* [patch v3 1/2] ASoC: add support for alc562[123] codecs
2010-10-21 10:44 [patch v3 0/2] ASoC: alc562[123] / HP t5325 support - try 3 Arnaud Patard
@ 2010-10-21 10:44 ` Arnaud Patard
2010-10-21 16:28 ` Mark Brown
2010-10-21 10:44 ` [patch v3 2/2] ASoC: kirkwood: Add audio support to hp t5325 thin clients Arnaud Patard
1 sibling, 1 reply; 4+ messages in thread
From: Arnaud Patard @ 2010-10-21 10:44 UTC (permalink / raw)
To: alsa-devel
Cc: Arnaud Patard, Mark Brown, Martin Michlmayr, Saeed Bishara,
Liam Girdwood
[-- Attachment #1: alc56xx.patch --]
[-- Type: text/plain, Size: 45992 bytes --]
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 <arnaud.patard@rtp-net.org>
Index: sound-2.6/sound/soc/codecs/Kconfig
===================================================================
--- sound-2.6.orig/sound/soc/codecs/Kconfig 2010-10-17 11:07:13.000000000 +0200
+++ sound-2.6/sound/soc/codecs/Kconfig 2010-10-17 11:07:35.000000000 +0200
@@ -22,6 +22,7 @@
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_AK4671
tristate
+config SND_SOC_ALC5623
+ tristate
+
config SND_SOC_CQ0093VC
tristate
@@ -317,3 +321,4 @@
config SND_SOC_WM9090
tristate
+
Index: sound-2.6/sound/soc/codecs/Makefile
===================================================================
--- sound-2.6.orig/sound/soc/codecs/Makefile 2010-10-17 11:07:13.000000000 +0200
+++ sound-2.6/sound/soc/codecs/Makefile 2010-10-17 11:07:35.000000000 +0200
@@ -17,6 +17,7 @@
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-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o
obj-$(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
Index: sound-2.6/sound/soc/codecs/alc5623.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
+++ sound-2.6/sound/soc/codecs/alc5623.c 2010-10-17 11:07:35.000000000 +0200
@@ -0,0 +1,1148 @@
+/*
+ * alc5623.c -- alc562[123] ALSA Soc Audio driver
+ *
+ * Copyright 2008 Realtek Microelectronics
+ * Author: flove <flove@realtek.com> Ethan <eku@marvell.com>
+ *
+ * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ *
+ * 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 <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/alc5623.h>
+
+#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 void alc5623_work(struct work_struct *work)
+{
+ struct snd_soc_codec *codec =
+ container_of(work, struct snd_soc_codec, delayed_work.work);
+ alc5623_set_bias_level(codec, codec->bias_level);
+}
+
+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;
+
+ INIT_DELAYED_WORK(&codec->delayed_work, alc5623_work);
+
+ 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;
+}
+
+/*
+ * This function forces any delayed work to be queued and run.
+ */
+static int run_delayed_work(struct delayed_work *dwork)
+{
+ int ret;
+
+ /* cancel any work waiting to be queued. */
+ ret = cancel_delayed_work(dwork);
+
+ /* if there was any work waiting then we run it now and
+ * wait for it's completion */
+ if (ret) {
+ schedule_delayed_work(dwork, 0);
+ flush_scheduled_work();
+ }
+ return ret;
+}
+
+/* power down chip */
+static int alc5623_remove(struct snd_soc_codec *codec)
+{
+ run_delayed_work(&codec->delayed_work);
+ 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 <arnaud.patard@rtp-net.org>");
+MODULE_LICENSE("GPL");
+
Index: sound-2.6/sound/soc/codecs/alc5623.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
+++ sound-2.6/sound/soc/codecs/alc5623.h 2010-10-17 11:07:35.000000000 +0200
@@ -0,0 +1,161 @@
+/*
+ * alc5623.h -- alc562[123] ALSA Soc Audio driver
+ *
+ * Copyright 2008 Realtek Microelectronics
+ * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ * Author: flove <flove@realtek.com>
+ * Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ * 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
Index: sound-2.6/include/sound/alc5623.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
+++ sound-2.6/include/sound/alc5623.h 2010-10-17 11:07:35.000000000 +0200
@@ -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
+
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