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sound/soc/codecs/wm9713.c 41.2 KB
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
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  /*
   * wm9713.c  --  ALSA Soc WM9713 codec support
   *
   * Copyright 2006 Wolfson Microelectronics PLC.
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   * Author: Liam Girdwood <lrg@slimlogic.co.uk>
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   *
   *  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.
   *
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   *  Features:-
   *
   *   o Support for AC97 Codec, Voice DAC and Aux DAC
   *   o Support for DAPM
   */
  
  #include <linux/init.h>
  #include <linux/module.h>
  #include <linux/device.h>
  #include <sound/core.h>
  #include <sound/pcm.h>
  #include <sound/ac97_codec.h>
  #include <sound/initval.h>
  #include <sound/pcm_params.h>
  #include <sound/soc.h>
  #include <sound/soc-dapm.h>
  
  #include "wm9713.h"
  
  #define WM9713_VERSION "0.15"
  
  struct wm9713_priv {
  	u32 pll_in; /* PLL input frequency */
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  };
  
  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);
  
  /*
   * WM9713 register cache
   * Reg 0x3c bit 15 is used by touch driver.
   */
  static const u16 wm9713_reg[] = {
  	0x6174, 0x8080, 0x8080, 0x8080,
  	0xc880, 0xe808, 0xe808, 0x0808,
  	0x00da, 0x8000, 0xd600, 0xaaa0,
  	0xaaa0, 0xaaa0, 0x0000, 0x0000,
  	0x0f0f, 0x0040, 0x0000, 0x7f00,
  	0x0405, 0x0410, 0xbb80, 0xbb80,
  	0x0000, 0xbb80, 0x0000, 0x4523,
  	0x0000, 0x2000, 0x7eff, 0xffff,
  	0x0000, 0x0000, 0x0080, 0x0000,
  	0x0000, 0x0000, 0xfffe, 0xffff,
  	0x0000, 0x0000, 0x0000, 0xfffe,
  	0x4000, 0x0000, 0x0000, 0x0000,
  	0xb032, 0x3e00, 0x0000, 0x0000,
  	0x0000, 0x0000, 0x0000, 0x0000,
  	0x0000, 0x0000, 0x0000, 0x0006,
  	0x0001, 0x0000, 0x574d, 0x4c13,
  	0x0000, 0x0000, 0x0000
  };
  
  /* virtual HP mixers regs */
  #define HPL_MIXER	0x80
  #define HPR_MIXER	0x82
  #define MICB_MUX	0x82
  
  static const char *wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"};
  static const char *wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"};
  static const char *wm9713_rec_src[] =
  	{"Mic 1", "Mic 2", "Line", "Mono In", "Headphone", "Speaker",
  	"Mono Out", "Zh"};
  static const char *wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"};
  static const char *wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"};
  static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv",
  	"Mono Vmid", "Inv Vmid"};
  static const char *wm9713_spk_pga[] =
  	{"Vmid", "Zh", "Headphone", "Speaker", "Inv", "Headphone Vmid",
  	"Speaker Vmid", "Inv Vmid"};
  static const char *wm9713_hp_pga[] = {"Vmid", "Zh", "Headphone",
  	"Headphone Vmid"};
  static const char *wm9713_out3_pga[] = {"Vmid", "Zh", "Inv 1", "Inv 1 Vmid"};
  static const char *wm9713_out4_pga[] = {"Vmid", "Zh", "Inv 2", "Inv 2 Vmid"};
  static const char *wm9713_dac_inv[] =
  	{"Off", "Mono", "Speaker", "Left Headphone", "Right Headphone",
  	"Headphone Mono", "NC", "Vmid"};
  static const char *wm9713_bass[] = {"Linear Control", "Adaptive Boost"};
  static const char *wm9713_ng_type[] = {"Constant Gain", "Mute"};
  static const char *wm9713_mic_select[] = {"Mic 1", "Mic 2 A", "Mic 2 B"};
  static const char *wm9713_micb_select[] = {"MPB", "MPA"};
  
  static const struct soc_enum wm9713_enum[] = {
  SOC_ENUM_SINGLE(AC97_LINE, 3, 4, wm9713_mic_mixer), /* record mic mixer 0 */
  SOC_ENUM_SINGLE(AC97_VIDEO, 14, 4, wm9713_rec_mux), /* record mux hp 1 */
  SOC_ENUM_SINGLE(AC97_VIDEO, 9, 4, wm9713_rec_mux),  /* record mux mono 2 */
  SOC_ENUM_SINGLE(AC97_VIDEO, 3, 8, wm9713_rec_src),  /* record mux left 3 */
  SOC_ENUM_SINGLE(AC97_VIDEO, 0, 8, wm9713_rec_src),  /* record mux right 4*/
  SOC_ENUM_DOUBLE(AC97_CD, 14, 6, 2, wm9713_rec_gain), /* record step size 5 */
  SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9713_alc_select), /* alc source select 6*/
  SOC_ENUM_SINGLE(AC97_REC_GAIN, 14, 4, wm9713_mono_pga), /* mono input select 7 */
  SOC_ENUM_SINGLE(AC97_REC_GAIN, 11, 8, wm9713_spk_pga), /* speaker left input select 8 */
  SOC_ENUM_SINGLE(AC97_REC_GAIN, 8, 8, wm9713_spk_pga), /* speaker right input select 9 */
  SOC_ENUM_SINGLE(AC97_REC_GAIN, 6, 3, wm9713_hp_pga), /* headphone left input 10 */
  SOC_ENUM_SINGLE(AC97_REC_GAIN, 4, 3, wm9713_hp_pga), /* headphone right input 11 */
  SOC_ENUM_SINGLE(AC97_REC_GAIN, 2, 4, wm9713_out3_pga), /* out 3 source 12 */
  SOC_ENUM_SINGLE(AC97_REC_GAIN, 0, 4, wm9713_out4_pga), /* out 4 source 13 */
  SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 13, 8, wm9713_dac_inv), /* dac invert 1 14 */
  SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 10, 8, wm9713_dac_inv), /* dac invert 2 15 */
  SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_bass), /* bass control 16 */
  SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type), /* noise gate type 17 */
  SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 */
  SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */
  };
  
  static const struct snd_kcontrol_new wm9713_snd_ac97_controls[] = {
  SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
  SOC_DOUBLE("Speaker Playback Switch", AC97_MASTER, 15, 7, 1, 1),
  SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
  SOC_DOUBLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 7, 1, 1),
  SOC_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1),
  SOC_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1),
  SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
  SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
  
  SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0),
  SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1),
  
  SOC_SINGLE("Capture Switch", AC97_CD, 15, 1, 1),
  SOC_ENUM("Capture Volume Steps", wm9713_enum[5]),
  SOC_DOUBLE("Capture Volume", AC97_CD, 8, 0, 31, 0),
  SOC_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0),
  
  SOC_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1),
  SOC_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0),
  SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0),
  
  SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
  SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
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  SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0),
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  SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
  SOC_ENUM("ALC Function", wm9713_enum[6]),
  SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
  SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 0),
  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", wm9713_enum[17]),
  SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 0),
  
  SOC_DOUBLE("Speaker Playback ZC Switch", AC97_MASTER, 14, 6, 1, 0),
  SOC_DOUBLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0),
  
  SOC_SINGLE("Out4 Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
  SOC_SINGLE("Out4 Playback ZC Switch", AC97_MASTER_MONO, 14, 1, 0),
  SOC_SINGLE("Out4 Playback Volume", AC97_MASTER_MONO, 8, 63, 1),
  
  SOC_SINGLE("Out3 Playback Switch", AC97_MASTER_MONO, 7, 1, 1),
  SOC_SINGLE("Out3 Playback ZC Switch", AC97_MASTER_MONO, 6, 1, 0),
  SOC_SINGLE("Out3 Playback Volume", AC97_MASTER_MONO, 0, 63, 1),
  
  SOC_SINGLE("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1),
  SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1),
  SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0),
  SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1),
  
  SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1),
  SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1),
  SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1),
  
  SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1),
  SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1),
  SOC_SINGLE("Voice Playback Mono Volume", AC97_PCM, 4, 7, 1),
  
  SOC_SINGLE("Aux Playback Headphone Volume", AC97_REC_SEL, 12, 7, 1),
  SOC_SINGLE("Aux Playback Master Volume", AC97_REC_SEL, 8, 7, 1),
  SOC_SINGLE("Aux Playback Mono Volume", AC97_REC_SEL, 4, 7, 1),
  
  SOC_ENUM("Bass Control", wm9713_enum[16]),
  SOC_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1),
  SOC_SINGLE("Tone Cut-off Switch", AC97_GENERAL_PURPOSE, 4, 1, 1),
  SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_GENERAL_PURPOSE, 6, 1, 0),
  SOC_SINGLE("Bass Volume", AC97_GENERAL_PURPOSE, 8, 15, 1),
  SOC_SINGLE("Tone Volume", AC97_GENERAL_PURPOSE, 0, 15, 1),
  
  SOC_SINGLE("3D Upper Cut-off Switch", AC97_REC_GAIN_MIC, 5, 1, 0),
  SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0),
  SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
  };
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  static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w,
  				 struct snd_kcontrol *kcontrol, int event)
  {
  	struct snd_soc_codec *codec = w->codec;
  	u16 status, rate;
  
  	BUG_ON(event != SND_SOC_DAPM_PRE_PMD);
  
  	/* Gracefully shut down the voice interface. */
  	status = ac97_read(codec, AC97_EXTENDED_MID) | 0x1000;
  	rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF;
  	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200);
  	schedule_timeout_interruptible(msecs_to_jiffies(1));
  	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00);
  	ac97_write(codec, AC97_EXTENDED_MID, status);
  
  	return 0;
  }
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  /* 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 using the current
   * register map, thus we add a new (virtual) register to help determine the
   * audio route within the device.
   */
  static int mixer_event(struct snd_soc_dapm_widget *w,
  	struct snd_kcontrol *kcontrol, int event)
  {
  	u16 l, r, beep, tone, phone, rec, pcm, aux;
  
  	l = ac97_read(w->codec, HPL_MIXER);
  	r = ac97_read(w->codec, HPR_MIXER);
  	beep = ac97_read(w->codec, AC97_PC_BEEP);
  	tone = ac97_read(w->codec, AC97_MASTER_TONE);
  	phone = ac97_read(w->codec, AC97_PHONE);
  	rec = ac97_read(w->codec, AC97_REC_SEL);
  	pcm = ac97_read(w->codec, AC97_PCM);
  	aux = ac97_read(w->codec, AC97_AUX);
  
  	if (event & SND_SOC_DAPM_PRE_REG)
  		return 0;
  	if ((l & 0x1) || (r & 0x1))
  		ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff);
  	else
  		ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000);
  
  	if ((l & 0x2) || (r & 0x2))
  		ac97_write(w->codec, AC97_MASTER_TONE, tone & 0x7fff);
  	else
  		ac97_write(w->codec, AC97_MASTER_TONE, tone | 0x8000);
  
  	if ((l & 0x4) || (r & 0x4))
  		ac97_write(w->codec, AC97_PHONE, phone & 0x7fff);
  	else
  		ac97_write(w->codec, AC97_PHONE, phone | 0x8000);
  
  	if ((l & 0x8) || (r & 0x8))
  		ac97_write(w->codec, AC97_REC_SEL, rec & 0x7fff);
  	else
  		ac97_write(w->codec, AC97_REC_SEL, rec | 0x8000);
  
  	if ((l & 0x10) || (r & 0x10))
  		ac97_write(w->codec, AC97_PCM, pcm & 0x7fff);
  	else
  		ac97_write(w->codec, AC97_PCM, pcm | 0x8000);
  
  	if ((l & 0x20) || (r & 0x20))
  		ac97_write(w->codec, AC97_AUX, aux & 0x7fff);
  	else
  		ac97_write(w->codec, AC97_AUX, aux | 0x8000);
  
  	return 0;
  }
  
  /* Left Headphone Mixers */
  static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = {
  SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0),
  SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0),
  SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0),
  SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0),
  SOC_DAPM_SINGLE("MonoIn Playback Switch", HPL_MIXER, 1, 1, 0),
  SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0),
  };
  
  /* Right Headphone Mixers */
  static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = {
  SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0),
  SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0),
  SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0),
  SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0),
  SOC_DAPM_SINGLE("MonoIn Playback Switch", HPR_MIXER, 1, 1, 0),
  SOC_DAPM_SINGLE("Bypass Playback Switch", HPR_MIXER, 0, 1, 0),
  };
  
  /* headphone capture mux */
  static const struct snd_kcontrol_new wm9713_hp_rec_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[1]);
  
  /* headphone mic mux */
  static const struct snd_kcontrol_new wm9713_hp_mic_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[0]);
  
  /* Speaker Mixer */
  static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = {
  SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1),
  SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1),
  SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1),
  SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1),
  SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 14, 1, 1),
  SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1),
  };
  
  /* Mono Mixer */
  static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = {
  SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1),
  SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1),
  SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1),
  SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1),
  SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 13, 1, 1),
  SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 13, 1, 1),
  SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_LINE, 7, 1, 1),
  SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_LINE, 6, 1, 1),
  };
  
  /* mono mic mux */
  static const struct snd_kcontrol_new wm9713_mono_mic_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[2]);
  
  /* mono output mux */
  static const struct snd_kcontrol_new wm9713_mono_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[7]);
  
  /* speaker left output mux */
  static const struct snd_kcontrol_new wm9713_hp_spkl_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[8]);
  
  /* speaker right output mux */
  static const struct snd_kcontrol_new wm9713_hp_spkr_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[9]);
  
  /* headphone left output mux */
  static const struct snd_kcontrol_new wm9713_hpl_out_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[10]);
  
  /* headphone right output mux */
  static const struct snd_kcontrol_new wm9713_hpr_out_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[11]);
  
  /* Out3 mux */
  static const struct snd_kcontrol_new wm9713_out3_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[12]);
  
  /* Out4 mux */
  static const struct snd_kcontrol_new wm9713_out4_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[13]);
  
  /* DAC inv mux 1 */
  static const struct snd_kcontrol_new wm9713_dac_inv1_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[14]);
  
  /* DAC inv mux 2 */
  static const struct snd_kcontrol_new wm9713_dac_inv2_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[15]);
  
  /* Capture source left */
  static const struct snd_kcontrol_new wm9713_rec_srcl_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[3]);
  
  /* Capture source right */
  static const struct snd_kcontrol_new wm9713_rec_srcr_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[4]);
  
  /* mic source */
  static const struct snd_kcontrol_new wm9713_mic_sel_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[18]);
  
  /* mic source B virtual control */
  static const struct snd_kcontrol_new wm9713_micb_sel_mux_controls =
  SOC_DAPM_ENUM("Route", wm9713_enum[19]);
  
  static const struct snd_soc_dapm_widget wm9713_dapm_widgets[] = {
  SND_SOC_DAPM_MUX("Capture Headphone Mux", SND_SOC_NOPM, 0, 0,
  	&wm9713_hp_rec_mux_controls),
  SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0,
  	&wm9713_hp_mic_mux_controls),
  SND_SOC_DAPM_MUX("Capture Mono Mux", SND_SOC_NOPM, 0, 0,
  	&wm9713_mono_mic_mux_controls),
  SND_SOC_DAPM_MUX("Mono Out Mux", SND_SOC_NOPM, 0, 0,
  	&wm9713_mono_mux_controls),
  SND_SOC_DAPM_MUX("Left Speaker Out Mux", SND_SOC_NOPM, 0, 0,
  	&wm9713_hp_spkl_mux_controls),
  SND_SOC_DAPM_MUX("Right Speaker Out Mux", SND_SOC_NOPM, 0, 0,
  	&wm9713_hp_spkr_mux_controls),
  SND_SOC_DAPM_MUX("Left Headphone Out Mux", SND_SOC_NOPM, 0, 0,
  	&wm9713_hpl_out_mux_controls),
  SND_SOC_DAPM_MUX("Right Headphone Out Mux", SND_SOC_NOPM, 0, 0,
  	&wm9713_hpr_out_mux_controls),
  SND_SOC_DAPM_MUX("Out 3 Mux", SND_SOC_NOPM, 0, 0,
  	&wm9713_out3_mux_controls),
  SND_SOC_DAPM_MUX("Out 4 Mux", SND_SOC_NOPM, 0, 0,
  	&wm9713_out4_mux_controls),
  SND_SOC_DAPM_MUX("DAC Inv Mux 1", SND_SOC_NOPM, 0, 0,
  	&wm9713_dac_inv1_mux_controls),
  SND_SOC_DAPM_MUX("DAC Inv Mux 2", SND_SOC_NOPM, 0, 0,
  	&wm9713_dac_inv2_mux_controls),
  SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0,
  	&wm9713_rec_srcl_mux_controls),
  SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0,
  	&wm9713_rec_srcr_mux_controls),
  SND_SOC_DAPM_MUX("Mic A Source", SND_SOC_NOPM, 0, 0,
  	&wm9713_mic_sel_mux_controls),
  SND_SOC_DAPM_MUX("Mic B Source", SND_SOC_NOPM, 0, 0,
  	&wm9713_micb_sel_mux_controls),
  SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_EXTENDED_MID, 3, 1,
  	&wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls),
  	mixer_event, SND_SOC_DAPM_POST_REG),
  SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_EXTENDED_MID, 2, 1,
  	&wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls),
  	mixer_event, SND_SOC_DAPM_POST_REG),
  SND_SOC_DAPM_MIXER("Mono Mixer", AC97_EXTENDED_MID, 0, 1,
  	&wm9713_mono_mixer_controls[0], ARRAY_SIZE(wm9713_mono_mixer_controls)),
  SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_EXTENDED_MID, 1, 1,
  	&wm9713_speaker_mixer_controls[0],
  	ARRAY_SIZE(wm9713_speaker_mixer_controls)),
  SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_EXTENDED_MID, 7, 1),
  SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_EXTENDED_MID, 6, 1),
  SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
  SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
  SND_SOC_DAPM_MIXER("Line Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
  SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
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  SND_SOC_DAPM_DAC_E("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1,
  		   wm9713_voice_shutdown, SND_SOC_DAPM_PRE_PMD),
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  SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1),
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  SND_SOC_DAPM_PGA("Left ADC", AC97_EXTENDED_MID, 5, 1, NULL, 0),
  SND_SOC_DAPM_PGA("Right ADC", AC97_EXTENDED_MID, 4, 1, NULL, 0),
  SND_SOC_DAPM_ADC("Left HiFi ADC", "Left HiFi Capture", SND_SOC_NOPM, 0, 0),
  SND_SOC_DAPM_ADC("Right HiFi ADC", "Right HiFi Capture", SND_SOC_NOPM, 0, 0),
  SND_SOC_DAPM_ADC("Left Voice ADC", "Left Voice Capture", SND_SOC_NOPM, 0, 0),
  SND_SOC_DAPM_ADC("Right Voice ADC", "Right Voice Capture", SND_SOC_NOPM, 0, 0),
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  SND_SOC_DAPM_PGA("Left Headphone", AC97_EXTENDED_MSTATUS, 10, 1, NULL, 0),
  SND_SOC_DAPM_PGA("Right Headphone", AC97_EXTENDED_MSTATUS, 9, 1, NULL, 0),
  SND_SOC_DAPM_PGA("Left Speaker", AC97_EXTENDED_MSTATUS, 8, 1, NULL, 0),
  SND_SOC_DAPM_PGA("Right Speaker", AC97_EXTENDED_MSTATUS, 7, 1, NULL, 0),
  SND_SOC_DAPM_PGA("Out 3", AC97_EXTENDED_MSTATUS, 11, 1, NULL, 0),
  SND_SOC_DAPM_PGA("Out 4", AC97_EXTENDED_MSTATUS, 12, 1, NULL, 0),
  SND_SOC_DAPM_PGA("Mono Out", AC97_EXTENDED_MSTATUS, 13, 1, NULL, 0),
  SND_SOC_DAPM_PGA("Left Line In", AC97_EXTENDED_MSTATUS, 6, 1, NULL, 0),
  SND_SOC_DAPM_PGA("Right Line In", AC97_EXTENDED_MSTATUS, 5, 1, NULL, 0),
  SND_SOC_DAPM_PGA("Mono In", AC97_EXTENDED_MSTATUS, 4, 1, NULL, 0),
  SND_SOC_DAPM_PGA("Mic A PGA", AC97_EXTENDED_MSTATUS, 3, 1, NULL, 0),
  SND_SOC_DAPM_PGA("Mic B PGA", AC97_EXTENDED_MSTATUS, 2, 1, NULL, 0),
  SND_SOC_DAPM_PGA("Mic A Pre Amp", AC97_EXTENDED_MSTATUS, 1, 1, NULL, 0),
  SND_SOC_DAPM_PGA("Mic B Pre Amp", AC97_EXTENDED_MSTATUS, 0, 1, NULL, 0),
  SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_EXTENDED_MSTATUS, 14, 1),
  SND_SOC_DAPM_OUTPUT("MONO"),
  SND_SOC_DAPM_OUTPUT("HPL"),
  SND_SOC_DAPM_OUTPUT("HPR"),
  SND_SOC_DAPM_OUTPUT("SPKL"),
  SND_SOC_DAPM_OUTPUT("SPKR"),
  SND_SOC_DAPM_OUTPUT("OUT3"),
  SND_SOC_DAPM_OUTPUT("OUT4"),
  SND_SOC_DAPM_INPUT("LINEL"),
  SND_SOC_DAPM_INPUT("LINER"),
  SND_SOC_DAPM_INPUT("MONOIN"),
  SND_SOC_DAPM_INPUT("PCBEEP"),
  SND_SOC_DAPM_INPUT("MIC1"),
  SND_SOC_DAPM_INPUT("MIC2A"),
  SND_SOC_DAPM_INPUT("MIC2B"),
  SND_SOC_DAPM_VMID("VMID"),
  };
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  static const struct snd_soc_dapm_route audio_map[] = {
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  	/* left HP mixer */
  	{"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
  	{"Left HP Mixer", "Voice Playback Switch",   "Voice DAC"},
  	{"Left HP Mixer", "Aux Playback Switch",     "Aux DAC"},
  	{"Left HP Mixer", "Bypass Playback Switch",  "Left Line In"},
  	{"Left HP Mixer", "PCM Playback Switch",     "Left DAC"},
  	{"Left HP Mixer", "MonoIn Playback Switch",  "Mono In"},
  	{"Left HP Mixer", NULL,  "Capture Headphone Mux"},
  
  	/* right HP mixer */
  	{"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"},
  	{"Right HP Mixer", "Voice Playback Switch",   "Voice DAC"},
  	{"Right HP Mixer", "Aux Playback Switch",     "Aux DAC"},
  	{"Right HP Mixer", "Bypass Playback Switch",  "Right Line In"},
  	{"Right HP Mixer", "PCM Playback Switch",     "Right DAC"},
  	{"Right HP Mixer", "MonoIn Playback Switch",  "Mono In"},
  	{"Right HP Mixer", NULL,  "Capture Headphone Mux"},
  
  	/* virtual mixer - mixes left & right channels for spk and mono */
  	{"AC97 Mixer", NULL, "Left DAC"},
  	{"AC97 Mixer", NULL, "Right DAC"},
  	{"Line Mixer", NULL, "Right Line In"},
  	{"Line Mixer", NULL, "Left Line In"},
  	{"HP Mixer", NULL, "Left HP Mixer"},
  	{"HP Mixer", NULL, "Right HP Mixer"},
  	{"Capture Mixer", NULL, "Left Capture Source"},
  	{"Capture Mixer", NULL, "Right Capture Source"},
  
  	/* speaker mixer */
  	{"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"},
  	{"Speaker Mixer", "Voice Playback Switch",   "Voice DAC"},
  	{"Speaker Mixer", "Aux Playback Switch",     "Aux DAC"},
  	{"Speaker Mixer", "Bypass Playback Switch",  "Line Mixer"},
  	{"Speaker Mixer", "PCM Playback Switch",     "AC97 Mixer"},
  	{"Speaker Mixer", "MonoIn Playback Switch",  "Mono In"},
  
  	/* mono mixer */
  	{"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"},
  	{"Mono Mixer", "Voice Playback Switch",   "Voice DAC"},
  	{"Mono Mixer", "Aux Playback Switch",     "Aux DAC"},
  	{"Mono Mixer", "Bypass Playback Switch",  "Line Mixer"},
  	{"Mono Mixer", "PCM Playback Switch",     "AC97 Mixer"},
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  	{"Mono Mixer", "Mic 1 Sidetone Switch", "Mic A PGA"},
  	{"Mono Mixer", "Mic 2 Sidetone Switch", "Mic B PGA"},
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  	{"Mono Mixer", NULL,  "Capture Mono Mux"},
  
  	/* DAC inv mux 1 */
  	{"DAC Inv Mux 1", "Mono", "Mono Mixer"},
  	{"DAC Inv Mux 1", "Speaker", "Speaker Mixer"},
  	{"DAC Inv Mux 1", "Left Headphone", "Left HP Mixer"},
  	{"DAC Inv Mux 1", "Right Headphone", "Right HP Mixer"},
  	{"DAC Inv Mux 1", "Headphone Mono", "HP Mixer"},
  
  	/* DAC inv mux 2 */
  	{"DAC Inv Mux 2", "Mono", "Mono Mixer"},
  	{"DAC Inv Mux 2", "Speaker", "Speaker Mixer"},
  	{"DAC Inv Mux 2", "Left Headphone", "Left HP Mixer"},
  	{"DAC Inv Mux 2", "Right Headphone", "Right HP Mixer"},
  	{"DAC Inv Mux 2", "Headphone Mono", "HP Mixer"},
  
  	/* headphone left mux */
  	{"Left Headphone Out Mux", "Headphone", "Left HP Mixer"},
  
  	/* headphone right mux */
  	{"Right Headphone Out Mux", "Headphone", "Right HP Mixer"},
  
  	/* speaker left mux */
  	{"Left Speaker Out Mux", "Headphone", "Left HP Mixer"},
  	{"Left Speaker Out Mux", "Speaker", "Speaker Mixer"},
  	{"Left Speaker Out Mux", "Inv", "DAC Inv Mux 1"},
  
  	/* speaker right mux */
  	{"Right Speaker Out Mux", "Headphone", "Right HP Mixer"},
  	{"Right Speaker Out Mux", "Speaker", "Speaker Mixer"},
  	{"Right Speaker Out Mux", "Inv", "DAC Inv Mux 2"},
  
  	/* mono mux */
  	{"Mono Out Mux", "Mono", "Mono Mixer"},
  	{"Mono Out Mux", "Inv", "DAC Inv Mux 1"},
  
  	/* out 3 mux */
  	{"Out 3 Mux", "Inv 1", "DAC Inv Mux 1"},
  
  	/* out 4 mux */
  	{"Out 4 Mux", "Inv 2", "DAC Inv Mux 2"},
  
  	/* output pga */
  	{"HPL", NULL, "Left Headphone"},
  	{"Left Headphone", NULL, "Left Headphone Out Mux"},
  	{"HPR", NULL, "Right Headphone"},
  	{"Right Headphone", NULL, "Right Headphone Out Mux"},
  	{"OUT3", NULL, "Out 3"},
  	{"Out 3", NULL, "Out 3 Mux"},
  	{"OUT4", NULL, "Out 4"},
  	{"Out 4", NULL, "Out 4 Mux"},
  	{"SPKL", NULL, "Left Speaker"},
  	{"Left Speaker", NULL, "Left Speaker Out Mux"},
  	{"SPKR", NULL, "Right Speaker"},
  	{"Right Speaker", NULL, "Right Speaker Out Mux"},
  	{"MONO", NULL, "Mono Out"},
  	{"Mono Out", NULL, "Mono Out Mux"},
  
  	/* input pga */
  	{"Left Line In", NULL, "LINEL"},
  	{"Right Line In", NULL, "LINER"},
  	{"Mono In", NULL, "MONOIN"},
  	{"Mic A PGA", NULL, "Mic A Pre Amp"},
  	{"Mic B PGA", NULL, "Mic B Pre Amp"},
  
  	/* left capture select */
  	{"Left Capture Source", "Mic 1", "Mic A Pre Amp"},
  	{"Left Capture Source", "Mic 2", "Mic B Pre Amp"},
  	{"Left Capture Source", "Line", "LINEL"},
  	{"Left Capture Source", "Mono In", "MONOIN"},
  	{"Left Capture Source", "Headphone", "Left HP Mixer"},
  	{"Left Capture Source", "Speaker", "Speaker Mixer"},
  	{"Left Capture Source", "Mono Out", "Mono Mixer"},
  
  	/* right capture select */
  	{"Right Capture Source", "Mic 1", "Mic A Pre Amp"},
  	{"Right Capture Source", "Mic 2", "Mic B Pre Amp"},
  	{"Right Capture Source", "Line", "LINER"},
  	{"Right Capture Source", "Mono In", "MONOIN"},
  	{"Right Capture Source", "Headphone", "Right HP Mixer"},
  	{"Right Capture Source", "Speaker", "Speaker Mixer"},
  	{"Right Capture Source", "Mono Out", "Mono Mixer"},
  
  	/* left ADC */
  	{"Left ADC", NULL, "Left Capture Source"},
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  	{"Left Voice ADC", NULL, "Left ADC"},
  	{"Left HiFi ADC", NULL, "Left ADC"},
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  	/* right ADC */
  	{"Right ADC", NULL, "Right Capture Source"},
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  	{"Right Voice ADC", NULL, "Right ADC"},
  	{"Right HiFi ADC", NULL, "Right ADC"},
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  	/* mic */
  	{"Mic A Pre Amp", NULL, "Mic A Source"},
  	{"Mic A Source", "Mic 1", "MIC1"},
  	{"Mic A Source", "Mic 2 A", "MIC2A"},
  	{"Mic A Source", "Mic 2 B", "Mic B Source"},
  	{"Mic B Pre Amp", "MPB", "Mic B Source"},
  	{"Mic B Source", NULL, "MIC2B"},
  
  	/* headphone capture */
  	{"Capture Headphone Mux", "Stereo", "Capture Mixer"},
  	{"Capture Headphone Mux", "Left", "Left Capture Source"},
  	{"Capture Headphone Mux", "Right", "Right Capture Source"},
  
  	/* mono capture */
  	{"Capture Mono Mux", "Stereo", "Capture Mixer"},
  	{"Capture Mono Mux", "Left", "Left Capture Source"},
  	{"Capture Mono Mux", "Right", "Right Capture Source"},
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  };
  
  static int wm9713_add_widgets(struct snd_soc_codec *codec)
  {
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  	snd_soc_dapm_new_controls(codec, wm9713_dapm_widgets,
  				  ARRAY_SIZE(wm9713_dapm_widgets));
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  	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
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  	return 0;
  }
  
  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_CD)
  		return soc_ac97_ops.read(codec->ac97, reg);
  	else {
  		reg = reg >> 1;
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  		if (reg >= (ARRAY_SIZE(wm9713_reg)))
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  			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;
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  	if (reg < (ARRAY_SIZE(wm9713_reg)))
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  		cache[reg] = val;
  
  	return 0;
  }
  
  /* PLL divisors */
  struct _pll_div {
  	u32 divsel:1;
  	u32 divctl:1;
  	u32 lf:1;
  	u32 n:4;
  	u32 k:24;
  };
  
  /* The size in bits of the PLL divide multiplied by 10
   * to allow rounding later */
  #define FIXED_PLL_SIZE ((1 << 22) * 10)
  
  static void pll_factors(struct _pll_div *pll_div, unsigned int source)
  {
  	u64 Kpart;
  	unsigned int K, Ndiv, Nmod, target;
  
  	/* The the PLL output is always 98.304MHz. */
  	target = 98304000;
  
  	/* If the input frequency is over 14.4MHz then scale it down. */
  	if (source > 14400000) {
  		source >>= 1;
  		pll_div->divsel = 1;
  
  		if (source > 14400000) {
  			source >>= 1;
  			pll_div->divctl = 1;
  		} else
  			pll_div->divctl = 0;
  
  	} else {
  		pll_div->divsel = 0;
  		pll_div->divctl = 0;
  	}
  
  	/* Low frequency sources require an additional divide in the
  	 * loop.
  	 */
  	if (source < 8192000) {
  		pll_div->lf = 1;
  		target >>= 2;
  	} else
  		pll_div->lf = 0;
  
  	Ndiv = target / source;
  	if ((Ndiv < 5) || (Ndiv > 12))
  		printk(KERN_WARNING
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  			"WM9713 PLL N value %u out of recommended range!
  ",
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  			Ndiv);
  
  	pll_div->n = Ndiv;
  	Nmod = target % source;
  	Kpart = FIXED_PLL_SIZE * (long long)Nmod;
  
  	do_div(Kpart, source);
  
  	K = Kpart & 0xFFFFFFFF;
  
  	/* Check if we need to round */
  	if ((K % 10) >= 5)
  		K += 5;
  
  	/* Move down to proper range now rounding is done */
  	K /= 10;
  
  	pll_div->k = K;
  }
  
  /**
   * Please note that changing the PLL input frequency may require
   * resynchronisation with the AC97 controller.
   */
  static int wm9713_set_pll(struct snd_soc_codec *codec,
  	int pll_id, unsigned int freq_in, unsigned int freq_out)
  {
  	struct wm9713_priv *wm9713 = codec->private_data;
  	u16 reg, reg2;
  	struct _pll_div pll_div;
  
  	/* turn PLL off ? */
c42f69bb0   Mark Brown   ASoC: Ignore outp...
741
  	if (freq_in == 0) {
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
742
743
744
745
746
  		/* disable PLL power and select ext source */
  		reg = ac97_read(codec, AC97_HANDSET_RATE);
  		ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080);
  		reg = ac97_read(codec, AC97_EXTENDED_MID);
  		ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200);
c42f69bb0   Mark Brown   ASoC: Ignore outp...
747
  		wm9713->pll_in = 0;
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
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769
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772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
  		return 0;
  	}
  
  	pll_factors(&pll_div, freq_in);
  
  	if (pll_div.k == 0) {
  		reg = (pll_div.n << 12) | (pll_div.lf << 11) |
  			(pll_div.divsel << 9) | (pll_div.divctl << 8);
  		ac97_write(codec, AC97_LINE1_LEVEL, reg);
  	} else {
  		/* write the fractional k to the reg 0x46 pages */
  		reg2 = (pll_div.n << 12) | (pll_div.lf << 11) | (1 << 10) |
  			(pll_div.divsel << 9) | (pll_div.divctl << 8);
  
  		/* K [21:20] */
  		reg = reg2 | (0x5 << 4) | (pll_div.k >> 20);
  		ac97_write(codec, AC97_LINE1_LEVEL, reg);
  
  		/* K [19:16] */
  		reg = reg2 | (0x4 << 4) | ((pll_div.k >> 16) & 0xf);
  		ac97_write(codec, AC97_LINE1_LEVEL, reg);
  
  		/* K [15:12] */
  		reg = reg2 | (0x3 << 4) | ((pll_div.k >> 12) & 0xf);
  		ac97_write(codec, AC97_LINE1_LEVEL, reg);
  
  		/* K [11:8] */
  		reg = reg2 | (0x2 << 4) | ((pll_div.k >> 8) & 0xf);
  		ac97_write(codec, AC97_LINE1_LEVEL, reg);
  
  		/* K [7:4] */
  		reg = reg2 | (0x1 << 4) | ((pll_div.k >> 4) & 0xf);
  		ac97_write(codec, AC97_LINE1_LEVEL, reg);
  
  		reg = reg2 | (0x0 << 4) | (pll_div.k & 0xf); /* K [3:0] */
  		ac97_write(codec, AC97_LINE1_LEVEL, reg);
  	}
  
  	/* turn PLL on and select as source */
  	reg = ac97_read(codec, AC97_EXTENDED_MID);
  	ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff);
  	reg = ac97_read(codec, AC97_HANDSET_RATE);
  	ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f);
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
791
792
793
794
795
796
  	wm9713->pll_in = freq_in;
  
  	/* wait 10ms AC97 link frames for the link to stabilise */
  	schedule_timeout_interruptible(msecs_to_jiffies(10));
  	return 0;
  }
85488037b   Mark Brown   ASoC: Add source ...
797
798
  static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
  		int source, unsigned int freq_in, unsigned int freq_out)
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
799
800
801
802
803
804
805
806
807
  {
  	struct snd_soc_codec *codec = codec_dai->codec;
  	return wm9713_set_pll(codec, pll_id, freq_in, freq_out);
  }
  
  /*
   * Tristate the PCM DAI lines, tristate can be disabled by calling
   * wm9713_set_dai_fmt()
   */
e550e17ff   Liam Girdwood   ALSA: asoc: codec...
808
  static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai,
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
  	int tristate)
  {
  	struct snd_soc_codec *codec = codec_dai->codec;
  	u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0x9fff;
  
  	if (tristate)
  		ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
  
  	return 0;
  }
  
  /*
   * Configure WM9713 clock dividers.
   * Voice DAC needs 256 FS
   */
e550e17ff   Liam Girdwood   ALSA: asoc: codec...
824
  static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
  		int div_id, int div)
  {
  	struct snd_soc_codec *codec = codec_dai->codec;
  	u16 reg;
  
  	switch (div_id) {
  	case WM9713_PCMCLK_DIV:
  		reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xf0ff;
  		ac97_write(codec, AC97_HANDSET_RATE, reg | div);
  		break;
  	case WM9713_CLKA_MULT:
  		reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffd;
  		ac97_write(codec, AC97_HANDSET_RATE, reg | div);
  		break;
  	case WM9713_CLKB_MULT:
  		reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffb;
  		ac97_write(codec, AC97_HANDSET_RATE, reg | div);
  		break;
  	case WM9713_HIFI_DIV:
  		reg = ac97_read(codec, AC97_HANDSET_RATE) & 0x8fff;
  		ac97_write(codec, AC97_HANDSET_RATE, reg | div);
  		break;
  	case WM9713_PCMBCLK_DIV:
  		reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xf1ff;
  		ac97_write(codec, AC97_CENTER_LFE_MASTER, reg | div);
  		break;
  	case WM9713_PCMCLK_PLL_DIV:
  		reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80;
  		ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x60 | div);
  		break;
  	case WM9713_HIFI_PLL_DIV:
  		reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80;
  		ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x70 | div);
  		break;
  	default:
  		return -EINVAL;
  	}
  
  	return 0;
  }
e550e17ff   Liam Girdwood   ALSA: asoc: codec...
865
  static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai,
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
  		unsigned int fmt)
  {
  	struct snd_soc_codec *codec = codec_dai->codec;
  	u16 gpio = ac97_read(codec, AC97_GPIO_CFG) & 0xffc5;
  	u16 reg = 0x8000;
  
  	/* clock masters */
  	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
  	case SND_SOC_DAIFMT_CBM_CFM:
  		reg |= 0x4000;
  		gpio |= 0x0010;
  		break;
  	case SND_SOC_DAIFMT_CBM_CFS:
  		reg |= 0x6000;
  		gpio |= 0x0018;
  		break;
  	case SND_SOC_DAIFMT_CBS_CFS:
d133b0ce6   Mark Brown   ALSA: ASoC: Fix W...
883
  		reg |= 0x2000;
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
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911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
  		gpio |= 0x001a;
  		break;
  	case SND_SOC_DAIFMT_CBS_CFM:
  		gpio |= 0x0012;
  		break;
  	}
  
  	/* clock inversion */
  	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
  	case SND_SOC_DAIFMT_IB_IF:
  		reg |= 0x00c0;
  		break;
  	case SND_SOC_DAIFMT_IB_NF:
  		reg |= 0x0080;
  		break;
  	case SND_SOC_DAIFMT_NB_IF:
  		reg |= 0x0040;
  		break;
  	}
  
  	/* DAI format */
  	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
  	case SND_SOC_DAIFMT_I2S:
  		reg |= 0x0002;
  		break;
  	case SND_SOC_DAIFMT_RIGHT_J:
  		break;
  	case SND_SOC_DAIFMT_LEFT_J:
  		reg |= 0x0001;
  		break;
  	case SND_SOC_DAIFMT_DSP_A:
  		reg |= 0x0003;
  		break;
  	case SND_SOC_DAIFMT_DSP_B:
  		reg |= 0x0043;
  		break;
  	}
  
  	ac97_write(codec, AC97_GPIO_CFG, gpio);
  	ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
  	return 0;
  }
  
  static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
dee89c4d9   Mark Brown   ASoC: Merge snd_s...
928
929
  				struct snd_pcm_hw_params *params,
  				struct snd_soc_dai *dai)
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
930
  {
55b8bac50   Mark Brown   ASoC: Use supplie...
931
  	struct snd_soc_codec *codec = dai->codec;
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
  	u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3;
  
  	switch (params_format(params)) {
  	case SNDRV_PCM_FORMAT_S16_LE:
  		break;
  	case SNDRV_PCM_FORMAT_S20_3LE:
  		reg |= 0x0004;
  		break;
  	case SNDRV_PCM_FORMAT_S24_LE:
  		reg |= 0x0008;
  		break;
  	case SNDRV_PCM_FORMAT_S32_LE:
  		reg |= 0x000c;
  		break;
  	}
  
  	/* enable PCM interface in master mode */
  	ac97_write(codec, AC97_CENTER_LFE_MASTER, reg);
  	return 0;
  }
dee89c4d9   Mark Brown   ASoC: Merge snd_s...
952
953
  static int ac97_hifi_prepare(struct snd_pcm_substream *substream,
  			     struct snd_soc_dai *dai)
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
954
  {
55b8bac50   Mark Brown   ASoC: Use supplie...
955
  	struct snd_soc_codec *codec = dai->codec;
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
956
  	struct snd_pcm_runtime *runtime = substream->runtime;
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
957
958
959
960
961
962
963
964
965
966
967
968
969
  	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);
  }
dee89c4d9   Mark Brown   ASoC: Merge snd_s...
970
971
  static int ac97_aux_prepare(struct snd_pcm_substream *substream,
  			    struct snd_soc_dai *dai)
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
972
  {
55b8bac50   Mark Brown   ASoC: Use supplie...
973
  	struct snd_soc_codec *codec = dai->codec;
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
974
  	struct snd_pcm_runtime *runtime = substream->runtime;
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
975
976
977
978
979
980
981
982
983
984
985
986
  	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);
  }
5415552d2   Mark Brown   ALSA: ASoC: Adver...
987
988
989
990
991
992
993
994
995
996
997
998
  #define WM9713_RATES (SNDRV_PCM_RATE_8000  |	\
  		      SNDRV_PCM_RATE_11025 |	\
  		      SNDRV_PCM_RATE_22050 |	\
  		      SNDRV_PCM_RATE_44100 |	\
  		      SNDRV_PCM_RATE_48000)
  
  #define WM9713_PCM_RATES (SNDRV_PCM_RATE_8000  |	\
  			  SNDRV_PCM_RATE_11025 |	\
  			  SNDRV_PCM_RATE_16000 |	\
  			  SNDRV_PCM_RATE_22050 |	\
  			  SNDRV_PCM_RATE_44100 |	\
  			  SNDRV_PCM_RATE_48000)
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
999
1000
1001
1002
  
  #define WM9713_PCM_FORMATS \
  	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
  	 SNDRV_PCM_FORMAT_S24_LE)
6335d0554   Eric Miao   ASoC: make ops a ...
1003
1004
1005
1006
1007
1008
1009
1010
1011
1012
1013
1014
1015
1016
  static struct snd_soc_dai_ops wm9713_dai_ops_hifi = {
  	.prepare	= ac97_hifi_prepare,
  	.set_clkdiv	= wm9713_set_dai_clkdiv,
  	.set_pll	= wm9713_set_dai_pll,
  };
  
  static struct snd_soc_dai_ops wm9713_dai_ops_aux = {
  	.prepare	= ac97_aux_prepare,
  	.set_clkdiv	= wm9713_set_dai_clkdiv,
  	.set_pll	= wm9713_set_dai_pll,
  };
  
  static struct snd_soc_dai_ops wm9713_dai_ops_voice = {
  	.hw_params	= wm9713_pcm_hw_params,
6335d0554   Eric Miao   ASoC: make ops a ...
1017
1018
1019
1020
1021
  	.set_clkdiv	= wm9713_set_dai_clkdiv,
  	.set_pll	= wm9713_set_dai_pll,
  	.set_fmt	= wm9713_set_dai_fmt,
  	.set_tristate	= wm9713_set_dai_tristate,
  };
e550e17ff   Liam Girdwood   ALSA: asoc: codec...
1022
  struct snd_soc_dai wm9713_dai[] = {
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1023
1024
  {
  	.name = "AC97 HiFi",
3ba9e10a6   Mark Brown   ASoC: Remove DAI ...
1025
  	.ac97_control = 1,
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1026
1027
1028
1029
1030
  	.playback = {
  		.stream_name = "HiFi Playback",
  		.channels_min = 1,
  		.channels_max = 2,
  		.rates = WM9713_RATES,
33f503c96   Mark Brown   ASoC: Use a share...
1031
  		.formats = SND_SOC_STD_AC97_FMTS,},
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1032
1033
1034
1035
1036
  	.capture = {
  		.stream_name = "HiFi Capture",
  		.channels_min = 1,
  		.channels_max = 2,
  		.rates = WM9713_RATES,
33f503c96   Mark Brown   ASoC: Use a share...
1037
  		.formats = SND_SOC_STD_AC97_FMTS,},
6335d0554   Eric Miao   ASoC: make ops a ...
1038
  	.ops = &wm9713_dai_ops_hifi,
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1039
1040
1041
1042
1043
1044
1045
1046
  	},
  	{
  	.name = "AC97 Aux",
  	.playback = {
  		.stream_name = "Aux Playback",
  		.channels_min = 1,
  		.channels_max = 1,
  		.rates = WM9713_RATES,
33f503c96   Mark Brown   ASoC: Use a share...
1047
  		.formats = SND_SOC_STD_AC97_FMTS,},
6335d0554   Eric Miao   ASoC: make ops a ...
1048
  	.ops = &wm9713_dai_ops_aux,
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1049
1050
1051
1052
1053
1054
1055
  	},
  	{
  	.name = "WM9713 Voice",
  	.playback = {
  		.stream_name = "Voice Playback",
  		.channels_min = 1,
  		.channels_max = 1,
5415552d2   Mark Brown   ALSA: ASoC: Adver...
1056
  		.rates = WM9713_PCM_RATES,
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1057
1058
1059
1060
1061
  		.formats = WM9713_PCM_FORMATS,},
  	.capture = {
  		.stream_name = "Voice Capture",
  		.channels_min = 1,
  		.channels_max = 2,
5415552d2   Mark Brown   ALSA: ASoC: Adver...
1062
  		.rates = WM9713_PCM_RATES,
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1063
  		.formats = WM9713_PCM_FORMATS,},
6335d0554   Eric Miao   ASoC: make ops a ...
1064
  	.ops = &wm9713_dai_ops_voice,
f4976116a   Mark Brown   ASoC: WM9713 requ...
1065
  	.symmetric_rates = 1,
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1066
1067
1068
1069
1070
1071
1072
1073
  	},
  };
  EXPORT_SYMBOL_GPL(wm9713_dai);
  
  int wm9713_reset(struct snd_soc_codec *codec, int try_warm)
  {
  	if (try_warm && soc_ac97_ops.warm_reset) {
  		soc_ac97_ops.warm_reset(codec->ac97);
abb68c26b   Mark Brown   ALSA: ASoC: Check...
1074
  		if (ac97_read(codec, 0) == wm9713_reg[0])
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1075
1076
1077
1078
  			return 1;
  	}
  
  	soc_ac97_ops.reset(codec->ac97);
e775f6c0f   Mark Brown   ASoC: Do a warm r...
1079
1080
  	if (soc_ac97_ops.warm_reset)
  		soc_ac97_ops.warm_reset(codec->ac97);
abb68c26b   Mark Brown   ALSA: ASoC: Check...
1081
  	if (ac97_read(codec, 0) != wm9713_reg[0])
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1082
1083
1084
1085
  		return -EIO;
  	return 0;
  }
  EXPORT_SYMBOL_GPL(wm9713_reset);
0be9898ad   Mark Brown   [ALSA] ASoC: Clar...
1086
1087
  static int wm9713_set_bias_level(struct snd_soc_codec *codec,
  				 enum snd_soc_bias_level level)
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1088
1089
  {
  	u16 reg;
0be9898ad   Mark Brown   [ALSA] ASoC: Clar...
1090
1091
  	switch (level) {
  	case SND_SOC_BIAS_ON:
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1092
1093
1094
1095
  		/* enable thermal shutdown */
  		reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff;
  		ac97_write(codec, AC97_EXTENDED_MID, reg);
  		break;
0be9898ad   Mark Brown   [ALSA] ASoC: Clar...
1096
  	case SND_SOC_BIAS_PREPARE:
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1097
  		break;
0be9898ad   Mark Brown   [ALSA] ASoC: Clar...
1098
  	case SND_SOC_BIAS_STANDBY:
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1099
1100
1101
1102
1103
  		/* enable master bias and vmid */
  		reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff;
  		ac97_write(codec, AC97_EXTENDED_MID, reg);
  		ac97_write(codec, AC97_POWERDOWN, 0x0000);
  		break;
0be9898ad   Mark Brown   [ALSA] ASoC: Clar...
1104
  	case SND_SOC_BIAS_OFF:
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1105
1106
1107
1108
1109
1110
  		/* disable everything including AC link */
  		ac97_write(codec, AC97_EXTENDED_MID, 0xffff);
  		ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
  		ac97_write(codec, AC97_POWERDOWN, 0xffff);
  		break;
  	}
0be9898ad   Mark Brown   [ALSA] ASoC: Clar...
1111
  	codec->bias_level = level;
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1112
1113
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1116
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1118
  	return 0;
  }
  
  static int wm9713_soc_suspend(struct platform_device *pdev,
  	pm_message_t state)
  {
  	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
6627a653b   Mark Brown   ASoC: Push the co...
1119
  	struct snd_soc_codec *codec = socdev->card->codec;
87b57fe2d   Mark Brown   [ALSA] wm9713: Do...
1120
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  	u16 reg;
  
  	/* Disable everything except touchpanel - that will be handled
  	 * by the touch driver and left disabled if touch is not in
  	 * use. */
  	reg = ac97_read(codec, AC97_EXTENDED_MID);
  	ac97_write(codec, AC97_EXTENDED_MID, reg | 0x7fff);
  	ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
  	ac97_write(codec, AC97_POWERDOWN, 0x6f00);
  	ac97_write(codec, AC97_POWERDOWN, 0xffff);
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1130

83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1131
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1133
1134
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1136
  	return 0;
  }
  
  static int wm9713_soc_resume(struct platform_device *pdev)
  {
  	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
6627a653b   Mark Brown   ASoC: Push the co...
1137
  	struct snd_soc_codec *codec = socdev->card->codec;
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1138
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  	struct wm9713_priv *wm9713 = codec->private_data;
  	int i, ret;
  	u16 *cache = codec->reg_cache;
  
  	ret = wm9713_reset(codec, 1);
  	if (ret < 0) {
  		printk(KERN_ERR "could not reset AC97 codec
  ");
  		return ret;
  	}
0be9898ad   Mark Brown   [ALSA] ASoC: Clar...
1148
  	wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1149
1150
  
  	/* do we need to re-start the PLL ? */
c42f69bb0   Mark Brown   ASoC: Ignore outp...
1151
1152
  	if (wm9713->pll_in)
  		wm9713_set_pll(codec, 0, wm9713->pll_in, 0);
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1153
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  	/* only synchronise the codec if warm reset failed */
  	if (ret == 0) {
  		for (i = 2; i < ARRAY_SIZE(wm9713_reg) << 1; i += 2) {
  			if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID ||
  				i == AC97_EXTENDED_MSTATUS || i > 0x66)
  				continue;
  			soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
  		}
  	}
0be9898ad   Mark Brown   [ALSA] ASoC: Clar...
1163
1164
  	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
  		wm9713_set_bias_level(codec, SND_SOC_BIAS_ON);
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1165
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  	return ret;
  }
  
  static int wm9713_soc_probe(struct platform_device *pdev)
  {
  	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
  	struct snd_soc_codec *codec;
  	int ret = 0, reg;
  
  	printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s
  ", WM9713_VERSION);
6627a653b   Mark Brown   ASoC: Push the co...
1177
1178
1179
  	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec),
  				      GFP_KERNEL);
  	if (socdev->card->codec == NULL)
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1180
  		return -ENOMEM;
6627a653b   Mark Brown   ASoC: Push the co...
1181
  	codec = socdev->card->codec;
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
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  	mutex_init(&codec->mutex);
  
  	codec->reg_cache = kmemdup(wm9713_reg, sizeof(wm9713_reg), GFP_KERNEL);
  	if (codec->reg_cache == NULL) {
  		ret = -ENOMEM;
  		goto cache_err;
  	}
  	codec->reg_cache_size = sizeof(wm9713_reg);
  	codec->reg_cache_step = 2;
  
  	codec->private_data = kzalloc(sizeof(struct wm9713_priv), GFP_KERNEL);
  	if (codec->private_data == NULL) {
  		ret = -ENOMEM;
  		goto priv_err;
  	}
  
  	codec->name = "WM9713";
  	codec->owner = THIS_MODULE;
  	codec->dai = wm9713_dai;
  	codec->num_dai = ARRAY_SIZE(wm9713_dai);
  	codec->write = ac97_write;
  	codec->read = ac97_read;
0be9898ad   Mark Brown   [ALSA] ASoC: Clar...
1204
  	codec->set_bias_level = wm9713_set_bias_level;
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1205
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  	INIT_LIST_HEAD(&codec->dapm_widgets);
  	INIT_LIST_HEAD(&codec->dapm_paths);
  
  	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
  	if (ret < 0)
  		goto codec_err;
  
  	/* register pcms */
  	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
  	if (ret < 0)
  		goto pcm_err;
  
  	/* do a cold reset for the controller and then try
  	 * a warm reset followed by an optional cold reset for codec */
  	wm9713_reset(codec, 0);
  	ret = wm9713_reset(codec, 1);
  	if (ret < 0) {
39639faba   Mark Brown   ASoC: Improve err...
1222
1223
  		printk(KERN_ERR "Failed to reset WM9713: AC97 link error
  ");
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1224
1225
  		goto reset_err;
  	}
0be9898ad   Mark Brown   [ALSA] ASoC: Clar...
1226
  	wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1227
1228
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1230
  
  	/* unmute the adc - move to kcontrol */
  	reg = ac97_read(codec, AC97_CD) & 0x7fff;
  	ac97_write(codec, AC97_CD, reg);
3e8e1952e   Ian Molton   ASoC: cleanup dup...
1231
1232
  	snd_soc_add_controls(codec, wm9713_snd_ac97_controls,
  				ARRAY_SIZE(wm9713_snd_ac97_controls));
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1233
  	wm9713_add_widgets(codec);
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1234

fe3e78e07   Mark Brown   ASoC: Factor out ...
1235
  	return 0;
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1236
1237
1238
1239
1240
1241
1242
1243
1244
1245
1246
  
  pcm_err:
  	snd_soc_free_ac97_codec(codec);
  
  codec_err:
  	kfree(codec->private_data);
  
  priv_err:
  	kfree(codec->reg_cache);
  
  cache_err:
6627a653b   Mark Brown   ASoC: Push the co...
1247
1248
  	kfree(socdev->card->codec);
  	socdev->card->codec = NULL;
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1249
1250
1251
1252
1253
1254
  	return ret;
  }
  
  static int wm9713_soc_remove(struct platform_device *pdev)
  {
  	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
6627a653b   Mark Brown   ASoC: Push the co...
1255
  	struct snd_soc_codec *codec = socdev->card->codec;
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1256
1257
1258
1259
1260
1261
1262
1263
1264
  
  	if (codec == NULL)
  		return 0;
  
  	snd_soc_dapm_free(socdev);
  	snd_soc_free_pcms(socdev);
  	snd_soc_free_ac97_codec(codec);
  	kfree(codec->private_data);
  	kfree(codec->reg_cache);
83ac08c08   Liam Girdwood   [ALSA] ASoC: WM97...
1265
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1279
  	kfree(codec);
  	return 0;
  }
  
  struct snd_soc_codec_device soc_codec_dev_wm9713 = {
  	.probe = 	wm9713_soc_probe,
  	.remove = 	wm9713_soc_remove,
  	.suspend =	wm9713_soc_suspend,
  	.resume = 	wm9713_soc_resume,
  };
  EXPORT_SYMBOL_GPL(soc_codec_dev_wm9713);
  
  MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver");
  MODULE_AUTHOR("Liam Girdwood");
  MODULE_LICENSE("GPL");