AuxXxilium/linux_dsm_epyc7002
1
0
Fork 0
mirror of https://github.com/AuxXxilium/linux_dsm_epyc7002.git synced 2024-12-21 07:09:15 +07:00
Code Issues Actions Packages Projects Releases Wiki Activity
3d3db94328
linux_dsm_epyc7002/sound/soc/codecs/wm9712.c
Robert Jarzmik 6bd25aae3a ASoC: wm97xx: fix compilation corner case 
When the old AC97 is not used, CONFIG_SND_SOC_AC97_BUS is not
defined. As a consequence, in the error path, snd_soc_free_ac97_codec()
is not defined and triggers a compilation error.

Fix it for wm9705 and wm9712, as wm9713 is correctly written.

Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2017-10-18 11:25:15 +01:00

740 lines
24 KiB
C
Raw Blame History

/*
* 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/mfd/wm97xx.h>
#include <linux/module.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <linux/regmap.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/ac97/codec.h>
#include <sound/ac97/compat.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include <sound/tlv.h>
#define WM9712_VENDOR_ID 0x574d4c12
#define WM9712_VENDOR_ID_MASK 0xffffffff
struct wm9712_priv {
struct snd_ac97 *ac97;
unsigned int hp_mixer[2];
struct mutex lock;
struct wm97xx_platform_data *mfd_pdata;
};
static const struct reg_default wm9712_reg_defaults[] = {
{ 0x02, 0x8000 },
{ 0x04, 0x8000 },
{ 0x06, 0x8000 },
{ 0x08, 0x0f0f },
{ 0x0a, 0xaaa0 },
{ 0x0c, 0xc008 },
{ 0x0e, 0x6808 },
{ 0x10, 0xe808 },
{ 0x12, 0xaaa0 },
{ 0x14, 0xad00 },
{ 0x16, 0x8000 },
{ 0x18, 0xe808 },
{ 0x1a, 0x3000 },
{ 0x1c, 0x8000 },
{ 0x20, 0x0000 },
{ 0x22, 0x0000 },
{ 0x26, 0x000f },
{ 0x28, 0x0605 },
{ 0x2a, 0x0410 },
{ 0x2c, 0xbb80 },
{ 0x2e, 0xbb80 },
{ 0x32, 0xbb80 },
{ 0x34, 0x2000 },
{ 0x4c, 0xf83e },
{ 0x4e, 0xffff },
{ 0x50, 0x0000 },
{ 0x52, 0x0000 },
{ 0x56, 0xf83e },
{ 0x58, 0x0008 },
{ 0x5c, 0x0000 },
{ 0x60, 0xb032 },
{ 0x62, 0x3e00 },
{ 0x64, 0x0000 },
{ 0x76, 0x0006 },
{ 0x78, 0x0001 },
{ 0x7a, 0x0000 },
};
static bool wm9712_volatile_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
case AC97_REC_GAIN:
return true;
default:
return regmap_ac97_default_volatile(dev, reg);
}
}
static const struct regmap_config wm9712_regmap_config = {
.reg_bits = 16,
.reg_stride = 2,
.val_bits = 16,
.max_register = 0x7e,
.cache_type = REGCACHE_RBTREE,
.volatile_reg = wm9712_volatile_reg,
.reg_defaults = wm9712_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(wm9712_reg_defaults),
};
#define HPL_MIXER 0x0
#define HPR_MIXER 0x1
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),
};
static const unsigned int wm9712_mixer_mute_regs[] = {
AC97_VIDEO,
AC97_PCM,
AC97_LINE,
AC97_PHONE,
AC97_CD,
AC97_PC_BEEP,
};
/* 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 wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
unsigned int val = ucontrol->value.integer.value[0];
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
unsigned int mixer, mask, shift, old;
struct snd_soc_dapm_update update = { 0 };
bool change;
mixer = mc->shift >> 8;
shift = mc->shift & 0xff;
mask = 1 << shift;
mutex_lock(&wm9712->lock);
old = wm9712->hp_mixer[mixer];
if (ucontrol->value.integer.value[0])
wm9712->hp_mixer[mixer] |= mask;
else
wm9712->hp_mixer[mixer] &= ~mask;
change = old != wm9712->hp_mixer[mixer];
if (change) {
update.kcontrol = kcontrol;
update.reg = wm9712_mixer_mute_regs[shift];
update.mask = 0x8000;
if ((wm9712->hp_mixer[0] & mask) ||
(wm9712->hp_mixer[1] & mask))
update.val = 0x0;
else
update.val = 0x8000;
snd_soc_dapm_mixer_update_power(dapm, kcontrol, val,
&update);
}
mutex_unlock(&wm9712->lock);
return change;
}
static int wm9712_hp_mixer_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
unsigned int shift, mixer;
mixer = mc->shift >> 8;
shift = mc->shift & 0xff;
ucontrol->value.integer.value[0] =
(wm9712->hp_mixer[mixer] >> shift) & 1;
return 0;
}
#define WM9712_HP_MIXER_CTRL(xname, xmixer, xshift) { \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_volsw, \
.get = wm9712_hp_mixer_get, .put = wm9712_hp_mixer_put, \
.private_value = SOC_SINGLE_VALUE(SND_SOC_NOPM, \
(xmixer << 8) | xshift, 1, 0, 0) \
}
/* Left Headphone Mixers */
static const struct snd_kcontrol_new wm9712_hpl_mixer_controls[] = {
WM9712_HP_MIXER_CTRL("PCBeep Bypass Switch", HPL_MIXER, 5),
WM9712_HP_MIXER_CTRL("Aux Playback Switch", HPL_MIXER, 4),
WM9712_HP_MIXER_CTRL("Phone Bypass Switch", HPL_MIXER, 3),
WM9712_HP_MIXER_CTRL("Line Bypass Switch", HPL_MIXER, 2),
WM9712_HP_MIXER_CTRL("PCM Playback Switch", HPL_MIXER, 1),
WM9712_HP_MIXER_CTRL("Mic Sidetone Switch", HPL_MIXER, 0),
};
/* Right Headphone Mixers */
static const struct snd_kcontrol_new wm9712_hpr_mixer_controls[] = {
WM9712_HP_MIXER_CTRL("PCBeep Bypass Switch", HPR_MIXER, 5),
WM9712_HP_MIXER_CTRL("Aux Playback Switch", HPR_MIXER, 4),
WM9712_HP_MIXER_CTRL("Phone Bypass Switch", HPR_MIXER, 3),
WM9712_HP_MIXER_CTRL("Line Bypass Switch", HPR_MIXER, 2),
WM9712_HP_MIXER_CTRL("PCM Playback Switch", HPR_MIXER, 1),
WM9712_HP_MIXER_CTRL("Mic Sidetone Switch", HPR_MIXER, 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("Left HP Mixer", AC97_INT_PAGING, 9, 1,
&wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls)),
SND_SOC_DAPM_MIXER("Right HP Mixer", AC97_INT_PAGING, 8, 1,
&wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls)),
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 int ac97_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
int reg;
struct snd_pcm_runtime *runtime = substream->runtime;
snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x1, 0x1);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
reg = AC97_PCM_FRONT_DAC_RATE;
else
reg = AC97_PCM_LR_ADC_RATE;
return snd_soc_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;
struct snd_pcm_runtime *runtime = substream->runtime;
snd_soc_update_bits(codec, AC97_EXTENDED_STATUS, 0x1, 0x1);
snd_soc_update_bits(codec, AC97_PCI_SID, 0x8000, 0x8000);
if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
return -ENODEV;
return snd_soc_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",
.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:
snd_soc_write(codec, AC97_POWERDOWN, 0x0000);
break;
case SND_SOC_BIAS_OFF:
/* disable everything including AC link */
snd_soc_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
snd_soc_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
return 0;
}
static int wm9712_soc_resume(struct snd_soc_codec *codec)
{
struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
int ret;
ret = snd_ac97_reset(wm9712->ac97, true, WM9712_VENDOR_ID,
WM9712_VENDOR_ID_MASK);
if (ret < 0)
return ret;
snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
if (ret == 0)
regcache_sync(codec->component.regmap);
return ret;
}
static int wm9712_soc_probe(struct snd_soc_codec *codec)
{
struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
struct regmap *regmap;
int ret;
if (wm9712->mfd_pdata) {
wm9712->ac97 = wm9712->mfd_pdata->ac97;
regmap = wm9712->mfd_pdata->regmap;
} else {
#ifdef CONFIG_SND_SOC_AC97_BUS
wm9712->ac97 = snd_soc_new_ac97_codec(codec, WM9712_VENDOR_ID,
WM9712_VENDOR_ID_MASK);
if (IS_ERR(wm9712->ac97)) {
ret = PTR_ERR(wm9712->ac97);
dev_err(codec->dev,
"Failed to register AC97 codec: %d\n", ret);
return ret;
}
regmap = regmap_init_ac97(wm9712->ac97, &wm9712_regmap_config);
if (IS_ERR(regmap)) {
snd_soc_free_ac97_codec(wm9712->ac97);
return PTR_ERR(regmap);
}
#endif
}
snd_soc_codec_init_regmap(codec, regmap);
/* set alc mux to none */
snd_soc_update_bits(codec, AC97_VIDEO, 0x3000, 0x3000);
return 0;
}
static int wm9712_soc_remove(struct snd_soc_codec *codec)
{
#ifdef CONFIG_SND_SOC_AC97_BUS
struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
if (!wm9712->mfd_pdata) {
snd_soc_codec_exit_regmap(codec);
snd_soc_free_ac97_codec(wm9712->ac97);
}
#endif
return 0;
}
static const struct snd_soc_codec_driver soc_codec_dev_wm9712 = {
.probe = wm9712_soc_probe,
.remove = wm9712_soc_remove,
.resume = wm9712_soc_resume,
.set_bias_level = wm9712_set_bias_level,
.suspend_bias_off = true,
.component_driver = {
.controls = wm9712_snd_ac97_controls,
.num_controls = ARRAY_SIZE(wm9712_snd_ac97_controls),
.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)
{
struct wm9712_priv *wm9712;
wm9712 = devm_kzalloc(&pdev->dev, sizeof(*wm9712), GFP_KERNEL);
if (wm9712 == NULL)
return -ENOMEM;
mutex_init(&wm9712->lock);
wm9712->mfd_pdata = dev_get_platdata(&pdev->dev);
platform_set_drvdata(pdev, wm9712);
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",
},
.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");
Reference in New Issue View Git Blame Copy Permalink