init android origin source code

1 files changed, 700 insertions, 0 deletions

diff --git a/ANDROID_3.4.5/sound/soc/codecs/wm9712.c b/ANDROID_3.4.5/sound/soc/codecs/wm9712.c
new file mode 100644
index 00000000..b342ae50
--- /dev/null
+++ b/ANDROID_3.4.5/sound/soc/codecs/wm9712.c
@@ -0,0 +1,700 @@
+/*
+ * wm9712.c  --  ALSA Soc WM9712 codec support
+ *
+ * Copyright 2006 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("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0),
+SOC_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1),
+
+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 ADC 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, 1),
+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("Route", 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("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_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"},
+
+	/* 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_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec =rtd->codec;
+	int reg;
+	u16 vra;
+
+	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_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+	u16 vra, xsle;
+
+	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 __devinit 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 __devexit 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 = __devexit_p(wm9712_remove),
+};
+
+module_platform_driver(wm9712_codec_driver);
+
+MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");