forked from Minki/linux
0d895e14df
The DAI ID defines are back from the time when DAIs were referenced by a
numerical ID. These days a string is used for matching instead and the
defines are unused. The last user of these defines was removed in commit
f0fba2ad1b
("ASoC: multi-component - ASoC Multi-Component Support"). So
remove the defines as well.
This also means the wm9712.h file no longer has any content and can be
removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
726 lines
24 KiB
C
726 lines
24 KiB
C
/*
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* wm9712.c -- ALSA Soc WM9712 codec support
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*
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* Copyright 2006-12 Wolfson Microelectronics PLC.
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* Author: Liam Girdwood <lrg@slimlogic.co.uk>
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*
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* This program is free software; you can redistribute it and/or modify it
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* under the terms of the GNU General Public License as published by the
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* Free Software Foundation; either version 2 of the License, or (at your
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* option) any later version.
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*/
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#include <linux/init.h>
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#include <linux/slab.h>
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#include <linux/module.h>
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#include <linux/kernel.h>
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#include <linux/device.h>
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#include <linux/regmap.h>
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/ac97_codec.h>
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#include <sound/initval.h>
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#include <sound/soc.h>
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#include <sound/tlv.h>
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#define WM9712_VENDOR_ID 0x574d4c12
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#define WM9712_VENDOR_ID_MASK 0xffffffff
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struct wm9712_priv {
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struct snd_ac97 *ac97;
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unsigned int hp_mixer[2];
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struct mutex lock;
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};
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static const struct reg_default wm9712_reg_defaults[] = {
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{ 0x02, 0x8000 },
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{ 0x04, 0x8000 },
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{ 0x06, 0x8000 },
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{ 0x08, 0x0f0f },
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{ 0x0a, 0xaaa0 },
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{ 0x0c, 0xc008 },
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{ 0x0e, 0x6808 },
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{ 0x10, 0xe808 },
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{ 0x12, 0xaaa0 },
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{ 0x14, 0xad00 },
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{ 0x16, 0x8000 },
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{ 0x18, 0xe808 },
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{ 0x1a, 0x3000 },
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{ 0x1c, 0x8000 },
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{ 0x20, 0x0000 },
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{ 0x22, 0x0000 },
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{ 0x26, 0x000f },
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{ 0x28, 0x0605 },
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{ 0x2a, 0x0410 },
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{ 0x2c, 0xbb80 },
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{ 0x2e, 0xbb80 },
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{ 0x32, 0xbb80 },
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{ 0x34, 0x2000 },
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{ 0x4c, 0xf83e },
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{ 0x4e, 0xffff },
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{ 0x50, 0x0000 },
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{ 0x52, 0x0000 },
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{ 0x56, 0xf83e },
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{ 0x58, 0x0008 },
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{ 0x5c, 0x0000 },
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{ 0x60, 0xb032 },
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{ 0x62, 0x3e00 },
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{ 0x64, 0x0000 },
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{ 0x76, 0x0006 },
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{ 0x78, 0x0001 },
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{ 0x7a, 0x0000 },
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};
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static bool wm9712_volatile_reg(struct device *dev, unsigned int reg)
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{
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switch (reg) {
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case AC97_REC_GAIN:
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return true;
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default:
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return regmap_ac97_default_volatile(dev, reg);
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}
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}
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static const struct regmap_config wm9712_regmap_config = {
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.reg_bits = 16,
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.reg_stride = 2,
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.val_bits = 16,
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.max_register = 0x7e,
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.cache_type = REGCACHE_RBTREE,
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.volatile_reg = wm9712_volatile_reg,
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.reg_defaults = wm9712_reg_defaults,
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.num_reg_defaults = ARRAY_SIZE(wm9712_reg_defaults),
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};
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#define HPL_MIXER 0x0
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#define HPR_MIXER 0x1
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static const char *wm9712_alc_select[] = {"None", "Left", "Right", "Stereo"};
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static const char *wm9712_alc_mux[] = {"Stereo", "Left", "Right", "None"};
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static const char *wm9712_out3_src[] = {"Left", "VREF", "Left + Right",
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"Mono"};
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static const char *wm9712_spk_src[] = {"Speaker Mix", "Headphone Mix"};
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static const char *wm9712_rec_adc[] = {"Stereo", "Left", "Right", "Mute"};
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static const char *wm9712_base[] = {"Linear Control", "Adaptive Boost"};
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static const char *wm9712_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"};
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static const char *wm9712_mic[] = {"Mic 1", "Differential", "Mic 2",
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"Stereo"};
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static const char *wm9712_rec_sel[] = {"Mic", "NC", "NC", "Speaker Mixer",
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"Line", "Headphone Mixer", "Phone Mixer", "Phone"};
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static const char *wm9712_ng_type[] = {"Constant Gain", "Mute"};
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static const char *wm9712_diff_sel[] = {"Mic", "Line"};
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static const DECLARE_TLV_DB_SCALE(main_tlv, -3450, 150, 0);
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static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 2000, 0);
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static const struct soc_enum wm9712_enum[] = {
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SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9712_alc_select),
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SOC_ENUM_SINGLE(AC97_VIDEO, 12, 4, wm9712_alc_mux),
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SOC_ENUM_SINGLE(AC97_AUX, 9, 4, wm9712_out3_src),
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SOC_ENUM_SINGLE(AC97_AUX, 8, 2, wm9712_spk_src),
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SOC_ENUM_SINGLE(AC97_REC_SEL, 12, 4, wm9712_rec_adc),
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SOC_ENUM_SINGLE(AC97_MASTER_TONE, 15, 2, wm9712_base),
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SOC_ENUM_DOUBLE(AC97_REC_GAIN, 14, 6, 2, wm9712_rec_gain),
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SOC_ENUM_SINGLE(AC97_MIC, 5, 4, wm9712_mic),
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SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9712_rec_sel),
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SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9712_rec_sel),
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SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9712_ng_type),
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SOC_ENUM_SINGLE(0x5c, 8, 2, wm9712_diff_sel),
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};
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static const struct snd_kcontrol_new wm9712_snd_ac97_controls[] = {
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SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
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SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1),
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SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
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SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
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SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
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SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0),
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SOC_SINGLE("Speaker Playback Invert Switch", AC97_MASTER, 6, 1, 0),
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SOC_SINGLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 7, 1, 0),
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SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_MONO, 7, 1, 0),
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SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1),
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SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
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SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
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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),
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SOC_ENUM("ALC Function", wm9712_enum[0]),
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SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
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SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 1),
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SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0),
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SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0),
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SOC_ENUM("ALC NG Type", wm9712_enum[10]),
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SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 1),
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SOC_SINGLE("Mic Headphone Volume", AC97_VIDEO, 12, 7, 1),
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SOC_SINGLE("ALC Headphone Volume", AC97_VIDEO, 7, 7, 1),
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SOC_SINGLE("Out3 Switch", AC97_AUX, 15, 1, 1),
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SOC_SINGLE("Out3 ZC Switch", AC97_AUX, 7, 1, 1),
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SOC_SINGLE("Out3 Volume", AC97_AUX, 0, 31, 1),
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SOC_SINGLE("PCBeep Bypass Headphone Volume", AC97_PC_BEEP, 12, 7, 1),
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SOC_SINGLE("PCBeep Bypass Speaker Volume", AC97_PC_BEEP, 8, 7, 1),
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SOC_SINGLE("PCBeep Bypass Phone Volume", AC97_PC_BEEP, 4, 7, 1),
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SOC_SINGLE("Aux Playback Headphone Volume", AC97_CD, 12, 7, 1),
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SOC_SINGLE("Aux Playback Speaker Volume", AC97_CD, 8, 7, 1),
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SOC_SINGLE("Aux Playback Phone Volume", AC97_CD, 4, 7, 1),
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SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 1),
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SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1),
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SOC_SINGLE_TLV("Capture Boost Switch", AC97_REC_SEL, 14, 1, 0, boost_tlv),
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SOC_SINGLE_TLV("Capture to Phone Boost Switch", AC97_REC_SEL, 11, 1, 1,
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boost_tlv),
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SOC_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1),
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SOC_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1),
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SOC_SINGLE("3D Playback Volume", AC97_3D_CONTROL, 0, 15, 0),
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SOC_ENUM("Bass Control", wm9712_enum[5]),
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SOC_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1),
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SOC_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1),
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SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0),
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SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 1),
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SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1),
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SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1),
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SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),
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SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0),
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SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
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SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv),
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SOC_SINGLE_TLV("Mic 2 Volume", AC97_MIC, 0, 31, 1, main_tlv),
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SOC_SINGLE_TLV("Mic Boost Volume", AC97_MIC, 7, 1, 0, boost_tlv),
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};
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static const unsigned int wm9712_mixer_mute_regs[] = {
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AC97_VIDEO,
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AC97_PCM,
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AC97_LINE,
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AC97_PHONE,
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AC97_CD,
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AC97_PC_BEEP,
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};
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/* We have to create a fake left and right HP mixers because
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* the codec only has a single control that is shared by both channels.
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* This makes it impossible to determine the audio path.
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*/
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static int wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
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struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
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struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
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unsigned int val = ucontrol->value.integer.value[0];
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struct soc_mixer_control *mc =
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(struct soc_mixer_control *)kcontrol->private_value;
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unsigned int mixer, mask, shift, old;
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struct snd_soc_dapm_update update;
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bool change;
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mixer = mc->shift >> 8;
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shift = mc->shift & 0xff;
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mask = 1 << shift;
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mutex_lock(&wm9712->lock);
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old = wm9712->hp_mixer[mixer];
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if (ucontrol->value.integer.value[0])
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wm9712->hp_mixer[mixer] |= mask;
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else
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wm9712->hp_mixer[mixer] &= ~mask;
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change = old != wm9712->hp_mixer[mixer];
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if (change) {
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update.kcontrol = kcontrol;
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update.reg = wm9712_mixer_mute_regs[shift];
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update.mask = 0x8000;
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if ((wm9712->hp_mixer[0] & mask) ||
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(wm9712->hp_mixer[1] & mask))
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update.val = 0x0;
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else
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update.val = 0x8000;
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snd_soc_dapm_mixer_update_power(dapm, kcontrol, val,
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&update);
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}
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mutex_unlock(&wm9712->lock);
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return change;
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}
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static int wm9712_hp_mixer_get(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
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struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm);
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struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
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struct soc_mixer_control *mc =
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(struct soc_mixer_control *)kcontrol->private_value;
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unsigned int shift, mixer;
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mixer = mc->shift >> 8;
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shift = mc->shift & 0xff;
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ucontrol->value.integer.value[0] =
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(wm9712->hp_mixer[mixer] >> shift) & 1;
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return 0;
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}
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#define WM9712_HP_MIXER_CTRL(xname, xmixer, xshift) { \
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.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
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.info = snd_soc_info_volsw, \
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.get = wm9712_hp_mixer_get, .put = wm9712_hp_mixer_put, \
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.private_value = SOC_SINGLE_VALUE(SND_SOC_NOPM, \
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(xmixer << 8) | xshift, 1, 0, 0) \
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}
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/* Left Headphone Mixers */
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static const struct snd_kcontrol_new wm9712_hpl_mixer_controls[] = {
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WM9712_HP_MIXER_CTRL("PCBeep Bypass Switch", HPL_MIXER, 5),
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WM9712_HP_MIXER_CTRL("Aux Playback Switch", HPL_MIXER, 4),
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WM9712_HP_MIXER_CTRL("Phone Bypass Switch", HPL_MIXER, 3),
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WM9712_HP_MIXER_CTRL("Line Bypass Switch", HPL_MIXER, 2),
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WM9712_HP_MIXER_CTRL("PCM Playback Switch", HPL_MIXER, 1),
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WM9712_HP_MIXER_CTRL("Mic Sidetone Switch", HPL_MIXER, 0),
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};
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/* Right Headphone Mixers */
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static const struct snd_kcontrol_new wm9712_hpr_mixer_controls[] = {
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WM9712_HP_MIXER_CTRL("PCBeep Bypass Switch", HPR_MIXER, 5),
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WM9712_HP_MIXER_CTRL("Aux Playback Switch", HPR_MIXER, 4),
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WM9712_HP_MIXER_CTRL("Phone Bypass Switch", HPR_MIXER, 3),
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WM9712_HP_MIXER_CTRL("Line Bypass Switch", HPR_MIXER, 2),
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WM9712_HP_MIXER_CTRL("PCM Playback Switch", HPR_MIXER, 1),
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WM9712_HP_MIXER_CTRL("Mic Sidetone Switch", HPR_MIXER, 0),
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};
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/* Speaker Mixer */
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static const struct snd_kcontrol_new wm9712_speaker_mixer_controls[] = {
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SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 11, 1, 1),
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SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 11, 1, 1),
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SOC_DAPM_SINGLE("Phone Bypass Switch", AC97_PHONE, 14, 1, 1),
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SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 14, 1, 1),
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SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 14, 1, 1),
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};
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/* Phone Mixer */
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static const struct snd_kcontrol_new wm9712_phone_mixer_controls[] = {
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SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 7, 1, 1),
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SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 7, 1, 1),
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SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 13, 1, 1),
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SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 13, 1, 1),
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SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_MIC, 14, 1, 1),
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SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_MIC, 13, 1, 1),
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};
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/* ALC headphone mux */
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static const struct snd_kcontrol_new wm9712_alc_mux_controls =
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SOC_DAPM_ENUM("Route", wm9712_enum[1]);
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/* out 3 mux */
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static const struct snd_kcontrol_new wm9712_out3_mux_controls =
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SOC_DAPM_ENUM("Route", wm9712_enum[2]);
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/* spk mux */
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static const struct snd_kcontrol_new wm9712_spk_mux_controls =
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SOC_DAPM_ENUM("Route", wm9712_enum[3]);
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/* Capture to Phone mux */
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static const struct snd_kcontrol_new wm9712_capture_phone_mux_controls =
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SOC_DAPM_ENUM("Route", wm9712_enum[4]);
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/* Capture left select */
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static const struct snd_kcontrol_new wm9712_capture_selectl_controls =
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SOC_DAPM_ENUM("Route", wm9712_enum[8]);
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/* Capture right select */
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static const struct snd_kcontrol_new wm9712_capture_selectr_controls =
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SOC_DAPM_ENUM("Route", wm9712_enum[9]);
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/* Mic select */
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static const struct snd_kcontrol_new wm9712_mic_src_controls =
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SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]);
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/* diff select */
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static const struct snd_kcontrol_new wm9712_diff_sel_controls =
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SOC_DAPM_ENUM("Route", wm9712_enum[11]);
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static const struct snd_soc_dapm_widget wm9712_dapm_widgets[] = {
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SND_SOC_DAPM_MUX("ALC Sidetone Mux", SND_SOC_NOPM, 0, 0,
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&wm9712_alc_mux_controls),
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SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0,
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&wm9712_out3_mux_controls),
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SND_SOC_DAPM_MUX("Speaker Mux", SND_SOC_NOPM, 0, 0,
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&wm9712_spk_mux_controls),
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SND_SOC_DAPM_MUX("Capture Phone Mux", SND_SOC_NOPM, 0, 0,
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&wm9712_capture_phone_mux_controls),
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SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0,
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&wm9712_capture_selectl_controls),
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SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0,
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&wm9712_capture_selectr_controls),
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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;
|
|
|
|
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)) {
|
|
ret = PTR_ERR(regmap);
|
|
goto err_free_ac97_codec;
|
|
}
|
|
|
|
snd_soc_codec_init_regmap(codec, regmap);
|
|
|
|
/* set alc mux to none */
|
|
snd_soc_update_bits(codec, AC97_VIDEO, 0x3000, 0x3000);
|
|
|
|
return 0;
|
|
err_free_ac97_codec:
|
|
snd_soc_free_ac97_codec(wm9712->ac97);
|
|
return ret;
|
|
}
|
|
|
|
static int wm9712_soc_remove(struct snd_soc_codec *codec)
|
|
{
|
|
struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec);
|
|
|
|
snd_soc_codec_exit_regmap(codec);
|
|
snd_soc_free_ac97_codec(wm9712->ac97);
|
|
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);
|
|
|
|
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");
|