mirror of
https://github.com/AuxXxilium/linux_dsm_epyc7002.git
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d71f4cece4
Conflicts: sound/soc/codecs/ad1938.c
740 lines
24 KiB
C
740 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 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 <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/soc-dapm.h>
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#include "wm9712.h"
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#define WM9712_VERSION "0.4"
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static unsigned int ac97_read(struct snd_soc_codec *codec,
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unsigned int reg);
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static int ac97_write(struct snd_soc_codec *codec,
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unsigned int reg, unsigned int val);
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/*
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* WM9712 register cache
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*/
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static const u16 wm9712_reg[] = {
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0x6174, 0x8000, 0x8000, 0x8000, /* 6 */
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0x0f0f, 0xaaa0, 0xc008, 0x6808, /* e */
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0xe808, 0xaaa0, 0xad00, 0x8000, /* 16 */
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0xe808, 0x3000, 0x8000, 0x0000, /* 1e */
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0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
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0x0405, 0x0410, 0xbb80, 0xbb80, /* 2e */
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0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
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0x0000, 0x2000, 0x0000, 0x0000, /* 3e */
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0x0000, 0x0000, 0x0000, 0x0000, /* 46 */
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0x0000, 0x0000, 0xf83e, 0xffff, /* 4e */
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0x0000, 0x0000, 0x0000, 0xf83e, /* 56 */
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0x0008, 0x0000, 0x0000, 0x0000, /* 5e */
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0xb032, 0x3e00, 0x0000, 0x0000, /* 66 */
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0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
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0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
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0x0001, 0x0000, 0x574d, 0x4c12, /* 7e */
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0x0000, 0x0000 /* virtual hp mixers */
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};
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/* virtual HP mixers regs */
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#define HPL_MIXER 0x80
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#define HPR_MIXER 0x82
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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 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("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0),
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SOC_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1),
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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 ADC 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, 1),
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SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
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SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
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SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
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SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0),
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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 mixer_event(struct snd_soc_dapm_widget *w,
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struct snd_kcontrol *k, int event)
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{
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u16 l, r, beep, line, phone, mic, pcm, aux;
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l = ac97_read(w->codec, HPL_MIXER);
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r = ac97_read(w->codec, HPR_MIXER);
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beep = ac97_read(w->codec, AC97_PC_BEEP);
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mic = ac97_read(w->codec, AC97_VIDEO);
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phone = ac97_read(w->codec, AC97_PHONE);
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line = ac97_read(w->codec, AC97_LINE);
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pcm = ac97_read(w->codec, AC97_PCM);
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aux = ac97_read(w->codec, AC97_CD);
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if (l & 0x1 || r & 0x1)
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ac97_write(w->codec, AC97_VIDEO, mic & 0x7fff);
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else
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ac97_write(w->codec, AC97_VIDEO, mic | 0x8000);
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if (l & 0x2 || r & 0x2)
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ac97_write(w->codec, AC97_PCM, pcm & 0x7fff);
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else
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ac97_write(w->codec, AC97_PCM, pcm | 0x8000);
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if (l & 0x4 || r & 0x4)
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ac97_write(w->codec, AC97_LINE, line & 0x7fff);
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else
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ac97_write(w->codec, AC97_LINE, line | 0x8000);
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if (l & 0x8 || r & 0x8)
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ac97_write(w->codec, AC97_PHONE, phone & 0x7fff);
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else
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ac97_write(w->codec, AC97_PHONE, phone | 0x8000);
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if (l & 0x10 || r & 0x10)
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ac97_write(w->codec, AC97_CD, aux & 0x7fff);
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else
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ac97_write(w->codec, AC97_CD, aux | 0x8000);
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if (l & 0x20 || r & 0x20)
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ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff);
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else
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ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000);
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return 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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SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPL_MIXER, 5, 1, 0),
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SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 4, 1, 0),
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SOC_DAPM_SINGLE("Phone Bypass Switch", HPL_MIXER, 3, 1, 0),
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SOC_DAPM_SINGLE("Line Bypass Switch", HPL_MIXER, 2, 1, 0),
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SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 1, 1, 0),
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SOC_DAPM_SINGLE("Mic Sidetone Switch", HPL_MIXER, 0, 1, 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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SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPR_MIXER, 5, 1, 0),
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SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 4, 1, 0),
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SOC_DAPM_SINGLE("Phone Bypass Switch", HPR_MIXER, 3, 1, 0),
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SOC_DAPM_SINGLE("Line Bypass Switch", HPR_MIXER, 2, 1, 0),
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SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 1, 1, 0),
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SOC_DAPM_SINGLE("Mic Sidetone Switch", HPR_MIXER, 0, 1, 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("Route", 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("Mic Select Source", SND_SOC_NOPM, 0, 0,
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&wm9712_mic_src_controls),
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SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0,
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&wm9712_diff_sel_controls),
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SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
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SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_INT_PAGING, 9, 1,
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&wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls),
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mixer_event, SND_SOC_DAPM_POST_REG),
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SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_INT_PAGING, 8, 1,
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&wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls),
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mixer_event, SND_SOC_DAPM_POST_REG),
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SND_SOC_DAPM_MIXER("Phone Mixer", AC97_INT_PAGING, 6, 1,
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&wm9712_phone_mixer_controls[0], ARRAY_SIZE(wm9712_phone_mixer_controls)),
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SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_INT_PAGING, 7, 1,
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&wm9712_speaker_mixer_controls[0],
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ARRAY_SIZE(wm9712_speaker_mixer_controls)),
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SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
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SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_INT_PAGING, 14, 1),
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SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_INT_PAGING, 13, 1),
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SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", SND_SOC_NOPM, 0, 0),
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SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_INT_PAGING, 12, 1),
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SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_INT_PAGING, 11, 1),
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SND_SOC_DAPM_PGA("Headphone PGA", AC97_INT_PAGING, 4, 1, NULL, 0),
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SND_SOC_DAPM_PGA("Speaker PGA", AC97_INT_PAGING, 3, 1, NULL, 0),
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SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0),
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SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0),
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SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0),
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SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0),
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SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1),
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SND_SOC_DAPM_OUTPUT("MONOOUT"),
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SND_SOC_DAPM_OUTPUT("HPOUTL"),
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SND_SOC_DAPM_OUTPUT("HPOUTR"),
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SND_SOC_DAPM_OUTPUT("LOUT2"),
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SND_SOC_DAPM_OUTPUT("ROUT2"),
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SND_SOC_DAPM_OUTPUT("OUT3"),
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SND_SOC_DAPM_INPUT("LINEINL"),
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SND_SOC_DAPM_INPUT("LINEINR"),
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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 audio_map[] = {
|
|
/* virtual mixer - mixes left & right channels for spk and mono */
|
|
{"AC97 Mixer", NULL, "Left DAC"},
|
|
{"AC97 Mixer", NULL, "Right DAC"},
|
|
|
|
/* Left HP mixer */
|
|
{"Left HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
|
|
{"Left HP Mixer", "Aux Playback Switch", "Aux DAC"},
|
|
{"Left HP Mixer", "Phone Bypass Switch", "Phone PGA"},
|
|
{"Left HP Mixer", "Line Bypass Switch", "Line PGA"},
|
|
{"Left HP Mixer", "PCM Playback Switch", "Left DAC"},
|
|
{"Left HP Mixer", "Mic Sidetone Switch", "Mic PGA"},
|
|
{"Left HP Mixer", NULL, "ALC Sidetone Mux"},
|
|
|
|
/* Right HP mixer */
|
|
{"Right HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
|
|
{"Right HP Mixer", "Aux Playback Switch", "Aux DAC"},
|
|
{"Right HP Mixer", "Phone Bypass Switch", "Phone PGA"},
|
|
{"Right HP Mixer", "Line Bypass Switch", "Line PGA"},
|
|
{"Right HP Mixer", "PCM Playback Switch", "Right DAC"},
|
|
{"Right HP Mixer", "Mic Sidetone Switch", "Mic PGA"},
|
|
{"Right HP Mixer", NULL, "ALC Sidetone Mux"},
|
|
|
|
/* speaker mixer */
|
|
{"Speaker Mixer", "PCBeep Bypass Switch", "PCBEEP"},
|
|
{"Speaker Mixer", "Line Bypass Switch", "Line PGA"},
|
|
{"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"},
|
|
{"Speaker Mixer", "Phone Bypass Switch", "Phone PGA"},
|
|
{"Speaker Mixer", "Aux Playback Switch", "Aux DAC"},
|
|
|
|
/* Phone mixer */
|
|
{"Phone Mixer", "PCBeep Bypass Switch", "PCBEEP"},
|
|
{"Phone Mixer", "Line Bypass Switch", "Line PGA"},
|
|
{"Phone Mixer", "Aux Playback Switch", "Aux DAC"},
|
|
{"Phone Mixer", "PCM Playback Switch", "AC97 Mixer"},
|
|
{"Phone Mixer", "Mic 1 Sidetone Switch", "Mic PGA"},
|
|
{"Phone Mixer", "Mic 2 Sidetone Switch", "Mic PGA"},
|
|
|
|
/* inputs */
|
|
{"Line PGA", NULL, "LINEINL"},
|
|
{"Line PGA", NULL, "LINEINR"},
|
|
{"Phone PGA", NULL, "PHONE"},
|
|
{"Mic PGA", NULL, "MIC1"},
|
|
{"Mic PGA", NULL, "MIC2"},
|
|
|
|
/* left capture selector */
|
|
{"Left Capture Select", "Mic", "MIC1"},
|
|
{"Left Capture Select", "Speaker Mixer", "Speaker Mixer"},
|
|
{"Left Capture Select", "Line", "LINEINL"},
|
|
{"Left Capture Select", "Headphone Mixer", "Left HP Mixer"},
|
|
{"Left Capture Select", "Phone Mixer", "Phone Mixer"},
|
|
{"Left Capture Select", "Phone", "PHONE"},
|
|
|
|
/* right capture selector */
|
|
{"Right Capture Select", "Mic", "MIC2"},
|
|
{"Right Capture Select", "Speaker Mixer", "Speaker Mixer"},
|
|
{"Right Capture Select", "Line", "LINEINR"},
|
|
{"Right Capture Select", "Headphone Mixer", "Right HP Mixer"},
|
|
{"Right Capture Select", "Phone Mixer", "Phone Mixer"},
|
|
{"Right Capture Select", "Phone", "PHONE"},
|
|
|
|
/* ALC Sidetone */
|
|
{"ALC Sidetone Mux", "Stereo", "Left Capture Select"},
|
|
{"ALC Sidetone Mux", "Stereo", "Right Capture Select"},
|
|
{"ALC Sidetone Mux", "Left", "Left Capture Select"},
|
|
{"ALC Sidetone Mux", "Right", "Right Capture Select"},
|
|
|
|
/* ADC's */
|
|
{"Left ADC", NULL, "Left Capture Select"},
|
|
{"Right ADC", NULL, "Right Capture Select"},
|
|
|
|
/* outputs */
|
|
{"MONOOUT", NULL, "Phone Mixer"},
|
|
{"HPOUTL", NULL, "Headphone PGA"},
|
|
{"Headphone PGA", NULL, "Left HP Mixer"},
|
|
{"HPOUTR", NULL, "Headphone PGA"},
|
|
{"Headphone PGA", NULL, "Right HP Mixer"},
|
|
|
|
/* mono mixer */
|
|
{"Mono Mixer", NULL, "Left HP Mixer"},
|
|
{"Mono Mixer", NULL, "Right HP Mixer"},
|
|
|
|
/* Out3 Mux */
|
|
{"Out3 Mux", "Left", "Left HP Mixer"},
|
|
{"Out3 Mux", "Mono", "Phone Mixer"},
|
|
{"Out3 Mux", "Left + Right", "Mono Mixer"},
|
|
{"Out 3 PGA", NULL, "Out3 Mux"},
|
|
{"OUT3", NULL, "Out 3 PGA"},
|
|
|
|
/* speaker Mux */
|
|
{"Speaker Mux", "Speaker Mix", "Speaker Mixer"},
|
|
{"Speaker Mux", "Headphone Mix", "Mono Mixer"},
|
|
{"Speaker PGA", NULL, "Speaker Mux"},
|
|
{"LOUT2", NULL, "Speaker PGA"},
|
|
{"ROUT2", NULL, "Speaker PGA"},
|
|
};
|
|
|
|
static int wm9712_add_widgets(struct snd_soc_codec *codec)
|
|
{
|
|
snd_soc_dapm_new_controls(codec, wm9712_dapm_widgets,
|
|
ARRAY_SIZE(wm9712_dapm_widgets));
|
|
|
|
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
|
|
|
|
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_REC_GAIN)
|
|
return soc_ac97_ops.read(codec->ac97, reg);
|
|
else {
|
|
reg = reg >> 1;
|
|
|
|
if (reg >= (ARRAY_SIZE(wm9712_reg)))
|
|
return -EIO;
|
|
|
|
return cache[reg];
|
|
}
|
|
}
|
|
|
|
static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
|
|
unsigned int val)
|
|
{
|
|
u16 *cache = codec->reg_cache;
|
|
|
|
if (reg < 0x7c)
|
|
soc_ac97_ops.write(codec->ac97, reg, val);
|
|
reg = reg >> 1;
|
|
if (reg < (ARRAY_SIZE(wm9712_reg)))
|
|
cache[reg] = val;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int ac97_prepare(struct snd_pcm_substream *substream,
|
|
struct snd_soc_dai *dai)
|
|
{
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
|
struct snd_soc_device *socdev = rtd->socdev;
|
|
struct snd_soc_codec *codec = socdev->card->codec;
|
|
int reg;
|
|
u16 vra;
|
|
|
|
vra = ac97_read(codec, AC97_EXTENDED_STATUS);
|
|
ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
|
|
|
|
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
|
|
reg = AC97_PCM_FRONT_DAC_RATE;
|
|
else
|
|
reg = AC97_PCM_LR_ADC_RATE;
|
|
|
|
return ac97_write(codec, reg, runtime->rate);
|
|
}
|
|
|
|
static int ac97_aux_prepare(struct snd_pcm_substream *substream,
|
|
struct snd_soc_dai *dai)
|
|
{
|
|
struct snd_pcm_runtime *runtime = substream->runtime;
|
|
struct snd_soc_pcm_runtime *rtd = substream->private_data;
|
|
struct snd_soc_device *socdev = rtd->socdev;
|
|
struct snd_soc_codec *codec = socdev->card->codec;
|
|
u16 vra, xsle;
|
|
|
|
vra = ac97_read(codec, AC97_EXTENDED_STATUS);
|
|
ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
|
|
xsle = ac97_read(codec, AC97_PCI_SID);
|
|
ac97_write(codec, AC97_PCI_SID, xsle | 0x8000);
|
|
|
|
if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
|
|
return -ENODEV;
|
|
|
|
return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate);
|
|
}
|
|
|
|
#define WM9712_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
|
|
SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
|
|
SNDRV_PCM_RATE_48000)
|
|
|
|
static struct snd_soc_dai_ops wm9712_dai_ops_hifi = {
|
|
.prepare = ac97_prepare,
|
|
};
|
|
|
|
static struct snd_soc_dai_ops wm9712_dai_ops_aux = {
|
|
.prepare = ac97_aux_prepare,
|
|
};
|
|
|
|
struct snd_soc_dai wm9712_dai[] = {
|
|
{
|
|
.name = "AC97 HiFi",
|
|
.ac97_control = 1,
|
|
.playback = {
|
|
.stream_name = "HiFi Playback",
|
|
.channels_min = 1,
|
|
.channels_max = 2,
|
|
.rates = WM9712_AC97_RATES,
|
|
.formats = SND_SOC_STD_AC97_FMTS,},
|
|
.capture = {
|
|
.stream_name = "HiFi Capture",
|
|
.channels_min = 1,
|
|
.channels_max = 2,
|
|
.rates = WM9712_AC97_RATES,
|
|
.formats = SND_SOC_STD_AC97_FMTS,},
|
|
.ops = &wm9712_dai_ops_hifi,
|
|
},
|
|
{
|
|
.name = "AC97 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,
|
|
}
|
|
};
|
|
EXPORT_SYMBOL_GPL(wm9712_dai);
|
|
|
|
static int wm9712_set_bias_level(struct snd_soc_codec *codec,
|
|
enum snd_soc_bias_level level)
|
|
{
|
|
switch (level) {
|
|
case SND_SOC_BIAS_ON:
|
|
case SND_SOC_BIAS_PREPARE:
|
|
break;
|
|
case SND_SOC_BIAS_STANDBY:
|
|
ac97_write(codec, AC97_POWERDOWN, 0x0000);
|
|
break;
|
|
case SND_SOC_BIAS_OFF:
|
|
/* disable everything including AC link */
|
|
ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
|
|
ac97_write(codec, AC97_POWERDOWN, 0xffff);
|
|
break;
|
|
}
|
|
codec->bias_level = level;
|
|
return 0;
|
|
}
|
|
|
|
static int wm9712_reset(struct snd_soc_codec *codec, int try_warm)
|
|
{
|
|
if (try_warm && soc_ac97_ops.warm_reset) {
|
|
soc_ac97_ops.warm_reset(codec->ac97);
|
|
if (ac97_read(codec, 0) == wm9712_reg[0])
|
|
return 1;
|
|
}
|
|
|
|
soc_ac97_ops.reset(codec->ac97);
|
|
if (soc_ac97_ops.warm_reset)
|
|
soc_ac97_ops.warm_reset(codec->ac97);
|
|
if (ac97_read(codec, 0) != wm9712_reg[0])
|
|
goto err;
|
|
return 0;
|
|
|
|
err:
|
|
printk(KERN_ERR "WM9712 AC97 reset failed\n");
|
|
return -EIO;
|
|
}
|
|
|
|
static int wm9712_soc_suspend(struct platform_device *pdev,
|
|
pm_message_t state)
|
|
{
|
|
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
|
|
struct snd_soc_codec *codec = socdev->card->codec;
|
|
|
|
wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF);
|
|
return 0;
|
|
}
|
|
|
|
static int wm9712_soc_resume(struct platform_device *pdev)
|
|
{
|
|
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
|
|
struct snd_soc_codec *codec = socdev->card->codec;
|
|
int i, ret;
|
|
u16 *cache = codec->reg_cache;
|
|
|
|
ret = wm9712_reset(codec, 1);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "could not reset AC97 codec\n");
|
|
return ret;
|
|
}
|
|
|
|
wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
|
|
|
|
if (ret == 0) {
|
|
/* Sync reg_cache with the hardware after cold reset */
|
|
for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i += 2) {
|
|
if (i == AC97_INT_PAGING || i == AC97_POWERDOWN ||
|
|
(i > 0x58 && i != 0x5c))
|
|
continue;
|
|
soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
|
|
}
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static int wm9712_soc_probe(struct platform_device *pdev)
|
|
{
|
|
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
|
|
struct snd_soc_codec *codec;
|
|
int ret = 0;
|
|
|
|
printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION);
|
|
|
|
socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec),
|
|
GFP_KERNEL);
|
|
if (socdev->card->codec == NULL)
|
|
return -ENOMEM;
|
|
codec = socdev->card->codec;
|
|
mutex_init(&codec->mutex);
|
|
|
|
codec->reg_cache = kmemdup(wm9712_reg, sizeof(wm9712_reg), GFP_KERNEL);
|
|
|
|
if (codec->reg_cache == NULL) {
|
|
ret = -ENOMEM;
|
|
goto cache_err;
|
|
}
|
|
codec->reg_cache_size = sizeof(wm9712_reg);
|
|
codec->reg_cache_step = 2;
|
|
|
|
codec->name = "WM9712";
|
|
codec->owner = THIS_MODULE;
|
|
codec->dai = wm9712_dai;
|
|
codec->num_dai = ARRAY_SIZE(wm9712_dai);
|
|
codec->write = ac97_write;
|
|
codec->read = ac97_read;
|
|
codec->set_bias_level = wm9712_set_bias_level;
|
|
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) {
|
|
printk(KERN_ERR "wm9712: failed to register AC97 codec\n");
|
|
goto codec_err;
|
|
}
|
|
|
|
/* register pcms */
|
|
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
|
|
if (ret < 0)
|
|
goto pcm_err;
|
|
|
|
ret = wm9712_reset(codec, 0);
|
|
if (ret < 0) {
|
|
printk(KERN_ERR "Failed to reset WM9712: AC97 link error\n");
|
|
goto reset_err;
|
|
}
|
|
|
|
/* set alc mux to none */
|
|
ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
|
|
|
|
wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
|
|
snd_soc_add_controls(codec, wm9712_snd_ac97_controls,
|
|
ARRAY_SIZE(wm9712_snd_ac97_controls));
|
|
wm9712_add_widgets(codec);
|
|
|
|
return 0;
|
|
|
|
reset_err:
|
|
snd_soc_free_pcms(socdev);
|
|
pcm_err:
|
|
snd_soc_free_ac97_codec(codec);
|
|
|
|
codec_err:
|
|
kfree(codec->reg_cache);
|
|
|
|
cache_err:
|
|
kfree(socdev->card->codec);
|
|
socdev->card->codec = NULL;
|
|
return ret;
|
|
}
|
|
|
|
static int wm9712_soc_remove(struct platform_device *pdev)
|
|
{
|
|
struct snd_soc_device *socdev = platform_get_drvdata(pdev);
|
|
struct snd_soc_codec *codec = socdev->card->codec;
|
|
|
|
if (codec == NULL)
|
|
return 0;
|
|
|
|
snd_soc_dapm_free(socdev);
|
|
snd_soc_free_pcms(socdev);
|
|
snd_soc_free_ac97_codec(codec);
|
|
kfree(codec->reg_cache);
|
|
kfree(codec);
|
|
return 0;
|
|
}
|
|
|
|
struct snd_soc_codec_device soc_codec_dev_wm9712 = {
|
|
.probe = wm9712_soc_probe,
|
|
.remove = wm9712_soc_remove,
|
|
.suspend = wm9712_soc_suspend,
|
|
.resume = wm9712_soc_resume,
|
|
};
|
|
EXPORT_SYMBOL_GPL(soc_codec_dev_wm9712);
|
|
|
|
MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver");
|
|
MODULE_AUTHOR("Liam Girdwood");
|
|
MODULE_LICENSE("GPL");
|