Merge branch 'for-2.6.39' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h
index 9eab263..466b1c7 100644
--- a/include/linux/mfd/wm8994/pdata.h
+++ b/include/linux/mfd/wm8994/pdata.h
@@ -103,13 +103,21 @@
unsigned int lineout1fb:1;
unsigned int lineout2fb:1;
- /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */
+ /* IRQ for microphone detection if brought out directly as a
+ * signal.
+ */
+ int micdet_irq;
+
+ /* WM8994 microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */
unsigned int micbias1_lvl:1;
unsigned int micbias2_lvl:1;
- /* Jack detect threashold levels, see datasheet for values */
+ /* WM8994 jack detect threashold levels, see datasheet for values */
unsigned int jd_scthr:2;
unsigned int jd_thr:2;
+
+ /* WM8958 microphone bias configuration */
+ int micbias[2];
};
#endif
diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h
index be072fa..f3ee842 100644
--- a/include/linux/mfd/wm8994/registers.h
+++ b/include/linux/mfd/wm8994/registers.h
@@ -63,6 +63,8 @@
#define WM8994_MICBIAS 0x3A
#define WM8994_LDO_1 0x3B
#define WM8994_LDO_2 0x3C
+#define WM8958_MICBIAS1 0x3D
+#define WM8958_MICBIAS2 0x3E
#define WM8994_CHARGE_PUMP_1 0x4C
#define WM8958_CHARGE_PUMP_2 0x4D
#define WM8994_CLASS_W_1 0x51
diff --git a/include/sound/control.h b/include/sound/control.h
index 7715e6f..e67db28 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -115,6 +115,8 @@
int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol);
int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id);
int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id);
+int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id,
+ int active);
struct snd_kcontrol *snd_ctl_find_numid(struct snd_card * card, unsigned int numid);
struct snd_kcontrol *snd_ctl_find_id(struct snd_card * card, struct snd_ctl_elem_id *id);
diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h
index 16f8d32..50a059e 100644
--- a/include/sound/cs4271.h
+++ b/include/sound/cs4271.h
@@ -19,7 +19,6 @@
struct cs4271_platform_data {
int gpio_nreset; /* GPIO driving Reset pin, if any */
- int gpio_disable; /* GPIO that disable serial bus, if any */
};
#endif /* __CS4271_H */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 4b6c0a8..bfa4836 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -234,6 +234,7 @@
struct snd_soc_codec_driver;
struct soc_enum;
struct snd_soc_jack;
+struct snd_soc_jack_zone;
struct snd_soc_jack_pin;
struct snd_soc_cache_ops;
#include <sound/soc-dapm.h>
@@ -258,6 +259,11 @@
SND_SOC_RBTREE_COMPRESSION
};
+int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+ unsigned int freq, int dir);
+int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
+ unsigned int freq_in, unsigned int freq_out);
+
int snd_soc_register_card(struct snd_soc_card *card);
int snd_soc_unregister_card(struct snd_soc_card *card);
int snd_soc_suspend(struct device *dev);
@@ -307,6 +313,9 @@
struct notifier_block *nb);
void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack,
struct notifier_block *nb);
+int snd_soc_jack_add_zones(struct snd_soc_jack *jack, int count,
+ struct snd_soc_jack_zone *zones);
+int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage);
#ifdef CONFIG_GPIOLIB
int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
struct snd_soc_jack_gpio *gpios);
@@ -331,7 +340,8 @@
*Controls
*/
struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
- void *data, char *long_name);
+ void *data, char *long_name,
+ const char *prefix);
int snd_soc_add_controls(struct snd_soc_codec *codec,
const struct snd_kcontrol_new *controls, int num_controls);
int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
@@ -407,6 +417,24 @@
};
/**
+ * struct snd_soc_jack_zone - Describes voltage zones of jack detection
+ *
+ * @min_mv: start voltage in mv
+ * @max_mv: end voltage in mv
+ * @jack_type: type of jack that is expected for this voltage
+ * @debounce_time: debounce_time for jack, codec driver should wait for this
+ * duration before reading the adc for voltages
+ * @:list: list container
+ */
+struct snd_soc_jack_zone {
+ unsigned int min_mv;
+ unsigned int max_mv;
+ unsigned int jack_type;
+ unsigned int debounce_time;
+ struct list_head list;
+};
+
+/**
* struct snd_soc_jack_gpio - Describes a gpio pin for jack detection
*
* @gpio: gpio number
@@ -414,6 +442,10 @@
* @report: value to report when jack detected
* @invert: report presence in low state
* @debouce_time: debouce time in ms
+ * @wake: enable as wake source
+ * @jack_status_check: callback function which overrides the detection
+ * to provide more complex checks (eg, reading an
+ * ADC).
*/
#ifdef CONFIG_GPIOLIB
struct snd_soc_jack_gpio {
@@ -422,6 +454,8 @@
int report;
int invert;
int debounce_time;
+ bool wake;
+
struct snd_soc_jack *jack;
struct delayed_work work;
@@ -435,6 +469,7 @@
struct list_head pins;
int status;
struct blocking_notifier_head notifier;
+ struct list_head jack_zones;
};
/* SoC PCM stream information */
@@ -533,6 +568,18 @@
pm_message_t state);
int (*resume)(struct snd_soc_codec *);
+ /* Default DAPM setup, added after probe() is run */
+ const struct snd_soc_dapm_widget *dapm_widgets;
+ int num_dapm_widgets;
+ const struct snd_soc_dapm_route *dapm_routes;
+ int num_dapm_routes;
+
+ /* codec wide operations */
+ int (*set_sysclk)(struct snd_soc_codec *codec,
+ int clk_id, unsigned int freq, int dir);
+ int (*set_pll)(struct snd_soc_codec *codec, int pll_id, int source,
+ unsigned int freq_in, unsigned int freq_out);
+
/* codec IO */
unsigned int (*read)(struct snd_soc_codec *, unsigned int);
int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
@@ -653,6 +700,7 @@
bool instantiated;
int (*probe)(struct snd_soc_card *card);
+ int (*late_probe)(struct snd_soc_card *card);
int (*remove)(struct snd_soc_card *card);
/* the pre and post PM functions are used to do any PM work before and
@@ -689,6 +737,14 @@
struct snd_soc_pcm_runtime *rtd_aux;
int num_aux_rtd;
+ /*
+ * Card-specific routes and widgets.
+ */
+ struct snd_soc_dapm_widget *dapm_widgets;
+ int num_dapm_widgets;
+ struct snd_soc_dapm_route *dapm_routes;
+ int num_dapm_routes;
+
struct work_struct deferred_resume_work;
/* lists of probed devices belonging to this card */
@@ -700,6 +756,9 @@
struct list_head paths;
struct list_head dapm_list;
+ /* Generic DAPM context for the card */
+ struct snd_soc_dapm_context dapm;
+
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_card_root;
struct dentry *debugfs_pop_time;
diff --git a/include/sound/tlv320aic32x4.h b/include/sound/tlv320aic32x4.h
new file mode 100644
index 0000000..c009f70
--- /dev/null
+++ b/include/sound/tlv320aic32x4.h
@@ -0,0 +1,31 @@
+/*
+ * tlv320aic32x4.h -- TLV320AIC32X4 Soc Audio driver platform data
+ *
+ * Copyright 2011 Vista Silicon S.L.
+ *
+ * Author: Javier Martin <javier.martin@vista-silicon.com>
+ *
+ * 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 _AIC32X4_PDATA_H
+#define _AIC32X4_PDATA_H
+
+#define AIC32X4_PWR_MICBIAS_2075_LDOIN 0x00000001
+#define AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE 0x00000002
+#define AIC32X4_PWR_AIC32X4_LDO_ENABLE 0x00000004
+#define AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36 0x00000008
+#define AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED 0x00000010
+
+#define AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K 0x00000001
+#define AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K 0x00000002
+
+struct aic32x4_pdata {
+ u32 power_cfg;
+ u32 micpga_routing;
+ bool swapdacs;
+};
+
+#endif
diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h
index 86172cf..cf7ccb7 100644
--- a/include/sound/wm8903.h
+++ b/include/sound/wm8903.h
@@ -17,13 +17,9 @@
/*
* R6 (0x06) - Mic Bias Control 0
*/
-#define WM8903_MICDET_HYST_ENA 0x0080 /* MICDET_HYST_ENA */
-#define WM8903_MICDET_HYST_ENA_MASK 0x0080 /* MICDET_HYST_ENA */
-#define WM8903_MICDET_HYST_ENA_SHIFT 7 /* MICDET_HYST_ENA */
-#define WM8903_MICDET_HYST_ENA_WIDTH 1 /* MICDET_HYST_ENA */
-#define WM8903_MICDET_THR_MASK 0x0070 /* MICDET_THR - [6:4] */
-#define WM8903_MICDET_THR_SHIFT 4 /* MICDET_THR - [6:4] */
-#define WM8903_MICDET_THR_WIDTH 3 /* MICDET_THR - [6:4] */
+#define WM8903_MICDET_THR_MASK 0x0030 /* MICDET_THR - [5:4] */
+#define WM8903_MICDET_THR_SHIFT 4 /* MICDET_THR - [5:4] */
+#define WM8903_MICDET_THR_WIDTH 2 /* MICDET_THR - [5:4] */
#define WM8903_MICSHORT_THR_MASK 0x000C /* MICSHORT_THR - [3:2] */
#define WM8903_MICSHORT_THR_SHIFT 2 /* MICSHORT_THR - [3:2] */
#define WM8903_MICSHORT_THR_WIDTH 2 /* MICSHORT_THR - [3:2] */
diff --git a/include/sound/wm9081.h b/include/sound/wm9081.h
index e173ddb..f34b0b1 100644
--- a/include/sound/wm9081.h
+++ b/include/sound/wm9081.h
@@ -17,9 +17,12 @@
u16 config[20];
};
-struct wm9081_retune_mobile_config {
- struct wm9081_retune_mobile_setting *configs;
- int num_configs;
+struct wm9081_pdata {
+ bool irq_high; /* IRQ is active high */
+ bool irq_cmos; /* IRQ is in CMOS mode */
+
+ struct wm9081_retune_mobile_setting *retune_configs;
+ int num_retune_configs;
};
#endif
diff --git a/sound/core/control.c b/sound/core/control.c
index 9ce00ed..db51e4e 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -466,6 +466,52 @@
}
/**
+ * snd_ctl_activate_id - activate/inactivate the control of the given id
+ * @card: the card instance
+ * @id: the control id to activate/inactivate
+ * @active: non-zero to activate
+ *
+ * Finds the control instance with the given id, and activate or
+ * inactivate the control together with notification, if changed.
+ *
+ * Returns 0 if unchanged, 1 if changed, or a negative error code on failure.
+ */
+int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id,
+ int active)
+{
+ struct snd_kcontrol *kctl;
+ struct snd_kcontrol_volatile *vd;
+ unsigned int index_offset;
+ int ret;
+
+ down_write(&card->controls_rwsem);
+ kctl = snd_ctl_find_id(card, id);
+ if (kctl == NULL) {
+ ret = -ENOENT;
+ goto unlock;
+ }
+ index_offset = snd_ctl_get_ioff(kctl, &kctl->id);
+ vd = &kctl->vd[index_offset];
+ ret = 0;
+ if (active) {
+ if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_INACTIVE))
+ goto unlock;
+ vd->access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ } else {
+ if (vd->access & SNDRV_CTL_ELEM_ACCESS_INACTIVE)
+ goto unlock;
+ vd->access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ }
+ ret = 1;
+ unlock:
+ up_write(&card->controls_rwsem);
+ if (ret > 0)
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, id);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(snd_ctl_activate_id);
+
+/**
* snd_ctl_rename_id - replace the id of a control on the card
* @card: the card instance
* @src_id: the old id
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index e239345..d63c175 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -29,16 +29,21 @@
select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
select SND_SOC_CX20442
select SND_SOC_DA7210 if I2C
+ select SND_SOC_DFBMCS320
select SND_SOC_JZ4740_CODEC if SOC_JZ4740
+ select SND_SOC_LM4857 if I2C
select SND_SOC_MAX98088 if I2C
+ select SND_SOC_MAX9850 if I2C
select SND_SOC_MAX9877 if I2C
select SND_SOC_PCM3008
+ select SND_SOC_SGTL5000 if I2C
select SND_SOC_SN95031 if INTEL_SCU_IPC
select SND_SOC_SPDIF
select SND_SOC_SSM2602 if I2C
select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
select SND_SOC_TLV320AIC23 if I2C
select SND_SOC_TLV320AIC26 if SPI_MASTER
+ select SND_SOC_TVL320AIC32X4 if I2C
select SND_SOC_TLV320AIC3X if I2C
select SND_SOC_TPA6130A2 if I2C
select SND_SOC_TLV320DAC33 if I2C
@@ -173,15 +178,25 @@
config SND_SOC_DA7210
tristate
+config SND_SOC_DFBMCS320
+ tristate
+
config SND_SOC_DMIC
tristate
config SND_SOC_MAX98088
tristate
+config SND_SOC_MAX9850
+ tristate
+
config SND_SOC_PCM3008
tristate
+#Freescale sgtl5000 codec
+config SND_SOC_SGTL5000
+ tristate
+
config SND_SOC_SN95031
tristate
@@ -201,6 +216,9 @@
tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE
depends on SPI
+config SND_SOC_TVL320AIC32X4
+ tristate
+
config SND_SOC_TLV320AIC3X
tristate
@@ -338,6 +356,9 @@
tristate
# Amp
+config SND_SOC_LM4857
+ tristate
+
config SND_SOC_MAX9877
tristate
@@ -349,4 +370,3 @@
config SND_SOC_WM9090
tristate
-
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 83b7acc..379bc55 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -15,10 +15,13 @@
snd-soc-cs4271-objs := cs4271.o
snd-soc-cx20442-objs := cx20442.o
snd-soc-da7210-objs := da7210.o
+snd-soc-dfbmcs320-objs := dfbmcs320.o
snd-soc-dmic-objs := dmic.o
snd-soc-l3-objs := l3.o
snd-soc-max98088-objs := max98088.o
+snd-soc-max9850-objs := max9850.o
snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-sgtl5000-objs := sgtl5000.o
snd-soc-alc5623-objs := alc5623.o
snd-soc-sn95031-objs := sn95031.o
snd-soc-spdif-objs := spdif_transciever.o
@@ -27,6 +30,7 @@
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
+snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o
snd-soc-tlv320dac33-objs := tlv320dac33.o
snd-soc-twl4030-objs := twl4030.o
snd-soc-twl6040-objs := twl6040.o
@@ -75,6 +79,7 @@
snd-soc-jz4740-codec-objs := jz4740.o
# Amp
+snd-soc-lm4857-objs := lm4857.o
snd-soc-max9877-objs := max9877.o
snd-soc-tpa6130a2-objs := tpa6130a2.o
snd-soc-wm2000-objs := wm2000.o
@@ -91,18 +96,21 @@
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o
obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o
+obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o
obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o
obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o
+obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o
obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o
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_MAX9850) += snd-soc-max9850.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
-obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o
+obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o
obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o
obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
@@ -110,6 +118,7 @@
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
+obj-$(CONFIG_SND_SOC_TVL320AIC32X4) += snd-soc-tlv320aic32x4.o
obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o
obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o
obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o
@@ -157,6 +166,7 @@
obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o
# Amp
+obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o
obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o
obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o
obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index c27f8f5..cbf0b6d 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -294,7 +294,6 @@
static int __init ak4104_init(void)
{
- pr_info("Asahi Kasei AK4104 ALSA SoC Codec Driver\n");
return spi_register_driver(&ak4104_spi_driver);
}
module_init(ak4104_init);
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index c0fccad..0206a17 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -719,7 +719,7 @@
/*
* cs4270_id - I2C device IDs supported by this driver
*/
-static struct i2c_device_id cs4270_id[] = {
+static const struct i2c_device_id cs4270_id[] = {
{"cs4270", 0},
{}
};
@@ -743,8 +743,6 @@
static int __init cs4270_init(void)
{
- pr_info("Cirrus Logic CS4270 ALSA SoC Codec Driver\n");
-
return i2c_add_driver(&cs4270_i2c_driver);
}
module_init(cs4270_init);
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 9c5b7db..083aab9 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -33,6 +33,7 @@
#define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_LE | \
SNDRV_PCM_FMTBIT_S32_LE)
+#define CS4271_PCM_RATES SNDRV_PCM_RATE_8000_192000
/*
* CS4271 registers
@@ -167,27 +168,6 @@
int gpio_disable;
};
-struct cs4271_clk_cfg {
- unsigned int ratio; /* MCLK / sample rate */
- u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */
- u8 mclk_master; /* ratio bit mask for Master mode */
- u8 mclk_slave; /* ratio bit mask for Slave mode */
-};
-
-static struct cs4271_clk_cfg cs4271_clk_tab[] = {
- {64, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1},
- {96, CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1},
- {128, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1},
- {192, CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1},
- {256, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_1, CS4271_MODE1_DIV_1},
- {384, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1},
- {512, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_2, CS4271_MODE1_DIV_1},
- {768, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3},
- {1024, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3, CS4271_MODE1_DIV_3}
-};
-
-#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab)
-
/*
* @freq is the desired MCLK rate
* MCLK rate should (c) be the sample rate, multiplied by one of the
@@ -296,6 +276,45 @@
return cs4271_set_deemph(codec);
}
+struct cs4271_clk_cfg {
+ bool master; /* codec mode */
+ u8 speed_mode; /* codec speed mode: 1x, 2x, 4x */
+ unsigned short ratio; /* MCLK / sample rate */
+ u8 ratio_mask; /* ratio bit mask for Master mode */
+};
+
+static struct cs4271_clk_cfg cs4271_clk_tab[] = {
+ {1, CS4271_MODE1_MODE_1X, 256, CS4271_MODE1_DIV_1},
+ {1, CS4271_MODE1_MODE_1X, 384, CS4271_MODE1_DIV_15},
+ {1, CS4271_MODE1_MODE_1X, 512, CS4271_MODE1_DIV_2},
+ {1, CS4271_MODE1_MODE_1X, 768, CS4271_MODE1_DIV_3},
+ {1, CS4271_MODE1_MODE_2X, 128, CS4271_MODE1_DIV_1},
+ {1, CS4271_MODE1_MODE_2X, 192, CS4271_MODE1_DIV_15},
+ {1, CS4271_MODE1_MODE_2X, 256, CS4271_MODE1_DIV_2},
+ {1, CS4271_MODE1_MODE_2X, 384, CS4271_MODE1_DIV_3},
+ {1, CS4271_MODE1_MODE_4X, 64, CS4271_MODE1_DIV_1},
+ {1, CS4271_MODE1_MODE_4X, 96, CS4271_MODE1_DIV_15},
+ {1, CS4271_MODE1_MODE_4X, 128, CS4271_MODE1_DIV_2},
+ {1, CS4271_MODE1_MODE_4X, 192, CS4271_MODE1_DIV_3},
+ {0, CS4271_MODE1_MODE_1X, 256, CS4271_MODE1_DIV_1},
+ {0, CS4271_MODE1_MODE_1X, 384, CS4271_MODE1_DIV_1},
+ {0, CS4271_MODE1_MODE_1X, 512, CS4271_MODE1_DIV_1},
+ {0, CS4271_MODE1_MODE_1X, 768, CS4271_MODE1_DIV_2},
+ {0, CS4271_MODE1_MODE_1X, 1024, CS4271_MODE1_DIV_2},
+ {0, CS4271_MODE1_MODE_2X, 128, CS4271_MODE1_DIV_1},
+ {0, CS4271_MODE1_MODE_2X, 192, CS4271_MODE1_DIV_1},
+ {0, CS4271_MODE1_MODE_2X, 256, CS4271_MODE1_DIV_1},
+ {0, CS4271_MODE1_MODE_2X, 384, CS4271_MODE1_DIV_2},
+ {0, CS4271_MODE1_MODE_2X, 512, CS4271_MODE1_DIV_2},
+ {0, CS4271_MODE1_MODE_4X, 64, CS4271_MODE1_DIV_1},
+ {0, CS4271_MODE1_MODE_4X, 96, CS4271_MODE1_DIV_1},
+ {0, CS4271_MODE1_MODE_4X, 128, CS4271_MODE1_DIV_1},
+ {0, CS4271_MODE1_MODE_4X, 192, CS4271_MODE1_DIV_2},
+ {0, CS4271_MODE1_MODE_4X, 256, CS4271_MODE1_DIV_2},
+};
+
+#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab)
+
static int cs4271_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
@@ -307,23 +326,28 @@
unsigned int ratio, val;
cs4271->rate = params_rate(params);
+
+ /* Configure DAC */
+ if (cs4271->rate < 50000)
+ val = CS4271_MODE1_MODE_1X;
+ else if (cs4271->rate < 100000)
+ val = CS4271_MODE1_MODE_2X;
+ else
+ val = CS4271_MODE1_MODE_4X;
+
ratio = cs4271->mclk / cs4271->rate;
for (i = 0; i < CS4171_NR_RATIOS; i++)
- if (cs4271_clk_tab[i].ratio == ratio)
+ if ((cs4271_clk_tab[i].master == cs4271->master) &&
+ (cs4271_clk_tab[i].speed_mode == val) &&
+ (cs4271_clk_tab[i].ratio == ratio))
break;
- if ((i == CS4171_NR_RATIOS) || ((ratio == 1024) && cs4271->master)) {
+ if (i == CS4171_NR_RATIOS) {
dev_err(codec->dev, "Invalid sample rate\n");
return -EINVAL;
}
- /* Configure DAC */
- val = cs4271_clk_tab[i].speed_mode;
-
- if (cs4271->master)
- val |= cs4271_clk_tab[i].mclk_master;
- else
- val |= cs4271_clk_tab[i].mclk_slave;
+ val |= cs4271_clk_tab[i].ratio_mask;
ret = snd_soc_update_bits(codec, CS4271_MODE1,
CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val);
@@ -392,14 +416,14 @@
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_96000,
+ .rates = CS4271_PCM_RATES,
.formats = CS4271_PCM_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_96000,
+ .rates = CS4271_PCM_RATES,
.formats = CS4271_PCM_FORMATS,
},
.ops = &cs4271_dai_ops,
@@ -441,22 +465,11 @@
struct cs4271_platform_data *cs4271plat = codec->dev->platform_data;
int ret;
int gpio_nreset = -EINVAL;
- int gpio_disable = -EINVAL;
codec->control_data = cs4271->control_data;
- if (cs4271plat) {
- if (gpio_is_valid(cs4271plat->gpio_nreset))
- gpio_nreset = cs4271plat->gpio_nreset;
- if (gpio_is_valid(cs4271plat->gpio_disable))
- gpio_disable = cs4271plat->gpio_disable;
- }
-
- if (gpio_disable >= 0)
- if (gpio_request(gpio_disable, "CS4271 Disable"))
- gpio_disable = -EINVAL;
- if (gpio_disable >= 0)
- gpio_direction_output(gpio_disable, 0);
+ if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset))
+ gpio_nreset = cs4271plat->gpio_nreset;
if (gpio_nreset >= 0)
if (gpio_request(gpio_nreset, "CS4271 Reset"))
@@ -471,7 +484,6 @@
}
cs4271->gpio_nreset = gpio_nreset;
- cs4271->gpio_disable = gpio_disable;
/*
* In case of I2C, chip address specified in board data.
@@ -509,10 +521,9 @@
static int cs4271_remove(struct snd_soc_codec *codec)
{
struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
- int gpio_nreset, gpio_disable;
+ int gpio_nreset;
gpio_nreset = cs4271->gpio_nreset;
- gpio_disable = cs4271->gpio_disable;
if (gpio_is_valid(gpio_nreset)) {
/* Set codec to the reset state */
@@ -520,9 +531,6 @@
gpio_free(gpio_nreset);
}
- if (gpio_is_valid(gpio_disable))
- gpio_free(gpio_disable);
-
return 0;
};
@@ -571,7 +579,7 @@
#endif /* defined(CONFIG_SPI_MASTER) */
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-static struct i2c_device_id cs4271_i2c_id[] = {
+static const struct i2c_device_id cs4271_i2c_id[] = {
{"cs4271", 0},
{}
};
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index bb4bf65..0bb424a 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -367,7 +367,7 @@
return 0;
}
-static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC;
+static const u8 cx20442_reg;
static struct snd_soc_codec_driver cx20442_codec_dev = {
.probe = cx20442_codec_probe,
diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/dfbmcs320.c
new file mode 100644
index 0000000..704bbde
--- /dev/null
+++ b/sound/soc/codecs/dfbmcs320.c
@@ -0,0 +1,72 @@
+/*
+ * Driver for the DFBM-CS320 bluetooth module
+ * Copyright 2011 Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * 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 <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+
+#include <sound/soc.h>
+
+static struct snd_soc_dai_driver dfbmcs320_dai = {
+ .name = "dfbmcs320-pcm",
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_dfbmcs320;
+
+static int __devinit dfbmcs320_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_dfbmcs320,
+ &dfbmcs320_dai, 1);
+}
+
+static int __devexit dfbmcs320_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+
+ return 0;
+}
+
+static struct platform_driver dfmcs320_driver = {
+ .driver = {
+ .name = "dfbmcs320",
+ .owner = THIS_MODULE,
+ },
+ .probe = dfbmcs320_probe,
+ .remove = __devexit_p(dfbmcs320_remove),
+};
+
+static int __init dfbmcs320_init(void)
+{
+ return platform_driver_register(&dfmcs320_driver);
+}
+module_init(dfbmcs320_init);
+
+static void __exit dfbmcs320_exit(void)
+{
+ platform_driver_unregister(&dfmcs320_driver);
+}
+module_exit(dfbmcs320_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
new file mode 100644
index 0000000..72de47e
--- /dev/null
+++ b/sound/soc/codecs/lm4857.c
@@ -0,0 +1,276 @@
+/*
+ * LM4857 AMP driver
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Copyright 2011 Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * 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 <linux/init.h>
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+struct lm4857 {
+ struct i2c_client *i2c;
+ uint8_t mode;
+};
+
+static const uint8_t lm4857_default_regs[] = {
+ 0x00, 0x00, 0x00, 0x00,
+};
+
+/* The register offsets in the cache array */
+#define LM4857_MVOL 0
+#define LM4857_LVOL 1
+#define LM4857_RVOL 2
+#define LM4857_CTRL 3
+
+/* the shifts required to set these bits */
+#define LM4857_3D 5
+#define LM4857_WAKEUP 5
+#define LM4857_EPGAIN 4
+
+static int lm4857_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ uint8_t data;
+ int ret;
+
+ ret = snd_soc_cache_write(codec, reg, value);
+ if (ret < 0)
+ return ret;
+
+ data = (reg << 6) | value;
+ ret = i2c_master_send(codec->control_data, &data, 1);
+ if (ret != 1) {
+ dev_err(codec->dev, "Failed to write register: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static unsigned int lm4857_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ unsigned int val;
+ int ret;
+
+ ret = snd_soc_cache_read(codec, reg, &val);
+ if (ret)
+ return -1;
+
+ return val;
+}
+
+static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.integer.value[0] = lm4857->mode;
+
+ return 0;
+}
+
+static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
+ uint8_t value = ucontrol->value.integer.value[0];
+
+ lm4857->mode = value;
+
+ if (codec->dapm.bias_level == SND_SOC_BIAS_ON)
+ snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, value + 6);
+
+ return 1;
+}
+
+static int lm4857_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, lm4857->mode + 6);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, 0);
+ break;
+ default:
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+
+ return 0;
+}
+
+static const char *lm4857_mode[] = {
+ "Earpiece",
+ "Loudspeaker",
+ "Loudspeaker + Headphone",
+ "Headphone",
+};
+
+static const struct soc_enum lm4857_mode_enum =
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode);
+
+static const struct snd_soc_dapm_widget lm4857_dapm_widgets[] = {
+ SND_SOC_DAPM_INPUT("IN"),
+
+ SND_SOC_DAPM_OUTPUT("LS"),
+ SND_SOC_DAPM_OUTPUT("HP"),
+ SND_SOC_DAPM_OUTPUT("EP"),
+};
+
+static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0);
+static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0);
+
+static const struct snd_kcontrol_new lm4857_controls[] = {
+ SOC_SINGLE_TLV("Left Playback Volume", LM4857_LVOL, 0, 31, 0,
+ stereo_tlv),
+ SOC_SINGLE_TLV("Right Playback Volume", LM4857_RVOL, 0, 31, 0,
+ stereo_tlv),
+ SOC_SINGLE_TLV("Mono Playback Volume", LM4857_MVOL, 0, 31, 0,
+ mono_tlv),
+ SOC_SINGLE("Spk 3D Playback Switch", LM4857_LVOL, LM4857_3D, 1, 0),
+ SOC_SINGLE("HP 3D Playback Switch", LM4857_RVOL, LM4857_3D, 1, 0),
+ SOC_SINGLE("Fast Wakeup Playback Switch", LM4857_CTRL,
+ LM4857_WAKEUP, 1, 0),
+ SOC_SINGLE("Earpiece 6dB Playback Switch", LM4857_CTRL,
+ LM4857_EPGAIN, 1, 0),
+
+ SOC_ENUM_EXT("Mode", lm4857_mode_enum,
+ lm4857_get_mode, lm4857_set_mode),
+};
+
+/* There is a demux inbetween the the input signal and the output signals.
+ * Currently there is no easy way to model it in ASoC and since it does not make
+ * much of a difference in practice simply connect the input direclty to the
+ * outputs. */
+static const struct snd_soc_dapm_route lm4857_routes[] = {
+ {"LS", NULL, "IN"},
+ {"HP", NULL, "IN"},
+ {"EP", NULL, "IN"},
+};
+
+static int lm4857_probe(struct snd_soc_codec *codec)
+{
+ struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ int ret;
+
+ codec->control_data = lm4857->i2c;
+
+ ret = snd_soc_add_controls(codec, lm4857_controls,
+ ARRAY_SIZE(lm4857_controls));
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_new_controls(dapm, lm4857_dapm_widgets,
+ ARRAY_SIZE(lm4857_dapm_widgets));
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_add_routes(dapm, lm4857_routes,
+ ARRAY_SIZE(lm4857_routes));
+ if (ret)
+ return ret;
+
+ snd_soc_dapm_new_widgets(dapm);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_lm4857 = {
+ .write = lm4857_write,
+ .read = lm4857_read,
+ .probe = lm4857_probe,
+ .reg_cache_size = ARRAY_SIZE(lm4857_default_regs),
+ .reg_word_size = sizeof(uint8_t),
+ .reg_cache_default = lm4857_default_regs,
+ .set_bias_level = lm4857_set_bias_level,
+};
+
+static int __devinit lm4857_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct lm4857 *lm4857;
+ int ret;
+
+ lm4857 = kzalloc(sizeof(*lm4857), GFP_KERNEL);
+ if (!lm4857)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, lm4857);
+
+ lm4857->i2c = i2c;
+
+ ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0);
+
+ if (ret) {
+ kfree(lm4857);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int __devexit lm4857_i2c_remove(struct i2c_client *i2c)
+{
+ struct lm4857 *lm4857 = i2c_get_clientdata(i2c);
+
+ snd_soc_unregister_codec(&i2c->dev);
+ kfree(lm4857);
+
+ return 0;
+}
+
+static const struct i2c_device_id lm4857_i2c_id[] = {
+ { "lm4857", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, lm4857_i2c_id);
+
+static struct i2c_driver lm4857_i2c_driver = {
+ .driver = {
+ .name = "lm4857",
+ .owner = THIS_MODULE,
+ },
+ .probe = lm4857_i2c_probe,
+ .remove = __devexit_p(lm4857_i2c_remove),
+ .id_table = lm4857_i2c_id,
+};
+
+static int __init lm4857_init(void)
+{
+ return i2c_add_driver(&lm4857_i2c_driver);
+}
+module_init(lm4857_init);
+
+static void __exit lm4857_exit(void)
+{
+ i2c_del_driver(&lm4857_i2c_driver);
+}
+module_exit(lm4857_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("LM4857 amplifier driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
new file mode 100644
index 0000000..208d2ee
--- /dev/null
+++ b/sound/soc/codecs/max9850.c
@@ -0,0 +1,389 @@
+/*
+ * max9850.c -- codec driver for max9850
+ *
+ * Copyright (C) 2011 taskit GmbH
+ *
+ * Author: Christian Glindkamp <christian.glindkamp@taskit.de>
+ *
+ * Initial development of this code was funded by
+ * MICRONIC Computer Systeme GmbH, http://www.mcsberlin.de/
+ *
+ * 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 <linux/module.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "max9850.h"
+
+struct max9850_priv {
+ unsigned int sysclk;
+};
+
+/* max9850 register cache */
+static const u8 max9850_reg[MAX9850_CACHEREGNUM] = {
+ 0x00, 0x00, 0x0c, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
+};
+
+/* these registers are not used at the moment but provided for the sake of
+ * completeness */
+static int max9850_volatile_register(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ switch (reg) {
+ case MAX9850_STATUSA:
+ case MAX9850_STATUSB:
+ return 1;
+ default:
+ return 0;
+ }
+}
+
+static const unsigned int max9850_tlv[] = {
+ TLV_DB_RANGE_HEAD(4),
+ 0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0),
+ 0x20, 0x33, TLV_DB_SCALE_ITEM(-4150, 200, 0),
+ 0x34, 0x37, TLV_DB_SCALE_ITEM(-150, 100, 0),
+ 0x38, 0x3f, TLV_DB_SCALE_ITEM(250, 50, 0),
+};
+
+static const struct snd_kcontrol_new max9850_controls[] = {
+SOC_SINGLE_TLV("Headphone Volume", MAX9850_VOLUME, 0, 0x3f, 1, max9850_tlv),
+SOC_SINGLE("Headphone Switch", MAX9850_VOLUME, 7, 1, 1),
+SOC_SINGLE("Mono Switch", MAX9850_GENERAL_PURPOSE, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new max9850_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Line In Switch", MAX9850_ENABLE, 1, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget max9850_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("Charge Pump 1", MAX9850_ENABLE, 4, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("Charge Pump 2", MAX9850_ENABLE, 5, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MCLK", MAX9850_ENABLE, 6, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("SHDN", MAX9850_ENABLE, 7, 0, NULL, 0),
+SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", MAX9850_ENABLE, 2, 0,
+ &max9850_mixer_controls[0],
+ ARRAY_SIZE(max9850_mixer_controls)),
+SND_SOC_DAPM_PGA("Headphone Output", MAX9850_ENABLE, 3, 0, NULL, 0),
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", MAX9850_ENABLE, 0, 0),
+SND_SOC_DAPM_OUTPUT("OUTL"),
+SND_SOC_DAPM_OUTPUT("HPL"),
+SND_SOC_DAPM_OUTPUT("OUTR"),
+SND_SOC_DAPM_OUTPUT("HPR"),
+SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_INPUT("INL"),
+SND_SOC_DAPM_INPUT("INR"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ /* output mixer */
+ {"Output Mixer", NULL, "DAC"},
+ {"Output Mixer", "Line In Switch", "Line Input"},
+
+ /* outputs */
+ {"Headphone Output", NULL, "Output Mixer"},
+ {"HPL", NULL, "Headphone Output"},
+ {"HPR", NULL, "Headphone Output"},
+ {"OUTL", NULL, "Output Mixer"},
+ {"OUTR", NULL, "Output Mixer"},
+
+ /* inputs */
+ {"Line Input", NULL, "INL"},
+ {"Line Input", NULL, "INR"},
+
+ /* supplies */
+ {"Output Mixer", NULL, "Charge Pump 1"},
+ {"Output Mixer", NULL, "Charge Pump 2"},
+ {"Output Mixer", NULL, "SHDN"},
+ {"DAC", NULL, "MCLK"},
+};
+
+static int max9850_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec);
+ u64 lrclk_div;
+ u8 sf, da;
+
+ if (!max9850->sysclk)
+ return -EINVAL;
+
+ /* lrclk_div = 2^22 * rate / iclk with iclk = mclk / sf */
+ sf = (snd_soc_read(codec, MAX9850_CLOCK) >> 2) + 1;
+ lrclk_div = (1 << 22);
+ lrclk_div *= params_rate(params);
+ lrclk_div *= sf;
+ do_div(lrclk_div, max9850->sysclk);
+
+ snd_soc_write(codec, MAX9850_LRCLK_MSB, (lrclk_div >> 8) & 0x7f);
+ snd_soc_write(codec, MAX9850_LRCLK_LSB, lrclk_div & 0xff);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ da = 0;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ da = 0x2;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ da = 0x3;
+ break;
+ default:
+ return -EINVAL;
+ }
+ snd_soc_update_bits(codec, MAX9850_DIGITAL_AUDIO, 0x3, da);
+
+ return 0;
+}
+
+static int max9850_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 max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec);
+
+ /* calculate mclk -> iclk divider */
+ if (freq <= 13000000)
+ snd_soc_write(codec, MAX9850_CLOCK, 0x0);
+ else if (freq <= 26000000)
+ snd_soc_write(codec, MAX9850_CLOCK, 0x4);
+ else if (freq <= 40000000)
+ snd_soc_write(codec, MAX9850_CLOCK, 0x8);
+ else
+ return -EINVAL;
+
+ max9850->sysclk = freq;
+ return 0;
+}
+
+static int max9850_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u8 da = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ da |= MAX9850_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ da |= MAX9850_DLY;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ da |= MAX9850_RTJ;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ 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:
+ da |= MAX9850_BCINV | MAX9850_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ da |= MAX9850_BCINV;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ da |= MAX9850_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set da */
+ snd_soc_write(codec, MAX9850_DIGITAL_AUDIO, da);
+
+ return 0;
+}
+
+static int max9850_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+ ret = snd_soc_cache_sync(codec);
+ if (ret) {
+ dev_err(codec->dev,
+ "Failed to sync cache: %d\n", ret);
+ return ret;
+ }
+ }
+ break;
+ case SND_SOC_BIAS_OFF:
+ break;
+ }
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+#define MAX9850_RATES SNDRV_PCM_RATE_8000_48000
+
+#define MAX9850_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops max9850_dai_ops = {
+ .hw_params = max9850_hw_params,
+ .set_sysclk = max9850_set_dai_sysclk,
+ .set_fmt = max9850_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver max9850_dai = {
+ .name = "max9850-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = MAX9850_RATES,
+ .formats = MAX9850_FORMATS
+ },
+ .ops = &max9850_dai_ops,
+};
+
+#ifdef CONFIG_PM
+static int max9850_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+ max9850_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+static int max9850_resume(struct snd_soc_codec *codec)
+{
+ max9850_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+#else
+#define max9850_suspend NULL
+#define max9850_resume NULL
+#endif
+
+static int max9850_probe(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ int ret;
+
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ /* enable zero-detect */
+ snd_soc_update_bits(codec, MAX9850_GENERAL_PURPOSE, 1, 1);
+ /* enable slew-rate control */
+ snd_soc_update_bits(codec, MAX9850_VOLUME, 0x40, 0x40);
+ /* set slew-rate 125ms */
+ snd_soc_update_bits(codec, MAX9850_CHARGE_PUMP, 0xff, 0xc0);
+
+ snd_soc_dapm_new_controls(dapm, max9850_dapm_widgets,
+ ARRAY_SIZE(max9850_dapm_widgets));
+ snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
+
+ snd_soc_add_controls(codec, max9850_controls,
+ ARRAY_SIZE(max9850_controls));
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_max9850 = {
+ .probe = max9850_probe,
+ .suspend = max9850_suspend,
+ .resume = max9850_resume,
+ .set_bias_level = max9850_set_bias_level,
+ .reg_cache_size = ARRAY_SIZE(max9850_reg),
+ .reg_word_size = sizeof(u8),
+ .reg_cache_default = max9850_reg,
+ .volatile_register = max9850_volatile_register,
+};
+
+static int __devinit max9850_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct max9850_priv *max9850;
+ int ret;
+
+ max9850 = kzalloc(sizeof(struct max9850_priv), GFP_KERNEL);
+ if (max9850 == NULL)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, max9850);
+
+ ret = snd_soc_register_codec(&i2c->dev,
+ &soc_codec_dev_max9850, &max9850_dai, 1);
+ if (ret < 0)
+ kfree(max9850);
+ return ret;
+}
+
+static __devexit int max9850_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ kfree(i2c_get_clientdata(client));
+ return 0;
+}
+
+static const struct i2c_device_id max9850_i2c_id[] = {
+ { "max9850", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, max9850_i2c_id);
+
+static struct i2c_driver max9850_i2c_driver = {
+ .driver = {
+ .name = "max9850",
+ .owner = THIS_MODULE,
+ },
+ .probe = max9850_i2c_probe,
+ .remove = __devexit_p(max9850_i2c_remove),
+ .id_table = max9850_i2c_id,
+};
+
+static int __init max9850_init(void)
+{
+ return i2c_add_driver(&max9850_i2c_driver);
+}
+module_init(max9850_init);
+
+static void __exit max9850_exit(void)
+{
+ i2c_del_driver(&max9850_i2c_driver);
+}
+module_exit(max9850_exit);
+
+MODULE_AUTHOR("Christian Glindkamp <christian.glindkamp@taskit.de>");
+MODULE_DESCRIPTION("ASoC MAX9850 codec driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/max9850.h b/sound/soc/codecs/max9850.h
new file mode 100644
index 0000000..72b1ddb
--- /dev/null
+++ b/sound/soc/codecs/max9850.h
@@ -0,0 +1,38 @@
+/*
+ * max9850.h -- codec driver for max9850
+ *
+ * Copyright (C) 2011 taskit GmbH
+ * Author: Christian Glindkamp <christian.glindkamp@taskit.de>
+ *
+ * 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.
+ *
+ */
+
+#ifndef _MAX9850_H
+#define _MAX9850_H
+
+#define MAX9850_STATUSA 0x00
+#define MAX9850_STATUSB 0x01
+#define MAX9850_VOLUME 0x02
+#define MAX9850_GENERAL_PURPOSE 0x03
+#define MAX9850_INTERRUPT 0x04
+#define MAX9850_ENABLE 0x05
+#define MAX9850_CLOCK 0x06
+#define MAX9850_CHARGE_PUMP 0x07
+#define MAX9850_LRCLK_MSB 0x08
+#define MAX9850_LRCLK_LSB 0x09
+#define MAX9850_DIGITAL_AUDIO 0x0a
+
+#define MAX9850_CACHEREGNUM 11
+
+/* MAX9850_DIGITAL_AUDIO */
+#define MAX9850_MASTER (1<<7)
+#define MAX9850_INV (1<<6)
+#define MAX9850_BCINV (1<<5)
+#define MAX9850_DLY (1<<3)
+#define MAX9850_RTJ (1<<2)
+
+#endif
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
new file mode 100644
index 0000000..1f7217f
--- /dev/null
+++ b/sound/soc/codecs/sgtl5000.c
@@ -0,0 +1,1513 @@
+/*
+ * sgtl5000.c -- SGTL5000 ALSA SoC Audio driver
+ *
+ * Copyright 2010-2011 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * 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/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <linux/regulator/driver.h>
+#include <linux/regulator/machine.h>
+#include <linux/regulator/consumer.h>
+#include <sound/core.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "sgtl5000.h"
+
+#define SGTL5000_DAP_REG_OFFSET 0x0100
+#define SGTL5000_MAX_REG_OFFSET 0x013A
+
+/* default value of sgtl5000 registers except DAP */
+static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] = {
+ 0xa011, /* 0x0000, CHIP_ID. 11 stand for revison 17 */
+ 0x0000, /* 0x0002, CHIP_DIG_POWER. */
+ 0x0008, /* 0x0004, CHIP_CKL_CTRL */
+ 0x0010, /* 0x0006, CHIP_I2S_CTRL */
+ 0x0000, /* 0x0008, reserved */
+ 0x0008, /* 0x000A, CHIP_SSS_CTRL */
+ 0x0000, /* 0x000C, reserved */
+ 0x020c, /* 0x000E, CHIP_ADCDAC_CTRL */
+ 0x3c3c, /* 0x0010, CHIP_DAC_VOL */
+ 0x0000, /* 0x0012, reserved */
+ 0x015f, /* 0x0014, CHIP_PAD_STRENGTH */
+ 0x0000, /* 0x0016, reserved */
+ 0x0000, /* 0x0018, reserved */
+ 0x0000, /* 0x001A, reserved */
+ 0x0000, /* 0x001E, reserved */
+ 0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */
+ 0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */
+ 0x0111, /* 0x0024, CHIP_ANN_CTRL */
+ 0x0000, /* 0x0026, CHIP_LINREG_CTRL */
+ 0x0000, /* 0x0028, CHIP_REF_CTRL */
+ 0x0000, /* 0x002A, CHIP_MIC_CTRL */
+ 0x0000, /* 0x002C, CHIP_LINE_OUT_CTRL */
+ 0x0404, /* 0x002E, CHIP_LINE_OUT_VOL */
+ 0x7060, /* 0x0030, CHIP_ANA_POWER */
+ 0x5000, /* 0x0032, CHIP_PLL_CTRL */
+ 0x0000, /* 0x0034, CHIP_CLK_TOP_CTRL */
+ 0x0000, /* 0x0036, CHIP_ANA_STATUS */
+ 0x0000, /* 0x0038, reserved */
+ 0x0000, /* 0x003A, CHIP_ANA_TEST2 */
+ 0x0000, /* 0x003C, CHIP_SHORT_CTRL */
+ 0x0000, /* reserved */
+};
+
+/* default value of dap registers */
+static const u16 sgtl5000_dap_regs[] = {
+ 0x0000, /* 0x0100, DAP_CONTROL */
+ 0x0000, /* 0x0102, DAP_PEQ */
+ 0x0040, /* 0x0104, DAP_BASS_ENHANCE */
+ 0x051f, /* 0x0106, DAP_BASS_ENHANCE_CTRL */
+ 0x0000, /* 0x0108, DAP_AUDIO_EQ */
+ 0x0040, /* 0x010A, DAP_SGTL_SURROUND */
+ 0x0000, /* 0x010C, DAP_FILTER_COEF_ACCESS */
+ 0x0000, /* 0x010E, DAP_COEF_WR_B0_MSB */
+ 0x0000, /* 0x0110, DAP_COEF_WR_B0_LSB */
+ 0x0000, /* 0x0112, reserved */
+ 0x0000, /* 0x0114, reserved */
+ 0x002f, /* 0x0116, DAP_AUDIO_EQ_BASS_BAND0 */
+ 0x002f, /* 0x0118, DAP_AUDIO_EQ_BAND0 */
+ 0x002f, /* 0x011A, DAP_AUDIO_EQ_BAND2 */
+ 0x002f, /* 0x011C, DAP_AUDIO_EQ_BAND3 */
+ 0x002f, /* 0x011E, DAP_AUDIO_EQ_TREBLE_BAND4 */
+ 0x8000, /* 0x0120, DAP_MAIN_CHAN */
+ 0x0000, /* 0x0122, DAP_MIX_CHAN */
+ 0x0510, /* 0x0124, DAP_AVC_CTRL */
+ 0x1473, /* 0x0126, DAP_AVC_THRESHOLD */
+ 0x0028, /* 0x0128, DAP_AVC_ATTACK */
+ 0x0050, /* 0x012A, DAP_AVC_DECAY */
+ 0x0000, /* 0x012C, DAP_COEF_WR_B1_MSB */
+ 0x0000, /* 0x012E, DAP_COEF_WR_B1_LSB */
+ 0x0000, /* 0x0130, DAP_COEF_WR_B2_MSB */
+ 0x0000, /* 0x0132, DAP_COEF_WR_B2_LSB */
+ 0x0000, /* 0x0134, DAP_COEF_WR_A1_MSB */
+ 0x0000, /* 0x0136, DAP_COEF_WR_A1_LSB */
+ 0x0000, /* 0x0138, DAP_COEF_WR_A2_MSB */
+ 0x0000, /* 0x013A, DAP_COEF_WR_A2_LSB */
+};
+
+/* regulator supplies for sgtl5000, VDDD is an optional external supply */
+enum sgtl5000_regulator_supplies {
+ VDDA,
+ VDDIO,
+ VDDD,
+ SGTL5000_SUPPLY_NUM
+};
+
+/* vddd is optional supply */
+static const char *supply_names[SGTL5000_SUPPLY_NUM] = {
+ "VDDA",
+ "VDDIO",
+ "VDDD"
+};
+
+#define LDO_CONSUMER_NAME "VDDD_LDO"
+#define LDO_VOLTAGE 1200000
+
+static struct regulator_consumer_supply ldo_consumer[] = {
+ REGULATOR_SUPPLY(LDO_CONSUMER_NAME, NULL),
+};
+
+static struct regulator_init_data ldo_init_data = {
+ .constraints = {
+ .min_uV = 850000,
+ .max_uV = 1600000,
+ .valid_modes_mask = REGULATOR_MODE_NORMAL,
+ .valid_ops_mask = REGULATOR_CHANGE_STATUS,
+ },
+ .num_consumer_supplies = 1,
+ .consumer_supplies = &ldo_consumer[0],
+};
+
+/*
+ * sgtl5000 internal ldo regulator,
+ * enabled when VDDD not provided
+ */
+struct ldo_regulator {
+ struct regulator_desc desc;
+ struct regulator_dev *dev;
+ int voltage;
+ void *codec_data;
+ bool enabled;
+};
+
+/* sgtl5000 private structure in codec */
+struct sgtl5000_priv {
+ int sysclk; /* sysclk rate */
+ int master; /* i2s master or not */
+ int fmt; /* i2s data format */
+ struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM];
+ struct ldo_regulator *ldo;
+};
+
+/*
+ * mic_bias power on/off share the same register bits with
+ * output impedance of mic bias, when power on mic bias, we
+ * need reclaim it to impedance value.
+ * 0x0 = Powered off
+ * 0x1 = 2Kohm
+ * 0x2 = 4Kohm
+ * 0x3 = 8Kohm
+ */
+static int mic_bias_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ /* change mic bias resistor to 4Kohm */
+ snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL,
+ SGTL5000_BIAS_R_4k, SGTL5000_BIAS_R_4k);
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ /*
+ * SGTL5000_BIAS_R_8k as mask to clean the two bits
+ * of mic bias and output impedance
+ */
+ snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL,
+ SGTL5000_BIAS_R_8k, 0);
+ break;
+ }
+ return 0;
+}
+
+/*
+ * using codec assist to small pop, hp_powerup or lineout_powerup
+ * should stay setting until vag_powerup is fully ramped down,
+ * vag fully ramped down require 400ms.
+ */
+static int small_pop_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
+ SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
+ SGTL5000_VAG_POWERUP, 0);
+ msleep(400);
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+/* input sources for ADC */
+static const char *adc_mux_text[] = {
+ "MIC_IN", "LINE_IN"
+};
+
+static const struct soc_enum adc_enum =
+SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 2, 2, adc_mux_text);
+
+static const struct snd_kcontrol_new adc_mux =
+SOC_DAPM_ENUM("Capture Mux", adc_enum);
+
+/* input sources for DAC */
+static const char *dac_mux_text[] = {
+ "DAC", "LINE_IN"
+};
+
+static const struct soc_enum dac_enum =
+SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 6, 2, dac_mux_text);
+
+static const struct snd_kcontrol_new dac_mux =
+SOC_DAPM_ENUM("Headphone Mux", dac_enum);
+
+static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
+ SND_SOC_DAPM_INPUT("LINE_IN"),
+ SND_SOC_DAPM_INPUT("MIC_IN"),
+
+ SND_SOC_DAPM_OUTPUT("HP_OUT"),
+ SND_SOC_DAPM_OUTPUT("LINE_OUT"),
+
+ SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0,
+ mic_bias_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+ SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0,
+ small_pop_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+ SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0,
+ small_pop_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+
+ SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux),
+ SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux),
+
+ /* aif for i2s input */
+ SND_SOC_DAPM_AIF_IN("AIFIN", "Playback",
+ 0, SGTL5000_CHIP_DIG_POWER,
+ 0, 0),
+
+ /* aif for i2s output */
+ SND_SOC_DAPM_AIF_OUT("AIFOUT", "Capture",
+ 0, SGTL5000_CHIP_DIG_POWER,
+ 1, 0),
+
+ SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0),
+
+ SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0),
+};
+
+/* routes for sgtl5000 */
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */
+ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */
+
+ {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */
+ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */
+
+ {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */
+ {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */
+ {"LO", NULL, "DAC"}, /* dac --> line_out */
+
+ {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */
+ {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */
+
+ {"LINE_OUT", NULL, "LO"},
+ {"HP_OUT", NULL, "HP"},
+};
+
+/* custom function to fetch info of PCM playback volume */
+static int dac_info_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0xfc - 0x3c;
+ return 0;
+}
+
+/*
+ * custom function to get of PCM playback volume
+ *
+ * dac volume register
+ * 15-------------8-7--------------0
+ * | R channel vol | L channel vol |
+ * -------------------------------
+ *
+ * PCM volume with 0.5017 dB steps from 0 to -90 dB
+ *
+ * register values map to dB
+ * 0x3B and less = Reserved
+ * 0x3C = 0 dB
+ * 0x3D = -0.5 dB
+ * 0xF0 = -90 dB
+ * 0xFC and greater = Muted
+ *
+ * register value map to userspace value
+ *
+ * register value 0x3c(0dB) 0xf0(-90dB)0xfc
+ * ------------------------------
+ * userspace value 0xc0 0
+ */
+static int dac_get_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int reg;
+ int l;
+ int r;
+
+ reg = snd_soc_read(codec, SGTL5000_CHIP_DAC_VOL);
+
+ /* get left channel volume */
+ l = (reg & SGTL5000_DAC_VOL_LEFT_MASK) >> SGTL5000_DAC_VOL_LEFT_SHIFT;
+
+ /* get right channel volume */
+ r = (reg & SGTL5000_DAC_VOL_RIGHT_MASK) >> SGTL5000_DAC_VOL_RIGHT_SHIFT;
+
+ /* make sure value fall in (0x3c,0xfc) */
+ l = clamp(l, 0x3c, 0xfc);
+ r = clamp(r, 0x3c, 0xfc);
+
+ /* invert it and map to userspace value */
+ l = 0xfc - l;
+ r = 0xfc - r;
+
+ ucontrol->value.integer.value[0] = l;
+ ucontrol->value.integer.value[1] = r;
+
+ return 0;
+}
+
+/*
+ * custom function to put of PCM playback volume
+ *
+ * dac volume register
+ * 15-------------8-7--------------0
+ * | R channel vol | L channel vol |
+ * -------------------------------
+ *
+ * PCM volume with 0.5017 dB steps from 0 to -90 dB
+ *
+ * register values map to dB
+ * 0x3B and less = Reserved
+ * 0x3C = 0 dB
+ * 0x3D = -0.5 dB
+ * 0xF0 = -90 dB
+ * 0xFC and greater = Muted
+ *
+ * userspace value map to register value
+ *
+ * userspace value 0xc0 0
+ * ------------------------------
+ * register value 0x3c(0dB) 0xf0(-90dB)0xfc
+ */
+static int dac_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ int reg;
+ int l;
+ int r;
+
+ l = ucontrol->value.integer.value[0];
+ r = ucontrol->value.integer.value[1];
+
+ /* make sure userspace volume fall in (0, 0xfc-0x3c) */
+ l = clamp(l, 0, 0xfc - 0x3c);
+ r = clamp(r, 0, 0xfc - 0x3c);
+
+ /* invert it, get the value can be set to register */
+ l = 0xfc - l;
+ r = 0xfc - r;
+
+ /* shift to get the register value */
+ reg = l << SGTL5000_DAC_VOL_LEFT_SHIFT |
+ r << SGTL5000_DAC_VOL_RIGHT_SHIFT;
+
+ snd_soc_write(codec, SGTL5000_CHIP_DAC_VOL, reg);
+
+ return 0;
+}
+
+static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0);
+
+/* tlv for mic gain, 0db 20db 30db 40db */
+static const unsigned int mic_gain_tlv[] = {
+ TLV_DB_RANGE_HEAD(4),
+ 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+ 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0),
+};
+
+/* tlv for hp volume, -51.5db to 12.0db, step .5db */
+static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0);
+
+static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
+ /* SOC_DOUBLE_S8_TLV with invert */
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = dac_info_volsw,
+ .get = dac_get_volsw,
+ .put = dac_put_volsw,
+ },
+
+ SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0),
+ SOC_SINGLE_TLV("Capture Attenuate Switch (-6dB)",
+ SGTL5000_CHIP_ANA_ADC_CTRL,
+ 8, 2, 0, capture_6db_attenuate),
+ SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0),
+
+ SOC_DOUBLE_TLV("Headphone Playback Volume",
+ SGTL5000_CHIP_ANA_HP_CTRL,
+ 0, 8,
+ 0x7f, 1,
+ headphone_volume),
+ SOC_SINGLE("Headphone Playback ZC Switch", SGTL5000_CHIP_ANA_CTRL,
+ 5, 1, 0),
+
+ SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL,
+ 0, 4, 0, mic_gain_tlv),
+};
+
+/* mute the codec used by alsa core */
+static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 adcdac_ctrl = SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT;
+
+ snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL,
+ adcdac_ctrl, mute ? adcdac_ctrl : 0);
+
+ return 0;
+}
+
+/* set codec format */
+static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+ u16 i2sctl = 0;
+
+ sgtl5000->master = 0;
+ /*
+ * i2s clock and frame master setting.
+ * ONLY support:
+ * - clock and frame slave,
+ * - clock and frame master
+ */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ i2sctl |= SGTL5000_I2S_MASTER;
+ sgtl5000->master = 1;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* setting i2s data format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ i2sctl |= SGTL5000_I2S_MODE_PCM;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ i2sctl |= SGTL5000_I2S_MODE_PCM;
+ i2sctl |= SGTL5000_I2S_LRALIGN;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ i2sctl |= SGTL5000_I2S_MODE_I2S_LJ;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ i2sctl |= SGTL5000_I2S_MODE_RJ;
+ i2sctl |= SGTL5000_I2S_LRPOL;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ i2sctl |= SGTL5000_I2S_MODE_I2S_LJ;
+ i2sctl |= SGTL5000_I2S_LRALIGN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ sgtl5000->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
+ /* Clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ i2sctl |= SGTL5000_I2S_SCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, SGTL5000_CHIP_I2S_CTRL, i2sctl);
+
+ return 0;
+}
+
+/* set codec sysclk */
+static int sgtl5000_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 sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+ switch (clk_id) {
+ case SGTL5000_SYSCLK:
+ sgtl5000->sysclk = freq;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+/*
+ * set clock according to i2s frame clock,
+ * sgtl5000 provide 2 clock sources.
+ * 1. sys_mclk. sample freq can only configure to
+ * 1/256, 1/384, 1/512 of sys_mclk.
+ * 2. pll. can derive any audio clocks.
+ *
+ * clock setting rules:
+ * 1. in slave mode, only sys_mclk can use.
+ * 2. as constraint by sys_mclk, sample freq should
+ * set to 32k, 44.1k and above.
+ * 3. using sys_mclk prefer to pll to save power.
+ */
+static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate)
+{
+ struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+ int clk_ctl = 0;
+ int sys_fs; /* sample freq */
+
+ /*
+ * sample freq should be divided by frame clock,
+ * if frame clock lower than 44.1khz, sample feq should set to
+ * 32khz or 44.1khz.
+ */
+ switch (frame_rate) {
+ case 8000:
+ case 16000:
+ sys_fs = 32000;
+ break;
+ case 11025:
+ case 22050:
+ sys_fs = 44100;
+ break;
+ default:
+ sys_fs = frame_rate;
+ break;
+ }
+
+ /* set divided factor of frame clock */
+ switch (sys_fs / frame_rate) {
+ case 4:
+ clk_ctl |= SGTL5000_RATE_MODE_DIV_4 << SGTL5000_RATE_MODE_SHIFT;
+ break;
+ case 2:
+ clk_ctl |= SGTL5000_RATE_MODE_DIV_2 << SGTL5000_RATE_MODE_SHIFT;
+ break;
+ case 1:
+ clk_ctl |= SGTL5000_RATE_MODE_DIV_1 << SGTL5000_RATE_MODE_SHIFT;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set the sys_fs according to frame rate */
+ switch (sys_fs) {
+ case 32000:
+ clk_ctl |= SGTL5000_SYS_FS_32k << SGTL5000_SYS_FS_SHIFT;
+ break;
+ case 44100:
+ clk_ctl |= SGTL5000_SYS_FS_44_1k << SGTL5000_SYS_FS_SHIFT;
+ break;
+ case 48000:
+ clk_ctl |= SGTL5000_SYS_FS_48k << SGTL5000_SYS_FS_SHIFT;
+ break;
+ case 96000:
+ clk_ctl |= SGTL5000_SYS_FS_96k << SGTL5000_SYS_FS_SHIFT;
+ break;
+ default:
+ dev_err(codec->dev, "frame rate %d not supported\n",
+ frame_rate);
+ return -EINVAL;
+ }
+
+ /*
+ * calculate the divider of mclk/sample_freq,
+ * factor of freq =96k can only be 256, since mclk in range (12m,27m)
+ */
+ switch (sgtl5000->sysclk / sys_fs) {
+ case 256:
+ clk_ctl |= SGTL5000_MCLK_FREQ_256FS <<
+ SGTL5000_MCLK_FREQ_SHIFT;
+ break;
+ case 384:
+ clk_ctl |= SGTL5000_MCLK_FREQ_384FS <<
+ SGTL5000_MCLK_FREQ_SHIFT;
+ break;
+ case 512:
+ clk_ctl |= SGTL5000_MCLK_FREQ_512FS <<
+ SGTL5000_MCLK_FREQ_SHIFT;
+ break;
+ default:
+ /* if mclk not satisify the divider, use pll */
+ if (sgtl5000->master) {
+ clk_ctl |= SGTL5000_MCLK_FREQ_PLL <<
+ SGTL5000_MCLK_FREQ_SHIFT;
+ } else {
+ dev_err(codec->dev,
+ "PLL not supported in slave mode\n");
+ return -EINVAL;
+ }
+ }
+
+ /* if using pll, please check manual 6.4.2 for detail */
+ if ((clk_ctl & SGTL5000_MCLK_FREQ_MASK) == SGTL5000_MCLK_FREQ_PLL) {
+ u64 out, t;
+ int div2;
+ int pll_ctl;
+ unsigned int in, int_div, frac_div;
+
+ if (sgtl5000->sysclk > 17000000) {
+ div2 = 1;
+ in = sgtl5000->sysclk / 2;
+ } else {
+ div2 = 0;
+ in = sgtl5000->sysclk;
+ }
+ if (sys_fs == 44100)
+ out = 180633600;
+ else
+ out = 196608000;
+ t = do_div(out, in);
+ int_div = out;
+ t *= 2048;
+ do_div(t, in);
+ frac_div = t;
+ pll_ctl = int_div << SGTL5000_PLL_INT_DIV_SHIFT |
+ frac_div << SGTL5000_PLL_FRAC_DIV_SHIFT;
+
+ snd_soc_write(codec, SGTL5000_CHIP_PLL_CTRL, pll_ctl);
+ if (div2)
+ snd_soc_update_bits(codec,
+ SGTL5000_CHIP_CLK_TOP_CTRL,
+ SGTL5000_INPUT_FREQ_DIV2,
+ SGTL5000_INPUT_FREQ_DIV2);
+ else
+ snd_soc_update_bits(codec,
+ SGTL5000_CHIP_CLK_TOP_CTRL,
+ SGTL5000_INPUT_FREQ_DIV2,
+ 0);
+
+ /* power up pll */
+ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+ SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP,
+ SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP);
+ } else {
+ /* power down pll */
+ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+ SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP,
+ 0);
+ }
+
+ /* if using pll, clk_ctrl must be set after pll power up */
+ snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl);
+
+ return 0;
+}
+
+/*
+ * Set PCM DAI bit size and sample rate.
+ * input: params_rate, params_fmt
+ */
+static int sgtl5000_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 sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+ int channels = params_channels(params);
+ int i2s_ctl = 0;
+ int stereo;
+ int ret;
+
+ /* sysclk should already set */
+ if (!sgtl5000->sysclk) {
+ dev_err(codec->dev, "%s: set sysclk first!\n", __func__);
+ return -EFAULT;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ stereo = SGTL5000_DAC_STEREO;
+ else
+ stereo = SGTL5000_ADC_STEREO;
+
+ /* set mono to save power */
+ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, stereo,
+ channels == 1 ? 0 : stereo);
+
+ /* set codec clock base on lrclk */
+ ret = sgtl5000_set_clock(codec, params_rate(params));
+ if (ret)
+ return ret;
+
+ /* set i2s data format */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J)
+ return -EINVAL;
+ i2s_ctl |= SGTL5000_I2S_DLEN_16 << SGTL5000_I2S_DLEN_SHIFT;
+ i2s_ctl |= SGTL5000_I2S_SCLKFREQ_32FS <<
+ SGTL5000_I2S_SCLKFREQ_SHIFT;
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ i2s_ctl |= SGTL5000_I2S_DLEN_20 << SGTL5000_I2S_DLEN_SHIFT;
+ i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS <<
+ SGTL5000_I2S_SCLKFREQ_SHIFT;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ i2s_ctl |= SGTL5000_I2S_DLEN_24 << SGTL5000_I2S_DLEN_SHIFT;
+ i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS <<
+ SGTL5000_I2S_SCLKFREQ_SHIFT;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J)
+ return -EINVAL;
+ i2s_ctl |= SGTL5000_I2S_DLEN_32 << SGTL5000_I2S_DLEN_SHIFT;
+ i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS <<
+ SGTL5000_I2S_SCLKFREQ_SHIFT;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, i2s_ctl, i2s_ctl);
+
+ return 0;
+}
+
+static int ldo_regulator_is_enabled(struct regulator_dev *dev)
+{
+ struct ldo_regulator *ldo = rdev_get_drvdata(dev);
+
+ return ldo->enabled;
+}
+
+static int ldo_regulator_enable(struct regulator_dev *dev)
+{
+ struct ldo_regulator *ldo = rdev_get_drvdata(dev);
+ struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data;
+ int reg;
+
+ if (ldo_regulator_is_enabled(dev))
+ return 0;
+
+ /* set regulator value firstly */
+ reg = (1600 - ldo->voltage / 1000) / 50;
+ reg = clamp(reg, 0x0, 0xf);
+
+ /* amend the voltage value, unit: uV */
+ ldo->voltage = (1600 - reg * 50) * 1000;
+
+ /* set voltage to register */
+ snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL,
+ (0x1 << 4) - 1, reg);
+
+ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+ SGTL5000_LINEREG_D_POWERUP,
+ SGTL5000_LINEREG_D_POWERUP);
+
+ /* when internal ldo enabled, simple digital power can be disabled */
+ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+ SGTL5000_LINREG_SIMPLE_POWERUP,
+ 0);
+
+ ldo->enabled = 1;
+ return 0;
+}
+
+static int ldo_regulator_disable(struct regulator_dev *dev)
+{
+ struct ldo_regulator *ldo = rdev_get_drvdata(dev);
+ struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data;
+
+ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+ SGTL5000_LINEREG_D_POWERUP,
+ 0);
+
+ /* clear voltage info */
+ snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL,
+ (0x1 << 4) - 1, 0);
+
+ ldo->enabled = 0;
+
+ return 0;
+}
+
+static int ldo_regulator_get_voltage(struct regulator_dev *dev)
+{
+ struct ldo_regulator *ldo = rdev_get_drvdata(dev);
+
+ return ldo->voltage;
+}
+
+static struct regulator_ops ldo_regulator_ops = {
+ .is_enabled = ldo_regulator_is_enabled,
+ .enable = ldo_regulator_enable,
+ .disable = ldo_regulator_disable,
+ .get_voltage = ldo_regulator_get_voltage,
+};
+
+static int ldo_regulator_register(struct snd_soc_codec *codec,
+ struct regulator_init_data *init_data,
+ int voltage)
+{
+ struct ldo_regulator *ldo;
+
+ ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL);
+
+ if (!ldo) {
+ dev_err(codec->dev, "failed to allocate ldo_regulator\n");
+ return -ENOMEM;
+ }
+
+ ldo->desc.name = kstrdup(dev_name(codec->dev), GFP_KERNEL);
+ if (!ldo->desc.name) {
+ kfree(ldo);
+ dev_err(codec->dev, "failed to allocate decs name memory\n");
+ return -ENOMEM;
+ }
+
+ ldo->desc.type = REGULATOR_VOLTAGE;
+ ldo->desc.owner = THIS_MODULE;
+ ldo->desc.ops = &ldo_regulator_ops;
+ ldo->desc.n_voltages = 1;
+
+ ldo->codec_data = codec;
+ ldo->voltage = voltage;
+
+ ldo->dev = regulator_register(&ldo->desc, codec->dev,
+ init_data, ldo);
+ if (IS_ERR(ldo->dev)) {
+ int ret = PTR_ERR(ldo->dev);
+
+ dev_err(codec->dev, "failed to register regulator\n");
+ kfree(ldo->desc.name);
+ kfree(ldo);
+
+ return ret;
+ }
+
+ return 0;
+}
+
+static int ldo_regulator_remove(struct snd_soc_codec *codec)
+{
+ struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+ struct ldo_regulator *ldo = sgtl5000->ldo;
+
+ if (!ldo)
+ return 0;
+
+ regulator_unregister(ldo->dev);
+ kfree(ldo->desc.name);
+ kfree(ldo);
+
+ return 0;
+}
+
+/*
+ * set dac bias
+ * common state changes:
+ * startup:
+ * off --> standby --> prepare --> on
+ * standby --> prepare --> on
+ *
+ * stop:
+ * on --> prepare --> standby
+ */
+static int sgtl5000_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ int ret;
+ struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+ ret = regulator_bulk_enable(
+ ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
+ if (ret)
+ return ret;
+ udelay(10);
+ }
+
+ break;
+ case SND_SOC_BIAS_OFF:
+ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
+ break;
+ }
+
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+#define SGTL5000_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE |\
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops sgtl5000_ops = {
+ .hw_params = sgtl5000_pcm_hw_params,
+ .digital_mute = sgtl5000_digital_mute,
+ .set_fmt = sgtl5000_set_dai_fmt,
+ .set_sysclk = sgtl5000_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver sgtl5000_dai = {
+ .name = "sgtl5000",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ /*
+ * only support 8~48K + 96K,
+ * TODO modify hw_param to support more
+ */
+ .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000,
+ .formats = SGTL5000_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000,
+ .formats = SGTL5000_FORMATS,
+ },
+ .ops = &sgtl5000_ops,
+ .symmetric_rates = 1,
+};
+
+static int sgtl5000_volatile_register(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ switch (reg) {
+ case SGTL5000_CHIP_ID:
+ case SGTL5000_CHIP_ADCDAC_CTRL:
+ case SGTL5000_CHIP_ANA_STATUS:
+ return 1;
+ }
+
+ return 0;
+}
+
+#ifdef CONFIG_SUSPEND
+static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+ sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ return 0;
+}
+
+/*
+ * restore all sgtl5000 registers,
+ * since a big hole between dap and regular registers,
+ * we will restore them respectively.
+ */
+static int sgtl5000_restore_regs(struct snd_soc_codec *codec)
+{
+ u16 *cache = codec->reg_cache;
+ int i;
+ int regular_regs = SGTL5000_CHIP_SHORT_CTRL >> 1;
+
+ /* restore regular registers */
+ for (i = 0; i < regular_regs; i++) {
+ int reg = i << 1;
+
+ /* this regs depends on the others */
+ if (reg == SGTL5000_CHIP_ANA_POWER ||
+ reg == SGTL5000_CHIP_CLK_CTRL ||
+ reg == SGTL5000_CHIP_LINREG_CTRL ||
+ reg == SGTL5000_CHIP_LINE_OUT_CTRL ||
+ reg == SGTL5000_CHIP_CLK_CTRL)
+ continue;
+
+ snd_soc_write(codec, reg, cache[i]);
+ }
+
+ /* restore dap registers */
+ for (i = SGTL5000_DAP_REG_OFFSET >> 1;
+ i < SGTL5000_MAX_REG_OFFSET >> 1; i++) {
+ int reg = i << 1;
+
+ snd_soc_write(codec, reg, cache[i]);
+ }
+
+ /*
+ * restore power and other regs according
+ * to set_power() and set_clock()
+ */
+ snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL,
+ cache[SGTL5000_CHIP_LINREG_CTRL >> 1]);
+
+ snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER,
+ cache[SGTL5000_CHIP_ANA_POWER >> 1]);
+
+ snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL,
+ cache[SGTL5000_CHIP_CLK_CTRL >> 1]);
+
+ snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL,
+ cache[SGTL5000_CHIP_REF_CTRL >> 1]);
+
+ snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL,
+ cache[SGTL5000_CHIP_LINE_OUT_CTRL >> 1]);
+ return 0;
+}
+
+static int sgtl5000_resume(struct snd_soc_codec *codec)
+{
+ /* Bring the codec back up to standby to enable regulators */
+ sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* Restore registers by cached in memory */
+ sgtl5000_restore_regs(codec);
+ return 0;
+}
+#else
+#define sgtl5000_suspend NULL
+#define sgtl5000_resume NULL
+#endif /* CONFIG_SUSPEND */
+
+/*
+ * sgtl5000 has 3 internal power supplies:
+ * 1. VAG, normally set to vdda/2
+ * 2. chargepump, set to different value
+ * according to voltage of vdda and vddio
+ * 3. line out VAG, normally set to vddio/2
+ *
+ * and should be set according to:
+ * 1. vddd provided by external or not
+ * 2. vdda and vddio voltage value. > 3.1v or not
+ * 3. chip revision >=0x11 or not. If >=0x11, not use external vddd.
+ */
+static int sgtl5000_set_power_regs(struct snd_soc_codec *codec)
+{
+ int vddd;
+ int vdda;
+ int vddio;
+ u16 ana_pwr;
+ u16 lreg_ctrl;
+ int vag;
+ struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+ vdda = regulator_get_voltage(sgtl5000->supplies[VDDA].consumer);
+ vddio = regulator_get_voltage(sgtl5000->supplies[VDDIO].consumer);
+ vddd = regulator_get_voltage(sgtl5000->supplies[VDDD].consumer);
+
+ vdda = vdda / 1000;
+ vddio = vddio / 1000;
+ vddd = vddd / 1000;
+
+ if (vdda <= 0 || vddio <= 0 || vddd < 0) {
+ dev_err(codec->dev, "regulator voltage not set correctly\n");
+
+ return -EINVAL;
+ }
+
+ /* according to datasheet, maximum voltage of supplies */
+ if (vdda > 3600 || vddio > 3600 || vddd > 1980) {
+ dev_err(codec->dev,
+ "exceed max voltage vdda %dmv vddio %dma vddd %dma\n",
+ vdda, vddio, vddd);
+
+ return -EINVAL;
+ }
+
+ /* reset value */
+ ana_pwr = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER);
+ ana_pwr |= SGTL5000_DAC_STEREO |
+ SGTL5000_ADC_STEREO |
+ SGTL5000_REFTOP_POWERUP;
+ lreg_ctrl = snd_soc_read(codec, SGTL5000_CHIP_LINREG_CTRL);
+
+ if (vddio < 3100 && vdda < 3100) {
+ /* enable internal oscillator used for charge pump */
+ snd_soc_update_bits(codec, SGTL5000_CHIP_CLK_TOP_CTRL,
+ SGTL5000_INT_OSC_EN,
+ SGTL5000_INT_OSC_EN);
+ /* Enable VDDC charge pump */
+ ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP;
+ } else if (vddio >= 3100 && vdda >= 3100) {
+ /*
+ * if vddio and vddd > 3.1v,
+ * charge pump should be clean before set ana_pwr
+ */
+ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+ SGTL5000_VDDC_CHRGPMP_POWERUP, 0);
+
+ /* VDDC use VDDIO rail */
+ lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD;
+ lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO <<
+ SGTL5000_VDDC_MAN_ASSN_SHIFT;
+ }
+
+ snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl);
+
+ snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr);
+
+ /* set voltage to register */
+ snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL,
+ (0x1 << 4) - 1, 0x8);
+
+ /*
+ * if vddd linear reg has been enabled,
+ * simple digital supply should be clear to get
+ * proper VDDD voltage.
+ */
+ if (ana_pwr & SGTL5000_LINEREG_D_POWERUP)
+ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+ SGTL5000_LINREG_SIMPLE_POWERUP,
+ 0);
+ else
+ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+ SGTL5000_LINREG_SIMPLE_POWERUP |
+ SGTL5000_STARTUP_POWERUP,
+ 0);
+
+ /*
+ * set ADC/DAC VAG to vdda / 2,
+ * should stay in range (0.8v, 1.575v)
+ */
+ vag = vdda / 2;
+ if (vag <= SGTL5000_ANA_GND_BASE)
+ vag = 0;
+ else if (vag >= SGTL5000_ANA_GND_BASE + SGTL5000_ANA_GND_STP *
+ (SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT))
+ vag = SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT;
+ else
+ vag = (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP;
+
+ snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL,
+ vag << SGTL5000_ANA_GND_SHIFT,
+ vag << SGTL5000_ANA_GND_SHIFT);
+
+ /* set line out VAG to vddio / 2, in range (0.8v, 1.675v) */
+ vag = vddio / 2;
+ if (vag <= SGTL5000_LINE_OUT_GND_BASE)
+ vag = 0;
+ else if (vag >= SGTL5000_LINE_OUT_GND_BASE +
+ SGTL5000_LINE_OUT_GND_STP * SGTL5000_LINE_OUT_GND_MAX)
+ vag = SGTL5000_LINE_OUT_GND_MAX;
+ else
+ vag = (vag - SGTL5000_LINE_OUT_GND_BASE) /
+ SGTL5000_LINE_OUT_GND_STP;
+
+ snd_soc_update_bits(codec, SGTL5000_CHIP_LINE_OUT_CTRL,
+ vag << SGTL5000_LINE_OUT_GND_SHIFT |
+ SGTL5000_LINE_OUT_CURRENT_360u <<
+ SGTL5000_LINE_OUT_CURRENT_SHIFT,
+ vag << SGTL5000_LINE_OUT_GND_SHIFT |
+ SGTL5000_LINE_OUT_CURRENT_360u <<
+ SGTL5000_LINE_OUT_CURRENT_SHIFT);
+
+ return 0;
+}
+
+static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
+{
+ u16 reg;
+ int ret;
+ int rev;
+ int i;
+ int external_vddd = 0;
+ struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+ for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++)
+ sgtl5000->supplies[i].supply = supply_names[i];
+
+ ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
+ if (!ret)
+ external_vddd = 1;
+ else {
+ /* set internal ldo to 1.2v */
+ int voltage = LDO_VOLTAGE;
+
+ ret = ldo_regulator_register(codec, &ldo_init_data, voltage);
+ if (ret) {
+ dev_err(codec->dev,
+ "Failed to register vddd internal supplies: %d\n",
+ ret);
+ return ret;
+ }
+
+ sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME;
+
+ ret = regulator_bulk_get(codec->dev,
+ ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
+
+ if (ret) {
+ ldo_regulator_remove(codec);
+ dev_err(codec->dev,
+ "Failed to request supplies: %d\n", ret);
+
+ return ret;
+ }
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
+ if (ret)
+ goto err_regulator_free;
+
+ /* wait for all power rails bring up */
+ udelay(10);
+
+ /* read chip information */
+ reg = snd_soc_read(codec, SGTL5000_CHIP_ID);
+ if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) !=
+ SGTL5000_PARTID_PART_ID) {
+ dev_err(codec->dev,
+ "Device with ID register %x is not a sgtl5000\n", reg);
+ ret = -ENODEV;
+ goto err_regulator_disable;
+ }
+
+ rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT;
+ dev_info(codec->dev, "sgtl5000 revision %d\n", rev);
+
+ /*
+ * workaround for revision 0x11 and later,
+ * roll back to use internal LDO
+ */
+ if (external_vddd && rev >= 0x11) {
+ int voltage = LDO_VOLTAGE;
+ /* disable all regulator first */
+ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
+ /* free VDDD regulator */
+ regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
+
+ ret = ldo_regulator_register(codec, &ldo_init_data, voltage);
+ if (ret)
+ return ret;
+
+ sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME;
+
+ ret = regulator_bulk_get(codec->dev,
+ ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
+ if (ret) {
+ ldo_regulator_remove(codec);
+ dev_err(codec->dev,
+ "Failed to request supplies: %d\n", ret);
+
+ return ret;
+ }
+
+ ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
+ if (ret)
+ goto err_regulator_free;
+
+ /* wait for all power rails bring up */
+ udelay(10);
+ }
+
+ return 0;
+
+err_regulator_disable:
+ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
+err_regulator_free:
+ regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
+ if (external_vddd)
+ ldo_regulator_remove(codec);
+ return ret;
+
+}
+
+static int sgtl5000_probe(struct snd_soc_codec *codec)
+{
+ int ret;
+ struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+ /* setup i2c data ops */
+ ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ ret = sgtl5000_enable_regulators(codec);
+ if (ret)
+ return ret;
+
+ /* power up sgtl5000 */
+ ret = sgtl5000_set_power_regs(codec);
+ if (ret)
+ goto err;
+
+ /* enable small pop, introduce 400ms delay in turning off */
+ snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL,
+ SGTL5000_SMALL_POP,
+ SGTL5000_SMALL_POP);
+
+ /* disable short cut detector */
+ snd_soc_write(codec, SGTL5000_CHIP_SHORT_CTRL, 0);
+
+ /*
+ * set i2s as default input of sound switch
+ * TODO: add sound switch to control and dapm widge.
+ */
+ snd_soc_write(codec, SGTL5000_CHIP_SSS_CTRL,
+ SGTL5000_DAC_SEL_I2S_IN << SGTL5000_DAC_SEL_SHIFT);
+ snd_soc_write(codec, SGTL5000_CHIP_DIG_POWER,
+ SGTL5000_ADC_EN | SGTL5000_DAC_EN);
+
+ /* enable dac volume ramp by default */
+ snd_soc_write(codec, SGTL5000_CHIP_ADCDAC_CTRL,
+ SGTL5000_DAC_VOL_RAMP_EN |
+ SGTL5000_DAC_MUTE_RIGHT |
+ SGTL5000_DAC_MUTE_LEFT);
+
+ snd_soc_write(codec, SGTL5000_CHIP_PAD_STRENGTH, 0x015f);
+
+ snd_soc_write(codec, SGTL5000_CHIP_ANA_CTRL,
+ SGTL5000_HP_ZCD_EN |
+ SGTL5000_ADC_ZCD_EN);
+
+ snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0);
+
+ /*
+ * disable DAP
+ * TODO:
+ * Enable DAP in kcontrol and dapm.
+ */
+ snd_soc_write(codec, SGTL5000_DAP_CTRL, 0);
+
+ /* leading to standby state */
+ ret = sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ if (ret)
+ goto err;
+
+ snd_soc_add_controls(codec, sgtl5000_snd_controls,
+ ARRAY_SIZE(sgtl5000_snd_controls));
+
+ snd_soc_dapm_new_controls(&codec->dapm, sgtl5000_dapm_widgets,
+ ARRAY_SIZE(sgtl5000_dapm_widgets));
+
+ snd_soc_dapm_add_routes(&codec->dapm, audio_map,
+ ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_new_widgets(&codec->dapm);
+
+ return 0;
+
+err:
+ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
+ regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
+ ldo_regulator_remove(codec);
+
+ return ret;
+}
+
+static int sgtl5000_remove(struct snd_soc_codec *codec)
+{
+ struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+ sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
+ regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+ sgtl5000->supplies);
+ ldo_regulator_remove(codec);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver sgtl5000_driver = {
+ .probe = sgtl5000_probe,
+ .remove = sgtl5000_remove,
+ .suspend = sgtl5000_suspend,
+ .resume = sgtl5000_resume,
+ .set_bias_level = sgtl5000_set_bias_level,
+ .reg_cache_size = ARRAY_SIZE(sgtl5000_regs),
+ .reg_word_size = sizeof(u16),
+ .reg_cache_step = 2,
+ .reg_cache_default = sgtl5000_regs,
+ .volatile_register = sgtl5000_volatile_register,
+};
+
+static __devinit int sgtl5000_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct sgtl5000_priv *sgtl5000;
+ int ret;
+
+ sgtl5000 = kzalloc(sizeof(struct sgtl5000_priv), GFP_KERNEL);
+ if (!sgtl5000)
+ return -ENOMEM;
+
+ /*
+ * copy DAP default values to default value array.
+ * sgtl5000 register space has a big hole, merge it
+ * at init phase makes life easy.
+ * FIXME: should we drop 'const' of sgtl5000_regs?
+ */
+ memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)),
+ sgtl5000_dap_regs,
+ SGTL5000_MAX_REG_OFFSET - SGTL5000_DAP_REG_OFFSET);
+
+ i2c_set_clientdata(client, sgtl5000);
+
+ ret = snd_soc_register_codec(&client->dev,
+ &sgtl5000_driver, &sgtl5000_dai, 1);
+ if (ret) {
+ dev_err(&client->dev, "Failed to register codec: %d\n", ret);
+ kfree(sgtl5000);
+ return ret;
+ }
+
+ return 0;
+}
+
+static __devexit int sgtl5000_i2c_remove(struct i2c_client *client)
+{
+ struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client);
+
+ snd_soc_unregister_codec(&client->dev);
+
+ kfree(sgtl5000);
+ return 0;
+}
+
+static const struct i2c_device_id sgtl5000_id[] = {
+ {"sgtl5000", 0},
+ {},
+};
+
+MODULE_DEVICE_TABLE(i2c, sgtl5000_id);
+
+static struct i2c_driver sgtl5000_i2c_driver = {
+ .driver = {
+ .name = "sgtl5000",
+ .owner = THIS_MODULE,
+ },
+ .probe = sgtl5000_i2c_probe,
+ .remove = __devexit_p(sgtl5000_i2c_remove),
+ .id_table = sgtl5000_id,
+};
+
+static int __init sgtl5000_modinit(void)
+{
+ return i2c_add_driver(&sgtl5000_i2c_driver);
+}
+module_init(sgtl5000_modinit);
+
+static void __exit sgtl5000_exit(void)
+{
+ i2c_del_driver(&sgtl5000_i2c_driver);
+}
+module_exit(sgtl5000_exit);
+
+MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver");
+MODULE_AUTHOR("Zeng Zhaoming <zhaoming.zeng@freescale.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h
new file mode 100644
index 0000000..eec3ab3
--- /dev/null
+++ b/sound/soc/codecs/sgtl5000.h
@@ -0,0 +1,400 @@
+/*
+ * sgtl5000.h - SGTL5000 audio codec interface
+ *
+ * Copyright 2010-2011 Freescale Semiconductor, Inc.
+ *
+ * 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 _SGTL5000_H
+#define _SGTL5000_H
+
+/*
+ * Register values.
+ */
+#define SGTL5000_CHIP_ID 0x0000
+#define SGTL5000_CHIP_DIG_POWER 0x0002
+#define SGTL5000_CHIP_CLK_CTRL 0x0004
+#define SGTL5000_CHIP_I2S_CTRL 0x0006
+#define SGTL5000_CHIP_SSS_CTRL 0x000a
+#define SGTL5000_CHIP_ADCDAC_CTRL 0x000e
+#define SGTL5000_CHIP_DAC_VOL 0x0010
+#define SGTL5000_CHIP_PAD_STRENGTH 0x0014
+#define SGTL5000_CHIP_ANA_ADC_CTRL 0x0020
+#define SGTL5000_CHIP_ANA_HP_CTRL 0x0022
+#define SGTL5000_CHIP_ANA_CTRL 0x0024
+#define SGTL5000_CHIP_LINREG_CTRL 0x0026
+#define SGTL5000_CHIP_REF_CTRL 0x0028
+#define SGTL5000_CHIP_MIC_CTRL 0x002a
+#define SGTL5000_CHIP_LINE_OUT_CTRL 0x002c
+#define SGTL5000_CHIP_LINE_OUT_VOL 0x002e
+#define SGTL5000_CHIP_ANA_POWER 0x0030
+#define SGTL5000_CHIP_PLL_CTRL 0x0032
+#define SGTL5000_CHIP_CLK_TOP_CTRL 0x0034
+#define SGTL5000_CHIP_ANA_STATUS 0x0036
+#define SGTL5000_CHIP_SHORT_CTRL 0x003c
+#define SGTL5000_CHIP_ANA_TEST2 0x003a
+#define SGTL5000_DAP_CTRL 0x0100
+#define SGTL5000_DAP_PEQ 0x0102
+#define SGTL5000_DAP_BASS_ENHANCE 0x0104
+#define SGTL5000_DAP_BASS_ENHANCE_CTRL 0x0106
+#define SGTL5000_DAP_AUDIO_EQ 0x0108
+#define SGTL5000_DAP_SURROUND 0x010a
+#define SGTL5000_DAP_FLT_COEF_ACCESS 0x010c
+#define SGTL5000_DAP_COEF_WR_B0_MSB 0x010e
+#define SGTL5000_DAP_COEF_WR_B0_LSB 0x0110
+#define SGTL5000_DAP_EQ_BASS_BAND0 0x0116
+#define SGTL5000_DAP_EQ_BASS_BAND1 0x0118
+#define SGTL5000_DAP_EQ_BASS_BAND2 0x011a
+#define SGTL5000_DAP_EQ_BASS_BAND3 0x011c
+#define SGTL5000_DAP_EQ_BASS_BAND4 0x011e
+#define SGTL5000_DAP_MAIN_CHAN 0x0120
+#define SGTL5000_DAP_MIX_CHAN 0x0122
+#define SGTL5000_DAP_AVC_CTRL 0x0124
+#define SGTL5000_DAP_AVC_THRESHOLD 0x0126
+#define SGTL5000_DAP_AVC_ATTACK 0x0128
+#define SGTL5000_DAP_AVC_DECAY 0x012a
+#define SGTL5000_DAP_COEF_WR_B1_MSB 0x012c
+#define SGTL5000_DAP_COEF_WR_B1_LSB 0x012e
+#define SGTL5000_DAP_COEF_WR_B2_MSB 0x0130
+#define SGTL5000_DAP_COEF_WR_B2_LSB 0x0132
+#define SGTL5000_DAP_COEF_WR_A1_MSB 0x0134
+#define SGTL5000_DAP_COEF_WR_A1_LSB 0x0136
+#define SGTL5000_DAP_COEF_WR_A2_MSB 0x0138
+#define SGTL5000_DAP_COEF_WR_A2_LSB 0x013a
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * SGTL5000_CHIP_ID
+ */
+#define SGTL5000_PARTID_MASK 0xff00
+#define SGTL5000_PARTID_SHIFT 8
+#define SGTL5000_PARTID_WIDTH 8
+#define SGTL5000_PARTID_PART_ID 0xa0
+#define SGTL5000_REVID_MASK 0x00ff
+#define SGTL5000_REVID_SHIFT 0
+#define SGTL5000_REVID_WIDTH 8
+
+/*
+ * SGTL5000_CHIP_DIG_POWER
+ */
+#define SGTL5000_ADC_EN 0x0040
+#define SGTL5000_DAC_EN 0x0020
+#define SGTL5000_DAP_POWERUP 0x0010
+#define SGTL5000_I2S_OUT_POWERUP 0x0002
+#define SGTL5000_I2S_IN_POWERUP 0x0001
+
+/*
+ * SGTL5000_CHIP_CLK_CTRL
+ */
+#define SGTL5000_RATE_MODE_MASK 0x0030
+#define SGTL5000_RATE_MODE_SHIFT 4
+#define SGTL5000_RATE_MODE_WIDTH 2
+#define SGTL5000_RATE_MODE_DIV_1 0
+#define SGTL5000_RATE_MODE_DIV_2 1
+#define SGTL5000_RATE_MODE_DIV_4 2
+#define SGTL5000_RATE_MODE_DIV_6 3
+#define SGTL5000_SYS_FS_MASK 0x000c
+#define SGTL5000_SYS_FS_SHIFT 2
+#define SGTL5000_SYS_FS_WIDTH 2
+#define SGTL5000_SYS_FS_32k 0x0
+#define SGTL5000_SYS_FS_44_1k 0x1
+#define SGTL5000_SYS_FS_48k 0x2
+#define SGTL5000_SYS_FS_96k 0x3
+#define SGTL5000_MCLK_FREQ_MASK 0x0003
+#define SGTL5000_MCLK_FREQ_SHIFT 0
+#define SGTL5000_MCLK_FREQ_WIDTH 2
+#define SGTL5000_MCLK_FREQ_256FS 0x0
+#define SGTL5000_MCLK_FREQ_384FS 0x1
+#define SGTL5000_MCLK_FREQ_512FS 0x2
+#define SGTL5000_MCLK_FREQ_PLL 0x3
+
+/*
+ * SGTL5000_CHIP_I2S_CTRL
+ */
+#define SGTL5000_I2S_SCLKFREQ_MASK 0x0100
+#define SGTL5000_I2S_SCLKFREQ_SHIFT 8
+#define SGTL5000_I2S_SCLKFREQ_WIDTH 1
+#define SGTL5000_I2S_SCLKFREQ_64FS 0x0
+#define SGTL5000_I2S_SCLKFREQ_32FS 0x1 /* Not for RJ mode */
+#define SGTL5000_I2S_MASTER 0x0080
+#define SGTL5000_I2S_SCLK_INV 0x0040
+#define SGTL5000_I2S_DLEN_MASK 0x0030
+#define SGTL5000_I2S_DLEN_SHIFT 4
+#define SGTL5000_I2S_DLEN_WIDTH 2
+#define SGTL5000_I2S_DLEN_32 0x0
+#define SGTL5000_I2S_DLEN_24 0x1
+#define SGTL5000_I2S_DLEN_20 0x2
+#define SGTL5000_I2S_DLEN_16 0x3
+#define SGTL5000_I2S_MODE_MASK 0x000c
+#define SGTL5000_I2S_MODE_SHIFT 2
+#define SGTL5000_I2S_MODE_WIDTH 2
+#define SGTL5000_I2S_MODE_I2S_LJ 0x0
+#define SGTL5000_I2S_MODE_RJ 0x1
+#define SGTL5000_I2S_MODE_PCM 0x2
+#define SGTL5000_I2S_LRALIGN 0x0002
+#define SGTL5000_I2S_LRPOL 0x0001 /* set for which mode */
+
+/*
+ * SGTL5000_CHIP_SSS_CTRL
+ */
+#define SGTL5000_DAP_MIX_LRSWAP 0x4000
+#define SGTL5000_DAP_LRSWAP 0x2000
+#define SGTL5000_DAC_LRSWAP 0x1000
+#define SGTL5000_I2S_OUT_LRSWAP 0x0400
+#define SGTL5000_DAP_MIX_SEL_MASK 0x0300
+#define SGTL5000_DAP_MIX_SEL_SHIFT 8
+#define SGTL5000_DAP_MIX_SEL_WIDTH 2
+#define SGTL5000_DAP_MIX_SEL_ADC 0x0
+#define SGTL5000_DAP_MIX_SEL_I2S_IN 0x1
+#define SGTL5000_DAP_SEL_MASK 0x00c0
+#define SGTL5000_DAP_SEL_SHIFT 6
+#define SGTL5000_DAP_SEL_WIDTH 2
+#define SGTL5000_DAP_SEL_ADC 0x0
+#define SGTL5000_DAP_SEL_I2S_IN 0x1
+#define SGTL5000_DAC_SEL_MASK 0x0030
+#define SGTL5000_DAC_SEL_SHIFT 4
+#define SGTL5000_DAC_SEL_WIDTH 2
+#define SGTL5000_DAC_SEL_ADC 0x0
+#define SGTL5000_DAC_SEL_I2S_IN 0x1
+#define SGTL5000_DAC_SEL_DAP 0x3
+#define SGTL5000_I2S_OUT_SEL_MASK 0x0003
+#define SGTL5000_I2S_OUT_SEL_SHIFT 0
+#define SGTL5000_I2S_OUT_SEL_WIDTH 2
+#define SGTL5000_I2S_OUT_SEL_ADC 0x0
+#define SGTL5000_I2S_OUT_SEL_I2S_IN 0x1
+#define SGTL5000_I2S_OUT_SEL_DAP 0x3
+
+/*
+ * SGTL5000_CHIP_ADCDAC_CTRL
+ */
+#define SGTL5000_VOL_BUSY_DAC_RIGHT 0x2000
+#define SGTL5000_VOL_BUSY_DAC_LEFT 0x1000
+#define SGTL5000_DAC_VOL_RAMP_EN 0x0200
+#define SGTL5000_DAC_VOL_RAMP_EXPO 0x0100
+#define SGTL5000_DAC_MUTE_RIGHT 0x0008
+#define SGTL5000_DAC_MUTE_LEFT 0x0004
+#define SGTL5000_ADC_HPF_FREEZE 0x0002
+#define SGTL5000_ADC_HPF_BYPASS 0x0001
+
+/*
+ * SGTL5000_CHIP_DAC_VOL
+ */
+#define SGTL5000_DAC_VOL_RIGHT_MASK 0xff00
+#define SGTL5000_DAC_VOL_RIGHT_SHIFT 8
+#define SGTL5000_DAC_VOL_RIGHT_WIDTH 8
+#define SGTL5000_DAC_VOL_LEFT_MASK 0x00ff
+#define SGTL5000_DAC_VOL_LEFT_SHIFT 0
+#define SGTL5000_DAC_VOL_LEFT_WIDTH 8
+
+/*
+ * SGTL5000_CHIP_PAD_STRENGTH
+ */
+#define SGTL5000_PAD_I2S_LRCLK_MASK 0x0300
+#define SGTL5000_PAD_I2S_LRCLK_SHIFT 8
+#define SGTL5000_PAD_I2S_LRCLK_WIDTH 2
+#define SGTL5000_PAD_I2S_SCLK_MASK 0x00c0
+#define SGTL5000_PAD_I2S_SCLK_SHIFT 6
+#define SGTL5000_PAD_I2S_SCLK_WIDTH 2
+#define SGTL5000_PAD_I2S_DOUT_MASK 0x0030
+#define SGTL5000_PAD_I2S_DOUT_SHIFT 4
+#define SGTL5000_PAD_I2S_DOUT_WIDTH 2
+#define SGTL5000_PAD_I2C_SDA_MASK 0x000c
+#define SGTL5000_PAD_I2C_SDA_SHIFT 2
+#define SGTL5000_PAD_I2C_SDA_WIDTH 2
+#define SGTL5000_PAD_I2C_SCL_MASK 0x0003
+#define SGTL5000_PAD_I2C_SCL_SHIFT 0
+#define SGTL5000_PAD_I2C_SCL_WIDTH 2
+
+/*
+ * SGTL5000_CHIP_ANA_ADC_CTRL
+ */
+#define SGTL5000_ADC_VOL_M6DB 0x0100
+#define SGTL5000_ADC_VOL_RIGHT_MASK 0x00f0
+#define SGTL5000_ADC_VOL_RIGHT_SHIFT 4
+#define SGTL5000_ADC_VOL_RIGHT_WIDTH 4
+#define SGTL5000_ADC_VOL_LEFT_MASK 0x000f
+#define SGTL5000_ADC_VOL_LEFT_SHIFT 0
+#define SGTL5000_ADC_VOL_LEFT_WIDTH 4
+
+/*
+ * SGTL5000_CHIP_ANA_HP_CTRL
+ */
+#define SGTL5000_HP_VOL_RIGHT_MASK 0x7f00
+#define SGTL5000_HP_VOL_RIGHT_SHIFT 8
+#define SGTL5000_HP_VOL_RIGHT_WIDTH 7
+#define SGTL5000_HP_VOL_LEFT_MASK 0x007f
+#define SGTL5000_HP_VOL_LEFT_SHIFT 0
+#define SGTL5000_HP_VOL_LEFT_WIDTH 7
+
+/*
+ * SGTL5000_CHIP_ANA_CTRL
+ */
+#define SGTL5000_LINE_OUT_MUTE 0x0100
+#define SGTL5000_HP_SEL_MASK 0x0040
+#define SGTL5000_HP_SEL_SHIFT 6
+#define SGTL5000_HP_SEL_WIDTH 1
+#define SGTL5000_HP_SEL_DAC 0x0
+#define SGTL5000_HP_SEL_LINE_IN 0x1
+#define SGTL5000_HP_ZCD_EN 0x0020
+#define SGTL5000_HP_MUTE 0x0010
+#define SGTL5000_ADC_SEL_MASK 0x0004
+#define SGTL5000_ADC_SEL_SHIFT 2
+#define SGTL5000_ADC_SEL_WIDTH 1
+#define SGTL5000_ADC_SEL_MIC 0x0
+#define SGTL5000_ADC_SEL_LINE_IN 0x1
+#define SGTL5000_ADC_ZCD_EN 0x0002
+#define SGTL5000_ADC_MUTE 0x0001
+
+/*
+ * SGTL5000_CHIP_LINREG_CTRL
+ */
+#define SGTL5000_VDDC_MAN_ASSN_MASK 0x0040
+#define SGTL5000_VDDC_MAN_ASSN_SHIFT 6
+#define SGTL5000_VDDC_MAN_ASSN_WIDTH 1
+#define SGTL5000_VDDC_MAN_ASSN_VDDA 0x0
+#define SGTL5000_VDDC_MAN_ASSN_VDDIO 0x1
+#define SGTL5000_VDDC_ASSN_OVRD 0x0020
+#define SGTL5000_LINREG_VDDD_MASK 0x000f
+#define SGTL5000_LINREG_VDDD_SHIFT 0
+#define SGTL5000_LINREG_VDDD_WIDTH 4
+
+/*
+ * SGTL5000_CHIP_REF_CTRL
+ */
+#define SGTL5000_ANA_GND_MASK 0x01f0
+#define SGTL5000_ANA_GND_SHIFT 4
+#define SGTL5000_ANA_GND_WIDTH 5
+#define SGTL5000_ANA_GND_BASE 800 /* mv */
+#define SGTL5000_ANA_GND_STP 25 /*mv */
+#define SGTL5000_BIAS_CTRL_MASK 0x000e
+#define SGTL5000_BIAS_CTRL_SHIFT 1
+#define SGTL5000_BIAS_CTRL_WIDTH 3
+#define SGTL5000_SMALL_POP 0x0001
+
+/*
+ * SGTL5000_CHIP_MIC_CTRL
+ */
+#define SGTL5000_BIAS_R_MASK 0x0200
+#define SGTL5000_BIAS_R_SHIFT 8
+#define SGTL5000_BIAS_R_WIDTH 2
+#define SGTL5000_BIAS_R_off 0x0
+#define SGTL5000_BIAS_R_2K 0x1
+#define SGTL5000_BIAS_R_4k 0x2
+#define SGTL5000_BIAS_R_8k 0x3
+#define SGTL5000_BIAS_VOLT_MASK 0x0070
+#define SGTL5000_BIAS_VOLT_SHIFT 4
+#define SGTL5000_BIAS_VOLT_WIDTH 3
+#define SGTL5000_MIC_GAIN_MASK 0x0003
+#define SGTL5000_MIC_GAIN_SHIFT 0
+#define SGTL5000_MIC_GAIN_WIDTH 2
+
+/*
+ * SGTL5000_CHIP_LINE_OUT_CTRL
+ */
+#define SGTL5000_LINE_OUT_CURRENT_MASK 0x0f00
+#define SGTL5000_LINE_OUT_CURRENT_SHIFT 8
+#define SGTL5000_LINE_OUT_CURRENT_WIDTH 4
+#define SGTL5000_LINE_OUT_CURRENT_180u 0x0
+#define SGTL5000_LINE_OUT_CURRENT_270u 0x1
+#define SGTL5000_LINE_OUT_CURRENT_360u 0x3
+#define SGTL5000_LINE_OUT_CURRENT_450u 0x7
+#define SGTL5000_LINE_OUT_CURRENT_540u 0xf
+#define SGTL5000_LINE_OUT_GND_MASK 0x003f
+#define SGTL5000_LINE_OUT_GND_SHIFT 0
+#define SGTL5000_LINE_OUT_GND_WIDTH 6
+#define SGTL5000_LINE_OUT_GND_BASE 800 /* mv */
+#define SGTL5000_LINE_OUT_GND_STP 25
+#define SGTL5000_LINE_OUT_GND_MAX 0x23
+
+/*
+ * SGTL5000_CHIP_LINE_OUT_VOL
+ */
+#define SGTL5000_LINE_OUT_VOL_RIGHT_MASK 0x1f00
+#define SGTL5000_LINE_OUT_VOL_RIGHT_SHIFT 8
+#define SGTL5000_LINE_OUT_VOL_RIGHT_WIDTH 5
+#define SGTL5000_LINE_OUT_VOL_LEFT_MASK 0x001f
+#define SGTL5000_LINE_OUT_VOL_LEFT_SHIFT 0
+#define SGTL5000_LINE_OUT_VOL_LEFT_WIDTH 5
+
+/*
+ * SGTL5000_CHIP_ANA_POWER
+ */
+#define SGTL5000_DAC_STEREO 0x4000
+#define SGTL5000_LINREG_SIMPLE_POWERUP 0x2000
+#define SGTL5000_STARTUP_POWERUP 0x1000
+#define SGTL5000_VDDC_CHRGPMP_POWERUP 0x0800
+#define SGTL5000_PLL_POWERUP 0x0400
+#define SGTL5000_LINEREG_D_POWERUP 0x0200
+#define SGTL5000_VCOAMP_POWERUP 0x0100
+#define SGTL5000_VAG_POWERUP 0x0080
+#define SGTL5000_ADC_STEREO 0x0040
+#define SGTL5000_REFTOP_POWERUP 0x0020
+#define SGTL5000_HP_POWERUP 0x0010
+#define SGTL5000_DAC_POWERUP 0x0008
+#define SGTL5000_CAPLESS_HP_POWERUP 0x0004
+#define SGTL5000_ADC_POWERUP 0x0002
+#define SGTL5000_LINE_OUT_POWERUP 0x0001
+
+/*
+ * SGTL5000_CHIP_PLL_CTRL
+ */
+#define SGTL5000_PLL_INT_DIV_MASK 0xf800
+#define SGTL5000_PLL_INT_DIV_SHIFT 11
+#define SGTL5000_PLL_INT_DIV_WIDTH 5
+#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700
+#define SGTL5000_PLL_FRAC_DIV_SHIFT 0
+#define SGTL5000_PLL_FRAC_DIV_WIDTH 11
+
+/*
+ * SGTL5000_CHIP_CLK_TOP_CTRL
+ */
+#define SGTL5000_INT_OSC_EN 0x0800
+#define SGTL5000_INPUT_FREQ_DIV2 0x0008
+
+/*
+ * SGTL5000_CHIP_ANA_STATUS
+ */
+#define SGTL5000_HP_LRSHORT 0x0200
+#define SGTL5000_CAPLESS_SHORT 0x0100
+#define SGTL5000_PLL_LOCKED 0x0010
+
+/*
+ * SGTL5000_CHIP_SHORT_CTRL
+ */
+#define SGTL5000_LVLADJR_MASK 0x7000
+#define SGTL5000_LVLADJR_SHIFT 12
+#define SGTL5000_LVLADJR_WIDTH 3
+#define SGTL5000_LVLADJL_MASK 0x0700
+#define SGTL5000_LVLADJL_SHIFT 8
+#define SGTL5000_LVLADJL_WIDTH 3
+#define SGTL5000_LVLADJC_MASK 0x0070
+#define SGTL5000_LVLADJC_SHIFT 4
+#define SGTL5000_LVLADJC_WIDTH 3
+#define SGTL5000_LR_SHORT_MOD_MASK 0x000c
+#define SGTL5000_LR_SHORT_MOD_SHIFT 2
+#define SGTL5000_LR_SHORT_MOD_WIDTH 2
+#define SGTL5000_CM_SHORT_MOD_MASK 0x0003
+#define SGTL5000_CM_SHORT_MOD_SHIFT 0
+#define SGTL5000_CM_SHORT_MOD_WIDTH 2
+
+/*
+ *SGTL5000_CHIP_ANA_TEST2
+ */
+#define SGTL5000_MONO_DAC 0x1000
+
+/*
+ * SGTL5000_DAP_CTRL
+ */
+#define SGTL5000_DAP_MIX_EN 0x0010
+#define SGTL5000_DAP_EN 0x0001
+
+#define SGTL5000_SYSCLK 0x00
+#define SGTL5000_LRCLK 0x01
+
+#endif
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 40e285d..2a30eae 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -34,17 +34,135 @@
#include <sound/soc-dapm.h>
#include <sound/initval.h>
#include <sound/tlv.h>
+#include <sound/jack.h>
#include "sn95031.h"
#define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100)
#define SN95031_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
+/* adc helper functions */
+
+/* enables mic bias voltage */
+static void sn95031_enable_mic_bias(struct snd_soc_codec *codec)
+{
+ snd_soc_write(codec, SN95031_VAUD, BIT(2)|BIT(1)|BIT(0));
+ snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(2), BIT(2));
+}
+
+/* Enable/Disable the ADC depending on the argument */
+static void configure_adc(struct snd_soc_codec *sn95031_codec, int val)
+{
+ int value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1);
+
+ if (val) {
+ /* Enable and start the ADC */
+ value |= (SN95031_ADC_ENBL | SN95031_ADC_START);
+ value &= (~SN95031_ADC_NO_LOOP);
+ } else {
+ /* Just stop the ADC */
+ value &= (~SN95031_ADC_START);
+ }
+ snd_soc_write(sn95031_codec, SN95031_ADC1CNTL1, value);
+}
+
/*
- * todo:
- * capture paths
- * jack detection
- * PM functions
+ * finds an empty channel for conversion
+ * If the ADC is not enabled then start using 0th channel
+ * itself. Otherwise find an empty channel by looking for a
+ * channel in which the stopbit is set to 1. returns the index
+ * of the first free channel if succeeds or an error code.
+ *
+ * Context: can sleep
+ *
*/
+static int find_free_channel(struct snd_soc_codec *sn95031_codec)
+{
+ int ret = 0, i, value;
+
+ /* check whether ADC is enabled */
+ value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1);
+
+ if ((value & SN95031_ADC_ENBL) == 0)
+ return 0;
+
+ /* ADC is already enabled; Looking for an empty channel */
+ for (i = 0; i < SN95031_ADC_CHANLS_MAX; i++) {
+ value = snd_soc_read(sn95031_codec,
+ SN95031_ADC_CHNL_START_ADDR + i);
+ if (value & SN95031_STOPBIT_MASK) {
+ ret = i;
+ break;
+ }
+ }
+ return (ret > SN95031_ADC_LOOP_MAX) ? (-EINVAL) : ret;
+}
+
+/* Initialize the ADC for reading micbias values. Can sleep. */
+static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec)
+{
+ int base_addr, chnl_addr;
+ int value;
+ static int channel_index;
+
+ /* Index of the first channel in which the stop bit is set */
+ channel_index = find_free_channel(sn95031_codec);
+ if (channel_index < 0) {
+ pr_err("No free ADC channels");
+ return channel_index;
+ }
+
+ base_addr = SN95031_ADC_CHNL_START_ADDR + channel_index;
+
+ if (!(channel_index == 0 || channel_index == SN95031_ADC_LOOP_MAX)) {
+ /* Reset stop bit for channels other than 0 and 12 */
+ value = snd_soc_read(sn95031_codec, base_addr);
+ /* Set the stop bit to zero */
+ snd_soc_write(sn95031_codec, base_addr, value & 0xEF);
+ /* Index of the first free channel */
+ base_addr++;
+ channel_index++;
+ }
+
+ /* Since this is the last channel, set the stop bit
+ to 1 by ORing the DIE_SENSOR_CODE with 0x10 */
+ snd_soc_write(sn95031_codec, base_addr,
+ SN95031_AUDIO_DETECT_CODE | 0x10);
+
+ chnl_addr = SN95031_ADC_DATA_START_ADDR + 2 * channel_index;
+ pr_debug("mid_initialize : %x", chnl_addr);
+ configure_adc(sn95031_codec, 1);
+ return chnl_addr;
+}
+
+
+/* reads the ADC registers and gets the mic bias value in mV. */
+static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec)
+{
+ u16 adc_adr = sn95031_initialize_adc(codec);
+ u16 adc_val1, adc_val2;
+ unsigned int mic_bias;
+
+ sn95031_enable_mic_bias(codec);
+
+ /* Enable the sound card for conversion before reading */
+ snd_soc_write(codec, SN95031_ADC1CNTL3, 0x05);
+ /* Re-toggle the RRDATARD bit */
+ snd_soc_write(codec, SN95031_ADC1CNTL3, 0x04);
+
+ /* Read the higher bits of data */
+ msleep(1000);
+ adc_val1 = snd_soc_read(codec, adc_adr);
+ adc_adr++;
+ adc_val2 = snd_soc_read(codec, adc_adr);
+
+ /* Adding lower two bits to the higher bits */
+ mic_bias = (adc_val1 << 2) + (adc_val2 & 3);
+ mic_bias = (mic_bias * SN95031_ADC_ONE_LSB_MULTIPLIER) / 1000;
+ pr_debug("mic bias = %dmV\n", mic_bias);
+ return mic_bias;
+}
+EXPORT_SYMBOL_GPL(sn95031_get_mic_bias);
+/*end - adc helper functions */
static inline unsigned int sn95031_read(struct snd_soc_codec *codec,
unsigned int reg)
@@ -241,7 +359,7 @@
static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"};
/* 0dB to 30dB in 10dB steps */
-static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 30);
+static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 0);
static const struct soc_enum sn95031_micmode1_enum =
SOC_ENUM_SINGLE(SN95031_MICAMP1, 1, 2, sn95031_micmode_text);
@@ -401,6 +519,8 @@
static const struct snd_soc_dapm_route sn95031_audio_map[] = {
/* headset and earpiece map */
+ { "HPOUTL", NULL, "Headset Rail"},
+ { "HPOUTR", NULL, "Headset Rail"},
{ "HPOUTL", NULL, "Headset Left Playback" },
{ "HPOUTR", NULL, "Headset Right Playback" },
{ "EPOUT", NULL, "Earpiece Playback" },
@@ -409,18 +529,16 @@
{ "Earpiece Playback", NULL, "Headset Left Filter"},
{ "Headset Left Filter", NULL, "HSDAC Left"},
{ "Headset Right Filter", NULL, "HSDAC Right"},
- { "HSDAC Left", NULL, "Headset Rail"},
- { "HSDAC Right", NULL, "Headset Rail"},
/* speaker map */
+ { "IHFOUTL", NULL, "Speaker Rail"},
+ { "IHFOUTR", NULL, "Speaker Rail"},
{ "IHFOUTL", "NULL", "Speaker Left Playback"},
{ "IHFOUTR", "NULL", "Speaker Right Playback"},
{ "Speaker Left Playback", NULL, "Speaker Left Filter"},
{ "Speaker Right Playback", NULL, "Speaker Right Filter"},
{ "Speaker Left Filter", NULL, "IHFDAC Left"},
{ "Speaker Right Filter", NULL, "IHFDAC Right"},
- { "IHFDAC Left", NULL, "Speaker Rail"},
- { "IHFDAC Right", NULL, "Speaker Rail"},
/* vibra map */
{ "VIB1OUT", NULL, "Vibra1 Playback"},
@@ -484,30 +602,30 @@
{ "Txpath2 Capture Route", "ADC Right", "ADC Right"},
{ "Txpath3 Capture Route", "ADC Right", "ADC Right"},
{ "Txpath4 Capture Route", "ADC Right", "ADC Right"},
- { "Txpath1 Capture Route", NULL, "DMIC1"},
- { "Txpath2 Capture Route", NULL, "DMIC1"},
- { "Txpath3 Capture Route", NULL, "DMIC1"},
- { "Txpath4 Capture Route", NULL, "DMIC1"},
- { "Txpath1 Capture Route", NULL, "DMIC2"},
- { "Txpath2 Capture Route", NULL, "DMIC2"},
- { "Txpath3 Capture Route", NULL, "DMIC2"},
- { "Txpath4 Capture Route", NULL, "DMIC2"},
- { "Txpath1 Capture Route", NULL, "DMIC3"},
- { "Txpath2 Capture Route", NULL, "DMIC3"},
- { "Txpath3 Capture Route", NULL, "DMIC3"},
- { "Txpath4 Capture Route", NULL, "DMIC3"},
- { "Txpath1 Capture Route", NULL, "DMIC4"},
- { "Txpath2 Capture Route", NULL, "DMIC4"},
- { "Txpath3 Capture Route", NULL, "DMIC4"},
- { "Txpath4 Capture Route", NULL, "DMIC4"},
- { "Txpath1 Capture Route", NULL, "DMIC5"},
- { "Txpath2 Capture Route", NULL, "DMIC5"},
- { "Txpath3 Capture Route", NULL, "DMIC5"},
- { "Txpath4 Capture Route", NULL, "DMIC5"},
- { "Txpath1 Capture Route", NULL, "DMIC6"},
- { "Txpath2 Capture Route", NULL, "DMIC6"},
- { "Txpath3 Capture Route", NULL, "DMIC6"},
- { "Txpath4 Capture Route", NULL, "DMIC6"},
+ { "Txpath1 Capture Route", "DMIC1", "DMIC1"},
+ { "Txpath2 Capture Route", "DMIC1", "DMIC1"},
+ { "Txpath3 Capture Route", "DMIC1", "DMIC1"},
+ { "Txpath4 Capture Route", "DMIC1", "DMIC1"},
+ { "Txpath1 Capture Route", "DMIC2", "DMIC2"},
+ { "Txpath2 Capture Route", "DMIC2", "DMIC2"},
+ { "Txpath3 Capture Route", "DMIC2", "DMIC2"},
+ { "Txpath4 Capture Route", "DMIC2", "DMIC2"},
+ { "Txpath1 Capture Route", "DMIC3", "DMIC3"},
+ { "Txpath2 Capture Route", "DMIC3", "DMIC3"},
+ { "Txpath3 Capture Route", "DMIC3", "DMIC3"},
+ { "Txpath4 Capture Route", "DMIC3", "DMIC3"},
+ { "Txpath1 Capture Route", "DMIC4", "DMIC4"},
+ { "Txpath2 Capture Route", "DMIC4", "DMIC4"},
+ { "Txpath3 Capture Route", "DMIC4", "DMIC4"},
+ { "Txpath4 Capture Route", "DMIC4", "DMIC4"},
+ { "Txpath1 Capture Route", "DMIC5", "DMIC5"},
+ { "Txpath2 Capture Route", "DMIC5", "DMIC5"},
+ { "Txpath3 Capture Route", "DMIC5", "DMIC5"},
+ { "Txpath4 Capture Route", "DMIC5", "DMIC5"},
+ { "Txpath1 Capture Route", "DMIC6", "DMIC6"},
+ { "Txpath2 Capture Route", "DMIC6", "DMIC6"},
+ { "Txpath3 Capture Route", "DMIC6", "DMIC6"},
+ { "Txpath4 Capture Route", "DMIC6", "DMIC6"},
/* tx path */
{ "TX1 Enable", NULL, "Txpath1 Capture Route"},
@@ -649,6 +767,61 @@
},
};
+static inline void sn95031_disable_jack_btn(struct snd_soc_codec *codec)
+{
+ snd_soc_write(codec, SN95031_BTNCTRL2, 0x00);
+}
+
+static inline void sn95031_enable_jack_btn(struct snd_soc_codec *codec)
+{
+ snd_soc_write(codec, SN95031_BTNCTRL1, 0x77);
+ snd_soc_write(codec, SN95031_BTNCTRL2, 0x01);
+}
+
+static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack)
+{
+ int micbias = sn95031_get_mic_bias(mfld_jack->codec);
+
+ int jack_type = snd_soc_jack_get_type(mfld_jack, micbias);
+
+ pr_debug("jack type detected = %d\n", jack_type);
+ if (jack_type == SND_JACK_HEADSET)
+ sn95031_enable_jack_btn(mfld_jack->codec);
+ return jack_type;
+}
+
+void sn95031_jack_detection(struct mfld_jack_data *jack_data)
+{
+ unsigned int status;
+ unsigned int mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_HEADSET;
+
+ pr_debug("interrupt id read in sram = 0x%x\n", jack_data->intr_id);
+ if (jack_data->intr_id & 0x1) {
+ pr_debug("short_push detected\n");
+ status = SND_JACK_HEADSET | SND_JACK_BTN_0;
+ } else if (jack_data->intr_id & 0x2) {
+ pr_debug("long_push detected\n");
+ status = SND_JACK_HEADSET | SND_JACK_BTN_1;
+ } else if (jack_data->intr_id & 0x4) {
+ pr_debug("headset or headphones inserted\n");
+ status = sn95031_get_headset_state(jack_data->mfld_jack);
+ } else if (jack_data->intr_id & 0x8) {
+ pr_debug("headset or headphones removed\n");
+ status = 0;
+ sn95031_disable_jack_btn(jack_data->mfld_jack->codec);
+ } else {
+ pr_err("unidentified interrupt\n");
+ return;
+ }
+
+ snd_soc_jack_report(jack_data->mfld_jack, status, mask);
+ /*button pressed and released so we send explicit button release */
+ if ((status & SND_JACK_BTN_0) | (status & SND_JACK_BTN_1))
+ snd_soc_jack_report(jack_data->mfld_jack,
+ SND_JACK_HEADSET, mask);
+}
+EXPORT_SYMBOL_GPL(sn95031_jack_detection);
+
/* codec registration */
static int sn95031_codec_probe(struct snd_soc_codec *codec)
{
diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h
index e2b17d9..20376d2 100644
--- a/sound/soc/codecs/sn95031.h
+++ b/sound/soc/codecs/sn95031.h
@@ -96,4 +96,37 @@
#define SN95031_SSR5 0x384
#define SN95031_SSR6 0x385
+/* ADC registers */
+
+#define SN95031_ADC1CNTL1 0x1C0
+#define SN95031_ADC_ENBL 0x10
+#define SN95031_ADC_START 0x08
+#define SN95031_ADC1CNTL3 0x1C2
+#define SN95031_ADCTHERM_ENBL 0x04
+#define SN95031_ADCRRDATA_ENBL 0x05
+#define SN95031_STOPBIT_MASK 16
+#define SN95031_ADCTHERM_MASK 4
+#define SN95031_ADC_CHANLS_MAX 15 /* Number of ADC channels */
+#define SN95031_ADC_LOOP_MAX (SN95031_ADC_CHANLS_MAX - 1)
+#define SN95031_ADC_NO_LOOP 0x07
+#define SN95031_AUDIO_GPIO_CTRL 0x070
+
+/* ADC channel code values */
+#define SN95031_AUDIO_DETECT_CODE 0x06
+
+/* ADC base addresses */
+#define SN95031_ADC_CHNL_START_ADDR 0x1C5 /* increments by 1 */
+#define SN95031_ADC_DATA_START_ADDR 0x1D4 /* increments by 2 */
+/* multipier to convert to mV */
+#define SN95031_ADC_ONE_LSB_MULTIPLIER 2346
+
+
+struct mfld_jack_data {
+ int intr_id;
+ int micbias_vol;
+ struct snd_soc_jack *mfld_jack;
+};
+
+extern void sn95031_jack_detection(struct mfld_jack_data *jack_data);
+
#endif
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
new file mode 100644
index 0000000..e93b9d1
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -0,0 +1,794 @@
+/*
+ * linux/sound/soc/codecs/tlv320aic32x4.c
+ *
+ * Copyright 2011 Vista Silicon S.L.
+ *
+ * Author: Javier Martin <javier.martin@vista-silicon.com>
+ *
+ * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27.
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
+ * MA 02110-1301, USA.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/cdev.h>
+#include <linux/slab.h>
+
+#include <sound/tlv320aic32x4.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "tlv320aic32x4.h"
+
+struct aic32x4_rate_divs {
+ u32 mclk;
+ u32 rate;
+ u8 p_val;
+ u8 pll_j;
+ u16 pll_d;
+ u16 dosr;
+ u8 ndac;
+ u8 mdac;
+ u8 aosr;
+ u8 nadc;
+ u8 madc;
+ u8 blck_N;
+};
+
+struct aic32x4_priv {
+ u32 sysclk;
+ s32 master;
+ u8 page_no;
+ void *control_data;
+ u32 power_cfg;
+ u32 micpga_routing;
+ bool swapdacs;
+};
+
+/* 0dB min, 1dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_step_1, 0, 100, 0);
+/* 0dB min, 0.5dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_step_0_5, 0, 50, 0);
+
+static const struct snd_kcontrol_new aic32x4_snd_controls[] = {
+ SOC_DOUBLE_R_TLV("PCM Playback Volume", AIC32X4_LDACVOL,
+ AIC32X4_RDACVOL, 0, 0x30, 0, tlv_step_0_5),
+ SOC_DOUBLE_R_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN,
+ AIC32X4_HPRGAIN, 0, 0x1D, 0, tlv_step_1),
+ SOC_DOUBLE_R_TLV("LO Driver Gain Volume", AIC32X4_LOLGAIN,
+ AIC32X4_LORGAIN, 0, 0x1D, 0, tlv_step_1),
+ SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN,
+ AIC32X4_HPRGAIN, 6, 0x01, 1),
+ SOC_DOUBLE_R("LO DAC Playback Switch", AIC32X4_LOLGAIN,
+ AIC32X4_LORGAIN, 6, 0x01, 1),
+ SOC_DOUBLE_R("Mic PGA Switch", AIC32X4_LMICPGAVOL,
+ AIC32X4_RMICPGAVOL, 7, 0x01, 1),
+
+ SOC_SINGLE("ADCFGA Left Mute Switch", AIC32X4_ADCFGA, 7, 1, 0),
+ SOC_SINGLE("ADCFGA Right Mute Switch", AIC32X4_ADCFGA, 3, 1, 0),
+
+ SOC_DOUBLE_R_TLV("ADC Level Volume", AIC32X4_LADCVOL,
+ AIC32X4_RADCVOL, 0, 0x28, 0, tlv_step_0_5),
+ SOC_DOUBLE_R_TLV("PGA Level Volume", AIC32X4_LMICPGAVOL,
+ AIC32X4_RMICPGAVOL, 0, 0x5f, 0, tlv_step_0_5),
+
+ SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0),
+
+ SOC_SINGLE("AGC Left Switch", AIC32X4_LAGC1, 7, 1, 0),
+ SOC_SINGLE("AGC Right Switch", AIC32X4_RAGC1, 7, 1, 0),
+ SOC_DOUBLE_R("AGC Target Level", AIC32X4_LAGC1, AIC32X4_RAGC1,
+ 4, 0x07, 0),
+ SOC_DOUBLE_R("AGC Gain Hysteresis", AIC32X4_LAGC1, AIC32X4_RAGC1,
+ 0, 0x03, 0),
+ SOC_DOUBLE_R("AGC Hysteresis", AIC32X4_LAGC2, AIC32X4_RAGC2,
+ 6, 0x03, 0),
+ SOC_DOUBLE_R("AGC Noise Threshold", AIC32X4_LAGC2, AIC32X4_RAGC2,
+ 1, 0x1F, 0),
+ SOC_DOUBLE_R("AGC Max PGA", AIC32X4_LAGC3, AIC32X4_RAGC3,
+ 0, 0x7F, 0),
+ SOC_DOUBLE_R("AGC Attack Time", AIC32X4_LAGC4, AIC32X4_RAGC4,
+ 3, 0x1F, 0),
+ SOC_DOUBLE_R("AGC Decay Time", AIC32X4_LAGC5, AIC32X4_RAGC5,
+ 3, 0x1F, 0),
+ SOC_DOUBLE_R("AGC Noise Debounce", AIC32X4_LAGC6, AIC32X4_RAGC6,
+ 0, 0x1F, 0),
+ SOC_DOUBLE_R("AGC Signal Debounce", AIC32X4_LAGC7, AIC32X4_RAGC7,
+ 0, 0x0F, 0),
+};
+
+static const struct aic32x4_rate_divs aic32x4_divs[] = {
+ /* 8k rate */
+ {AIC32X4_FREQ_12000000, 8000, 1, 7, 6800, 768, 5, 3, 128, 5, 18, 24},
+ {AIC32X4_FREQ_24000000, 8000, 2, 7, 6800, 768, 15, 1, 64, 45, 4, 24},
+ {AIC32X4_FREQ_25000000, 8000, 2, 7, 3728, 768, 15, 1, 64, 45, 4, 24},
+ /* 11.025k rate */
+ {AIC32X4_FREQ_12000000, 11025, 1, 7, 5264, 512, 8, 2, 128, 8, 8, 16},
+ {AIC32X4_FREQ_24000000, 11025, 2, 7, 5264, 512, 16, 1, 64, 32, 4, 16},
+ /* 16k rate */
+ {AIC32X4_FREQ_12000000, 16000, 1, 7, 6800, 384, 5, 3, 128, 5, 9, 12},
+ {AIC32X4_FREQ_24000000, 16000, 2, 7, 6800, 384, 15, 1, 64, 18, 5, 12},
+ {AIC32X4_FREQ_25000000, 16000, 2, 7, 3728, 384, 15, 1, 64, 18, 5, 12},
+ /* 22.05k rate */
+ {AIC32X4_FREQ_12000000, 22050, 1, 7, 5264, 256, 4, 4, 128, 4, 8, 8},
+ {AIC32X4_FREQ_24000000, 22050, 2, 7, 5264, 256, 16, 1, 64, 16, 4, 8},
+ {AIC32X4_FREQ_25000000, 22050, 2, 7, 2253, 256, 16, 1, 64, 16, 4, 8},
+ /* 32k rate */
+ {AIC32X4_FREQ_12000000, 32000, 1, 7, 1680, 192, 2, 7, 64, 2, 21, 6},
+ {AIC32X4_FREQ_24000000, 32000, 2, 7, 1680, 192, 7, 2, 64, 7, 6, 6},
+ /* 44.1k rate */
+ {AIC32X4_FREQ_12000000, 44100, 1, 7, 5264, 128, 2, 8, 128, 2, 8, 4},
+ {AIC32X4_FREQ_24000000, 44100, 2, 7, 5264, 128, 8, 2, 64, 8, 4, 4},
+ {AIC32X4_FREQ_25000000, 44100, 2, 7, 2253, 128, 8, 2, 64, 8, 4, 4},
+ /* 48k rate */
+ {AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4},
+ {AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4},
+ {AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4}
+};
+
+static const struct snd_kcontrol_new hpl_output_mixer_controls[] = {
+ SOC_DAPM_SINGLE("L_DAC Switch", AIC32X4_HPLROUTE, 3, 1, 0),
+ SOC_DAPM_SINGLE("IN1_L Switch", AIC32X4_HPLROUTE, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new hpr_output_mixer_controls[] = {
+ SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_HPRROUTE, 3, 1, 0),
+ SOC_DAPM_SINGLE("IN1_R Switch", AIC32X4_HPRROUTE, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new lol_output_mixer_controls[] = {
+ SOC_DAPM_SINGLE("L_DAC Switch", AIC32X4_LOLROUTE, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new lor_output_mixer_controls[] = {
+ SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_LORROUTE, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new left_input_mixer_controls[] = {
+ SOC_DAPM_SINGLE("IN1_L P Switch", AIC32X4_LMICPGAPIN, 6, 1, 0),
+ SOC_DAPM_SINGLE("IN2_L P Switch", AIC32X4_LMICPGAPIN, 4, 1, 0),
+ SOC_DAPM_SINGLE("IN3_L P Switch", AIC32X4_LMICPGAPIN, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_input_mixer_controls[] = {
+ SOC_DAPM_SINGLE("IN1_R P Switch", AIC32X4_RMICPGAPIN, 6, 1, 0),
+ SOC_DAPM_SINGLE("IN2_R P Switch", AIC32X4_RMICPGAPIN, 4, 1, 0),
+ SOC_DAPM_SINGLE("IN3_R P Switch", AIC32X4_RMICPGAPIN, 2, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", AIC32X4_DACSETUP, 7, 0),
+ SND_SOC_DAPM_MIXER("HPL Output Mixer", SND_SOC_NOPM, 0, 0,
+ &hpl_output_mixer_controls[0],
+ ARRAY_SIZE(hpl_output_mixer_controls)),
+ SND_SOC_DAPM_PGA("HPL Power", AIC32X4_OUTPWRCTL, 5, 0, NULL, 0),
+
+ SND_SOC_DAPM_MIXER("LOL Output Mixer", SND_SOC_NOPM, 0, 0,
+ &lol_output_mixer_controls[0],
+ ARRAY_SIZE(lol_output_mixer_controls)),
+ SND_SOC_DAPM_PGA("LOL Power", AIC32X4_OUTPWRCTL, 3, 0, NULL, 0),
+
+ SND_SOC_DAPM_DAC("Right DAC", "Right Playback", AIC32X4_DACSETUP, 6, 0),
+ SND_SOC_DAPM_MIXER("HPR Output Mixer", SND_SOC_NOPM, 0, 0,
+ &hpr_output_mixer_controls[0],
+ ARRAY_SIZE(hpr_output_mixer_controls)),
+ SND_SOC_DAPM_PGA("HPR Power", AIC32X4_OUTPWRCTL, 4, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("LOR Output Mixer", SND_SOC_NOPM, 0, 0,
+ &lor_output_mixer_controls[0],
+ ARRAY_SIZE(lor_output_mixer_controls)),
+ SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0,
+ &left_input_mixer_controls[0],
+ ARRAY_SIZE(left_input_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0,
+ &right_input_mixer_controls[0],
+ ARRAY_SIZE(right_input_mixer_controls)),
+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0),
+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0),
+ SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0),
+
+ SND_SOC_DAPM_OUTPUT("HPL"),
+ SND_SOC_DAPM_OUTPUT("HPR"),
+ SND_SOC_DAPM_OUTPUT("LOL"),
+ SND_SOC_DAPM_OUTPUT("LOR"),
+ SND_SOC_DAPM_INPUT("IN1_L"),
+ SND_SOC_DAPM_INPUT("IN1_R"),
+ SND_SOC_DAPM_INPUT("IN2_L"),
+ SND_SOC_DAPM_INPUT("IN2_R"),
+ SND_SOC_DAPM_INPUT("IN3_L"),
+ SND_SOC_DAPM_INPUT("IN3_R"),
+};
+
+static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = {
+ /* Left Output */
+ {"HPL Output Mixer", "L_DAC Switch", "Left DAC"},
+ {"HPL Output Mixer", "IN1_L Switch", "IN1_L"},
+
+ {"HPL Power", NULL, "HPL Output Mixer"},
+ {"HPL", NULL, "HPL Power"},
+
+ {"LOL Output Mixer", "L_DAC Switch", "Left DAC"},
+
+ {"LOL Power", NULL, "LOL Output Mixer"},
+ {"LOL", NULL, "LOL Power"},
+
+ /* Right Output */
+ {"HPR Output Mixer", "R_DAC Switch", "Right DAC"},
+ {"HPR Output Mixer", "IN1_R Switch", "IN1_R"},
+
+ {"HPR Power", NULL, "HPR Output Mixer"},
+ {"HPR", NULL, "HPR Power"},
+
+ {"LOR Output Mixer", "R_DAC Switch", "Right DAC"},
+
+ {"LOR Power", NULL, "LOR Output Mixer"},
+ {"LOR", NULL, "LOR Power"},
+
+ /* Left input */
+ {"Left Input Mixer", "IN1_L P Switch", "IN1_L"},
+ {"Left Input Mixer", "IN2_L P Switch", "IN2_L"},
+ {"Left Input Mixer", "IN3_L P Switch", "IN3_L"},
+
+ {"Left ADC", NULL, "Left Input Mixer"},
+
+ /* Right Input */
+ {"Right Input Mixer", "IN1_R P Switch", "IN1_R"},
+ {"Right Input Mixer", "IN2_R P Switch", "IN2_R"},
+ {"Right Input Mixer", "IN3_R P Switch", "IN3_R"},
+
+ {"Right ADC", NULL, "Right Input Mixer"},
+};
+
+static inline int aic32x4_change_page(struct snd_soc_codec *codec,
+ unsigned int new_page)
+{
+ struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+ u8 data[2];
+ int ret;
+
+ data[0] = 0x00;
+ data[1] = new_page & 0xff;
+
+ ret = codec->hw_write(codec->control_data, data, 2);
+ if (ret == 2) {
+ aic32x4->page_no = new_page;
+ return 0;
+ } else {
+ return ret;
+ }
+}
+
+static int aic32x4_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int val)
+{
+ struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+ unsigned int page = reg / 128;
+ unsigned int fixed_reg = reg % 128;
+ u8 data[2];
+ int ret;
+
+ /* A write to AIC32X4_PSEL is really a non-explicit page change */
+ if (reg == AIC32X4_PSEL)
+ return aic32x4_change_page(codec, val);
+
+ if (aic32x4->page_no != page) {
+ ret = aic32x4_change_page(codec, page);
+ if (ret != 0)
+ return ret;
+ }
+
+ data[0] = fixed_reg & 0xff;
+ data[1] = val & 0xff;
+
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+static unsigned int aic32x4_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+ struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+ unsigned int page = reg / 128;
+ unsigned int fixed_reg = reg % 128;
+ int ret;
+
+ if (aic32x4->page_no != page) {
+ ret = aic32x4_change_page(codec, page);
+ if (ret != 0)
+ return ret;
+ }
+ return i2c_smbus_read_byte_data(codec->control_data, fixed_reg & 0xff);
+}
+
+static inline int aic32x4_get_divs(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(aic32x4_divs); i++) {
+ if ((aic32x4_divs[i].rate == rate)
+ && (aic32x4_divs[i].mclk == mclk)) {
+ return i;
+ }
+ }
+ printk(KERN_ERR "aic32x4: master clock and sample rate is not supported\n");
+ return -EINVAL;
+}
+
+static int aic32x4_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets,
+ ARRAY_SIZE(aic32x4_dapm_widgets));
+
+ snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes,
+ ARRAY_SIZE(aic32x4_dapm_routes));
+
+ snd_soc_dapm_new_widgets(&codec->dapm);
+ return 0;
+}
+
+static int aic32x4_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 aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+
+ switch (freq) {
+ case AIC32X4_FREQ_12000000:
+ case AIC32X4_FREQ_24000000:
+ case AIC32X4_FREQ_25000000:
+ aic32x4->sysclk = freq;
+ return 0;
+ }
+ printk(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n");
+ return -EINVAL;
+}
+
+static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+ u8 iface_reg_1;
+ u8 iface_reg_2;
+ u8 iface_reg_3;
+
+ iface_reg_1 = snd_soc_read(codec, AIC32X4_IFACE1);
+ iface_reg_1 = iface_reg_1 & ~(3 << 6 | 3 << 2);
+ iface_reg_2 = snd_soc_read(codec, AIC32X4_IFACE2);
+ iface_reg_2 = 0;
+ iface_reg_3 = snd_soc_read(codec, AIC32X4_IFACE3);
+ iface_reg_3 = iface_reg_3 & ~(1 << 3);
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ aic32x4->master = 1;
+ iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ aic32x4->master = 0;
+ break;
+ default:
+ printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n");
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT);
+ iface_reg_3 |= (1 << 3); /* invert bit clock */
+ iface_reg_2 = 0x01; /* add offset 1 */
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT);
+ iface_reg_3 |= (1 << 3); /* invert bit clock */
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ iface_reg_1 |=
+ (AIC32X4_RIGHT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT);
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface_reg_1 |=
+ (AIC32X4_LEFT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT);
+ break;
+ default:
+ printk(KERN_ERR "aic32x4: invalid DAI interface format\n");
+ return -EINVAL;
+ }
+
+ snd_soc_write(codec, AIC32X4_IFACE1, iface_reg_1);
+ snd_soc_write(codec, AIC32X4_IFACE2, iface_reg_2);
+ snd_soc_write(codec, AIC32X4_IFACE3, iface_reg_3);
+ return 0;
+}
+
+static int aic32x4_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+ u8 data;
+ int i;
+
+ i = aic32x4_get_divs(aic32x4->sysclk, params_rate(params));
+ if (i < 0) {
+ printk(KERN_ERR "aic32x4: sampling rate not supported\n");
+ return i;
+ }
+
+ /* Use PLL as CODEC_CLKIN and DAC_MOD_CLK as BDIV_CLKIN */
+ snd_soc_write(codec, AIC32X4_CLKMUX, AIC32X4_PLLCLKIN);
+ snd_soc_write(codec, AIC32X4_IFACE3, AIC32X4_DACMOD2BCLK);
+
+ /* We will fix R value to 1 and will make P & J=K.D as varialble */
+ data = snd_soc_read(codec, AIC32X4_PLLPR);
+ data &= ~(7 << 4);
+ snd_soc_write(codec, AIC32X4_PLLPR,
+ (data | (aic32x4_divs[i].p_val << 4) | 0x01));
+
+ snd_soc_write(codec, AIC32X4_PLLJ, aic32x4_divs[i].pll_j);
+
+ snd_soc_write(codec, AIC32X4_PLLDMSB, (aic32x4_divs[i].pll_d >> 8));
+ snd_soc_write(codec, AIC32X4_PLLDLSB,
+ (aic32x4_divs[i].pll_d & 0xff));
+
+ /* NDAC divider value */
+ data = snd_soc_read(codec, AIC32X4_NDAC);
+ data &= ~(0x7f);
+ snd_soc_write(codec, AIC32X4_NDAC, data | aic32x4_divs[i].ndac);
+
+ /* MDAC divider value */
+ data = snd_soc_read(codec, AIC32X4_MDAC);
+ data &= ~(0x7f);
+ snd_soc_write(codec, AIC32X4_MDAC, data | aic32x4_divs[i].mdac);
+
+ /* DOSR MSB & LSB values */
+ snd_soc_write(codec, AIC32X4_DOSRMSB, aic32x4_divs[i].dosr >> 8);
+ snd_soc_write(codec, AIC32X4_DOSRLSB,
+ (aic32x4_divs[i].dosr & 0xff));
+
+ /* NADC divider value */
+ data = snd_soc_read(codec, AIC32X4_NADC);
+ data &= ~(0x7f);
+ snd_soc_write(codec, AIC32X4_NADC, data | aic32x4_divs[i].nadc);
+
+ /* MADC divider value */
+ data = snd_soc_read(codec, AIC32X4_MADC);
+ data &= ~(0x7f);
+ snd_soc_write(codec, AIC32X4_MADC, data | aic32x4_divs[i].madc);
+
+ /* AOSR value */
+ snd_soc_write(codec, AIC32X4_AOSR, aic32x4_divs[i].aosr);
+
+ /* BCLK N divider */
+ data = snd_soc_read(codec, AIC32X4_BCLKN);
+ data &= ~(0x7f);
+ snd_soc_write(codec, AIC32X4_BCLKN, data | aic32x4_divs[i].blck_N);
+
+ data = snd_soc_read(codec, AIC32X4_IFACE1);
+ data = data & ~(3 << 4);
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ data |= (AIC32X4_WORD_LEN_20BITS << AIC32X4_DOSRMSB_SHIFT);
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ data |= (AIC32X4_WORD_LEN_24BITS << AIC32X4_DOSRMSB_SHIFT);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ data |= (AIC32X4_WORD_LEN_32BITS << AIC32X4_DOSRMSB_SHIFT);
+ break;
+ }
+ snd_soc_write(codec, AIC32X4_IFACE1, data);
+
+ return 0;
+}
+
+static int aic32x4_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 dac_reg;
+
+ dac_reg = snd_soc_read(codec, AIC32X4_DACMUTE) & ~AIC32X4_MUTEON;
+ if (mute)
+ snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg | AIC32X4_MUTEON);
+ else
+ snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg);
+ return 0;
+}
+
+static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+ u8 value;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ if (aic32x4->master) {
+ /* Switch on PLL */
+ value = snd_soc_read(codec, AIC32X4_PLLPR);
+ snd_soc_write(codec, AIC32X4_PLLPR,
+ (value | AIC32X4_PLLEN));
+
+ /* Switch on NDAC Divider */
+ value = snd_soc_read(codec, AIC32X4_NDAC);
+ snd_soc_write(codec, AIC32X4_NDAC,
+ value | AIC32X4_NDACEN);
+
+ /* Switch on MDAC Divider */
+ value = snd_soc_read(codec, AIC32X4_MDAC);
+ snd_soc_write(codec, AIC32X4_MDAC,
+ value | AIC32X4_MDACEN);
+
+ /* Switch on NADC Divider */
+ value = snd_soc_read(codec, AIC32X4_NADC);
+ snd_soc_write(codec, AIC32X4_NADC,
+ value | AIC32X4_MDACEN);
+
+ /* Switch on MADC Divider */
+ value = snd_soc_read(codec, AIC32X4_MADC);
+ snd_soc_write(codec, AIC32X4_MADC,
+ value | AIC32X4_MDACEN);
+
+ /* Switch on BCLK_N Divider */
+ value = snd_soc_read(codec, AIC32X4_BCLKN);
+ snd_soc_write(codec, AIC32X4_BCLKN,
+ value | AIC32X4_BCLKEN);
+ }
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (aic32x4->master) {
+ /* Switch off PLL */
+ value = snd_soc_read(codec, AIC32X4_PLLPR);
+ snd_soc_write(codec, AIC32X4_PLLPR,
+ (value & ~AIC32X4_PLLEN));
+
+ /* Switch off NDAC Divider */
+ value = snd_soc_read(codec, AIC32X4_NDAC);
+ snd_soc_write(codec, AIC32X4_NDAC,
+ value & ~AIC32X4_NDACEN);
+
+ /* Switch off MDAC Divider */
+ value = snd_soc_read(codec, AIC32X4_MDAC);
+ snd_soc_write(codec, AIC32X4_MDAC,
+ value & ~AIC32X4_MDACEN);
+
+ /* Switch off NADC Divider */
+ value = snd_soc_read(codec, AIC32X4_NADC);
+ snd_soc_write(codec, AIC32X4_NADC,
+ value & ~AIC32X4_NDACEN);
+
+ /* Switch off MADC Divider */
+ value = snd_soc_read(codec, AIC32X4_MADC);
+ snd_soc_write(codec, AIC32X4_MADC,
+ value & ~AIC32X4_MDACEN);
+ value = snd_soc_read(codec, AIC32X4_BCLKN);
+
+ /* Switch off BCLK_N Divider */
+ snd_soc_write(codec, AIC32X4_BCLKN,
+ value & ~AIC32X4_BCLKEN);
+ }
+ break;
+ case SND_SOC_BIAS_OFF:
+ break;
+ }
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+#define AIC32X4_RATES SNDRV_PCM_RATE_8000_48000
+#define AIC32X4_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
+ | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops aic32x4_ops = {
+ .hw_params = aic32x4_hw_params,
+ .digital_mute = aic32x4_mute,
+ .set_fmt = aic32x4_set_dai_fmt,
+ .set_sysclk = aic32x4_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver aic32x4_dai = {
+ .name = "tlv320aic32x4-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AIC32X4_RATES,
+ .formats = AIC32X4_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = AIC32X4_RATES,
+ .formats = AIC32X4_FORMATS,},
+ .ops = &aic32x4_ops,
+ .symmetric_rates = 1,
+};
+
+static int aic32x4_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+ aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int aic32x4_resume(struct snd_soc_codec *codec)
+{
+ aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ return 0;
+}
+
+static int aic32x4_probe(struct snd_soc_codec *codec)
+{
+ struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+ u32 tmp_reg;
+
+ codec->hw_write = (hw_write_t) i2c_master_send;
+ codec->control_data = aic32x4->control_data;
+
+ snd_soc_write(codec, AIC32X4_RESET, 0x01);
+
+ /* Power platform configuration */
+ if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) {
+ snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN |
+ AIC32X4_MICBIAS_2075V);
+ }
+ if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) {
+ snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE);
+ }
+ if (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) {
+ snd_soc_write(codec, AIC32X4_LDOCTL, AIC32X4_LDOCTLEN);
+ }
+ tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE);
+ if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) {
+ tmp_reg |= AIC32X4_LDOIN_18_36;
+ }
+ if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) {
+ tmp_reg |= AIC32X4_LDOIN2HP;
+ }
+ snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg);
+
+ /* Do DACs need to be swapped? */
+ if (aic32x4->swapdacs) {
+ snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2RCHN | AIC32X4_RDAC2LCHN);
+ } else {
+ snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2LCHN | AIC32X4_RDAC2RCHN);
+ }
+
+ /* Mic PGA routing */
+ if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) {
+ snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K);
+ }
+ if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) {
+ snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K);
+ }
+
+ aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ snd_soc_add_controls(codec, aic32x4_snd_controls,
+ ARRAY_SIZE(aic32x4_snd_controls));
+ aic32x4_add_widgets(codec);
+
+ return 0;
+}
+
+static int aic32x4_remove(struct snd_soc_codec *codec)
+{
+ aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
+ .read = aic32x4_read,
+ .write = aic32x4_write,
+ .probe = aic32x4_probe,
+ .remove = aic32x4_remove,
+ .suspend = aic32x4_suspend,
+ .resume = aic32x4_resume,
+ .set_bias_level = aic32x4_set_bias_level,
+};
+
+static __devinit int aic32x4_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct aic32x4_pdata *pdata = i2c->dev.platform_data;
+ struct aic32x4_priv *aic32x4;
+ int ret;
+
+ aic32x4 = kzalloc(sizeof(struct aic32x4_priv), GFP_KERNEL);
+ if (aic32x4 == NULL)
+ return -ENOMEM;
+
+ aic32x4->control_data = i2c;
+ i2c_set_clientdata(i2c, aic32x4);
+
+ if (pdata) {
+ aic32x4->power_cfg = pdata->power_cfg;
+ aic32x4->swapdacs = pdata->swapdacs;
+ aic32x4->micpga_routing = pdata->micpga_routing;
+ } else {
+ aic32x4->power_cfg = 0;
+ aic32x4->swapdacs = false;
+ aic32x4->micpga_routing = 0;
+ }
+
+ ret = snd_soc_register_codec(&i2c->dev,
+ &soc_codec_dev_aic32x4, &aic32x4_dai, 1);
+ if (ret < 0)
+ kfree(aic32x4);
+ return ret;
+}
+
+static __devexit int aic32x4_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ kfree(i2c_get_clientdata(client));
+ return 0;
+}
+
+static const struct i2c_device_id aic32x4_i2c_id[] = {
+ { "tlv320aic32x4", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
+
+static struct i2c_driver aic32x4_i2c_driver = {
+ .driver = {
+ .name = "tlv320aic32x4",
+ .owner = THIS_MODULE,
+ },
+ .probe = aic32x4_i2c_probe,
+ .remove = __devexit_p(aic32x4_i2c_remove),
+ .id_table = aic32x4_i2c_id,
+};
+
+static int __init aic32x4_modinit(void)
+{
+ int ret = 0;
+
+ ret = i2c_add_driver(&aic32x4_i2c_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "Failed to register aic32x4 I2C driver: %d\n",
+ ret);
+ }
+ return ret;
+}
+module_init(aic32x4_modinit);
+
+static void __exit aic32x4_exit(void)
+{
+ i2c_del_driver(&aic32x4_i2c_driver);
+}
+module_exit(aic32x4_exit);
+
+MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver");
+MODULE_AUTHOR("Javier Martin <javier.martin@vista-silicon.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h
new file mode 100644
index 0000000..aae2b24
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic32x4.h
@@ -0,0 +1,143 @@
+/*
+ * tlv320aic32x4.h
+ *
+ * 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 _TLV320AIC32X4_H
+#define _TLV320AIC32X4_H
+
+/* tlv320aic32x4 register space (in decimal to match datasheet) */
+
+#define AIC32X4_PAGE1 128
+
+#define AIC32X4_PSEL 0
+#define AIC32X4_RESET 1
+#define AIC32X4_CLKMUX 4
+#define AIC32X4_PLLPR 5
+#define AIC32X4_PLLJ 6
+#define AIC32X4_PLLDMSB 7
+#define AIC32X4_PLLDLSB 8
+#define AIC32X4_NDAC 11
+#define AIC32X4_MDAC 12
+#define AIC32X4_DOSRMSB 13
+#define AIC32X4_DOSRLSB 14
+#define AIC32X4_NADC 18
+#define AIC32X4_MADC 19
+#define AIC32X4_AOSR 20
+#define AIC32X4_CLKMUX2 25
+#define AIC32X4_CLKOUTM 26
+#define AIC32X4_IFACE1 27
+#define AIC32X4_IFACE2 28
+#define AIC32X4_IFACE3 29
+#define AIC32X4_BCLKN 30
+#define AIC32X4_IFACE4 31
+#define AIC32X4_IFACE5 32
+#define AIC32X4_IFACE6 33
+#define AIC32X4_DOUTCTL 53
+#define AIC32X4_DINCTL 54
+#define AIC32X4_DACSPB 60
+#define AIC32X4_ADCSPB 61
+#define AIC32X4_DACSETUP 63
+#define AIC32X4_DACMUTE 64
+#define AIC32X4_LDACVOL 65
+#define AIC32X4_RDACVOL 66
+#define AIC32X4_ADCSETUP 81
+#define AIC32X4_ADCFGA 82
+#define AIC32X4_LADCVOL 83
+#define AIC32X4_RADCVOL 84
+#define AIC32X4_LAGC1 86
+#define AIC32X4_LAGC2 87
+#define AIC32X4_LAGC3 88
+#define AIC32X4_LAGC4 89
+#define AIC32X4_LAGC5 90
+#define AIC32X4_LAGC6 91
+#define AIC32X4_LAGC7 92
+#define AIC32X4_RAGC1 94
+#define AIC32X4_RAGC2 95
+#define AIC32X4_RAGC3 96
+#define AIC32X4_RAGC4 97
+#define AIC32X4_RAGC5 98
+#define AIC32X4_RAGC6 99
+#define AIC32X4_RAGC7 100
+#define AIC32X4_PWRCFG (AIC32X4_PAGE1 + 1)
+#define AIC32X4_LDOCTL (AIC32X4_PAGE1 + 2)
+#define AIC32X4_OUTPWRCTL (AIC32X4_PAGE1 + 9)
+#define AIC32X4_CMMODE (AIC32X4_PAGE1 + 10)
+#define AIC32X4_HPLROUTE (AIC32X4_PAGE1 + 12)
+#define AIC32X4_HPRROUTE (AIC32X4_PAGE1 + 13)
+#define AIC32X4_LOLROUTE (AIC32X4_PAGE1 + 14)
+#define AIC32X4_LORROUTE (AIC32X4_PAGE1 + 15)
+#define AIC32X4_HPLGAIN (AIC32X4_PAGE1 + 16)
+#define AIC32X4_HPRGAIN (AIC32X4_PAGE1 + 17)
+#define AIC32X4_LOLGAIN (AIC32X4_PAGE1 + 18)
+#define AIC32X4_LORGAIN (AIC32X4_PAGE1 + 19)
+#define AIC32X4_HEADSTART (AIC32X4_PAGE1 + 20)
+#define AIC32X4_MICBIAS (AIC32X4_PAGE1 + 51)
+#define AIC32X4_LMICPGAPIN (AIC32X4_PAGE1 + 52)
+#define AIC32X4_LMICPGANIN (AIC32X4_PAGE1 + 54)
+#define AIC32X4_RMICPGAPIN (AIC32X4_PAGE1 + 55)
+#define AIC32X4_RMICPGANIN (AIC32X4_PAGE1 + 57)
+#define AIC32X4_FLOATINGINPUT (AIC32X4_PAGE1 + 58)
+#define AIC32X4_LMICPGAVOL (AIC32X4_PAGE1 + 59)
+#define AIC32X4_RMICPGAVOL (AIC32X4_PAGE1 + 60)
+
+#define AIC32X4_FREQ_12000000 12000000
+#define AIC32X4_FREQ_24000000 24000000
+#define AIC32X4_FREQ_25000000 25000000
+
+#define AIC32X4_WORD_LEN_16BITS 0x00
+#define AIC32X4_WORD_LEN_20BITS 0x01
+#define AIC32X4_WORD_LEN_24BITS 0x02
+#define AIC32X4_WORD_LEN_32BITS 0x03
+
+#define AIC32X4_I2S_MODE 0x00
+#define AIC32X4_DSP_MODE 0x01
+#define AIC32X4_RIGHT_JUSTIFIED_MODE 0x02
+#define AIC32X4_LEFT_JUSTIFIED_MODE 0x03
+
+#define AIC32X4_AVDDWEAKDISABLE 0x08
+#define AIC32X4_LDOCTLEN 0x01
+
+#define AIC32X4_LDOIN_18_36 0x01
+#define AIC32X4_LDOIN2HP 0x02
+
+#define AIC32X4_DACSPBLOCK_MASK 0x1f
+#define AIC32X4_ADCSPBLOCK_MASK 0x1f
+
+#define AIC32X4_PLLJ_SHIFT 6
+#define AIC32X4_DOSRMSB_SHIFT 4
+
+#define AIC32X4_PLLCLKIN 0x03
+
+#define AIC32X4_MICBIAS_LDOIN 0x08
+#define AIC32X4_MICBIAS_2075V 0x60
+
+#define AIC32X4_LMICPGANIN_IN2R_10K 0x10
+#define AIC32X4_RMICPGANIN_IN1L_10K 0x10
+
+#define AIC32X4_LMICPGAVOL_NOGAIN 0x80
+#define AIC32X4_RMICPGAVOL_NOGAIN 0x80
+
+#define AIC32X4_BCLKMASTER 0x08
+#define AIC32X4_WCLKMASTER 0x04
+#define AIC32X4_PLLEN (0x01 << 7)
+#define AIC32X4_NDACEN (0x01 << 7)
+#define AIC32X4_MDACEN (0x01 << 7)
+#define AIC32X4_NADCEN (0x01 << 7)
+#define AIC32X4_MADCEN (0x01 << 7)
+#define AIC32X4_BCLKEN (0x01 << 7)
+#define AIC32X4_DACEN (0x03 << 6)
+#define AIC32X4_RDAC2LCHN (0x02 << 2)
+#define AIC32X4_LDAC2RCHN (0x02 << 4)
+#define AIC32X4_LDAC2LCHN (0x01 << 4)
+#define AIC32X4_RDAC2RCHN (0x01 << 2)
+
+#define AIC32X4_SSTEP2WCLK 0x01
+#define AIC32X4_MUTEON 0x0C
+#define AIC32X4_DACMOD2BCLK 0x01
+
+#endif /* _TLV320AIC32X4_H */
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 71d7be8..00b6d87 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1615,6 +1615,7 @@
},
{ },
};
+MODULE_DEVICE_TABLE(i2c, tlv320dac33_i2c_id);
static struct i2c_driver tlv320dac33_i2c_driver = {
.driver = {
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 80ddf4f..a3b9cbb 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -836,24 +836,25 @@
}
#ifdef CONFIG_PM
-static int wm2000_i2c_suspend(struct i2c_client *i2c, pm_message_t mesg)
+static int wm2000_i2c_suspend(struct device *dev)
{
+ struct i2c_client *i2c = to_i2c_client(dev);
struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev);
return wm2000_anc_transition(wm2000, ANC_OFF);
}
-static int wm2000_i2c_resume(struct i2c_client *i2c)
+static int wm2000_i2c_resume(struct device *dev)
{
+ struct i2c_client *i2c = to_i2c_client(dev);
struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev);
return wm2000_anc_set_mode(wm2000);
}
-#else
-#define wm2000_i2c_suspend NULL
-#define wm2000_i2c_resume NULL
#endif
+static SIMPLE_DEV_PM_OPS(wm2000_pm, wm2000_i2c_suspend, wm2000_i2c_resume);
+
static const struct i2c_device_id wm2000_i2c_id[] = {
{ "wm2000", 0 },
{ }
@@ -864,11 +865,10 @@
.driver = {
.name = "wm2000",
.owner = THIS_MODULE,
+ .pm = &wm2000_pm,
},
.probe = wm2000_i2c_probe,
.remove = __devexit_p(wm2000_i2c_remove),
- .suspend = wm2000_i2c_suspend,
- .resume = wm2000_i2c_resume,
.shutdown = wm2000_i2c_shutdown,
.id_table = wm2000_i2c_id,
};
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 79b02ae..3f09dee 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -55,8 +55,10 @@
module_param(caps_charge, int, 0);
MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
-static void wm8753_set_dai_mode(struct snd_soc_codec *codec,
- struct snd_soc_dai *dai, unsigned int hifi);
+static int wm8753_hifi_write_dai_fmt(struct snd_soc_codec *codec,
+ unsigned int fmt);
+static int wm8753_voice_write_dai_fmt(struct snd_soc_codec *codec,
+ unsigned int fmt);
/*
* wm8753 register cache
@@ -87,6 +89,10 @@
enum snd_soc_control_type control_type;
unsigned int sysclk;
unsigned int pcmclk;
+
+ unsigned int voice_fmt;
+ unsigned int hifi_fmt;
+
int dai_func;
};
@@ -170,9 +176,9 @@
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- int mode = snd_soc_read(codec, WM8753_IOCTL);
+ struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
- ucontrol->value.integer.value[0] = (mode & 0xc) >> 2;
+ ucontrol->value.integer.value[0] = wm8753->dai_func;
return 0;
}
@@ -180,16 +186,26 @@
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- int mode = snd_soc_read(codec, WM8753_IOCTL);
struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
+ u16 ioctl;
- if (((mode & 0xc) >> 2) == ucontrol->value.integer.value[0])
- return 0;
+ if (codec->active)
+ return -EBUSY;
- mode &= 0xfff3;
- mode |= (ucontrol->value.integer.value[0] << 2);
+ ioctl = snd_soc_read(codec, WM8753_IOCTL);
- wm8753->dai_func = ucontrol->value.integer.value[0];
+ wm8753->dai_func = ucontrol->value.integer.value[0];
+
+ if (((ioctl >> 2) & 0x3) == wm8753->dai_func)
+ return 1;
+
+ ioctl = (ioctl & 0x1f3) | (wm8753->dai_func << 2);
+ snd_soc_write(codec, WM8753_IOCTL, ioctl);
+
+
+ wm8753_hifi_write_dai_fmt(codec, wm8753->hifi_fmt);
+ wm8753_voice_write_dai_fmt(codec, wm8753->voice_fmt);
+
return 1;
}
@@ -828,10 +844,9 @@
/*
* Set's ADC and Voice DAC format.
*/
-static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec *codec,
unsigned int fmt)
{
- struct snd_soc_codec *codec = codec_dai->codec;
u16 voice = snd_soc_read(codec, WM8753_PCM) & 0x01ec;
/* interface format */
@@ -858,13 +873,6 @@
return 0;
}
-static int wm8753_pcm_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- wm8753_set_dai_mode(dai->codec, dai, 0);
- return 0;
-}
-
/*
* Set PCM DAI bit size and sample rate.
*/
@@ -905,10 +913,9 @@
/*
* Set's PCM dai fmt and BCLK.
*/
-static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec *codec,
unsigned int fmt)
{
- struct snd_soc_codec *codec = codec_dai->codec;
u16 voice, ioctl;
voice = snd_soc_read(codec, WM8753_PCM) & 0x011f;
@@ -999,10 +1006,9 @@
/*
* Set's HiFi DAC format.
*/
-static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec *codec,
unsigned int fmt)
{
- struct snd_soc_codec *codec = codec_dai->codec;
u16 hifi = snd_soc_read(codec, WM8753_HIFI) & 0x01e0;
/* interface format */
@@ -1032,10 +1038,9 @@
/*
* Set's I2S DAI format.
*/
-static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec *codec,
unsigned int fmt)
{
- struct snd_soc_codec *codec = codec_dai->codec;
u16 ioctl, hifi;
hifi = snd_soc_read(codec, WM8753_HIFI) & 0x011f;
@@ -1098,13 +1103,6 @@
return 0;
}
-static int wm8753_i2s_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- wm8753_set_dai_mode(dai->codec, dai, 1);
- return 0;
-}
-
/*
* Set PCM DAI bit size and sample rate.
*/
@@ -1147,61 +1145,117 @@
return 0;
}
-static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec *codec,
unsigned int fmt)
{
- struct snd_soc_codec *codec = codec_dai->codec;
u16 clock;
/* set clk source as pcmclk */
clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb;
snd_soc_write(codec, WM8753_CLOCK, clock);
- if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
- return -EINVAL;
- return wm8753_pcm_set_dai_fmt(codec_dai, fmt);
+ return wm8753_vdac_adc_set_dai_fmt(codec, fmt);
}
-static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec *codec,
unsigned int fmt)
{
- if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
- return -EINVAL;
- return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
+ return wm8753_hdac_set_dai_fmt(codec, fmt);
}
-static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec *codec,
unsigned int fmt)
{
- struct snd_soc_codec *codec = codec_dai->codec;
u16 clock;
/* set clk source as pcmclk */
clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb;
snd_soc_write(codec, WM8753_CLOCK, clock);
- if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
- return -EINVAL;
- return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
+ return wm8753_vdac_adc_set_dai_fmt(codec, fmt);
}
-static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec *codec,
unsigned int fmt)
{
- struct snd_soc_codec *codec = codec_dai->codec;
u16 clock;
/* set clk source as mclk */
clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb;
snd_soc_write(codec, WM8753_CLOCK, clock | 0x4);
- if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
+ if (wm8753_hdac_set_dai_fmt(codec, fmt) < 0)
return -EINVAL;
- if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
- return -EINVAL;
- return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
+ return wm8753_vdac_adc_set_dai_fmt(codec, fmt);
}
+static int wm8753_hifi_write_dai_fmt(struct snd_soc_codec *codec,
+ unsigned int fmt)
+{
+ struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+
+ switch (wm8753->dai_func) {
+ case 0:
+ ret = wm8753_mode1h_set_dai_fmt(codec, fmt);
+ break;
+ case 1:
+ ret = wm8753_mode2_set_dai_fmt(codec, fmt);
+ break;
+ case 2:
+ case 3:
+ ret = wm8753_mode3_4_set_dai_fmt(codec, fmt);
+ break;
+ default:
+ break;
+ }
+ if (ret)
+ return ret;
+
+ return wm8753_i2s_set_dai_fmt(codec, fmt);
+}
+
+static int wm8753_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
+
+ wm8753->hifi_fmt = fmt;
+
+ return wm8753_hifi_write_dai_fmt(codec, fmt);
+};
+
+static int wm8753_voice_write_dai_fmt(struct snd_soc_codec *codec,
+ unsigned int fmt)
+{
+ struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+
+ if (wm8753->dai_func != 0)
+ return 0;
+
+ ret = wm8753_mode1v_set_dai_fmt(codec, fmt);
+ if (ret)
+ return ret;
+ ret = wm8753_pcm_set_dai_fmt(codec, fmt);
+ if (ret)
+ return ret;
+
+ return 0;
+};
+
+static int wm8753_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
+
+ wm8753->voice_fmt = fmt;
+
+ return wm8753_voice_write_dai_fmt(codec, fmt);
+};
+
static int wm8753_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
@@ -1268,57 +1322,25 @@
* 3. Voice disabled - HIFI over HIFI
* 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture
*/
-static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode1 = {
- .startup = wm8753_i2s_startup,
+static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode = {
.hw_params = wm8753_i2s_hw_params,
.digital_mute = wm8753_mute,
- .set_fmt = wm8753_mode1h_set_dai_fmt,
+ .set_fmt = wm8753_hifi_set_dai_fmt,
.set_clkdiv = wm8753_set_dai_clkdiv,
.set_pll = wm8753_set_dai_pll,
.set_sysclk = wm8753_set_dai_sysclk,
};
-static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode1 = {
- .startup = wm8753_pcm_startup,
+static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode = {
.hw_params = wm8753_pcm_hw_params,
.digital_mute = wm8753_mute,
- .set_fmt = wm8753_mode1v_set_dai_fmt,
+ .set_fmt = wm8753_voice_set_dai_fmt,
.set_clkdiv = wm8753_set_dai_clkdiv,
.set_pll = wm8753_set_dai_pll,
.set_sysclk = wm8753_set_dai_sysclk,
};
-static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode2 = {
- .startup = wm8753_pcm_startup,
- .hw_params = wm8753_pcm_hw_params,
- .digital_mute = wm8753_mute,
- .set_fmt = wm8753_mode2_set_dai_fmt,
- .set_clkdiv = wm8753_set_dai_clkdiv,
- .set_pll = wm8753_set_dai_pll,
- .set_sysclk = wm8753_set_dai_sysclk,
-};
-
-static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode3 = {
- .startup = wm8753_i2s_startup,
- .hw_params = wm8753_i2s_hw_params,
- .digital_mute = wm8753_mute,
- .set_fmt = wm8753_mode3_4_set_dai_fmt,
- .set_clkdiv = wm8753_set_dai_clkdiv,
- .set_pll = wm8753_set_dai_pll,
- .set_sysclk = wm8753_set_dai_sysclk,
-};
-
-static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode4 = {
- .startup = wm8753_i2s_startup,
- .hw_params = wm8753_i2s_hw_params,
- .digital_mute = wm8753_mute,
- .set_fmt = wm8753_mode3_4_set_dai_fmt,
- .set_clkdiv = wm8753_set_dai_clkdiv,
- .set_pll = wm8753_set_dai_pll,
- .set_sysclk = wm8753_set_dai_sysclk,
-};
-
-static struct snd_soc_dai_driver wm8753_all_dai[] = {
+static struct snd_soc_dai_driver wm8753_dai[] = {
/* DAI HiFi mode 1 */
{ .name = "wm8753-hifi",
.playback = {
@@ -1326,14 +1348,16 @@
.channels_min = 1,
.channels_max = 2,
.rates = WM8753_RATES,
- .formats = WM8753_FORMATS},
+ .formats = WM8753_FORMATS
+ },
.capture = { /* dummy for fast DAI switching */
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8753_RATES,
- .formats = WM8753_FORMATS},
- .ops = &wm8753_dai_ops_hifi_mode1,
+ .formats = WM8753_FORMATS
+ },
+ .ops = &wm8753_dai_ops_hifi_mode,
},
/* DAI Voice mode 1 */
{ .name = "wm8753-voice",
@@ -1342,97 +1366,19 @@
.channels_min = 1,
.channels_max = 1,
.rates = WM8753_RATES,
- .formats = WM8753_FORMATS,},
+ .formats = WM8753_FORMATS,
+ },
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8753_RATES,
- .formats = WM8753_FORMATS,},
- .ops = &wm8753_dai_ops_voice_mode1,
-},
-/* DAI HiFi mode 2 - dummy */
-{ .name = "wm8753-hifi",
-},
-/* DAI Voice mode 2 */
-{ .name = "wm8753-voice",
- .playback = {
- .stream_name = "Voice Playback",
- .channels_min = 1,
- .channels_max = 1,
- .rates = WM8753_RATES,
- .formats = WM8753_FORMATS,},
- .capture = {
- .stream_name = "Capture",
- .channels_min = 1,
- .channels_max = 2,
- .rates = WM8753_RATES,
- .formats = WM8753_FORMATS,},
- .ops = &wm8753_dai_ops_voice_mode2,
-},
-/* DAI HiFi mode 3 */
-{ .name = "wm8753-hifi",
- .playback = {
- .stream_name = "HiFi Playback",
- .channels_min = 1,
- .channels_max = 2,
- .rates = WM8753_RATES,
- .formats = WM8753_FORMATS,},
- .capture = {
- .stream_name = "Capture",
- .channels_min = 1,
- .channels_max = 2,
- .rates = WM8753_RATES,
- .formats = WM8753_FORMATS,},
- .ops = &wm8753_dai_ops_hifi_mode3,
-},
-/* DAI Voice mode 3 - dummy */
-{ .name = "wm8753-voice",
-},
-/* DAI HiFi mode 4 */
-{ .name = "wm8753-hifi",
- .playback = {
- .stream_name = "HiFi Playback",
- .channels_min = 1,
- .channels_max = 2,
- .rates = WM8753_RATES,
- .formats = WM8753_FORMATS,},
- .capture = {
- .stream_name = "Capture",
- .channels_min = 1,
- .channels_max = 2,
- .rates = WM8753_RATES,
- .formats = WM8753_FORMATS,},
- .ops = &wm8753_dai_ops_hifi_mode4,
-},
-/* DAI Voice mode 4 - dummy */
-{ .name = "wm8753-voice",
+ .formats = WM8753_FORMATS,
+ },
+ .ops = &wm8753_dai_ops_voice_mode,
},
};
-static struct snd_soc_dai_driver wm8753_dai[] = {
- {
- .name = "wm8753-aif0",
- },
- {
- .name = "wm8753-aif1",
- },
-};
-
-static void wm8753_set_dai_mode(struct snd_soc_codec *codec,
- struct snd_soc_dai *dai, unsigned int hifi)
-{
- struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
-
- if (wm8753->dai_func < 4) {
- if (hifi)
- dai->driver = &wm8753_all_dai[wm8753->dai_func << 1];
- else
- dai->driver = &wm8753_all_dai[(wm8753->dai_func << 1) + 1];
- }
- snd_soc_write(codec, WM8753_IOCTL, wm8753->dai_func);
-}
-
static void wm8753_work(struct work_struct *work)
{
struct snd_soc_dapm_context *dapm =
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 3d4c55f..ae1cadf 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -223,11 +223,12 @@
int fs;
int deemph;
+ int dcs_pending;
+ int dcs_cache[4];
+
/* Reference count */
int class_w_users;
- struct completion wseq;
-
struct snd_soc_jack *mic_jack;
int mic_det;
int mic_short;
@@ -246,6 +247,12 @@
case WM8903_REVISION_NUMBER:
case WM8903_INTERRUPT_STATUS_1:
case WM8903_WRITE_SEQUENCER_4:
+ case WM8903_POWER_MANAGEMENT_3:
+ case WM8903_POWER_MANAGEMENT_2:
+ case WM8903_DC_SERVO_READBACK_1:
+ case WM8903_DC_SERVO_READBACK_2:
+ case WM8903_DC_SERVO_READBACK_3:
+ case WM8903_DC_SERVO_READBACK_4:
return 1;
default:
@@ -253,50 +260,6 @@
}
}
-static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start)
-{
- u16 reg[5];
- struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-
- BUG_ON(start > 48);
-
- /* Enable the sequencer if it's not already on */
- reg[0] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_0);
- snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0,
- reg[0] | WM8903_WSEQ_ENA);
-
- dev_dbg(codec->dev, "Starting sequence at %d\n", start);
-
- snd_soc_write(codec, WM8903_WRITE_SEQUENCER_3,
- start | WM8903_WSEQ_START);
-
- /* Wait for it to complete. If we have the interrupt wired up then
- * that will break us out of the poll early.
- */
- do {
- wait_for_completion_timeout(&wm8903->wseq,
- msecs_to_jiffies(10));
-
- reg[4] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_4);
- } while (reg[4] & WM8903_WSEQ_BUSY);
-
- dev_dbg(codec->dev, "Sequence complete\n");
-
- /* Disable the sequencer again if we enabled it */
- snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, reg[0]);
-
- return 0;
-}
-
-static void wm8903_sync_reg_cache(struct snd_soc_codec *codec, u16 *cache)
-{
- int i;
-
- /* There really ought to be something better we can do here :/ */
- for (i = 0; i < ARRAY_SIZE(wm8903_reg_defaults); i++)
- cache[i] = codec->hw_read(codec, i);
-}
-
static void wm8903_reset(struct snd_soc_codec *codec)
{
snd_soc_write(codec, WM8903_SW_RESET_AND_ID, 0);
@@ -304,11 +267,6 @@
sizeof(wm8903_reg_defaults));
}
-#define WM8903_OUTPUT_SHORT 0x8
-#define WM8903_OUTPUT_OUT 0x4
-#define WM8903_OUTPUT_INT 0x2
-#define WM8903_OUTPUT_IN 0x1
-
static int wm8903_cp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
@@ -318,99 +276,103 @@
return 0;
}
-/*
- * Event for headphone and line out amplifier power changes. Special
- * power up/down sequences are required in order to maximise pop/click
- * performance.
- */
-static int wm8903_output_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
+static int wm8903_dcs_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
- u16 val;
- u16 reg;
- u16 dcs_reg;
- u16 dcs_bit;
- int shift;
+ struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
- switch (w->reg) {
- case WM8903_POWER_MANAGEMENT_2:
- reg = WM8903_ANALOGUE_HP_0;
- dcs_bit = 0 + w->shift;
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ wm8903->dcs_pending |= 1 << w->shift;
break;
- case WM8903_POWER_MANAGEMENT_3:
- reg = WM8903_ANALOGUE_LINEOUT_0;
- dcs_bit = 2 + w->shift;
+ case SND_SOC_DAPM_PRE_PMD:
+ snd_soc_update_bits(codec, WM8903_DC_SERVO_0,
+ 1 << w->shift, 0);
break;
- default:
- BUG();
- return -EINVAL; /* Spurious warning from some compilers */
- }
-
- switch (w->shift) {
- case 0:
- shift = 0;
- break;
- case 1:
- shift = 4;
- break;
- default:
- BUG();
- return -EINVAL; /* Spurious warning from some compilers */
- }
-
- if (event & SND_SOC_DAPM_PRE_PMU) {
- val = snd_soc_read(codec, reg);
-
- /* Short the output */
- val &= ~(WM8903_OUTPUT_SHORT << shift);
- snd_soc_write(codec, reg, val);
- }
-
- if (event & SND_SOC_DAPM_POST_PMU) {
- val = snd_soc_read(codec, reg);
-
- val |= (WM8903_OUTPUT_IN << shift);
- snd_soc_write(codec, reg, val);
-
- val |= (WM8903_OUTPUT_INT << shift);
- snd_soc_write(codec, reg, val);
-
- /* Turn on the output ENA_OUTP */
- val |= (WM8903_OUTPUT_OUT << shift);
- snd_soc_write(codec, reg, val);
-
- /* Enable the DC servo */
- dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0);
- dcs_reg |= dcs_bit;
- snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg);
-
- /* Remove the short */
- val |= (WM8903_OUTPUT_SHORT << shift);
- snd_soc_write(codec, reg, val);
- }
-
- if (event & SND_SOC_DAPM_PRE_PMD) {
- val = snd_soc_read(codec, reg);
-
- /* Short the output */
- val &= ~(WM8903_OUTPUT_SHORT << shift);
- snd_soc_write(codec, reg, val);
-
- /* Disable the DC servo */
- dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0);
- dcs_reg &= ~dcs_bit;
- snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg);
-
- /* Then disable the intermediate and output stages */
- val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT |
- WM8903_OUTPUT_IN) << shift);
- snd_soc_write(codec, reg, val);
}
return 0;
}
+#define WM8903_DCS_MODE_WRITE_STOP 0
+#define WM8903_DCS_MODE_START_STOP 2
+
+static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm,
+ enum snd_soc_dapm_type event, int subseq)
+{
+ struct snd_soc_codec *codec = container_of(dapm,
+ struct snd_soc_codec, dapm);
+ struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
+ int dcs_mode = WM8903_DCS_MODE_WRITE_STOP;
+ int i, val;
+
+ /* Complete any pending DC servo starts */
+ if (wm8903->dcs_pending) {
+ dev_dbg(codec->dev, "Starting DC servo for %x\n",
+ wm8903->dcs_pending);
+
+ /* If we've no cached values then we need to do startup */
+ for (i = 0; i < ARRAY_SIZE(wm8903->dcs_cache); i++) {
+ if (!(wm8903->dcs_pending & (1 << i)))
+ continue;
+
+ if (wm8903->dcs_cache[i]) {
+ dev_dbg(codec->dev,
+ "Restore DC servo %d value %x\n",
+ 3 - i, wm8903->dcs_cache[i]);
+
+ snd_soc_write(codec, WM8903_DC_SERVO_4 + i,
+ wm8903->dcs_cache[i] & 0xff);
+ } else {
+ dev_dbg(codec->dev,
+ "Calibrate DC servo %d\n", 3 - i);
+ dcs_mode = WM8903_DCS_MODE_START_STOP;
+ }
+ }
+
+ /* Don't trust the cache for analogue */
+ if (wm8903->class_w_users)
+ dcs_mode = WM8903_DCS_MODE_START_STOP;
+
+ snd_soc_update_bits(codec, WM8903_DC_SERVO_2,
+ WM8903_DCS_MODE_MASK, dcs_mode);
+
+ snd_soc_update_bits(codec, WM8903_DC_SERVO_0,
+ WM8903_DCS_ENA_MASK, wm8903->dcs_pending);
+
+ switch (dcs_mode) {
+ case WM8903_DCS_MODE_WRITE_STOP:
+ break;
+
+ case WM8903_DCS_MODE_START_STOP:
+ msleep(270);
+
+ /* Cache the measured offsets for digital */
+ if (wm8903->class_w_users)
+ break;
+
+ for (i = 0; i < ARRAY_SIZE(wm8903->dcs_cache); i++) {
+ if (!(wm8903->dcs_pending & (1 << i)))
+ continue;
+
+ val = snd_soc_read(codec,
+ WM8903_DC_SERVO_READBACK_1 + i);
+ dev_dbg(codec->dev, "DC servo %d: %x\n",
+ 3 - i, val);
+ wm8903->dcs_cache[i] = val;
+ }
+ break;
+
+ default:
+ pr_warn("DCS mode %d delay not set\n", dcs_mode);
+ break;
+ }
+
+ wm8903->dcs_pending = 0;
+ }
+}
+
/*
* When used with DAC outputs only the WM8903 charge pump supports
* operation in class W mode, providing very low power consumption
@@ -674,6 +636,22 @@
static const struct soc_enum rsidetone_enum =
SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text);
+static const char *aif_text[] = {
+ "Left", "Right"
+};
+
+static const struct soc_enum lcapture_enum =
+ SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 7, 2, aif_text);
+
+static const struct soc_enum rcapture_enum =
+ SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 6, 2, aif_text);
+
+static const struct soc_enum lplay_enum =
+ SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 5, 2, aif_text);
+
+static const struct soc_enum rplay_enum =
+ SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 4, 2, aif_text);
+
static const struct snd_kcontrol_new wm8903_snd_controls[] = {
/* Input PGAs - No TLV since the scale depends on PGA mode */
@@ -791,6 +769,18 @@
static const struct snd_kcontrol_new rsidetone_mux =
SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum);
+static const struct snd_kcontrol_new lcapture_mux =
+ SOC_DAPM_ENUM("Left Capture Mux", lcapture_enum);
+
+static const struct snd_kcontrol_new rcapture_mux =
+ SOC_DAPM_ENUM("Right Capture Mux", rcapture_enum);
+
+static const struct snd_kcontrol_new lplay_mux =
+ SOC_DAPM_ENUM("Left Playback Mux", lplay_enum);
+
+static const struct snd_kcontrol_new rplay_mux =
+ SOC_DAPM_ENUM("Right Playback Mux", rplay_enum);
+
static const struct snd_kcontrol_new left_output_mixer[] = {
SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0),
SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0),
@@ -854,14 +844,26 @@
SND_SOC_DAPM_PGA("Left Input PGA", WM8903_POWER_MANAGEMENT_0, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0),
-SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8903_POWER_MANAGEMENT_6, 1, 0),
-SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8903_POWER_MANAGEMENT_6, 0, 0),
+SND_SOC_DAPM_ADC("ADCL", NULL, WM8903_POWER_MANAGEMENT_6, 1, 0),
+SND_SOC_DAPM_ADC("ADCR", NULL, WM8903_POWER_MANAGEMENT_6, 0, 0),
+
+SND_SOC_DAPM_MUX("Left Capture Mux", SND_SOC_NOPM, 0, 0, &lcapture_mux),
+SND_SOC_DAPM_MUX("Right Capture Mux", SND_SOC_NOPM, 0, 0, &rcapture_mux),
+
+SND_SOC_DAPM_AIF_OUT("AIFTXL", "Left HiFi Capture", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIFTXR", "Right HiFi Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &lsidetone_mux),
SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &rsidetone_mux),
-SND_SOC_DAPM_DAC("DACL", "Left Playback", WM8903_POWER_MANAGEMENT_6, 3, 0),
-SND_SOC_DAPM_DAC("DACR", "Right Playback", WM8903_POWER_MANAGEMENT_6, 2, 0),
+SND_SOC_DAPM_AIF_IN("AIFRXL", "Left Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_IN("AIFRXR", "Right Playback", 0, SND_SOC_NOPM, 0, 0),
+
+SND_SOC_DAPM_MUX("Left Playback Mux", SND_SOC_NOPM, 0, 0, &lplay_mux),
+SND_SOC_DAPM_MUX("Right Playback Mux", SND_SOC_NOPM, 0, 0, &rplay_mux),
+
+SND_SOC_DAPM_DAC("DACL", NULL, WM8903_POWER_MANAGEMENT_6, 3, 0),
+SND_SOC_DAPM_DAC("DACR", NULL, WM8903_POWER_MANAGEMENT_6, 2, 0),
SND_SOC_DAPM_MIXER("Left Output Mixer", WM8903_POWER_MANAGEMENT_1, 1, 0,
left_output_mixer, ARRAY_SIZE(left_output_mixer)),
@@ -873,23 +875,45 @@
SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0,
right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
-SND_SOC_DAPM_PGA_E("Left Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
- 1, 0, NULL, 0, wm8903_output_event,
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
- SND_SOC_DAPM_PRE_PMD),
-SND_SOC_DAPM_PGA_E("Right Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
- 0, 0, NULL, 0, wm8903_output_event,
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
- SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
+ 4, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
+ 0, 0, NULL, 0),
-SND_SOC_DAPM_PGA_E("Left Line Output PGA", WM8903_POWER_MANAGEMENT_3, 1, 0,
- NULL, 0, wm8903_output_event,
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
- SND_SOC_DAPM_PRE_PMD),
-SND_SOC_DAPM_PGA_E("Right Line Output PGA", WM8903_POWER_MANAGEMENT_3, 0, 0,
- NULL, 0, wm8903_output_event,
- SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
- SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_DCS", 3, SND_SOC_NOPM, 3, 0, wm8903_dcs_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA_S("HPR_DCS", 3, SND_SOC_NOPM, 2, 0, wm8903_dcs_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA_S("LINEOUTL_DCS", 3, SND_SOC_NOPM, 1, 0, wm8903_dcs_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA_S("LINEOUTR_DCS", 3, SND_SOC_NOPM, 0, 0, wm8903_dcs_event,
+ SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0,
NULL, 0),
@@ -899,10 +923,18 @@
SND_SOC_DAPM_SUPPLY("Charge Pump", WM8903_CHARGE_PUMP_0, 0, 0,
wm8903_cp_event, SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8903_CLOCK_RATES_2, 1, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8903_CLOCK_RATES_2, 2, 0, NULL, 0),
};
static const struct snd_soc_dapm_route intercon[] = {
+ { "CLK_DSP", NULL, "CLK_SYS" },
+ { "Mic Bias", NULL, "CLK_SYS" },
+ { "HPL_DCS", NULL, "CLK_SYS" },
+ { "HPR_DCS", NULL, "CLK_SYS" },
+ { "LINEOUTL_DCS", NULL, "CLK_SYS" },
+ { "LINEOUTR_DCS", NULL, "CLK_SYS" },
+
{ "Left Input Mux", "IN1L", "IN1L" },
{ "Left Input Mux", "IN2L", "IN2L" },
{ "Left Input Mux", "IN3L", "IN3L" },
@@ -943,18 +975,36 @@
{ "Left Input PGA", NULL, "Left Input Mode Mux" },
{ "Right Input PGA", NULL, "Right Input Mode Mux" },
+ { "Left Capture Mux", "Left", "ADCL" },
+ { "Left Capture Mux", "Right", "ADCR" },
+
+ { "Right Capture Mux", "Left", "ADCL" },
+ { "Right Capture Mux", "Right", "ADCR" },
+
+ { "AIFTXL", NULL, "Left Capture Mux" },
+ { "AIFTXR", NULL, "Right Capture Mux" },
+
{ "ADCL", NULL, "Left Input PGA" },
{ "ADCL", NULL, "CLK_DSP" },
{ "ADCR", NULL, "Right Input PGA" },
{ "ADCR", NULL, "CLK_DSP" },
+ { "Left Playback Mux", "Left", "AIFRXL" },
+ { "Left Playback Mux", "Right", "AIFRXR" },
+
+ { "Right Playback Mux", "Left", "AIFRXL" },
+ { "Right Playback Mux", "Right", "AIFRXR" },
+
{ "DACL Sidetone", "Left", "ADCL" },
{ "DACL Sidetone", "Right", "ADCR" },
{ "DACR Sidetone", "Left", "ADCL" },
{ "DACR Sidetone", "Right", "ADCR" },
+ { "DACL", NULL, "Left Playback Mux" },
{ "DACL", NULL, "DACL Sidetone" },
{ "DACL", NULL, "CLK_DSP" },
+
+ { "DACR", NULL, "Right Playback Mux" },
{ "DACR", NULL, "DACR Sidetone" },
{ "DACR", NULL, "CLK_DSP" },
@@ -987,11 +1037,35 @@
{ "Left Speaker PGA", NULL, "Left Speaker Mixer" },
{ "Right Speaker PGA", NULL, "Right Speaker Mixer" },
- { "HPOUTL", NULL, "Left Headphone Output PGA" },
- { "HPOUTR", NULL, "Right Headphone Output PGA" },
+ { "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" },
+ { "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" },
+ { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" },
+ { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" },
- { "LINEOUTL", NULL, "Left Line Output PGA" },
- { "LINEOUTR", NULL, "Right Line Output PGA" },
+ { "HPL_DCS", NULL, "DCS Master" },
+ { "HPR_DCS", NULL, "DCS Master" },
+ { "LINEOUTL_DCS", NULL, "DCS Master" },
+ { "LINEOUTR_DCS", NULL, "DCS Master" },
+
+ { "HPL_DCS", NULL, "HPL_ENA_DLY" },
+ { "HPR_DCS", NULL, "HPR_ENA_DLY" },
+ { "LINEOUTL_DCS", NULL, "LINEOUTL_ENA_DLY" },
+ { "LINEOUTR_DCS", NULL, "LINEOUTR_ENA_DLY" },
+
+ { "HPL_ENA_OUTP", NULL, "HPL_DCS" },
+ { "HPR_ENA_OUTP", NULL, "HPR_DCS" },
+ { "LINEOUTL_ENA_OUTP", NULL, "LINEOUTL_DCS" },
+ { "LINEOUTR_ENA_OUTP", NULL, "LINEOUTR_DCS" },
+
+ { "HPL_RMV_SHORT", NULL, "HPL_ENA_OUTP" },
+ { "HPR_RMV_SHORT", NULL, "HPR_ENA_OUTP" },
+ { "LINEOUTL_RMV_SHORT", NULL, "LINEOUTL_ENA_OUTP" },
+ { "LINEOUTR_RMV_SHORT", NULL, "LINEOUTR_ENA_OUTP" },
+
+ { "HPOUTL", NULL, "HPL_RMV_SHORT" },
+ { "HPOUTR", NULL, "HPR_RMV_SHORT" },
+ { "LINEOUTL", NULL, "LINEOUTL_RMV_SHORT" },
+ { "LINEOUTR", NULL, "LINEOUTR_RMV_SHORT" },
{ "LOP", NULL, "Left Speaker PGA" },
{ "LON", NULL, "Left Speaker PGA" },
@@ -1019,29 +1093,71 @@
static int wm8903_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
- u16 reg;
-
switch (level) {
case SND_SOC_BIAS_ON:
+ break;
+
case SND_SOC_BIAS_PREPARE:
- reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0);
- reg &= ~(WM8903_VMID_RES_MASK);
- reg |= WM8903_VMID_RES_50K;
- snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg);
+ snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+ WM8903_VMID_RES_MASK,
+ WM8903_VMID_RES_50K);
break;
case SND_SOC_BIAS_STANDBY:
if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
- snd_soc_write(codec, WM8903_CLOCK_RATES_2,
- WM8903_CLK_SYS_ENA);
+ snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0,
+ WM8903_POBCTRL | WM8903_ISEL_MASK |
+ WM8903_STARTUP_BIAS_ENA |
+ WM8903_BIAS_ENA,
+ WM8903_POBCTRL |
+ (2 << WM8903_ISEL_SHIFT) |
+ WM8903_STARTUP_BIAS_ENA);
- /* Change DC servo dither level in startup sequence */
- snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11);
- snd_soc_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257);
- snd_soc_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2);
+ snd_soc_update_bits(codec,
+ WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0,
+ WM8903_SPK_DISCHARGE,
+ WM8903_SPK_DISCHARGE);
- wm8903_run_sequence(codec, 0);
- wm8903_sync_reg_cache(codec, codec->reg_cache);
+ msleep(33);
+
+ snd_soc_update_bits(codec, WM8903_POWER_MANAGEMENT_5,
+ WM8903_SPKL_ENA | WM8903_SPKR_ENA,
+ WM8903_SPKL_ENA | WM8903_SPKR_ENA);
+
+ snd_soc_update_bits(codec,
+ WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0,
+ WM8903_SPK_DISCHARGE, 0);
+
+ snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+ WM8903_VMID_TIE_ENA |
+ WM8903_BUFIO_ENA |
+ WM8903_VMID_IO_ENA |
+ WM8903_VMID_SOFT_MASK |
+ WM8903_VMID_RES_MASK |
+ WM8903_VMID_BUF_ENA,
+ WM8903_VMID_TIE_ENA |
+ WM8903_BUFIO_ENA |
+ WM8903_VMID_IO_ENA |
+ (2 << WM8903_VMID_SOFT_SHIFT) |
+ WM8903_VMID_RES_250K |
+ WM8903_VMID_BUF_ENA);
+
+ msleep(129);
+
+ snd_soc_update_bits(codec, WM8903_POWER_MANAGEMENT_5,
+ WM8903_SPKL_ENA | WM8903_SPKR_ENA,
+ 0);
+
+ snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+ WM8903_VMID_SOFT_MASK, 0);
+
+ snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+ WM8903_VMID_RES_MASK,
+ WM8903_VMID_RES_50K);
+
+ snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0,
+ WM8903_BIAS_ENA | WM8903_POBCTRL,
+ WM8903_BIAS_ENA);
/* By default no bypass paths are enabled so
* enable Class W support.
@@ -1054,17 +1170,32 @@
WM8903_CP_DYN_V);
}
- reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0);
- reg &= ~(WM8903_VMID_RES_MASK);
- reg |= WM8903_VMID_RES_250K;
- snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg);
+ snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+ WM8903_VMID_RES_MASK,
+ WM8903_VMID_RES_250K);
break;
case SND_SOC_BIAS_OFF:
- wm8903_run_sequence(codec, 32);
- reg = snd_soc_read(codec, WM8903_CLOCK_RATES_2);
- reg &= ~WM8903_CLK_SYS_ENA;
- snd_soc_write(codec, WM8903_CLOCK_RATES_2, reg);
+ snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0,
+ WM8903_BIAS_ENA, 0);
+
+ snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+ WM8903_VMID_SOFT_MASK,
+ 2 << WM8903_VMID_SOFT_SHIFT);
+
+ snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+ WM8903_VMID_BUF_ENA, 0);
+
+ msleep(290);
+
+ snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+ WM8903_VMID_TIE_ENA | WM8903_BUFIO_ENA |
+ WM8903_VMID_IO_ENA | WM8903_VMID_RES_MASK |
+ WM8903_VMID_SOFT_MASK |
+ WM8903_VMID_BUF_ENA, 0);
+
+ snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0,
+ WM8903_STARTUP_BIAS_ENA, 0);
break;
}
@@ -1489,7 +1620,7 @@
WM8903_MICDET_EINT | WM8903_MICSHRT_EINT,
irq_mask);
- if (det && shrt) {
+ if (det || shrt) {
/* Enable mic detection, this may not have been set through
* platform data (eg, if the defaults are OK). */
snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0,
@@ -1517,8 +1648,7 @@
int_val = snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1) & mask;
if (int_val & WM8903_WSEQ_BUSY_EINT) {
- dev_dbg(codec->dev, "Write sequencer done\n");
- complete(&wm8903->wseq);
+ dev_warn(codec->dev, "Write sequencer done\n");
}
/*
@@ -1765,7 +1895,6 @@
u16 val;
wm8903->codec = codec;
- init_completion(&wm8903->wseq);
ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
if (ret != 0) {
@@ -1781,19 +1910,33 @@
}
val = snd_soc_read(codec, WM8903_REVISION_NUMBER);
- dev_info(codec->dev, "WM8903 revision %d\n",
- val & WM8903_CHIP_REV_MASK);
+ dev_info(codec->dev, "WM8903 revision %c\n",
+ (val & WM8903_CHIP_REV_MASK) + 'A');
wm8903_reset(codec);
/* Set up GPIOs and microphone detection */
if (pdata) {
+ bool mic_gpio = false;
+
for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) {
if (pdata->gpio_cfg[i] == WM8903_GPIO_NO_CONFIG)
continue;
snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i,
pdata->gpio_cfg[i] & 0xffff);
+
+ val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK)
+ >> WM8903_GP1_FN_SHIFT;
+
+ switch (val) {
+ case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT:
+ case WM8903_GPn_FN_MICBIAS_SHORT_DETECT:
+ mic_gpio = true;
+ break;
+ default:
+ break;
+ }
}
snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0,
@@ -1804,6 +1947,14 @@
snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0,
WM8903_WSEQ_ENA, WM8903_WSEQ_ENA);
+ /* If microphone detection is enabled by pdata but
+ * detected via IRQ then interrupts can be lost before
+ * the machine driver has set up microphone detection
+ * IRQs as the IRQs are clear on read. The detection
+ * will be enabled when the machine driver configures.
+ */
+ WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA));
+
wm8903->mic_delay = pdata->micdet_delay;
}
@@ -1863,9 +2014,9 @@
snd_soc_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val);
/* Enable DAC soft mute by default */
- val = snd_soc_read(codec, WM8903_DAC_DIGITAL_1);
- val |= WM8903_DAC_MUTEMODE;
- snd_soc_write(codec, WM8903_DAC_DIGITAL_1, val);
+ snd_soc_update_bits(codec, WM8903_DAC_DIGITAL_1,
+ WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE,
+ WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE);
snd_soc_add_controls(codec, wm8903_snd_controls,
ARRAY_SIZE(wm8903_snd_controls));
@@ -1894,6 +2045,7 @@
.reg_word_size = sizeof(u16),
.reg_cache_default = wm8903_reg_defaults,
.volatile_register = wm8903_volatile_register,
+ .seq_notifier = wm8903_seq_notifier,
};
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
@@ -1932,7 +2084,7 @@
static struct i2c_driver wm8903_i2c_driver = {
.driver = {
- .name = "wm8903-codec",
+ .name = "wm8903",
.owner = THIS_MODULE,
},
.probe = wm8903_i2c_probe,
diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h
index e8490f3..db94931 100644
--- a/sound/soc/codecs/wm8903.h
+++ b/sound/soc/codecs/wm8903.h
@@ -75,6 +75,14 @@
#define WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0 0x41
#define WM8903_DC_SERVO_0 0x43
#define WM8903_DC_SERVO_2 0x45
+#define WM8903_DC_SERVO_4 0x47
+#define WM8903_DC_SERVO_5 0x48
+#define WM8903_DC_SERVO_6 0x49
+#define WM8903_DC_SERVO_7 0x4A
+#define WM8903_DC_SERVO_READBACK_1 0x51
+#define WM8903_DC_SERVO_READBACK_2 0x52
+#define WM8903_DC_SERVO_READBACK_3 0x53
+#define WM8903_DC_SERVO_READBACK_4 0x54
#define WM8903_ANALOGUE_HP_0 0x5A
#define WM8903_ANALOGUE_LINEOUT_0 0x5E
#define WM8903_CHARGE_PUMP_0 0x62
@@ -165,7 +173,7 @@
#define WM8903_VMID_RES_50K 2
#define WM8903_VMID_RES_250K 3
-#define WM8903_VMID_RES_5K 4
+#define WM8903_VMID_RES_5K 6
/*
* R8 (0x08) - Analogue DAC 0
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 30fb48e..85e3e63 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -93,6 +93,7 @@
static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0);
static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0);
static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1);
+static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0);
static const struct snd_kcontrol_new wm8978_snd_controls[] = {
@@ -144,19 +145,19 @@
SOC_SINGLE("DAC Playback Limiter Threshold",
WM8978_DAC_LIMITER_2, 4, 7, 0),
- SOC_SINGLE("DAC Playback Limiter Boost",
- WM8978_DAC_LIMITER_2, 0, 15, 0),
+ SOC_SINGLE_TLV("DAC Playback Limiter Volume",
+ WM8978_DAC_LIMITER_2, 0, 12, 0, limiter_tlv),
SOC_ENUM("ALC Enable Switch", alc1),
SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0),
SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0),
- SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 7, 0),
+ SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 10, 0),
SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0),
SOC_ENUM("ALC Capture Mode", alc3),
- SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 15, 0),
- SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 15, 0),
+ SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 10, 0),
+ SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 10, 0),
SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0),
SOC_SINGLE("ALC Capture Noise Gate Threshold",
@@ -211,8 +212,10 @@
WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1),
/* DAC / ADC oversampling */
- SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, 8, 1, 0),
- SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, 8, 1, 0),
+ SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL,
+ 5, 1, 0),
+ SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL,
+ 5, 1, 0),
};
/* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */
diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c
index 68e9b02..a87adbd 100644
--- a/sound/soc/codecs/wm8994-tables.c
+++ b/sound/soc/codecs/wm8994-tables.c
@@ -62,8 +62,8 @@
{ 0x00FF, 0x00FF }, /* R58 - MICBIAS */
{ 0x000F, 0x000F }, /* R59 - LDO 1 */
{ 0x0007, 0x0007 }, /* R60 - LDO 2 */
- { 0x0000, 0x0000 }, /* R61 */
- { 0x0000, 0x0000 }, /* R62 */
+ { 0xFFFF, 0xFFFF }, /* R61 */
+ { 0xFFFF, 0xFFFF }, /* R62 */
{ 0x0000, 0x0000 }, /* R63 */
{ 0x0000, 0x0000 }, /* R64 */
{ 0x0000, 0x0000 }, /* R65 */
@@ -209,9 +209,9 @@
{ 0x0000, 0x0000 }, /* R205 */
{ 0x0000, 0x0000 }, /* R206 */
{ 0x0000, 0x0000 }, /* R207 */
- { 0x0000, 0x0000 }, /* R208 */
- { 0x0000, 0x0000 }, /* R209 */
- { 0x0000, 0x0000 }, /* R210 */
+ { 0xFFFF, 0xFFFF }, /* R208 */
+ { 0xFFFF, 0xFFFF }, /* R209 */
+ { 0xFFFF, 0xFFFF }, /* R210 */
{ 0x0000, 0x0000 }, /* R211 */
{ 0x0000, 0x0000 }, /* R212 */
{ 0x0000, 0x0000 }, /* R213 */
@@ -1573,7 +1573,7 @@
{ 0x03C3, 0x03C3 }, /* R1569 - Sidetone */
};
-const __devinitdata u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = {
+const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = {
0x8994, /* R0 - Software Reset */
0x0000, /* R1 - Power Management (1) */
0x6000, /* R2 - Power Management (2) */
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 0ca81d3..3dc64c8 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -102,11 +102,16 @@
wm8958_micdet_cb jack_cb;
void *jack_cb_data;
- bool jack_is_mic;
- bool jack_is_video;
+ int micdet_irq;
int revision;
struct wm8994_pdata *pdata;
+
+ unsigned int aif1clk_enable:1;
+ unsigned int aif2clk_enable:1;
+
+ unsigned int aif1clk_disable:1;
+ unsigned int aif2clk_disable:1;
};
static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg)
@@ -523,7 +528,7 @@
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
- struct wm8994_priv *wm8994 =snd_soc_codec_get_drvdata(codec);
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int block = wm8994_get_retune_mobile_block(kcontrol->id.name);
ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block];
@@ -1004,6 +1009,117 @@
}
}
+static int late_enable_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ if (wm8994->aif1clk_enable) {
+ snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
+ WM8994_AIF1CLK_ENA_MASK,
+ WM8994_AIF1CLK_ENA);
+ wm8994->aif1clk_enable = 0;
+ }
+ if (wm8994->aif2clk_enable) {
+ snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
+ WM8994_AIF2CLK_ENA_MASK,
+ WM8994_AIF2CLK_ENA);
+ wm8994->aif2clk_enable = 0;
+ }
+ break;
+ }
+
+ return 0;
+}
+
+static int late_disable_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMD:
+ if (wm8994->aif1clk_disable) {
+ snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
+ WM8994_AIF1CLK_ENA_MASK, 0);
+ wm8994->aif1clk_disable = 0;
+ }
+ if (wm8994->aif2clk_disable) {
+ snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
+ WM8994_AIF2CLK_ENA_MASK, 0);
+ wm8994->aif2clk_disable = 0;
+ }
+ break;
+ }
+
+ return 0;
+}
+
+static int aif1clk_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ wm8994->aif1clk_enable = 1;
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ wm8994->aif1clk_disable = 1;
+ break;
+ }
+
+ return 0;
+}
+
+static int aif2clk_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ wm8994->aif2clk_enable = 1;
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ wm8994->aif2clk_disable = 1;
+ break;
+ }
+
+ return 0;
+}
+
+static int adc_mux_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ late_enable_ev(w, kcontrol, event);
+ return 0;
+}
+
+static int micbias_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ late_enable_ev(w, kcontrol, event);
+ return 0;
+}
+
+static int dac_ev(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ unsigned int mask = 1 << w->shift;
+
+ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
+ mask, mask);
+ return 0;
+}
+
static const char *hp_mux_text[] = {
"Mixer",
"DAC",
@@ -1272,11 +1388,68 @@
static const struct snd_kcontrol_new aif2dacr_src_mux =
SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum);
+static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = {
+SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_ev,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_ev,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+ late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("Late DAC1R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+ late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("Late DAC2L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+ late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+ late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+
+SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev)
+};
+
+static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = {
+SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0)
+};
+
+static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = {
+SND_SOC_DAPM_DAC_E("DAC2L", NULL, SND_SOC_NOPM, 3, 0,
+ dac_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_DAC_E("DAC2R", NULL, SND_SOC_NOPM, 2, 0,
+ dac_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_DAC_E("DAC1L", NULL, SND_SOC_NOPM, 1, 0,
+ dac_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_DAC_E("DAC1R", NULL, SND_SOC_NOPM, 0, 0,
+ dac_ev, SND_SOC_DAPM_PRE_PMU),
+};
+
+static const struct snd_soc_dapm_widget wm8994_dac_widgets[] = {
+SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0),
+SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0),
+SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0),
+SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8994_adc_revd_widgets[] = {
+SND_SOC_DAPM_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux,
+ adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux,
+ adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+};
+
+static const struct snd_soc_dapm_widget wm8994_adc_widgets[] = {
+SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
+SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
+};
+
static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("DMIC1DAT"),
SND_SOC_DAPM_INPUT("DMIC2DAT"),
SND_SOC_DAPM_INPUT("Clock"),
+SND_SOC_DAPM_MICBIAS("MICBIAS", WM8994_MICBIAS, 2, 0),
+SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev,
+ SND_SOC_DAPM_PRE_PMU),
+
SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@@ -1284,12 +1457,9 @@
SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0),
-SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
-SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0),
-
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL,
0, WM8994_POWER_MANAGEMENT_4, 9, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL,
0, WM8994_POWER_MANAGEMENT_4, 8, 0),
SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0,
WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev,
@@ -1298,9 +1468,9 @@
WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL,
0, WM8994_POWER_MANAGEMENT_4, 11, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL,
0, WM8994_POWER_MANAGEMENT_4, 10, 0),
SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0,
WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev,
@@ -1345,6 +1515,7 @@
SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux),
@@ -1368,14 +1539,6 @@
SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0),
SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0),
-SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
-SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
-
-SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0),
-SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0),
-SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0),
-SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0),
-
SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux),
SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux),
@@ -1515,14 +1678,12 @@
{ "AIF2ADC Mux", "AIF3DACDAT", "AIF3ADCDAT" },
/* DAC1 inputs */
- { "DAC1L", NULL, "DAC1L Mixer" },
{ "DAC1L Mixer", "AIF2 Switch", "AIF2DACL" },
{ "DAC1L Mixer", "AIF1.2 Switch", "AIF1DAC2L" },
{ "DAC1L Mixer", "AIF1.1 Switch", "AIF1DAC1L" },
{ "DAC1L Mixer", "Left Sidetone Switch", "Left Sidetone" },
{ "DAC1L Mixer", "Right Sidetone Switch", "Right Sidetone" },
- { "DAC1R", NULL, "DAC1R Mixer" },
{ "DAC1R Mixer", "AIF2 Switch", "AIF2DACR" },
{ "DAC1R Mixer", "AIF1.2 Switch", "AIF1DAC2R" },
{ "DAC1R Mixer", "AIF1.1 Switch", "AIF1DAC1R" },
@@ -1531,7 +1692,6 @@
/* DAC2/AIF2 outputs */
{ "AIF2ADCL", NULL, "AIF2DAC2L Mixer" },
- { "DAC2L", NULL, "AIF2DAC2L Mixer" },
{ "AIF2DAC2L Mixer", "AIF2 Switch", "AIF2DACL" },
{ "AIF2DAC2L Mixer", "AIF1.2 Switch", "AIF1DAC2L" },
{ "AIF2DAC2L Mixer", "AIF1.1 Switch", "AIF1DAC1L" },
@@ -1539,13 +1699,17 @@
{ "AIF2DAC2L Mixer", "Right Sidetone Switch", "Right Sidetone" },
{ "AIF2ADCR", NULL, "AIF2DAC2R Mixer" },
- { "DAC2R", NULL, "AIF2DAC2R Mixer" },
{ "AIF2DAC2R Mixer", "AIF2 Switch", "AIF2DACR" },
{ "AIF2DAC2R Mixer", "AIF1.2 Switch", "AIF1DAC2R" },
{ "AIF2DAC2R Mixer", "AIF1.1 Switch", "AIF1DAC1R" },
{ "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" },
{ "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" },
+ { "AIF1ADCDAT", NULL, "AIF1ADC1L" },
+ { "AIF1ADCDAT", NULL, "AIF1ADC1R" },
+ { "AIF1ADCDAT", NULL, "AIF1ADC2L" },
+ { "AIF1ADCDAT", NULL, "AIF1ADC2R" },
+
{ "AIF2ADCDAT", NULL, "AIF2ADC Mux" },
/* AIF3 output */
@@ -1578,6 +1742,33 @@
{ "Right Headphone Mux", "DAC", "DAC1R" },
};
+static const struct snd_soc_dapm_route wm8994_lateclk_revd_intercon[] = {
+ { "DAC1L", NULL, "Late DAC1L Enable PGA" },
+ { "Late DAC1L Enable PGA", NULL, "DAC1L Mixer" },
+ { "DAC1R", NULL, "Late DAC1R Enable PGA" },
+ { "Late DAC1R Enable PGA", NULL, "DAC1R Mixer" },
+ { "DAC2L", NULL, "Late DAC2L Enable PGA" },
+ { "Late DAC2L Enable PGA", NULL, "AIF2DAC2L Mixer" },
+ { "DAC2R", NULL, "Late DAC2R Enable PGA" },
+ { "Late DAC2R Enable PGA", NULL, "AIF2DAC2R Mixer" }
+};
+
+static const struct snd_soc_dapm_route wm8994_lateclk_intercon[] = {
+ { "DAC1L", NULL, "DAC1L Mixer" },
+ { "DAC1R", NULL, "DAC1R Mixer" },
+ { "DAC2L", NULL, "AIF2DAC2L Mixer" },
+ { "DAC2R", NULL, "AIF2DAC2R Mixer" },
+};
+
+static const struct snd_soc_dapm_route wm8994_revd_intercon[] = {
+ { "AIF1DACDAT", NULL, "AIF2DACDAT" },
+ { "AIF2DACDAT", NULL, "AIF1DACDAT" },
+ { "AIF1ADCDAT", NULL, "AIF2ADCDAT" },
+ { "AIF2ADCDAT", NULL, "AIF1ADCDAT" },
+ { "MICBIAS", NULL, "CLK_SYS" },
+ { "MICBIAS", NULL, "MICBIAS Supply" },
+};
+
static const struct snd_soc_dapm_route wm8994_intercon[] = {
{ "AIF2DACL", NULL, "AIF2DAC Mux" },
{ "AIF2DACR", NULL, "AIF2DAC Mux" },
@@ -2501,6 +2692,22 @@
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int i, ret;
+ unsigned int val, mask;
+
+ if (wm8994->revision < 4) {
+ /* force a HW read */
+ val = wm8994_reg_read(codec->control_data,
+ WM8994_POWER_MANAGEMENT_5);
+
+ /* modify the cache only */
+ codec->cache_only = 1;
+ mask = WM8994_DAC1R_ENA | WM8994_DAC1L_ENA |
+ WM8994_DAC2R_ENA | WM8994_DAC2L_ENA;
+ val &= mask;
+ snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
+ mask, val);
+ codec->cache_only = 0;
+ }
/* Restore the registers */
ret = snd_soc_cache_sync(codec);
@@ -2688,6 +2895,13 @@
else
snd_soc_add_controls(wm8994->codec, wm8994_eq_controls,
ARRAY_SIZE(wm8994_eq_controls));
+
+ for (i = 0; i < ARRAY_SIZE(pdata->micbias); i++) {
+ if (pdata->micbias[i]) {
+ snd_soc_write(codec, WM8958_MICBIAS1 + i,
+ pdata->micbias[i] & 0xffff);
+ }
+ }
}
/**
@@ -2798,47 +3012,18 @@
int report = 0;
/* If nothing present then clear our statuses */
- if (!(status & WM8958_MICD_STS)) {
- wm8994->jack_is_video = false;
- wm8994->jack_is_mic = false;
+ if (!(status & WM8958_MICD_STS))
goto done;
- }
- /* Assume anything over 475 ohms is a microphone and remember
- * that we've seen one (since buttons override it) */
- if (status & 0x600)
- wm8994->jack_is_mic = true;
- if (wm8994->jack_is_mic)
- report |= SND_JACK_MICROPHONE;
-
- /* Video has an impedence of approximately 75 ohms; assume
- * this isn't used as a button and remember it since buttons
- * override it. */
- if (status & 0x40)
- wm8994->jack_is_video = true;
- if (wm8994->jack_is_video)
- report |= SND_JACK_VIDEOOUT;
+ report = SND_JACK_MICROPHONE;
/* Everything else is buttons; just assign slots */
- if (status & 0x4)
+ if (status & 0x1c0)
report |= SND_JACK_BTN_0;
- if (status & 0x8)
- report |= SND_JACK_BTN_1;
- if (status & 0x10)
- report |= SND_JACK_BTN_2;
- if (status & 0x20)
- report |= SND_JACK_BTN_3;
- if (status & 0x80)
- report |= SND_JACK_BTN_4;
- if (status & 0x100)
- report |= SND_JACK_BTN_5;
done:
- snd_soc_jack_report(wm8994->micdet[0].jack,
- SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 |
- SND_JACK_BTN_3 | SND_JACK_BTN_4 | SND_JACK_BTN_5 |
- SND_JACK_MICROPHONE | SND_JACK_VIDEOOUT,
- report);
+ snd_soc_jack_report(wm8994->micdet[0].jack, report,
+ SND_JACK_BTN_0 | SND_JACK_MICROPHONE);
}
/**
@@ -2937,6 +3122,12 @@
wm8994->pdata = dev_get_platdata(codec->dev->parent);
wm8994->codec = codec;
+ if (wm8994->pdata && wm8994->pdata->micdet_irq)
+ wm8994->micdet_irq = wm8994->pdata->micdet_irq;
+ else if (wm8994->pdata && wm8994->pdata->irq_base)
+ wm8994->micdet_irq = wm8994->pdata->irq_base +
+ WM8994_IRQ_MIC1_DET;
+
pm_runtime_enable(codec->dev);
pm_runtime_resume(codec->dev);
@@ -2985,14 +3176,17 @@
switch (control->type) {
case WM8994:
- ret = wm8994_request_irq(codec->control_data,
- WM8994_IRQ_MIC1_DET,
- wm8994_mic_irq, "Mic 1 detect",
- wm8994);
- if (ret != 0)
- dev_warn(codec->dev,
- "Failed to request Mic1 detect IRQ: %d\n",
- ret);
+ if (wm8994->micdet_irq) {
+ ret = request_threaded_irq(wm8994->micdet_irq, NULL,
+ wm8994_mic_irq,
+ IRQF_TRIGGER_RISING,
+ "Mic1 detect",
+ wm8994);
+ if (ret != 0)
+ dev_warn(codec->dev,
+ "Failed to request Mic1 detect IRQ: %d\n",
+ ret);
+ }
ret = wm8994_request_irq(codec->control_data,
WM8994_IRQ_MIC1_SHRT,
@@ -3023,15 +3217,17 @@
break;
case WM8958:
- ret = wm8994_request_irq(codec->control_data,
- WM8994_IRQ_MIC1_DET,
- wm8958_mic_irq, "Mic detect",
- wm8994);
- if (ret != 0)
- dev_warn(codec->dev,
- "Failed to request Mic detect IRQ: %d\n",
- ret);
- break;
+ if (wm8994->micdet_irq) {
+ ret = request_threaded_irq(wm8994->micdet_irq, NULL,
+ wm8958_mic_irq,
+ IRQF_TRIGGER_RISING,
+ "Mic detect",
+ wm8994);
+ if (ret != 0)
+ dev_warn(codec->dev,
+ "Failed to request Mic detect IRQ: %d\n",
+ ret);
+ }
}
/* Remember if AIFnLRCLK is configured as a GPIO. This should be
@@ -3112,10 +3308,31 @@
case WM8994:
snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets,
ARRAY_SIZE(wm8994_specific_dapm_widgets));
+ if (wm8994->revision < 4) {
+ snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets,
+ ARRAY_SIZE(wm8994_lateclk_revd_widgets));
+ snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets,
+ ARRAY_SIZE(wm8994_adc_revd_widgets));
+ snd_soc_dapm_new_controls(dapm, wm8994_dac_revd_widgets,
+ ARRAY_SIZE(wm8994_dac_revd_widgets));
+ } else {
+ snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets,
+ ARRAY_SIZE(wm8994_lateclk_widgets));
+ snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets,
+ ARRAY_SIZE(wm8994_adc_widgets));
+ snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets,
+ ARRAY_SIZE(wm8994_dac_widgets));
+ }
break;
case WM8958:
snd_soc_add_controls(codec, wm8958_snd_controls,
ARRAY_SIZE(wm8958_snd_controls));
+ snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets,
+ ARRAY_SIZE(wm8994_lateclk_widgets));
+ snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets,
+ ARRAY_SIZE(wm8994_adc_widgets));
+ snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets,
+ ARRAY_SIZE(wm8994_dac_widgets));
snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets,
ARRAY_SIZE(wm8958_dapm_widgets));
break;
@@ -3129,8 +3346,20 @@
case WM8994:
snd_soc_dapm_add_routes(dapm, wm8994_intercon,
ARRAY_SIZE(wm8994_intercon));
+
+ if (wm8994->revision < 4) {
+ snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
+ ARRAY_SIZE(wm8994_revd_intercon));
+ snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon,
+ ARRAY_SIZE(wm8994_lateclk_revd_intercon));
+ } else {
+ snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon,
+ ARRAY_SIZE(wm8994_lateclk_intercon));
+ }
break;
case WM8958:
+ snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon,
+ ARRAY_SIZE(wm8994_lateclk_intercon));
snd_soc_dapm_add_routes(dapm, wm8958_intercon,
ARRAY_SIZE(wm8958_intercon));
break;
@@ -3142,7 +3371,8 @@
wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT, wm8994);
wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994);
wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994);
- wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, wm8994);
+ if (wm8994->micdet_irq)
+ free_irq(wm8994->micdet_irq, wm8994);
err:
kfree(wm8994);
return ret;
@@ -3159,8 +3389,8 @@
switch (control->type) {
case WM8994:
- wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT,
- wm8994);
+ if (wm8994->micdet_irq)
+ free_irq(wm8994->micdet_irq, wm8994);
wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET,
wm8994);
wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT,
@@ -3170,8 +3400,8 @@
break;
case WM8958:
- wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET,
- wm8994);
+ if (wm8994->micdet_irq)
+ free_irq(wm8994->micdet_irq, wm8994);
break;
}
kfree(wm8994->retune_mobile_texts);
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index 0c355bf..999b885 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -43,6 +43,6 @@
};
extern const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE];
-extern const __devinitdata u16 wm8994_reg_defaults[WM8994_CACHE_SIZE];
+extern const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE];
#endif
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 5c224dd..55cdf29 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -15,6 +15,7 @@
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
+#include <linux/device.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
@@ -166,7 +167,7 @@
int fll_fref;
int fll_fout;
int tdm_width;
- struct wm9081_retune_mobile_config *retune;
+ struct wm9081_pdata pdata;
};
static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int reg)
@@ -388,27 +389,6 @@
SOC_DAPM_SINGLE("Playback Switch", WM9081_ANALOGUE_MIXER, 4, 1, 0),
};
-static int speaker_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- struct snd_soc_codec *codec = w->codec;
- unsigned int reg = snd_soc_read(codec, WM9081_POWER_MANAGEMENT);
-
- switch (event) {
- case SND_SOC_DAPM_POST_PMU:
- reg |= WM9081_SPK_ENA;
- break;
-
- case SND_SOC_DAPM_PRE_PMD:
- reg &= ~WM9081_SPK_ENA;
- break;
- }
-
- snd_soc_write(codec, WM9081_POWER_MANAGEMENT, reg);
-
- return 0;
-}
-
struct _fll_div {
u16 fll_fratio;
u16 fll_outdiv;
@@ -746,9 +726,8 @@
SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0),
-SND_SOC_DAPM_PGA_E("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0,
- speaker_event,
- SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("LINEOUT"),
SND_SOC_DAPM_OUTPUT("SPKN"),
@@ -761,7 +740,7 @@
};
-static const struct snd_soc_dapm_route audio_paths[] = {
+static const struct snd_soc_dapm_route wm9081_audio_paths[] = {
{ "DAC", NULL, "CLK_SYS" },
{ "DAC", NULL, "CLK_DSP" },
@@ -779,8 +758,10 @@
{ "Speaker PGA", NULL, "TOCLK" },
{ "Speaker PGA", NULL, "CLK_SYS" },
- { "SPKN", NULL, "Speaker PGA" },
- { "SPKP", NULL, "Speaker PGA" },
+ { "Speaker", NULL, "Speaker PGA" },
+
+ { "SPKN", NULL, "Speaker" },
+ { "SPKP", NULL, "Speaker" },
};
static int wm9081_set_bias_level(struct snd_soc_codec *codec,
@@ -1081,21 +1062,22 @@
aif4 |= wm9081->bclk / wm9081->fs;
/* Apply a ReTune Mobile configuration if it's in use */
- if (wm9081->retune) {
- struct wm9081_retune_mobile_config *retune = wm9081->retune;
+ if (wm9081->pdata.num_retune_configs) {
+ struct wm9081_pdata *pdata = &wm9081->pdata;
struct wm9081_retune_mobile_setting *s;
int eq1;
best = 0;
- best_val = abs(retune->configs[0].rate - wm9081->fs);
- for (i = 0; i < retune->num_configs; i++) {
- cur_val = abs(retune->configs[i].rate - wm9081->fs);
+ best_val = abs(pdata->retune_configs[0].rate - wm9081->fs);
+ for (i = 0; i < pdata->num_retune_configs; i++) {
+ cur_val = abs(pdata->retune_configs[i].rate -
+ wm9081->fs);
if (cur_val < best_val) {
best_val = cur_val;
best = i;
}
}
- s = &retune->configs[best];
+ s = &pdata->retune_configs[best];
dev_dbg(codec->dev, "ReTune Mobile %s tuned for %dHz\n",
s->name, s->rate);
@@ -1138,10 +1120,9 @@
return 0;
}
-static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai,
+static int wm9081_set_sysclk(struct snd_soc_codec *codec,
int clk_id, unsigned int freq, int dir)
{
- struct snd_soc_codec *codec = codec_dai->codec;
struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
switch (clk_id) {
@@ -1206,7 +1187,6 @@
static struct snd_soc_dai_ops wm9081_dai_ops = {
.hw_params = wm9081_hw_params,
- .set_sysclk = wm9081_set_sysclk,
.set_fmt = wm9081_set_dai_fmt,
.digital_mute = wm9081_digital_mute,
.set_tdm_slot = wm9081_set_tdm_slot,
@@ -1230,7 +1210,6 @@
static int wm9081_probe(struct snd_soc_codec *codec)
{
struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
- struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret;
u16 reg;
@@ -1254,6 +1233,14 @@
return ret;
}
+ reg = 0;
+ if (wm9081->pdata.irq_high)
+ reg |= WM9081_IRQ_POL;
+ if (!wm9081->pdata.irq_cmos)
+ reg |= WM9081_IRQ_OP_CTRL;
+ snd_soc_update_bits(codec, WM9081_INTERRUPT_CONTROL,
+ WM9081_IRQ_POL | WM9081_IRQ_OP_CTRL, reg);
+
wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
/* Enable zero cross by default */
@@ -1265,17 +1252,13 @@
snd_soc_add_controls(codec, wm9081_snd_controls,
ARRAY_SIZE(wm9081_snd_controls));
- if (!wm9081->retune) {
+ if (!wm9081->pdata.num_retune_configs) {
dev_dbg(codec->dev,
"No ReTune Mobile data, using normal EQ\n");
snd_soc_add_controls(codec, wm9081_eq_controls,
ARRAY_SIZE(wm9081_eq_controls));
}
- snd_soc_dapm_new_controls(dapm, wm9081_dapm_widgets,
- ARRAY_SIZE(wm9081_dapm_widgets));
- snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
-
return ret;
}
@@ -1319,11 +1302,19 @@
.remove = wm9081_remove,
.suspend = wm9081_suspend,
.resume = wm9081_resume,
+
+ .set_sysclk = wm9081_set_sysclk,
.set_bias_level = wm9081_set_bias_level,
+
.reg_cache_size = ARRAY_SIZE(wm9081_reg_defaults),
.reg_word_size = sizeof(u16),
.reg_cache_default = wm9081_reg_defaults,
.volatile_register = wm9081_volatile_register,
+
+ .dapm_widgets = wm9081_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(wm9081_dapm_widgets),
+ .dapm_routes = wm9081_audio_paths,
+ .num_dapm_routes = ARRAY_SIZE(wm9081_audio_paths),
};
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
@@ -1341,6 +1332,10 @@
wm9081->control_type = SND_SOC_I2C;
wm9081->control_data = i2c;
+ if (dev_get_platdata(&i2c->dev))
+ memcpy(&wm9081->pdata, dev_get_platdata(&i2c->dev),
+ sizeof(wm9081->pdata));
+
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_wm9081, &wm9081_dai, 1);
if (ret < 0)
@@ -1363,7 +1358,7 @@
static struct i2c_driver wm9081_i2c_driver = {
.driver = {
- .name = "wm9081-codec",
+ .name = "wm9081",
.owner = THIS_MODULE,
},
.probe = wm9081_i2c_probe,
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 613df5d..7b6b3c1 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -82,7 +82,8 @@
} while (reg & op && count < 400);
if (reg & op)
- dev_err(codec->dev, "Timed out waiting for DC Servo\n");
+ dev_err(codec->dev, "Timed out waiting for DC Servo %x\n",
+ op);
}
/*
@@ -674,6 +675,9 @@
};
static const struct snd_soc_dapm_route analogue_routes[] = {
+ { "MICBIAS1", NULL, "CLK_SYS" },
+ { "MICBIAS2", NULL, "CLK_SYS" },
+
{ "IN1L PGA", "IN1LP Switch", "IN1LP" },
{ "IN1L PGA", "IN1LN Switch", "IN1LN" },
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index b36f0b3..fe79842 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -218,7 +218,19 @@
.ops = &evm_spdif_ops,
},
};
-static struct snd_soc_dai_link da8xx_evm_dai = {
+
+static struct snd_soc_dai_link da830_evm_dai = {
+ .name = "TLV320AIC3X",
+ .stream_name = "AIC3X",
+ .cpu_dai_name = "davinci-mcasp.1",
+ .codec_dai_name = "tlv320aic3x-hifi",
+ .codec_name = "tlv320aic3x-codec.1-0018",
+ .platform_name = "davinci-pcm-audio",
+ .init = evm_aic3x_init,
+ .ops = &evm_ops,
+};
+
+static struct snd_soc_dai_link da850_evm_dai = {
.name = "TLV320AIC3X",
.stream_name = "AIC3X",
.cpu_dai_name= "davinci-mcasp.0",
@@ -259,13 +271,13 @@
static struct snd_soc_card da830_snd_soc_card = {
.name = "DA830/OMAP-L137 EVM",
- .dai_link = &da8xx_evm_dai,
+ .dai_link = &da830_evm_dai,
.num_links = 1,
};
static struct snd_soc_card da850_snd_soc_card = {
.name = "DA850/OMAP-L138 EVM",
- .dai_link = &da8xx_evm_dai,
+ .dai_link = &da850_evm_dai,
.num_links = 1,
};
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 9e0e565..d0d60b8 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -658,7 +658,7 @@
return -ENODEV;
}
- ioarea = request_mem_region(mem->start, (mem->end - mem->start) + 1,
+ ioarea = request_mem_region(mem->start, resource_size(mem),
pdev->name);
if (!ioarea) {
dev_err(&pdev->dev, "McBSP region already claimed\n");
@@ -694,20 +694,25 @@
}
clk_enable(dev->clk);
- dev->base = (void __iomem *)IO_ADDRESS(mem->start);
+ dev->base = ioremap(mem->start, resource_size(mem));
+ if (!dev->base) {
+ dev_err(&pdev->dev, "ioremap failed\n");
+ ret = -ENOMEM;
+ goto err_release_clk;
+ }
dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr =
- (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG);
+ (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG);
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr =
- (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG);
+ (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG);
/* first TX, then RX */
res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
if (!res) {
dev_err(&pdev->dev, "no DMA resource\n");
ret = -ENXIO;
- goto err_free_mem;
+ goto err_iounmap;
}
dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start;
@@ -715,7 +720,7 @@
if (!res) {
dev_err(&pdev->dev, "no DMA resource\n");
ret = -ENXIO;
- goto err_free_mem;
+ goto err_iounmap;
}
dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
dev->dev = &pdev->dev;
@@ -724,14 +729,19 @@
ret = snd_soc_register_dai(&pdev->dev, &davinci_i2s_dai);
if (ret != 0)
- goto err_free_mem;
+ goto err_iounmap;
return 0;
+err_iounmap:
+ iounmap(dev->base);
+err_release_clk:
+ clk_disable(dev->clk);
+ clk_put(dev->clk);
err_free_mem:
kfree(dev);
err_release_region:
- release_mem_region(mem->start, (mem->end - mem->start) + 1);
+ release_mem_region(mem->start, resource_size(mem));
return ret;
}
@@ -747,7 +757,7 @@
dev->clk = NULL;
kfree(dev);
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- release_mem_region(mem->start, (mem->end - mem->start) + 1);
+ release_mem_region(mem->start, resource_size(mem));
return 0;
}
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index fb55d2c..a5af834 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -868,7 +868,7 @@
}
ioarea = request_mem_region(mem->start,
- (mem->end - mem->start) + 1, pdev->name);
+ resource_size(mem), pdev->name);
if (!ioarea) {
dev_err(&pdev->dev, "Audio region already claimed\n");
ret = -EBUSY;
@@ -885,7 +885,13 @@
clk_enable(dev->clk);
dev->clk_active = 1;
- dev->base = (void __iomem *)IO_ADDRESS(mem->start);
+ dev->base = ioremap(mem->start, resource_size(mem));
+ if (!dev->base) {
+ dev_err(&pdev->dev, "ioremap failed\n");
+ ret = -ENOMEM;
+ goto err_release_clk;
+ }
+
dev->op_mode = pdata->op_mode;
dev->tdm_slots = pdata->tdm_slots;
dev->num_serializer = pdata->num_serializer;
@@ -899,14 +905,14 @@
dma_data->asp_chan_q = pdata->asp_chan_q;
dma_data->ram_chan_q = pdata->ram_chan_q;
dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
- io_v2p(dev->base));
+ mem->start);
/* first TX, then RX */
res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
if (!res) {
dev_err(&pdev->dev, "no DMA resource\n");
ret = -ENODEV;
- goto err_release_region;
+ goto err_iounmap;
}
dma_data->channel = res->start;
@@ -915,13 +921,13 @@
dma_data->asp_chan_q = pdata->asp_chan_q;
dma_data->ram_chan_q = pdata->ram_chan_q;
dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
- io_v2p(dev->base));
+ mem->start);
res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
if (!res) {
dev_err(&pdev->dev, "no DMA resource\n");
ret = -ENODEV;
- goto err_release_region;
+ goto err_iounmap;
}
dma_data->channel = res->start;
@@ -929,11 +935,16 @@
ret = snd_soc_register_dai(&pdev->dev, &davinci_mcasp_dai[pdata->op_mode]);
if (ret != 0)
- goto err_release_region;
+ goto err_iounmap;
return 0;
+err_iounmap:
+ iounmap(dev->base);
+err_release_clk:
+ clk_disable(dev->clk);
+ clk_put(dev->clk);
err_release_region:
- release_mem_region(mem->start, (mem->end - mem->start) + 1);
+ release_mem_region(mem->start, resource_size(mem));
err_release_data:
kfree(dev);
@@ -951,7 +962,7 @@
dev->clk = NULL;
mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
- release_mem_region(mem->start, (mem->end - mem->start) + 1);
+ release_mem_region(mem->start, resource_size(mem));
kfree(dev);
diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c
index b270085..d3aa151 100644
--- a/sound/soc/ep93xx/edb93xx.c
+++ b/sound/soc/ep93xx/edb93xx.c
@@ -41,17 +41,17 @@
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int err;
+ unsigned int mclk_rate;
unsigned int rate = params_rate(params);
+
/*
- * We set LRCLK equal to `rate' and SCLK = LRCLK * 64,
- * because our sample size is 32 bit * 2 channels.
- * I2S standard permits us to transmit more bits than
- * the codec uses.
- * MCLK = SCLK * 4 is the best recommended value,
- * but we have to fall back to ratio 2 for higher
- * sample rates.
+ * According to CS4271 datasheet we use MCLK/LRCK=256 for
+ * rates below 50kHz and 128 for higher sample rates
*/
- unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2);
+ if (rate < 50000)
+ mclk_rate = rate * 64 * 4;
+ else
+ mclk_rate = rate * 64 * 2;
err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_IF |
diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c
index 68a0bae..104e95c 100644
--- a/sound/soc/ep93xx/ep93xx-ac97.c
+++ b/sound/soc/ep93xx/ep93xx-ac97.c
@@ -253,7 +253,6 @@
struct ep93xx_ac97_info *info = snd_soc_dai_get_drvdata(dai);
unsigned v = 0;
-
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index fff579a..042f4e9 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -242,7 +242,7 @@
{
struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
unsigned word_len, div, sdiv, lrdiv;
- int found = 0, err;
+ int err;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
@@ -275,15 +275,14 @@
* the codec uses.
*/
div = clk_get_rate(info->mclk) / params_rate(params);
- for (sdiv = 2; sdiv <= 4; sdiv += 2)
- for (lrdiv = 64; lrdiv <= 128; lrdiv <<= 1)
- if (sdiv * lrdiv == div) {
- found = 1;
- goto out;
- }
-out:
- if (!found)
- return -EINVAL;
+ sdiv = 4;
+ if (div > (256 + 512) / 2) {
+ lrdiv = 128;
+ } else {
+ lrdiv = 64;
+ if (div < (128 + 256) / 2)
+ sdiv = 2;
+ }
err = clk_set_rate(info->sclk, clk_get_rate(info->mclk) / sdiv);
if (err)
@@ -314,10 +313,12 @@
struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
if (!dai->active)
- return;
+ return 0;
ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK);
ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE);
+
+ return 0;
}
static int ep93xx_i2s_resume(struct snd_soc_dai *dai)
@@ -325,10 +326,12 @@
struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
if (!dai->active)
- return;
+ return 0;
ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK);
ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE);
+
+ return 0;
}
#else
#define ep93xx_i2s_suspend NULL
@@ -352,13 +355,13 @@
.playback = {
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_96000,
+ .rates = SNDRV_PCM_RATE_8000_192000,
.formats = EP93XX_I2S_FORMATS,
},
.capture = {
.channels_min = 2,
.channels_max = 2,
- .rates = SNDRV_PCM_RATE_8000_96000,
+ .rates = SNDRV_PCM_RATE_8000_192000,
.formats = EP93XX_I2S_FORMATS,
},
.ops = &ep93xx_i2s_dai_ops,
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index 0667077..a456e49 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -35,9 +35,9 @@
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER),
- .rates = SNDRV_PCM_RATE_8000_96000,
+ .rates = SNDRV_PCM_RATE_8000_192000,
.rate_min = SNDRV_PCM_RATE_8000,
- .rate_max = SNDRV_PCM_RATE_96000,
+ .rate_max = SNDRV_PCM_RATE_192000,
.formats = (SNDRV_PCM_FMTBIT_S16_LE |
SNDRV_PCM_FMTBIT_S24_LE |
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index 642270a..d8f130d 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -30,6 +30,16 @@
Enable support for audio on the i.MX31ADS with the WM1133-EV1
PMIC board with WM8835x fitted.
+config SND_SOC_MX27VIS_AIC32X4
+ tristate "SoC audio support for Visstrim M10 boards"
+ depends on MACH_IMX27_VISSTRIM_M10
+ select SND_SOC_TVL320AIC32X4
+ select SND_MXC_SOC_SSI
+ select SND_MXC_SOC_MX2
+ help
+ Say Y if you want to add support for SoC audio on Visstrim SM10
+ board with TLV320AIC32X4 codec.
+
config SND_SOC_PHYCORE_AC97
tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
depends on MACH_PCM043 || MACH_PCA100
@@ -44,7 +54,8 @@
tristate "Eukrea TLV320"
depends on MACH_EUKREA_MBIMX27_BASEBOARD \
|| MACH_EUKREA_MBIMXSD25_BASEBOARD \
- || MACH_EUKREA_MBIMXSD35_BASEBOARD
+ || MACH_EUKREA_MBIMXSD35_BASEBOARD \
+ || MACH_EUKREA_MBIMXSD51_BASEBOARD
select SND_SOC_TLV320AIC23
select SND_MXC_SOC_SSI
select SND_MXC_SOC_FIQ
diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile
index b67fc02..d6d609b 100644
--- a/sound/soc/imx/Makefile
+++ b/sound/soc/imx/Makefile
@@ -10,8 +10,10 @@
# i.MX Machine Support
snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
snd-soc-phycore-ac97-objs := phycore-ac97.o
+snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
snd-soc-wm1133-ev1-objs := wm1133-ev1.o
obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
+obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c
index e20c9e1..75fb4b8 100644
--- a/sound/soc/imx/eukrea-tlv320.c
+++ b/sound/soc/imx/eukrea-tlv320.c
@@ -79,7 +79,7 @@
.name = "tlv320aic23",
.stream_name = "TLV320AIC23",
.codec_dai_name = "tlv320aic23-hifi",
- .platform_name = "imx-pcm-audio.0",
+ .platform_name = "imx-fiq-pcm-audio.0",
.codec_name = "tlv320aic23-codec.0-001a",
.cpu_dai_name = "imx-ssi.0",
.ops = &eukrea_tlv320_snd_ops,
@@ -98,7 +98,8 @@
int ret;
if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd()
- && !machine_is_eukrea_cpuimx35sd())
+ && !machine_is_eukrea_cpuimx35sd()
+ && !machine_is_eukrea_cpuimx51sd())
/* return happy. We might run on a totally different machine */
return 0;
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 30894ea..bc92ec6 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -108,7 +108,7 @@
break;
case SND_SOC_DAIFMT_DSP_B:
/* data on rising edge of bclk, frame high with data */
- strcr |= SSI_STCR_TFSL;
+ strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0;
break;
case SND_SOC_DAIFMT_DSP_A:
/* data on rising edge of bclk, frame high 1clk before data */
@@ -656,6 +656,9 @@
ssi->dma_params_rx.dma_addr = res->start + SSI_SRX0;
ssi->dma_params_tx.dma_addr = res->start + SSI_STX0;
+ ssi->dma_params_tx.burstsize = 4;
+ ssi->dma_params_rx.burstsize = 4;
+
res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0");
if (res)
ssi->dma_params_tx.dma = res->start;
diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c
new file mode 100644
index 0000000..054110b
--- /dev/null
+++ b/sound/soc/imx/mx27vis-aic32x4.c
@@ -0,0 +1,137 @@
+/*
+ * mx27vis-aic32x4.c
+ *
+ * Copyright 2011 Vista Silicon S.L.
+ *
+ * Author: Javier Martin <javier.martin@vista-silicon.com>
+ *
+ * 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.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
+ * MA 02110-1301, USA.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <asm/mach-types.h>
+#include <mach/audmux.h>
+
+#include "../codecs/tlv320aic32x4.h"
+#include "imx-ssi.h"
+
+static int mx27vis_aic32x4_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;
+ u32 dai_format;
+
+ dai_format = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+
+ /* set codec DAI configuration */
+ snd_soc_dai_set_fmt(codec_dai, dai_format);
+
+ /* set cpu DAI configuration */
+ snd_soc_dai_set_fmt(cpu_dai, dai_format);
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, 0,
+ 25000000, SND_SOC_CLOCK_OUT);
+ if (ret) {
+ pr_err("%s: failed setting codec sysclk\n", __func__);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
+ SND_SOC_CLOCK_IN);
+ if (ret) {
+ pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops mx27vis_aic32x4_snd_ops = {
+ .hw_params = mx27vis_aic32x4_hw_params,
+};
+
+static struct snd_soc_dai_link mx27vis_aic32x4_dai = {
+ .name = "tlv320aic32x4",
+ .stream_name = "TLV320AIC32X4",
+ .codec_dai_name = "tlv320aic32x4-hifi",
+ .platform_name = "imx-pcm-audio.0",
+ .codec_name = "tlv320aic32x4.0-0018",
+ .cpu_dai_name = "imx-ssi.0",
+ .ops = &mx27vis_aic32x4_snd_ops,
+};
+
+static struct snd_soc_card mx27vis_aic32x4 = {
+ .name = "visstrim_m10-audio",
+ .dai_link = &mx27vis_aic32x4_dai,
+ .num_links = 1,
+};
+
+static struct platform_device *mx27vis_aic32x4_snd_device;
+
+static int __init mx27vis_aic32x4_init(void)
+{
+ int ret;
+
+ mx27vis_aic32x4_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!mx27vis_aic32x4_snd_device)
+ return -ENOMEM;
+
+ platform_set_drvdata(mx27vis_aic32x4_snd_device, &mx27vis_aic32x4);
+ ret = platform_device_add(mx27vis_aic32x4_snd_device);
+
+ if (ret) {
+ printk(KERN_ERR "ASoC: Platform device allocation failed\n");
+ platform_device_put(mx27vis_aic32x4_snd_device);
+ }
+
+ /* Connect SSI0 as clock slave to SSI1 external pins */
+ mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
+ MXC_AUDMUX_V1_PCR_SYN |
+ MXC_AUDMUX_V1_PCR_TFSDIR |
+ MXC_AUDMUX_V1_PCR_TCLKDIR |
+ MXC_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) |
+ MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1)
+ );
+ mxc_audmux_v1_configure_port(MX27_AUDMUX_PPCR1_SSI_PINS_1,
+ MXC_AUDMUX_V1_PCR_SYN |
+ MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0)
+ );
+
+ return ret;
+}
+
+static void __exit mx27vis_aic32x4_exit(void)
+{
+ platform_device_unregister(mx27vis_aic32x4_snd_device);
+}
+
+module_init(mx27vis_aic32x4_init);
+module_exit(mx27vis_aic32x4_exit);
+
+MODULE_AUTHOR("Javier Martin <javier.martin@vista-silicon.com>");
+MODULE_DESCRIPTION("ALSA SoC AIC32X4 mx27 visstrim");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/mid-x86/Kconfig
index 1ad7538..2935042 100644
--- a/sound/soc/mid-x86/Kconfig
+++ b/sound/soc/mid-x86/Kconfig
@@ -1,6 +1,7 @@
config SND_MFLD_MACHINE
tristate "SOC Machine Audio driver for Intel Medfield MID platform"
depends on INTEL_SCU_IPC
+ depends on SND_INTEL_SST
select SND_SOC_SN95031
select SND_SST_PLATFORM
help
@@ -11,4 +12,3 @@
config SND_SST_PLATFORM
tristate
- depends on SND_INTEL_SST
diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c
index 7925851..429aa1b 100644
--- a/sound/soc/mid-x86/mfld_machine.c
+++ b/sound/soc/mid-x86/mfld_machine.c
@@ -27,18 +27,59 @@
#include <linux/init.h>
#include <linux/device.h>
#include <linux/slab.h>
+#include <linux/io.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include <sound/jack.h>
#include "../codecs/sn95031.h"
#define MID_MONO 1
#define MID_STEREO 2
#define MID_MAX_CAP 5
+#define MFLD_JACK_INSERT 0x04
+
+enum soc_mic_bias_zones {
+ MFLD_MV_START = 0,
+ /* mic bias volutage range for Headphones*/
+ MFLD_MV_HP = 400,
+ /* mic bias volutage range for American Headset*/
+ MFLD_MV_AM_HS = 650,
+ /* mic bias volutage range for Headset*/
+ MFLD_MV_HS = 2000,
+ MFLD_MV_UNDEFINED,
+};
static unsigned int hs_switch;
static unsigned int lo_dac;
+struct mfld_mc_private {
+ struct platform_device *socdev;
+ void __iomem *int_base;
+ struct snd_soc_codec *codec;
+ u8 interrupt_status;
+};
+
+struct snd_soc_jack mfld_jack;
+
+/*Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin mfld_jack_pins[] = {
+ {
+ .pin = "Headphones",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ {
+ .pin = "AMIC1",
+ .mask = SND_JACK_MICROPHONE,
+ },
+};
+
+/* jack detection voltage zones */
+static struct snd_soc_jack_zone mfld_zones[] = {
+ {MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE},
+ {MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET},
+};
+
/* sound card controls */
static const char *headset_switch_text[] = {"Earpiece", "Headset"};
@@ -67,13 +108,11 @@
if (ucontrol->value.integer.value[0]) {
pr_debug("hs_set HS path\n");
- snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL");
- snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR");
+ snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
} else {
pr_debug("hs_set EP path\n");
- snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL");
- snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR");
+ snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT");
}
snd_soc_dapm_sync(&codec->dapm);
@@ -91,12 +130,10 @@
snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT");
snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT");
if (hs_switch) {
- snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL");
- snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR");
+ snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
} else {
- snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL");
- snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR");
+ snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT");
}
}
@@ -130,8 +167,7 @@
case 1:
pr_debug("set hs path\n");
- snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL");
- snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR");
+ snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22);
break;
@@ -162,12 +198,45 @@
lo_get_switch, lo_set_switch),
};
+static const struct snd_soc_dapm_widget mfld_widgets[] = {
+ SND_SOC_DAPM_HP("Headphones", NULL),
+ SND_SOC_DAPM_MIC("Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route mfld_map[] = {
+ {"Headphones", NULL, "HPOUTR"},
+ {"Headphones", NULL, "HPOUTL"},
+ {"Mic", NULL, "AMIC1"},
+};
+
+static void mfld_jack_check(unsigned int intr_status)
+{
+ struct mfld_jack_data jack_data;
+
+ jack_data.mfld_jack = &mfld_jack;
+ jack_data.intr_id = intr_status;
+
+ sn95031_jack_detection(&jack_data);
+ /* TODO: add american headset detection post gpiolib support */
+}
+
static int mfld_init(struct snd_soc_pcm_runtime *runtime)
{
struct snd_soc_codec *codec = runtime->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
int ret_val;
+ /* Add jack sense widgets */
+ snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets));
+
+ /* Set up the map */
+ snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map));
+
+ /* always connected */
+ snd_soc_dapm_enable_pin(dapm, "Headphones");
+ snd_soc_dapm_enable_pin(dapm, "Mic");
+ snd_soc_dapm_sync(dapm);
+
ret_val = snd_soc_add_controls(codec, mfld_snd_controls,
ARRAY_SIZE(mfld_snd_controls));
if (ret_val) {
@@ -175,8 +244,7 @@
return ret_val;
}
/* default is earpiece pin, userspace sets it explcitly */
- snd_soc_dapm_disable_pin(dapm, "HPOUTL");
- snd_soc_dapm_disable_pin(dapm, "HPOUTR");
+ snd_soc_dapm_disable_pin(dapm, "Headphones");
/* default is lineout NC, userspace sets it explcitly */
snd_soc_dapm_disable_pin(dapm, "LINEOUTL");
snd_soc_dapm_disable_pin(dapm, "LINEOUTR");
@@ -185,7 +253,35 @@
/* we dont use linein in this so set to NC */
snd_soc_dapm_disable_pin(dapm, "LINEINL");
snd_soc_dapm_disable_pin(dapm, "LINEINR");
- return snd_soc_dapm_sync(dapm);
+ snd_soc_dapm_sync(dapm);
+
+ /* Headset and button jack detection */
+ ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack",
+ SND_JACK_HEADSET | SND_JACK_BTN_0 |
+ SND_JACK_BTN_1, &mfld_jack);
+ if (ret_val) {
+ pr_err("jack creation failed\n");
+ return ret_val;
+ }
+
+ ret_val = snd_soc_jack_add_pins(&mfld_jack,
+ ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins);
+ if (ret_val) {
+ pr_err("adding jack pins failed\n");
+ return ret_val;
+ }
+ ret_val = snd_soc_jack_add_zones(&mfld_jack,
+ ARRAY_SIZE(mfld_zones), mfld_zones);
+ if (ret_val) {
+ pr_err("adding jack zones failed\n");
+ return ret_val;
+ }
+
+ /* we want to check if anything is inserted at boot,
+ * so send a fake event to codec and it will read adc
+ * to find if anything is there or not */
+ mfld_jack_check(MFLD_JACK_INSERT);
+ return ret_val;
}
struct snd_soc_dai_link mfld_msic_dailink[] = {
@@ -234,37 +330,94 @@
.num_links = ARRAY_SIZE(mfld_msic_dailink),
};
+static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev)
+{
+ struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev;
+
+ memcpy_fromio(&mc_private->interrupt_status,
+ ((void *)(mc_private->int_base)),
+ sizeof(u8));
+ return IRQ_WAKE_THREAD;
+}
+
+static irqreturn_t snd_mfld_jack_detection(int irq, void *data)
+{
+ struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data;
+
+ if (mfld_jack.codec == NULL)
+ return IRQ_HANDLED;
+ mfld_jack_check(mc_drv_ctx->interrupt_status);
+
+ return IRQ_HANDLED;
+}
+
static int __devinit snd_mfld_mc_probe(struct platform_device *pdev)
{
- struct platform_device *socdev;
- int ret_val = 0;
+ int ret_val = 0, irq;
+ struct mfld_mc_private *mc_drv_ctx;
+ struct resource *irq_mem;
pr_debug("snd_mfld_mc_probe called\n");
- socdev = platform_device_alloc("soc-audio", -1);
- if (!socdev) {
- pr_err("soc-audio device allocation failed\n");
+ /* retrive the irq number */
+ irq = platform_get_irq(pdev, 0);
+
+ /* audio interrupt base of SRAM location where
+ * interrupts are stored by System FW */
+ mc_drv_ctx = kzalloc(sizeof(*mc_drv_ctx), GFP_ATOMIC);
+ if (!mc_drv_ctx) {
+ pr_err("allocation failed\n");
return -ENOMEM;
}
- platform_set_drvdata(socdev, &snd_soc_card_mfld);
- ret_val = platform_device_add(socdev);
- if (ret_val) {
- pr_err("Unable to add soc-audio device, err %d\n", ret_val);
- platform_device_put(socdev);
+
+ irq_mem = platform_get_resource_byname(
+ pdev, IORESOURCE_MEM, "IRQ_BASE");
+ if (!irq_mem) {
+ pr_err("no mem resource given\n");
+ ret_val = -ENODEV;
+ goto unalloc;
}
-
- platform_set_drvdata(pdev, socdev);
-
+ mc_drv_ctx->int_base = ioremap_nocache(irq_mem->start,
+ resource_size(irq_mem));
+ if (!mc_drv_ctx->int_base) {
+ pr_err("Mapping of cache failed\n");
+ ret_val = -ENOMEM;
+ goto unalloc;
+ }
+ /* register for interrupt */
+ ret_val = request_threaded_irq(irq, snd_mfld_jack_intr_handler,
+ snd_mfld_jack_detection,
+ IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx);
+ if (ret_val) {
+ pr_err("cannot register IRQ\n");
+ goto unalloc;
+ }
+ /* register the soc card */
+ snd_soc_card_mfld.dev = &pdev->dev;
+ ret_val = snd_soc_register_card(&snd_soc_card_mfld);
+ if (ret_val) {
+ pr_debug("snd_soc_register_card failed %d\n", ret_val);
+ goto freeirq;
+ }
+ platform_set_drvdata(pdev, mc_drv_ctx);
pr_debug("successfully exited probe\n");
return ret_val;
+
+freeirq:
+ free_irq(irq, mc_drv_ctx);
+unalloc:
+ kfree(mc_drv_ctx);
+ return ret_val;
}
static int __devexit snd_mfld_mc_remove(struct platform_device *pdev)
{
- struct platform_device *socdev = platform_get_drvdata(pdev);
- pr_debug("snd_mfld_mc_remove called\n");
+ struct mfld_mc_private *mc_drv_ctx = platform_get_drvdata(pdev);
- platform_device_unregister(socdev);
+ pr_debug("snd_mfld_mc_remove called\n");
+ free_irq(platform_get_irq(pdev, 0), mc_drv_ctx);
+ snd_soc_unregister_card(&snd_soc_card_mfld);
+ kfree(mc_drv_ctx);
platform_set_drvdata(pdev, NULL);
return 0;
}
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index 96e6e9c..ee2c224 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -365,6 +365,14 @@
return stream->stream_info.buffer_ptr;
}
+static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+ memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
+
+ return 0;
+}
static struct snd_pcm_ops sst_platform_ops = {
.open = sst_platform_open,
@@ -373,6 +381,7 @@
.prepare = sst_platform_pcm_prepare,
.trigger = sst_platform_pcm_trigger,
.pointer = sst_platform_pcm_pointer,
+ .hw_params = sst_platform_pcm_hw_params,
};
static void sst_pcm_free(struct snd_pcm *pcm)
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index 28333e7..dc65650 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -117,7 +117,7 @@
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9705-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9705-codec",
@@ -126,7 +126,7 @@
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name = "wm9705-aux",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9705-codec",
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index 01bf316..51897fc 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -99,7 +99,7 @@
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9705-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9705-codec",
@@ -109,7 +109,7 @@
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name ="wm9705-aux",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9705-codec",
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index c6a37c6..053ed20 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -89,7 +89,7 @@
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9712-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
@@ -98,7 +98,7 @@
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name ="wm9712-aux",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index fc22e6e..b13a425 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -37,7 +37,7 @@
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9712-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
@@ -45,7 +45,7 @@
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name ="wm9712-aux",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 0d70fc8..38ca675 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -162,7 +162,7 @@
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9713-hifi",
.codec_name = "wm9713-codec",
.init = mioa701_wm9713_init,
@@ -172,7 +172,7 @@
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name ="wm9713-aux",
.codec_name = "wm9713-codec",
.platform_name = "pxa-pcm-audio",
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 857db96..504e400 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -132,7 +132,7 @@
{
.name = "AC97 HiFi",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9712-hifi",
.codec_name = "wm9712-codec",
.platform_name = "pxa-pcm-audio",
@@ -141,7 +141,7 @@
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name = "wm9712-aux",
.codec_name = "wm9712-codec",
.platform_name = "pxa-pcm-audio",
diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c
index db1dd56..2afabaf 100644
--- a/sound/soc/pxa/raumfeld.c
+++ b/sound/soc/pxa/raumfeld.c
@@ -229,19 +229,19 @@
{
.name = "ak4104",
.stream_name = "Playback",
- .cpu_dai_name = "pxa-ssp-dai.1",
- .codec_dai_name = "ak4104-hifi",
- .platform_name = "pxa-pcm-audio",
+ .cpu_dai_name = "pxa-ssp-dai.1",
+ .codec_dai_name = "ak4104-hifi",
+ .platform_name = "pxa-pcm-audio",
.ops = &raumfeld_ak4104_ops,
- .codec_name = "ak4104-codec.0",
+ .codec_name = "ak4104-codec.0",
},
{
.name = "CS4270",
.stream_name = "CS4270",
- .cpu_dai_name = "pxa-ssp-dai.0",
- .platform_name = "pxa-pcm-audio",
- .codec_dai_name = "cs4270-hifi",
- .codec_name = "cs4270-codec.0-0048",
+ .cpu_dai_name = "pxa-ssp-dai.0",
+ .platform_name = "pxa-pcm-audio",
+ .codec_dai_name = "cs4270-hifi",
+ .codec_name = "cs4270-codec.0-0048",
.ops = &raumfeld_cs4270_ops,
},};
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 489139a..9a23513 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -219,7 +219,7 @@
{
.name = "AC97",
.stream_name = "AC97 HiFi",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_dai_name = "wm9712-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
@@ -229,7 +229,7 @@
{
.name = "AC97 Aux",
.stream_name = "AC97 Aux",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_dai_name = "wm9712-aux",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm9712-codec",
diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
index 3ceaef6..d69d9fc 100644
--- a/sound/soc/pxa/z2.c
+++ b/sound/soc/pxa/z2.c
@@ -95,6 +95,11 @@
.pin = "Headphone Jack",
.mask = SND_JACK_HEADPHONE,
},
+ {
+ .pin = "Ext Spk",
+ .mask = SND_JACK_HEADPHONE,
+ .invert = 1
+ },
};
/* Headset jack detection gpios */
@@ -147,7 +152,7 @@
snd_soc_dapm_disable_pin(dapm, "LINPUT3");
snd_soc_dapm_disable_pin(dapm, "RINPUT3");
snd_soc_dapm_disable_pin(dapm, "OUT3");
- snd_soc_dapm_disable_pin(dapm, "MONO");
+ snd_soc_dapm_disable_pin(dapm, "MONO1");
/* Add z2 specific widgets */
snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets,
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index c585829..ac57726 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -166,7 +166,7 @@
.stream_name = "AC97 HiFi",
.codec_name = "wm9713-codec",
.platform_name = "pxa-pcm-audio",
- .cpu_dai_name = "pxa-ac97.0",
+ .cpu_dai_name = "pxa2xx-ac97",
.codec_name = "wm9713-hifi",
.init = zylonite_wm9713_init,
},
@@ -175,7 +175,7 @@
.stream_name = "AC97 Aux",
.codec_name = "wm9713-codec",
.platform_name = "pxa-pcm-audio",
- .cpu_dai_name = "pxa-ac97.1",
+ .cpu_dai_name = "pxa2xx-ac97-aux",
.codec_name = "wm9713-aux",
},
{
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index a6a6b5f..a08237a 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -35,23 +35,16 @@
tristate
config SND_SOC_SAMSUNG_NEO1973_WM8753
- tristate "SoC I2S Audio support for NEO1973 - WM8753"
- depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA01
+ tristate "Audio support for Openmoko Neo1973 Smartphones (GTA01/GTA02)"
+ depends on SND_SOC_SAMSUNG && (MACH_NEO1973_GTA01 || MACH_NEO1973_GTA02)
select SND_S3C24XX_I2S
select SND_SOC_WM8753
+ select SND_SOC_LM4857 if MACH_NEO1973_GTA01
+ select SND_SOC_DFBMCS320
help
- Say Y if you want to add support for SoC audio on smdk2440
- with the WM8753.
+ Say Y here to enable audio support for the Openmoko Neo1973
+ Smartphones.
-config SND_SOC_SAMSUNG_NEO1973_GTA02_WM8753
- tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)"
- depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02
- select SND_S3C24XX_I2S
- select SND_SOC_WM8753
- help
- This driver provides audio support for the Openmoko Neo FreeRunner
- smartphone.
-
config SND_SOC_SAMSUNG_JIVE_WM8750
tristate "SoC I2S Audio support for Jive"
depends on SND_SOC_SAMSUNG && MACH_JIVE
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 705d4e8..294dec0 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -20,7 +20,6 @@
# S3C24XX Machine Support
snd-soc-jive-wm8750-objs := jive_wm8750.o
snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
-snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o
snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
@@ -38,7 +37,6 @@
obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o
obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
-obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o
obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
obj-$(CONFIG_SND_SOC_SAMSUNG_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
obj-$(CONFIG_SND_SOC_SAMSUNG_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index 9bce1df..5cb3b88 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -310,7 +310,7 @@
/* we seem to be getting the odd error from the pcm library due
* to out-of-bounds pointers. this is maybe due to the dma engine
* not having loaded the new values for the channel before being
- * callled... (todo - fix )
+ * called... (todo - fix )
*/
if (res >= snd_pcm_lib_buffer_bytes(substream)) {
diff --git a/sound/soc/samsung/lm4857.h b/sound/soc/samsung/lm4857.h
deleted file mode 100644
index 0cf5b70..0000000
--- a/sound/soc/samsung/lm4857.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * lm4857.h -- ALSA Soc Audio Layer
- *
- * Copyright 2007 Wolfson Microelectronics PLC.
- * Author: Graeme Gregory
- * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
- *
- * 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.
- *
- * Revision history
- * 18th Jun 2007 Initial version.
- */
-
-#ifndef LM4857_H_
-#define LM4857_H_
-
-/* The register offsets in the cache array */
-#define LM4857_MVOL 0
-#define LM4857_LVOL 1
-#define LM4857_RVOL 2
-#define LM4857_CTRL 3
-
-/* the shifts required to set these bits */
-#define LM4857_3D 5
-#define LM4857_WAKEUP 5
-#define LM4857_EPGAIN 4
-
-#endif /*LM4857_H_*/
-
diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c
deleted file mode 100644
index 95ebf81..0000000
--- a/sound/soc/samsung/neo1973_gta02_wm8753.c
+++ /dev/null
@@ -1,494 +0,0 @@
-/*
- * neo1973_gta02_wm8753.c -- SoC audio for Openmoko Freerunner(GTA02)
- *
- * Copyright 2007 Openmoko Inc
- * Author: Graeme Gregory <graeme@openmoko.org>
- * Copyright 2007 Wolfson Microelectronics PLC.
- * Author: Graeme Gregory <linux@wolfsonmicro.com>
- * Copyright 2009 Wolfson Microelectronics
- *
- * 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 <linux/gpio.h>
-
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <plat/regs-iis.h>
-#include <mach/gta02.h>
-
-#include "../codecs/wm8753.h"
-#include "s3c24xx-i2s.h"
-
-static struct snd_soc_card neo1973_gta02;
-
-static int neo1973_gta02_hifi_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;
- unsigned int pll_out = 0, bclk = 0;
- int ret = 0;
- unsigned long iis_clkrate;
-
- iis_clkrate = s3c24xx_i2s_get_clockrate();
-
- switch (params_rate(params)) {
- case 8000:
- case 16000:
- pll_out = 12288000;
- break;
- case 48000:
- bclk = WM8753_BCLK_DIV_4;
- pll_out = 12288000;
- break;
- case 96000:
- bclk = WM8753_BCLK_DIV_2;
- pll_out = 12288000;
- break;
- case 11025:
- bclk = WM8753_BCLK_DIV_16;
- pll_out = 11289600;
- break;
- case 22050:
- bclk = WM8753_BCLK_DIV_8;
- pll_out = 11289600;
- break;
- case 44100:
- bclk = WM8753_BCLK_DIV_4;
- pll_out = 11289600;
- break;
- case 88200:
- bclk = WM8753_BCLK_DIV_2;
- pll_out = 11289600;
- break;
- }
-
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai,
- SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai,
- SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- /* set the codec system clock for DAC and ADC */
- ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out,
- SND_SOC_CLOCK_IN);
- if (ret < 0)
- return ret;
-
- /* set MCLK division for sample rate */
- ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
- S3C2410_IISMOD_32FS);
- if (ret < 0)
- return ret;
-
- /* set codec BCLK division for sample rate */
- ret = snd_soc_dai_set_clkdiv(codec_dai,
- WM8753_BCLKDIV, bclk);
- if (ret < 0)
- return ret;
-
- /* set prescaler division for sample rate */
- ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
- S3C24XX_PRESCALE(4, 4));
- if (ret < 0)
- return ret;
-
- /* codec PLL input is PCLK/4 */
- ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0,
- iis_clkrate / 4, pll_out);
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
-
- /* disable the PLL */
- return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0);
-}
-
-/*
- * Neo1973 WM8753 HiFi DAI opserations.
- */
-static struct snd_soc_ops neo1973_gta02_hifi_ops = {
- .hw_params = neo1973_gta02_hifi_hw_params,
- .hw_free = neo1973_gta02_hifi_hw_free,
-};
-
-static int neo1973_gta02_voice_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;
- unsigned int pcmdiv = 0;
- int ret = 0;
- unsigned long iis_clkrate;
-
- iis_clkrate = s3c24xx_i2s_get_clockrate();
-
- if (params_rate(params) != 8000)
- return -EINVAL;
- if (params_channels(params) != 1)
- return -EINVAL;
-
- pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
-
- /* todo: gg check mode (DSP_B) against CSR datasheet */
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
- if (ret < 0)
- return ret;
-
- /* set the codec system clock for DAC and ADC */
- ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK,
- 12288000, SND_SOC_CLOCK_IN);
- if (ret < 0)
- return ret;
-
- /* set codec PCM division for sample rate */
- ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV,
- pcmdiv);
- if (ret < 0)
- return ret;
-
- /* configure and enable PLL for 12.288MHz output */
- ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0,
- iis_clkrate / 4, 12288000);
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
-
- /* disable the PLL */
- return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0);
-}
-
-static struct snd_soc_ops neo1973_gta02_voice_ops = {
- .hw_params = neo1973_gta02_voice_hw_params,
- .hw_free = neo1973_gta02_voice_hw_free,
-};
-
-#define LM4853_AMP 1
-#define LM4853_SPK 2
-
-static u8 lm4853_state;
-
-/* This has no effect, it exists only to maintain compatibility with
- * existing ALSA state files.
- */
-static int lm4853_set_state(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- int val = ucontrol->value.integer.value[0];
-
- if (val)
- lm4853_state |= LM4853_AMP;
- else
- lm4853_state &= ~LM4853_AMP;
-
- return 0;
-}
-
-static int lm4853_get_state(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP;
-
- return 0;
-}
-
-static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- int val = ucontrol->value.integer.value[0];
-
- if (val) {
- lm4853_state |= LM4853_SPK;
- gpio_set_value(GTA02_GPIO_HP_IN, 0);
- } else {
- lm4853_state &= ~LM4853_SPK;
- gpio_set_value(GTA02_GPIO_HP_IN, 1);
- }
-
- return 0;
-}
-
-static int lm4853_get_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1;
-
- return 0;
-}
-
-static int lm4853_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k,
- int event)
-{
- gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(event));
-
- return 0;
-}
-
-static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
- SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
- SND_SOC_DAPM_LINE("GSM Line Out", NULL),
- SND_SOC_DAPM_LINE("GSM Line In", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_MIC("Handset Mic", NULL),
- SND_SOC_DAPM_SPK("Handset Spk", NULL),
-};
-
-
-/* example machine audio_mapnections */
-static const struct snd_soc_dapm_route audio_map[] = {
-
- /* Connections to the lm4853 amp */
- {"Stereo Out", NULL, "LOUT1"},
- {"Stereo Out", NULL, "ROUT1"},
-
- /* Connections to the GSM Module */
- {"GSM Line Out", NULL, "MONO1"},
- {"GSM Line Out", NULL, "MONO2"},
- {"RXP", NULL, "GSM Line In"},
- {"RXN", NULL, "GSM Line In"},
-
- /* Connections to Headset */
- {"MIC1", NULL, "Mic Bias"},
- {"Mic Bias", NULL, "Headset Mic"},
-
- /* Call Mic */
- {"MIC2", NULL, "Mic Bias"},
- {"MIC2N", NULL, "Mic Bias"},
- {"Mic Bias", NULL, "Handset Mic"},
-
- /* Call Speaker */
- {"Handset Spk", NULL, "LOUT2"},
- {"Handset Spk", NULL, "ROUT2"},
-
- /* Connect the ALC pins */
- {"ACIN", NULL, "ACOP"},
-};
-
-static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = {
- SOC_DAPM_PIN_SWITCH("Stereo Out"),
- SOC_DAPM_PIN_SWITCH("GSM Line Out"),
- SOC_DAPM_PIN_SWITCH("GSM Line In"),
- SOC_DAPM_PIN_SWITCH("Headset Mic"),
- SOC_DAPM_PIN_SWITCH("Handset Mic"),
- SOC_DAPM_PIN_SWITCH("Handset Spk"),
-
- /* This has no effect, it exists only to maintain compatibility with
- * existing ALSA state files.
- */
- SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0,
- lm4853_get_state,
- lm4853_set_state),
- SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0,
- lm4853_get_spk,
- lm4853_set_spk),
-};
-
-/*
- * This is an example machine initialisation for a wm8753 connected to a
- * neo1973 GTA02.
- */
-static int neo1973_gta02_wm8753_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
- int err;
-
- /* set up NC codec pins */
- snd_soc_dapm_nc_pin(dapm, "OUT3");
- snd_soc_dapm_nc_pin(dapm, "OUT4");
- snd_soc_dapm_nc_pin(dapm, "LINE1");
- snd_soc_dapm_nc_pin(dapm, "LINE2");
-
- /* Add neo1973 gta02 specific widgets */
- snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets,
- ARRAY_SIZE(wm8753_dapm_widgets));
-
- /* add neo1973 gta02 specific controls */
- err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls,
- ARRAY_SIZE(wm8753_neo1973_gta02_controls));
-
- if (err < 0)
- return err;
-
- /* set up neo1973 gta02 specific audio path audio_map */
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
- /* set endpoints to default off mode */
- snd_soc_dapm_disable_pin(dapm, "Stereo Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line In");
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_disable_pin(dapm, "Handset Mic");
- snd_soc_dapm_disable_pin(dapm, "Handset Spk");
-
- /* allow audio paths from the GSM modem to run during suspend */
- snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
- snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out");
- snd_soc_dapm_ignore_suspend(dapm, "GSM Line In");
- snd_soc_dapm_ignore_suspend(dapm, "Headset Mic");
- snd_soc_dapm_ignore_suspend(dapm, "Handset Mic");
- snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
-
- snd_soc_dapm_sync(dapm);
-
- return 0;
-}
-
-/*
- * BT Codec DAI
- */
-static struct snd_soc_dai_driver bt_dai = {
- .name = "bluetooth-dai",
- .playback = {
- .channels_min = 1,
- .channels_max = 1,
- .rates = SNDRV_PCM_RATE_8000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
- .capture = {
- .channels_min = 1,
- .channels_max = 1,
- .rates = SNDRV_PCM_RATE_8000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
-};
-
-static struct snd_soc_dai_link neo1973_gta02_dai[] = {
-{ /* Hifi Playback - for similatious use with voice below */
- .name = "WM8753",
- .stream_name = "WM8753 HiFi",
- .cpu_dai_name = "s3c24xx-iis",
- .codec_dai_name = "wm8753-hifi",
- .init = neo1973_gta02_wm8753_init,
- .platform_name = "samsung-audio",
- .codec_name = "wm8753-codec.0-001a",
- .ops = &neo1973_gta02_hifi_ops,
-},
-{ /* Voice via BT */
- .name = "Bluetooth",
- .stream_name = "Voice",
- .cpu_dai_name = "bluetooth-dai",
- .codec_dai_name = "wm8753-voice",
- .ops = &neo1973_gta02_voice_ops,
- .codec_name = "wm8753-codec.0-001a",
- .platform_name = "samsung-audio",
-},
-};
-
-static struct snd_soc_card neo1973_gta02 = {
- .name = "neo1973-gta02",
- .dai_link = neo1973_gta02_dai,
- .num_links = ARRAY_SIZE(neo1973_gta02_dai),
-};
-
-static struct platform_device *neo1973_gta02_snd_device;
-
-static int __init neo1973_gta02_init(void)
-{
- int ret;
-
- if (!machine_is_neo1973_gta02()) {
- printk(KERN_INFO
- "Only GTA02 is supported by this ASoC driver\n");
- return -ENODEV;
- }
-
- neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1);
- if (!neo1973_gta02_snd_device)
- return -ENOMEM;
-
- /* register bluetooth DAI here */
- ret = snd_soc_register_dai(&neo1973_gta02_snd_device->dev, &bt_dai);
- if (ret)
- goto err_put_device;
-
- platform_set_drvdata(neo1973_gta02_snd_device, &neo1973_gta02);
- ret = platform_device_add(neo1973_gta02_snd_device);
-
- if (ret)
- goto err_unregister_dai;
-
- /* Initialise GPIOs used by amp */
- ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN");
- if (ret) {
- pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN);
- goto err_del_device;
- }
-
- ret = gpio_direction_output(GTA02_GPIO_HP_IN, 1);
- if (ret) {
- pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN);
- goto err_free_gpio_hp_in;
- }
-
- ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT");
- if (ret) {
- pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT);
- goto err_free_gpio_hp_in;
- }
-
- ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1);
- if (ret) {
- pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT);
- goto err_free_gpio_amp_shut;
- }
-
- return 0;
-
-err_free_gpio_amp_shut:
- gpio_free(GTA02_GPIO_AMP_SHUT);
-err_free_gpio_hp_in:
- gpio_free(GTA02_GPIO_HP_IN);
-err_del_device:
- platform_device_del(neo1973_gta02_snd_device);
-err_unregister_dai:
- snd_soc_unregister_dai(&neo1973_gta02_snd_device->dev);
-err_put_device:
- platform_device_put(neo1973_gta02_snd_device);
- return ret;
-}
-module_init(neo1973_gta02_init);
-
-static void __exit neo1973_gta02_exit(void)
-{
- snd_soc_unregister_dai(&neo1973_gta02_snd_device->dev);
- platform_device_unregister(neo1973_gta02_snd_device);
- gpio_free(GTA02_GPIO_HP_IN);
- gpio_free(GTA02_GPIO_AMP_SHUT);
-}
-module_exit(neo1973_gta02_exit);
-
-/* Module information */
-MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org");
-MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index d3cd688..78bfdb3 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -1,56 +1,32 @@
/*
- * neo1973_wm8753.c -- SoC audio for Neo1973
+ * neo1973_wm8753.c -- SoC audio for Openmoko Neo1973 and Freerunner devices
*
+ * Copyright 2007 Openmoko Inc
+ * Author: Graeme Gregory <graeme@openmoko.org>
* Copyright 2007 Wolfson Microelectronics PLC.
* Author: Graeme Gregory
* graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Copyright 2009 Wolfson Microelectronics
*
* 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 <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
#include <linux/platform_device.h>
-#include <linux/i2c.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
+#include <linux/gpio.h>
+
#include <sound/soc.h>
-#include <sound/tlv.h>
#include <asm/mach-types.h>
-#include <mach/regs-clock.h>
-#include <mach/regs-gpio.h>
-#include <mach/hardware.h>
-#include <linux/io.h>
-#include <mach/spi-gpio.h>
-
#include <plat/regs-iis.h>
+#include <mach/gta02.h>
#include "../codecs/wm8753.h"
-#include "lm4857.h"
-#include "dma.h"
#include "s3c24xx-i2s.h"
-/* define the scenarios */
-#define NEO_AUDIO_OFF 0
-#define NEO_GSM_CALL_AUDIO_HANDSET 1
-#define NEO_GSM_CALL_AUDIO_HEADSET 2
-#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3
-#define NEO_STEREO_TO_SPEAKERS 4
-#define NEO_STEREO_TO_HEADPHONES 5
-#define NEO_CAPTURE_HANDSET 6
-#define NEO_CAPTURE_HEADSET 7
-#define NEO_CAPTURE_BLUETOOTH 8
-
-static struct snd_soc_card neo1973;
-static struct i2c_client *i2c;
-
static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -61,8 +37,6 @@
int ret = 0;
unsigned long iis_clkrate;
- pr_debug("Entered %s\n", __func__);
-
iis_clkrate = s3c24xx_i2s_get_clockrate();
switch (params_rate(params)) {
@@ -147,8 +121,6 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- pr_debug("Entered %s\n", __func__);
-
/* disable the PLL */
return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0);
}
@@ -170,8 +142,6 @@
int ret = 0;
unsigned long iis_clkrate;
- pr_debug("Entered %s\n", __func__);
-
iis_clkrate = s3c24xx_i2s_get_clockrate();
if (params_rate(params) != 8000)
@@ -213,8 +183,6 @@
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- pr_debug("Entered %s\n", __func__);
-
/* disable the PLL */
return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0);
}
@@ -224,205 +192,16 @@
.hw_free = neo1973_voice_hw_free,
};
-static int neo1973_scenario;
+/* Shared routes and controls */
-static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.integer.value[0] = neo1973_scenario;
- return 0;
-}
-
-static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
-{
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- pr_debug("Entered %s\n", __func__);
-
- switch (neo1973_scenario) {
- case NEO_AUDIO_OFF:
- snd_soc_dapm_disable_pin(dapm, "Audio Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line In");
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_disable_pin(dapm, "Call Mic");
- break;
- case NEO_GSM_CALL_AUDIO_HANDSET:
- snd_soc_dapm_enable_pin(dapm, "Audio Out");
- snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
- snd_soc_dapm_enable_pin(dapm, "GSM Line In");
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_enable_pin(dapm, "Call Mic");
- break;
- case NEO_GSM_CALL_AUDIO_HEADSET:
- snd_soc_dapm_enable_pin(dapm, "Audio Out");
- snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
- snd_soc_dapm_enable_pin(dapm, "GSM Line In");
- snd_soc_dapm_enable_pin(dapm, "Headset Mic");
- snd_soc_dapm_disable_pin(dapm, "Call Mic");
- break;
- case NEO_GSM_CALL_AUDIO_BLUETOOTH:
- snd_soc_dapm_disable_pin(dapm, "Audio Out");
- snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
- snd_soc_dapm_enable_pin(dapm, "GSM Line In");
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_disable_pin(dapm, "Call Mic");
- break;
- case NEO_STEREO_TO_SPEAKERS:
- snd_soc_dapm_enable_pin(dapm, "Audio Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line In");
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_disable_pin(dapm, "Call Mic");
- break;
- case NEO_STEREO_TO_HEADPHONES:
- snd_soc_dapm_enable_pin(dapm, "Audio Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line In");
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_disable_pin(dapm, "Call Mic");
- break;
- case NEO_CAPTURE_HANDSET:
- snd_soc_dapm_disable_pin(dapm, "Audio Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line In");
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_enable_pin(dapm, "Call Mic");
- break;
- case NEO_CAPTURE_HEADSET:
- snd_soc_dapm_disable_pin(dapm, "Audio Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line In");
- snd_soc_dapm_enable_pin(dapm, "Headset Mic");
- snd_soc_dapm_disable_pin(dapm, "Call Mic");
- break;
- case NEO_CAPTURE_BLUETOOTH:
- snd_soc_dapm_disable_pin(dapm, "Audio Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line In");
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_disable_pin(dapm, "Call Mic");
- break;
- default:
- snd_soc_dapm_disable_pin(dapm, "Audio Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
- snd_soc_dapm_disable_pin(dapm, "GSM Line In");
- snd_soc_dapm_disable_pin(dapm, "Headset Mic");
- snd_soc_dapm_disable_pin(dapm, "Call Mic");
- }
-
- snd_soc_dapm_sync(dapm);
-
- return 0;
-}
-
-static int neo1973_set_scenario(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-
- pr_debug("Entered %s\n", __func__);
-
- if (neo1973_scenario == ucontrol->value.integer.value[0])
- return 0;
-
- neo1973_scenario = ucontrol->value.integer.value[0];
- set_scenario_endpoints(codec, neo1973_scenario);
- return 1;
-}
-
-static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0};
-
-static void lm4857_write_regs(void)
-{
- pr_debug("Entered %s\n", __func__);
-
- if (i2c_master_send(i2c, lm4857_regs, 4) != 4)
- printk(KERN_ERR "lm4857: i2c write failed\n");
-}
-
-static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct soc_mixer_control *mc =
- (struct soc_mixer_control *)kcontrol->private_value;
- int reg = mc->reg;
- int shift = mc->shift;
- int mask = mc->max;
-
- pr_debug("Entered %s\n", __func__);
-
- ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask;
- return 0;
-}
-
-static int lm4857_set_reg(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct soc_mixer_control *mc =
- (struct soc_mixer_control *)kcontrol->private_value;
- int reg = mc->reg;
- int shift = mc->shift;
- int mask = mc->max;
-
- if (((lm4857_regs[reg] >> shift) & mask) ==
- ucontrol->value.integer.value[0])
- return 0;
-
- lm4857_regs[reg] &= ~(mask << shift);
- lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift;
- lm4857_write_regs();
- return 1;
-}
-
-static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- u8 value = lm4857_regs[LM4857_CTRL] & 0x0F;
-
- pr_debug("Entered %s\n", __func__);
-
- if (value)
- value -= 5;
-
- ucontrol->value.integer.value[0] = value;
- return 0;
-}
-
-static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- u8 value = ucontrol->value.integer.value[0];
-
- pr_debug("Entered %s\n", __func__);
-
- if (value)
- value += 5;
-
- if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value)
- return 0;
-
- lm4857_regs[LM4857_CTRL] &= 0xF0;
- lm4857_regs[LM4857_CTRL] |= value;
- lm4857_write_regs();
- return 1;
-}
-
-static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
- SND_SOC_DAPM_LINE("Audio Out", NULL),
+static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = {
SND_SOC_DAPM_LINE("GSM Line Out", NULL),
SND_SOC_DAPM_LINE("GSM Line In", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_MIC("Call Mic", NULL),
+ SND_SOC_DAPM_MIC("Handset Mic", NULL),
};
-
-static const struct snd_soc_dapm_route dapm_routes[] = {
-
- /* Connections to the lm4857 amp */
- {"Audio Out", NULL, "LOUT1"},
- {"Audio Out", NULL, "ROUT1"},
-
+static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = {
/* Connections to the GSM Module */
{"GSM Line Out", NULL, "MONO1"},
{"GSM Line Out", NULL, "MONO2"},
@@ -436,123 +215,209 @@
/* Call Mic */
{"MIC2", NULL, "Mic Bias"},
{"MIC2N", NULL, "Mic Bias"},
- {"Mic Bias", NULL, "Call Mic"},
+ {"Mic Bias", NULL, "Handset Mic"},
/* Connect the ALC pins */
{"ACIN", NULL, "ACOP"},
};
-static const char *lm4857_mode[] = {
- "Off",
- "Call Speaker",
- "Stereo Speakers",
- "Stereo Speakers + Headphones",
- "Headphones"
+static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
+ SOC_DAPM_PIN_SWITCH("GSM Line Out"),
+ SOC_DAPM_PIN_SWITCH("GSM Line In"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Handset Mic"),
};
-static const struct soc_enum lm4857_mode_enum[] = {
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode),
+/* GTA02 specific routes and controlls */
+
+#ifdef CONFIG_MACH_NEO1973_GTA02
+
+static int gta02_speaker_enabled;
+
+static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ gta02_speaker_enabled = ucontrol->value.integer.value[0];
+
+ gpio_set_value(GTA02_GPIO_HP_IN, !gta02_speaker_enabled);
+
+ return 0;
+}
+
+static int lm4853_get_spk(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = gta02_speaker_enabled;
+ return 0;
+}
+
+static int lm4853_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(event));
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_route neo1973_gta02_routes[] = {
+ /* Connections to the amp */
+ {"Stereo Out", NULL, "LOUT1"},
+ {"Stereo Out", NULL, "ROUT1"},
+
+ /* Call Speaker */
+ {"Handset Spk", NULL, "LOUT2"},
+ {"Handset Spk", NULL, "ROUT2"},
};
-static const char *neo_scenarios[] = {
- "Off",
- "GSM Handset",
- "GSM Headset",
- "GSM Bluetooth",
- "Speakers",
- "Headphones",
- "Capture Handset",
- "Capture Headset",
- "Capture Bluetooth"
+static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Handset Spk"),
+ SOC_DAPM_PIN_SWITCH("Stereo Out"),
+
+ SOC_SINGLE_BOOL_EXT("Amp Spk Switch", 0,
+ lm4853_get_spk,
+ lm4853_set_spk),
};
-static const struct soc_enum neo_scenario_enum[] = {
- SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios),
+static const struct snd_soc_dapm_widget neo1973_gta02_wm8753_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Handset Spk", NULL),
+ SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
};
-static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0);
-static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0);
+static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ int ret;
-static const struct snd_kcontrol_new wm8753_neo1973_controls[] = {
- SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0,
- lm4857_get_reg, lm4857_set_reg, stereo_tlv),
- SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0,
- lm4857_get_reg, lm4857_set_reg, stereo_tlv),
- SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0,
- lm4857_get_reg, lm4857_set_reg, mono_tlv),
- SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0],
- lm4857_get_mode, lm4857_set_mode),
- SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0],
- neo1973_get_scenario, neo1973_set_scenario),
- SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0,
- lm4857_get_reg, lm4857_set_reg),
- SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0,
- lm4857_get_reg, lm4857_set_reg),
- SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0,
- lm4857_get_reg, lm4857_set_reg),
- SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0,
- lm4857_get_reg, lm4857_set_reg),
-};
+ ret = snd_soc_dapm_new_controls(dapm, neo1973_gta02_wm8753_dapm_widgets,
+ ARRAY_SIZE(neo1973_gta02_wm8753_dapm_widgets));
+ if (ret)
+ return ret;
-/*
- * This is an example machine initialisation for a wm8753 connected to a
- * neo1973 II. It is missing logic to detect hp/mic insertions and logic
- * to re-route the audio in such an event.
- */
+ ret = snd_soc_dapm_add_routes(dapm, neo1973_gta02_routes,
+ ARRAY_SIZE(neo1973_gta02_routes));
+ if (ret)
+ return ret;
+
+ ret = snd_soc_add_controls(codec, neo1973_gta02_wm8753_controls,
+ ARRAY_SIZE(neo1973_gta02_wm8753_controls));
+ if (ret)
+ return ret;
+
+ snd_soc_dapm_disable_pin(dapm, "Stereo Out");
+ snd_soc_dapm_disable_pin(dapm, "Handset Spk");
+ snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
+ snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
+
+ return 0;
+}
+
+#else
+static int neo1973_gta02_wm8753_init(struct snd_soc_code *codec) { return 0; }
+#endif
+
static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- int err;
-
- pr_debug("Entered %s\n", __func__);
+ int ret;
/* set up NC codec pins */
- snd_soc_dapm_nc_pin(dapm, "LOUT2");
- snd_soc_dapm_nc_pin(dapm, "ROUT2");
+ if (machine_is_neo1973_gta01()) {
+ snd_soc_dapm_nc_pin(dapm, "LOUT2");
+ snd_soc_dapm_nc_pin(dapm, "ROUT2");
+ }
snd_soc_dapm_nc_pin(dapm, "OUT3");
snd_soc_dapm_nc_pin(dapm, "OUT4");
snd_soc_dapm_nc_pin(dapm, "LINE1");
snd_soc_dapm_nc_pin(dapm, "LINE2");
/* Add neo1973 specific widgets */
- snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets,
- ARRAY_SIZE(wm8753_dapm_widgets));
-
- /* set endpoints to default mode */
- set_scenario_endpoints(codec, NEO_AUDIO_OFF);
+ ret = snd_soc_dapm_new_controls(dapm, neo1973_wm8753_dapm_widgets,
+ ARRAY_SIZE(neo1973_wm8753_dapm_widgets));
+ if (ret)
+ return ret;
/* add neo1973 specific controls */
- err = snd_soc_add_controls(codec, wm8753_neo1973_controls,
- ARRAY_SIZE(8753_neo1973_controls));
- if (err < 0)
- return err;
+ ret = snd_soc_add_controls(codec, neo1973_wm8753_controls,
+ ARRAY_SIZE(neo1973_wm8753_controls));
+ if (ret)
+ return ret;
/* set up neo1973 specific audio routes */
- err = snd_soc_dapm_add_routes(dapm, dapm_routes,
- ARRAY_SIZE(dapm_routes));
+ ret = snd_soc_dapm_add_routes(dapm, neo1973_wm8753_routes,
+ ARRAY_SIZE(neo1973_wm8753_routes));
+ if (ret)
+ return ret;
+
+ /* set endpoints to default off mode */
+ snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+ snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+ snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+ snd_soc_dapm_disable_pin(dapm, "Handset Mic");
+
+ /* allow audio paths from the GSM modem to run during suspend */
+ snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out");
+ snd_soc_dapm_ignore_suspend(dapm, "GSM Line In");
+ snd_soc_dapm_ignore_suspend(dapm, "Headset Mic");
+ snd_soc_dapm_ignore_suspend(dapm, "Handset Mic");
+
+ if (machine_is_neo1973_gta02()) {
+ ret = neo1973_gta02_wm8753_init(codec);
+ if (ret)
+ return ret;
+ }
snd_soc_dapm_sync(dapm);
+
return 0;
}
-/*
- * BT Codec DAI
- */
-static struct snd_soc_dai bt_dai = {
- .name = "bluetooth-dai",
- .playback = {
- .channels_min = 1,
- .channels_max = 1,
- .rates = SNDRV_PCM_RATE_8000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
- .capture = {
- .channels_min = 1,
- .channels_max = 1,
- .rates = SNDRV_PCM_RATE_8000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+/* GTA01 specific controlls */
+
+#ifdef CONFIG_MACH_NEO1973_GTA01
+
+static const struct snd_soc_dapm_route neo1973_lm4857_routes[] = {
+ {"Amp IN", NULL, "ROUT1"},
+ {"Amp IN", NULL, "LOUT1"},
+
+ {"Handset Spk", NULL, "Amp EP"},
+ {"Stereo Out", NULL, "Amp LS"},
+ {"Headphone", NULL, "Amp HP"},
};
+static const struct snd_soc_dapm_widget neo1973_lm4857_dapm_widgets[] = {
+ SND_SOC_DAPM_SPK("Handset Spk", NULL),
+ SND_SOC_DAPM_SPK("Stereo Out", NULL),
+ SND_SOC_DAPM_HP("Headphone", NULL),
+};
+
+static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm)
+{
+ int ret;
+
+ ret = snd_soc_dapm_new_controls(dapm, neo1973_lm4857_dapm_widgets,
+ ARRAY_SIZE(neo1973_lm4857_dapm_widgets));
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_add_routes(dapm, neo1973_lm4857_routes,
+ ARRAY_SIZE(neo1973_lm4857_routes));
+ if (ret)
+ return ret;
+
+ snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
+ snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
+ snd_soc_dapm_ignore_suspend(dapm, "Headphone");
+
+ snd_soc_dapm_sync(dapm);
+
+ return 0;
+}
+
+#else
+static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) { return 0; };
+#endif
+
static struct snd_soc_dai_link neo1973_dai[] = {
{ /* Hifi Playback - for similatious use with voice below */
.name = "WM8753",
@@ -568,90 +433,49 @@
.name = "Bluetooth",
.stream_name = "Voice",
.platform_name = "samsung-audio",
- .cpu_dai_name = "bluetooth-dai",
+ .cpu_dai_name = "dfbmcs320-pcm",
.codec_dai_name = "wm8753-voice",
.codec_name = "wm8753-codec.0-001a",
.ops = &neo1973_voice_ops,
},
};
+static struct snd_soc_aux_dev neo1973_aux_devs[] = {
+ {
+ .name = "dfbmcs320",
+ .codec_name = "dfbmcs320.0",
+ },
+ {
+ .name = "lm4857",
+ .codec_name = "lm4857.0-007c",
+ .init = neo1973_lm4857_init,
+ },
+};
+
+static struct snd_soc_codec_conf neo1973_codec_conf[] = {
+ {
+ .dev_name = "lm4857.0-007c",
+ .name_prefix = "Amp",
+ },
+};
+
+#ifdef CONFIG_MACH_NEO1973_GTA02
+static const struct gpio neo1973_gta02_gpios[] = {
+ { GTA02_GPIO_HP_IN, GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" },
+ { GTA02_GPIO_AMP_SHUT, GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" },
+};
+#else
+static const struct gpio neo1973_gta02_gpios[] = {};
+#endif
+
static struct snd_soc_card neo1973 = {
.name = "neo1973",
.dai_link = neo1973_dai,
.num_links = ARRAY_SIZE(neo1973_dai),
-};
-
-static int lm4857_i2c_probe(struct i2c_client *client,
- const struct i2c_device_id *id)
-{
- pr_debug("Entered %s\n", __func__);
-
- i2c = client;
-
- lm4857_write_regs();
- return 0;
-}
-
-static int lm4857_i2c_remove(struct i2c_client *client)
-{
- pr_debug("Entered %s\n", __func__);
-
- i2c = NULL;
-
- return 0;
-}
-
-static u8 lm4857_state;
-
-static int lm4857_suspend(struct i2c_client *dev, pm_message_t state)
-{
- pr_debug("Entered %s\n", __func__);
-
- dev_dbg(&dev->dev, "lm4857_suspend\n");
- lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf;
- if (lm4857_state) {
- lm4857_regs[LM4857_CTRL] &= 0xf0;
- lm4857_write_regs();
- }
- return 0;
-}
-
-static int lm4857_resume(struct i2c_client *dev)
-{
- pr_debug("Entered %s\n", __func__);
-
- if (lm4857_state) {
- lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f);
- lm4857_write_regs();
- }
- return 0;
-}
-
-static void lm4857_shutdown(struct i2c_client *dev)
-{
- pr_debug("Entered %s\n", __func__);
-
- dev_dbg(&dev->dev, "lm4857_shutdown\n");
- lm4857_regs[LM4857_CTRL] &= 0xf0;
- lm4857_write_regs();
-}
-
-static const struct i2c_device_id lm4857_i2c_id[] = {
- { "neo1973_lm4857", 0 },
- { }
-};
-
-static struct i2c_driver lm4857_i2c_driver = {
- .driver = {
- .name = "LM4857 I2C Amp",
- .owner = THIS_MODULE,
- },
- .suspend = lm4857_suspend,
- .resume = lm4857_resume,
- .shutdown = lm4857_shutdown,
- .probe = lm4857_i2c_probe,
- .remove = lm4857_i2c_remove,
- .id_table = lm4857_i2c_id,
+ .aux_dev = neo1973_aux_devs,
+ .num_aux_devs = ARRAY_SIZE(neo1973_aux_devs),
+ .codec_conf = neo1973_codec_conf,
+ .num_configs = ARRAY_SIZE(neo1973_codec_conf),
};
static struct platform_device *neo1973_snd_device;
@@ -660,46 +484,56 @@
{
int ret;
- pr_debug("Entered %s\n", __func__);
-
- if (!machine_is_neo1973_gta01()) {
- printk(KERN_INFO
- "Only GTA01 hardware supported by ASoC driver\n");
+ if (!machine_is_neo1973_gta01() && !machine_is_neo1973_gta02())
return -ENODEV;
+
+ if (machine_is_neo1973_gta02()) {
+ neo1973.name = "neo1973gta02";
+ neo1973.num_aux_devs = 1;
+
+ ret = gpio_request_array(neo1973_gta02_gpios,
+ ARRAY_SIZE(neo1973_gta02_gpios));
+ if (ret)
+ return ret;
}
neo1973_snd_device = platform_device_alloc("soc-audio", -1);
- if (!neo1973_snd_device)
- return -ENOMEM;
+ if (!neo1973_snd_device) {
+ ret = -ENOMEM;
+ goto err_gpio_free;
+ }
platform_set_drvdata(neo1973_snd_device, &neo1973);
ret = platform_device_add(neo1973_snd_device);
- if (ret) {
- platform_device_put(neo1973_snd_device);
- return ret;
+ if (ret)
+ goto err_put_device;
+
+ return 0;
+
+err_put_device:
+ platform_device_put(neo1973_snd_device);
+err_gpio_free:
+ if (machine_is_neo1973_gta02()) {
+ gpio_free_array(neo1973_gta02_gpios,
+ ARRAY_SIZE(neo1973_gta02_gpios));
}
-
- ret = i2c_add_driver(&lm4857_i2c_driver);
-
- if (ret != 0)
- platform_device_unregister(neo1973_snd_device);
-
return ret;
}
+module_init(neo1973_init);
static void __exit neo1973_exit(void)
{
- pr_debug("Entered %s\n", __func__);
-
- i2c_del_driver(&lm4857_i2c_driver);
platform_device_unregister(neo1973_snd_device);
-}
-module_init(neo1973_init);
+ if (machine_is_neo1973_gta02()) {
+ gpio_free_array(neo1973_gta02_gpios,
+ ARRAY_SIZE(neo1973_gta02_gpios));
+ }
+}
module_exit(neo1973_exit);
/* Module information */
MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org, www.openmoko.org");
-MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973");
+MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 and Frerunner");
MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index db66dc4..5d76da4 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -1609,24 +1609,23 @@
return 0;
}
+ if (!codec->cache_ops || !codec->cache_ops->sync)
+ return -EINVAL;
+
if (codec->cache_ops->name)
name = codec->cache_ops->name;
else
name = "unknown";
- if (codec->cache_ops && codec->cache_ops->sync) {
- if (codec->cache_ops->name)
- dev_dbg(codec->dev, "Syncing %s cache for %s codec\n",
- codec->cache_ops->name, codec->name);
- trace_snd_soc_cache_sync(codec, name, "start");
- ret = codec->cache_ops->sync(codec);
- if (!ret)
- codec->cache_sync = 0;
- trace_snd_soc_cache_sync(codec, name, "end");
- return ret;
- }
-
- return -EINVAL;
+ if (codec->cache_ops->name)
+ dev_dbg(codec->dev, "Syncing %s cache for %s codec\n",
+ codec->cache_ops->name, codec->name);
+ trace_snd_soc_cache_sync(codec, name, "start");
+ ret = codec->cache_ops->sync(codec);
+ if (!ret)
+ codec->cache_sync = 0;
+ trace_snd_soc_cache_sync(codec, name, "end");
+ return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_cache_sync);
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 205cbd7..17efacd 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -87,15 +87,56 @@
return c;
}
-/* codec register dump */
-static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
+/* fill buf which is 'len' bytes with a formatted
+ * string of the form 'reg: value\n' */
+static int format_register_str(struct snd_soc_codec *codec,
+ unsigned int reg, char *buf, size_t len)
{
- int ret, i, step = 1, count = 0;
+ int wordsize = codec->driver->reg_word_size * 2;
+ int regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+ int ret;
+ char tmpbuf[len + 1];
+ char regbuf[regsize + 1];
+
+ /* since tmpbuf is allocated on the stack, warn the callers if they
+ * try to abuse this function */
+ WARN_ON(len > 63);
+
+ /* +2 for ': ' and + 1 for '\n' */
+ if (wordsize + regsize + 2 + 1 != len)
+ return -EINVAL;
+
+ ret = snd_soc_read(codec , reg);
+ if (ret < 0) {
+ memset(regbuf, 'X', regsize);
+ regbuf[regsize] = '\0';
+ } else {
+ snprintf(regbuf, regsize + 1, "%.*x", regsize, ret);
+ }
+
+ /* prepare the buffer */
+ snprintf(tmpbuf, len + 1, "%.*x: %s\n", wordsize, reg, regbuf);
+ /* copy it back to the caller without the '\0' */
+ memcpy(buf, tmpbuf, len);
+
+ return 0;
+}
+
+/* codec register dump */
+static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf,
+ size_t count, loff_t pos)
+{
+ int i, step = 1;
int wordsize, regsize;
+ int len;
+ size_t total = 0;
+ loff_t p = 0;
wordsize = codec->driver->reg_word_size * 2;
regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+ len = wordsize + regsize + 2 + 1;
+
if (!codec->driver->reg_cache_size)
return 0;
@@ -105,51 +146,34 @@
for (i = 0; i < codec->driver->reg_cache_size; i += step) {
if (codec->readable_register && !codec->readable_register(codec, i))
continue;
-
- count += sprintf(buf + count, "%.*x: ", regsize, i);
- if (count >= PAGE_SIZE - 1)
- break;
-
if (codec->driver->display_register) {
count += codec->driver->display_register(codec, buf + count,
PAGE_SIZE - count, i);
} else {
- /* If the read fails it's almost certainly due to
- * the register being volatile and the device being
- * powered off.
- */
- ret = snd_soc_read(codec, i);
- if (ret >= 0)
- count += snprintf(buf + count,
- PAGE_SIZE - count,
- "%.*x", wordsize, ret);
- else
- count += snprintf(buf + count,
- PAGE_SIZE - count,
- "<no data: %d>", ret);
+ /* only support larger than PAGE_SIZE bytes debugfs
+ * entries for the default case */
+ if (p >= pos) {
+ if (total + len >= count - 1)
+ break;
+ format_register_str(codec, i, buf + total, len);
+ total += len;
+ }
+ p += len;
}
-
- if (count >= PAGE_SIZE - 1)
- break;
-
- count += snprintf(buf + count, PAGE_SIZE - count, "\n");
- if (count >= PAGE_SIZE - 1)
- break;
}
- /* Truncate count; min() would cause a warning */
- if (count >= PAGE_SIZE)
- count = PAGE_SIZE - 1;
+ total = min(total, count - 1);
- return count;
+ return total;
}
+
static ssize_t codec_reg_show(struct device *dev,
struct device_attribute *attr, char *buf)
{
struct snd_soc_pcm_runtime *rtd =
container_of(dev, struct snd_soc_pcm_runtime, dev);
- return soc_codec_reg_show(rtd->codec, buf);
+ return soc_codec_reg_show(rtd->codec, buf, PAGE_SIZE, 0);
}
static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
@@ -188,16 +212,28 @@
}
static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
- size_t count, loff_t *ppos)
+ size_t count, loff_t *ppos)
{
ssize_t ret;
struct snd_soc_codec *codec = file->private_data;
- char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
+ char *buf;
+
+ if (*ppos < 0 || !count)
+ return -EINVAL;
+
+ buf = kmalloc(count, GFP_KERNEL);
if (!buf)
return -ENOMEM;
- ret = soc_codec_reg_show(codec, buf);
- if (ret >= 0)
- ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
+
+ ret = soc_codec_reg_show(codec, buf, count, *ppos);
+ if (ret >= 0) {
+ if (copy_to_user(user_buf, buf, ret)) {
+ kfree(buf);
+ return -EFAULT;
+ }
+ *ppos += ret;
+ }
+
kfree(buf);
return ret;
}
@@ -464,20 +500,30 @@
struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret;
- if (codec_dai->driver->symmetric_rates || cpu_dai->driver->symmetric_rates ||
- rtd->dai_link->symmetric_rates) {
- dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n",
- rtd->rate);
+ if (!codec_dai->driver->symmetric_rates &&
+ !cpu_dai->driver->symmetric_rates &&
+ !rtd->dai_link->symmetric_rates)
+ return 0;
- ret = snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_RATE,
- rtd->rate,
- rtd->rate);
- if (ret < 0) {
- dev_err(&rtd->dev,
- "Unable to apply rate symmetry constraint: %d\n", ret);
- return ret;
- }
+ /* This can happen if multiple streams are starting simultaneously -
+ * the second can need to get its constraints before the first has
+ * picked a rate. Complain and allow the application to carry on.
+ */
+ if (!rtd->rate) {
+ dev_warn(&rtd->dev,
+ "Not enforcing symmetric_rates due to race\n");
+ return 0;
+ }
+
+ dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate);
+
+ ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ rtd->rate, rtd->rate);
+ if (ret < 0) {
+ dev_err(&rtd->dev,
+ "Unable to apply rate symmetry constraint: %d\n", ret);
+ return ret;
}
return 0;
@@ -1428,6 +1474,7 @@
struct snd_soc_codec *codec)
{
int ret = 0;
+ const struct snd_soc_codec_driver *driver = codec->driver;
codec->card = card;
codec->dapm.card = card;
@@ -1436,8 +1483,8 @@
if (!try_module_get(codec->dev->driver->owner))
return -ENODEV;
- if (codec->driver->probe) {
- ret = codec->driver->probe(codec);
+ if (driver->probe) {
+ ret = driver->probe(codec);
if (ret < 0) {
dev_err(codec->dev,
"asoc: failed to probe CODEC %s: %d\n",
@@ -1446,6 +1493,13 @@
}
}
+ if (driver->dapm_widgets)
+ snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets,
+ driver->num_dapm_widgets);
+ if (driver->dapm_routes)
+ snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes,
+ driver->num_dapm_routes);
+
soc_init_codec_debugfs(codec);
/* mark codec as probed and add to card codec list */
@@ -1482,6 +1536,7 @@
rtd = &card->rtd_aux[num];
name = aux_dev->name;
}
+ rtd->card = card;
/* machine controls, routes and widgets are not prefixed */
temp = codec->name_prefix;
@@ -1503,7 +1558,6 @@
/* register the rtd device */
rtd->codec = codec;
- rtd->card = card;
rtd->dev.parent = card->dev;
rtd->dev.release = rtd_release;
rtd->dev.init_name = name;
@@ -1801,7 +1855,12 @@
}
card->snd_card->dev = card->dev;
-#ifdef CONFIG_PM
+ card->dapm.bias_level = SND_SOC_BIAS_OFF;
+ card->dapm.dev = card->dev;
+ card->dapm.card = card;
+ list_add(&card->dapm.list, &card->dapm_list);
+
+#ifdef CONFIG_PM_SLEEP
/* deferred resume work */
INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
#endif
@@ -1831,11 +1890,37 @@
}
}
+ if (card->dapm_widgets)
+ snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets,
+ card->num_dapm_widgets);
+ if (card->dapm_routes)
+ snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
+ card->num_dapm_routes);
+
+#ifdef CONFIG_DEBUG_FS
+ card->dapm.debugfs_dapm = debugfs_create_dir("dapm",
+ card->debugfs_card_root);
+ if (!card->dapm.debugfs_dapm)
+ printk(KERN_WARNING
+ "Failed to create card DAPM debugfs directory\n");
+
+ snd_soc_dapm_debugfs_init(&card->dapm);
+#endif
+
snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname),
"%s", card->name);
snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
"%s", card->name);
+ if (card->late_probe) {
+ ret = card->late_probe(card);
+ if (ret < 0) {
+ dev_err(card->dev, "%s late_probe() failed: %d\n",
+ card->name, ret);
+ goto probe_aux_dev_err;
+ }
+ }
+
ret = snd_card_register(card->snd_card);
if (ret < 0) {
printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name);
@@ -2259,22 +2344,45 @@
* @_template: control template
* @data: control private data
* @long_name: control long name
+ * @prefix: control name prefix
*
* Create a new mixer control from a template control.
*
* Returns 0 for success, else error.
*/
struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
- void *data, char *long_name)
+ void *data, char *long_name,
+ const char *prefix)
{
struct snd_kcontrol_new template;
+ struct snd_kcontrol *kcontrol;
+ char *name = NULL;
+ int name_len;
memcpy(&template, _template, sizeof(template));
- if (long_name)
- template.name = long_name;
template.index = 0;
- return snd_ctl_new1(&template, data);
+ if (!long_name)
+ long_name = template.name;
+
+ if (prefix) {
+ name_len = strlen(long_name) + strlen(prefix) + 2;
+ name = kmalloc(name_len, GFP_ATOMIC);
+ if (!name)
+ return NULL;
+
+ snprintf(name, name_len, "%s %s", prefix, long_name);
+
+ template.name = name;
+ } else {
+ template.name = long_name;
+ }
+
+ kcontrol = snd_ctl_new1(&template, data);
+
+ kfree(name);
+
+ return kcontrol;
}
EXPORT_SYMBOL_GPL(snd_soc_cnew);
@@ -2293,22 +2401,16 @@
const struct snd_kcontrol_new *controls, int num_controls)
{
struct snd_card *card = codec->card->snd_card;
- char prefixed_name[44], *name;
int err, i;
for (i = 0; i < num_controls; i++) {
const struct snd_kcontrol_new *control = &controls[i];
- if (codec->name_prefix) {
- snprintf(prefixed_name, sizeof(prefixed_name), "%s %s",
- codec->name_prefix, control->name);
- name = prefixed_name;
- } else {
- name = control->name;
- }
- err = snd_ctl_add(card, snd_soc_cnew(control, codec, name));
+ err = snd_ctl_add(card, snd_soc_cnew(control, codec,
+ control->name,
+ codec->name_prefix));
if (err < 0) {
dev_err(codec->dev, "%s: Failed to add %s: %d\n",
- codec->name, name, err);
+ codec->name, control->name, err);
return err;
}
}
@@ -2989,12 +3091,34 @@
{
if (dai->driver && dai->driver->ops->set_sysclk)
return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir);
+ else if (dai->codec && dai->codec->driver->set_sysclk)
+ return dai->codec->driver->set_sysclk(dai->codec, clk_id,
+ freq, dir);
else
return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
/**
+ * snd_soc_codec_set_sysclk - configure CODEC system or master clock.
+ * @codec: CODEC
+ * @clk_id: DAI specific clock ID
+ * @freq: new clock frequency in Hz
+ * @dir: new clock direction - input/output.
+ *
+ * Configures the CODEC master (MCLK) or system (SYSCLK) clocking.
+ */
+int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+ unsigned int freq, int dir)
+{
+ if (codec->driver->set_sysclk)
+ return codec->driver->set_sysclk(codec, clk_id, freq, dir);
+ else
+ return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk);
+
+/**
* snd_soc_dai_set_clkdiv - configure DAI clock dividers.
* @dai: DAI
* @div_id: DAI specific clock divider ID
@@ -3030,11 +3154,35 @@
if (dai->driver && dai->driver->ops->set_pll)
return dai->driver->ops->set_pll(dai, pll_id, source,
freq_in, freq_out);
+ else if (dai->codec && dai->codec->driver->set_pll)
+ return dai->codec->driver->set_pll(dai->codec, pll_id, source,
+ freq_in, freq_out);
else
return -EINVAL;
}
EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
+/*
+ * snd_soc_codec_set_pll - configure codec PLL.
+ * @codec: CODEC
+ * @pll_id: DAI specific PLL ID
+ * @source: DAI specific source for the PLL
+ * @freq_in: PLL input clock frequency in Hz
+ * @freq_out: requested PLL output clock frequency in Hz
+ *
+ * Configures and enables PLL to generate output clock based on input clock.
+ */
+int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
+ unsigned int freq_in, unsigned int freq_out)
+{
+ if (codec->driver->set_pll)
+ return codec->driver->set_pll(codec, pll_id, source,
+ freq_in, freq_out);
+ else
+ return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll);
+
/**
* snd_soc_dai_set_fmt - configure DAI hardware audio format.
* @dai: DAI
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d0342aa..81c4052 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -32,6 +32,7 @@
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
+#include <linux/async.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/bitops.h>
@@ -125,17 +126,17 @@
/**
* snd_soc_dapm_set_bias_level - set the bias level for the system
- * @card: audio device
+ * @dapm: DAPM context
* @level: level to configure
*
* Configure the bias (power) levels for the SoC audio device.
*
* Returns 0 for success else error.
*/
-static int snd_soc_dapm_set_bias_level(struct snd_soc_card *card,
- struct snd_soc_dapm_context *dapm,
+static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
{
+ struct snd_soc_card *card = dapm->card;
int ret = 0;
switch (level) {
@@ -365,9 +366,20 @@
struct snd_soc_dapm_widget *w)
{
int i, ret = 0;
- size_t name_len;
+ size_t name_len, prefix_len;
struct snd_soc_dapm_path *path;
- struct snd_card *card = dapm->codec->card->snd_card;
+ struct snd_card *card = dapm->card->snd_card;
+ const char *prefix;
+
+ if (dapm->codec)
+ prefix = dapm->codec->name_prefix;
+ else
+ prefix = NULL;
+
+ if (prefix)
+ prefix_len = strlen(prefix) + 1;
+ else
+ prefix_len = 0;
/* add kcontrol */
for (i = 0; i < w->num_kcontrols; i++) {
@@ -396,8 +408,15 @@
switch (w->id) {
default:
+ /* The control will get a prefix from
+ * the control creation process but
+ * we're also using the same prefix
+ * for widgets so cut the prefix off
+ * the front of the widget name.
+ */
snprintf(path->long_name, name_len, "%s %s",
- w->name, w->kcontrols[i].name);
+ w->name + prefix_len,
+ w->kcontrols[i].name);
break;
case snd_soc_dapm_mixer_named_ctl:
snprintf(path->long_name, name_len, "%s",
@@ -408,7 +427,7 @@
path->long_name[name_len - 1] = '\0';
path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
- path->long_name);
+ path->long_name, prefix);
ret = snd_ctl_add(card, path->kcontrol);
if (ret < 0) {
dev_err(dapm->dev,
@@ -429,7 +448,9 @@
{
struct snd_soc_dapm_path *path = NULL;
struct snd_kcontrol *kcontrol;
- struct snd_card *card = dapm->codec->card->snd_card;
+ struct snd_card *card = dapm->card->snd_card;
+ const char *prefix;
+ size_t prefix_len;
int ret = 0;
if (!w->num_kcontrols) {
@@ -437,7 +458,22 @@
return -EINVAL;
}
- kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
+ if (dapm->codec)
+ prefix = dapm->codec->name_prefix;
+ else
+ prefix = NULL;
+
+ if (prefix)
+ prefix_len = strlen(prefix) + 1;
+ else
+ prefix_len = 0;
+
+ /* The control will get a prefix from the control creation
+ * process but we're also using the same prefix for widgets so
+ * cut the prefix off the front of the widget name.
+ */
+ kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name + prefix_len,
+ prefix);
ret = snd_ctl_add(card, kcontrol);
if (ret < 0)
@@ -479,7 +515,7 @@
*/
static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget)
{
- int level = snd_power_get_state(widget->dapm->codec->card->snd_card);
+ int level = snd_power_get_state(widget->dapm->card->snd_card);
switch (level) {
case SNDRV_CTL_POWER_D3hot:
@@ -712,7 +748,15 @@
!path->connected(path->source, path->sink))
continue;
- if (path->sink && path->sink->power_check &&
+ if (!path->sink)
+ continue;
+
+ if (path->sink->force) {
+ power = 1;
+ break;
+ }
+
+ if (path->sink->power_check &&
path->sink->power_check(path->sink)) {
power = 1;
break;
@@ -963,7 +1007,7 @@
}
if (!list_empty(&pending))
- dapm_seq_run_coalesced(dapm, &pending);
+ dapm_seq_run_coalesced(cur_dapm, &pending);
if (cur_dapm && cur_dapm->seq_notifier) {
for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++)
@@ -1006,7 +1050,62 @@
}
}
+/* Async callback run prior to DAPM sequences - brings to _PREPARE if
+ * they're changing state.
+ */
+static void dapm_pre_sequence_async(void *data, async_cookie_t cookie)
+{
+ struct snd_soc_dapm_context *d = data;
+ int ret;
+ if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) {
+ ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY);
+ if (ret != 0)
+ dev_err(d->dev,
+ "Failed to turn on bias: %d\n", ret);
+ }
+
+ /* If we're changing to all on or all off then prepare */
+ if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) ||
+ (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) {
+ ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE);
+ if (ret != 0)
+ dev_err(d->dev,
+ "Failed to prepare bias: %d\n", ret);
+ }
+}
+
+/* Async callback run prior to DAPM sequences - brings to their final
+ * state.
+ */
+static void dapm_post_sequence_async(void *data, async_cookie_t cookie)
+{
+ struct snd_soc_dapm_context *d = data;
+ int ret;
+
+ /* If we just powered the last thing off drop to standby bias */
+ if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) {
+ ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY);
+ if (ret != 0)
+ dev_err(d->dev, "Failed to apply standby bias: %d\n",
+ ret);
+ }
+
+ /* If we're in standby and can support bias off then do that */
+ if (d->bias_level == SND_SOC_BIAS_STANDBY && d->idle_bias_off) {
+ ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_OFF);
+ if (ret != 0)
+ dev_err(d->dev, "Failed to turn off bias: %d\n", ret);
+ }
+
+ /* If we just powered up then move to active bias */
+ if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) {
+ ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_ON);
+ if (ret != 0)
+ dev_err(d->dev, "Failed to apply active bias: %d\n",
+ ret);
+ }
+}
/*
* Scan each dapm widget for complete audio path.
@@ -1019,12 +1118,12 @@
*/
static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
{
- struct snd_soc_card *card = dapm->codec->card;
+ struct snd_soc_card *card = dapm->card;
struct snd_soc_dapm_widget *w;
struct snd_soc_dapm_context *d;
LIST_HEAD(up_list);
LIST_HEAD(down_list);
- int ret = 0;
+ LIST_HEAD(async_domain);
int power;
trace_snd_soc_dapm_start(card);
@@ -1102,25 +1201,11 @@
}
}
- list_for_each_entry(d, &dapm->card->dapm_list, list) {
- if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) {
- ret = snd_soc_dapm_set_bias_level(card, d,
- SND_SOC_BIAS_STANDBY);
- if (ret != 0)
- dev_err(d->dev,
- "Failed to turn on bias: %d\n", ret);
- }
-
- /* If we're changing to all on or all off then prepare */
- if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) ||
- (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) {
- ret = snd_soc_dapm_set_bias_level(card, d,
- SND_SOC_BIAS_PREPARE);
- if (ret != 0)
- dev_err(d->dev,
- "Failed to prepare bias: %d\n", ret);
- }
- }
+ /* Run all the bias changes in parallel */
+ list_for_each_entry(d, &dapm->card->dapm_list, list)
+ async_schedule_domain(dapm_pre_sequence_async, d,
+ &async_domain);
+ async_synchronize_full_domain(&async_domain);
/* Power down widgets first; try to avoid amplifying pops. */
dapm_seq_run(dapm, &down_list, event, false);
@@ -1130,37 +1215,11 @@
/* Now power up. */
dapm_seq_run(dapm, &up_list, event, true);
- list_for_each_entry(d, &dapm->card->dapm_list, list) {
- /* If we just powered the last thing off drop to standby bias */
- if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) {
- ret = snd_soc_dapm_set_bias_level(card, d,
- SND_SOC_BIAS_STANDBY);
- if (ret != 0)
- dev_err(d->dev,
- "Failed to apply standby bias: %d\n",
- ret);
- }
-
- /* If we're in standby and can support bias off then do that */
- if (d->bias_level == SND_SOC_BIAS_STANDBY &&
- d->idle_bias_off) {
- ret = snd_soc_dapm_set_bias_level(card, d,
- SND_SOC_BIAS_OFF);
- if (ret != 0)
- dev_err(d->dev,
- "Failed to turn off bias: %d\n", ret);
- }
-
- /* If we just powered up then move to active bias */
- if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) {
- ret = snd_soc_dapm_set_bias_level(card, d,
- SND_SOC_BIAS_ON);
- if (ret != 0)
- dev_err(d->dev,
- "Failed to apply active bias: %d\n",
- ret);
- }
- }
+ /* Run all the bias changes in parallel */
+ list_for_each_entry(d, &dapm->card->dapm_list, list)
+ async_schedule_domain(dapm_post_sequence_async, d,
+ &async_domain);
+ async_synchronize_full_domain(&async_domain);
pop_dbg(dapm->dev, card->pop_time,
"DAPM sequencing finished, waiting %dms\n", card->pop_time);
@@ -1218,7 +1277,7 @@
if (p->connect)
ret += snprintf(buf + ret, PAGE_SIZE - ret,
- " in %s %s\n",
+ " in \"%s\" \"%s\"\n",
p->name ? p->name : "static",
p->source->name);
}
@@ -1228,7 +1287,7 @@
if (p->connect)
ret += snprintf(buf + ret, PAGE_SIZE - ret,
- " out %s %s\n",
+ " out \"%s\" \"%s\"\n",
p->name ? p->name : "static",
p->sink->name);
}
@@ -1493,7 +1552,7 @@
char prefixed_source[80];
int ret = 0;
- if (dapm->codec->name_prefix) {
+ if (dapm->codec && dapm->codec->name_prefix) {
snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s",
dapm->codec->name_prefix, route->sink);
sink = prefixed_sink;
@@ -1664,6 +1723,7 @@
int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
{
struct snd_soc_dapm_widget *w;
+ unsigned int val;
list_for_each_entry(w, &dapm->card->widgets, list)
{
@@ -1712,6 +1772,18 @@
case snd_soc_dapm_post:
break;
}
+
+ /* Read the initial power state from the device */
+ if (w->reg >= 0) {
+ val = snd_soc_read(w->codec, w->reg);
+ val &= 1 << w->shift;
+ if (w->invert)
+ val = !val;
+
+ if (val)
+ w->power = 1;
+ }
+
w->new = 1;
}
@@ -2130,14 +2202,14 @@
return -ENOMEM;
name_len = strlen(widget->name) + 1;
- if (dapm->codec->name_prefix)
+ if (dapm->codec && dapm->codec->name_prefix)
name_len += 1 + strlen(dapm->codec->name_prefix);
w->name = kmalloc(name_len, GFP_KERNEL);
if (w->name == NULL) {
kfree(w);
return -ENOMEM;
}
- if (dapm->codec->name_prefix)
+ if (dapm->codec && dapm->codec->name_prefix)
snprintf(w->name, name_len, "%s %s",
dapm->codec->name_prefix, widget->name);
else
@@ -2242,7 +2314,6 @@
mutex_unlock(&codec->mutex);
return 0;
}
-EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
/**
* snd_soc_dapm_enable_pin - enable pin.
@@ -2419,9 +2490,9 @@
* standby.
*/
if (powerdown) {
- snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_PREPARE);
+ snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE);
dapm_seq_run(dapm, &down_list, 0, false);
- snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_STANDBY);
+ snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY);
}
}
@@ -2434,7 +2505,7 @@
list_for_each_entry(codec, &card->codec_dev_list, list) {
soc_dapm_shutdown_codec(&codec->dapm);
- snd_soc_dapm_set_bias_level(card, &codec->dapm, SND_SOC_BIAS_OFF);
+ snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF);
}
}
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index ac5a5bc..fcab80b 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -37,6 +37,7 @@
{
jack->codec = codec;
INIT_LIST_HEAD(&jack->pins);
+ INIT_LIST_HEAD(&jack->jack_zones);
BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier);
return snd_jack_new(codec->card->snd_card, id, type, &jack->jack);
@@ -100,7 +101,7 @@
}
/* Report before the DAPM sync to help users updating micbias status */
- blocking_notifier_call_chain(&jack->notifier, status, NULL);
+ blocking_notifier_call_chain(&jack->notifier, status, jack);
snd_soc_dapm_sync(dapm);
@@ -112,6 +113,51 @@
EXPORT_SYMBOL_GPL(snd_soc_jack_report);
/**
+ * snd_soc_jack_add_zones - Associate voltage zones with jack
+ *
+ * @jack: ASoC jack
+ * @count: Number of zones
+ * @zone: Array of zones
+ *
+ * After this function has been called the zones specified in the
+ * array will be associated with the jack.
+ */
+int snd_soc_jack_add_zones(struct snd_soc_jack *jack, int count,
+ struct snd_soc_jack_zone *zones)
+{
+ int i;
+
+ for (i = 0; i < count; i++) {
+ INIT_LIST_HEAD(&zones[i].list);
+ list_add(&(zones[i].list), &jack->jack_zones);
+ }
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_add_zones);
+
+/**
+ * snd_soc_jack_get_type - Based on the mic bias value, this function returns
+ * the type of jack from the zones delcared in the jack type
+ *
+ * @micbias_voltage: mic bias voltage at adc channel when jack is plugged in
+ *
+ * Based on the mic bias value passed, this function helps identify
+ * the type of jack from the already delcared jack zones
+ */
+int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage)
+{
+ struct snd_soc_jack_zone *zone;
+
+ list_for_each_entry(zone, &jack->jack_zones, list) {
+ if (micbias_voltage >= zone->min_mv &&
+ micbias_voltage < zone->max_mv)
+ return zone->jack_type;
+ }
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_get_type);
+
+/**
* snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack
*
* @jack: ASoC jack
@@ -194,7 +240,7 @@
int enable;
int report;
- enable = gpio_get_value(gpio->gpio);
+ enable = gpio_get_value_cansleep(gpio->gpio);
if (gpio->invert)
enable = !enable;
@@ -284,6 +330,14 @@
if (ret)
goto err;
+ if (gpios[i].wake) {
+ ret = set_irq_wake(gpio_to_irq(gpios[i].gpio), 1);
+ if (ret != 0)
+ printk(KERN_ERR
+ "Failed to mark GPIO %d as wake source: %d\n",
+ gpios[i].gpio, ret);
+ }
+
#ifdef CONFIG_GPIO_SYSFS
/* Expose GPIO value over sysfs for diagnostic purposes */
gpio_export(gpios[i].gpio, false);
diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile
index dfd2ab9..fd183d3 100644
--- a/sound/soc/tegra/Makefile
+++ b/sound/soc/tegra/Makefile
@@ -2,13 +2,14 @@
snd-soc-tegra-das-objs := tegra_das.o
snd-soc-tegra-pcm-objs := tegra_pcm.o
snd-soc-tegra-i2s-objs := tegra_i2s.o
+snd-soc-tegra-utils-objs += tegra_asoc_utils.o
+obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-utils.o
obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-das.o
obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-pcm.o
obj-$(CONFIG_SND_TEGRA_SOC_I2S) += snd-soc-tegra-i2s.o
# Tegra machine Support
snd-soc-tegra-harmony-objs := harmony.o
-snd-soc-tegra-harmony-objs += tegra_asoc_utils.o
obj-$(CONFIG_SND_TEGRA_SOC_HARMONY) += snd-soc-tegra-harmony.o
diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c
index 11e2cb8..8585957 100644
--- a/sound/soc/tegra/harmony.c
+++ b/sound/soc/tegra/harmony.c
@@ -38,10 +38,13 @@
#include <mach/harmony_audio.h>
#include <sound/core.h>
+#include <sound/jack.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
+#include "../codecs/wm8903.h"
+
#include "tegra_das.h"
#include "tegra_i2s.h"
#include "tegra_pcm.h"
@@ -49,10 +52,14 @@
#define DRV_NAME "tegra-snd-harmony"
+#define GPIO_SPKR_EN BIT(0)
+#define GPIO_INT_MIC_EN BIT(1)
+#define GPIO_EXT_MIC_EN BIT(2)
+
struct tegra_harmony {
struct tegra_asoc_utils_data util_data;
struct harmony_audio_platform_data *pdata;
- int gpio_spkr_en_requested;
+ int gpio_requested;
};
static int harmony_asoc_hw_params(struct snd_pcm_substream *substream,
@@ -123,6 +130,33 @@
.hw_params = harmony_asoc_hw_params,
};
+static struct snd_soc_jack harmony_hp_jack;
+
+static struct snd_soc_jack_pin harmony_hp_jack_pins[] = {
+ {
+ .pin = "Headphone Jack",
+ .mask = SND_JACK_HEADPHONE,
+ },
+};
+
+static struct snd_soc_jack_gpio harmony_hp_jack_gpios[] = {
+ {
+ .name = "headphone detect",
+ .report = SND_JACK_HEADPHONE,
+ .debounce_time = 150,
+ .invert = 1,
+ }
+};
+
+static struct snd_soc_jack harmony_mic_jack;
+
+static struct snd_soc_jack_pin harmony_mic_jack_pins[] = {
+ {
+ .pin = "Mic Jack",
+ .mask = SND_JACK_MICROPHONE,
+ },
+};
+
static int harmony_event_int_spk(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
@@ -154,6 +188,10 @@
{"IN1L", NULL, "Mic Bias"},
};
+static const struct snd_kcontrol_new harmony_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Int Spk"),
+};
+
static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
@@ -168,19 +206,65 @@
dev_err(card->dev, "cannot get spkr_en gpio\n");
return ret;
}
- harmony->gpio_spkr_en_requested = 1;
+ harmony->gpio_requested |= GPIO_SPKR_EN;
gpio_direction_output(pdata->gpio_spkr_en, 0);
+ ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en");
+ if (ret) {
+ dev_err(card->dev, "cannot get int_mic_en gpio\n");
+ return ret;
+ }
+ harmony->gpio_requested |= GPIO_INT_MIC_EN;
+
+ /* Disable int mic; enable signal is active-high */
+ gpio_direction_output(pdata->gpio_int_mic_en, 0);
+
+ ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en");
+ if (ret) {
+ dev_err(card->dev, "cannot get ext_mic_en gpio\n");
+ return ret;
+ }
+ harmony->gpio_requested |= GPIO_EXT_MIC_EN;
+
+ /* Enable ext mic; enable signal is active-low */
+ gpio_direction_output(pdata->gpio_ext_mic_en, 0);
+
+ ret = snd_soc_add_controls(codec, harmony_controls,
+ ARRAY_SIZE(harmony_controls));
+ if (ret < 0)
+ return ret;
+
snd_soc_dapm_new_controls(dapm, harmony_dapm_widgets,
ARRAY_SIZE(harmony_dapm_widgets));
snd_soc_dapm_add_routes(dapm, harmony_audio_map,
ARRAY_SIZE(harmony_audio_map));
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
- snd_soc_dapm_enable_pin(dapm, "Int Spk");
- snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+ harmony_hp_jack_gpios[0].gpio = pdata->gpio_hp_det;
+ snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
+ &harmony_hp_jack);
+ snd_soc_jack_add_pins(&harmony_hp_jack,
+ ARRAY_SIZE(harmony_hp_jack_pins),
+ harmony_hp_jack_pins);
+ snd_soc_jack_add_gpios(&harmony_hp_jack,
+ ARRAY_SIZE(harmony_hp_jack_gpios),
+ harmony_hp_jack_gpios);
+
+ snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE,
+ &harmony_mic_jack);
+ snd_soc_jack_add_pins(&harmony_mic_jack,
+ ARRAY_SIZE(harmony_mic_jack_pins),
+ harmony_mic_jack_pins);
+ wm8903_mic_detect(codec, &harmony_mic_jack, SND_JACK_MICROPHONE, 0);
+
+ snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
+
+ snd_soc_dapm_nc_pin(dapm, "IN3L");
+ snd_soc_dapm_nc_pin(dapm, "IN3R");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUTL");
+ snd_soc_dapm_nc_pin(dapm, "LINEOUTR");
+
snd_soc_dapm_sync(dapm);
return 0;
@@ -189,7 +273,7 @@
static struct snd_soc_dai_link harmony_wm8903_dai = {
.name = "WM8903",
.stream_name = "WM8903 PCM",
- .codec_name = "wm8903-codec.0-001a",
+ .codec_name = "wm8903.0-001a",
.platform_name = "tegra-pcm-audio",
.cpu_dai_name = "tegra-i2s.0",
.codec_dai_name = "wm8903-hifi",
@@ -270,7 +354,11 @@
tegra_asoc_utils_fini(&harmony->util_data);
- if (harmony->gpio_spkr_en_requested)
+ if (harmony->gpio_requested & GPIO_EXT_MIC_EN)
+ gpio_free(pdata->gpio_ext_mic_en);
+ if (harmony->gpio_requested & GPIO_INT_MIC_EN)
+ gpio_free(pdata->gpio_int_mic_en);
+ if (harmony->gpio_requested & GPIO_SPKR_EN)
gpio_free(pdata->gpio_spkr_en);
kfree(harmony);
@@ -302,3 +390,4 @@
MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
MODULE_DESCRIPTION("Harmony machine ASoC driver");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c
index cb4fc13..52f0a3f 100644
--- a/sound/soc/tegra/tegra_asoc_utils.c
+++ b/sound/soc/tegra/tegra_asoc_utils.c
@@ -101,6 +101,7 @@
return 0;
}
+EXPORT_SYMBOL_GPL(tegra_asoc_utils_set_rate);
int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data,
struct device *dev)
@@ -139,6 +140,7 @@
err:
return ret;
}
+EXPORT_SYMBOL_GPL(tegra_asoc_utils_init);
void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data)
{
@@ -146,4 +148,8 @@
clk_put(data->clk_pll_a_out0);
clk_put(data->clk_pll_a);
}
+EXPORT_SYMBOL_GPL(tegra_asoc_utils_fini);
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Tegra ASoC utility code");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c
index 01eb9c9..9f24ef7 100644
--- a/sound/soc/tegra/tegra_das.c
+++ b/sound/soc/tegra/tegra_das.c
@@ -262,3 +262,4 @@
MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
MODULE_DESCRIPTION("Tegra DAS driver");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 870ee36..4f5e2c9 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -500,3 +500,4 @@
MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
MODULE_DESCRIPTION("Tegra I2S ASoC driver");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 663ea9f..3c271f9 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -39,6 +39,8 @@
#include "tegra_pcm.h"
+#define DRV_NAME "tegra-pcm-audio"
+
static const struct snd_pcm_hardware tegra_pcm_hardware = {
.info = SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_MMAP_VALID |
@@ -159,8 +161,8 @@
prtd->dma_req[1].dev = prtd;
prtd->dma_chan = tegra_dma_allocate_channel(TEGRA_DMA_MODE_ONESHOT);
- if (IS_ERR(prtd->dma_chan)) {
- ret = PTR_ERR(prtd->dma_chan);
+ if (prtd->dma_chan == NULL) {
+ ret = -ENOMEM;
goto err;
}
@@ -377,7 +379,7 @@
static struct platform_driver tegra_pcm_driver = {
.driver = {
- .name = "tegra-pcm-audio",
+ .name = DRV_NAME,
.owner = THIS_MODULE,
},
.probe = tegra_pcm_platform_probe,
@@ -399,3 +401,4 @@
MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
MODULE_DESCRIPTION("Tegra PCM ASoC driver");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);