diff options
Diffstat (limited to '')
| -rw-r--r-- | sound/pci/hda/patch_ca0132.c | 10123 |
1 files changed, 10123 insertions, 0 deletions
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c new file mode 100644 index 000000000..748a3c409 --- /dev/null +++ b/sound/pci/hda/patch_ca0132.c @@ -0,0 +1,10123 @@ +// 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(0x3842, 0x104b, "EVGA X299 Dark", QUIRK_R3DI), + SND_PCI_QUIRK(0x3842, 0x1055, "EVGA Z390 DARK", 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, ®); + + 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, ®); + + 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) +{ + 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; + + return dsp_allocate_ports(codec, num_chans, rate_multi, port_map); +} + +/* + * 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) + goto found; + + return -EINVAL; +found: + 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); |