From 6f4bc952c60b26ecfcb013fb9a7e9474023e046e Mon Sep 17 00:00:00 2001 From: "Arnaud Patard (Rtp)" Date: Thu, 21 Oct 2010 19:40:02 +0200 Subject: [PATCH 01/10] ASoC: add support for alc562[123] codecs This patch is adding support for alc562[123] codecs. It's based on the source code available in HP source code and other places. Signed-off-by: Arnaud Patard Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/alc5623.h | 15 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/alc5623.c | 1118 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/alc5623.h | 161 ++++++ 5 files changed, 1301 insertions(+) create mode 100644 include/sound/alc5623.h create mode 100644 sound/soc/codecs/alc5623.c create mode 100644 sound/soc/codecs/alc5623.h diff --git a/include/sound/alc5623.h b/include/sound/alc5623.h new file mode 100644 index 000000000000..422c97d43df3 --- /dev/null +++ b/include/sound/alc5623.h @@ -0,0 +1,15 @@ +#ifndef _INCLUDE_SOUND_ALC5623_H +#define _INCLUDE_SOUND_ALC5623_H +struct alc5623_platform_data { + /* configure : */ + /* Lineout/Speaker Amps Vmid ratio control */ + /* enable/disable adc/dac high pass filters */ + unsigned int add_ctrl; + /* configure : */ + /* output to enable when jack is low */ + /* output to enable when jack is high */ + /* jack detect (gpio/nc/jack detect [12] */ + unsigned int jack_det_ctrl; +}; +#endif + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 94a9d06b9027..658cbe07fb72 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -22,6 +22,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_ALC562 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C @@ -129,6 +130,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_ALC5623 + tristate + config SND_SOC_CQ0093VC tristate @@ -317,3 +321,4 @@ config SND_SOC_WM2000 config SND_SOC_WM9090 tristate + diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f67a2d6f7a46..0dcaed3e73f3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -17,6 +17,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-alc5623-objs := alc5623.o snd-soc-spdif-objs := spdif_transciever.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-stac9766-objs := stac9766.o @@ -92,6 +93,7 @@ obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c new file mode 100644 index 000000000000..fac61744f8c7 --- /dev/null +++ b/sound/soc/codecs/alc5623.c @@ -0,0 +1,1118 @@ +/* + * alc5623.c -- alc562[123] ALSA Soc Audio driver + * + * Copyright 2008 Realtek Microelectronics + * Author: flove Ethan + * + * Copyright 2010 Arnaud Patard + * + * + * Based on WM8753.c + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "alc5623.h" + +static int caps_charge = 2000; +module_param(caps_charge, int, 0); +MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)"); + +/* codec private data */ +struct alc5623_priv { + enum snd_soc_control_type control_type; + void *control_data; + struct mutex mutex; + u8 id; + unsigned int sysclk; + u16 reg_cache[ALC5623_VENDOR_ID2+2]; + unsigned int add_ctrl; + unsigned int jack_det_ctrl; +}; + +static void alc5623_fill_cache(struct snd_soc_codec *codec) +{ + int i, step = codec->driver->reg_cache_step; + u16 *cache = codec->reg_cache; + + /* not really efficient ... */ + for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) + cache[i] = codec->hw_read(codec, i); +} + +static inline int alc5623_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, ALC5623_RESET, 0); +} + +static int amp_mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + /* to power-on/off class-d amp generators/speaker */ + /* need to write to 'index-46h' register : */ + /* so write index num (here 0x46) to reg 0x6a */ + /* and then 0xffff/0 to reg 0x6c */ + snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0); + break; + } + + return 0; +} + +/* + * ALC5623 Controls + */ + +static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0); +static const unsigned int boost_tlv[] = { + TLV_DB_RANGE_HEAD(3), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), +}; +static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = { + SOC_DOUBLE_TLV("Speaker Playback Volume", + ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Speaker Playback Switch", + ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("Headphone Playback Volume", + ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Headphone Playback Switch", + ALC5623_HP_OUT_VOL, 15, 7, 1, 1), +}; + +static const struct snd_kcontrol_new rt5622_vol_snd_controls[] = { + SOC_DOUBLE_TLV("Speaker Playback Volume", + ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Speaker Playback Switch", + ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("Line Playback Volume", + ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Line Playback Switch", + ALC5623_HP_OUT_VOL, 15, 7, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = { + SOC_DOUBLE_TLV("Line Playback Volume", + ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Line Playback Switch", + ALC5623_SPK_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("Headphone Playback Volume", + ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Headphone Playback Switch", + ALC5623_HP_OUT_VOL, 15, 7, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_snd_controls[] = { + SOC_DOUBLE_TLV("Auxout Playback Volume", + ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv), + SOC_DOUBLE("Auxout Playback Switch", + ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1), + SOC_DOUBLE_TLV("PCM Playback Volume", + ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("AuxI Capture Volume", + ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("LineIn Capture Volume", + ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv), + SOC_SINGLE_TLV("Mic1 Capture Volume", + ALC5623_MIC_VOL, 8, 31, 1, vol_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", + ALC5623_MIC_VOL, 0, 31, 1, vol_tlv), + SOC_DOUBLE_TLV("Rec Capture Volume", + ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv), + SOC_SINGLE_TLV("Mic 1 Boost Volume", + ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Mic 2 Boost Volume", + ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv), + SOC_SINGLE_TLV("Digital Boost Volume", + ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv), +}; + +/* + * DAPM Controls + */ +static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1), +SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1), +SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1), +SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1), +SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1), +SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1), +SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1), +SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1), +}; + +static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1), +SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1), +SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1), +}; + +/* Left Record Mixer */ +static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1), +SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1), +SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1), +SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1), +}; + +/* Right Record Mixer */ +static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = { +SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1), +SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1), +SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1), +SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1), +SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1), +SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1), +SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1), +}; + +static const char *alc5623_spk_n_sour_sel[] = { + "RN/-R", "RP/+R", "LN/-R", "Vmid" }; +static const char *alc5623_hpl_out_input_sel[] = { + "Vmid", "HP Left Mix"}; +static const char *alc5623_hpr_out_input_sel[] = { + "Vmid", "HP Right Mix"}; +static const char *alc5623_spkout_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; +static const char *alc5623_aux_out_input_sel[] = { + "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; + +/* auxout output mux */ +static const struct soc_enum alc5623_aux_out_input_enum = +SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel); +static const struct snd_kcontrol_new alc5623_auxout_mux_controls = +SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum); + +/* speaker output mux */ +static const struct soc_enum alc5623_spkout_input_enum = +SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel); +static const struct snd_kcontrol_new alc5623_spkout_mux_controls = +SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum); + +/* headphone left output mux */ +static const struct soc_enum alc5623_hpl_out_input_enum = +SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel); +static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls = +SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum); + +/* headphone right output mux */ +static const struct soc_enum alc5623_hpr_out_input_enum = +SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel); +static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls = +SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum); + +/* speaker output N select */ +static const struct soc_enum alc5623_spk_n_sour_enum = +SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel); +static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls = +SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum); + +static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = { +/* Muxes */ +SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0, + &alc5623_auxout_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0, + &alc5623_spkout_mux_controls), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5623_hpl_out_mux_controls), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, + &alc5623_hpr_out_mux_controls), +SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0, + &alc5623_spkoutn_mux_controls), + +/* output mixers */ +SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0, + &alc5623_hp_mixer_controls[0], + ARRAY_SIZE(alc5623_hp_mixer_controls)), +SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0, + &alc5623_hpr_mixer_controls[0], + ARRAY_SIZE(alc5623_hpr_mixer_controls)), +SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0, + &alc5623_hpl_mixer_controls[0], + ARRAY_SIZE(alc5623_hpl_mixer_controls)), +SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0, + &alc5623_mono_mixer_controls[0], + ARRAY_SIZE(alc5623_mono_mixer_controls)), +SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0, + &alc5623_speaker_mixer_controls[0], + ARRAY_SIZE(alc5623_speaker_mixer_controls)), + +/* input mixers */ +SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0, + &alc5623_captureL_mixer_controls[0], + ARRAY_SIZE(alc5623_captureL_mixer_controls)), +SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0, + &alc5623_captureR_mixer_controls[0], + ARRAY_SIZE(alc5623_captureR_mixer_controls)), + +SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", + ALC5623_PWR_MANAG_ADD2, 9, 0), +SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", + ALC5623_PWR_MANAG_ADD2, 8, 0), +SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0), +SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", + ALC5623_PWR_MANAG_ADD2, 7, 0), +SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", + ALC5623_PWR_MANAG_ADD2, 6, 0), +SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0), +SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0), +SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0), + +SND_SOC_DAPM_OUTPUT("AUXOUTL"), +SND_SOC_DAPM_OUTPUT("AUXOUTR"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_OUTPUT("SPKOUT"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_INPUT("LINEINL"), +SND_SOC_DAPM_INPUT("LINEINR"), +SND_SOC_DAPM_INPUT("AUXINL"), +SND_SOC_DAPM_INPUT("AUXINR"), +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2"), +SND_SOC_DAPM_VMID("Vmid"), +}; + +static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"}; +static const struct soc_enum alc5623_amp_enum = + SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names); +static const struct snd_kcontrol_new alc5623_amp_mux_controls = + SOC_DAPM_ENUM("Route", alc5623_amp_enum); + +static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = { +SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0, + amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0), +SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0, + &alc5623_amp_mux_controls), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* virtual mixer - mixes left & right channels */ + {"I2S Mix", NULL, "Left DAC"}, + {"I2S Mix", NULL, "Right DAC"}, + {"Line Mix", NULL, "Right LineIn"}, + {"Line Mix", NULL, "Left LineIn"}, + {"AuxI Mix", NULL, "Left AuxI"}, + {"AuxI Mix", NULL, "Right AuxI"}, + {"AUXOUTL", NULL, "Left AuxOut"}, + {"AUXOUTR", NULL, "Right AuxOut"}, + + /* HP mixer */ + {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"}, + {"HPL Mix", NULL, "HP Mix"}, + {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"}, + {"HPR Mix", NULL, "HP Mix"}, + {"HP Mix", "LI2HP Playback Switch", "Line Mix"}, + {"HP Mix", "AUXI2HP Playback Switch", "AuxI Mix"}, + {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"}, + {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"}, + {"HP Mix", "DAC2HP Playback Switch", "I2S Mix"}, + + /* speaker mixer */ + {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"}, + {"Speaker Mix", "AUXI2SPK Playback Switch", "AuxI Mix"}, + {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"}, + {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"}, + {"Speaker Mix", "DAC2SPK Playback Switch", "I2S Mix"}, + + /* mono mixer */ + {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"}, + {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"}, + {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"}, + {"Mono Mix", "AUXI2MONO Playback Switch", "AuxI Mix"}, + {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"}, + {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"}, + {"Mono Mix", "DAC2MONO Playback Switch", "I2S Mix"}, + + /* Left record mixer */ + {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"}, + {"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"}, + {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"}, + {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /*Right record mixer */ + {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"}, + {"Right Capture Mix", "Right AuxI Capture Switch", "AUXINR"}, + {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"}, + {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"}, + {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"}, + {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"}, + {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"}, + + /* headphone left mux */ + {"Left Headphone Mux", "HP Left Mix", "HPL Mix"}, + {"Left Headphone Mux", "Vmid", "Vmid"}, + + /* headphone right mux */ + {"Right Headphone Mux", "HP Right Mix", "HPR Mix"}, + {"Right Headphone Mux", "Vmid", "Vmid"}, + + /* speaker out mux */ + {"SpeakerOut Mux", "Vmid", "Vmid"}, + {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"}, + {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"}, + {"SpeakerOut Mux", "Mono Mix", "Mono Mix"}, + + /* Mono/Aux Out mux */ + {"AuxOut Mux", "Vmid", "Vmid"}, + {"AuxOut Mux", "HPOut Mix", "HPOut Mix"}, + {"AuxOut Mux", "Speaker Mix", "Speaker Mix"}, + {"AuxOut Mux", "Mono Mix", "Mono Mix"}, + + /* output pga */ + {"HPL", NULL, "Left Headphone"}, + {"Left Headphone", NULL, "Left Headphone Mux"}, + {"HPR", NULL, "Right Headphone"}, + {"Right Headphone", NULL, "Right Headphone Mux"}, + {"Left AuxOut", NULL, "AuxOut Mux"}, + {"Right AuxOut", NULL, "AuxOut Mux"}, + + /* input pga */ + {"Left LineIn", NULL, "LINEINL"}, + {"Right LineIn", NULL, "LINEINR"}, + {"Left AuxI", NULL, "AUXINL"}, + {"Right AuxI", NULL, "AUXINR"}, + {"MIC1 Pre Amp", NULL, "MIC1"}, + {"MIC2 Pre Amp", NULL, "MIC2"}, + {"MIC1 PGA", NULL, "MIC1 Pre Amp"}, + {"MIC2 PGA", NULL, "MIC2 Pre Amp"}, + + /* left ADC */ + {"Left ADC", NULL, "Left Capture Mix"}, + + /* right ADC */ + {"Right ADC", NULL, "Right Capture Mix"}, + + {"SpeakerOut N Mux", "RN/-R", "SpeakerOut"}, + {"SpeakerOut N Mux", "RP/+R", "SpeakerOut"}, + {"SpeakerOut N Mux", "LN/-R", "SpeakerOut"}, + {"SpeakerOut N Mux", "Vmid", "Vmid"}, + + {"SPKOUT", NULL, "SpeakerOut"}, + {"SPKOUTN", NULL, "SpeakerOut N Mux"}, +}; + +static const struct snd_soc_dapm_route intercon_spk[] = { + {"SpeakerOut", NULL, "SpeakerOut Mux"}, +}; + +static const struct snd_soc_dapm_route intercon_amp_spk[] = { + {"AB Amp", NULL, "SpeakerOut Mux"}, + {"D Amp", NULL, "SpeakerOut Mux"}, + {"AB-D Amp Mux", "AB Amp", "AB Amp"}, + {"AB-D Amp Mux", "D Amp", "D Amp"}, + {"SpeakerOut", NULL, "AB-D Amp Mux"}, +}; + +/* PLL divisors */ +struct _pll_div { + u32 pll_in; + u32 pll_out; + u16 regvalue; +}; + +/* Note : pll code from original alc5623 driver. Not sure of how good it is */ +/* usefull only for master mode */ +static const struct _pll_div codec_master_pll_div[] = { + + { 2048000, 8192000, 0x0ea0}, + { 3686400, 8192000, 0x4e27}, + { 12000000, 8192000, 0x456b}, + { 13000000, 8192000, 0x495f}, + { 13100000, 8192000, 0x0320}, + { 2048000, 11289600, 0xf637}, + { 3686400, 11289600, 0x2f22}, + { 12000000, 11289600, 0x3e2f}, + { 13000000, 11289600, 0x4d5b}, + { 13100000, 11289600, 0x363b}, + { 2048000, 16384000, 0x1ea0}, + { 3686400, 16384000, 0x9e27}, + { 12000000, 16384000, 0x452b}, + { 13000000, 16384000, 0x542f}, + { 13100000, 16384000, 0x03a0}, + { 2048000, 16934400, 0xe625}, + { 3686400, 16934400, 0x9126}, + { 12000000, 16934400, 0x4d2c}, + { 13000000, 16934400, 0x742f}, + { 13100000, 16934400, 0x3c27}, + { 2048000, 22579200, 0x2aa0}, + { 3686400, 22579200, 0x2f20}, + { 12000000, 22579200, 0x7e2f}, + { 13000000, 22579200, 0x742f}, + { 13100000, 22579200, 0x3c27}, + { 2048000, 24576000, 0x2ea0}, + { 3686400, 24576000, 0xee27}, + { 12000000, 24576000, 0x2915}, + { 13000000, 24576000, 0x772e}, + { 13100000, 24576000, 0x0d20}, +}; + +static const struct _pll_div codec_slave_pll_div[] = { + + { 1024000, 16384000, 0x3ea0}, + { 1411200, 22579200, 0x3ea0}, + { 1536000, 24576000, 0x3ea0}, + { 2048000, 16384000, 0x1ea0}, + { 2822400, 22579200, 0x1ea0}, + { 3072000, 24576000, 0x1ea0}, + +}; + +static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + int i; + struct snd_soc_codec *codec = codec_dai->codec; + int gbl_clk = 0, pll_div = 0; + u16 reg; + + if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK) + return -ENODEV; + + /* Disable PLL power */ + snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2, + ALC5623_PWR_ADD2_PLL, + 0); + + /* pll is not used in slave mode */ + reg = snd_soc_read(codec, ALC5623_DAI_CONTROL); + if (reg & ALC5623_DAI_SDP_SLAVE_MODE) + return 0; + + if (!freq_in || !freq_out) + return 0; + + switch (pll_id) { + case ALC5623_PLL_FR_MCLK: + for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) { + if (codec_master_pll_div[i].pll_in == freq_in + && codec_master_pll_div[i].pll_out == freq_out) { + /* PLL source from MCLK */ + pll_div = codec_master_pll_div[i].regvalue; + break; + } + } + break; + case ALC5623_PLL_FR_BCK: + for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) { + if (codec_slave_pll_div[i].pll_in == freq_in + && codec_slave_pll_div[i].pll_out == freq_out) { + /* PLL source from Bitclk */ + gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK; + pll_div = codec_slave_pll_div[i].regvalue; + break; + } + } + break; + default: + return -EINVAL; + } + + if (!pll_div) + return -EINVAL; + + snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk); + snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div); + snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2, + ALC5623_PWR_ADD2_PLL, + ALC5623_PWR_ADD2_PLL); + gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL; + snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk); + + return 0; +} + +struct _coeff_div { + u16 fs; + u16 regvalue; +}; + +/* codec hifi mclk (after PLL) clock divider coefficients */ +/* values inspired from column BCLK=32Fs of Appendix A table */ +static const struct _coeff_div coeff_div[] = { + {256*8, 0x3a69}, + {384*8, 0x3c6b}, + {256*4, 0x2a69}, + {384*4, 0x2c6b}, + {256*2, 0x1a69}, + {384*2, 0x1c6b}, + {256*1, 0x0a69}, + {384*1, 0x0c6b}, +}; + +static int get_coeff(struct snd_soc_codec *codec, int rate) +{ + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].fs * rate == alc5623->sysclk) + return i; + } + return -EINVAL; +} + +/* + * Clock after PLL and dividers + */ +static int alc5623_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case 8192000: + case 11289600: + case 12288000: + case 16384000: + case 16934400: + case 18432000: + case 22579200: + case 24576000: + alc5623->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = ALC5623_DAI_SDP_MASTER_MODE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iface = ALC5623_DAI_SDP_SLAVE_MODE; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= ALC5623_DAI_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface |= ALC5623_DAI_I2S_DF_RIGHT; + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ALC5623_DAI_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ALC5623_DAI_I2S_DF_PCM; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL; + break; + case SND_SOC_DAIFMT_NB_IF: + break; + default: + return -EINVAL; + } + + return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface); +} + +static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + int coeff, rate; + u16 iface; + + iface = snd_soc_read(codec, ALC5623_DAI_CONTROL); + iface &= ~ALC5623_DAI_I2S_DL_MASK; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + iface |= ALC5623_DAI_I2S_DL_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= ALC5623_DAI_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= ALC5623_DAI_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= ALC5623_DAI_I2S_DL_32; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_write(codec, ALC5623_DAI_CONTROL, iface); + rate = params_rate(params); + coeff = get_coeff(codec, rate); + if (coeff < 0) + return -EINVAL; + + coeff = coeff_div[coeff].regvalue; + dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n", + __func__, alc5623->sysclk, rate, coeff); + snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff); + + return 0; +} + +static int alc5623_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT; + u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute; + + if (mute) + mute_reg |= hp_mute; + + return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg); +} + +#define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \ + | ALC5623_PWR_ADD2_DAC_REF_CIR) + +#define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \ + | ALC5623_PWR_ADD3_MIC1_BOOST_AD) + +#define ALC5623_ADD1_POWER_EN \ + (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \ + | ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \ + | ALC5623_PWR_ADD1_HP_OUT_ENH_AMP) + +#define ALC5623_ADD1_POWER_EN_5622 \ + (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \ + | ALC5623_PWR_ADD1_HP_OUT_AMP) + +static void enable_power_depop(struct snd_soc_codec *codec) +{ + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + + snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1, + ALC5623_PWR_ADD1_SOFTGEN_EN, + ALC5623_PWR_ADD1_SOFTGEN_EN); + + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN); + + snd_soc_update_bits(codec, ALC5623_MISC_CTRL, + ALC5623_MISC_HP_DEPOP_MODE2_EN, + ALC5623_MISC_HP_DEPOP_MODE2_EN); + + msleep(500); + + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN); + + /* avoid writing '1' into 5622 reserved bits */ + if (alc5623->id == 0x22) + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, + ALC5623_ADD1_POWER_EN_5622); + else + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, + ALC5623_ADD1_POWER_EN); + + /* disable HP Depop2 */ + snd_soc_update_bits(codec, ALC5623_MISC_CTRL, + ALC5623_MISC_HP_DEPOP_MODE2_EN, + 0); + +} + +static int alc5623_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + enable_power_depop(codec); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + /* everything off except vref/vmid, */ + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, + ALC5623_PWR_ADD2_VREF); + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, + ALC5623_PWR_ADD3_MAIN_BIAS); + break; + case SND_SOC_BIAS_OFF: + /* everything off, dac mute, inactive */ + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0); + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0); + snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0); + break; + } + codec->bias_level = level; + return 0; +} + +#define ALC5623_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \ + | SNDRV_PCM_FMTBIT_S24_LE \ + | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops alc5623_dai_ops = { + .hw_params = alc5623_pcm_hw_params, + .digital_mute = alc5623_mute, + .set_fmt = alc5623_set_dai_fmt, + .set_sysclk = alc5623_set_dai_sysclk, + .set_pll = alc5623_set_dai_pll, +}; + +static struct snd_soc_dai_driver alc5623_dai = { + .name = "alc5623-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5623_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ALC5623_FORMATS,}, + + .ops = &alc5623_dai_ops, +}; + +static int alc5623_suspend(struct snd_soc_codec *codec, pm_message_t mesg) +{ + alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int alc5623_resume(struct snd_soc_codec *codec) +{ + int i, step = codec->driver->reg_cache_step; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 2 ; i < codec->driver->reg_cache_size ; i += step) + snd_soc_write(codec, i, cache[i]); + + alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* charge alc5623 caps */ + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) { + alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + codec->bias_level = SND_SOC_BIAS_ON; + alc5623_set_bias_level(codec, codec->bias_level); + } + + return 0; +} + +static int alc5623_probe(struct snd_soc_codec *codec) +{ + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + alc5623_reset(codec); + alc5623_fill_cache(codec); + + /* power on device */ + alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (alc5623->add_ctrl) { + snd_soc_write(codec, ALC5623_ADD_CTRL_REG, + alc5623->add_ctrl); + } + + if (alc5623->jack_det_ctrl) { + snd_soc_write(codec, ALC5623_JACK_DET_CTRL, + alc5623->jack_det_ctrl); + } + + switch (alc5623->id) { + default: + case 0x21: + snd_soc_add_controls(codec, rt5621_vol_snd_controls, + ARRAY_SIZE(rt5621_vol_snd_controls)); + break; + case 0x22: + snd_soc_add_controls(codec, rt5622_vol_snd_controls, + ARRAY_SIZE(rt5622_vol_snd_controls)); + break; + case 0x23: + snd_soc_add_controls(codec, alc5623_vol_snd_controls, + ARRAY_SIZE(alc5623_vol_snd_controls)); + break; + } + + snd_soc_add_controls(codec, alc5623_snd_controls, + ARRAY_SIZE(alc5623_snd_controls)); + + snd_soc_dapm_new_controls(codec, alc5623_dapm_widgets, + ARRAY_SIZE(alc5623_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + switch (alc5623->id) { + default: + case 0x21: + case 0x22: + snd_soc_dapm_new_controls(codec, alc5623_dapm_amp_widgets, + ARRAY_SIZE(alc5623_dapm_amp_widgets)); + snd_soc_dapm_add_routes(codec, intercon_amp_spk, + ARRAY_SIZE(intercon_amp_spk)); + break; + case 0x23: + snd_soc_dapm_add_routes(codec, intercon_spk, + ARRAY_SIZE(intercon_spk)); + break; + } + + return ret; +} + +/* power down chip */ +static int alc5623_remove(struct snd_soc_codec *codec) +{ + alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_device_alc5623 = { + .probe = alc5623_probe, + .remove = alc5623_remove, + .suspend = alc5623_suspend, + .resume = alc5623_resume, + .set_bias_level = alc5623_set_bias_level, + .reg_cache_size = ALC5623_VENDOR_ID2+2, + .reg_word_size = sizeof(u16), + .reg_cache_step = 2, +}; + +/* + * ALC5623 2 wire address is determined by A1 pin + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int alc5623_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct alc5623_platform_data *pdata; + struct alc5623_priv *alc5623; + int ret, vid1, vid2; + + vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1); + if (vid1 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } + vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); + + vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2); + if (vid2 < 0) { + dev_err(&client->dev, "failed to read I2C\n"); + return -EIO; + } + + if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { + dev_err(&client->dev, "unknown or wrong codec\n"); + dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n", + 0x10ec, id->driver_data, + vid1, vid2); + return -ENODEV; + } + + dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2); + + alc5623 = kzalloc(sizeof(struct alc5623_priv), GFP_KERNEL); + if (alc5623 == NULL) { + ret = -ENOMEM; + goto err; + } + + pdata = client->dev.platform_data; + if (pdata) { + alc5623->add_ctrl = pdata->add_ctrl; + alc5623->jack_det_ctrl = pdata->jack_det_ctrl; + } + + alc5623->id = vid2; + switch (alc5623->id) { + case 0x21: + alc5623_dai.name = "alc5621-hifi"; + break; + case 0x22: + alc5623_dai.name = "alc5622-hifi"; + break; + default: + case 0x23: + alc5623_dai.name = "alc5623-hifi"; + break; + } + + i2c_set_clientdata(client, alc5623); + alc5623->control_data = client; + alc5623->control_type = SND_SOC_I2C; + mutex_init(&alc5623->mutex); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_device_alc5623, &alc5623_dai, 1); + if (ret != 0) { + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + return 0; + +err: + return ret; +} + +static int alc5623_i2c_remove(struct i2c_client *client) +{ + struct alc5623_priv *alc5623 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + kfree(alc5623); + return 0; +} + +static const struct i2c_device_id alc5623_i2c_table[] = { + {"alc5621", 0x21}, + {"alc5622", 0x22}, + {"alc5623", 0x23}, + {} +}; +MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table); + +/* i2c codec control layer */ +static struct i2c_driver alc5623_i2c_driver = { + .driver = { + .name = "alc562x-codec", + .owner = THIS_MODULE, + }, + .probe = alc5623_i2c_probe, + .remove = __devexit_p(alc5623_i2c_remove), + .id_table = alc5623_i2c_table, +}; + +static int __init alc5623_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&alc5623_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver", __func__); + return ret; + } + + return ret; +} +module_init(alc5623_modinit); + +static void __exit alc5623_modexit(void) +{ + i2c_del_driver(&alc5623_i2c_driver); +} +module_exit(alc5623_modexit); + +MODULE_DESCRIPTION("ASoC alc5621/2/3 driver"); +MODULE_AUTHOR("Arnaud Patard "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/alc5623.h b/sound/soc/codecs/alc5623.h new file mode 100644 index 000000000000..f3d68260d425 --- /dev/null +++ b/sound/soc/codecs/alc5623.h @@ -0,0 +1,161 @@ +/* + * alc5623.h -- alc562[123] ALSA Soc Audio driver + * + * Copyright 2008 Realtek Microelectronics + * Copyright 2010 Arnaud Patard + * + * Author: flove + * Arnaud Patard + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef _ALC5623_H +#define _ALC5623_H + +#define ALC5623_RESET 0x00 +/* 5621 5622 5623 */ +/* speaker output vol 2 2 */ +/* line output vol 4 2 */ +/* HP output vol 4 0 4 */ +#define ALC5623_SPK_OUT_VOL 0x02 +#define ALC5623_HP_OUT_VOL 0x04 +#define ALC5623_MONO_AUX_OUT_VOL 0x06 +#define ALC5623_AUXIN_VOL 0x08 +#define ALC5623_LINE_IN_VOL 0x0A +#define ALC5623_STEREO_DAC_VOL 0x0C +#define ALC5623_MIC_VOL 0x0E +#define ALC5623_MIC_ROUTING_CTRL 0x10 +#define ALC5623_ADC_REC_GAIN 0x12 +#define ALC5623_ADC_REC_MIXER 0x14 +#define ALC5623_SOFT_VOL_CTRL_TIME 0x16 +/* ALC5623_OUTPUT_MIXER_CTRL : */ +/* same remark as for reg 2 line vs speaker */ +#define ALC5623_OUTPUT_MIXER_CTRL 0x1C +#define ALC5623_MIC_CTRL 0x22 + +#define ALC5623_DAI_CONTROL 0x34 +#define ALC5623_DAI_SDP_MASTER_MODE (0 << 15) +#define ALC5623_DAI_SDP_SLAVE_MODE (1 << 15) +#define ALC5623_DAI_I2S_PCM_MODE (1 << 14) +#define ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL (1 << 7) +#define ALC5623_DAI_ADC_DATA_L_R_SWAP (1 << 5) +#define ALC5623_DAI_DAC_DATA_L_R_SWAP (1 << 4) +#define ALC5623_DAI_I2S_DL_MASK (3 << 2) +#define ALC5623_DAI_I2S_DL_32 (3 << 2) +#define ALC5623_DAI_I2S_DL_24 (2 << 2) +#define ALC5623_DAI_I2S_DL_20 (1 << 2) +#define ALC5623_DAI_I2S_DL_16 (0 << 2) +#define ALC5623_DAI_I2S_DF_PCM (3 << 0) +#define ALC5623_DAI_I2S_DF_LEFT (2 << 0) +#define ALC5623_DAI_I2S_DF_RIGHT (1 << 0) +#define ALC5623_DAI_I2S_DF_I2S (0 << 0) + +#define ALC5623_STEREO_AD_DA_CLK_CTRL 0x36 +#define ALC5623_COMPANDING_CTRL 0x38 + +#define ALC5623_PWR_MANAG_ADD1 0x3A +#define ALC5623_PWR_ADD1_MAIN_I2S_EN (1 << 15) +#define ALC5623_PWR_ADD1_ZC_DET_PD_EN (1 << 14) +#define ALC5623_PWR_ADD1_MIC1_BIAS_EN (1 << 11) +#define ALC5623_PWR_ADD1_SHORT_CURR_DET_EN (1 << 10) +#define ALC5623_PWR_ADD1_SOFTGEN_EN (1 << 8) /* rsvd on 5622 */ +#define ALC5623_PWR_ADD1_DEPOP_BUF_HP (1 << 6) /* rsvd on 5622 */ +#define ALC5623_PWR_ADD1_HP_OUT_AMP (1 << 5) +#define ALC5623_PWR_ADD1_HP_OUT_ENH_AMP (1 << 4) /* rsvd on 5622 */ +#define ALC5623_PWR_ADD1_DEPOP_BUF_AUX (1 << 2) +#define ALC5623_PWR_ADD1_AUX_OUT_AMP (1 << 1) +#define ALC5623_PWR_ADD1_AUX_OUT_ENH_AMP (1 << 0) /* rsvd on 5622 */ + +#define ALC5623_PWR_MANAG_ADD2 0x3C +#define ALC5623_PWR_ADD2_LINEOUT (1 << 15) /* rt5623 */ +#define ALC5623_PWR_ADD2_CLASS_AB (1 << 15) /* rt5621 */ +#define ALC5623_PWR_ADD2_CLASS_D (1 << 14) /* rt5621 */ +#define ALC5623_PWR_ADD2_VREF (1 << 13) +#define ALC5623_PWR_ADD2_PLL (1 << 12) +#define ALC5623_PWR_ADD2_DAC_REF_CIR (1 << 10) +#define ALC5623_PWR_ADD2_L_DAC_CLK (1 << 9) +#define ALC5623_PWR_ADD2_R_DAC_CLK (1 << 8) +#define ALC5623_PWR_ADD2_L_ADC_CLK_GAIN (1 << 7) +#define ALC5623_PWR_ADD2_R_ADC_CLK_GAIN (1 << 6) +#define ALC5623_PWR_ADD2_L_HP_MIXER (1 << 5) +#define ALC5623_PWR_ADD2_R_HP_MIXER (1 << 4) +#define ALC5623_PWR_ADD2_SPK_MIXER (1 << 3) +#define ALC5623_PWR_ADD2_MONO_MIXER (1 << 2) +#define ALC5623_PWR_ADD2_L_ADC_REC_MIXER (1 << 1) +#define ALC5623_PWR_ADD2_R_ADC_REC_MIXER (1 << 0) + +#define ALC5623_PWR_MANAG_ADD3 0x3E +#define ALC5623_PWR_ADD3_MAIN_BIAS (1 << 15) +#define ALC5623_PWR_ADD3_AUXOUT_L_VOL_AMP (1 << 14) +#define ALC5623_PWR_ADD3_AUXOUT_R_VOL_AMP (1 << 13) +#define ALC5623_PWR_ADD3_SPK_OUT (1 << 12) +#define ALC5623_PWR_ADD3_HP_L_OUT_VOL (1 << 10) +#define ALC5623_PWR_ADD3_HP_R_OUT_VOL (1 << 9) +#define ALC5623_PWR_ADD3_LINEIN_L_VOL (1 << 7) +#define ALC5623_PWR_ADD3_LINEIN_R_VOL (1 << 6) +#define ALC5623_PWR_ADD3_AUXIN_L_VOL (1 << 5) +#define ALC5623_PWR_ADD3_AUXIN_R_VOL (1 << 4) +#define ALC5623_PWR_ADD3_MIC1_FUN_CTRL (1 << 3) +#define ALC5623_PWR_ADD3_MIC2_FUN_CTRL (1 << 2) +#define ALC5623_PWR_ADD3_MIC1_BOOST_AD (1 << 1) +#define ALC5623_PWR_ADD3_MIC2_BOOST_AD (1 << 0) + +#define ALC5623_ADD_CTRL_REG 0x40 + +#define ALC5623_GLOBAL_CLK_CTRL_REG 0x42 +#define ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL (1 << 15) +#define ALC5623_GBL_CLK_SYS_SOUR_SEL_MCLK (0 << 15) +#define ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK (1 << 14) +#define ALC5623_GBL_CLK_PLL_SOUR_SEL_MCLK (0 << 14) +#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV8 (3 << 1) +#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV4 (2 << 1) +#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV2 (1 << 1) +#define ALC5623_GBL_CLK_PLL_DIV_RATIO_DIV1 (0 << 1) +#define ALC5623_GBL_CLK_PLL_PRE_DIV2 (1 << 0) +#define ALC5623_GBL_CLK_PLL_PRE_DIV1 (0 << 0) + +#define ALC5623_PLL_CTRL 0x44 +#define ALC5623_PLL_CTRL_N_VAL(n) (((n)&0xff) << 8) +#define ALC5623_PLL_CTRL_K_VAL(k) (((k)&0x7) << 4) +#define ALC5623_PLL_CTRL_M_VAL(m) ((m)&0xf) + +#define ALC5623_GPIO_OUTPUT_PIN_CTRL 0x4A +#define ALC5623_GPIO_PIN_CONFIG 0x4C +#define ALC5623_GPIO_PIN_POLARITY 0x4E +#define ALC5623_GPIO_PIN_STICKY 0x50 +#define ALC5623_GPIO_PIN_WAKEUP 0x52 +#define ALC5623_GPIO_PIN_STATUS 0x54 +#define ALC5623_GPIO_PIN_SHARING 0x56 +#define ALC5623_OVER_CURR_STATUS 0x58 +#define ALC5623_JACK_DET_CTRL 0x5A + +#define ALC5623_MISC_CTRL 0x5E +#define ALC5623_MISC_DISABLE_FAST_VREG (1 << 15) +#define ALC5623_MISC_SPK_CLASS_AB_OC_PD (1 << 13) /* 5621 */ +#define ALC5623_MISC_SPK_CLASS_AB_OC_DET (1 << 12) /* 5621 */ +#define ALC5623_MISC_HP_DEPOP_MODE3_EN (1 << 10) +#define ALC5623_MISC_HP_DEPOP_MODE2_EN (1 << 9) +#define ALC5623_MISC_HP_DEPOP_MODE1_EN (1 << 8) +#define ALC5623_MISC_AUXOUT_DEPOP_MODE3_EN (1 << 6) +#define ALC5623_MISC_AUXOUT_DEPOP_MODE2_EN (1 << 5) +#define ALC5623_MISC_AUXOUT_DEPOP_MODE1_EN (1 << 4) +#define ALC5623_MISC_M_DAC_L_INPUT (1 << 3) +#define ALC5623_MISC_M_DAC_R_INPUT (1 << 2) +#define ALC5623_MISC_IRQOUT_INV_CTRL (1 << 0) + +#define ALC5623_PSEDUEO_SPATIAL_CTRL 0x60 +#define ALC5623_EQ_CTRL 0x62 +#define ALC5623_EQ_MODE_ENABLE 0x66 +#define ALC5623_AVC_CTRL 0x68 +#define ALC5623_HID_CTRL_INDEX 0x6A +#define ALC5623_HID_CTRL_DATA 0x6C +#define ALC5623_VENDOR_ID1 0x7C +#define ALC5623_VENDOR_ID2 0x7E + +#define ALC5623_PLL_FR_MCLK 0 +#define ALC5623_PLL_FR_BCK 1 +#endif From d906401114861585c990ff0290c002b5d22fc71a Mon Sep 17 00:00:00 2001 From: "Arnaud Patard (Rtp)" Date: Thu, 21 Oct 2010 19:40:03 +0200 Subject: [PATCH 02/10] ASoC: kirkwood: Add audio support to hp t5325 thin clients This patch is adding support for hp t5325 thin clients. There's a alc5623 codec connected to the i2s interface. Signed-off-by: Arnaud Patard Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/kirkwood/Kconfig | 9 ++ sound/soc/kirkwood/Makefile | 2 + sound/soc/kirkwood/kirkwood-t5325.c | 141 ++++++++++++++++++++++++++++ 3 files changed, 152 insertions(+) create mode 100644 sound/soc/kirkwood/kirkwood-t5325.c diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 16ec2a2dba4d..54258fd9797f 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -18,3 +18,12 @@ config SND_KIRKWOOD_SOC_OPENRD Say Y if you want to add support for SoC audio on Openrd Client. +config SND_KIRKWOOD_SOC_T5325 + tristate "SoC Audio support for HP t5325" + depends on SND_KIRKWOOD_SOC && MACH_T5325 + select SND_KIRKWOOD_SOC_I2S + select SND_SOC_ALC5623 + help + Say Y if you want to add support for SoC audio on + the HP t5325 thin client. + diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 33a16dcab5b5..3e62ae9e7bbe 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -5,5 +5,7 @@ obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o obj-$(CONFIG_SND_KIRKWOOD_SOC_I2S) += snd-soc-kirkwood-i2s.o snd-soc-openrd-objs := kirkwood-openrd.o +snd-soc-t5325-objs := kirkwood-t5325.o obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o +obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c new file mode 100644 index 000000000000..51b52e31cb0b --- /dev/null +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -0,0 +1,141 @@ +/* + * kirkwood-t5325.c + * + * (c) 2010 Arnaud Patard + * + * 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 "../codecs/alc5623.h" + +static int t5325_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int ret; + unsigned int freq, fmt; + + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + freq = params_rate(params) * 256; + + return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); + +} + +static struct snd_soc_ops t5325_ops = { + .hw_params = t5325_hw_params, +}; + +static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route t5325_route[] = { + { "Headphone Jack", NULL, "HPL" }, + { "Headphone Jack", NULL, "HPR" }, + + {"Speaker", NULL, "SPKOUT"}, + {"Speaker", NULL, "SPKOUTN"}, + + { "MIC1", NULL, "Mic Jack" }, + { "MIC2", NULL, "Mic Jack" }, +}; + +static int t5325_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + + snd_soc_dapm_new_controls(codec, t5325_dapm_widgets, + ARRAY_SIZE(t5325_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, t5325_route, ARRAY_SIZE(t5325_route)); + + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Speaker"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link t5325_dai[] = { +{ + .name = "ALC5621", + .stream_name = "ALC5621 HiFi", + .cpu_dai_name = "kirkwood-i2s", + .platform_name = "kirkwood-pcm-audio", + .codec_dai_name = "alc5621-hifi", + .codec_name = "alc562x-codec.0-001a", + .ops = &t5325_ops, + .init = t5325_dai_init, +}, +}; + + +static struct snd_soc_card t5325 = { + .name = "t5325", + .dai_link = t5325_dai, + .num_links = ARRAY_SIZE(t5325_dai), +}; + +static struct platform_device *t5325_snd_device; + +static int __init t5325_init(void) +{ + int ret; + + if (!machine_is_t5325()) + return 0; + + t5325_snd_device = platform_device_alloc("soc-audio", -1); + if (!t5325_snd_device) + return -ENOMEM; + + platform_set_drvdata(t5325_snd_device, + &t5325); + + ret = platform_device_add(t5325_snd_device); + if (ret) { + printk(KERN_ERR "%s: platform_device_add failed\n", __func__); + platform_device_put(t5325_snd_device); + } + + return ret; +} +module_init(t5325_init); + +static void __exit t5325_exit(void) +{ + platform_device_unregister(t5325_snd_device); +} +module_exit(t5325_exit); + +MODULE_AUTHOR("Arnaud Patard "); +MODULE_DESCRIPTION("ALSA SoC t5325 audio client"); +MODULE_LICENSE("GPL"); From c593b520cf70b0672680da04cc1e8c5f93bd739d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 27 Oct 2010 20:11:17 -0700 Subject: [PATCH 03/10] ASoC: Check return value of struct_strtoul() in pmdown_time_set() strict_strtoul() has just been made must check so do so. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-core.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 70d9a7394b2b..805343fe903b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -165,8 +165,11 @@ static ssize_t pmdown_time_set(struct device *dev, { struct snd_soc_pcm_runtime *rtd = container_of(dev, struct snd_soc_pcm_runtime, dev); + int ret; - strict_strtol(buf, 10, &rtd->pmdown_time); + ret = strict_strtol(buf, 10, &rtd->pmdown_time); + if (ret) + return ret; return count; } From fec6dd833e733b5d9588a1f1e4d81118b79b5774 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 27 Oct 2010 13:48:36 -0700 Subject: [PATCH 04/10] ASoC: Store DC offset correction for wm_hubs devices in class W mode Providing the analogue configuration of the output path remains the same the DC offset corrected by the DC servo will remain identical so we can skip the callibration, reducing the startup time for the headphone output. Implement this for the wm_hubs devices as has been done for several other CODECs. Don't do this if we have any analogue paths enabled since offsets may be being introduced by the analogue paths which could vary outside the control of the driver. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8993.c | 2 ++ sound/soc/codecs/wm8994.c | 3 ++ sound/soc/codecs/wm_hubs.c | 69 +++++++++++++++++++++++++------------- sound/soc/codecs/wm_hubs.h | 3 ++ 4 files changed, 54 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 589e3fa24734..67fe5ccc6082 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -735,6 +735,7 @@ static int class_w_put(struct snd_kcontrol *kcontrol, 0); } wm8993->class_w_users++; + wm8993->hubs_data.class_w = true; } /* Implement the change */ @@ -751,6 +752,7 @@ static int class_w_put(struct snd_kcontrol *kcontrol, WM8993_CP_DYN_V); } wm8993->class_w_users--; + wm8993->hubs_data.class_w = false; } dev_dbg(codec->dev, "Indirect DAC use count now %d\n", diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0db59c3aa5d4..3f70dee048b0 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2228,6 +2228,7 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, static void wm8994_update_class_w(struct snd_soc_codec *codec) { + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int enable = 1; int source = 0; /* GCC flow analysis can't track enable */ int reg, reg_r; @@ -2278,11 +2279,13 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) WM8994_CP_DYN_PWR | WM8994_CP_DYN_SRC_SEL_MASK, source | WM8994_CP_DYN_PWR); + wm8994->hubs.class_w = true; } else { dev_dbg(codec->dev, "Class W disabled\n"); snd_soc_update_bits(codec, WM8994_CLASS_W_1, WM8994_CP_DYN_PWR, 0); + wm8994->hubs.class_w = false; } } diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2cb81538cd91..31c2a5724d85 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -94,6 +94,18 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); u16 reg, reg_l, reg_r, dcs_cfg; + /* If we're using a digital only path and have a previously + * callibrated DC servo offset stored then use that. */ + if (hubs->class_w && hubs->class_w_dcs) { + dev_dbg(codec->dev, "Using cached DC servo offset %x\n", + hubs->class_w_dcs); + snd_soc_write(codec, WM8993_DC_SERVO_3, hubs->class_w_dcs); + wait_for_dc_servo(codec, + WM8993_DCS_TRIG_DAC_WR_0 | + WM8993_DCS_TRIG_DAC_WR_1); + return; + } + /* Set for 32 series updates */ snd_soc_update_bits(codec, WM8993_DC_SERVO_1, WM8993_DCS_SERIES_NO_01_MASK, @@ -101,34 +113,34 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) wait_for_dc_servo(codec, WM8993_DCS_TRIG_SERIES_0 | WM8993_DCS_TRIG_SERIES_1); + /* Different chips in the family support different readback + * methods. + */ + switch (hubs->dcs_readback_mode) { + case 0: + reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) + & WM8993_DCS_INTEG_CHAN_0_MASK;; + reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) + & WM8993_DCS_INTEG_CHAN_1_MASK; + break; + case 1: + reg = snd_soc_read(codec, WM8993_DC_SERVO_3); + reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) + >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; + reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; + break; + default: + WARN(1, "Unknown DCS readback method"); + break; + } + + dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r); + /* Apply correction to DC servo result */ if (hubs->dcs_codes) { dev_dbg(codec->dev, "Applying %d code DC servo correction\n", hubs->dcs_codes); - /* Different chips in the family support different - * readback methods. - */ - switch (hubs->dcs_readback_mode) { - case 0: - reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) - & WM8993_DCS_INTEG_CHAN_0_MASK;; - reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) - & WM8993_DCS_INTEG_CHAN_1_MASK; - break; - case 1: - reg = snd_soc_read(codec, WM8993_DC_SERVO_3); - reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) - >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; - reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; - break; - default: - WARN(1, "Unknown DCS readback method"); - break; - } - - dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r); - /* HPOUT1L */ if (reg_l + hubs->dcs_codes > 0 && reg_l + hubs->dcs_codes < 0xff) @@ -148,7 +160,15 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) wait_for_dc_servo(codec, WM8993_DCS_TRIG_DAC_WR_0 | WM8993_DCS_TRIG_DAC_WR_1); + } else { + dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT; + dcs_cfg |= reg_r; } + + /* Save the callibrated offset if we're in class W mode and + * therefore don't have any analogue signal mixed in. */ + if (hubs->class_w) + hubs->class_w_dcs = dcs_cfg; } /* @@ -163,6 +183,9 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + /* Updating the analogue gains invalidates the DC servo cache */ + hubs->class_w_dcs = 0; + /* If we're applying an offset correction then updating the * callibration would be likely to introduce further offsets. */ if (hubs->dcs_codes) diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index e51c16683589..f8a5e976b5e6 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -23,6 +23,9 @@ struct wm_hubs_data { int dcs_codes; int dcs_readback_mode; int hp_startup_mode; + + bool class_w; + u16 class_w_dcs; }; extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *); From 6d212d8e86fb4221bd91b9266b7567ee2b83bd01 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 29 Oct 2010 15:41:17 -0700 Subject: [PATCH 05/10] ASoC: Remove volatility from WM8900 POWER1 register Not all bits can be read back from POWER1 so avoid corruption when using a read/modify/write cycle by marking it non-volatile - the only thing we read back from it is the chip revision which has diagnostic value only. We can re-add later but that's a more invasive change than is suitable for a bugfix. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8900.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index b4f11724a63f..aca4b1ea10bb 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -186,7 +186,6 @@ static int wm8900_volatile_register(unsigned int reg) { switch (reg) { case WM8900_REG_ID: - case WM8900_REG_POWER1: return 1; default: return 0; @@ -1200,11 +1199,6 @@ static int wm8900_probe(struct snd_soc_codec *codec) return -ENODEV; } - /* Read back from the chip */ - reg = snd_soc_read(codec, WM8900_REG_POWER1); - reg = (reg >> 12) & 0xf; - dev_info(codec->dev, "WM8900 revision %d\n", reg); - wm8900_reset(codec); /* Turn the chip on */ From 703dde6219346bc3b7d41d4fa2c36846d728e52c Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 29 Oct 2010 16:47:44 +0300 Subject: [PATCH 06/10] ASoC: Fix SND_SOC_ALL_CODECS typo for jz4740 Include jz4740.c to SND_SOC_ALL_CODECS when the dependencies are met. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 94a9d06b9027..02a9751bf149 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,7 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_DA7210 if I2C - select SND_SOC_JZ4740 if SOC_JZ4740 + select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_MAX98088 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 From 76a6106f124e375df0ea6ba6bcf204b8caff786a Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 29 Oct 2010 16:47:45 +0300 Subject: [PATCH 07/10] ASoC: Include cx20442 to SND_SOC_ALL_CODECS Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 02a9751bf149..3b5690d28b8b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -25,6 +25,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C + select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_JZ4740_CODEC if SOC_JZ4740 select SND_SOC_MAX98088 if I2C From 473f89fff76568a9f30c53b458e6323d48b0ab95 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 29 Oct 2010 16:47:44 +0300 Subject: [PATCH 08/10] ASoC: Fix SND_SOC_ALL_CODECS typo for alc5623 Include alc5623.c in SND_SOC_ALL_CODECS when dependencies are met. Signed-off-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8d11fbd1b1aa..e61fbab48aa2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -22,7 +22,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C - select SND_SOC_ALC562 if I2C + select SND_SOC_ALC5623 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C select SND_SOC_CS4270 if I2C From 5a0b07433ddd808ecbb5f4287b61be6fa7af1b57 Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Sat, 30 Oct 2010 14:08:56 -0700 Subject: [PATCH 09/10] ASoC: Update WARN uses in wm_hubs Add missing newlines. Signed-off-by: Joe Perches Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2cb81538cd91..19ca782ac970 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -123,7 +123,7 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; break; default: - WARN(1, "Unknown DCS readback method"); + WARN(1, "Unknown DCS readback method\n"); break; } From cb9906229595941d632fc4022b05da4f9533856a Mon Sep 17 00:00:00 2001 From: Eric Miao Date: Tue, 2 Nov 2010 05:10:07 +0800 Subject: [PATCH 10/10] ASoC: fix the building issue of missing codec field in 'struct snd_soc_card' Signed-off-by: Mark Brown --- sound/soc/pxa/tosa.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index a3bfb2e8b70f..73d0edd8ded9 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -79,7 +79,7 @@ static void tosa_ext_control(struct snd_soc_codec *codec) static int tosa_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->card->codec; + struct snd_soc_codec *codec = rtd->codec; /* check the jack status at stream startup */ tosa_ext_control(codec);