|  | /* | 
|  | * wm9712.c  --  ALSA Soc WM9712 codec support | 
|  | * | 
|  | * Copyright 2006-12 Wolfson Microelectronics PLC. | 
|  | * Author: Liam Girdwood <lrg@slimlogic.co.uk> | 
|  | * | 
|  | *  This program is free software; you can redistribute  it and/or modify it | 
|  | *  under  the terms of  the GNU General  Public License as published by the | 
|  | *  Free Software Foundation;  either version 2 of the  License, or (at your | 
|  | *  option) any later version. | 
|  | */ | 
|  |  | 
|  | #include <linux/init.h> | 
|  | #include <linux/slab.h> | 
|  | #include <linux/module.h> | 
|  | #include <linux/kernel.h> | 
|  | #include <linux/device.h> | 
|  | #include <sound/core.h> | 
|  | #include <sound/pcm.h> | 
|  | #include <sound/ac97_codec.h> | 
|  | #include <sound/initval.h> | 
|  | #include <sound/soc.h> | 
|  | #include <sound/tlv.h> | 
|  | #include "wm9712.h" | 
|  |  | 
|  | static unsigned int ac97_read(struct snd_soc_codec *codec, | 
|  | unsigned int reg); | 
|  | static int ac97_write(struct snd_soc_codec *codec, | 
|  | unsigned int reg, unsigned int val); | 
|  |  | 
|  | /* | 
|  | * WM9712 register cache | 
|  | */ | 
|  | static const u16 wm9712_reg[] = { | 
|  | 0x6174, 0x8000, 0x8000, 0x8000, /*  6 */ | 
|  | 0x0f0f, 0xaaa0, 0xc008, 0x6808, /*  e */ | 
|  | 0xe808, 0xaaa0, 0xad00, 0x8000, /* 16 */ | 
|  | 0xe808, 0x3000, 0x8000, 0x0000, /* 1e */ | 
|  | 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ | 
|  | 0x0405, 0x0410, 0xbb80, 0xbb80, /* 2e */ | 
|  | 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ | 
|  | 0x0000, 0x2000, 0x0000, 0x0000, /* 3e */ | 
|  | 0x0000, 0x0000, 0x0000, 0x0000, /* 46 */ | 
|  | 0x0000, 0x0000, 0xf83e, 0xffff, /* 4e */ | 
|  | 0x0000, 0x0000, 0x0000, 0xf83e, /* 56 */ | 
|  | 0x0008, 0x0000, 0x0000, 0x0000, /* 5e */ | 
|  | 0xb032, 0x3e00, 0x0000, 0x0000, /* 66 */ | 
|  | 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ | 
|  | 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ | 
|  | 0x0001, 0x0000, 0x574d, 0x4c12, /* 7e */ | 
|  | 0x0000, 0x0000 /* virtual hp mixers */ | 
|  | }; | 
|  |  | 
|  | /* virtual HP mixers regs */ | 
|  | #define HPL_MIXER	0x80 | 
|  | #define HPR_MIXER	0x82 | 
|  |  | 
|  | static const char *wm9712_alc_select[] = {"None", "Left", "Right", "Stereo"}; | 
|  | static const char *wm9712_alc_mux[] = {"Stereo", "Left", "Right", "None"}; | 
|  | static const char *wm9712_out3_src[] = {"Left", "VREF", "Left + Right", | 
|  | "Mono"}; | 
|  | static const char *wm9712_spk_src[] = {"Speaker Mix", "Headphone Mix"}; | 
|  | static const char *wm9712_rec_adc[] = {"Stereo", "Left", "Right", "Mute"}; | 
|  | static const char *wm9712_base[] = {"Linear Control", "Adaptive Boost"}; | 
|  | static const char *wm9712_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; | 
|  | static const char *wm9712_mic[] = {"Mic 1", "Differential", "Mic 2", | 
|  | "Stereo"}; | 
|  | static const char *wm9712_rec_sel[] = {"Mic", "NC", "NC", "Speaker Mixer", | 
|  | "Line", "Headphone Mixer", "Phone Mixer", "Phone"}; | 
|  | static const char *wm9712_ng_type[] = {"Constant Gain", "Mute"}; | 
|  | static const char *wm9712_diff_sel[] = {"Mic", "Line"}; | 
|  |  | 
|  | static const DECLARE_TLV_DB_SCALE(main_tlv, -3450, 150, 0); | 
|  | static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 2000, 0); | 
|  |  | 
|  | static const struct soc_enum wm9712_enum[] = { | 
|  | SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9712_alc_select), | 
|  | SOC_ENUM_SINGLE(AC97_VIDEO, 12, 4, wm9712_alc_mux), | 
|  | SOC_ENUM_SINGLE(AC97_AUX, 9, 4, wm9712_out3_src), | 
|  | SOC_ENUM_SINGLE(AC97_AUX, 8, 2, wm9712_spk_src), | 
|  | SOC_ENUM_SINGLE(AC97_REC_SEL, 12, 4, wm9712_rec_adc), | 
|  | SOC_ENUM_SINGLE(AC97_MASTER_TONE, 15, 2, wm9712_base), | 
|  | SOC_ENUM_DOUBLE(AC97_REC_GAIN, 14, 6, 2, wm9712_rec_gain), | 
|  | SOC_ENUM_SINGLE(AC97_MIC, 5, 4, wm9712_mic), | 
|  | SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9712_rec_sel), | 
|  | SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9712_rec_sel), | 
|  | SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9712_ng_type), | 
|  | SOC_ENUM_SINGLE(0x5c, 8, 2, wm9712_diff_sel), | 
|  | }; | 
|  |  | 
|  | static const struct snd_kcontrol_new wm9712_snd_ac97_controls[] = { | 
|  | SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1), | 
|  | SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1), | 
|  | SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), | 
|  | SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1), | 
|  | SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1), | 
|  |  | 
|  | SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0), | 
|  | SOC_SINGLE("Speaker Playback Invert Switch", AC97_MASTER, 6, 1, 0), | 
|  | SOC_SINGLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 7, 1, 0), | 
|  | SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_MONO, 7, 1, 0), | 
|  | SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), | 
|  | SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), | 
|  |  | 
|  | SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), | 
|  | SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), | 
|  | SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0), | 
|  | SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0), | 
|  | SOC_ENUM("ALC Function", wm9712_enum[0]), | 
|  | SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0), | 
|  | SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 1), | 
|  | SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0), | 
|  | SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0), | 
|  | SOC_ENUM("ALC NG Type", wm9712_enum[10]), | 
|  | SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 1), | 
|  |  | 
|  | SOC_SINGLE("Mic Headphone  Volume", AC97_VIDEO, 12, 7, 1), | 
|  | SOC_SINGLE("ALC Headphone Volume", AC97_VIDEO, 7, 7, 1), | 
|  |  | 
|  | SOC_SINGLE("Out3 Switch", AC97_AUX, 15, 1, 1), | 
|  | SOC_SINGLE("Out3 ZC Switch", AC97_AUX, 7, 1, 1), | 
|  | SOC_SINGLE("Out3 Volume", AC97_AUX, 0, 31, 1), | 
|  |  | 
|  | SOC_SINGLE("PCBeep Bypass Headphone Volume", AC97_PC_BEEP, 12, 7, 1), | 
|  | SOC_SINGLE("PCBeep Bypass Speaker Volume", AC97_PC_BEEP, 8, 7, 1), | 
|  | SOC_SINGLE("PCBeep Bypass Phone Volume", AC97_PC_BEEP, 4, 7, 1), | 
|  |  | 
|  | SOC_SINGLE("Aux Playback Headphone Volume", AC97_CD, 12, 7, 1), | 
|  | SOC_SINGLE("Aux Playback Speaker Volume", AC97_CD, 8, 7, 1), | 
|  | SOC_SINGLE("Aux Playback Phone Volume", AC97_CD, 4, 7, 1), | 
|  |  | 
|  | SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 1), | 
|  | SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1), | 
|  |  | 
|  | SOC_SINGLE_TLV("Capture Boost Switch", AC97_REC_SEL, 14, 1, 0, boost_tlv), | 
|  | SOC_SINGLE_TLV("Capture to Phone Boost Switch", AC97_REC_SEL, 11, 1, 1, | 
|  | boost_tlv), | 
|  |  | 
|  | SOC_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1), | 
|  | SOC_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1), | 
|  | SOC_SINGLE("3D Playback Volume", AC97_3D_CONTROL, 0, 15, 0), | 
|  |  | 
|  | SOC_ENUM("Bass Control", wm9712_enum[5]), | 
|  | SOC_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1), | 
|  | SOC_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1), | 
|  | SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0), | 
|  | SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 1), | 
|  | SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1), | 
|  |  | 
|  | SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1), | 
|  | SOC_ENUM("Capture Volume Steps", wm9712_enum[6]), | 
|  | SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0), | 
|  | SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0), | 
|  |  | 
|  | SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv), | 
|  | SOC_SINGLE_TLV("Mic 2 Volume", AC97_MIC, 0, 31, 1, main_tlv), | 
|  | SOC_SINGLE_TLV("Mic Boost Volume", AC97_MIC, 7, 1, 0, boost_tlv), | 
|  | }; | 
|  |  | 
|  | /* We have to create a fake left and right HP mixers because | 
|  | * the codec only has a single control that is shared by both channels. | 
|  | * This makes it impossible to determine the audio path. | 
|  | */ | 
|  | static int mixer_event(struct snd_soc_dapm_widget *w, | 
|  | struct snd_kcontrol *k, int event) | 
|  | { | 
|  | u16 l, r, beep, line, phone, mic, pcm, aux; | 
|  |  | 
|  | l = ac97_read(w->codec, HPL_MIXER); | 
|  | r = ac97_read(w->codec, HPR_MIXER); | 
|  | beep = ac97_read(w->codec, AC97_PC_BEEP); | 
|  | mic = ac97_read(w->codec, AC97_VIDEO); | 
|  | phone = ac97_read(w->codec, AC97_PHONE); | 
|  | line = ac97_read(w->codec, AC97_LINE); | 
|  | pcm = ac97_read(w->codec, AC97_PCM); | 
|  | aux = ac97_read(w->codec, AC97_CD); | 
|  |  | 
|  | if (l & 0x1 || r & 0x1) | 
|  | ac97_write(w->codec, AC97_VIDEO, mic & 0x7fff); | 
|  | else | 
|  | ac97_write(w->codec, AC97_VIDEO, mic | 0x8000); | 
|  |  | 
|  | if (l & 0x2 || r & 0x2) | 
|  | ac97_write(w->codec, AC97_PCM, pcm & 0x7fff); | 
|  | else | 
|  | ac97_write(w->codec, AC97_PCM, pcm | 0x8000); | 
|  |  | 
|  | if (l & 0x4 || r & 0x4) | 
|  | ac97_write(w->codec, AC97_LINE, line & 0x7fff); | 
|  | else | 
|  | ac97_write(w->codec, AC97_LINE, line | 0x8000); | 
|  |  | 
|  | if (l & 0x8 || r & 0x8) | 
|  | ac97_write(w->codec, AC97_PHONE, phone & 0x7fff); | 
|  | else | 
|  | ac97_write(w->codec, AC97_PHONE, phone | 0x8000); | 
|  |  | 
|  | if (l & 0x10 || r & 0x10) | 
|  | ac97_write(w->codec, AC97_CD, aux & 0x7fff); | 
|  | else | 
|  | ac97_write(w->codec, AC97_CD, aux | 0x8000); | 
|  |  | 
|  | if (l & 0x20 || r & 0x20) | 
|  | ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff); | 
|  | else | 
|  | ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000); | 
|  |  | 
|  | return 0; | 
|  | } | 
|  |  | 
|  | /* Left Headphone Mixers */ | 
|  | static const struct snd_kcontrol_new wm9712_hpl_mixer_controls[] = { | 
|  | SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPL_MIXER, 5, 1, 0), | 
|  | SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 4, 1, 0), | 
|  | SOC_DAPM_SINGLE("Phone Bypass Switch", HPL_MIXER, 3, 1, 0), | 
|  | SOC_DAPM_SINGLE("Line Bypass Switch", HPL_MIXER, 2, 1, 0), | 
|  | SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 1, 1, 0), | 
|  | SOC_DAPM_SINGLE("Mic Sidetone Switch", HPL_MIXER, 0, 1, 0), | 
|  | }; | 
|  |  | 
|  | /* Right Headphone Mixers */ | 
|  | static const struct snd_kcontrol_new wm9712_hpr_mixer_controls[] = { | 
|  | SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPR_MIXER, 5, 1, 0), | 
|  | SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 4, 1, 0), | 
|  | SOC_DAPM_SINGLE("Phone Bypass Switch", HPR_MIXER, 3, 1, 0), | 
|  | SOC_DAPM_SINGLE("Line Bypass Switch", HPR_MIXER, 2, 1, 0), | 
|  | SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 1, 1, 0), | 
|  | SOC_DAPM_SINGLE("Mic Sidetone Switch", HPR_MIXER, 0, 1, 0), | 
|  | }; | 
|  |  | 
|  | /* Speaker Mixer */ | 
|  | static const struct snd_kcontrol_new wm9712_speaker_mixer_controls[] = { | 
|  | SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 11, 1, 1), | 
|  | SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 11, 1, 1), | 
|  | SOC_DAPM_SINGLE("Phone Bypass Switch", AC97_PHONE, 14, 1, 1), | 
|  | SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 14, 1, 1), | 
|  | SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 14, 1, 1), | 
|  | }; | 
|  |  | 
|  | /* Phone Mixer */ | 
|  | static const struct snd_kcontrol_new wm9712_phone_mixer_controls[] = { | 
|  | SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 7, 1, 1), | 
|  | SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 7, 1, 1), | 
|  | SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 13, 1, 1), | 
|  | SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 13, 1, 1), | 
|  | SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_MIC, 14, 1, 1), | 
|  | SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_MIC, 13, 1, 1), | 
|  | }; | 
|  |  | 
|  | /* ALC headphone mux */ | 
|  | static const struct snd_kcontrol_new wm9712_alc_mux_controls = | 
|  | SOC_DAPM_ENUM("Route", wm9712_enum[1]); | 
|  |  | 
|  | /* out 3 mux */ | 
|  | static const struct snd_kcontrol_new wm9712_out3_mux_controls = | 
|  | SOC_DAPM_ENUM("Route", wm9712_enum[2]); | 
|  |  | 
|  | /* spk mux */ | 
|  | static const struct snd_kcontrol_new wm9712_spk_mux_controls = | 
|  | SOC_DAPM_ENUM("Route", wm9712_enum[3]); | 
|  |  | 
|  | /* Capture to Phone mux */ | 
|  | static const struct snd_kcontrol_new wm9712_capture_phone_mux_controls = | 
|  | SOC_DAPM_ENUM("Route", wm9712_enum[4]); | 
|  |  | 
|  | /* Capture left select */ | 
|  | static const struct snd_kcontrol_new wm9712_capture_selectl_controls = | 
|  | SOC_DAPM_ENUM("Route", wm9712_enum[8]); | 
|  |  | 
|  | /* Capture right select */ | 
|  | static const struct snd_kcontrol_new wm9712_capture_selectr_controls = | 
|  | SOC_DAPM_ENUM("Route", wm9712_enum[9]); | 
|  |  | 
|  | /* Mic select */ | 
|  | static const struct snd_kcontrol_new wm9712_mic_src_controls = | 
|  | SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]); | 
|  |  | 
|  | /* diff select */ | 
|  | static const struct snd_kcontrol_new wm9712_diff_sel_controls = | 
|  | SOC_DAPM_ENUM("Route", wm9712_enum[11]); | 
|  |  | 
|  | static const struct snd_soc_dapm_widget wm9712_dapm_widgets[] = { | 
|  | SND_SOC_DAPM_MUX("ALC Sidetone Mux", SND_SOC_NOPM, 0, 0, | 
|  | &wm9712_alc_mux_controls), | 
|  | SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, | 
|  | &wm9712_out3_mux_controls), | 
|  | SND_SOC_DAPM_MUX("Speaker Mux", SND_SOC_NOPM, 0, 0, | 
|  | &wm9712_spk_mux_controls), | 
|  | SND_SOC_DAPM_MUX("Capture Phone Mux", SND_SOC_NOPM, 0, 0, | 
|  | &wm9712_capture_phone_mux_controls), | 
|  | SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0, | 
|  | &wm9712_capture_selectl_controls), | 
|  | SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0, | 
|  | &wm9712_capture_selectr_controls), | 
|  | SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0, | 
|  | &wm9712_mic_src_controls), | 
|  | SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0, | 
|  | &wm9712_mic_src_controls), | 
|  | SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0, | 
|  | &wm9712_diff_sel_controls), | 
|  | SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), | 
|  | SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_INT_PAGING, 9, 1, | 
|  | &wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls), | 
|  | mixer_event, SND_SOC_DAPM_POST_REG), | 
|  | SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_INT_PAGING, 8, 1, | 
|  | &wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls), | 
|  | mixer_event, SND_SOC_DAPM_POST_REG), | 
|  | SND_SOC_DAPM_MIXER("Phone Mixer", AC97_INT_PAGING, 6, 1, | 
|  | &wm9712_phone_mixer_controls[0], ARRAY_SIZE(wm9712_phone_mixer_controls)), | 
|  | SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_INT_PAGING, 7, 1, | 
|  | &wm9712_speaker_mixer_controls[0], | 
|  | ARRAY_SIZE(wm9712_speaker_mixer_controls)), | 
|  | SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), | 
|  | SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_INT_PAGING, 14, 1), | 
|  | SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_INT_PAGING, 13, 1), | 
|  | SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", SND_SOC_NOPM, 0, 0), | 
|  | SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_INT_PAGING, 12, 1), | 
|  | SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_INT_PAGING, 11, 1), | 
|  | SND_SOC_DAPM_PGA("Headphone PGA", AC97_INT_PAGING, 4, 1, NULL, 0), | 
|  | SND_SOC_DAPM_PGA("Speaker PGA", AC97_INT_PAGING, 3, 1, NULL, 0), | 
|  | SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0), | 
|  | SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0), | 
|  | SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0), | 
|  | SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0), | 
|  | SND_SOC_DAPM_PGA("Differential Mic", SND_SOC_NOPM, 0, 0, NULL, 0), | 
|  | SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1), | 
|  | SND_SOC_DAPM_OUTPUT("MONOOUT"), | 
|  | SND_SOC_DAPM_OUTPUT("HPOUTL"), | 
|  | SND_SOC_DAPM_OUTPUT("HPOUTR"), | 
|  | SND_SOC_DAPM_OUTPUT("LOUT2"), | 
|  | SND_SOC_DAPM_OUTPUT("ROUT2"), | 
|  | SND_SOC_DAPM_OUTPUT("OUT3"), | 
|  | SND_SOC_DAPM_INPUT("LINEINL"), | 
|  | SND_SOC_DAPM_INPUT("LINEINR"), | 
|  | SND_SOC_DAPM_INPUT("PHONE"), | 
|  | SND_SOC_DAPM_INPUT("PCBEEP"), | 
|  | SND_SOC_DAPM_INPUT("MIC1"), | 
|  | SND_SOC_DAPM_INPUT("MIC2"), | 
|  | }; | 
|  |  | 
|  | static const struct snd_soc_dapm_route wm9712_audio_map[] = { | 
|  | /* virtual mixer - mixes left & right channels for spk and mono */ | 
|  | {"AC97 Mixer", NULL, "Left DAC"}, | 
|  | {"AC97 Mixer", NULL, "Right DAC"}, | 
|  |  | 
|  | /* Left HP mixer */ | 
|  | {"Left HP Mixer", "PCBeep Bypass Switch", "PCBEEP"}, | 
|  | {"Left HP Mixer", "Aux Playback Switch",  "Aux DAC"}, | 
|  | {"Left HP Mixer", "Phone Bypass Switch",  "Phone PGA"}, | 
|  | {"Left HP Mixer", "Line Bypass Switch",   "Line PGA"}, | 
|  | {"Left HP Mixer", "PCM Playback Switch",  "Left DAC"}, | 
|  | {"Left HP Mixer", "Mic Sidetone Switch",  "Mic PGA"}, | 
|  | {"Left HP Mixer", NULL,  "ALC Sidetone Mux"}, | 
|  |  | 
|  | /* Right HP mixer */ | 
|  | {"Right HP Mixer", "PCBeep Bypass Switch", "PCBEEP"}, | 
|  | {"Right HP Mixer", "Aux Playback Switch",  "Aux DAC"}, | 
|  | {"Right HP Mixer", "Phone Bypass Switch",  "Phone PGA"}, | 
|  | {"Right HP Mixer", "Line Bypass Switch",   "Line PGA"}, | 
|  | {"Right HP Mixer", "PCM Playback Switch",  "Right DAC"}, | 
|  | {"Right HP Mixer", "Mic Sidetone Switch",  "Mic PGA"}, | 
|  | {"Right HP Mixer", NULL,  "ALC Sidetone Mux"}, | 
|  |  | 
|  | /* speaker mixer */ | 
|  | {"Speaker Mixer", "PCBeep Bypass Switch", "PCBEEP"}, | 
|  | {"Speaker Mixer", "Line Bypass Switch",   "Line PGA"}, | 
|  | {"Speaker Mixer", "PCM Playback Switch",  "AC97 Mixer"}, | 
|  | {"Speaker Mixer", "Phone Bypass Switch",  "Phone PGA"}, | 
|  | {"Speaker Mixer", "Aux Playback Switch",  "Aux DAC"}, | 
|  |  | 
|  | /* Phone mixer */ | 
|  | {"Phone Mixer", "PCBeep Bypass Switch",  "PCBEEP"}, | 
|  | {"Phone Mixer", "Line Bypass Switch",    "Line PGA"}, | 
|  | {"Phone Mixer", "Aux Playback Switch",   "Aux DAC"}, | 
|  | {"Phone Mixer", "PCM Playback Switch",   "AC97 Mixer"}, | 
|  | {"Phone Mixer", "Mic 1 Sidetone Switch", "Mic PGA"}, | 
|  | {"Phone Mixer", "Mic 2 Sidetone Switch", "Mic PGA"}, | 
|  |  | 
|  | /* inputs */ | 
|  | {"Line PGA", NULL, "LINEINL"}, | 
|  | {"Line PGA", NULL, "LINEINR"}, | 
|  | {"Phone PGA", NULL, "PHONE"}, | 
|  | {"Mic PGA", NULL, "MIC1"}, | 
|  | {"Mic PGA", NULL, "MIC2"}, | 
|  |  | 
|  | /* microphones */ | 
|  | {"Differential Mic", NULL, "MIC1"}, | 
|  | {"Differential Mic", NULL, "MIC2"}, | 
|  | {"Left Mic Select Source", "Mic 1", "MIC1"}, | 
|  | {"Left Mic Select Source", "Mic 2", "MIC2"}, | 
|  | {"Left Mic Select Source", "Stereo", "MIC1"}, | 
|  | {"Left Mic Select Source", "Differential", "Differential Mic"}, | 
|  | {"Right Mic Select Source", "Mic 1", "MIC1"}, | 
|  | {"Right Mic Select Source", "Mic 2", "MIC2"}, | 
|  | {"Right Mic Select Source", "Stereo", "MIC2"}, | 
|  | {"Right Mic Select Source", "Differential", "Differential Mic"}, | 
|  |  | 
|  | /* left capture selector */ | 
|  | {"Left Capture Select", "Mic", "MIC1"}, | 
|  | {"Left Capture Select", "Speaker Mixer", "Speaker Mixer"}, | 
|  | {"Left Capture Select", "Line", "LINEINL"}, | 
|  | {"Left Capture Select", "Headphone Mixer", "Left HP Mixer"}, | 
|  | {"Left Capture Select", "Phone Mixer", "Phone Mixer"}, | 
|  | {"Left Capture Select", "Phone", "PHONE"}, | 
|  |  | 
|  | /* right capture selector */ | 
|  | {"Right Capture Select", "Mic", "MIC2"}, | 
|  | {"Right Capture Select", "Speaker Mixer", "Speaker Mixer"}, | 
|  | {"Right Capture Select", "Line", "LINEINR"}, | 
|  | {"Right Capture Select", "Headphone Mixer", "Right HP Mixer"}, | 
|  | {"Right Capture Select", "Phone Mixer", "Phone Mixer"}, | 
|  | {"Right Capture Select", "Phone", "PHONE"}, | 
|  |  | 
|  | /* ALC Sidetone */ | 
|  | {"ALC Sidetone Mux", "Stereo", "Left Capture Select"}, | 
|  | {"ALC Sidetone Mux", "Stereo", "Right Capture Select"}, | 
|  | {"ALC Sidetone Mux", "Left", "Left Capture Select"}, | 
|  | {"ALC Sidetone Mux", "Right", "Right Capture Select"}, | 
|  |  | 
|  | /* ADC's */ | 
|  | {"Left ADC", NULL, "Left Capture Select"}, | 
|  | {"Right ADC", NULL, "Right Capture Select"}, | 
|  |  | 
|  | /* outputs */ | 
|  | {"MONOOUT", NULL, "Phone Mixer"}, | 
|  | {"HPOUTL", NULL, "Headphone PGA"}, | 
|  | {"Headphone PGA", NULL, "Left HP Mixer"}, | 
|  | {"HPOUTR", NULL, "Headphone PGA"}, | 
|  | {"Headphone PGA", NULL, "Right HP Mixer"}, | 
|  |  | 
|  | /* mono mixer */ | 
|  | {"Mono Mixer", NULL, "Left HP Mixer"}, | 
|  | {"Mono Mixer", NULL, "Right HP Mixer"}, | 
|  |  | 
|  | /* Out3 Mux */ | 
|  | {"Out3 Mux", "Left", "Left HP Mixer"}, | 
|  | {"Out3 Mux", "Mono", "Phone Mixer"}, | 
|  | {"Out3 Mux", "Left + Right", "Mono Mixer"}, | 
|  | {"Out 3 PGA", NULL, "Out3 Mux"}, | 
|  | {"OUT3", NULL, "Out 3 PGA"}, | 
|  |  | 
|  | /* speaker Mux */ | 
|  | {"Speaker Mux", "Speaker Mix", "Speaker Mixer"}, | 
|  | {"Speaker Mux", "Headphone Mix", "Mono Mixer"}, | 
|  | {"Speaker PGA", NULL, "Speaker Mux"}, | 
|  | {"LOUT2", NULL, "Speaker PGA"}, | 
|  | {"ROUT2", NULL, "Speaker PGA"}, | 
|  | }; | 
|  |  | 
|  | static unsigned int ac97_read(struct snd_soc_codec *codec, | 
|  | unsigned int reg) | 
|  | { | 
|  | u16 *cache = codec->reg_cache; | 
|  |  | 
|  | if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || | 
|  | reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || | 
|  | reg == AC97_REC_GAIN) | 
|  | return soc_ac97_ops->read(codec->ac97, reg); | 
|  | else { | 
|  | reg = reg >> 1; | 
|  |  | 
|  | if (reg >= (ARRAY_SIZE(wm9712_reg))) | 
|  | return -EIO; | 
|  |  | 
|  | return cache[reg]; | 
|  | } | 
|  | } | 
|  |  | 
|  | static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, | 
|  | unsigned int val) | 
|  | { | 
|  | u16 *cache = codec->reg_cache; | 
|  |  | 
|  | if (reg < 0x7c) | 
|  | soc_ac97_ops->write(codec->ac97, reg, val); | 
|  | reg = reg >> 1; | 
|  | if (reg < (ARRAY_SIZE(wm9712_reg))) | 
|  | cache[reg] = val; | 
|  |  | 
|  | return 0; | 
|  | } | 
|  |  | 
|  | static int ac97_prepare(struct snd_pcm_substream *substream, | 
|  | struct snd_soc_dai *dai) | 
|  | { | 
|  | struct snd_soc_codec *codec = dai->codec; | 
|  | int reg; | 
|  | u16 vra; | 
|  | struct snd_pcm_runtime *runtime = substream->runtime; | 
|  |  | 
|  | vra = ac97_read(codec, AC97_EXTENDED_STATUS); | 
|  | ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); | 
|  |  | 
|  | if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) | 
|  | reg = AC97_PCM_FRONT_DAC_RATE; | 
|  | else | 
|  | reg = AC97_PCM_LR_ADC_RATE; | 
|  |  | 
|  | return ac97_write(codec, reg, runtime->rate); | 
|  | } | 
|  |  | 
|  | static int ac97_aux_prepare(struct snd_pcm_substream *substream, | 
|  | struct snd_soc_dai *dai) | 
|  | { | 
|  | struct snd_soc_codec *codec = dai->codec; | 
|  | u16 vra, xsle; | 
|  | struct snd_pcm_runtime *runtime = substream->runtime; | 
|  |  | 
|  | vra = ac97_read(codec, AC97_EXTENDED_STATUS); | 
|  | ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); | 
|  | xsle = ac97_read(codec, AC97_PCI_SID); | 
|  | ac97_write(codec, AC97_PCI_SID, xsle | 0x8000); | 
|  |  | 
|  | if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) | 
|  | return -ENODEV; | 
|  |  | 
|  | return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); | 
|  | } | 
|  |  | 
|  | #define WM9712_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ | 
|  | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ | 
|  | SNDRV_PCM_RATE_48000) | 
|  |  | 
|  | static const struct snd_soc_dai_ops wm9712_dai_ops_hifi = { | 
|  | .prepare	= ac97_prepare, | 
|  | }; | 
|  |  | 
|  | static const struct snd_soc_dai_ops wm9712_dai_ops_aux = { | 
|  | .prepare	= ac97_aux_prepare, | 
|  | }; | 
|  |  | 
|  | static struct snd_soc_dai_driver wm9712_dai[] = { | 
|  | { | 
|  | .name = "wm9712-hifi", | 
|  | .ac97_control = 1, | 
|  | .playback = { | 
|  | .stream_name = "HiFi Playback", | 
|  | .channels_min = 1, | 
|  | .channels_max = 2, | 
|  | .rates = WM9712_AC97_RATES, | 
|  | .formats = SND_SOC_STD_AC97_FMTS,}, | 
|  | .capture = { | 
|  | .stream_name = "HiFi Capture", | 
|  | .channels_min = 1, | 
|  | .channels_max = 2, | 
|  | .rates = WM9712_AC97_RATES, | 
|  | .formats = SND_SOC_STD_AC97_FMTS,}, | 
|  | .ops = &wm9712_dai_ops_hifi, | 
|  | }, | 
|  | { | 
|  | .name = "wm9712-aux", | 
|  | .playback = { | 
|  | .stream_name = "Aux Playback", | 
|  | .channels_min = 1, | 
|  | .channels_max = 1, | 
|  | .rates = WM9712_AC97_RATES, | 
|  | .formats = SND_SOC_STD_AC97_FMTS,}, | 
|  | .ops = &wm9712_dai_ops_aux, | 
|  | } | 
|  | }; | 
|  |  | 
|  | static int wm9712_set_bias_level(struct snd_soc_codec *codec, | 
|  | enum snd_soc_bias_level level) | 
|  | { | 
|  | switch (level) { | 
|  | case SND_SOC_BIAS_ON: | 
|  | case SND_SOC_BIAS_PREPARE: | 
|  | break; | 
|  | case SND_SOC_BIAS_STANDBY: | 
|  | ac97_write(codec, AC97_POWERDOWN, 0x0000); | 
|  | break; | 
|  | case SND_SOC_BIAS_OFF: | 
|  | /* disable everything including AC link */ | 
|  | ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); | 
|  | ac97_write(codec, AC97_POWERDOWN, 0xffff); | 
|  | break; | 
|  | } | 
|  | codec->dapm.bias_level = level; | 
|  | return 0; | 
|  | } | 
|  |  | 
|  | static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) | 
|  | { | 
|  | if (try_warm && soc_ac97_ops->warm_reset) { | 
|  | soc_ac97_ops->warm_reset(codec->ac97); | 
|  | if (ac97_read(codec, 0) == wm9712_reg[0]) | 
|  | return 1; | 
|  | } | 
|  |  | 
|  | soc_ac97_ops->reset(codec->ac97); | 
|  | if (soc_ac97_ops->warm_reset) | 
|  | soc_ac97_ops->warm_reset(codec->ac97); | 
|  | if (ac97_read(codec, 0) != wm9712_reg[0]) | 
|  | goto err; | 
|  | return 0; | 
|  |  | 
|  | err: | 
|  | printk(KERN_ERR "WM9712 AC97 reset failed\n"); | 
|  | return -EIO; | 
|  | } | 
|  |  | 
|  | static int wm9712_soc_suspend(struct snd_soc_codec *codec) | 
|  | { | 
|  | wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF); | 
|  | return 0; | 
|  | } | 
|  |  | 
|  | static int wm9712_soc_resume(struct snd_soc_codec *codec) | 
|  | { | 
|  | int i, ret; | 
|  | u16 *cache = codec->reg_cache; | 
|  |  | 
|  | ret = wm9712_reset(codec, 1); | 
|  | if (ret < 0) { | 
|  | printk(KERN_ERR "could not reset AC97 codec\n"); | 
|  | return ret; | 
|  | } | 
|  |  | 
|  | wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); | 
|  |  | 
|  | if (ret == 0) { | 
|  | /* Sync reg_cache with the hardware after cold reset */ | 
|  | for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i += 2) { | 
|  | if (i == AC97_INT_PAGING || i == AC97_POWERDOWN || | 
|  | (i > 0x58 && i != 0x5c)) | 
|  | continue; | 
|  | soc_ac97_ops->write(codec->ac97, i, cache[i>>1]); | 
|  | } | 
|  | } | 
|  |  | 
|  | return ret; | 
|  | } | 
|  |  | 
|  | static int wm9712_soc_probe(struct snd_soc_codec *codec) | 
|  | { | 
|  | int ret = 0; | 
|  |  | 
|  | ret = snd_soc_new_ac97_codec(codec, soc_ac97_ops, 0); | 
|  | if (ret < 0) { | 
|  | printk(KERN_ERR "wm9712: failed to register AC97 codec\n"); | 
|  | return ret; | 
|  | } | 
|  |  | 
|  | ret = wm9712_reset(codec, 0); | 
|  | if (ret < 0) { | 
|  | printk(KERN_ERR "Failed to reset WM9712: AC97 link error\n"); | 
|  | goto reset_err; | 
|  | } | 
|  |  | 
|  | /* set alc mux to none */ | 
|  | ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); | 
|  |  | 
|  | wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); | 
|  | snd_soc_add_codec_controls(codec, wm9712_snd_ac97_controls, | 
|  | ARRAY_SIZE(wm9712_snd_ac97_controls)); | 
|  |  | 
|  | return 0; | 
|  |  | 
|  | reset_err: | 
|  | snd_soc_free_ac97_codec(codec); | 
|  | return ret; | 
|  | } | 
|  |  | 
|  | static int wm9712_soc_remove(struct snd_soc_codec *codec) | 
|  | { | 
|  | snd_soc_free_ac97_codec(codec); | 
|  | return 0; | 
|  | } | 
|  |  | 
|  | static struct snd_soc_codec_driver soc_codec_dev_wm9712 = { | 
|  | .probe = 	wm9712_soc_probe, | 
|  | .remove = 	wm9712_soc_remove, | 
|  | .suspend =	wm9712_soc_suspend, | 
|  | .resume =	wm9712_soc_resume, | 
|  | .read = ac97_read, | 
|  | .write = ac97_write, | 
|  | .set_bias_level = wm9712_set_bias_level, | 
|  | .reg_cache_size = ARRAY_SIZE(wm9712_reg), | 
|  | .reg_word_size = sizeof(u16), | 
|  | .reg_cache_step = 2, | 
|  | .reg_cache_default = wm9712_reg, | 
|  | .dapm_widgets = wm9712_dapm_widgets, | 
|  | .num_dapm_widgets = ARRAY_SIZE(wm9712_dapm_widgets), | 
|  | .dapm_routes = wm9712_audio_map, | 
|  | .num_dapm_routes = ARRAY_SIZE(wm9712_audio_map), | 
|  | }; | 
|  |  | 
|  | static int wm9712_probe(struct platform_device *pdev) | 
|  | { | 
|  | return snd_soc_register_codec(&pdev->dev, | 
|  | &soc_codec_dev_wm9712, wm9712_dai, ARRAY_SIZE(wm9712_dai)); | 
|  | } | 
|  |  | 
|  | static int wm9712_remove(struct platform_device *pdev) | 
|  | { | 
|  | snd_soc_unregister_codec(&pdev->dev); | 
|  | return 0; | 
|  | } | 
|  |  | 
|  | static struct platform_driver wm9712_codec_driver = { | 
|  | .driver = { | 
|  | .name = "wm9712-codec", | 
|  | .owner = THIS_MODULE, | 
|  | }, | 
|  |  | 
|  | .probe = wm9712_probe, | 
|  | .remove = wm9712_remove, | 
|  | }; | 
|  |  | 
|  | module_platform_driver(wm9712_codec_driver); | 
|  |  | 
|  | MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver"); | 
|  | MODULE_AUTHOR("Liam Girdwood"); | 
|  | MODULE_LICENSE("GPL"); |