "dspio_scp bad resp buf len parms\n");
return -EINVAL;
}
scp_send.hdr = make_scp_header(mod_id, src_id, (dir == SCP_GET), req,
0, 0, 0, len/sizeof(unsigned int));
if (data != NULL && len > 0) {
len = min((unsigned int)(sizeof(scp_send.data)), len);
memcpy(scp_send.data, data, len);
}
ret_bytes = 0;
send_size = sizeof(unsigned int) + len;
status = dspio_send_scp_message(codec, (unsigned char *)&scp_send,
send_size, (unsigned char *)&scp_reply,
sizeof(scp_reply), &ret_bytes);
if (status < 0) {
codec_dbg(codec, "dspio_scp: send scp msg failed\n");
return status;
}
/* extract send and reply headers members */
extract_scp_header(scp_send.hdr, NULL, NULL, &send_get_flag,
NULL, NULL, NULL, NULL, NULL);
extract_scp_header(scp_reply.hdr, NULL, NULL, NULL, NULL, NULL,
&reply_resp_flag, &reply_error_flag,
&reply_data_size);
if (!send_get_flag)
return 0;
if (reply_resp_flag && !reply_error_flag) {
ret_size = (ret_bytes - sizeof(scp_reply.hdr))
/ sizeof(unsigned int);
if (*reply_len < ret_size*sizeof(unsigned int)) {
codec_dbg(codec, "reply too long for buf\n");
return -EINVAL;
} else if (ret_size != reply_data_size) {
codec_dbg(codec, "RetLen and HdrLen .NE.\n");
return -EINVAL;
} else if (!reply) {
codec_dbg(codec, "NULL reply\n");
return -EINVAL;
} else {
*reply_len = ret_size*sizeof(unsigned int);
memcpy(reply, scp_reply.data, *reply_len);
}
} else {
codec_dbg(codec, "reply ill-formed or errflag set\n");
return -EIO;
}
return status;
}
/*
* Set DSP parameters
*/
static int dspio_set_param(struct hda_codec *codec, int mod_id,
int src_id, int req, const void *data, unsigned int len)
{
return dspio_scp(codec, mod_id, src_id, req, SCP_SET, data, len, NULL,
NULL);
}
static int dspio_set_uint_param(struct hda_codec *codec, int mod_id,
int req, const unsigned int data)
{
return dspio_set_param(codec, mod_id, 0x20, req, &data,
sizeof(unsigned int));
}
/*
* Allocate a DSP DMA channel via an SCP message
*/
static int dspio_alloc_dma_chan(struct hda_codec *codec, unsigned int *dma_chan)
{
int status = 0;
unsigned int size = sizeof(dma_chan);
codec_dbg(codec, " dspio_alloc_dma_chan() -- begin\n");
status = dspio_scp(codec, MASTERCONTROL, 0x20,
MASTERCONTROL_ALLOC_DMA_CHAN, SCP_GET, NULL, 0,
dma_chan, &size);
if (status < 0) {
codec_dbg(codec, "dspio_alloc_dma_chan: SCP Failed\n");
return status;
}
if ((*dma_chan + 1) == 0) {
codec_dbg(codec, "no free dma channels to allocate\n");
return -EBUSY;
}
codec_dbg(codec, "dspio_alloc_dma_chan: chan=%d\n", *dma_chan);
codec_dbg(codec, " dspio_alloc_dma_chan() -- complete\n");
return status;
}
/*
* Free a DSP DMA via an SCP message
*/
static int dspio_free_dma_chan(struct hda_codec *codec, unsigned int dma_chan)
{
int status = 0;
unsigned int dummy = 0;
codec_dbg(codec, " dspio_free_dma_chan() -- begin\n");
codec_dbg(codec, "dspio_free_dma_chan: chan=%d\n", dma_chan);
status = dspio_scp(codec, MASTERCONTROL, 0x20,
MASTERCONTROL_ALLOC_DMA_CHAN, SCP_SET, &dma_chan,
sizeof(dma_chan), NULL, &dummy);
if (status < 0) {
codec_dbg(codec, "dspio_free_dma_chan: SCP Failed\n");
return status;
}
codec_dbg(codec, " dspio_free_dma_chan() -- complete\n");
return status;
}
/*
* (Re)start the DSP
*/
static int dsp_set_run_state(struct hda_codec *codec)
{
unsigned int dbg_ctrl_reg;
unsigned int halt_state;
int err;
err = chipio_read(codec, DSP_DBGCNTL_INST_OFFSET, &dbg_ctrl_reg);
if (err < 0)
return err;
halt_state = (dbg_ctrl_reg & DSP_DBGCNTL_STATE_MASK) >>
DSP_DBGCNTL_STATE_LOBIT;
if (halt_state != 0) {
dbg_ctrl_reg &= ~((halt_state << DSP_DBGCNTL_SS_LOBIT) &
DSP_DBGCNTL_SS_MASK);
err = chipio_write(codec, DSP_DBGCNTL_INST_OFFSET,
dbg_ctrl_reg);
if (err < 0)
return err;
dbg_ctrl_reg |= (halt_state << DSP_DBGCNTL_EXEC_LOBIT) &
DSP_DBGCNTL_EXEC_MASK;
err = chipio_write(codec, DSP_DBGCNTL_INST_OFFSET,
dbg_ctrl_reg);
if (err < 0)
return err;
}
return 0;
}
/*
* Reset the DSP
*/
static int dsp_reset(struct hda_codec *codec)
{
unsigned int res;
int retry = 20;
codec_dbg(codec, "dsp_reset\n");
do {
res = dspio_send(codec, VENDOR_DSPIO_DSP_INIT, 0);
retry--;
} while (res == -EIO && retry);
if (!retry) {
codec_dbg(codec, "dsp_reset timeout\n");
return -EIO;
}
return 0;
}
/*
* Convert chip address to DSP address
*/
static unsigned int dsp_chip_to_dsp_addx(unsigned int chip_addx,
bool *code, bool *yram)
{
*code = *yram = false;
if (UC_RANGE(chip_addx, 1)) {
*code = true;
return UC_OFF(chip_addx);
} else if (X_RANGE_ALL(chip_addx, 1)) {
return X_OFF(chip_addx);
} else if (Y_RANGE_ALL(chip_addx, 1)) {
*yram = true;
return Y_OFF(chip_addx);
}
return INVALID_CHIP_ADDRESS;
}
/*
* Check if the DSP DMA is active
*/
static bool dsp_is_dma_active(struct hda_codec *codec, unsigned int dma_chan)
{
unsigned int dma_chnlstart_reg;
chipio_read(codec, DSPDMAC_CHNLSTART_INST_OFFSET, &dma_chnlstart_reg);
return ((dma_chnlstart_reg & (1 <<
(DSPDMAC_CHNLSTART_EN_LOBIT + dma_chan))) != 0);
}
static int dsp_dma_setup_common(struct hda_codec *codec,
unsigned int chip_addx,
unsigned int dma_chan,
unsigned int port_map_mask,
bool ovly)
{
int status = 0;
unsigned int chnl_prop;
unsigned int dsp_addx;
unsigned int active;
bool code, yram;
codec_dbg(codec, "-- dsp_dma_setup_common() -- Begin ---------\n");
if (dma_chan >= DSPDMAC_DMA_CFG_CHANNEL_COUNT) {
codec_dbg(codec, "dma chan num invalid\n");
return -EINVAL;
}
if (dsp_is_dma_active(codec, dma_chan)) {
codec_dbg(codec, "dma already active\n");
return -EBUSY;
}
dsp_addx = dsp_chip_to_dsp_addx(chip_addx, &code, &yram);
if (dsp_addx == INVALID_CHIP_ADDRESS) {
codec_dbg(codec, "invalid chip addr\n");
return -ENXIO;
}
chnl_prop = DSPDMAC_CHNLPROP_AC_MASK;
active = 0;
codec_dbg(codec, " dsp_dma_setup_common() start reg pgm\n");
if (ovly) {
status = chipio_read(codec, DSPDMAC_CHNLPROP_INST_OFFSET,
&chnl_prop);
if (status < 0) {
codec_dbg(codec, "read CHNLPROP Reg fail\n");
return status;
}
codec_dbg(codec, "dsp_dma_setup_common() Read CHNLPROP\n");
}
if (!code)
chnl_prop &= ~(1 << (DSPDMAC_CHNLPROP_MSPCE_LOBIT + dma_chan));
else
chnl_prop |= (1 << (DSPDMAC_CHNLPROP_MSPCE_LOBIT + dma_chan));
chnl_prop &= ~(1 << (DSPDMAC_CHNLPROP_DCON_LOBIT + dma_chan));
status = chipio_write(codec, DSPDMAC_CHNLPROP_INST_OFFSET, chnl_prop);
if (status < 0) {
codec_dbg(codec, "write CHNLPROP Reg fail\n");
return status;
}
codec_dbg(codec, " dsp_dma_setup_common() Write CHNLPROP\n");
if (ovly) {
status = chipio_read(codec, DSPDMAC_ACTIVE_INST_OFFSET,
&active);
if (status < 0) {
codec_dbg(codec, "read ACTIVE Reg fail\n");
return status;
}
codec_dbg(codec, "dsp_dma_setup_common() Read ACTIVE\n");
}
active &= (~(1 << (DSPDMAC_ACTIVE_AAR_LOBIT + dma_chan))) &
DSPDMAC_ACTIVE_AAR_MASK;
status = chipio_write(codec, DSPDMAC_ACTIVE_INST_OFFSET, active);
if (status < 0) {
codec_dbg(codec, "write ACTIVE Reg fail\n");
return status;
}
codec_dbg(codec, " dsp_dma_setup_common() Write ACTIVE\n");
status = chipio_write(codec, DSPDMAC_AUDCHSEL_INST_OFFSET(dma_chan),
port_map_mask);
if (status < 0) {
codec_dbg(codec, "write AUDCHSEL Reg fail\n");
return status;
}
codec_dbg(codec, " dsp_dma_setup_common() Write AUDCHSEL\n");
status = chipio_write(codec, DSPDMAC_IRQCNT_INST_OFFSET(dma_chan),
DSPDMAC_IRQCNT_BICNT_MASK | DSPDMAC_IRQCNT_CICNT_MASK);
if (status < 0) {
codec_dbg(codec, "write IRQCNT Reg fail\n");
return status;
}
codec_dbg(codec, " dsp_dma_setup_common() Write IRQCNT\n");
codec_dbg(codec,
"ChipA=0x%x,DspA=0x%x,dmaCh=%u, "
"CHSEL=0x%x,CHPROP=0x%x,Active=0x%x\n",
chip_addx, dsp_addx, dma_chan,
port_map_mask, chnl_prop, active);
codec_dbg(codec, "-- dsp_dma_setup_common() -- Complete ------\n");
return 0;
}
/*
* Setup the DSP DMA per-transfer-specific registers
*/
static int dsp_dma_setup(struct hda_codec *codec,
unsigned int chip_addx,
unsigned int count,
unsigned int dma_chan)
{
int status = 0;
bool code, yram;
unsigned int dsp_addx;
unsigned int addr_field;
unsigned int incr_field;
unsigned int base_cnt;
unsigned int cur_cnt;
unsigned int dma_cfg = 0;
unsigned int adr_ofs = 0;
unsigned int xfr_cnt = 0;
const unsigned int max_dma_count = 1 << (DSPDMAC_XFRCNT_BCNT_HIBIT -
DSPDMAC_XFRCNT_BCNT_LOBIT + 1);
codec_dbg(codec, "-- dsp_dma_setup() -- Begin ---------\n");
if (count > max_dma_count) {
codec_dbg(codec, "count too big\n");
return -EINVAL;
}
dsp_addx = dsp_chip_to_dsp_addx(chip_addx, &code, &yram);
if (dsp_addx == INVALID_CHIP_ADDRESS) {
codec_dbg(codec, "invalid chip addr\n");
return -ENXIO;
}
codec_dbg(codec, " dsp_dma_setup() start reg pgm\n");
addr_field = dsp_addx << DSPDMAC_DMACFG_DBADR_LOBIT;
incr_field = 0;
if (!code) {
addr_field <<= 1;
if (yram)
addr_field |= (1 << DSPDMAC_DMACFG_DBADR_LOBIT);
incr_field = (1 << DSPDMAC_DMACFG_AINCR_LOBIT);
}
dma_cfg = addr_field + incr_field;
status = chipio_write(codec, DSPDMAC_DMACFG_INST_OFFSET(dma_chan),
dma_cfg);
if (status < 0) {
codec_dbg(codec, "write DMACFG Reg fail\n");
return status;
}
codec_dbg(codec, " dsp_dma_setup() Write DMACFG\n");
adr_ofs = (count - 1) << (DSPDMAC_DSPADROFS_BOFS_LOBIT +
(code ? 0 : 1));
status = chipio_write(codec, DSPDMAC_DSPADROFS_INST_OFFSET(dma_chan),
adr_ofs);
if (status < 0) {
codec_dbg(codec, "write DSPADROFS Reg fail\n");
return status;
}
codec_dbg(codec, " dsp_dma_setup() Write DSPADROFS\n");
base_cnt = (count - 1) << DSPDMAC_XFRCNT_BCNT_LOBIT;
cur_cnt = (count - 1) << DSPDMAC_XFRCNT_CCNT_LOBIT;
xfr_cnt = base_cnt | cur_cnt;
status = chipio_write(codec,
DSPDMAC_XFRCNT_INST_OFFSET(dma_chan), xfr_cnt);
if (status < 0) {
codec_dbg(codec, "write XFRCNT Reg fail\n");
return status;
}
codec_dbg(codec, " dsp_dma_setup() Write XFRCNT\n");
codec_dbg(codec,
"ChipA=0x%x, cnt=0x%x, DMACFG=0x%x, "
"ADROFS=0x%x, XFRCNT=0x%x\n",
chip_addx, count, dma_cfg, adr_ofs, xfr_cnt);
codec_dbg(codec, "-- dsp_dma_setup() -- Complete ---------\n");
return 0;
}
/*
* Start the DSP DMA
*/
static int dsp_dma_start(struct hda_codec *codec,
unsigned int dma_chan, bool ovly)
{
unsigned int reg = 0;
int status = 0;
codec_dbg(codec, "-- dsp_dma_start() -- Begin ---------\n");
if (ovly) {
status = chipio_read(codec,
DSPDMAC_CHNLSTART_INST_OFFSET, &reg);
if (status < 0) {
codec_dbg(codec, "read CHNLSTART reg fail\n");
return status;
}
codec_dbg(codec, "-- dsp_dma_start() Read CHNLSTART\n");
reg &= ~(DSPDMAC_CHNLSTART_EN_MASK |
DSPDMAC_CHNLSTART_DIS_MASK);
}
status = chipio_write(codec, DSPDMAC_CHNLSTART_INST_OFFSET,
reg | (1 << (dma_chan + DSPDMAC_CHNLSTART_EN_LOBIT)));
if (status < 0) {
codec_dbg(codec, "write CHNLSTART reg fail\n");
return status;
}
codec_dbg(codec, "-- dsp_dma_start() -- Complete ---------\n");
return status;
}
/*
* Stop the DSP DMA
*/
static int dsp_dma_stop(struct hda_codec *codec,
unsigned int dma_chan, bool ovly)
{
unsigned int reg = 0;
int status = 0;
codec_dbg(codec, "-- dsp_dma_stop() -- Begin ---------\n");
if (ovly) {
status = chipio_read(codec,
DSPDMAC_CHNLSTART_INST_OFFSET, &reg);
if (status < 0) {
codec_dbg(codec, "read CHNLSTART reg fail\n");
return status;
}
codec_dbg(codec, "-- dsp_dma_stop() Read CHNLSTART\n");
reg &= ~(DSPDMAC_CHNLSTART_EN_MASK |
DSPDMAC_CHNLSTART_DIS_MASK);
}
status = chipio_write(codec, DSPDMAC_CHNLSTART_INST_OFFSET,
reg | (1 << (dma_chan + DSPDMAC_CHNLSTART_DIS_LOBIT)));
if (status < 0) {
codec_dbg(codec, "write CHNLSTART reg fail\n");
return status;
}
codec_dbg(codec, "-- dsp_dma_stop() -- Complete ---------\n");
return status;
}
/**
* dsp_allocate_router_ports - Allocate router ports
*
* @codec: the HDA codec
* @num_chans: number of channels in the stream
* @ports_per_channel: number of ports per channel
* @start_device: start device
* @port_map: pointer to the port list to hold the allocated ports
*
* Returns zero or a negative error code.
*/
static int dsp_allocate_router_ports(struct hda_codec *codec,
unsigned int num_chans,
unsigned int ports_per_channel,
unsigned int start_device,
unsigned int *port_map)
{
int status = 0;
int res;
u8 val;
status = chipio_send(codec, VENDOR_CHIPIO_STATUS, 0);
if (status < 0)
return status;
val = start_device << 6;
val |= (ports_per_channel - 1) << 4;
val |= num_chans - 1;
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_PORT_ALLOC_CONFIG_SET,
val);
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_PORT_ALLOC_SET,
MEM_CONNID_DSP);
status = chipio_send(codec, VENDOR_CHIPIO_STATUS, 0);
if (status < 0)
return status;
res = snd_hda_codec_read(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_PORT_ALLOC_GET, 0);
*port_map = res;
return (res < 0) ? res : 0;
}
/*
* Free router ports
*/
static int dsp_free_router_ports(struct hda_codec *codec)
{
int status = 0;
status = chipio_send(codec, VENDOR_CHIPIO_STATUS, 0);
if (status < 0)
return status;
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_PORT_FREE_SET,
MEM_CONNID_DSP);
status = chipio_send(codec, VENDOR_CHIPIO_STATUS, 0);
return status;
}
/*
* Allocate DSP ports for the download stream
*/
static int dsp_allocate_ports(struct hda_codec *codec,
unsigned int num_chans,
unsigned int rate_multi, unsigned int *port_map)
{
int status;
codec_dbg(codec, " dsp_allocate_ports() -- begin\n");
if ((rate_multi != 1) && (rate_multi != 2) && (rate_multi != 4)) {
codec_dbg(codec, "bad rate multiple\n");
return -EINVAL;
}
status = dsp_allocate_router_ports(codec, num_chans,
rate_multi, 0, port_map);
codec_dbg(codec, " dsp_allocate_ports() -- complete\n");
return status;
}
static int dsp_allocate_ports_format(struct hda_codec *codec,
const unsigned short fmt,
unsigned int *port_map)
{
int status;
unsigned int num_chans;
unsigned int sample_rate_div = ((get_hdafmt_rate(fmt) >> 0) & 3) + 1;
unsigned int sample_rate_mul = ((get_hdafmt_rate(fmt) >> 3) & 3) + 1;
unsigned int rate_multi = sample_rate_mul / sample_rate_div;
if ((rate_multi != 1) && (rate_multi != 2) && (rate_multi != 4)) {
codec_dbg(codec, "bad rate multiple\n");
return -EINVAL;
}
num_chans = get_hdafmt_chs(fmt) + 1;
status = dsp_allocate_ports(codec, num_chans, rate_multi, port_map);
return status;
}
/*
* free DSP ports
*/
static int dsp_free_ports(struct hda_codec *codec)
{
int status;
codec_dbg(codec, " dsp_free_ports() -- begin\n");
status = dsp_free_router_ports(codec);
if (status < 0) {
codec_dbg(codec, "free router ports fail\n");
return status;
}
codec_dbg(codec, " dsp_free_ports() -- complete\n");
return status;
}
/*
* HDA DMA engine stuffs for DSP code download
*/
struct dma_engine {
struct hda_codec *codec;
unsigned short m_converter_format;
struct snd_dma_buffer *dmab;
unsigned int buf_size;
};
enum dma_state {
DMA_STATE_STOP = 0,
DMA_STATE_RUN = 1
};
static int dma_convert_to_hda_format(struct hda_codec *codec,
unsigned int sample_rate,
unsigned short channels,
unsigned short *hda_format)
{
unsigned int format_val;
format_val = snd_hdac_calc_stream_format(sample_rate,
channels, SNDRV_PCM_FORMAT_S32_LE, 32, 0);
if (hda_format)
*hda_format = (unsigned short)format_val;
return 0;
}
/*
* Reset DMA for DSP download
*/
static int dma_reset(struct dma_engine *dma)
{
struct hda_codec *codec = dma->codec;
struct ca0132_spec *spec = codec->spec;
int status;
if (dma->dmab->area)
snd_hda_codec_load_dsp_cleanup(codec, dma->dmab);
status = snd_hda_codec_load_dsp_prepare(codec,
dma->m_converter_format,
dma->buf_size,
dma->dmab);
if (status < 0)
return status;
spec->dsp_stream_id = status;
return 0;
}
static int dma_set_state(struct dma_engine *dma, enum dma_state state)
{
bool cmd;
switch (state) {
case DMA_STATE_STOP:
cmd = false;
break;
case DMA_STATE_RUN:
cmd = true;
break;
default:
return 0;
}
snd_hda_codec_load_dsp_trigger(dma->codec, cmd);
return 0;
}
static unsigned int dma_get_buffer_size(struct dma_engine *dma)
{
return dma->dmab->bytes;
}
static unsigned char *dma_get_buffer_addr(struct dma_engine *dma)
{
return dma->dmab->area;
}
static int dma_xfer(struct dma_engine *dma,
const unsigned int *data,
unsigned int count)
{
memcpy(dma->dmab->area, data, count);
return 0;
}
static void dma_get_converter_format(
struct dma_engine *dma,
unsigned short *format)
{
if (format)
*format = dma->m_converter_format;
}
static unsigned int dma_get_stream_id(struct dma_engine *dma)
{
struct ca0132_spec *spec = dma->codec->spec;
return spec->dsp_stream_id;
}
struct dsp_image_seg {
u32 magic;
u32 chip_addr;
u32 count;
u32 data[];
};
static const u32 g_magic_value = 0x4c46584d;
static const u32 g_chip_addr_magic_value = 0xFFFFFF01;
static bool is_valid(const struct dsp_image_seg *p)
{
return p->magic == g_magic_value;
}
static bool is_hci_prog_list_seg(const struct dsp_image_seg *p)
{
return g_chip_addr_magic_value == p->chip_addr;
}
static bool is_last(const struct dsp_image_seg *p)
{
return p->count == 0;
}
static size_t dsp_sizeof(const struct dsp_image_seg *p)
{
return struct_size(p, data, p->count);
}
static const struct dsp_image_seg *get_next_seg_ptr(
const struct dsp_image_seg *p)
{
return (struct dsp_image_seg *)((unsigned char *)(p) + dsp_sizeof(p));
}
/*
* CA0132 chip DSP transfer stuffs. For DSP download.
*/
#define INVALID_DMA_CHANNEL (~0U)
/*
* Program a list of address/data pairs via the ChipIO widget.
* The segment data is in the format of successive pairs of words.
* These are repeated as indicated by the segment's count field.
*/
static int dspxfr_hci_write(struct hda_codec *codec,
const struct dsp_image_seg *fls)
{
int status;
const u32 *data;
unsigned int count;
if (fls == NULL || fls->chip_addr != g_chip_addr_magic_value) {
codec_dbg(codec, "hci_write invalid params\n");
return -EINVAL;
}
count = fls->count;
data = (u32 *)(fls->data);
while (count >= 2) {
status = chipio_write(codec, data[0], data[1]);
if (status < 0) {
codec_dbg(codec, "hci_write chipio failed\n");
return status;
}
count -= 2;
data += 2;
}
return 0;
}
/**
* dspxfr_one_seg - Write a block of data into DSP code or data RAM using pre-allocated DMA engine.
*
* @codec: the HDA codec
* @fls: pointer to a fast load image
* @reloc: Relocation address for loading single-segment overlays, or 0 for
* no relocation
* @dma_engine: pointer to DMA engine to be used for DSP download
* @dma_chan: The number of DMA channels used for DSP download
* @port_map_mask: port mapping
* @ovly: TRUE if overlay format is required
*
* Returns zero or a negative error code.
*/
static int dspxfr_one_seg(struct hda_codec *codec,
const struct dsp_image_seg *fls,
unsigned int reloc,
struct dma_engine *dma_engine,
unsigned int dma_chan,
unsigned int port_map_mask,
bool ovly)
{
int status = 0;
bool comm_dma_setup_done = false;
const unsigned int *data;
unsigned int chip_addx;
unsigned int words_to_write;
unsigned int buffer_size_words;
unsigned char *buffer_addx;
unsigned short hda_format;
unsigned int sample_rate_div;
unsigned int sample_rate_mul;
unsigned int num_chans;
unsigned int hda_frame_size_words;
unsigned int remainder_words;
const u32 *data_remainder;
u32 chip_addx_remainder;
unsigned int run_size_words;
const struct dsp_image_seg *hci_write = NULL;
unsigned long timeout;
bool dma_active;
if (fls == NULL)
return -EINVAL;
if (is_hci_prog_list_seg(fls)) {
hci_write = fls;
fls = get_next_seg_ptr(fls);
}
if (hci_write && (!fls || is_last(fls))) {
codec_dbg(codec, "hci_write\n");
return dspxfr_hci_write(codec, hci_write);
}
if (fls == NULL || dma_engine == NULL || port_map_mask == 0) {
codec_dbg(codec, "Invalid Params\n");
return -EINVAL;
}
data = fls->data;
chip_addx = fls->chip_addr;
words_to_write = fls->count;
if (!words_to_write)
return hci_write ? dspxfr_hci_write(codec, hci_write) : 0;
if (reloc)
chip_addx = (chip_addx & (0xFFFF0000 << 2)) + (reloc << 2);
if (!UC_RANGE(chip_addx, words_to_write) &&
!X_RANGE_ALL(chip_addx, words_to_write) &&
!Y_RANGE_ALL(chip_addx, words_to_write)) {
codec_dbg(codec, "Invalid chip_addx Params\n");
return -EINVAL;
}
buffer_size_words = (unsigned int)dma_get_buffer_size(dma_engine) /
sizeof(u32);
buffer_addx = dma_get_buffer_addr(dma_engine);
if (buffer_addx == NULL) {
codec_dbg(codec, "dma_engine buffer NULL\n");
return -EINVAL;
}
dma_get_converter_format(dma_engine, &hda_format);
sample_rate_div = ((get_hdafmt_rate(hda_format) >> 0) & 3) + 1;
sample_rate_mul = ((get_hdafmt_rate(hda_format) >> 3) & 3) + 1;
num_chans = get_hdafmt_chs(hda_format) + 1;
hda_frame_size_words = ((sample_rate_div == 0) ? 0 :
(num_chans * sample_rate_mul / sample_rate_div));
if (hda_frame_size_words == 0) {
codec_dbg(codec, "frmsz zero\n");
return -EINVAL;
}
buffer_size_words = min(buffer_size_words,
(unsigned int)(UC_RANGE(chip_addx, 1) ?
65536 : 32768));
buffer_size_words -= buffer_size_words % hda_frame_size_words;
codec_dbg(codec,
"chpadr=0x%08x frmsz=%u nchan=%u "
"rate_mul=%u div=%u bufsz=%u\n",
chip_addx, hda_frame_size_words, num_chans,
sample_rate_mul, sample_rate_div, buffer_size_words);
if (buffer_size_words < hda_frame_size_words) {
codec_dbg(codec, "dspxfr_one_seg:failed\n");
return -EINVAL;
}
remainder_words = words_to_write % hda_frame_size_words;
data_remainder = data;
chip_addx_remainder = chip_addx;
data += remainder_words;
chip_addx += remainder_words*sizeof(u32);
words_to_write -= remainder_words;
while (words_to_write != 0) {
run_size_words = min(buffer_size_words, words_to_write);
codec_dbg(codec, "dspxfr (seg loop)cnt=%u rs=%u remainder=%u\n",
words_to_write, run_size_words, remainder_words);
dma_xfer(dma_engine, data, run_size_words*sizeof(u32));
if (!comm_dma_setup_done) {
status = dsp_dma_stop(codec, dma_chan, ovly);
if (status < 0)
return status;
status = dsp_dma_setup_common(codec, chip_addx,
dma_chan, port_map_mask, ovly);
if (status < 0)
return status;
comm_dma_setup_done = true;
}
status = dsp_dma_setup(codec, chip_addx,
run_size_words, dma_chan);
if (status < 0)
return status;
status = dsp_dma_start(codec, dma_chan, ovly);
if (status < 0)
return status;
if (!dsp_is_dma_active(codec, dma_chan)) {
codec_dbg(codec, "dspxfr:DMA did not start\n");
return -EIO;
}
status = dma_set_state(dma_engine, DMA_STATE_RUN);
if (status < 0)
return status;
if (remainder_words != 0) {
status = chipio_write_multiple(codec,
chip_addx_remainder,
data_remainder,
remainder_words);
if (status < 0)
return status;
remainder_words = 0;
}
if (hci_write) {
status = dspxfr_hci_write(codec, hci_write);
if (status < 0)
return status;
hci_write = NULL;
}
timeout = jiffies + msecs_to_jiffies(2000);
do {
dma_active = dsp_is_dma_active(codec, dma_chan);
if (!dma_active)
break;
msleep(20);
} while (time_before(jiffies, timeout));
if (dma_active)
break;
codec_dbg(codec, "+++++ DMA complete\n");
dma_set_state(dma_engine, DMA_STATE_STOP);
status = dma_reset(dma_engine);
if (status < 0)
return status;
data += run_size_words;
chip_addx += run_size_words*sizeof(u32);
words_to_write -= run_size_words;
}
if (remainder_words != 0) {
status = chipio_write_multiple(codec, chip_addx_remainder,
data_remainder, remainder_words);
}
return status;
}
/**
* dspxfr_image - Write the entire DSP image of a DSP code/data overlay to DSP memories
*
* @codec: the HDA codec
* @fls_data: pointer to a fast load image
* @reloc: Relocation address for loading single-segment overlays, or 0 for
* no relocation
* @sample_rate: sampling rate of the stream used for DSP download
* @channels: channels of the stream used for DSP download
* @ovly: TRUE if overlay format is required
*
* Returns zero or a negative error code.
*/
static int dspxfr_image(struct hda_codec *codec,
const struct dsp_image_seg *fls_data,
unsigned int reloc,
unsigned int sample_rate,
unsigned short channels,
bool ovly)
{
struct ca0132_spec *spec = codec->spec;
int status;
unsigned short hda_format = 0;
unsigned int response;
unsigned char stream_id = 0;
struct dma_engine *dma_engine;
unsigned int dma_chan;
unsigned int port_map_mask;
if (fls_data == NULL)
return -EINVAL;
dma_engine = kzalloc(sizeof(*dma_engine), GFP_KERNEL);
if (!dma_engine)
return -ENOMEM;
dma_engine->dmab = kzalloc(sizeof(*dma_engine->dmab), GFP_KERNEL);
if (!dma_engine->dmab) {
kfree(dma_engine);
return -ENOMEM;
}
dma_engine->codec = codec;
dma_convert_to_hda_format(codec, sample_rate, channels, &hda_format);
dma_engine->m_converter_format = hda_format;
dma_engine->buf_size = (ovly ? DSP_DMA_WRITE_BUFLEN_OVLY :
DSP_DMA_WRITE_BUFLEN_INIT) * 2;
dma_chan = ovly ? INVALID_DMA_CHANNEL : 0;
status = codec_set_converter_format(codec, WIDGET_CHIP_CTRL,
hda_format, &response);
if (status < 0) {
codec_dbg(codec, "set converter format fail\n");
goto exit;
}
status = snd_hda_codec_load_dsp_prepare(codec,
dma_engine->m_converter_format,
dma_engine->buf_size,
dma_engine->dmab);
if (status < 0)
goto exit;
spec->dsp_stream_id = status;
if (ovly) {
status = dspio_alloc_dma_chan(codec, &dma_chan);
if (status < 0) {
codec_dbg(codec, "alloc dmachan fail\n");
dma_chan = INVALID_DMA_CHANNEL;
goto exit;
}
}
port_map_mask = 0;
status = dsp_allocate_ports_format(codec, hda_format,
&port_map_mask);
if (status < 0) {
codec_dbg(codec, "alloc ports fail\n");
goto exit;
}
stream_id = dma_get_stream_id(dma_engine);
status = codec_set_converter_stream_channel(codec,
WIDGET_CHIP_CTRL, stream_id, 0, &response);
if (status < 0) {
codec_dbg(codec, "set stream chan fail\n");
goto exit;
}
while ((fls_data != NULL) && !is_last(fls_data)) {
if (!is_valid(fls_data)) {
codec_dbg(codec, "FLS check fail\n");
status = -EINVAL;
goto exit;
}
status = dspxfr_one_seg(codec, fls_data, reloc,
dma_engine, dma_chan,
port_map_mask, ovly);
if (status < 0)
break;
if (is_hci_prog_list_seg(fls_data))
fls_data = get_next_seg_ptr(fls_data);
if ((fls_data != NULL) && !is_last(fls_data))
fls_data = get_next_seg_ptr(fls_data);
}
if (port_map_mask != 0)
status = dsp_free_ports(codec);
if (status < 0)
goto exit;
status = codec_set_converter_stream_channel(codec,
WIDGET_CHIP_CTRL, 0, 0, &response);
exit:
if (ovly && (dma_chan != INVALID_DMA_CHANNEL))
dspio_free_dma_chan(codec, dma_chan);
if (dma_engine->dmab->area)
snd_hda_codec_load_dsp_cleanup(codec, dma_engine->dmab);
kfree(dma_engine->dmab);
kfree(dma_engine);
return status;
}
/*
* CA0132 DSP download stuffs.
*/
static void dspload_post_setup(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
codec_dbg(codec, "---- dspload_post_setup ------\n");
if (!ca0132_use_alt_functions(spec)) {
/*set DSP speaker to 2.0 configuration*/
chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x18), 0x08080080);
chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x19), 0x3f800000);
/*update write pointer*/
chipio_write(codec, XRAM_XRAM_INST_OFFSET(0x29), 0x00000002);
}
}
/**
* dspload_image - Download DSP from a DSP Image Fast Load structure.
*
* @codec: the HDA codec
* @fls: pointer to a fast load image
* @ovly: TRUE if overlay format is required
* @reloc: Relocation address for loading single-segment overlays, or 0 for
* no relocation
* @autostart: TRUE if DSP starts after loading; ignored if ovly is TRUE
* @router_chans: number of audio router channels to be allocated (0 means use
* internal defaults; max is 32)
*
* Download DSP from a DSP Image Fast Load structure. This structure is a
* linear, non-constant sized element array of structures, each of which
* contain the count of the data to be loaded, the data itself, and the
* corresponding starting chip address of the starting data location.
* Returns zero or a negative error code.
*/
static int dspload_image(struct hda_codec *codec,
const struct dsp_image_seg *fls,
bool ovly,
unsigned int reloc,
bool autostart,
int router_chans)
{
int status = 0;
unsigned int sample_rate;
unsigned short channels;
codec_dbg(codec, "---- dspload_image begin ------\n");
if (router_chans == 0) {
if (!ovly)
router_chans = DMA_TRANSFER_FRAME_SIZE_NWORDS;
else
router_chans = DMA_OVERLAY_FRAME_SIZE_NWORDS;
}
sample_rate = 48000;
channels = (unsigned short)router_chans;
while (channels > 16) {
sample_rate *= 2;
channels /= 2;
}
do {
codec_dbg(codec, "Ready to program DMA\n");
if (!ovly)
status = dsp_reset(codec);
if (status < 0)
break;
codec_dbg(codec, "dsp_reset() complete\n");
status = dspxfr_image(codec, fls, reloc, sample_rate, channels,
ovly);
if (status < 0)
break;
codec_dbg(codec, "dspxfr_image() complete\n");
if (autostart && !ovly) {
dspload_post_setup(codec);
status = dsp_set_run_state(codec);
}
codec_dbg(codec, "LOAD FINISHED\n");
} while (0);
return status;
}
#ifdef CONFIG_SND_HDA_CODEC_CA0132_DSP
static bool dspload_is_loaded(struct hda_codec *codec)
{
unsigned int data = 0;
int status = 0;
status = chipio_read(codec, 0x40004, &data);
if ((status < 0) || (data != 1))
return false;
return true;
}
#else
#define dspload_is_loaded(codec) false
#endif
static bool dspload_wait_loaded(struct hda_codec *codec)
{
unsigned long timeout = jiffies + msecs_to_jiffies(2000);
do {
if (dspload_is_loaded(codec)) {
codec_info(codec, "ca0132 DSP downloaded and running\n");
return true;
}
msleep(20);
} while (time_before(jiffies, timeout));
codec_err(codec, "ca0132 failed to download DSP\n");
return false;
}
/*
* ca0113 related functions. The ca0113 acts as the HDA bus for the pci-e
* based cards, and has a second mmio region, region2, that's used for special
* commands.
*/
/*
* For cards with PCI-E region2 (Sound Blaster Z/ZxR, Recon3D, and AE-5)
* the mmio address 0x320 is used to set GPIO pins. The format for the data
* The first eight bits are just the number of the pin. So far, I've only seen
* this number go to 7.
* AE-5 note: The AE-5 seems to use pins 2 and 3 to somehow set the color value
* of the on-card LED. It seems to use pin 2 for data, then toggles 3 to on and
* then off to send that bit.
*/
static void ca0113_mmio_gpio_set(struct hda_codec *codec, unsigned int gpio_pin,
bool enable)
{
struct ca0132_spec *spec = codec->spec;
unsigned short gpio_data;
gpio_data = gpio_pin & 0xF;
gpio_data |= ((enable << 8) & 0x100);
writew(gpio_data, spec->mem_base + 0x320);
}
/*
* Special pci region2 commands that are only used by the AE-5. They follow
* a set format, and require reads at certain points to seemingly 'clear'
* the response data. My first tests didn't do these reads, and would cause
* the card to get locked up until the memory was read. These commands
* seem to work with three distinct values that I've taken to calling group,
* target-id, and value.
*/
static void ca0113_mmio_command_set(struct hda_codec *codec, unsigned int group,
unsigned int target, unsigned int value)
{
struct ca0132_spec *spec = codec->spec;
unsigned int write_val;
writel(0x0000007e, spec->mem_base + 0x210);
readl(spec->mem_base + 0x210);
writel(0x0000005a, spec->mem_base + 0x210);
readl(spec->mem_base + 0x210);
readl(spec->mem_base + 0x210);
writel(0x00800005, spec->mem_base + 0x20c);
writel(group, spec->mem_base + 0x804);
writel(0x00800005, spec->mem_base + 0x20c);
write_val = (target & 0xff);
write_val |= (value << 8);
writel(write_val, spec->mem_base + 0x204);
/*
* Need delay here or else it goes too fast and works inconsistently.
*/
msleep(20);
readl(spec->mem_base + 0x860);
readl(spec->mem_base + 0x854);
readl(spec->mem_base + 0x840);
writel(0x00800004, spec->mem_base + 0x20c);
writel(0x00000000, spec->mem_base + 0x210);
readl(spec->mem_base + 0x210);
readl(spec->mem_base + 0x210);
}
/*
* This second type of command is used for setting the sound filter type.
*/
static void ca0113_mmio_command_set_type2(struct hda_codec *codec,
unsigned int group, unsigned int target, unsigned int value)
{
struct ca0132_spec *spec = codec->spec;
unsigned int write_val;
writel(0x0000007e, spec->mem_base + 0x210);
readl(spec->mem_base + 0x210);
writel(0x0000005a, spec->mem_base + 0x210);
readl(spec->mem_base + 0x210);
readl(spec->mem_base + 0x210);
writel(0x00800003, spec->mem_base + 0x20c);
writel(group, spec->mem_base + 0x804);
writel(0x00800005, spec->mem_base + 0x20c);
write_val = (target & 0xff);
write_val |= (value << 8);
writel(write_val, spec->mem_base + 0x204);
msleep(20);
readl(spec->mem_base + 0x860);
readl(spec->mem_base + 0x854);
readl(spec->mem_base + 0x840);
writel(0x00800004, spec->mem_base + 0x20c);
writel(0x00000000, spec->mem_base + 0x210);
readl(spec->mem_base + 0x210);
readl(spec->mem_base + 0x210);
}
/*
* Setup GPIO for the other variants of Core3D.
*/
/*
* Sets up the GPIO pins so that they are discoverable. If this isn't done,
* the card shows as having no GPIO pins.
*/
static void ca0132_gpio_init(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
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);
break;
case QUIRK_R3DI:
snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5B);
break;
default:
break;
}
}
/* Sets the GPIO for audio output. */
static void ca0132_gpio_setup(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
switch (ca0132_quirk(spec)) {
case QUIRK_SBZ:
snd_hda_codec_write(codec, 0x01, 0,
AC_VERB_SET_GPIO_DIRECTION, 0x07);
snd_hda_codec_write(codec, 0x01, 0,
AC_VERB_SET_GPIO_MASK, 0x07);
snd_hda_codec_write(codec, 0x01, 0,
AC_VERB_SET_GPIO_DATA, 0x04);
snd_hda_codec_write(codec, 0x01, 0,
AC_VERB_SET_GPIO_DATA, 0x06);
break;
case QUIRK_R3DI:
snd_hda_codec_write(codec, 0x01, 0,
AC_VERB_SET_GPIO_DIRECTION, 0x1E);
snd_hda_codec_write(codec, 0x01, 0,
AC_VERB_SET_GPIO_MASK, 0x1F);
snd_hda_codec_write(codec, 0x01, 0,
AC_VERB_SET_GPIO_DATA, 0x0C);
break;
default:
break;
}
}
/*
* GPIO control functions for the Recon3D integrated.
*/
enum r3di_gpio_bit {
/* Bit 1 - Switch between front/rear mic. 0 = rear, 1 = front */
R3DI_MIC_SELECT_BIT = 1,
/* Bit 2 - Switch between headphone/line out. 0 = Headphone, 1 = Line */
R3DI_OUT_SELECT_BIT = 2,
/*
* I dunno what this actually does, but it stays on until the dsp
* is downloaded.
*/
R3DI_GPIO_DSP_DOWNLOADING = 3,
/*
* Same as above, no clue what it does, but it comes on after the dsp
* is downloaded.
*/
R3DI_GPIO_DSP_DOWNLOADED = 4
};
enum r3di_mic_select {
/* Set GPIO bit 1 to 0 for rear mic */
R3DI_REAR_MIC = 0,
/* Set GPIO bit 1 to 1 for front microphone*/
R3DI_FRONT_MIC = 1
};
enum r3di_out_select {
/* Set GPIO bit 2 to 0 for headphone */
R3DI_HEADPHONE_OUT = 0,
/* Set GPIO bit 2 to 1 for speaker */
R3DI_LINE_OUT = 1
};
enum r3di_dsp_status {
/* Set GPIO bit 3 to 1 until DSP is downloaded */
R3DI_DSP_DOWNLOADING = 0,
/* Set GPIO bit 4 to 1 once DSP is downloaded */
R3DI_DSP_DOWNLOADED = 1
};
static void r3di_gpio_mic_set(struct hda_codec *codec,
enum r3di_mic_select cur_mic)
{
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_mic) {
case R3DI_REAR_MIC:
cur_gpio &= ~(1 << R3DI_MIC_SELECT_BIT);
break;
case R3DI_FRONT_MIC:
cur_gpio |= (1 << R3DI_MIC_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)
{
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 (dsp_status) {
case R3DI_DSP_DOWNLOADING:
cur_gpio |= (1 << R3DI_GPIO_DSP_DOWNLOADING);
snd_hda_codec_write(codec, codec->core.afg, 0,
AC_VERB_SET_GPIO_DATA, cur_gpio);
break;
case R3DI_DSP_DOWNLOADED:
/* Set DOWNLOADING bit to 0. */
cur_gpio &= ~(1 << R3DI_GPIO_DSP_DOWNLOADING);
snd_hda_codec_write(codec, codec->core.afg, 0,
AC_VERB_SET_GPIO_DATA, cur_gpio);
cur_gpio |= (1 << R3DI_GPIO_DSP_DOWNLOADED);
break;
}
snd_hda_codec_write(codec, codec->core.afg, 0,
AC_VERB_SET_GPIO_DATA, cur_gpio);
}
/*
* PCM callbacks
*/
static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
snd_hda_codec_setup_stream(codec, spec->dacs[0], stream_tag, 0, format);
return 0;
}
static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
if (spec->dsp_state == DSP_DOWNLOADING)
return 0;
/*If Playback effects are on, allow stream some time to flush
*effects tail*/
if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
msleep(50);
snd_hda_codec_cleanup_stream(codec, spec->dacs[0]);
return 0;
}
static unsigned int ca0132_playback_pcm_delay(struct hda_pcm_stream *info,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
unsigned int latency = DSP_PLAYBACK_INIT_LATENCY;
struct snd_pcm_runtime *runtime = substream->runtime;
if (spec->dsp_state != DSP_DOWNLOADED)
return 0;
/* Add latency if playback enhancement and either effect is enabled. */
if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) {
if ((spec->effects_switch[SURROUND - EFFECT_START_NID]) ||
(spec->effects_switch[DIALOG_PLUS - EFFECT_START_NID]))
latency += DSP_PLAY_ENHANCEMENT_LATENCY;
}
/* Applying Speaker EQ adds latency as well. */
if (spec->cur_out_type == SPEAKER_OUT)
latency += DSP_SPEAKER_OUT_LATENCY;
return (latency * runtime->rate) / 1000;
}
/*
* Digital out
*/
static int ca0132_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
return snd_hda_multi_out_dig_open(codec, &spec->multiout);
}
static int ca0132_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
return snd_hda_multi_out_dig_prepare(codec, &spec->multiout,
stream_tag, format, substream);
}
static int ca0132_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
}
static int ca0132_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
return snd_hda_multi_out_dig_close(codec, &spec->multiout);
}
/*
* Analog capture
*/
static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
unsigned int stream_tag,
unsigned int format,
struct snd_pcm_substream *substream)
{
snd_hda_codec_setup_stream(codec, hinfo->nid,
stream_tag, 0, format);
return 0;
}
static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
if (spec->dsp_state == DSP_DOWNLOADING)
return 0;
snd_hda_codec_cleanup_stream(codec, hinfo->nid);
return 0;
}
static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info,
struct hda_codec *codec,
struct snd_pcm_substream *substream)
{
struct ca0132_spec *spec = codec->spec;
unsigned int latency = DSP_CAPTURE_INIT_LATENCY;
struct snd_pcm_runtime *runtime = substream->runtime;
if (spec->dsp_state != DSP_DOWNLOADED)
return 0;
if (spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID])
latency += DSP_CRYSTAL_VOICE_LATENCY;
return (latency * runtime->rate) / 1000;
}
/*
* Controls stuffs.
*/
/*
* Mixer controls helpers.
*/
#define CA0132_CODEC_VOL_MONO(xname, nid, channel, dir) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
.subdevice = HDA_SUBDEV_AMP_FLAG, \
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
.info = ca0132_volume_info, \
.get = ca0132_volume_get, \
.put = ca0132_volume_put, \
.tlv = { .c = ca0132_volume_tlv }, \
.private_value = HDA_COMPOSE_AMP_VAL(nid, channel, 0, dir) }
/*
* Creates a mixer control that uses defaults of HDA_CODEC_VOL except for the
* volume put, which is used for setting the DSP volume. This was done because
* the ca0132 functions were taking too much time and causing lag.
*/
#define CA0132_ALT_CODEC_VOL_MONO(xname, nid, channel, dir) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
.subdevice = HDA_SUBDEV_AMP_FLAG, \
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \
SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \
.info = snd_hda_mixer_amp_volume_info, \
.get = snd_hda_mixer_amp_volume_get, \
.put = ca0132_alt_volume_put, \
.tlv = { .c = snd_hda_mixer_amp_tlv }, \
.private_value = HDA_COMPOSE_AMP_VAL(nid, channel, 0, dir) }
#define CA0132_CODEC_MUTE_MONO(xname, nid, channel, dir) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = xname, \
.subdevice = HDA_SUBDEV_AMP_FLAG, \
.info = snd_hda_mixer_amp_switch_info, \
.get = ca0132_switch_get, \
.put = ca0132_switch_put, \
.private_value = HDA_COMPOSE_AMP_VAL(nid, channel, 0, dir) }
/* stereo */
#define CA0132_CODEC_VOL(xname, nid, dir) \
CA0132_CODEC_VOL_MONO(xname, nid, 3, dir)
#define CA0132_ALT_CODEC_VOL(xname, nid, dir) \
CA0132_ALT_CODEC_VOL_MONO(xname, nid, 3, dir)
#define CA0132_CODEC_MUTE(xname, nid, dir) \
CA0132_CODEC_MUTE_MONO(xname, nid, 3, dir)
/* lookup tables */
/*
* Lookup table with decibel values for the DSP. When volume is changed in
* Windows, the DSP is also sent the dB value in floating point. In Windows,
* these values have decimal points, probably because the Windows driver
* actually uses floating point. We can't here, so I made a lookup table of
* values -90 to 9. -90 is the lowest decibel value for both the ADC's and the
* DAC's, and 9 is the maximum.
*/
static const unsigned int float_vol_db_lookup[] = {
0xC2B40000, 0xC2B20000, 0xC2B00000, 0xC2AE0000, 0xC2AC0000, 0xC2AA0000,
0xC2A80000, 0xC2A60000, 0xC2A40000, 0xC2A20000, 0xC2A00000, 0xC29E0000,
0xC29C0000, 0xC29A0000, 0xC2980000, 0xC2960000, 0xC2940000, 0xC2920000,
0xC2900000, 0xC28E0000, 0xC28C0000, 0xC28A0000, 0xC2880000, 0xC2860000,
0xC2840000, 0xC2820000, 0xC2800000, 0xC27C0000, 0xC2780000, 0xC2740000,
0xC2700000, 0xC26C0000, 0xC2680000, 0xC2640000, 0xC2600000, 0xC25C0000,
0xC2580000, 0xC2540000, 0xC2500000, 0xC24C0000, 0xC2480000, 0xC2440000,
0xC2400000, 0xC23C0000, 0xC2380000, 0xC2340000, 0xC2300000, 0xC22C0000,
0xC2280000, 0xC2240000, 0xC2200000, 0xC21C0000, 0xC2180000, 0xC2140000,
0xC2100000, 0xC20C0000, 0xC2080000, 0xC2040000, 0xC2000000, 0xC1F80000,
0xC1F00000, 0xC1E80000, 0xC1E00000, 0xC1D80000, 0xC1D00000, 0xC1C80000,
0xC1C00000, 0xC1B80000, 0xC1B00000, 0xC1A80000, 0xC1A00000, 0xC1980000,
0xC1900000, 0xC1880000, 0xC1800000, 0xC1700000, 0xC1600000, 0xC1500000,
0xC1400000, 0xC1300000, 0xC1200000, 0xC1100000, 0xC1000000, 0xC0E00000,
0xC0C00000, 0xC0A00000, 0xC0800000, 0xC0400000, 0xC0000000, 0xBF800000,
0x00000000, 0x3F800000, 0x40000000, 0x40400000, 0x40800000, 0x40A00000,
0x40C00000, 0x40E00000, 0x41000000, 0x41100000
};
/*
* This table counts from float 0 to 1 in increments of .01, which is
* useful for a few different sliders.
*/
static const unsigned int float_zero_to_one_lookup[] = {
0x00000000, 0x3C23D70A, 0x3CA3D70A, 0x3CF5C28F, 0x3D23D70A, 0x3D4CCCCD,
0x3D75C28F, 0x3D8F5C29, 0x3DA3D70A, 0x3DB851EC, 0x3DCCCCCD, 0x3DE147AE,
0x3DF5C28F, 0x3E051EB8, 0x3E0F5C29, 0x3E19999A, 0x3E23D70A, 0x3E2E147B,
0x3E3851EC, 0x3E428F5C, 0x3E4CCCCD, 0x3E570A3D, 0x3E6147AE, 0x3E6B851F,
0x3E75C28F, 0x3E800000, 0x3E851EB8, 0x3E8A3D71, 0x3E8F5C29, 0x3E947AE1,
0x3E99999A, 0x3E9EB852, 0x3EA3D70A, 0x3EA8F5C3, 0x3EAE147B, 0x3EB33333,
0x3EB851EC, 0x3EBD70A4, 0x3EC28F5C, 0x3EC7AE14, 0x3ECCCCCD, 0x3ED1EB85,
0x3ED70A3D, 0x3EDC28F6, 0x3EE147AE, 0x3EE66666, 0x3EEB851F, 0x3EF0A3D7,
0x3EF5C28F, 0x3EFAE148, 0x3F000000, 0x3F028F5C, 0x3F051EB8, 0x3F07AE14,
0x3F0A3D71, 0x3F0CCCCD, 0x3F0F5C29, 0x3F11EB85, 0x3F147AE1, 0x3F170A3D,
0x3F19999A, 0x3F1C28F6, 0x3F1EB852, 0x3F2147AE, 0x3F23D70A, 0x3F266666,
0x3F28F5C3, 0x3F2B851F, 0x3F2E147B, 0x3F30A3D7, 0x3F333333, 0x3F35C28F,
0x3F3851EC, 0x3F3AE148, 0x3F3D70A4, 0x3F400000, 0x3F428F5C, 0x3F451EB8,
0x3F47AE14, 0x3F4A3D71, 0x3F4CCCCD, 0x3F4F5C29, 0x3F51EB85, 0x3F547AE1,
0x3F570A3D, 0x3F59999A, 0x3F5C28F6, 0x3F5EB852, 0x3F6147AE, 0x3F63D70A,
0x3F666666, 0x3F68F5C3, 0x3F6B851F, 0x3F6E147B, 0x3F70A3D7, 0x3F733333,
0x3F75C28F, 0x3F7851EC, 0x3F7AE148, 0x3F7D70A4, 0x3F800000
};
/*
* This table counts from float 10 to 1000, which is the range of the x-bass
* crossover slider in Windows.
*/
static const unsigned int float_xbass_xover_lookup[] = {
0x41200000, 0x41A00000, 0x41F00000, 0x42200000, 0x42480000, 0x42700000,
0x428C0000, 0x42A00000, 0x42B40000, 0x42C80000, 0x42DC0000, 0x42F00000,
0x43020000, 0x430C0000, 0x43160000, 0x43200000, 0x432A0000, 0x43340000,
0x433E0000, 0x43480000, 0x43520000, 0x435C0000, 0x43660000, 0x43700000,
0x437A0000, 0x43820000, 0x43870000, 0x438C0000, 0x43910000, 0x43960000,
0x439B0000, 0x43A00000, 0x43A50000, 0x43AA0000, 0x43AF0000, 0x43B40000,
0x43B90000, 0x43BE0000, 0x43C30000, 0x43C80000, 0x43CD0000, 0x43D20000,
0x43D70000, 0x43DC0000, 0x43E10000, 0x43E60000, 0x43EB0000, 0x43F00000,
0x43F50000, 0x43FA0000, 0x43FF0000, 0x44020000, 0x44048000, 0x44070000,
0x44098000, 0x440C0000, 0x440E8000, 0x44110000, 0x44138000, 0x44160000,
0x44188000, 0x441B0000, 0x441D8000, 0x44200000, 0x44228000, 0x44250000,
0x44278000, 0x442A0000, 0x442C8000, 0x442F0000, 0x44318000, 0x44340000,
0x44368000, 0x44390000, 0x443B8000, 0x443E0000, 0x44408000, 0x44430000,
0x44458000, 0x44480000, 0x444A8000, 0x444D0000, 0x444F8000, 0x44520000,
0x44548000, 0x44570000, 0x44598000, 0x445C0000, 0x445E8000, 0x44610000,
0x44638000, 0x44660000, 0x44688000, 0x446B0000, 0x446D8000, 0x44700000,
0x44728000, 0x44750000, 0x44778000, 0x447A0000
};
/* The following are for tuning of products */
#ifdef ENABLE_TUNING_CONTROLS
static const unsigned int voice_focus_vals_lookup[] = {
0x41A00000, 0x41A80000, 0x41B00000, 0x41B80000, 0x41C00000, 0x41C80000,
0x41D00000, 0x41D80000, 0x41E00000, 0x41E80000, 0x41F00000, 0x41F80000,
0x42000000, 0x42040000, 0x42080000, 0x420C0000, 0x42100000, 0x42140000,
0x42180000, 0x421C0000, 0x42200000, 0x42240000, 0x42280000, 0x422C0000,
0x42300000, 0x42340000, 0x42380000, 0x423C0000, 0x42400000, 0x42440000,
0x42480000, 0x424C0000, 0x42500000, 0x42540000, 0x42580000, 0x425C0000,
0x42600000, 0x42640000, 0x42680000, 0x426C0000, 0x42700000, 0x42740000,
0x42780000, 0x427C0000, 0x42800000, 0x42820000, 0x42840000, 0x42860000,
0x42880000, 0x428A0000, 0x428C0000, 0x428E0000, 0x42900000, 0x42920000,
0x42940000, 0x42960000, 0x42980000, 0x429A0000, 0x429C0000, 0x429E0000,
0x42A00000, 0x42A20000, 0x42A40000, 0x42A60000, 0x42A80000, 0x42AA0000,
0x42AC0000, 0x42AE0000, 0x42B00000, 0x42B20000, 0x42B40000, 0x42B60000,
0x42B80000, 0x42BA0000, 0x42BC0000, 0x42BE0000, 0x42C00000, 0x42C20000,
0x42C40000, 0x42C60000, 0x42C80000, 0x42CA0000, 0x42CC0000, 0x42CE0000,
0x42D00000, 0x42D20000, 0x42D40000, 0x42D60000, 0x42D80000, 0x42DA0000,
0x42DC0000, 0x42DE0000, 0x42E00000, 0x42E20000, 0x42E40000, 0x42E60000,
0x42E80000, 0x42EA0000, 0x42EC0000, 0x42EE0000, 0x42F00000, 0x42F20000,
0x42F40000, 0x42F60000, 0x42F80000, 0x42FA0000, 0x42FC0000, 0x42FE0000,
0x43000000, 0x43010000, 0x43020000, 0x43030000, 0x43040000, 0x43050000,
0x43060000, 0x43070000, 0x43080000, 0x43090000, 0x430A0000, 0x430B0000,
0x430C0000, 0x430D0000, 0x430E0000, 0x430F0000, 0x43100000, 0x43110000,
0x43120000, 0x43130000, 0x43140000, 0x43150000, 0x43160000, 0x43170000,
0x43180000, 0x43190000, 0x431A0000, 0x431B0000, 0x431C0000, 0x431D0000,
0x431E0000, 0x431F0000, 0x43200000, 0x43210000, 0x43220000, 0x43230000,
0x43240000, 0x43250000, 0x43260000, 0x43270000, 0x43280000, 0x43290000,
0x432A0000, 0x432B0000, 0x432C0000, 0x432D0000, 0x432E0000, 0x432F0000,
0x43300000, 0x43310000, 0x43320000, 0x43330000, 0x43340000
};
static const unsigned int mic_svm_vals_lookup[] = {
0x00000000, 0x3C23D70A, 0x3CA3D70A, 0x3CF5C28F, 0x3D23D70A, 0x3D4CCCCD,
0x3D75C28F, 0x3D8F5C29, 0x3DA3D70A, 0x3DB851EC, 0x3DCCCCCD, 0x3DE147AE,
0x3DF5C28F, 0x3E051EB8, 0x3E0F5C29, 0x3E19999A, 0x3E23D70A, 0x3E2E147B,
0x3E3851EC, 0x3E428F5C, 0x3E4CCCCD, 0x3E570A3D, 0x3E6147AE, 0x3E6B851F,
0x3E75C28F, 0x3E800000, 0x3E851EB8, 0x3E8A3D71, 0x3E8F5C29, 0x3E947AE1,
0x3E99999A, 0x3E9EB852, 0x3EA3D70A, 0x3EA8F5C3, 0x3EAE147B, 0x3EB33333,
0x3EB851EC, 0x3EBD70A4, 0x3EC28F5C, 0x3EC7AE14, 0x3ECCCCCD, 0x3ED1EB85,
0x3ED70A3D, 0x3EDC28F6, 0x3EE147AE, 0x3EE66666, 0x3EEB851F, 0x3EF0A3D7,
0x3EF5C28F, 0x3EFAE148, 0x3F000000, 0x3F028F5C, 0x3F051EB8, 0x3F07AE14,
0x3F0A3D71, 0x3F0CCCCD, 0x3F0F5C29, 0x3F11EB85, 0x3F147AE1, 0x3F170A3D,
0x3F19999A, 0x3F1C28F6, 0x3F1EB852, 0x3F2147AE, 0x3F23D70A, 0x3F266666,
0x3F28F5C3, 0x3F2B851F, 0x3F2E147B, 0x3F30A3D7, 0x3F333333, 0x3F35C28F,
0x3F3851EC, 0x3F3AE148, 0x3F3D70A4, 0x3F400000, 0x3F428F5C, 0x3F451EB8,
0x3F47AE14, 0x3F4A3D71, 0x3F4CCCCD, 0x3F4F5C29, 0x3F51EB85, 0x3F547AE1,
0x3F570A3D, 0x3F59999A, 0x3F5C28F6, 0x3F5EB852, 0x3F6147AE, 0x3F63D70A,
0x3F666666, 0x3F68F5C3, 0x3F6B851F, 0x3F6E147B, 0x3F70A3D7, 0x3F733333,
0x3F75C28F, 0x3F7851EC, 0x3F7AE148, 0x3F7D70A4, 0x3F800000
};
static const unsigned int equalizer_vals_lookup[] = {
0xC1C00000, 0xC1B80000, 0xC1B00000, 0xC1A80000, 0xC1A00000, 0xC1980000,
0xC1900000, 0xC1880000, 0xC1800000, 0xC1700000, 0xC1600000, 0xC1500000,
0xC1400000, 0xC1300000, 0xC1200000, 0xC1100000, 0xC1000000, 0xC0E00000,
0xC0C00000, 0xC0A00000, 0xC0800000, 0xC0400000, 0xC0000000, 0xBF800000,
0x00000000, 0x3F800000, 0x40000000, 0x40400000, 0x40800000, 0x40A00000,
0x40C00000, 0x40E00000, 0x41000000, 0x41100000, 0x41200000, 0x41300000,
0x41400000, 0x41500000, 0x41600000, 0x41700000, 0x41800000, 0x41880000,
0x41900000, 0x41980000, 0x41A00000, 0x41A80000, 0x41B00000, 0x41B80000,
0x41C00000
};
static int tuning_ctl_set(struct hda_codec *codec, hda_nid_t nid,
const unsigned int *lookup, int idx)
{
int i = 0;
for (i = 0; i < TUNING_CTLS_COUNT; i++)
if (nid == ca0132_tuning_ctls[i].nid)
break;
snd_hda_power_up(codec);
dspio_set_param(codec, ca0132_tuning_ctls[i].mid, 0x20,
ca0132_tuning_ctls[i].req,
&(lookup[idx]), sizeof(unsigned int));
snd_hda_power_down(codec);
return 1;
}
static int tuning_ctl_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;
hda_nid_t nid = get_amp_nid(kcontrol);
long *valp = ucontrol->value.integer.value;
int idx = nid - TUNING_CTL_START_NID;
*valp = spec->cur_ctl_vals[idx];
return 0;
}
static int voice_focus_ctl_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
int chs = get_amp_channels(kcontrol);
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = chs == 3 ? 2 : 1;
uinfo->value.integer.min = 20;
uinfo->value.integer.max = 180;
uinfo->value.integer.step = 1;
return 0;
}
static int voice_focus_ctl_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;
hda_nid_t nid = get_amp_nid(kcontrol);
long *valp = ucontrol->value.integer.value;
int idx;
idx = nid - TUNING_CTL_START_NID;
/* any change? */
if (spec->cur_ctl_vals[idx] == *valp)
return 0;
spec->cur_ctl_vals[idx] = *valp;
idx = *valp - 20;
tuning_ctl_set(codec, nid, voice_focus_vals_lookup, idx);
return 1;
}
static int mic_svm_ctl_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
int chs = get_amp_channels(kcontrol);
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = chs == 3 ? 2 : 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 100;
uinfo->value.integer.step = 1;
return 0;
}
static int mic_svm_ctl_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;
hda_nid_t nid = get_amp_nid(kcontrol);
long *valp = ucontrol->value.integer.value;
int idx;
idx = nid - TUNING_CTL_START_NID;
/* any change? */
if (spec->cur_ctl_vals[idx] == *valp)
return 0;
spec->cur_ctl_vals[idx] = *valp;
idx = *valp;
tuning_ctl_set(codec, nid, mic_svm_vals_lookup, idx);
return 0;
}
static int equalizer_ctl_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
int chs = get_amp_channels(kcontrol);
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = chs == 3 ? 2 : 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 48;
uinfo->value.integer.step = 1;
return 0;
}
static int equalizer_ctl_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;
hda_nid_t nid = get_amp_nid(kcontrol);
long *valp = ucontrol->value.integer.value;
int idx;
idx = nid - TUNING_CTL_START_NID;
/* any change? */
if (spec->cur_ctl_vals[idx] == *valp)
return 0;
spec->cur_ctl_vals[idx] = *valp;
idx = *valp;
tuning_ctl_set(codec, nid, equalizer_vals_lookup, idx);
return 1;
}
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(voice_focus_db_scale, 2000, 100, 0);
static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(eq_db_scale, -2400, 100, 0);
static int add_tuning_control(struct hda_codec *codec,
hda_nid_t pnid, hda_nid_t nid,
const char *name, int dir)
{
char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
int type = dir ? HDA_INPUT : HDA_OUTPUT;
struct snd_kcontrol_new knew =
HDA_CODEC_VOLUME_MONO(namestr, nid, 1, 0, type);
knew.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
SNDRV_CTL_ELEM_ACCESS_TLV_READ;
knew.tlv.c = 0;
knew.tlv.p = 0;
switch (pnid) {
case VOICE_FOCUS:
knew.info = voice_focus_ctl_info;
knew.get = tuning_ctl_get;
knew.put = voice_focus_ctl_put;
knew.tlv.p = voice_focus_db_scale;
break;
case MIC_SVM:
knew.info = mic_svm_ctl_info;
knew.get = tuning_ctl_get;
knew.put = mic_svm_ctl_put;
break;
case EQUALIZER:
knew.info = equalizer_ctl_info;
knew.get = tuning_ctl_get;
knew.put = equalizer_ctl_put;
knew.tlv.p = eq_db_scale;
break;
default:
return 0;
}
knew.private_value =
HDA_COMPOSE_AMP_VAL(nid, 1, 0, type);
sprintf(namestr, "%s %s Volume", name, dirstr[dir]);
return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
}
static int add_tuning_ctls(struct hda_codec *codec)
{
int i;
int err;
for (i = 0; i < TUNING_CTLS_COUNT; i++) {
err = add_tuning_control(codec,
ca0132_tuning_ctls[i].parent_nid,
ca0132_tuning_ctls[i].nid,
ca0132_tuning_ctls[i].name,
ca0132_tuning_ctls[i].direct);
if (err < 0)
return err;
}
return 0;
}
static void ca0132_init_tuning_defaults(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
int i;
/* Wedge Angle defaults to 30. 10 below is 30 - 20. 20 is min. */
spec->cur_ctl_vals[WEDGE_ANGLE - TUNING_CTL_START_NID] = 10;
/* SVM level defaults to 0.74. */
spec->cur_ctl_vals[SVM_LEVEL - TUNING_CTL_START_NID] = 74;
/* EQ defaults to 0dB. */
for (i = 2; i < TUNING_CTLS_COUNT; i++)
spec->cur_ctl_vals[i] = 24;
}
#endif /*ENABLE_TUNING_CONTROLS*/
/*
* Select the active output.
* If autodetect is enabled, output will be selected based on jack detection.
* If jack inserted, headphone will be selected, else built-in speakers
* If autodetect is disabled, output will be selected based on selection.
*/
static int ca0132_select_out(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
unsigned int pin_ctl;
int jack_present;
int auto_jack;
unsigned int tmp;
int err;
codec_dbg(codec, "ca0132_select_out\n");
snd_hda_power_up_pm(codec);
auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID];
if (auto_jack)
jack_present = snd_hda_jack_detect(codec, spec->unsol_tag_hp);
else
jack_present =
spec->vnode_lswitch[VNID_HP_SEL - VNODE_START_NID];
if (jack_present)
spec->cur_out_type = HEADPHONE_OUT;
else
spec->cur_out_type = SPEAKER_OUT;
if (spec->cur_out_type == SPEAKER_OUT) {
codec_dbg(codec, "ca0132_select_out speaker\n");
/*speaker out config*/
tmp = FLOAT_ONE;
err = dspio_set_uint_param(codec, 0x80, 0x04, tmp);
if (err < 0)
goto exit;
/*enable speaker EQ*/
tmp = FLOAT_ONE;
err = dspio_set_uint_param(codec, 0x8f, 0x00, tmp);
if (err < 0)
goto exit;
/* Setup EAPD */
snd_hda_codec_write(codec, spec->out_pins[1], 0,
VENDOR_CHIPIO_EAPD_SEL_SET, 0x02);
snd_hda_codec_write(codec, spec->out_pins[0], 0,
AC_VERB_SET_EAPD_BTLENABLE, 0x00);
snd_hda_codec_write(codec, spec->out_pins[0], 0,
VENDOR_CHIPIO_EAPD_SEL_SET, 0x00);
snd_hda_codec_write(codec, spec->out_pins[0], 0,
AC_VERB_SET_EAPD_BTLENABLE, 0x02);
/* 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 speaker 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);
} else {
codec_dbg(codec, "ca0132_select_out hp\n");
/*headphone out config*/
tmp = FLOAT_ZERO;
err = dspio_set_uint_param(codec, 0x80, 0x04, tmp);
if (err < 0)
goto exit;
/*disable speaker EQ*/
tmp = FLOAT_ZERO;
err = dspio_set_uint_param(codec, 0x8f, 0x00, tmp);
if (err < 0)
goto exit;
/* Setup EAPD */
snd_hda_codec_write(codec, spec->out_pins[0], 0,
VENDOR_CHIPIO_EAPD_SEL_SET, 0x00);
snd_hda_codec_write(codec, spec->out_pins[0], 0,
AC_VERB_SET_EAPD_BTLENABLE, 0x00);
snd_hda_codec_write(codec, spec->out_pins[1], 0,
VENDOR_CHIPIO_EAPD_SEL_SET, 0x02);
snd_hda_codec_write(codec, spec->out_pins[0], 0,
AC_VERB_SET_EAPD_BTLENABLE, 0x02);
/* 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);
/* enable headphone*/
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);
}
exit:
snd_hda_power_down_pm(codec);
return err < 0 ? err : 0;
}
static int ae5_headphone_gain_set(struct hda_codec *codec, long val);
static int zxr_headphone_gain_set(struct hda_codec *codec, long val);
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;
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;
/* 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 (quirk == QUIRK_AE5 || quirk == QUIRK_AE7) {
err = dspio_set_uint_param(codec, 0x96,
SPEAKER_FULL_RANGE_SURROUND_L_R, FLOAT_ONE);
if (err < 0)
return err;
}
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.
*/
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;
int quirk = ca0132_quirk(spec);
unsigned int i;
*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;
}
}
}
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]);
}
}
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,
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.
*/
static int ca0132_alt_select_out(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
unsigned int tmp, outfx_set;
int jack_present;
int auto_jack;
int err;
/* Default Headphone is rear headphone */
hda_nid_t headphone_nid = spec->out_pins[1];
codec_dbg(codec, "%s\n", __func__);
snd_hda_power_up_pm(codec);
auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID];
/*
* If headphone rear or front is plugged in, set to headphone.
* If neither is plugged in, set to rear line out. Only if
* hp/speaker auto detect is enabled.
*/
if (auto_jack) {
jack_present = snd_hda_jack_detect(codec, spec->unsol_tag_hp) ||
snd_hda_jack_detect(codec, spec->unsol_tag_front_hp);
if (jack_present)
spec->cur_out_type = HEADPHONE_OUT;
else
spec->cur_out_type = SPEAKER_OUT;
} else
spec->cur_out_type = spec->out_enum_val;
outfx_set = spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID];
/* Begin DSP output switch, mute DSP volume. */
err = dspio_set_uint_param(codec, 0x96, SPEAKER_TUNING_MUTE, FLOAT_ONE);
if (err < 0)
goto exit;
if (ca0132_alt_select_out_quirk_set(codec) < 0)
goto exit;
switch (spec->cur_out_type) {
case SPEAKER_OUT:
codec_dbg(codec, "%s speaker\n", __func__);
/* Enable EAPD */
snd_hda_codec_write(codec, spec->out_pins[0], 0,
AC_VERB_SET_EAPD_BTLENABLE, 0x01);
/* 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
* 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
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);
/* 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))
headphone_nid = spec->out_pins[2];
else if (snd_hda_jack_detect(codec, spec->unsol_tag_hp))
headphone_nid = spec->out_pins[1];
ca0132_set_out_node_pincfg(codec, headphone_nid, 1, 1);
if (outfx_set)
err = dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
else
err = dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ZERO);
if (err < 0)
goto exit;
break;
}
/*
* 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 (outfx_set)
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;
if (spec->cur_out_type == SPEAKER_OUT) {
err = ca0132_alt_set_full_range_speaker(codec);
if (err < 0)
goto exit;
}
exit:
snd_hda_power_down_pm(codec);
return err < 0 ? err : 0;
}
static void ca0132_unsol_hp_delayed(struct work_struct *work)
{
struct ca0132_spec *spec = container_of(
to_delayed_work(work), struct ca0132_spec, unsol_hp_work);
struct hda_jack_tbl *jack;
if (ca0132_use_alt_functions(spec))
ca0132_alt_select_out(spec->codec);
else
ca0132_select_out(spec->codec);
jack = snd_hda_jack_tbl_get(spec->codec, spec->unsol_tag_hp);
if (jack) {
jack->block_report = 0;
snd_hda_jack_report_sync(spec->codec);
}
}
static void ca0132_set_dmic(struct hda_codec *codec, int enable);
static int ca0132_mic_boost_set(struct hda_codec *codec, long val);
static void resume_mic1(struct hda_codec *codec, unsigned int oldval);
static int stop_mic1(struct hda_codec *codec);
static int ca0132_cvoice_switch_set(struct hda_codec *codec);
static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val);
/*
* Select the active VIP source
*/
static int ca0132_set_vipsource(struct hda_codec *codec, int val)
{
struct ca0132_spec *spec = codec->spec;
unsigned int tmp;
if (spec->dsp_state != DSP_DOWNLOADED)
return 0;
/* if CrystalVoice if off, vipsource should be 0 */
if (!spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID] ||
(val == 0)) {
chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, 0);
chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
if (spec->cur_mic_type == DIGITAL_MIC)
tmp = FLOAT_TWO;
else
tmp = FLOAT_ONE;
dspio_set_uint_param(codec, 0x80, 0x00, tmp);
tmp = FLOAT_ZERO;
dspio_set_uint_param(codec, 0x80, 0x05, tmp);
} else {
chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_16_000);
chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_16_000);
if (spec->cur_mic_type == DIGITAL_MIC)
tmp = FLOAT_TWO;
else
tmp = FLOAT_ONE;
dspio_set_uint_param(codec, 0x80, 0x00, tmp);
tmp = FLOAT_ONE;
dspio_set_uint_param(codec, 0x80, 0x05, tmp);
msleep(20);
chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, val);
}
return 1;
}
static int ca0132_alt_set_vipsource(struct hda_codec *codec, int val)
{
struct ca0132_spec *spec = codec->spec;
unsigned int tmp;
if (spec->dsp_state != DSP_DOWNLOADED)
return 0;
codec_dbg(codec, "%s\n", __func__);
chipio_set_stream_control(codec, 0x03, 0);
chipio_set_stream_control(codec, 0x04, 0);
/* if CrystalVoice is off, vipsource should be 0 */
if (!spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID] ||
(val == 0) || spec->in_enum_val == REAR_LINE_IN) {
codec_dbg(codec, "%s: off.", __func__);
chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, 0);
tmp = FLOAT_ZERO;
dspio_set_uint_param(codec, 0x80, 0x05, tmp);
chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
if (ca0132_quirk(spec) == QUIRK_R3DI)
chipio_set_conn_rate(codec, 0x0F, SR_96_000);
if (spec->in_enum_val == REAR_LINE_IN)
tmp = FLOAT_ZERO;
else {
if (ca0132_quirk(spec) == QUIRK_SBZ)
tmp = FLOAT_THREE;
else
tmp = FLOAT_ONE;
}
dspio_set_uint_param(codec, 0x80, 0x00, tmp);
} else {
codec_dbg(codec, "%s: on.", __func__);
chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_16_000);
chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_16_000);
if (ca0132_quirk(spec) == QUIRK_R3DI)
chipio_set_conn_rate(codec, 0x0F, SR_16_000);
if (spec->effects_switch[VOICE_FOCUS - EFFECT_START_NID])
tmp = FLOAT_TWO;
else
tmp = FLOAT_ONE;
dspio_set_uint_param(codec, 0x80, 0x00, tmp);
tmp = FLOAT_ONE;
dspio_set_uint_param(codec, 0x80, 0x05, tmp);
msleep(20);
chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, val);
}
chipio_set_stream_control(codec, 0x03, 1);
chipio_set_stream_control(codec, 0x04, 1);
return 1;
}
/*
* Select the active microphone.
* If autodetect is enabled, mic will be selected based on jack detection.
* If jack inserted, ext.mic will be selected, else built-in mic
* If autodetect is disabled, mic will be selected based on selection.
*/
static int ca0132_select_mic(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
int jack_present;
int auto_jack;
codec_dbg(codec, "ca0132_select_mic\n");
snd_hda_power_up_pm(codec);
auto_jack = spec->vnode_lswitch[VNID_AMIC1_ASEL - VNODE_START_NID];
if (auto_jack)
jack_present = snd_hda_jack_detect(codec, spec->unsol_tag_amic1);
else
jack_present =
spec->vnode_lswitch[VNID_AMIC1_SEL - VNODE_START_NID];
if (jack_present)
spec->cur_mic_type = LINE_MIC_IN;
else
spec->cur_mic_type = DIGITAL_MIC;
if (spec->cur_mic_type == DIGITAL_MIC) {
/* enable digital Mic */
chipio_set_conn_rate(codec, MEM_CONNID_DMIC, SR_32_000);
ca0132_set_dmic(codec, 1);
ca0132_mic_boost_set(codec, 0);
/* set voice focus */
ca0132_effects_set(codec, VOICE_FOCUS,
spec->effects_switch
[VOICE_FOCUS - EFFECT_START_NID]);
} else {
/* disable digital Mic */
chipio_set_conn_rate(codec, MEM_CONNID_DMIC, SR_96_000);
ca0132_set_dmic(codec, 0);
ca0132_mic_boost_set(codec, spec->cur_mic_boost);
/* disable voice focus */
ca0132_effects_set(codec, VOICE_FOCUS, 0);
}
snd_hda_power_down_pm(codec);
return 0;
}
/*
* Select the active input.
* Mic detection isn't used, because it's kind of pointless on the SBZ.
* The front mic has no jack-detection, so the only way to switch to it
* is to do it manually in alsamixer.
*/
static int ca0132_alt_select_in(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
unsigned int tmp;
codec_dbg(codec, "%s\n", __func__);
snd_hda_power_up_pm(codec);
chipio_set_stream_control(codec, 0x03, 0);
chipio_set_stream_control(codec, 0x04, 0);
spec->cur_mic_type = spec->in_enum_val;
switch (spec->cur_mic_type) {
case REAR_MIC:
switch (ca0132_quirk(spec)) {
case QUIRK_SBZ:
case QUIRK_R3D:
ca0113_mmio_gpio_set(codec, 0, false);
tmp = FLOAT_THREE;
break;
case QUIRK_ZXR:
tmp = FLOAT_THREE;
break;
case QUIRK_R3DI:
r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
tmp = FLOAT_ONE;
break;
case QUIRK_AE5:
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;
}
chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
if (ca0132_quirk(spec) == QUIRK_R3DI)
chipio_set_conn_rate(codec, 0x0F, SR_96_000);
dspio_set_uint_param(codec, 0x80, 0x00, tmp);
chipio_set_stream_control(codec, 0x03, 1);
chipio_set_stream_control(codec, 0x04, 1);
switch (ca0132_quirk(spec)) {
case QUIRK_SBZ:
chipio_write(codec, 0x18B098, 0x0000000C);
chipio_write(codec, 0x18B09C, 0x0000000C);
break;
case QUIRK_ZXR:
chipio_write(codec, 0x18B098, 0x0000000C);
chipio_write(codec, 0x18B09C, 0x000000CC);
break;
case QUIRK_AE5:
chipio_write(codec, 0x18B098, 0x0000000C);
chipio_write(codec, 0x18B09C, 0x0000004C);
break;
default:
break;
}
ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val);
break;
case REAR_LINE_IN:
ca0132_mic_boost_set(codec, 0);
switch (ca0132_quirk(spec)) {
case QUIRK_SBZ:
case QUIRK_R3D:
ca0113_mmio_gpio_set(codec, 0, false);
break;
case QUIRK_R3DI:
r3di_gpio_mic_set(codec, R3DI_REAR_MIC);
break;
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;
}
chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
if (ca0132_quirk(spec) == QUIRK_R3DI)
chipio_set_conn_rate(codec, 0x0F, SR_96_000);
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)) {
case QUIRK_SBZ:
case QUIRK_AE5:
chipio_write(codec, 0x18B098, 0x00000000);
chipio_write(codec, 0x18B09C, 0x00000000);
break;
default:
break;
}
chipio_set_stream_control(codec, 0x03, 1);
chipio_set_stream_control(codec, 0x04, 1);
break;
case FRONT_MIC:
switch (ca0132_quirk(spec)) {
case QUIRK_SBZ:
case QUIRK_R3D:
ca0113_mmio_gpio_set(codec, 0, true);
ca0113_mmio_gpio_set(codec, 5, false);
tmp = FLOAT_THREE;
break;
case QUIRK_R3DI:
r3di_gpio_mic_set(codec, R3DI_FRONT_MIC);
tmp = FLOAT_ONE;
break;
case QUIRK_AE5:
ca0113_mmio_command_set(codec, 0x30, 0x28, 0x3f);
tmp = FLOAT_THREE;
break;
default:
tmp = FLOAT_ONE;
break;
}
chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
if (ca0132_quirk(spec) == QUIRK_R3DI)
chipio_set_conn_rate(codec, 0x0F, SR_96_000);
dspio_set_uint_param(codec, 0x80, 0x00, tmp);
chipio_set_stream_control(codec, 0x03, 1);
chipio_set_stream_control(codec, 0x04, 1);
switch (ca0132_quirk(spec)) {
case QUIRK_SBZ:
chipio_write(codec, 0x18B098, 0x0000000C);
chipio_write(codec, 0x18B09C, 0x000000CC);
break;
case QUIRK_AE5:
chipio_write(codec, 0x18B098, 0x0000000C);
chipio_write(codec, 0x18B09C, 0x0000004C);
break;
default:
break;
}
ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val);
break;
}
ca0132_cvoice_switch_set(codec);
snd_hda_power_down_pm(codec);
return 0;
}
/*
* Check if VNODE settings take effect immediately.
*/
static bool ca0132_is_vnode_effective(struct hda_codec *codec,
hda_nid_t vnid,
hda_nid_t *shared_nid)
{
struct ca0132_spec *spec = codec->spec;
hda_nid_t nid;
switch (vnid) {
case VNID_SPK:
nid = spec->shared_out_nid;
break;
case VNID_MIC:
nid = spec->shared_mic_nid;
break;
default:
return false;
}
if (shared_nid)
*shared_nid = nid;
return true;
}
/*
* The following functions are control change helpers.
* They return 0 if no changed. Return 1 if changed.
*/
static int ca0132_voicefx_set(struct hda_codec *codec, int enable)
{
struct ca0132_spec *spec = codec->spec;
unsigned int tmp;
/* based on CrystalVoice state to enable VoiceFX. */
if (enable) {
tmp = spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID] ?
FLOAT_ONE : FLOAT_ZERO;
} else {
tmp = FLOAT_ZERO;
}
dspio_set_uint_param(codec, ca0132_voicefx.mid,
ca0132_voicefx.reqs[0], tmp);
return 1;
}
/*
* Set the effects parameters
*/
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, channel_cfg;
int num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT;
int err = 0;
int idx = nid - EFFECT_START_NID;
if ((idx < 0) || (idx >= num_fx))
return 0; /* no changed */
/* for out effect, qualify with PE */
if ((nid >= OUT_EFFECT_START_NID) && (nid < OUT_EFFECT_END_NID)) {
/* if PE if off, turn off out effects. */
if (!spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
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 */
if ((nid >= IN_EFFECT_START_NID) && (nid < IN_EFFECT_END_NID)) {
/* if CrystalVoice if off, turn off in effects. */
if (!spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID])
val = 0;
/* Voice Focus applies to 2-ch Mic, Digital Mic */
if ((nid == VOICE_FOCUS) && (spec->cur_mic_type != DIGITAL_MIC))
val = 0;
/* If Voice Focus on SBZ, set to two channel. */
if ((nid == VOICE_FOCUS) && ca0132_use_pci_mmio(spec)
&& (spec->cur_mic_type != REAR_LINE_IN)) {
if (spec->effects_switch[CRYSTAL_VOICE -
EFFECT_START_NID]) {
if (spec->effects_switch[VOICE_FOCUS -
EFFECT_START_NID]) {
tmp = FLOAT_TWO;
val = 1;
} else
tmp = FLOAT_ONE;
dspio_set_uint_param(codec, 0x80, 0x00, tmp);
}
}
/*
* For SBZ noise reduction, there's an extra command
* to module ID 0x47. No clue why.
*/
if ((nid == NOISE_REDUCTION) && ca0132_use_pci_mmio(spec)
&& (spec->cur_mic_type != REAR_LINE_IN)) {
if (spec->effects_switch[CRYSTAL_VOICE -
EFFECT_START_NID]) {
if (spec->effects_switch[NOISE_REDUCTION -
EFFECT_START_NID])
tmp = FLOAT_ONE;
else
tmp = FLOAT_ZERO;
} else
tmp = FLOAT_ZERO;
dspio_set_uint_param(codec, 0x47, 0x00, tmp);
}
/* If rear line in disable effects. */
if (ca0132_use_alt_functions(spec) &&
spec->in_enum_val == REAR_LINE_IN)
val = 0;
}
codec_dbg(codec, "ca0132_effect_set: nid=0x%x, val=%ld\n",
nid, val);
on = (val == 0) ? FLOAT_ZERO : FLOAT_ONE;
err = dspio_set_uint_param(codec, ca0132_effects[idx].mid,
ca0132_effects[idx].reqs[0], on);
if (err < 0)
return 0; /* no changed */
return 1;
}
/*
* Turn on/off Playback Enhancements
*/
static int ca0132_pe_switch_set(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
hda_nid_t nid;
int i, ret = 0;
codec_dbg(codec, "ca0132_pe_switch_set: val=%ld\n",
spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]);
if (ca0132_use_alt_functions(spec))
ca0132_alt_select_out(codec);
i = OUT_EFFECT_START_NID - EFFECT_START_NID;
nid = OUT_EFFECT_START_NID;
/* PE affects all out effects */
for (; nid < OUT_EFFECT_END_NID; nid++, i++)
ret |= ca0132_effects_set(codec, nid, spec->effects_switch[i]);
return ret;
}
/* Check if Mic1 is streaming, if so, stop streaming */
static int stop_mic1(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
unsigned int oldval = snd_hda_codec_read(codec, spec->adcs[0], 0,
AC_VERB_GET_CONV, 0);
if (oldval != 0)
snd_hda_codec_write(codec, spec->adcs[0], 0,
AC_VERB_SET_CHANNEL_STREAMID,
0);
return oldval;
}
/* Resume Mic1 streaming if it was stopped. */
static void resume_mic1(struct hda_codec *codec, unsigned int oldval)
{
struct ca0132_spec *spec = codec->spec;
/* Restore the previous stream and channel */
if (oldval != 0)
snd_hda_codec_write(codec, spec->adcs[0], 0,
AC_VERB_SET_CHANNEL_STREAMID,
oldval);
}
/*
* Turn on/off CrystalVoice
*/
static int ca0132_cvoice_switch_set(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
hda_nid_t nid;
int i, ret = 0;
unsigned int oldval;
codec_dbg(codec, "ca0132_cvoice_switch_set: val=%ld\n",
spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID]);
i = IN_EFFECT_START_NID - EFFECT_START_NID;
nid = IN_EFFECT_START_NID;
/* CrystalVoice affects all in effects */
for (; nid < IN_EFFECT_END_NID; nid++, i++)
ret |= ca0132_effects_set(codec, nid, spec->effects_switch[i]);
/* including VoiceFX */
ret |= ca0132_voicefx_set(codec, (spec->voicefx_val ? 1 : 0));
/* set correct vipsource */
oldval = stop_mic1(codec);
if (ca0132_use_alt_functions(spec))
ret |= ca0132_alt_set_vipsource(codec, 1);
else
ret |= ca0132_set_vipsource(codec, 1);
resume_mic1(codec, oldval);
return ret;
}
static int ca0132_mic_boost_set(struct hda_codec *codec, long val)
{
struct ca0132_spec *spec = codec->spec;
int ret = 0;
if (val) /* on */
ret = snd_hda_codec_amp_update(codec, spec->input_pins[0], 0,
HDA_INPUT, 0, HDA_AMP_VOLMASK, 3);
else /* off */
ret = snd_hda_codec_amp_update(codec, spec->input_pins[0], 0,
HDA_INPUT, 0, HDA_AMP_VOLMASK, 0);
return ret;
}
static int ca0132_alt_mic_boost_set(struct hda_codec *codec, long val)
{
struct ca0132_spec *spec = codec->spec;
int ret = 0;
ret = snd_hda_codec_amp_update(codec, spec->input_pins[0], 0,
HDA_INPUT, 0, HDA_AMP_VOLMASK, val);
return ret;
}
static int ae5_headphone_gain_set(struct hda_codec *codec, long val)
{
unsigned int i;
for (i = 0; i < 4; i++)
ca0113_mmio_command_set(codec, 0x48, 0x11 + i,
ae5_headphone_gain_presets[val].vals[i]);
return 0;
}
/*
* gpio pin 1 is a relay that switches on/off, apparently setting the headphone
* amplifier to handle a 600 ohm load.
*/
static int zxr_headphone_gain_set(struct hda_codec *codec, long val)
{
ca0113_mmio_gpio_set(codec, 1, val);
return 0;
}
static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
hda_nid_t nid = get_amp_nid(kcontrol);
hda_nid_t shared_nid = 0;
bool effective;
int ret = 0;
struct ca0132_spec *spec = codec->spec;
int auto_jack;
if (nid == VNID_HP_SEL) {
auto_jack =
spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID];
if (!auto_jack) {
if (ca0132_use_alt_functions(spec))
ca0132_alt_select_out(codec);
else
ca0132_select_out(codec);
}
return 1;
}
if (nid == VNID_AMIC1_SEL) {
auto_jack =
spec->vnode_lswitch[VNID_AMIC1_ASEL - VNODE_START_NID];
if (!auto_jack)
ca0132_select_mic(codec);
return 1;
}
if (nid == VNID_HP_ASEL) {
if (ca0132_use_alt_functions(spec))
ca0132_alt_select_out(codec);
else
ca0132_select_out(codec);
return 1;
}
if (nid == VNID_AMIC1_ASEL) {
ca0132_select_mic(codec);
return 1;
}
/* if effective conditions, then update hw immediately. */
effective = ca0132_is_vnode_effective(codec, nid, &shared_nid);
if (effective) {
int dir = get_amp_direction(kcontrol);
int ch = get_amp_channels(kcontrol);
unsigned long pval;
mutex_lock(&codec->control_mutex);
pval = kcontrol->private_value;
kcontrol->private_value = HDA_COMPOSE_AMP_VAL(shared_nid, ch,
0, dir);
ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
kcontrol->private_value = pval;
mutex_unlock(&codec->control_mutex);
}
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
* Chromebook. I figured they were probably tuned specifically for it, and left
* out for a reason.
*/
/* 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)
{
int i = 0;
unsigned int y;
/*
* For X_BASS, req 2 is actually crossover freq instead of
* effect level
*/
if (nid == X_BASS)
y = 2;
else
y = 1;
snd_hda_power_up(codec);
if (nid == XBASS_XOVER) {
for (i = 0; i < OUT_EFFECTS_COUNT; i++)
if (ca0132_effects[i].nid == X_BASS)
break;
dspio_set_param(codec, ca0132_effects[i].mid, 0x20,
ca0132_effects[i].reqs[1],
&(lookup[idx - 1]), sizeof(unsigned int));
} else {
/* Find the actual effect structure */
for (i = 0; i < OUT_EFFECTS_COUNT; i++)
if (nid == ca0132_effects[i].nid)
break;
dspio_set_param(codec, ca0132_effects[i].mid, 0x20,
ca0132_effects[i].reqs[y],
&(lookup[idx]), sizeof(unsigned int));
}
snd_hda_power_down(codec);
return 0;
}
static int ca0132_alt_xbass_xover_slider_ctl_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;
long *valp = ucontrol->value.integer.value;
hda_nid_t nid = get_amp_nid(kcontrol);
if (nid == BASS_REDIRECTION_XOVER)
*valp = spec->bass_redirect_xover_freq;
else
*valp = spec->xbass_xover_freq;
return 0;
}
static int ca0132_alt_slider_ctl_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;
hda_nid_t nid = get_amp_nid(kcontrol);
long *valp = ucontrol->value.integer.value;
int idx = nid - OUT_EFFECT_START_NID;
*valp = spec->fx_ctl_val[idx];
return 0;
}
/*
* The X-bass crossover starts at 10hz, so the min is 1. The
* frequency is set in multiples of 10.
*/
static int ca0132_alt_xbass_xover_slider_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 1;
uinfo->value.integer.min = 1;
uinfo->value.integer.max = 100;
uinfo->value.integer.step = 1;
return 0;
}
static int ca0132_alt_effect_slider_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
int chs = get_amp_channels(kcontrol);
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = chs == 3 ? 2 : 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 100;
uinfo->value.integer.step = 1;
return 0;
}
static int ca0132_alt_xbass_xover_slider_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;
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 (*cur_val == *valp)
return 0;
*cur_val = *valp;
idx = *valp;
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;
}
static int ca0132_alt_effect_slider_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;
hda_nid_t nid = get_amp_nid(kcontrol);
long *valp = ucontrol->value.integer.value;
int idx;
idx = nid - EFFECT_START_NID;
/* any change? */
if (spec->fx_ctl_val[idx] == *valp)
return 0;
spec->fx_ctl_val[idx] = *valp;
idx = *valp;
ca0132_alt_slider_ctl_set(codec, nid, float_zero_to_one_lookup, idx);
return 0;
}
/*
* Mic Boost Enum for alternative ca0132 codecs. I didn't like that the original
* only has off or full 30 dB, and didn't like making a volume slider that has
* traditional 0-100 in alsamixer that goes in big steps. I like enum better.
*/
#define MIC_BOOST_NUM_OF_STEPS 4
#define MIC_BOOST_ENUM_MAX_STRLEN 10
static int ca0132_alt_mic_boost_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
char *sfx = "dB";
char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = MIC_BOOST_NUM_OF_STEPS;
if (uinfo->value.enumerated.item >= MIC_BOOST_NUM_OF_STEPS)
uinfo->value.enumerated.item = MIC_BOOST_NUM_OF_STEPS - 1;
sprintf(namestr, "%d %s", (uinfo->value.enumerated.item * 10), sfx);
strcpy(uinfo->value.enumerated.name, namestr);
return 0;
}
static int ca0132_alt_mic_boost_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->mic_boost_enum_val;
return 0;
}
static int ca0132_alt_mic_boost_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 = MIC_BOOST_NUM_OF_STEPS;
if (sel >= items)
return 0;
codec_dbg(codec, "ca0132_alt_mic_boost: boost=%d\n",
sel);
spec->mic_boost_enum_val = sel;
if (spec->in_enum_val != REAR_LINE_IN)
ca0132_alt_mic_boost_set(codec, spec->mic_boost_enum_val);
return 1;
}
/*
* Sound BlasterX AE-5 Headphone Gain Controls.
*/
#define AE5_HEADPHONE_GAIN_MAX 3
static int ae5_headphone_gain_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
char *sfx = " Ohms)";
char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = AE5_HEADPHONE_GAIN_MAX;
if (uinfo->value.enumerated.item >= AE5_HEADPHONE_GAIN_MAX)
uinfo->value.enumerated.item = AE5_HEADPHONE_GAIN_MAX - 1;
sprintf(namestr, "%s %s",
ae5_headphone_gain_presets[uinfo->value.enumerated.item].name,
sfx);
strcpy(uinfo->value.enumerated.name, namestr);
return 0;
}
static int ae5_headphone_gain_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->ae5_headphone_gain_val;
return 0;
}
static int ae5_headphone_gain_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 = AE5_HEADPHONE_GAIN_MAX;
if (sel >= items)
return 0;
codec_dbg(codec, "ae5_headphone_gain: boost=%d\n",
sel);
spec->ae5_headphone_gain_val = sel;
if (spec->out_enum_val == HEADPHONE_OUT)
ae5_headphone_gain_set(codec, spec->ae5_headphone_gain_val);
return 1;
}
/*
* Sound BlasterX AE-5 sound filter enumerated control.
*/
#define AE5_SOUND_FILTER_MAX 3
static int ae5_sound_filter_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = AE5_SOUND_FILTER_MAX;
if (uinfo->value.enumerated.item >= AE5_SOUND_FILTER_MAX)
uinfo->value.enumerated.item = AE5_SOUND_FILTER_MAX - 1;
sprintf(namestr, "%s",
ae5_filter_presets[uinfo->value.enumerated.item].name);
strcpy(uinfo->value.enumerated.name, namestr);
return 0;
}
static int ae5_sound_filter_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->ae5_filter_val;
return 0;
}
static int ae5_sound_filter_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 = AE5_SOUND_FILTER_MAX;
if (sel >= items)
return 0;
codec_dbg(codec, "ae5_sound_filter: %s\n",
ae5_filter_presets[sel].name);
spec->ae5_filter_val = sel;
ca0113_mmio_command_set_type2(codec, 0x48, 0x07,
ae5_filter_presets[sel].val);
return 1;
}
/*
* Input Select Control for alternative ca0132 codecs. This exists because
* front microphone has no auto-detect, and we need a way to set the rear
* as line-in
*/
static int ca0132_alt_input_source_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = IN_SRC_NUM_OF_INPUTS;
if (uinfo->value.enumerated.item >= IN_SRC_NUM_OF_INPUTS)
uinfo->value.enumerated.item = IN_SRC_NUM_OF_INPUTS - 1;
strcpy(uinfo->value.enumerated.name,
in_src_str[uinfo->value.enumerated.item]);
return 0;
}
static int ca0132_alt_input_source_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->in_enum_val;
return 0;
}
static int ca0132_alt_input_source_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 = 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;
codec_dbg(codec, "ca0132_alt_input_select: sel=%d, preset=%s\n",
sel, in_src_str[sel]);
spec->in_enum_val = sel;
ca0132_alt_select_in(codec);
return 1;
}
/* Sound Blaster Z Output Select Control */
static int ca0132_alt_output_select_get_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = NUM_OF_OUTPUTS;
if (uinfo->value.enumerated.item >= NUM_OF_OUTPUTS)
uinfo->value.enumerated.item = NUM_OF_OUTPUTS - 1;
strcpy(uinfo->value.enumerated.name,
out_type_str[uinfo->value.enumerated.item]);
return 0;
}
static int ca0132_alt_output_select_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->out_enum_val;
return 0;
}
static int ca0132_alt_output_select_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 = NUM_OF_OUTPUTS;
unsigned int auto_jack;
if (sel >= items)
return 0;
codec_dbg(codec, "ca0132_alt_output_select: sel=%d, preset=%s\n",
sel, out_type_str[sel]);
spec->out_enum_val = sel;
auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID];
if (!auto_jack)
ca0132_alt_select_out(codec);
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 == SPEAKER_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
* and night, disregard the slider value and have uneditable values.
*/
#define NUM_OF_SVM_SETTINGS 3
static const char *const out_svm_set_enum_str[3] = {"Normal", "Loud", "Night" };
static int ca0132_alt_svm_setting_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = NUM_OF_SVM_SETTINGS;
if (uinfo->value.enumerated.item >= NUM_OF_SVM_SETTINGS)
uinfo->value.enumerated.item = NUM_OF_SVM_SETTINGS - 1;
strcpy(uinfo->value.enumerated.name,
out_svm_set_enum_str[uinfo->value.enumerated.item]);
return 0;
}
static int ca0132_alt_svm_setting_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->smart_volume_setting;
return 0;
}
static int ca0132_alt_svm_setting_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 = NUM_OF_SVM_SETTINGS;
unsigned int idx = SMART_VOLUME - EFFECT_START_NID;
unsigned int tmp;
if (sel >= items)
return 0;
codec_dbg(codec, "ca0132_alt_svm_setting: sel=%d, preset=%s\n",
sel, out_svm_set_enum_str[sel]);
spec->smart_volume_setting = sel;
switch (sel) {
case 0:
tmp = FLOAT_ZERO;
break;
case 1:
tmp = FLOAT_ONE;
break;
case 2:
tmp = FLOAT_TWO;
break;
default:
tmp = FLOAT_ZERO;
break;
}
/* Req 2 is the Smart Volume Setting req. */
dspio_set_uint_param(codec, ca0132_effects[idx].mid,
ca0132_effects[idx].reqs[2], tmp);
return 1;
}
/* Sound Blaster Z EQ preset controls */
static int ca0132_alt_eq_preset_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
unsigned int items = ARRAY_SIZE(ca0132_alt_eq_presets);
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,
ca0132_alt_eq_presets[uinfo->value.enumerated.item].name);
return 0;
}
static int ca0132_alt_eq_preset_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->eq_preset_val;
return 0;
}
static int ca0132_alt_eq_preset_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 i, err = 0;
int sel = ucontrol->value.enumerated.item[0];
unsigned int items = ARRAY_SIZE(ca0132_alt_eq_presets);
if (sel >= items)
return 0;
codec_dbg(codec, "%s: sel=%d, preset=%s\n", __func__, sel,
ca0132_alt_eq_presets[sel].name);
/*
* Idx 0 is default.
* Default needs to qualify with CrystalVoice state.
*/
for (i = 0; i < EQ_PRESET_MAX_PARAM_COUNT; i++) {
err = dspio_set_uint_param(codec, ca0132_alt_eq_enum.mid,
ca0132_alt_eq_enum.reqs[i],
ca0132_alt_eq_presets[sel].vals[i]);
if (err < 0)
break;
}
if (err >= 0)
spec->eq_preset_val = sel;
return 1;
}
static int ca0132_voicefx_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
unsigned int items = ARRAY_SIZE(ca0132_voicefx_presets);
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,
ca0132_voicefx_presets[uinfo->value.enumerated.item].name);
return 0;
}
static int ca0132_voicefx_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->voicefx_val;
return 0;
}
static int ca0132_voicefx_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 i, err = 0;
int sel = ucontrol->value.enumerated.item[0];
if (sel >= ARRAY_SIZE(ca0132_voicefx_presets))
return 0;
codec_dbg(codec, "ca0132_voicefx_put: sel=%d, preset=%s\n",
sel, ca0132_voicefx_presets[sel].name);
/*
* Idx 0 is default.
* Default needs to qualify with CrystalVoice state.
*/
for (i = 0; i < VOICEFX_MAX_PARAM_COUNT; i++) {
err = dspio_set_uint_param(codec, ca0132_voicefx.mid,
ca0132_voicefx.reqs[i],
ca0132_voicefx_presets[sel].vals[i]);
if (err < 0)
break;
}
if (err >= 0) {
spec->voicefx_val = sel;
/* enable voice fx */
ca0132_voicefx_set(codec, (sel ? 1 : 0));
}
return 1;
}
static int ca0132_switch_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;
hda_nid_t nid = get_amp_nid(kcontrol);
int ch = get_amp_channels(kcontrol);
long *valp = ucontrol->value.integer.value;
/* vnode */
if ((nid >= VNODE_START_NID) && (nid < VNODE_END_NID)) {
if (ch & 1) {
*valp = spec->vnode_lswitch[nid - VNODE_START_NID];
valp++;
}
if (ch & 2) {
*valp = spec->vnode_rswitch[nid - VNODE_START_NID];
valp++;
}
return 0;
}
/* effects, include PE and CrystalVoice */
if ((nid >= EFFECT_START_NID) && (nid < EFFECT_END_NID)) {
*valp = spec->effects_switch[nid - EFFECT_START_NID];
return 0;
}
/* mic boost */
if (nid == spec->input_pins[0]) {
*valp = spec->cur_mic_boost;
return 0;
}
if (nid == ZXR_HEADPHONE_GAIN) {
*valp = spec->zxr_gain_set;
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;
}
if (nid == BASS_REDIRECTION) {
*valp = spec->bass_redirection_val;
return 0;
}
return 0;
}
static int ca0132_switch_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;
hda_nid_t nid = get_amp_nid(kcontrol);
int ch = get_amp_channels(kcontrol);
long *valp = ucontrol->value.integer.value;
int changed = 1;
codec_dbg(codec, "ca0132_switch_put: nid=0x%x, val=%ld\n",
nid, *valp);
snd_hda_power_up(codec);
/* vnode */
if ((nid >= VNODE_START_NID) && (nid < VNODE_END_NID)) {
if (ch & 1) {
spec->vnode_lswitch[nid - VNODE_START_NID] = *valp;
valp++;
}
if (ch & 2) {
spec->vnode_rswitch[nid - VNODE_START_NID] = *valp;
valp++;
}
changed = ca0132_vnode_switch_set(kcontrol, ucontrol);
goto exit;
}
/* PE */
if (nid == PLAY_ENHANCEMENT) {
spec->effects_switch[nid - EFFECT_START_NID] = *valp;
changed = ca0132_pe_switch_set(codec);
goto exit;
}
/* CrystalVoice */
if (nid == CRYSTAL_VOICE) {
spec->effects_switch[nid - EFFECT_START_NID] = *valp;
changed = ca0132_cvoice_switch_set(codec);
goto exit;
}
/* out and in effects */
if (((nid >= OUT_EFFECT_START_NID) && (nid < OUT_EFFECT_END_NID)) ||
((nid >= IN_EFFECT_START_NID) && (nid < IN_EFFECT_END_NID))) {
spec->effects_switch[nid - EFFECT_START_NID] = *valp;
changed = ca0132_effects_set(codec, nid, *valp);
goto exit;
}
/* mic boost */
if (nid == spec->input_pins[0]) {
spec->cur_mic_boost = *valp;
if (ca0132_use_alt_functions(spec)) {
if (spec->in_enum_val != REAR_LINE_IN)
changed = ca0132_mic_boost_set(codec, *valp);
} else {
/* Mic boost does not apply to Digital Mic */
if (spec->cur_mic_type != DIGITAL_MIC)
changed = ca0132_mic_boost_set(codec, *valp);
}
goto exit;
}
if (nid == ZXR_HEADPHONE_GAIN) {
spec->zxr_gain_set = *valp;
if (spec->cur_out_type == HEADPHONE_OUT)
changed = zxr_headphone_gain_set(codec, *valp);
else
changed = 0;
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 == SPEAKER_OUT)
ca0132_alt_set_full_range_speaker(codec);
changed = 0;
}
if (nid == BASS_REDIRECTION) {
spec->bass_redirection_val = *valp;
if (spec->cur_out_type == SPEAKER_OUT)
ca0132_alt_surround_set_bass_redirection(codec, *valp);
changed = 0;
}
exit:
snd_hda_power_down(codec);
return changed;
}
/*
* Volume related
*/
/*
* Sets the internal DSP decibel level to match the DAC for output, and the
* ADC for input. Currently only the SBZ sets dsp capture volume level, and
* all alternative codecs set DSP playback volume.
*/
static void ca0132_alt_dsp_volume_put(struct hda_codec *codec, hda_nid_t nid)
{
struct ca0132_spec *spec = codec->spec;
unsigned int dsp_dir;
unsigned int lookup_val;
if (nid == VNID_SPK)
dsp_dir = DSP_VOL_OUT;
else
dsp_dir = DSP_VOL_IN;
lookup_val = spec->vnode_lvol[nid - VNODE_START_NID];
dspio_set_uint_param(codec,
ca0132_alt_vol_ctls[dsp_dir].mid,
ca0132_alt_vol_ctls[dsp_dir].reqs[0],
float_vol_db_lookup[lookup_val]);
lookup_val = spec->vnode_rvol[nid - VNODE_START_NID];
dspio_set_uint_param(codec,
ca0132_alt_vol_ctls[dsp_dir].mid,
ca0132_alt_vol_ctls[dsp_dir].reqs[1],
float_vol_db_lookup[lookup_val]);
dspio_set_uint_param(codec,
ca0132_alt_vol_ctls[dsp_dir].mid,
ca0132_alt_vol_ctls[dsp_dir].reqs[2], FLOAT_ZERO);
}
static int ca0132_volume_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct ca0132_spec *spec = codec->spec;
hda_nid_t nid = get_amp_nid(kcontrol);
int ch = get_amp_channels(kcontrol);
int dir = get_amp_direction(kcontrol);
unsigned long pval;
int err;
switch (nid) {
case VNID_SPK:
/* follow shared_out info */
nid = spec->shared_out_nid;
mutex_lock(&codec->control_mutex);
pval = kcontrol->private_value;
kcontrol->private_value = HDA_COMPOSE_AMP_VAL(nid, ch, 0, dir);
err = snd_hda_mixer_amp_volume_info(kcontrol, uinfo);
kcontrol->private_value = pval;
mutex_unlock(&codec->control_mutex);
break;
case VNID_MIC:
/* follow shared_mic info */
nid = spec->shared_mic_nid;
mutex_lock(&codec->control_mutex);
pval = kcontrol->private_value;
kcontrol->private_value = HDA_COMPOSE_AMP_VAL(nid, ch, 0, dir);
err = snd_hda_mixer_amp_volume_info(kcontrol, uinfo);
kcontrol->private_value = pval;
mutex_unlock(&codec->control_mutex);
break;
default:
err = snd_hda_mixer_amp_volume_info(kcontrol, uinfo);
}
return err;
}
static int ca0132_volume_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;
hda_nid_t nid = get_amp_nid(kcontrol);
int ch = get_amp_channels(kcontrol);
long *valp = ucontrol->value.integer.value;
/* store the left and right volume */
if (ch & 1) {
*valp = spec->vnode_lvol[nid - VNODE_START_NID];
valp++;
}
if (ch & 2) {
*valp = spec->vnode_rvol[nid - VNODE_START_NID];
valp++;
}
return 0;
}
static int ca0132_volume_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;
hda_nid_t nid = get_amp_nid(kcontrol);
int ch = get_amp_channels(kcontrol);
long *valp = ucontrol->value.integer.value;
hda_nid_t shared_nid = 0;
bool effective;
int changed = 1;
/* store the left and right volume */
if (ch & 1) {
spec->vnode_lvol[nid - VNODE_START_NID] = *valp;
valp++;
}
if (ch & 2) {
spec->vnode_rvol[nid - VNODE_START_NID] = *valp;
valp++;
}
/* if effective conditions, then update hw immediately. */
effective = ca0132_is_vnode_effective(codec, nid, &shared_nid);
if (effective) {
int dir = get_amp_direction(kcontrol);
unsigned long pval;
snd_hda_power_up(codec);
mutex_lock(&codec->control_mutex);
pval = kcontrol->private_value;
kcontrol->private_value = HDA_COMPOSE_AMP_VAL(shared_nid, ch,
0, dir);
changed = snd_hda_mixer_amp_volume_put(kcontrol, ucontrol);
kcontrol->private_value = pval;
mutex_unlock(&codec->control_mutex);
snd_hda_power_down(codec);
}
return changed;
}
/*
* This function is the same as the one above, because using an if statement
* inside of the above volume control for the DSP volume would cause too much
* lag. This is a lot more smooth.
*/
static int ca0132_alt_volume_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;
hda_nid_t nid = get_amp_nid(kcontrol);
int ch = get_amp_channels(kcontrol);
long *valp = ucontrol->value.integer.value;
hda_nid_t vnid = 0;
int changed;
switch (nid) {
case 0x02:
vnid = VNID_SPK;
break;
case 0x07:
vnid = VNID_MIC;
break;
}
/* store the left and right volume */
if (ch & 1) {
spec->vnode_lvol[vnid - VNODE_START_NID] = *valp;
valp++;
}
if (ch & 2) {
spec->vnode_rvol[vnid - VNODE_START_NID] = *valp;
valp++;
}
snd_hda_power_up(codec);
ca0132_alt_dsp_volume_put(codec, vnid);
mutex_lock(&codec->control_mutex);
changed = snd_hda_mixer_amp_volume_put(kcontrol, ucontrol);
mutex_unlock(&codec->control_mutex);
snd_hda_power_down(codec);
return changed;
}
static int ca0132_volume_tlv(struct snd_kcontrol *kcontrol, int op_flag,
unsigned int size, unsigned int __user *tlv)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct ca0132_spec *spec = codec->spec;
hda_nid_t nid = get_amp_nid(kcontrol);
int ch = get_amp_channels(kcontrol);
int dir = get_amp_direction(kcontrol);
unsigned long pval;
int err;
switch (nid) {
case VNID_SPK:
/* follow shared_out tlv */
nid = spec->shared_out_nid;
mutex_lock(&codec->control_mutex);
pval = kcontrol->private_value;
kcontrol->private_value = HDA_COMPOSE_AMP_VAL(nid, ch, 0, dir);
err = snd_hda_mixer_amp_tlv(kcontrol, op_flag, size, tlv);
kcontrol->private_value = pval;
mutex_unlock(&codec->control_mutex);
break;
case VNID_MIC:
/* follow shared_mic tlv */
nid = spec->shared_mic_nid;
mutex_lock(&codec->control_mutex);
pval = kcontrol->private_value;
kcontrol->private_value = HDA_COMPOSE_AMP_VAL(nid, ch, 0, dir);
err = snd_hda_mixer_amp_tlv(kcontrol, op_flag, size, tlv);
kcontrol->private_value = pval;
mutex_unlock(&codec->control_mutex);
break;
default:
err = snd_hda_mixer_amp_tlv(kcontrol, op_flag, size, tlv);
}
return err;
}
/* Add volume slider control for effect level */
static int ca0132_alt_add_effect_slider(struct hda_codec *codec, hda_nid_t nid,
const char *pfx, int dir)
{
char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
int type = dir ? HDA_INPUT : HDA_OUTPUT;
struct snd_kcontrol_new knew =
HDA_CODEC_VOLUME_MONO(namestr, nid, 1, 0, type);
sprintf(namestr, "FX: %s %s Volume", pfx, dirstr[dir]);
knew.tlv.c = NULL;
switch (nid) {
case XBASS_XOVER:
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;
break;
default:
knew.info = ca0132_alt_effect_slider_info;
knew.get = ca0132_alt_slider_ctl_get;
knew.put = ca0132_alt_effect_slider_put;
knew.private_value =
HDA_COMPOSE_AMP_VAL(nid, 1, 0, type);
break;
}
return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
}
/*
* Added FX: prefix for the alternative codecs, because otherwise the surround
* effect would conflict with the Surround sound volume control. Also seems more
* clear as to what the switches do. Left alone for others.
*/
static int add_fx_switch(struct hda_codec *codec, hda_nid_t nid,
const char *pfx, int dir)
{
struct ca0132_spec *spec = codec->spec;
char namestr[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
int type = dir ? HDA_INPUT : HDA_OUTPUT;
struct snd_kcontrol_new knew =
CA0132_CODEC_MUTE_MONO(namestr, nid, 1, type);
/* If using alt_controls, add FX: prefix. But, don't add FX:
* prefix to OutFX or InFX enable controls.
*/
if (ca0132_use_alt_controls(spec) && (nid <= IN_EFFECT_END_NID))
sprintf(namestr, "FX: %s %s Switch", pfx, dirstr[dir]);
else
sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]);
return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec));
}
static int add_voicefx(struct hda_codec *codec)
{
struct snd_kcontrol_new knew =
HDA_CODEC_MUTE_MONO(ca0132_voicefx.name,
VOICEFX, 1, 0, HDA_INPUT);
knew.info = ca0132_voicefx_info;
knew.get = ca0132_voicefx_get;
knew.put = ca0132_voicefx_put;
return snd_hda_ctl_add(codec, VOICEFX, snd_ctl_new1(&knew, codec));
}
/* Create the EQ Preset control */
static int add_ca0132_alt_eq_presets(struct hda_codec *codec)
{
struct snd_kcontrol_new knew =
HDA_CODEC_MUTE_MONO(ca0132_alt_eq_enum.name,
EQ_PRESET_ENUM, 1, 0, HDA_OUTPUT);
knew.info = ca0132_alt_eq_preset_info;
knew.get = ca0132_alt_eq_preset_get;
knew.put = ca0132_alt_eq_preset_put;
return snd_hda_ctl_add(codec, EQ_PRESET_ENUM,
snd_ctl_new1(&knew, codec));
}
/*
* Add enumerated control for the three different settings of the smart volume
* output effect. Normal just uses the slider value, and loud and night are
* their own things that ignore that value.
*/
static int ca0132_alt_add_svm_enum(struct hda_codec *codec)
{
struct snd_kcontrol_new knew =
HDA_CODEC_MUTE_MONO("FX: Smart Volume Setting",
SMART_VOLUME_ENUM, 1, 0, HDA_OUTPUT);
knew.info = ca0132_alt_svm_setting_info;
knew.get = ca0132_alt_svm_setting_get;
knew.put = ca0132_alt_svm_setting_put;
return snd_hda_ctl_add(codec, SMART_VOLUME_ENUM,
snd_ctl_new1(&knew, codec));
}
/*
* Create an Output Select enumerated control for codecs with surround
* out capabilities.
*/
static int ca0132_alt_add_output_enum(struct hda_codec *codec)
{
struct snd_kcontrol_new knew =
HDA_CODEC_MUTE_MONO("Output Select",
OUTPUT_SOURCE_ENUM, 1, 0, HDA_OUTPUT);
knew.info = ca0132_alt_output_select_get_info;
knew.get = ca0132_alt_output_select_get;
knew.put = ca0132_alt_output_select_put;
return snd_hda_ctl_add(codec, OUTPUT_SOURCE_ENUM,
snd_ctl_new1(&knew, codec));
}
/*
* Add a control for selecting channel count on speaker output. Setting this
* allows the DSP to do bass redirection and channel upmixing on surround
* configurations.
*/
static int ca0132_alt_add_speaker_channel_cfg_enum(struct hda_codec *codec)
{
struct snd_kcontrol_new knew =
HDA_CODEC_MUTE_MONO("Surround Channel Config",
SPEAKER_CHANNEL_CFG_ENUM, 1, 0, HDA_OUTPUT);
knew.info = ca0132_alt_speaker_channel_cfg_get_info;
knew.get = ca0132_alt_speaker_channel_cfg_get;
knew.put = ca0132_alt_speaker_channel_cfg_put;
return snd_hda_ctl_add(codec, SPEAKER_CHANNEL_CFG_ENUM,
snd_ctl_new1(&knew, codec));
}
/*
* Full range front stereo and rear surround switches. When these are set to
* full range, the lower frequencies from these channels are no longer
* redirected to the LFE channel.
*/
static int ca0132_alt_add_front_full_range_switch(struct hda_codec *codec)
{
struct snd_kcontrol_new knew =
CA0132_CODEC_MUTE_MONO("Full-Range Front Speakers",
SPEAKER_FULL_RANGE_FRONT, 1, HDA_OUTPUT);
return snd_hda_ctl_add(codec, SPEAKER_FULL_RANGE_FRONT,
snd_ctl_new1(&knew, codec));
}
static int ca0132_alt_add_rear_full_range_switch(struct hda_codec *codec)
{
struct snd_kcontrol_new knew =
CA0132_CODEC_MUTE_MONO("Full-Range Rear Speakers",
SPEAKER_FULL_RANGE_REAR, 1, HDA_OUTPUT);
return snd_hda_ctl_add(codec, SPEAKER_FULL_RANGE_REAR,
snd_ctl_new1(&knew, codec));
}
/*
* Bass redirection redirects audio below the crossover frequency to the LFE
* channel on speakers that are set as not being full-range. On configurations
* without an LFE channel, it does nothing. Bass redirection seems to be the
* replacement for X-Bass on configurations with an LFE channel.
*/
static int ca0132_alt_add_bass_redirection_crossover(struct hda_codec *codec)
{
const char *namestr = "Bass Redirection Crossover";
struct snd_kcontrol_new knew =
HDA_CODEC_VOLUME_MONO(namestr, BASS_REDIRECTION_XOVER, 1, 0,
HDA_OUTPUT);
knew.tlv.c = NULL;
knew.info = ca0132_alt_xbass_xover_slider_info;
knew.get = ca0132_alt_xbass_xover_slider_ctl_get;
knew.put = ca0132_alt_xbass_xover_slider_put;
return snd_hda_ctl_add(codec, BASS_REDIRECTION_XOVER,
snd_ctl_new1(&knew, codec));
}
static int ca0132_alt_add_bass_redirection_switch(struct hda_codec *codec)
{
const char *namestr = "Bass Redirection";
struct snd_kcontrol_new knew =
CA0132_CODEC_MUTE_MONO(namestr, BASS_REDIRECTION, 1,
HDA_OUTPUT);
return snd_hda_ctl_add(codec, BASS_REDIRECTION,
snd_ctl_new1(&knew, codec));
}
/*
* Create an Input Source enumerated control for the alternate ca0132 codecs
* because the front microphone has no auto-detect, and Line-in has to be set
* somehow.
*/
static int ca0132_alt_add_input_enum(struct hda_codec *codec)
{
struct snd_kcontrol_new knew =
HDA_CODEC_MUTE_MONO("Input Source",
INPUT_SOURCE_ENUM, 1, 0, HDA_INPUT);
knew.info = ca0132_alt_input_source_info;
knew.get = ca0132_alt_input_source_get;
knew.put = ca0132_alt_input_source_put;
return snd_hda_ctl_add(codec, INPUT_SOURCE_ENUM,
snd_ctl_new1(&knew, codec));
}
/*
* Add mic boost enumerated control. Switches through 0dB to 30dB. This adds
* more control than the original mic boost, which is either full 30dB or off.
*/
static int ca0132_alt_add_mic_boost_enum(struct hda_codec *codec)
{
struct snd_kcontrol_new knew =
HDA_CODEC_MUTE_MONO("Mic Boost Capture Switch",
MIC_BOOST_ENUM, 1, 0, HDA_INPUT);
knew.info = ca0132_alt_mic_boost_info;
knew.get = ca0132_alt_mic_boost_get;
knew.put = ca0132_alt_mic_boost_put;
return snd_hda_ctl_add(codec, MIC_BOOST_ENUM,
snd_ctl_new1(&knew, codec));
}
/*
* Add headphone gain enumerated control for the AE-5. This switches between
* three modes, low, medium, and high. When non-headphone outputs are selected,
* it is automatically set to high. This is the same behavior as Windows.
*/
static int ae5_add_headphone_gain_enum(struct hda_codec *codec)
{
struct snd_kcontrol_new knew =
HDA_CODEC_MUTE_MONO("AE-5: Headphone Gain",
AE5_HEADPHONE_GAIN_ENUM, 1, 0, HDA_OUTPUT);
knew.info = ae5_headphone_gain_info;
knew.get = ae5_headphone_gain_get;
knew.put = ae5_headphone_gain_put;
return snd_hda_ctl_add(codec, AE5_HEADPHONE_GAIN_ENUM,
snd_ctl_new1(&knew, codec));
}
/*
* Add sound filter enumerated control for the AE-5. This adds three different
* settings: Slow Roll Off, Minimum Phase, and Fast Roll Off. From what I've
* read into it, it changes the DAC's interpolation filter.
*/
static int ae5_add_sound_filter_enum(struct hda_codec *codec)
{
struct snd_kcontrol_new knew =
HDA_CODEC_MUTE_MONO("AE-5: Sound Filter",
AE5_SOUND_FILTER_ENUM, 1, 0, HDA_OUTPUT);
knew.info = ae5_sound_filter_info;
knew.get = ae5_sound_filter_get;
knew.put = ae5_sound_filter_put;
return snd_hda_ctl_add(codec, AE5_SOUND_FILTER_ENUM,
snd_ctl_new1(&knew, codec));
}
static int zxr_add_headphone_gain_switch(struct hda_codec *codec)
{
struct snd_kcontrol_new knew =
CA0132_CODEC_MUTE_MONO("ZxR: 600 Ohm Gain",
ZXR_HEADPHONE_GAIN, 1, HDA_OUTPUT);
return snd_hda_ctl_add(codec, ZXR_HEADPHONE_GAIN,
snd_ctl_new1(&knew, codec));
}
/*
* Need to create follower controls for the alternate codecs that have surround
* capabilities.
*/
static const char * const ca0132_alt_follower_pfxs[] = {
"Front", "Surround", "Center", "LFE", NULL,
};
/*
* Also need special channel map, because the default one is incorrect.
* I think this has to do with the pin for rear surround being 0x11,
* and the center/lfe being 0x10. Usually the pin order is the opposite.
*/
static const struct snd_pcm_chmap_elem ca0132_alt_chmaps[] = {
{ .channels = 2,
.map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } },
{ .channels = 4,
.map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR,
SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } },
{ .channels = 6,
.map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR,
SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE,
SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } },
{ }
};
/* Add the correct chmap for streams with 6 channels. */
static void ca0132_alt_add_chmap_ctls(struct hda_codec *codec)
{
int err = 0;
struct hda_pcm *pcm;
list_for_each_entry(pcm, &codec->pcm_list_head, list) {
struct hda_pcm_stream *hinfo =
&pcm->stream[SNDRV_PCM_STREAM_PLAYBACK];
struct snd_pcm_chmap *chmap;
const struct snd_pcm_chmap_elem *elem;
elem = ca0132_alt_chmaps;
if (hinfo->channels_max == 6) {
err = snd_pcm_add_chmap_ctls(pcm->pcm,
SNDRV_PCM_STREAM_PLAYBACK,
elem, hinfo->channels_max, 0, &chmap);
if (err < 0)
codec_dbg(codec, "snd_pcm_add_chmap_ctls failed!");
}
}
}
/*
* When changing Node IDs for Mixer Controls below, make sure to update
* Node IDs in ca0132_config() as well.
*/
static const struct snd_kcontrol_new ca0132_mixer[] = {
CA0132_CODEC_VOL("Master Playback Volume", VNID_SPK, HDA_OUTPUT),
CA0132_CODEC_MUTE("Master Playback Switch", VNID_SPK, HDA_OUTPUT),
CA0132_CODEC_VOL("Capture Volume", VNID_MIC, HDA_INPUT),
CA0132_CODEC_MUTE("Capture Switch", VNID_MIC, HDA_INPUT),
HDA_CODEC_VOLUME("Analog-Mic2 Capture Volume", 0x08, 0, HDA_INPUT),
HDA_CODEC_MUTE("Analog-Mic2 Capture Switch", 0x08, 0, HDA_INPUT),
HDA_CODEC_VOLUME("What U Hear Capture Volume", 0x0a, 0, HDA_INPUT),
HDA_CODEC_MUTE("What U Hear Capture Switch", 0x0a, 0, HDA_INPUT),
CA0132_CODEC_MUTE_MONO("Mic1-Boost (30dB) Capture Switch",
0x12, 1, HDA_INPUT),
CA0132_CODEC_MUTE_MONO("HP/Speaker Playback Switch",
VNID_HP_SEL, 1, HDA_OUTPUT),
CA0132_CODEC_MUTE_MONO("AMic1/DMic Capture Switch",
VNID_AMIC1_SEL, 1, HDA_INPUT),
CA0132_CODEC_MUTE_MONO("HP/Speaker Auto Detect Playback Switch",
VNID_HP_ASEL, 1, HDA_OUTPUT),
CA0132_CODEC_MUTE_MONO("AMic1/DMic Auto Detect Capture Switch",
VNID_AMIC1_ASEL, 1, HDA_INPUT),
{ } /* end */
};
/*
* Desktop specific control mixer. Removes auto-detect for mic, and adds
* surround controls. Also sets both the Front Playback and Capture Volume
* controls to alt so they set the DSP's decibel level.
*/
static const struct snd_kcontrol_new desktop_mixer[] = {
CA0132_ALT_CODEC_VOL("Front Playback Volume", 0x02, HDA_OUTPUT),
CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x04, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x03, 1, 0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x03, 1, 0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x03, 2, 0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x03, 2, 0, HDA_OUTPUT),
CA0132_ALT_CODEC_VOL("Capture Volume", 0x07, HDA_INPUT),
CA0132_CODEC_MUTE("Capture Switch", VNID_MIC, HDA_INPUT),
HDA_CODEC_VOLUME("What U Hear Capture Volume", 0x0a, 0, HDA_INPUT),
HDA_CODEC_MUTE("What U Hear Capture Switch", 0x0a, 0, HDA_INPUT),
CA0132_CODEC_MUTE_MONO("HP/Speaker Auto Detect Playback Switch",
VNID_HP_ASEL, 1, HDA_OUTPUT),
{ } /* end */
};
/*
* Same as the Sound Blaster Z, except doesn't use the alt volume for capture
* because it doesn't set decibel levels for the DSP for capture.
*/
static const struct snd_kcontrol_new r3di_mixer[] = {
CA0132_ALT_CODEC_VOL("Front Playback Volume", 0x02, HDA_OUTPUT),
CA0132_CODEC_MUTE("Front Playback Switch", VNID_SPK, HDA_OUTPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x04, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x03, 1, 0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x03, 1, 0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x03, 2, 0, HDA_OUTPUT),
HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x03, 2, 0, HDA_OUTPUT),
CA0132_CODEC_VOL("Capture Volume", VNID_MIC, HDA_INPUT),
CA0132_CODEC_MUTE("Capture Switch", VNID_MIC, HDA_INPUT),
HDA_CODEC_VOLUME("What U Hear Capture Volume", 0x0a, 0, HDA_INPUT),
HDA_CODEC_MUTE("What U Hear Capture Switch", 0x0a, 0, HDA_INPUT),
CA0132_CODEC_MUTE_MONO("HP/Speaker Auto Detect Playback Switch",
VNID_HP_ASEL, 1, HDA_OUTPUT),
{ } /* end */
};
static int ca0132_build_controls(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
int i, num_fx, num_sliders;
int err = 0;
/* Add Mixer controls */
for (i = 0; i < spec->num_mixers; i++) {
err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
if (err < 0)
return err;
}
/* Setup vmaster with surround followers for desktop ca0132 devices */
if (ca0132_use_alt_functions(spec)) {
snd_hda_set_vmaster_tlv(codec, spec->dacs[0], HDA_OUTPUT,
spec->tlv);
snd_hda_add_vmaster(codec, "Master Playback Volume",
spec->tlv, ca0132_alt_follower_pfxs,
"Playback Volume", 0);
err = __snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL, ca0132_alt_follower_pfxs,
"Playback Switch",
true, 0, &spec->vmaster_mute.sw_kctl);
if (err < 0)
return err;
}
/* Add in and out effects controls.
* VoiceFX, PE and CrystalVoice are added separately.
*/
num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT;
for (i = 0; i < num_fx; i++) {
/* Desktop cards break if Echo Cancellation is used. */
if (ca0132_use_pci_mmio(spec)) {
if (i == (ECHO_CANCELLATION - IN_EFFECT_START_NID +
OUT_EFFECTS_COUNT))
continue;
}
err = add_fx_switch(codec, ca0132_effects[i].nid,
ca0132_effects[i].name,
ca0132_effects[i].direct);
if (err < 0)
return err;
}
/*
* If codec has use_alt_controls set to true, add effect level sliders,
* EQ presets, and Smart Volume presets. Also, change names to add FX
* prefix, and change PlayEnhancement and CrystalVoice to match.
*/
if (ca0132_use_alt_controls(spec)) {
err = ca0132_alt_add_svm_enum(codec);
if (err < 0)
return err;
err = add_ca0132_alt_eq_presets(codec);
if (err < 0)
return err;
err = add_fx_switch(codec, PLAY_ENHANCEMENT,
"Enable OutFX", 0);
if (err < 0)
return err;
err = add_fx_switch(codec, CRYSTAL_VOICE,
"Enable InFX", 1);
if (err < 0)
return err;
num_sliders = OUT_EFFECTS_COUNT - 1;
for (i = 0; i < num_sliders; i++) {
err = ca0132_alt_add_effect_slider(codec,
ca0132_effects[i].nid,
ca0132_effects[i].name,
ca0132_effects[i].direct);
if (err < 0)
return err;
}
err = ca0132_alt_add_effect_slider(codec, XBASS_XOVER,
"X-Bass Crossover", EFX_DIR_OUT);
if (err < 0)
return err;
} else {
err = add_fx_switch(codec, PLAY_ENHANCEMENT,
"PlayEnhancement", 0);
if (err < 0)
return err;
err = add_fx_switch(codec, CRYSTAL_VOICE,
"CrystalVoice", 1);
if (err < 0)
return err;
}
err = add_voicefx(codec);
if (err < 0)
return err;
/*
* If the codec uses alt_functions, you need the enumerated controls
* to select the new outputs and inputs, plus add the new mic boost
* setting control.
*/
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_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_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);
if (err < 0)
return err;
/*
* ZxR only has microphone input, there is no front panel
* header on the card, and aux-in is handled by the DBPro board.
*/
if (ca0132_quirk(spec) != QUIRK_ZXR) {
err = ca0132_alt_add_input_enum(codec);
if (err < 0)
return err;
}
}
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;
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
err = snd_hda_jack_add_kctls(codec, &spec->autocfg);
if (err < 0)
return err;
if (spec->dig_out) {
err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out,
spec->dig_out);
if (err < 0)
return err;
err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);
if (err < 0)
return err;
/* spec->multiout.share_spdif = 1; */
}
if (spec->dig_in) {
err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
if (err < 0)
return err;
}
if (ca0132_use_alt_functions(spec))
ca0132_alt_add_chmap_ctls(codec);
return 0;
}
static int dbpro_build_controls(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
int err = 0;
if (spec->dig_out) {
err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out,
spec->dig_out);
if (err < 0)
return err;
}
if (spec->dig_in) {
err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
if (err < 0)
return err;
}
return 0;
}
/*
* PCM
*/
static const struct hda_pcm_stream ca0132_pcm_analog_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 6,
.ops = {
.prepare = ca0132_playback_pcm_prepare,
.cleanup = ca0132_playback_pcm_cleanup,
.get_delay = ca0132_playback_pcm_delay,
},
};
static const struct hda_pcm_stream ca0132_pcm_analog_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
.ops = {
.prepare = ca0132_capture_pcm_prepare,
.cleanup = ca0132_capture_pcm_cleanup,
.get_delay = ca0132_capture_pcm_delay,
},
};
static const struct hda_pcm_stream ca0132_pcm_digital_playback = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
.ops = {
.open = ca0132_dig_playback_pcm_open,
.close = ca0132_dig_playback_pcm_close,
.prepare = ca0132_dig_playback_pcm_prepare,
.cleanup = ca0132_dig_playback_pcm_cleanup
},
};
static const struct hda_pcm_stream ca0132_pcm_digital_capture = {
.substreams = 1,
.channels_min = 2,
.channels_max = 2,
};
static int ca0132_build_pcms(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
struct hda_pcm *info;
info = snd_hda_codec_pcm_new(codec, "CA0132 Analog");
if (!info)
return -ENOMEM;
if (ca0132_use_alt_functions(spec)) {
info->own_chmap = true;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].chmap
= ca0132_alt_chmaps;
}
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0132_pcm_analog_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0];
info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
spec->multiout.max_channels;
info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture;
info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1;
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0];
/* With the DSP enabled, desktops don't use this ADC. */
if (!ca0132_use_alt_functions(spec)) {
info = snd_hda_codec_pcm_new(codec, "CA0132 Analog Mic-In2");
if (!info)
return -ENOMEM;
info->stream[SNDRV_PCM_STREAM_CAPTURE] =
ca0132_pcm_analog_capture;
info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1;
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[1];
}
info = snd_hda_codec_pcm_new(codec, "CA0132 What U Hear");
if (!info)
return -ENOMEM;
info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture;
info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1;
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[2];
if (!spec->dig_out && !spec->dig_in)
return 0;
info = snd_hda_codec_pcm_new(codec, "CA0132 Digital");
if (!info)
return -ENOMEM;
info->pcm_type = HDA_PCM_TYPE_SPDIF;
if (spec->dig_out) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
ca0132_pcm_digital_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dig_out;
}
if (spec->dig_in) {
info->stream[SNDRV_PCM_STREAM_CAPTURE] =
ca0132_pcm_digital_capture;
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in;
}
return 0;
}
static int dbpro_build_pcms(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
struct hda_pcm *info;
info = snd_hda_codec_pcm_new(codec, "CA0132 Alt Analog");
if (!info)
return -ENOMEM;
info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0132_pcm_analog_capture;
info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = 1;
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0];
if (!spec->dig_out && !spec->dig_in)
return 0;
info = snd_hda_codec_pcm_new(codec, "CA0132 Digital");
if (!info)
return -ENOMEM;
info->pcm_type = HDA_PCM_TYPE_SPDIF;
if (spec->dig_out) {
info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
ca0132_pcm_digital_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dig_out;
}
if (spec->dig_in) {
info->stream[SNDRV_PCM_STREAM_CAPTURE] =
ca0132_pcm_digital_capture;
info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in;
}
return 0;
}
static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
{
if (pin) {
snd_hda_set_pin_ctl(codec, pin, PIN_HP);
if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
snd_hda_codec_write(codec, pin, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_OUT_UNMUTE);
}
if (dac && (get_wcaps(codec, dac) & AC_WCAP_OUT_AMP))
snd_hda_codec_write(codec, dac, 0,
AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO);
}
static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
{
if (pin) {
snd_hda_set_pin_ctl(codec, pin, PIN_VREF80);
if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP)
snd_hda_codec_write(codec, pin, 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_UNMUTE(0));
}
if (adc && (get_wcaps(codec, adc) & AC_WCAP_IN_AMP)) {
snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_UNMUTE(0));
/* init to 0 dB and unmute. */
snd_hda_codec_amp_stereo(codec, adc, HDA_INPUT, 0,
HDA_AMP_VOLMASK, 0x5a);
snd_hda_codec_amp_stereo(codec, adc, HDA_INPUT, 0,
HDA_AMP_MUTE, 0);
}
}
static void refresh_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir)
{
unsigned int caps;
caps = snd_hda_param_read(codec, nid, dir == HDA_OUTPUT ?
AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP);
snd_hda_override_amp_caps(codec, nid, dir, caps);
}
/*
* Switch between Digital built-in mic and analog mic.
*/
static void ca0132_set_dmic(struct hda_codec *codec, int enable)
{
struct ca0132_spec *spec = codec->spec;
unsigned int tmp;
u8 val;
unsigned int oldval;
codec_dbg(codec, "ca0132_set_dmic: enable=%d\n", enable);
oldval = stop_mic1(codec);
ca0132_set_vipsource(codec, 0);
if (enable) {
/* set DMic input as 2-ch */
tmp = FLOAT_TWO;
dspio_set_uint_param(codec, 0x80, 0x00, tmp);
val = spec->dmic_ctl;
val |= 0x80;
snd_hda_codec_write(codec, spec->input_pins[0], 0,
VENDOR_CHIPIO_DMIC_CTL_SET, val);
if (!(spec->dmic_ctl & 0x20))
chipio_set_control_flag(codec, CONTROL_FLAG_DMIC, 1);
} else {
/* set AMic input as mono */
tmp = FLOAT_ONE;
dspio_set_uint_param(codec, 0x80, 0x00, tmp);
val = spec->dmic_ctl;
/* clear bit7 and bit5 to disable dmic */
val &= 0x5f;
snd_hda_codec_write(codec, spec->input_pins[0], 0,
VENDOR_CHIPIO_DMIC_CTL_SET, val);
if (!(spec->dmic_ctl & 0x20))
chipio_set_control_flag(codec, CONTROL_FLAG_DMIC, 0);
}
ca0132_set_vipsource(codec, 1);
resume_mic1(codec, oldval);
}
/*
* Initialization for Digital Mic.
*/
static void ca0132_init_dmic(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
u8 val;
/* Setup Digital Mic here, but don't enable.
* Enable based on jack detect.
*/
/* MCLK uses MPIO1, set to enable.
* Bit 2-0: MPIO select
* Bit 3: set to disable
* Bit 7-4: reserved
*/
val = 0x01;
snd_hda_codec_write(codec, spec->input_pins[0], 0,
VENDOR_CHIPIO_DMIC_MCLK_SET, val);
/* Data1 uses MPIO3. Data2 not use
* Bit 2-0: Data1 MPIO select
* Bit 3: set disable Data1
* Bit 6-4: Data2 MPIO select
* Bit 7: set disable Data2
*/
val = 0x83;
snd_hda_codec_write(codec, spec->input_pins[0], 0,
VENDOR_CHIPIO_DMIC_PIN_SET, val);
/* Use Ch-0 and Ch-1. Rate is 48K, mode 1. Disable DMic first.
* Bit 3-0: Channel mask
* Bit 4: set for 48KHz, clear for 32KHz
* Bit 5: mode
* Bit 6: set to select Data2, clear for Data1
* Bit 7: set to enable DMic, clear for AMic
*/
if (ca0132_quirk(spec) == QUIRK_ALIENWARE_M17XR4)
val = 0x33;
else
val = 0x23;
/* keep a copy of dmic ctl val for enable/disable dmic purpuse */
spec->dmic_ctl = val;
snd_hda_codec_write(codec, spec->input_pins[0], 0,
VENDOR_CHIPIO_DMIC_CTL_SET, val);
}
/*
* Initialization for Analog Mic 2
*/
static void ca0132_init_analog_mic2(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
mutex_lock(&spec->chipio_mutex);
chipio_8051_write_exram_no_mutex(codec, 0x1920, 0x00);
chipio_8051_write_exram_no_mutex(codec, 0x192d, 0x00);
mutex_unlock(&spec->chipio_mutex);
}
static void ca0132_refresh_widget_caps(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
int i;
codec_dbg(codec, "ca0132_refresh_widget_caps.\n");
snd_hda_codec_update_widgets(codec);
for (i = 0; i < spec->multiout.num_dacs; i++)
refresh_amp_caps(codec, spec->dacs[i], HDA_OUTPUT);
for (i = 0; i < spec->num_outputs; i++)
refresh_amp_caps(codec, spec->out_pins[i], HDA_OUTPUT);
for (i = 0; i < spec->num_inputs; i++) {
refresh_amp_caps(codec, spec->adcs[i], HDA_INPUT);
refresh_amp_caps(codec, spec->input_pins[i], HDA_INPUT);
}
}
/* If there is an active channel for some reason, find it and free it. */
static void ca0132_alt_free_active_dma_channels(struct hda_codec *codec)
{
unsigned int i, tmp;
int status;
/* Read active DSPDMAC channel register. */
status = chipio_read(codec, DSPDMAC_CHNLSTART_MODULE_OFFSET, &tmp);
if (status >= 0) {
/* AND against 0xfff to get the active channel bits. */
tmp = tmp & 0xfff;
/* If there are no active channels, nothing to free. */
if (!tmp)
return;
} else {
codec_dbg(codec, "%s: Failed to read active DSP DMA channel register.\n",
__func__);
return;
}
/*
* Check each DSP DMA channel for activity, and if the channel is
* active, free it.
*/
for (i = 0; i < DSPDMAC_DMA_CFG_CHANNEL_COUNT; i++) {
if (dsp_is_dma_active(codec, i)) {
status = dspio_free_dma_chan(codec, i);
if (status < 0)
codec_dbg(codec, "%s: Failed to free active DSP DMA channel %d.\n",
__func__, i);
}
}
}
/*
* In the case of CT_EXTENSIONS_ENABLE being set to 1, and the DSP being in
* use, audio is no longer routed directly to the DAC/ADC from the HDA stream.
* Instead, audio is now routed through the DSP's DMA controllers, which
* the DSP is tasked with setting up itself. Through debugging, it seems the
* cause of most of the no-audio on startup issues were due to improperly
* configured DSP DMA channels.
*
* Normally, the DSP configures these the first time an HDA audio stream is
* started post DSP firmware download. That is why creating a 'dummy' stream
* worked in fixing the audio in some cases. This works most of the time, but
* sometimes if a stream is started/stopped before the DSP can setup the DMA
* configuration registers, it ends up in a broken state. Issues can also
* arise if streams are started in an unusual order, i.e the audio output dma
* channel being sandwiched between the mic1 and mic2 dma channels.
*
* The solution to this is to make sure that the DSP has no DMA channels
* in use post DSP firmware download, and then to manually start each default
* DSP stream that uses the DMA channels. These are 0x0c, the audio output
* stream, 0x03, analog mic 1, and 0x04, analog mic 2.
*/
static void ca0132_alt_start_dsp_audio_streams(struct hda_codec *codec)
{
static const unsigned int dsp_dma_stream_ids[] = { 0x0c, 0x03, 0x04 };
struct ca0132_spec *spec = codec->spec;
unsigned int i, tmp;
/*
* Check if any of the default streams are active, and if they are,
* stop them.
*/
mutex_lock(&spec->chipio_mutex);
for (i = 0; i < ARRAY_SIZE(dsp_dma_stream_ids); i++) {
chipio_get_stream_control(codec, dsp_dma_stream_ids[i], &tmp);
if (tmp) {
chipio_set_stream_control(codec,
dsp_dma_stream_ids[i], 0);
}
}
mutex_unlock(&spec->chipio_mutex);
/*
* If all DSP streams are inactive, there should be no active DSP DMA
* channels. Check and make sure this is the case, and if it isn't,
* free any active channels.
*/
ca0132_alt_free_active_dma_channels(codec);
mutex_lock(&spec->chipio_mutex);
/* Make sure stream 0x0c is six channels. */
chipio_set_stream_channels(codec, 0x0c, 6);
for (i = 0; i < ARRAY_SIZE(dsp_dma_stream_ids); i++) {
chipio_set_stream_control(codec,
dsp_dma_stream_ids[i], 1);
/* Give the DSP some time to setup the DMA channel. */
msleep(75);
}
mutex_unlock(&spec->chipio_mutex);
}
/*
* The region of ChipIO memory from 0x190000-0x1903fc is a sort of 'audio
* router', where each entry represents a 48khz audio channel, with a format
* of an 8-bit destination, an 8-bit source, and an unknown 2-bit number
* value. The 2-bit number value is seemingly 0 if inactive, 1 if active,
* and 3 if it's using Sample Rate Converter ports.
* An example is:
* 0x0001f8c0
* In this case, f8 is the destination, and c0 is the source. The number value
* is 1.
* This region of memory is normally managed internally by the 8051, where
* the region of exram memory from 0x1477-0x1575 has each byte represent an
* entry within the 0x190000 range, and when a range of entries is in use, the
* ending value is overwritten with 0xff.
* 0x1578 in exram is a table of 0x25 entries, corresponding to the ChipIO
* streamID's, where each entry is a starting 0x190000 port offset.
* 0x159d in exram is the same as 0x1578, except it contains the ending port
* offset for the corresponding streamID.
*
* On certain cards, such as the SBZ/ZxR/AE7, these are originally setup by
* the 8051, then manually overwritten to remap the ports to work with the
* new DACs.
*
* Currently known portID's:
* 0x00-0x1f: HDA audio stream input/output ports.
* 0x80-0xbf: Sample rate converter input/outputs. Only valid ports seem to
* have the lower-nibble set to 0x1, 0x2, and 0x9.
* 0xc0-0xdf: DSP DMA input/output ports. Dynamically assigned.
* 0xe0-0xff: DAC/ADC audio input/output ports.
*
* Currently known streamID's:
* 0x03: Mic1 ADC to DSP.
* 0x04: Mic2 ADC to DSP.
* 0x05: HDA node 0x02 audio stream to DSP.
* 0x0f: DSP Mic exit to HDA node 0x07.
* 0x0c: DSP processed audio to DACs.
* 0x14: DAC0, front L/R.
*
* It is possible to route the HDA audio streams directly to the DAC and
* bypass the DSP entirely, with the only downside being that since the DSP
* does volume control, the only volume control you'll get is through PCM on
* the PC side, in the same way volume is handled for optical out. This may be
* useful for debugging.
*/
static void chipio_remap_stream(struct hda_codec *codec,
const struct chipio_stream_remap_data *remap_data)
{
unsigned int i, stream_offset;
/* Get the starting port for the stream to be remapped. */
chipio_8051_read_exram(codec, 0x1578 + remap_data->stream_id,
&stream_offset);
/*
* Check if the stream's port value is 0xff, because the 8051 may not
* have gotten around to setting up the stream yet. Wait until it's
* setup to remap it's ports.
*/
if (stream_offset == 0xff) {
for (i = 0; i < 5; i++) {
msleep(25);
chipio_8051_read_exram(codec, 0x1578 + remap_data->stream_id,
&stream_offset);
if (stream_offset != 0xff)
break;
}
}
if (stream_offset == 0xff) {
codec_info(codec, "%s: Stream 0x%02x ports aren't allocated, remap failed!\n",
__func__, remap_data->stream_id);
return;
}
/* Offset isn't in bytes, its in 32-bit words, so multiply it by 4. */
stream_offset *= 0x04;
stream_offset += 0x190000;
for (i = 0; i < remap_data->count; i++) {
chipio_write_no_mutex(codec,
stream_offset + remap_data->offset[i],
remap_data->value[i]);
}
/* Update stream map configuration. */
chipio_write_no_mutex(codec, 0x19042c, 0x00000001);
}
/*
* 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:
case QUIRK_AE7:
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]);
}
/*
* Initialize mic for non-chromebook ca0132 implementations.
*/
static void ca0132_alt_init_analog_mics(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
unsigned int tmp;
/* Mic 1 Setup */
chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
if (ca0132_quirk(spec) == QUIRK_R3DI) {
chipio_set_conn_rate(codec, 0x0F, SR_96_000);
tmp = FLOAT_ONE;
} else
tmp = FLOAT_THREE;
dspio_set_uint_param(codec, 0x80, 0x00, tmp);
/* Mic 2 setup (not present on desktop cards) */
chipio_set_conn_rate(codec, MEM_CONNID_MICIN2, SR_96_000);
chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2, SR_96_000);
if (ca0132_quirk(spec) == QUIRK_R3DI)
chipio_set_conn_rate(codec, 0x0F, SR_96_000);
tmp = FLOAT_ZERO;
dspio_set_uint_param(codec, 0x80, 0x01, tmp);
}
/*
* Sets the source of stream 0x14 to connpointID 0x48, and the destination
* connpointID to 0x91. If this isn't done, the destination is 0x71, and
* you get no sound. I'm guessing this has to do with the Sound Blaster Z
* having an updated DAC, which changes the destination to that DAC.
*/
static void sbz_connect_streams(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
mutex_lock(&spec->chipio_mutex);
codec_dbg(codec, "Connect Streams entered, mutex locked and loaded.\n");
/* This value is 0x43 for 96khz, and 0x83 for 192khz. */
chipio_write_no_mutex(codec, 0x18a020, 0x00000043);
/* Setup stream 0x14 with it's source and destination points */
chipio_set_stream_source_dest(codec, 0x14, 0x48, 0x91);
chipio_set_conn_rate_no_mutex(codec, 0x48, SR_96_000);
chipio_set_conn_rate_no_mutex(codec, 0x91, SR_96_000);
chipio_set_stream_channels(codec, 0x14, 2);
chipio_set_stream_control(codec, 0x14, 1);
codec_dbg(codec, "Connect Streams exited, mutex released.\n");
mutex_unlock(&spec->chipio_mutex);
}
/*
* Write data through ChipIO to setup proper stream destinations.
* Not sure how it exactly works, but it seems to direct data
* to different destinations. Example is f8 to c0, e0 to c0.
* All I know is, if you don't set these, you get no sound.
*/
static void sbz_chipio_startup_data(struct hda_codec *codec)
{
const struct chipio_stream_remap_data *dsp_out_remap_data;
struct ca0132_spec *spec = codec->spec;
mutex_lock(&spec->chipio_mutex);
codec_dbg(codec, "Startup Data entered, mutex locked and loaded.\n");
/* Remap DAC0's output ports. */
chipio_remap_stream(codec, &stream_remap_data[0]);
/* Remap DSP audio output stream ports. */
switch (ca0132_quirk(spec)) {
case QUIRK_SBZ:
dsp_out_remap_data = &stream_remap_data[1];
break;
case QUIRK_ZXR:
dsp_out_remap_data = &stream_remap_data[2];
break;
default:
dsp_out_remap_data = NULL;
break;
}
if (dsp_out_remap_data)
chipio_remap_stream(codec, dsp_out_remap_data);
codec_dbg(codec, "Startup Data exited, mutex released.\n");
mutex_unlock(&spec->chipio_mutex);
}
static void ca0132_alt_dsp_initial_mic_setup(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
unsigned int tmp;
chipio_set_stream_control(codec, 0x03, 0);
chipio_set_stream_control(codec, 0x04, 0);
chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
tmp = FLOAT_THREE;
dspio_set_uint_param(codec, 0x80, 0x00, tmp);
chipio_set_stream_control(codec, 0x03, 1);
chipio_set_stream_control(codec, 0x04, 1);
switch (ca0132_quirk(spec)) {
case QUIRK_SBZ:
chipio_write(codec, 0x18b098, 0x0000000c);
chipio_write(codec, 0x18b09C, 0x0000000c);
break;
case QUIRK_AE5:
chipio_write(codec, 0x18b098, 0x0000000c);
chipio_write(codec, 0x18b09c, 0x0000004c);
break;
default:
break;
}
}
static void ae5_post_dsp_register_set(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
chipio_8051_write_direct(codec, 0x93, 0x10);
chipio_8051_write_pll_pmu(codec, 0x44, 0xc2);
writeb(0xff, spec->mem_base + 0x304);
writeb(0xff, spec->mem_base + 0x304);
writeb(0xff, spec->mem_base + 0x304);
writeb(0xff, spec->mem_base + 0x304);
writeb(0x00, spec->mem_base + 0x100);
writeb(0xff, spec->mem_base + 0x304);
writeb(0x00, spec->mem_base + 0x100);
writeb(0xff, spec->mem_base + 0x304);
writeb(0x00, spec->mem_base + 0x100);
writeb(0xff, spec->mem_base + 0x304);
writeb(0x00, spec->mem_base + 0x100);
writeb(0xff, spec->mem_base + 0x304);
ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x3f);
ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f);
ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
}
static void ae5_post_dsp_param_setup(struct hda_codec *codec)
{
/*
* Param3 in the 8051's memory is represented by the ascii string 'mch'
* which seems to be 'multichannel'. This is also mentioned in the
* AE-5's registry values in Windows.
*/
chipio_set_control_param(codec, 3, 0);
/*
* I believe ASI is 'audio serial interface' and that it's used to
* change colors on the external LED strip connected to the AE-5.
*/
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);
chipio_8051_write_exram(codec, 0xfa92, 0x22);
}
static void ae5_post_dsp_pll_setup(struct hda_codec *codec)
{
chipio_8051_write_pll_pmu(codec, 0x41, 0xc8);
chipio_8051_write_pll_pmu(codec, 0x45, 0xcc);
chipio_8051_write_pll_pmu(codec, 0x40, 0xcb);
chipio_8051_write_pll_pmu(codec, 0x43, 0xc7);
chipio_8051_write_pll_pmu(codec, 0x51, 0x8d);
}
static void ae5_post_dsp_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);
chipio_set_conn_rate_no_mutex(codec, 0x70, SR_96_000);
chipio_set_stream_source_dest(codec, 0x5, 0x43, 0x0);
chipio_set_stream_source_dest(codec, 0x18, 0x9, 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);
chipio_8051_write_pll_pmu_no_mutex(codec, 0x43, 0xc7);
ca0113_mmio_command_set(codec, 0x48, 0x01, 0x80);
mutex_unlock(&spec->chipio_mutex);
}
static void ae5_post_dsp_startup_data(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
mutex_lock(&spec->chipio_mutex);
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);
ca0113_mmio_command_set(codec, 0x48, 0x0a, 0x05);
chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 7);
ca0113_mmio_command_set(codec, 0x48, 0x0b, 0x12);
ca0113_mmio_command_set(codec, 0x48, 0x04, 0x00);
ca0113_mmio_command_set(codec, 0x48, 0x06, 0x48);
ca0113_mmio_command_set(codec, 0x48, 0x0a, 0x05);
ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00);
ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00);
ca0113_mmio_gpio_set(codec, 0, true);
ca0113_mmio_gpio_set(codec, 1, true);
ca0113_mmio_command_set(codec, 0x48, 0x07, 0x80);
chipio_write_no_mutex(codec, 0x18b03c, 0x00000012);
ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00);
ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00);
mutex_unlock(&spec->chipio_mutex);
}
static void ae7_post_dsp_setup_ports(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
mutex_lock(&spec->chipio_mutex);
/* Seems to share the same port remapping as the SBZ. */
chipio_remap_stream(codec, &stream_remap_data[1]);
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_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)
{
static const unsigned int addr[] = {
0x41, 0x45, 0x40, 0x43, 0x51
};
static const unsigned int data[] = {
0xc8, 0xcc, 0xcb, 0xc7, 0x8d
};
unsigned int i;
for (i = 0; i < ARRAY_SIZE(addr); i++)
chipio_8051_write_pll_pmu_no_mutex(codec, addr[i], data[i]);
}
static void ae7_post_dsp_asi_setup_ports(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
static const unsigned int target[] = {
0x0b, 0x04, 0x06, 0x0a, 0x0c, 0x11, 0x12, 0x13, 0x14
};
static const unsigned int data[] = {
0x12, 0x00, 0x48, 0x05, 0x5f, 0xff, 0xff, 0xff, 0x7f
};
unsigned int i;
mutex_lock(&spec->chipio_mutex);
chipio_8051_write_pll_pmu_no_mutex(codec, 0x43, 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);
chipio_8051_write_pll_pmu(codec, 0x44, 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);
chipio_8051_write_exram(codec, 0xfa92, 0x22);
ae7_post_dsp_pll_setup(codec);
ae7_post_dsp_asi_stream_setup(codec);
chipio_8051_write_pll_pmu(codec, 0x43, 0xc7);
ae7_post_dsp_asi_setup_ports(codec);
}
/*
* Setup default parameters for DSP
*/
static void ca0132_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;
/* 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]);
}
}
/*remove DSP headroom*/
tmp = FLOAT_ZERO;
dspio_set_uint_param(codec, 0x96, 0x3C, tmp);
/*set speaker EQ bypass attenuation*/
dspio_set_uint_param(codec, 0x8f, 0x01, tmp);
/* set AMic1 and AMic2 as mono mic */
tmp = FLOAT_ONE;
dspio_set_uint_param(codec, 0x80, 0x00, tmp);
dspio_set_uint_param(codec, 0x80, 0x01, tmp);
/* set AMic1 as CrystalVoice input */
tmp = FLOAT_ONE;
dspio_set_uint_param(codec, 0x80, 0x05, tmp);
/* set WUH source */
tmp = FLOAT_TWO;
dspio_set_uint_param(codec, 0x31, 0x00, tmp);
}
/*
* Setup default parameters for Recon3D/Recon3Di DSP.
*/
static void r3d_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_init_analog_mics(codec);
ca0132_alt_start_dsp_audio_streams(codec);
/*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);
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++) {
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]);
}
}
}
/*
* Setup default parameters for the Sound Blaster Z DSP. A lot more going on
* than the Chromebook setup.
*/
static void sbz_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_init_analog_mics(codec);
ca0132_alt_start_dsp_audio_streams(codec);
sbz_connect_streams(codec);
sbz_chipio_startup_data(codec);
/*
* Sets internal input loopback to off, used to have a switch to
* enable input loopback, but turned out to be way too buggy.
*/
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);
ca0132_alt_dsp_initial_mic_setup(codec);
/* 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);
}
/*
* Setup default parameters for the Sound BlasterX AE-5 DSP.
*/
static void ae5_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_init_analog_mics(codec);
ca0132_alt_start_dsp_audio_streams(codec);
/* New, unknown SCP req's */
tmp = FLOAT_ZERO;
dspio_set_uint_param(codec, 0x96, 0x29, tmp);
dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
dspio_set_uint_param(codec, 0x80, 0x0d, tmp);
dspio_set_uint_param(codec, 0x80, 0x0e, tmp);
ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f);
ca0113_mmio_gpio_set(codec, 0, false);
ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
/* 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);
ca0132_alt_dsp_initial_mic_setup(codec);
ae5_post_dsp_register_set(codec);
ae5_post_dsp_param_setup(codec);
ae5_post_dsp_pll_setup(codec);
ae5_post_dsp_stream_setup(codec);
ae5_post_dsp_startup_data(codec);
/* 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);
}
/*
* 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_init_analog_mics(codec);
ca0132_alt_start_dsp_audio_streams(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);
}
/*
* Initialization of flags in chip
*/
static void ca0132_init_flags(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
if (ca0132_use_alt_functions(spec)) {
chipio_set_control_flag(codec, CONTROL_FLAG_DSP_96KHZ, 1);
chipio_set_control_flag(codec, CONTROL_FLAG_DAC_96KHZ, 1);
chipio_set_control_flag(codec, CONTROL_FLAG_ADC_B_96KHZ, 1);
chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_96KHZ, 1);
chipio_set_control_flag(codec, CONTROL_FLAG_SRC_RATE_96KHZ, 1);
chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0);
chipio_set_control_flag(codec, CONTROL_FLAG_SPDIF2OUT, 0);
chipio_set_control_flag(codec,
CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0);
chipio_set_control_flag(codec,
CONTROL_FLAG_PORT_A_10KOHM_LOAD, 1);
} else {
chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0);
chipio_set_control_flag(codec,
CONTROL_FLAG_PORT_A_COMMON_MODE, 0);
chipio_set_control_flag(codec,
CONTROL_FLAG_PORT_D_COMMON_MODE, 0);
chipio_set_control_flag(codec,
CONTROL_FLAG_PORT_A_10KOHM_LOAD, 0);
chipio_set_control_flag(codec,
CONTROL_FLAG_PORT_D_10KOHM_LOAD, 0);
chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_HIGH_PASS, 1);
}
}
/*
* Initialization of parameters in chip
*/
static void ca0132_init_params(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
if (ca0132_use_alt_functions(spec)) {
chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000);
chipio_set_conn_rate(codec, 0x0B, SR_48_000);
chipio_set_control_param(codec, CONTROL_PARAM_SPDIF1_SOURCE, 0);
chipio_set_control_param(codec, 0, 0);
chipio_set_control_param(codec, CONTROL_PARAM_VIP_SOURCE, 0);
}
chipio_set_control_param(codec, CONTROL_PARAM_PORTA_160OHM_GAIN, 6);
chipio_set_control_param(codec, CONTROL_PARAM_PORTD_160OHM_GAIN, 6);
}
static void ca0132_set_dsp_msr(struct hda_codec *codec, bool is96k)
{
chipio_set_control_flag(codec, CONTROL_FLAG_DSP_96KHZ, is96k);
chipio_set_control_flag(codec, CONTROL_FLAG_DAC_96KHZ, is96k);
chipio_set_control_flag(codec, CONTROL_FLAG_SRC_RATE_96KHZ, is96k);
chipio_set_control_flag(codec, CONTROL_FLAG_SRC_CLOCK_196MHZ, is96k);
chipio_set_control_flag(codec, CONTROL_FLAG_ADC_B_96KHZ, is96k);
chipio_set_control_flag(codec, CONTROL_FLAG_ADC_C_96KHZ, is96k);
chipio_set_conn_rate(codec, MEM_CONNID_MICIN1, SR_96_000);
chipio_set_conn_rate(codec, MEM_CONNID_MICOUT1, SR_96_000);
chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000);
}
static bool ca0132_download_dsp_images(struct hda_codec *codec)
{
bool dsp_loaded = false;
struct ca0132_spec *spec = codec->spec;
const struct dsp_image_seg *dsp_os_image;
const struct firmware *fw_entry = NULL;
/*
* Alternate firmwares for different variants. The Recon3Di apparently
* can use the default firmware, but I'll leave the option in case
* it needs it again.
*/
switch (ca0132_quirk(spec)) {
case QUIRK_SBZ:
case QUIRK_R3D:
case QUIRK_AE5:
if (request_firmware(&fw_entry, DESKTOP_EFX_FILE,
codec->card->dev) != 0)
codec_dbg(codec, "Desktop firmware not found.");
else
codec_dbg(codec, "Desktop firmware selected.");
break;
case QUIRK_R3DI:
if (request_firmware(&fw_entry, R3DI_EFX_FILE,
codec->card->dev) != 0)
codec_dbg(codec, "Recon3Di alt firmware not detected.");
else
codec_dbg(codec, "Recon3Di firmware selected.");
break;
default:
break;
}
/*
* Use default ctefx.bin if no alt firmware is detected, or if none
* exists for your particular codec.
*/
if (!fw_entry) {
codec_dbg(codec, "Default firmware selected.");
if (request_firmware(&fw_entry, EFX_FILE,
codec->card->dev) != 0)
return false;
}
dsp_os_image = (struct dsp_image_seg *)(fw_entry->data);
if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) {
codec_err(codec, "ca0132 DSP load image failed\n");
goto exit_download;
}
dsp_loaded = dspload_wait_loaded(codec);
exit_download:
release_firmware(fw_entry);
return dsp_loaded;
}
static void ca0132_download_dsp(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
#ifndef CONFIG_SND_HDA_CODEC_CA0132_DSP
return; /* NOP */
#endif
if (spec->dsp_state == DSP_DOWNLOAD_FAILED)
return; /* don't retry failures */
chipio_enable_clocks(codec);
if (spec->dsp_state != DSP_DOWNLOADED) {
spec->dsp_state = DSP_DOWNLOADING;
if (!ca0132_download_dsp_images(codec))
spec->dsp_state = DSP_DOWNLOAD_FAILED;
else
spec->dsp_state = DSP_DOWNLOADED;
}
/* For codecs using alt functions, this is already done earlier */
if (spec->dsp_state == DSP_DOWNLOADED && !ca0132_use_alt_functions(spec))
ca0132_set_dsp_msr(codec, true);
}
static void ca0132_process_dsp_response(struct hda_codec *codec,
struct hda_jack_callback *callback)
{
struct ca0132_spec *spec = codec->spec;
codec_dbg(codec, "ca0132_process_dsp_response\n");
snd_hda_power_up_pm(codec);
if (spec->wait_scp) {
if (dspio_get_response_data(codec) >= 0)
spec->wait_scp = 0;
}
dspio_clear_response_queue(codec);
snd_hda_power_down_pm(codec);
}
static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb)
{
struct ca0132_spec *spec = codec->spec;
struct hda_jack_tbl *tbl;
/* Delay enabling the HP amp, to let the mic-detection
* state machine run.
*/
tbl = snd_hda_jack_tbl_get(codec, cb->nid);
if (tbl)
tbl->block_report = 1;
schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500));
}
static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb)
{
struct ca0132_spec *spec = codec->spec;
if (ca0132_use_alt_functions(spec))
ca0132_alt_select_in(codec);
else
ca0132_select_mic(codec);
}
static void ca0132_setup_unsol(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
snd_hda_jack_detect_enable_callback(codec, spec->unsol_tag_hp, hp_callback);
snd_hda_jack_detect_enable_callback(codec, spec->unsol_tag_amic1,
amic_callback);
snd_hda_jack_detect_enable_callback(codec, UNSOL_TAG_DSP,
ca0132_process_dsp_response);
/* Front headphone jack detection */
if (ca0132_use_alt_functions(spec))
snd_hda_jack_detect_enable_callback(codec,
spec->unsol_tag_front_hp, hp_callback);
}
/*
* Verbs tables.
*/
/* Sends before DSP download. */
static const struct hda_verb ca0132_base_init_verbs[] = {
/*enable ct extension*/
{0x15, VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE, 0x1},
{}
};
/* Send at exit. */
static const struct hda_verb ca0132_base_exit_verbs[] = {
/*set afg to D3*/
{0x01, AC_VERB_SET_POWER_STATE, 0x03},
/*disable ct extension*/
{0x15, VENDOR_CHIPIO_CT_EXTENSIONS_ENABLE, 0},
{}
};
/* Other verbs tables. Sends after DSP download. */
static const struct hda_verb ca0132_init_verbs0[] = {
/* chip init verbs */
{0x15, 0x70D, 0xF0},
{0x15, 0x70E, 0xFE},
{0x15, 0x707, 0x75},
{0x15, 0x707, 0xD3},
{0x15, 0x707, 0x09},
{0x15, 0x707, 0x53},
{0x15, 0x707, 0xD4},
{0x15, 0x707, 0xEF},
{0x15, 0x707, 0x75},
{0x15, 0x707, 0xD3},
{0x15, 0x707, 0x09},
{0x15, 0x707, 0x02},
{0x15, 0x707, 0x37},
{0x15, 0x707, 0x78},
{0x15, 0x53C, 0xCE},
{0x15, 0x575, 0xC9},
{0x15, 0x53D, 0xCE},
{0x15, 0x5B7, 0xC9},
{0x15, 0x70D, 0xE8},
{0x15, 0x70E, 0xFE},
{0x15, 0x707, 0x02},
{0x15, 0x707, 0x68},
{0x15, 0x707, 0x62},
{0x15, 0x53A, 0xCE},
{0x15, 0x546, 0xC9},
{0x15, 0x53B, 0xCE},
{0x15, 0x5E8, 0xC9},
{}
};
/* Extra init verbs for desktop cards. */
static const struct hda_verb ca0132_init_verbs1[] = {
{0x15, 0x70D, 0x20},
{0x15, 0x70E, 0x19},
{0x15, 0x707, 0x00},
{0x15, 0x539, 0xCE},
{0x15, 0x546, 0xC9},
{0x15, 0x70D, 0xB7},
{0x15, 0x70E, 0x09},
{0x15, 0x707, 0x10},
{0x15, 0x70D, 0xAF},
{0x15, 0x70E, 0x09},
{0x15, 0x707, 0x01},
{0x15, 0x707, 0x05},
{0x15, 0x70D, 0x73},
{0x15, 0x70E, 0x09},
{0x15, 0x707, 0x14},
{0x15, 0x6FF, 0xC4},
{}
};
static void ca0132_init_chip(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
int num_fx;
int i;
unsigned int on;
mutex_init(&spec->chipio_mutex);
/*
* The Windows driver always does this upon startup, which seems to
* clear out any previous configuration. This should help issues where
* a boot into Windows prior to a boot into Linux breaks things. Also,
* Windows always sends the reset twice.
*/
if (ca0132_use_alt_functions(spec)) {
chipio_set_control_flag(codec, CONTROL_FLAG_IDLE_ENABLE, 0);
chipio_write_no_mutex(codec, 0x18b0a4, 0x000000c2);
snd_hda_codec_write(codec, codec->core.afg, 0,
AC_VERB_SET_CODEC_RESET, 0);
snd_hda_codec_write(codec, codec->core.afg, 0,
AC_VERB_SET_CODEC_RESET, 0);
}
spec->cur_out_type = SPEAKER_OUT;
if (!ca0132_use_alt_functions(spec))
spec->cur_mic_type = DIGITAL_MIC;
else
spec->cur_mic_type = REAR_MIC;
spec->cur_mic_boost = 0;
for (i = 0; i < VNODES_COUNT; i++) {
spec->vnode_lvol[i] = 0x5a;
spec->vnode_rvol[i] = 0x5a;
spec->vnode_lswitch[i] = 0;
spec->vnode_rswitch[i] = 0;
}
/*
* Default states for effects are in ca0132_effects[].
*/
num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT;
for (i = 0; i < num_fx; i++) {
on = (unsigned int)ca0132_effects[i].reqs[0];
spec->effects_switch[i] = on ? 1 : 0;
}
/*
* Sets defaults for the effect slider controls, only for alternative
* 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];
spec->bass_redirect_xover_freq = 8;
}
spec->voicefx_val = 0;
spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID] = 1;
spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID] = 0;
/*
* The ZxR doesn't have a front panel header, and it's line-in is on
* the daughter board. So, there is no input enum control, and we need
* to make sure that spec->in_enum_val is set properly.
*/
if (ca0132_quirk(spec) == QUIRK_ZXR)
spec->in_enum_val = REAR_MIC;
#ifdef ENABLE_TUNING_CONTROLS
ca0132_init_tuning_defaults(codec);
#endif
}
/*
* Recon3Di exit specific commands.
*/
/* prevents popping noise on shutdown */
static void r3di_gpio_shutdown(struct hda_codec *codec)
{
snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0x00);
}
/*
* Sound Blaster Z exit specific commands.
*/
static void sbz_region2_exit(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
unsigned int i;
for (i = 0; i < 4; i++)
writeb(0x0, spec->mem_base + 0x100);
for (i = 0; i < 8; i++)
writeb(0xb3, spec->mem_base + 0x304);
ca0113_mmio_gpio_set(codec, 0, false);
ca0113_mmio_gpio_set(codec, 1, false);
ca0113_mmio_gpio_set(codec, 4, true);
ca0113_mmio_gpio_set(codec, 5, false);
ca0113_mmio_gpio_set(codec, 7, false);
}
static void sbz_set_pin_ctl_default(struct hda_codec *codec)
{
static const hda_nid_t pins[] = {0x0B, 0x0C, 0x0E, 0x12, 0x13};
unsigned int i;
snd_hda_codec_write(codec, 0x11, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40);
for (i = 0; i < ARRAY_SIZE(pins); i++)
snd_hda_codec_write(codec, pins[i], 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00);
}
static void ca0132_clear_unsolicited(struct hda_codec *codec)
{
static const hda_nid_t pins[] = {0x0B, 0x0E, 0x0F, 0x10, 0x11, 0x12, 0x13};
unsigned int i;
for (i = 0; i < ARRAY_SIZE(pins); i++) {
snd_hda_codec_write(codec, pins[i], 0,
AC_VERB_SET_UNSOLICITED_ENABLE, 0x00);
}
}
/* On shutdown, sends commands in sets of three */
static void sbz_gpio_shutdown_commands(struct hda_codec *codec, int dir,
int mask, int data)
{
if (dir >= 0)
snd_hda_codec_write(codec, 0x01, 0,
AC_VERB_SET_GPIO_DIRECTION, dir);
if (mask >= 0)
snd_hda_codec_write(codec, 0x01, 0,
AC_VERB_SET_GPIO_MASK, mask);
if (data >= 0)
snd_hda_codec_write(codec, 0x01, 0,
AC_VERB_SET_GPIO_DATA, data);
}
static void zxr_dbpro_power_state_shutdown(struct hda_codec *codec)
{
static const hda_nid_t pins[] = {0x05, 0x0c, 0x09, 0x0e, 0x08, 0x11, 0x01};
unsigned int i;
for (i = 0; i < ARRAY_SIZE(pins); i++)
snd_hda_codec_write(codec, pins[i], 0,
AC_VERB_SET_POWER_STATE, 0x03);
}
static void sbz_exit_chip(struct hda_codec *codec)
{
chipio_set_stream_control(codec, 0x03, 0);
chipio_set_stream_control(codec, 0x04, 0);
/* Mess with GPIO */
sbz_gpio_shutdown_commands(codec, 0x07, 0x07, -1);
sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x05);
sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x01);
chipio_set_stream_control(codec, 0x14, 0);
chipio_set_stream_control(codec, 0x0C, 0);
chipio_set_conn_rate(codec, 0x41, SR_192_000);
chipio_set_conn_rate(codec, 0x91, SR_192_000);
chipio_write(codec, 0x18a020, 0x00000083);
sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x03);
sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x07);
sbz_gpio_shutdown_commands(codec, 0x07, 0x07, 0x06);
chipio_set_stream_control(codec, 0x0C, 0);
chipio_set_control_param(codec, 0x0D, 0x24);
ca0132_clear_unsolicited(codec);
sbz_set_pin_ctl_default(codec);
snd_hda_codec_write(codec, 0x0B, 0,
AC_VERB_SET_EAPD_BTLENABLE, 0x00);
sbz_region2_exit(codec);
}
static void r3d_exit_chip(struct hda_codec *codec)
{
ca0132_clear_unsolicited(codec);
snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x5b);
}
static void ae5_exit_chip(struct hda_codec *codec)
{
chipio_set_stream_control(codec, 0x03, 0);
chipio_set_stream_control(codec, 0x04, 0);
ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
ca0113_mmio_command_set(codec, 0x30, 0x30, 0x00);
ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x00);
ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x00);
ca0113_mmio_gpio_set(codec, 0, false);
ca0113_mmio_gpio_set(codec, 1, false);
snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0);
chipio_set_stream_control(codec, 0x18, 0);
chipio_set_stream_control(codec, 0x0c, 0);
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);
chipio_set_stream_control(codec, 0x04, 0);
chipio_set_stream_control(codec, 0x14, 0);
chipio_set_stream_control(codec, 0x0C, 0);
chipio_set_conn_rate(codec, 0x41, SR_192_000);
chipio_set_conn_rate(codec, 0x91, SR_192_000);
chipio_write(codec, 0x18a020, 0x00000083);
snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
ca0132_clear_unsolicited(codec);
sbz_set_pin_ctl_default(codec);
snd_hda_codec_write(codec, 0x0B, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x00);
ca0113_mmio_gpio_set(codec, 5, false);
ca0113_mmio_gpio_set(codec, 2, false);
ca0113_mmio_gpio_set(codec, 3, false);
ca0113_mmio_gpio_set(codec, 0, false);
ca0113_mmio_gpio_set(codec, 4, true);
ca0113_mmio_gpio_set(codec, 0, true);
ca0113_mmio_gpio_set(codec, 5, true);
ca0113_mmio_gpio_set(codec, 2, false);
ca0113_mmio_gpio_set(codec, 3, false);
}
static void ca0132_exit_chip(struct hda_codec *codec)
{
/* put any chip cleanup stuffs here. */
if (dspload_is_loaded(codec))
dsp_reset(codec);
}
/*
* This fixes a problem that was hard to reproduce. Very rarely, I would
* boot up, and there would be no sound, but the DSP indicated it had loaded
* properly. I did a few memory dumps to see if anything was different, and
* there were a few areas of memory uninitialized with a1a2a3a4. This function
* checks if those areas are uninitialized, and if they are, it'll attempt to
* reload the card 3 times. Usually it fixes by the second.
*/
static void sbz_dsp_startup_check(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
unsigned int dsp_data_check[4];
unsigned int cur_address = 0x390;
unsigned int i;
unsigned int failure = 0;
unsigned int reload = 3;
if (spec->startup_check_entered)
return;
spec->startup_check_entered = true;
for (i = 0; i < 4; i++) {
chipio_read(codec, cur_address, &dsp_data_check[i]);
cur_address += 0x4;
}
for (i = 0; i < 4; i++) {
if (dsp_data_check[i] == 0xa1a2a3a4)
failure = 1;
}
codec_dbg(codec, "Startup Check: %d ", failure);
if (failure)
codec_info(codec, "DSP not initialized properly. Attempting to fix.");
/*
* While the failure condition is true, and we haven't reached our
* three reload limit, continue trying to reload the driver and
* fix the issue.
*/
while (failure && (reload != 0)) {
codec_info(codec, "Reloading... Tries left: %d", reload);
sbz_exit_chip(codec);
spec->dsp_state = DSP_DOWNLOAD_INIT;
codec->patch_ops.init(codec);
failure = 0;
for (i = 0; i < 4; i++) {
chipio_read(codec, cur_address, &dsp_data_check[i]);
cur_address += 0x4;
}
for (i = 0; i < 4; i++) {
if (dsp_data_check[i] == 0xa1a2a3a4)
failure = 1;
}
reload--;
}
if (!failure && reload < 3)
codec_info(codec, "DSP fixed.");
if (!failure)
return;
codec_info(codec, "DSP failed to initialize properly. Either try a full shutdown or a suspend to clear the internal memory.");
}
/*
* This is for the extra volume verbs 0x797 (left) and 0x798 (right). These add
* extra precision for decibel values. If you had the dB value in floating point
* you would take the value after the decimal point, multiply by 64, and divide
* by 2. So for 8.59, it's (59 * 64) / 100. Useful if someone wanted to
* implement fixed point or floating point dB volumes. For now, I'll set them
* to 0 just incase a value has lingered from a boot into Windows.
*/
static void ca0132_alt_vol_setup(struct hda_codec *codec)
{
snd_hda_codec_write(codec, 0x02, 0, 0x797, 0x00);
snd_hda_codec_write(codec, 0x02, 0, 0x798, 0x00);
snd_hda_codec_write(codec, 0x03, 0, 0x797, 0x00);
snd_hda_codec_write(codec, 0x03, 0, 0x798, 0x00);
snd_hda_codec_write(codec, 0x04, 0, 0x797, 0x00);
snd_hda_codec_write(codec, 0x04, 0, 0x798, 0x00);
snd_hda_codec_write(codec, 0x07, 0, 0x797, 0x00);
snd_hda_codec_write(codec, 0x07, 0, 0x798, 0x00);
}
/*
* Extra commands that don't really fit anywhere else.
*/
static void sbz_pre_dsp_setup(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
writel(0x00820680, spec->mem_base + 0x01C);
writel(0x00820680, spec->mem_base + 0x01C);
chipio_write(codec, 0x18b0a4, 0x000000c2);
snd_hda_codec_write(codec, 0x11, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, 0x44);
}
static void r3d_pre_dsp_setup(struct hda_codec *codec)
{
chipio_write(codec, 0x18b0a4, 0x000000c2);
chipio_8051_write_exram(codec, 0x1c1e, 0x5b);
snd_hda_codec_write(codec, 0x11, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, 0x44);
}
static void r3di_pre_dsp_setup(struct hda_codec *codec)
{
chipio_write(codec, 0x18b0a4, 0x000000c2);
chipio_8051_write_exram(codec, 0x1c1e, 0x5b);
chipio_8051_write_exram(codec, 0x1920, 0x00);
chipio_8051_write_exram(codec, 0x1921, 0x40);
snd_hda_codec_write(codec, 0x11, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL, 0x04);
}
/*
* The ZxR seems to use alternative DAC's for the surround channels, which
* require PLL PMU setup for the clock rate, I'm guessing. Without setting
* this up, we get no audio out of the surround jacks.
*/
static void zxr_pre_dsp_setup(struct hda_codec *codec)
{
static const unsigned int addr[] = { 0x43, 0x40, 0x41, 0x42, 0x45 };
static const unsigned int data[] = { 0x08, 0x0c, 0x0b, 0x07, 0x0d };
unsigned int i;
chipio_write(codec, 0x189000, 0x0001f100);
msleep(50);
chipio_write(codec, 0x18900c, 0x0001f100);
msleep(50);
/*
* This writes a RET instruction at the entry point of the function at
* 0xfa92 in exram. This function seems to have something to do with
* ASI. Might be some way to prevent the card from reconfiguring the
* ASI stuff itself.
*/
chipio_8051_write_exram(codec, 0xfa92, 0x22);
chipio_8051_write_pll_pmu(codec, 0x51, 0x98);
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x725, 0x82);
chipio_set_control_param(codec, CONTROL_PARAM_ASI, 3);
chipio_write(codec, 0x18902c, 0x00000000);
msleep(50);
chipio_write(codec, 0x18902c, 0x00000003);
msleep(50);
for (i = 0; i < ARRAY_SIZE(addr); i++)
chipio_8051_write_pll_pmu(codec, addr[i], data[i]);
}
/*
* These are sent before the DSP is downloaded. Not sure
* what they do, or if they're necessary. Could possibly
* be removed. Figure they're better to leave in.
*/
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;
addr = ca0113_mmio_init_address_sbz;
for (i = 0; i < 3; i++)
writel(0x00000000, spec->mem_base + addr[i]);
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;
}
for (i = 0; i < 2; i++)
writel(tmp[i], spec->mem_base + addr[cur_addr + i]);
cur_addr += i;
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;
}
for (i = 0; i < count; i++)
writel(data[i], spec->mem_base + addr[cur_addr + i]);
}
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;
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_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)
{
struct ca0132_spec *spec = codec->spec;
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;
default:
break;
}
}
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
* what it does.
*/
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)
chipio_8051_write_pll_pmu(codec, 0x41, 0xc8);
chipio_8051_write_direct(codec, 0x93, 0x10);
chipio_8051_write_pll_pmu(codec, 0x44, 0xc2);
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;
}
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);
if (ca0132_quirk(spec) == QUIRK_AE5)
ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
}
/*
* Extra init functions for alternative ca0132 codecs. Done
* here so they don't clutter up the main ca0132_init function
* anymore than they have to.
*/
static void ca0132_alt_init(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
ca0132_alt_vol_setup(codec);
switch (ca0132_quirk(spec)) {
case QUIRK_SBZ:
codec_dbg(codec, "SBZ alt_init");
ca0132_gpio_init(codec);
sbz_pre_dsp_setup(codec);
snd_hda_sequence_write(codec, spec->chip_init_verbs);
snd_hda_sequence_write(codec, spec->desktop_init_verbs);
break;
case QUIRK_R3DI:
codec_dbg(codec, "R3DI alt_init");
ca0132_gpio_init(codec);
ca0132_gpio_setup(codec);
r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADING);
r3di_pre_dsp_setup(codec);
snd_hda_sequence_write(codec, spec->chip_init_verbs);
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x6FF, 0xC4);
break;
case QUIRK_R3D:
r3d_pre_dsp_setup(codec);
snd_hda_sequence_write(codec, spec->chip_init_verbs);
snd_hda_sequence_write(codec, spec->desktop_init_verbs);
break;
case QUIRK_AE5:
ca0132_gpio_init(codec);
chipio_8051_write_pll_pmu(codec, 0x49, 0x88);
chipio_write(codec, 0x18b030, 0x00000020);
snd_hda_sequence_write(codec, spec->chip_init_verbs);
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);
chipio_8051_write_pll_pmu(codec, 0x49, 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:
chipio_8051_write_pll_pmu(codec, 0x49, 0x88);
snd_hda_sequence_write(codec, spec->chip_init_verbs);
snd_hda_sequence_write(codec, spec->desktop_init_verbs);
zxr_pre_dsp_setup(codec);
break;
default:
break;
}
}
static int ca0132_init(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
int i;
bool dsp_loaded;
/*
* If the DSP is already downloaded, and init has been entered again,
* there's only two reasons for it. One, the codec has awaken from a
* suspended state, and in that case dspload_is_loaded will return
* false, and the init will be ran again. The other reason it gets
* re entered is on startup for some reason it triggers a suspend and
* resume state. In this case, it will check if the DSP is downloaded,
* and not run the init function again. For codecs using alt_functions,
* it will check if the DSP is loaded properly.
*/
if (spec->dsp_state == DSP_DOWNLOADED) {
dsp_loaded = dspload_is_loaded(codec);
if (!dsp_loaded) {
spec->dsp_reload = true;
spec->dsp_state = DSP_DOWNLOAD_INIT;
} else {
if (ca0132_quirk(spec) == QUIRK_SBZ)
sbz_dsp_startup_check(codec);
return 0;
}
}
if (spec->dsp_state != DSP_DOWNLOAD_FAILED)
spec->dsp_state = DSP_DOWNLOAD_INIT;
spec->curr_chip_addx = INVALID_CHIP_ADDRESS;
if (ca0132_use_pci_mmio(spec))
ca0132_mmio_init(codec);
snd_hda_power_up_pm(codec);
if (ca0132_quirk(spec) == QUIRK_AE5 || ca0132_quirk(spec) == QUIRK_AE7)
ae5_register_set(codec);
ca0132_init_params(codec);
ca0132_init_flags(codec);
snd_hda_sequence_write(codec, spec->base_init_verbs);
if (ca0132_use_alt_functions(spec))
ca0132_alt_init(codec);
ca0132_download_dsp(codec);
ca0132_refresh_widget_caps(codec);
switch (ca0132_quirk(spec)) {
case QUIRK_R3DI:
case QUIRK_R3D:
r3d_setup_defaults(codec);
break;
case QUIRK_SBZ:
case QUIRK_ZXR:
sbz_setup_defaults(codec);
break;
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);
ca0132_init_dmic(codec);
break;
}
for (i = 0; i < spec->num_outputs; i++)
init_output(codec, spec->out_pins[i], spec->dacs[0]);
init_output(codec, cfg->dig_out_pins[0], spec->dig_out);
for (i = 0; i < spec->num_inputs; i++)
init_input(codec, spec->input_pins[i], spec->adcs[i]);
init_input(codec, cfg->dig_in_pin, spec->dig_in);
if (!ca0132_use_alt_functions(spec)) {
snd_hda_sequence_write(codec, spec->chip_init_verbs);
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_PARAM_EX_ID_SET, 0x0D);
snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
VENDOR_CHIPIO_PARAM_EX_VALUE_SET, 0x20);
}
if (ca0132_quirk(spec) == QUIRK_SBZ)
ca0132_gpio_setup(codec);
snd_hda_sequence_write(codec, spec->spec_init_verbs);
if (ca0132_use_alt_functions(spec)) {
ca0132_alt_select_out(codec);
ca0132_alt_select_in(codec);
} else {
ca0132_select_out(codec);
ca0132_select_mic(codec);
}
snd_hda_jack_report_sync(codec);
/*
* Re set the PlayEnhancement switch on a resume event, because the
* controls will not be reloaded.
*/
if (spec->dsp_reload) {
spec->dsp_reload = false;
ca0132_pe_switch_set(codec);
}
snd_hda_power_down_pm(codec);
return 0;
}
static int dbpro_init(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
unsigned int i;
init_output(codec, cfg->dig_out_pins[0], spec->dig_out);
init_input(codec, cfg->dig_in_pin, spec->dig_in);
for (i = 0; i < spec->num_inputs; i++)
init_input(codec, spec->input_pins[i], spec->adcs[i]);
return 0;
}
static void ca0132_free(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
cancel_delayed_work_sync(&spec->unsol_hp_work);
snd_hda_power_up(codec);
switch (ca0132_quirk(spec)) {
case QUIRK_SBZ:
sbz_exit_chip(codec);
break;
case QUIRK_ZXR:
zxr_exit_chip(codec);
break;
case QUIRK_R3D:
r3d_exit_chip(codec);
break;
case QUIRK_AE5:
ae5_exit_chip(codec);
break;
case QUIRK_AE7:
ae7_exit_chip(codec);
break;
case QUIRK_R3DI:
r3di_gpio_shutdown(codec);
break;
default:
break;
}
snd_hda_sequence_write(codec, spec->base_exit_verbs);
ca0132_exit_chip(codec);
snd_hda_power_down(codec);
#ifdef CONFIG_PCI
if (spec->mem_base)
pci_iounmap(codec->bus->pci, spec->mem_base);
#endif
kfree(spec->spec_init_verbs);
kfree(codec->spec);
}
static void dbpro_free(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
zxr_dbpro_power_state_shutdown(codec);
kfree(spec->spec_init_verbs);
kfree(codec->spec);
}
#ifdef CONFIG_PM
static int ca0132_suspend(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
cancel_delayed_work_sync(&spec->unsol_hp_work);
return 0;
}
#endif
static const struct hda_codec_ops ca0132_patch_ops = {
.build_controls = ca0132_build_controls,
.build_pcms = ca0132_build_pcms,
.init = ca0132_init,
.free = ca0132_free,
.unsol_event = snd_hda_jack_unsol_event,
#ifdef CONFIG_PM
.suspend = ca0132_suspend,
#endif
};
static const struct hda_codec_ops dbpro_patch_ops = {
.build_controls = dbpro_build_controls,
.build_pcms = dbpro_build_pcms,
.init = dbpro_init,
.free = dbpro_free,
};
static void ca0132_config(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
spec->dacs[0] = 0x2;
spec->dacs[1] = 0x3;
spec->dacs[2] = 0x4;
spec->multiout.dac_nids = spec->dacs;
spec->multiout.num_dacs = 3;
if (!ca0132_use_alt_functions(spec))
spec->multiout.max_channels = 2;
else
spec->multiout.max_channels = 6;
switch (ca0132_quirk(spec)) {
case QUIRK_ALIENWARE:
codec_dbg(codec, "%s: QUIRK_ALIENWARE applied.\n", __func__);
snd_hda_apply_pincfgs(codec, alienware_pincfgs);
break;
case QUIRK_SBZ:
codec_dbg(codec, "%s: QUIRK_SBZ applied.\n", __func__);
snd_hda_apply_pincfgs(codec, sbz_pincfgs);
break;
case QUIRK_ZXR:
codec_dbg(codec, "%s: QUIRK_ZXR applied.\n", __func__);
snd_hda_apply_pincfgs(codec, zxr_pincfgs);
break;
case QUIRK_R3D:
codec_dbg(codec, "%s: QUIRK_R3D applied.\n", __func__);
snd_hda_apply_pincfgs(codec, r3d_pincfgs);
break;
case QUIRK_R3DI:
codec_dbg(codec, "%s: QUIRK_R3DI applied.\n", __func__);
snd_hda_apply_pincfgs(codec, r3di_pincfgs);
break;
case QUIRK_AE5:
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;
}
switch (ca0132_quirk(spec)) {
case QUIRK_ALIENWARE:
spec->num_outputs = 2;
spec->out_pins[0] = 0x0b; /* speaker out */
spec->out_pins[1] = 0x0f;
spec->shared_out_nid = 0x2;
spec->unsol_tag_hp = 0x0f;
spec->adcs[0] = 0x7; /* digital mic / analog mic1 */
spec->adcs[1] = 0x8; /* analog mic2 */
spec->adcs[2] = 0xa; /* what u hear */
spec->num_inputs = 3;
spec->input_pins[0] = 0x12;
spec->input_pins[1] = 0x11;
spec->input_pins[2] = 0x13;
spec->shared_mic_nid = 0x7;
spec->unsol_tag_amic1 = 0x11;
break;
case QUIRK_SBZ:
case QUIRK_R3D:
spec->num_outputs = 2;
spec->out_pins[0] = 0x0B; /* Line out */
spec->out_pins[1] = 0x0F; /* Rear headphone out */
spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/
spec->out_pins[3] = 0x11; /* Rear surround */
spec->shared_out_nid = 0x2;
spec->unsol_tag_hp = spec->out_pins[1];
spec->unsol_tag_front_hp = spec->out_pins[2];
spec->adcs[0] = 0x7; /* Rear Mic / Line-in */
spec->adcs[1] = 0x8; /* Front Mic, but only if no DSP */
spec->adcs[2] = 0xa; /* what u hear */
spec->num_inputs = 2;
spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */
spec->input_pins[1] = 0x13; /* What U Hear */
spec->shared_mic_nid = 0x7;
spec->unsol_tag_amic1 = spec->input_pins[0];
/* SPDIF I/O */
spec->dig_out = 0x05;
spec->multiout.dig_out_nid = spec->dig_out;
spec->dig_in = 0x09;
break;
case QUIRK_ZXR:
spec->num_outputs = 2;
spec->out_pins[0] = 0x0B; /* Line out */
spec->out_pins[1] = 0x0F; /* Rear headphone out */
spec->out_pins[2] = 0x10; /* Center/LFE */
spec->out_pins[3] = 0x11; /* Rear surround */
spec->shared_out_nid = 0x2;
spec->unsol_tag_hp = spec->out_pins[1];
spec->unsol_tag_front_hp = spec->out_pins[2];
spec->adcs[0] = 0x7; /* Rear Mic / Line-in */
spec->adcs[1] = 0x8; /* Not connected, no front mic */
spec->adcs[2] = 0xa; /* what u hear */
spec->num_inputs = 2;
spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */
spec->input_pins[1] = 0x13; /* What U Hear */
spec->shared_mic_nid = 0x7;
spec->unsol_tag_amic1 = spec->input_pins[0];
break;
case QUIRK_ZXR_DBPRO:
spec->adcs[0] = 0x8; /* ZxR DBPro Aux In */
spec->num_inputs = 1;
spec->input_pins[0] = 0x11; /* RCA Line-in */
spec->dig_out = 0x05;
spec->multiout.dig_out_nid = spec->dig_out;
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 */
spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/
spec->out_pins[3] = 0x0F; /* Rear surround */
spec->shared_out_nid = 0x2;
spec->unsol_tag_hp = spec->out_pins[1];
spec->unsol_tag_front_hp = spec->out_pins[2];
spec->adcs[0] = 0x7; /* Rear Mic / Line-in */
spec->adcs[1] = 0x8; /* Front Mic, but only if no DSP */
spec->adcs[2] = 0xa; /* what u hear */
spec->num_inputs = 2;
spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */
spec->input_pins[1] = 0x13; /* What U Hear */
spec->shared_mic_nid = 0x7;
spec->unsol_tag_amic1 = spec->input_pins[0];
/* SPDIF I/O */
spec->dig_out = 0x05;
spec->multiout.dig_out_nid = spec->dig_out;
break;
case QUIRK_R3DI:
spec->num_outputs = 2;
spec->out_pins[0] = 0x0B; /* Line out */
spec->out_pins[1] = 0x0F; /* Rear headphone out */
spec->out_pins[2] = 0x10; /* Front Headphone / Center/LFE*/
spec->out_pins[3] = 0x11; /* Rear surround */
spec->shared_out_nid = 0x2;
spec->unsol_tag_hp = spec->out_pins[1];
spec->unsol_tag_front_hp = spec->out_pins[2];
spec->adcs[0] = 0x07; /* Rear Mic / Line-in */
spec->adcs[1] = 0x08; /* Front Mic, but only if no DSP */
spec->adcs[2] = 0x0a; /* what u hear */
spec->num_inputs = 2;
spec->input_pins[0] = 0x12; /* Rear Mic / Line-in */
spec->input_pins[1] = 0x13; /* What U Hear */
spec->shared_mic_nid = 0x7;
spec->unsol_tag_amic1 = spec->input_pins[0];
/* SPDIF I/O */
spec->dig_out = 0x05;
spec->multiout.dig_out_nid = spec->dig_out;
break;
default:
spec->num_outputs = 2;
spec->out_pins[0] = 0x0b; /* speaker out */
spec->out_pins[1] = 0x10; /* headphone out */
spec->shared_out_nid = 0x2;
spec->unsol_tag_hp = spec->out_pins[1];
spec->adcs[0] = 0x7; /* digital mic / analog mic1 */
spec->adcs[1] = 0x8; /* analog mic2 */
spec->adcs[2] = 0xa; /* what u hear */
spec->num_inputs = 3;
spec->input_pins[0] = 0x12;
spec->input_pins[1] = 0x11;
spec->input_pins[2] = 0x13;
spec->shared_mic_nid = 0x7;
spec->unsol_tag_amic1 = spec->input_pins[0];
/* SPDIF I/O */
spec->dig_out = 0x05;
spec->multiout.dig_out_nid = spec->dig_out;
spec->dig_in = 0x09;
break;
}
}
static int ca0132_prepare_verbs(struct hda_codec *codec)
{
/* Verbs + terminator (an empty element) */
#define NUM_SPEC_VERBS 2
struct ca0132_spec *spec = codec->spec;
spec->chip_init_verbs = ca0132_init_verbs0;
/*
* Since desktop cards use pci_mmio, this can be used to determine
* whether or not to use these verbs instead of a separate bool.
*/
if (ca0132_use_pci_mmio(spec))
spec->desktop_init_verbs = ca0132_init_verbs1;
spec->spec_init_verbs = kcalloc(NUM_SPEC_VERBS,
sizeof(struct hda_verb),
GFP_KERNEL);
if (!spec->spec_init_verbs)
return -ENOMEM;
/* config EAPD */
spec->spec_init_verbs[0].nid = 0x0b;
spec->spec_init_verbs[0].param = 0x78D;
spec->spec_init_verbs[0].verb = 0x00;
/* Previously commented configuration */
/*
spec->spec_init_verbs[2].nid = 0x0b;
spec->spec_init_verbs[2].param = AC_VERB_SET_EAPD_BTLENABLE;
spec->spec_init_verbs[2].verb = 0x02;
spec->spec_init_verbs[3].nid = 0x10;
spec->spec_init_verbs[3].param = 0x78D;
spec->spec_init_verbs[3].verb = 0x02;
spec->spec_init_verbs[4].nid = 0x10;
spec->spec_init_verbs[4].param = AC_VERB_SET_EAPD_BTLENABLE;
spec->spec_init_verbs[4].verb = 0x02;
*/
/* Terminator: spec->spec_init_verbs[NUM_SPEC_VERBS-1] */
return 0;
}
/*
* The Sound Blaster ZxR shares the same PCI subsystem ID as some regular
* Sound Blaster Z cards. However, they have different HDA codec subsystem
* ID's. So, we check for the ZxR's subsystem ID, as well as the DBPro
* daughter boards ID.
*/
static void sbz_detect_quirk(struct hda_codec *codec)
{
struct ca0132_spec *spec = codec->spec;
switch (codec->core.subsystem_id) {
case 0x11020033:
spec->quirk = QUIRK_ZXR;
break;
case 0x1102003f:
spec->quirk = QUIRK_ZXR_DBPRO;
break;
default:
spec->quirk = QUIRK_SBZ;
break;
}
}
static int patch_ca0132(struct hda_codec *codec)
{
struct ca0132_spec *spec;
int err;
const struct snd_pci_quirk *quirk;
codec_dbg(codec, "patch_ca0132\n");
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (!spec)
return -ENOMEM;
codec->spec = spec;
spec->codec = codec;
/* Detect codec quirk */
quirk = snd_pci_quirk_lookup(codec->bus->pci, ca0132_quirks);
if (quirk)
spec->quirk = quirk->value;
else
spec->quirk = QUIRK_NONE;
if (ca0132_quirk(spec) == QUIRK_SBZ)
sbz_detect_quirk(codec);
if (ca0132_quirk(spec) == QUIRK_ZXR_DBPRO)
codec->patch_ops = dbpro_patch_ops;
else
codec->patch_ops = ca0132_patch_ops;
codec->pcm_format_first = 1;
codec->no_sticky_stream = 1;
spec->dsp_state = DSP_DOWNLOAD_INIT;
spec->num_mixers = 1;
/* Set which mixers each quirk uses. */
switch (ca0132_quirk(spec)) {
case QUIRK_SBZ:
spec->mixers[0] = desktop_mixer;
snd_hda_codec_set_name(codec, "Sound Blaster Z");
break;
case QUIRK_ZXR:
spec->mixers[0] = desktop_mixer;
snd_hda_codec_set_name(codec, "Sound Blaster ZxR");
break;
case QUIRK_ZXR_DBPRO:
break;
case QUIRK_R3D:
spec->mixers[0] = desktop_mixer;
snd_hda_codec_set_name(codec, "Recon3D");
break;
case QUIRK_R3DI:
spec->mixers[0] = r3di_mixer;
snd_hda_codec_set_name(codec, "Recon3Di");
break;
case QUIRK_AE5:
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;
}
/* Setup whether or not to use alt functions/controls/pci_mmio */
switch (ca0132_quirk(spec)) {
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;
spec->use_pci_mmio = true;
break;
case QUIRK_R3DI:
spec->use_alt_controls = true;
spec->use_alt_functions = true;
spec->use_pci_mmio = false;
break;
default:
spec->use_alt_controls = false;
spec->use_alt_functions = false;
spec->use_pci_mmio = false;
break;
}
#ifdef CONFIG_PCI
if (spec->use_pci_mmio) {
spec->mem_base = pci_iomap(codec->bus->pci, 2, 0xC20);
if (spec->mem_base == NULL) {
codec_warn(codec, "pci_iomap failed! Setting quirk to QUIRK_NONE.");
spec->quirk = QUIRK_NONE;
}
}
#endif
spec->base_init_verbs = ca0132_base_init_verbs;
spec->base_exit_verbs = ca0132_base_exit_verbs;
INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed);
ca0132_init_chip(codec);
ca0132_config(codec);
err = ca0132_prepare_verbs(codec);
if (err < 0)
goto error;
err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
if (err < 0)
goto error;
ca0132_setup_unsol(codec);
return 0;
error:
ca0132_free(codec);
return err;
}
/*
* patch entries
*/
static const struct hda_device_id snd_hda_id_ca0132[] = {
HDA_CODEC_ENTRY(0x11020011, "CA0132", patch_ca0132),
{} /* terminator */
};
MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_ca0132);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Creative Sound Core3D codec");
static struct hda_codec_driver ca0132_driver = {
.id = snd_hda_id_ca0132,
};
module_hda_codec_driver(ca0132_driver);