| Author | Tokens | Token Proportion | Commits | Commit Proportion |
|---|---|---|---|---|
| Connor McAdams | 25210 | 54.89% | 86 | 43.43% |
| Ian Minett | 18497 | 40.28% | 18 | 9.09% |
| Takashi Iwai | 1373 | 2.99% | 45 | 22.73% |
| Gabriele Martino | 307 | 0.67% | 2 | 1.01% |
| Dylan Reid | 221 | 0.48% | 2 | 1.01% |
| Chih-Chung Chang | 56 | 0.12% | 1 | 0.51% |
| Alastair Bridgewater | 45 | 0.10% | 6 | 3.03% |
| Takashi Sakamoto | 22 | 0.05% | 5 | 2.53% |
| Michał Mirosław | 21 | 0.05% | 1 | 0.51% |
| Arnd Bergmann | 18 | 0.04% | 1 | 0.51% |
| Hsin-Yu Chao | 18 | 0.04% | 2 | 1.01% |
| Paweł Rekowski | 11 | 0.02% | 1 | 0.51% |
| Geoffrey Allott | 11 | 0.02% | 1 | 0.51% |
| Sven Hahne | 11 | 0.02% | 1 | 0.51% |
| Adam Stylinski | 11 | 0.02% | 1 | 0.51% |
| Simeon Simeonoff | 11 | 0.02% | 1 | 0.51% |
| Xian Wang | 11 | 0.02% | 1 | 0.51% |
| Kuninori Morimoto | 9 | 0.02% | 1 | 0.51% |
| Xi Wang | 9 | 0.02% | 1 | 0.51% |
| Jérémy Lefaure | 8 | 0.02% | 1 | 0.51% |
| Julia Lawall | 7 | 0.02% | 2 | 1.01% |
| Colin Ian King | 6 | 0.01% | 3 | 1.52% |
| Pierre-Louis Bossart | 5 | 0.01% | 2 | 1.01% |
| Jaroslav Kysela | 4 | 0.01% | 1 | 0.51% |
| Kees Cook | 3 | 0.01% | 1 | 0.51% |
| Paul Gortmaker | 3 | 0.01% | 1 | 0.51% |
| Mark Brown | 3 | 0.01% | 1 | 0.51% |
| Matthias Kaehlcke | 3 | 0.01% | 1 | 0.51% |
| Thomas Gleixner | 2 | 0.00% | 1 | 0.51% |
| Vitaliy Kulikov | 2 | 0.00% | 1 | 0.51% |
| Cezary Rojewski | 2 | 0.00% | 1 | 0.51% |
| Fengguang Wu | 2 | 0.00% | 1 | 0.51% |
| Alexey V. Vissarionov | 1 | 0.00% | 1 | 0.51% |
| Sachin Kamat | 1 | 0.00% | 1 | 0.51% |
| Gustavo A. R. Silva | 1 | 0.00% | 1 | 0.51% |
| ye xingchen | 1 | 0.00% | 1 | 0.51% |
| Total | 45926 | 198 |
// 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_stream_format(channels, 32, sample_rate); 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); } static int ca0132_suspend(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; cancel_delayed_work_sync(&spec->unsol_hp_work); return 0; } 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, .suspend = ca0132_suspend, }; 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);
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