/* * wm9712.c -- ALSA Soc WM9712 codec support * * Copyright 2006-12 Wolfson Microelectronics PLC. * Author: Liam Girdwood * * 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 #include #include #include #include #include #include #include #include #include #include #include "wm9712.h" struct wm9712_priv { struct snd_ac97 *ac97; unsigned int hp_mixer[2]; struct mutex lock; }; static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg); static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val); /* * WM9712 register cache */ static const u16 wm9712_reg[] = { 0x6174, 0x8000, 0x8000, 0x8000, /* 6 */ 0x0f0f, 0xaaa0, 0xc008, 0x6808, /* e */ 0xe808, 0xaaa0, 0xad00, 0x8000, /* 16 */ 0xe808, 0x3000, 0x8000, 0x0000, /* 1e */ 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */ 0x0405, 0x0410, 0xbb80, 0xbb80, /* 2e */ 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */ 0x0000, 0x2000, 0x0000, 0x0000, /* 3e */ 0x0000, 0x0000, 0x0000, 0x0000, /* 46 */ 0x0000, 0x0000, 0xf83e, 0xffff, /* 4e */ 0x0000, 0x0000, 0x0000, 0xf83e, /* 56 */ 0x0008, 0x0000, 0x0000, 0x0000, /* 5e */ 0xb032, 0x3e00, 0x0000, 0x0000, /* 66 */ 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */ 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */ 0x0001, 0x0000, 0x574d, 0x4c12, /* 7e */ }; #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.enumerated.item[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; 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.enumerated.item[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.enumerated.item[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 unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); 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(wm9712->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) { struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); u16 *cache = codec->reg_cache; soc_ac97_ops->write(wm9712->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_soc_codec *codec = dai->codec; int reg; u16 vra; struct snd_pcm_runtime *runtime = substream->runtime; 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_soc_codec *codec = dai->codec; u16 vra, xsle; struct snd_pcm_runtime *runtime = substream->runtime; vra = ac97_read(codec, AC97_EXTENDED_STATUS); ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); xsle = ac97_read(codec, AC97_PCI_SID); ac97_write(codec, AC97_PCI_SID, xsle | 0x8000); if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); } #define WM9712_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ SNDRV_PCM_RATE_48000) static const struct snd_soc_dai_ops wm9712_dai_ops_hifi = { .prepare = ac97_prepare, }; static const struct snd_soc_dai_ops wm9712_dai_ops_aux = { .prepare = ac97_aux_prepare, }; static struct snd_soc_dai_driver wm9712_dai[] = { { .name = "wm9712-hifi", .playback = { .stream_name = "HiFi Playback", .channels_min = 1, .channels_max = 2, .rates = WM9712_AC97_RATES, .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "HiFi Capture", .channels_min = 1, .channels_max = 2, .rates = WM9712_AC97_RATES, .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &wm9712_dai_ops_hifi, }, { .name = "wm9712-aux", .playback = { .stream_name = "Aux Playback", .channels_min = 1, .channels_max = 1, .rates = WM9712_AC97_RATES, .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &wm9712_dai_ops_aux, } }; static int wm9712_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: ac97_write(codec, AC97_POWERDOWN, 0x0000); break; case SND_SOC_BIAS_OFF: /* disable everything including AC link */ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); ac97_write(codec, AC97_POWERDOWN, 0xffff); break; } codec->dapm.bias_level = level; return 0; } static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); if (try_warm && soc_ac97_ops->warm_reset) { soc_ac97_ops->warm_reset(wm9712->ac97); if (ac97_read(codec, 0) == wm9712_reg[0]) return 1; } soc_ac97_ops->reset(wm9712->ac97); if (soc_ac97_ops->warm_reset) soc_ac97_ops->warm_reset(wm9712->ac97); if (ac97_read(codec, 0) != wm9712_reg[0]) goto err; return 0; err: dev_err(codec->dev, "Failed to reset: AC97 link error\n"); return -EIO; } static int wm9712_soc_suspend(struct snd_soc_codec *codec) { wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static int wm9712_soc_resume(struct snd_soc_codec *codec) { struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); int i, ret; u16 *cache = codec->reg_cache; ret = wm9712_reset(codec, 1); if (ret < 0) 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(wm9712->ac97, i, cache[i>>1]); } } return ret; } static int wm9712_soc_probe(struct snd_soc_codec *codec) { struct wm9712_priv *wm9712 = snd_soc_codec_get_drvdata(codec); int ret = 0; wm9712->ac97 = snd_soc_new_ac97_codec(codec); if (IS_ERR(wm9712->ac97)) { ret = PTR_ERR(wm9712->ac97); dev_err(codec->dev, "Failed to register AC97 codec: %d\n", ret); return ret; } ret = wm9712_reset(codec, 0); if (ret < 0) 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); return 0; reset_err: 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_free_ac97_codec(wm9712->ac97); return 0; } static struct snd_soc_codec_driver soc_codec_dev_wm9712 = { .probe = wm9712_soc_probe, .remove = wm9712_soc_remove, .suspend = wm9712_soc_suspend, .resume = wm9712_soc_resume, .read = ac97_read, .write = ac97_write, .set_bias_level = wm9712_set_bias_level, .reg_cache_size = ARRAY_SIZE(wm9712_reg), .reg_word_size = sizeof(u16), .reg_cache_step = 2, .reg_cache_default = wm9712_reg, .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", .owner = THIS_MODULE, }, .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");