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-rw-r--r--include/dt-bindings/sound/qcom,q6asm.h4
-rw-r--r--include/sound/control.h45
-rw-r--r--include/sound/gus.h4
-rw-r--r--include/sound/hda_codec.h4
-rw-r--r--include/sound/hdaudio.h3
-rw-r--r--include/sound/hdmi-codec.h8
-rw-r--r--include/sound/memalloc.h9
-rw-r--r--include/sound/omap-hdmi-audio.h2
-rw-r--r--include/sound/rt5670.h26
-rw-r--r--include/sound/simple_card_utils.h6
-rw-r--r--include/sound/soc-component.h30
-rw-r--r--include/sound/soc-dai.h14
-rw-r--r--include/sound/soc-dapm.h20
-rw-r--r--include/sound/soc-link.h1
-rw-r--r--include/sound/soc.h34
-rw-r--r--include/sound/wm8960.h17
16 files changed, 128 insertions, 99 deletions
diff --git a/include/dt-bindings/sound/qcom,q6asm.h b/include/dt-bindings/sound/qcom,q6asm.h
index 1eb77d87c2e8..f59d74f14395 100644
--- a/include/dt-bindings/sound/qcom,q6asm.h
+++ b/include/dt-bindings/sound/qcom,q6asm.h
@@ -19,4 +19,8 @@
#define MSM_FRONTEND_DAI_MULTIMEDIA15 14
#define MSM_FRONTEND_DAI_MULTIMEDIA16 15
+#define Q6ASM_DAI_TX_RX 0
+#define Q6ASM_DAI_TX 1
+#define Q6ASM_DAI_RX 2
+
#endif /* __DT_BINDINGS_Q6_ASM_H__ */
diff --git a/include/sound/control.h b/include/sound/control.h
index aeaed2a05bae..e128cff10dfa 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -188,20 +188,21 @@ int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels,
*/
struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
const unsigned int *tlv);
-int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
- unsigned int flags);
-/* optional flags for slave */
-#define SND_CTL_SLAVE_NEED_UPDATE (1 << 0)
+int _snd_ctl_add_follower(struct snd_kcontrol *master,
+ struct snd_kcontrol *follower,
+ unsigned int flags);
+/* optional flags for follower */
+#define SND_CTL_FOLLOWER_NEED_UPDATE (1 << 0)
/**
- * snd_ctl_add_slave - Add a virtual slave control
+ * snd_ctl_add_follower - Add a virtual follower control
* @master: vmaster element
- * @slave: slave element to add
+ * @follower: follower element to add
*
- * Add a virtual slave control to the given master element created via
+ * Add a virtual follower control to the given master element created via
* snd_ctl_create_virtual_master() beforehand.
*
- * All slaves must be the same type (returning the same information
+ * All followers must be the same type (returning the same information
* via info callback). The function doesn't check it, so it's your
* responsibility.
*
@@ -213,18 +214,18 @@ int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
* Return: Zero if successful or a negative error code.
*/
static inline int
-snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
+snd_ctl_add_follower(struct snd_kcontrol *master, struct snd_kcontrol *follower)
{
- return _snd_ctl_add_slave(master, slave, 0);
+ return _snd_ctl_add_follower(master, follower, 0);
}
/**
- * snd_ctl_add_slave_uncached - Add a virtual slave control
+ * snd_ctl_add_follower_uncached - Add a virtual follower control
* @master: vmaster element
- * @slave: slave element to add
+ * @follower: follower element to add
*
- * Add a virtual slave control to the given master.
- * Unlike snd_ctl_add_slave(), the element added via this function
+ * Add a virtual follower control to the given master.
+ * Unlike snd_ctl_add_follower(), the element added via this function
* is supposed to have volatile values, and get callback is called
* at each time queried from the master.
*
@@ -235,10 +236,10 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
* Return: Zero if successful or a negative error code.
*/
static inline int
-snd_ctl_add_slave_uncached(struct snd_kcontrol *master,
- struct snd_kcontrol *slave)
+snd_ctl_add_follower_uncached(struct snd_kcontrol *master,
+ struct snd_kcontrol *follower)
{
- return _snd_ctl_add_slave(master, slave, SND_CTL_SLAVE_NEED_UPDATE);
+ return _snd_ctl_add_follower(master, follower, SND_CTL_FOLLOWER_NEED_UPDATE);
}
int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kctl,
@@ -246,11 +247,11 @@ int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kctl,
void *private_data);
void snd_ctl_sync_vmaster(struct snd_kcontrol *kctl, bool hook_only);
#define snd_ctl_sync_vmaster_hook(kctl) snd_ctl_sync_vmaster(kctl, true)
-int snd_ctl_apply_vmaster_slaves(struct snd_kcontrol *kctl,
- int (*func)(struct snd_kcontrol *vslave,
- struct snd_kcontrol *slave,
- void *arg),
- void *arg);
+int snd_ctl_apply_vmaster_followers(struct snd_kcontrol *kctl,
+ int (*func)(struct snd_kcontrol *vfollower,
+ struct snd_kcontrol *follower,
+ void *arg),
+ void *arg);
/*
* Helper functions for jack-detection controls
diff --git a/include/sound/gus.h b/include/sound/gus.h
index 410939ecf3a5..cd8da68cab92 100644
--- a/include/sound/gus.h
+++ b/include/sound/gus.h
@@ -613,4 +613,8 @@ int snd_gus_dram_write(struct snd_gus_card *gus, char __user *ptr,
int snd_gus_dram_read(struct snd_gus_card *gus, char __user *ptr,
unsigned int addr, unsigned int size, int rom);
+/* gus_timer.c */
+void snd_gf1_timers_init(struct snd_gus_card *gus);
+void snd_gf1_timers_done(struct snd_gus_card *gus);
+
#endif /* __SOUND_GUS_H */
diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h
index d16a4229209b..0fea49bfc5e8 100644
--- a/include/sound/hda_codec.h
+++ b/include/sound/hda_codec.h
@@ -208,7 +208,7 @@ struct hda_codec {
struct mutex control_mutex;
struct snd_array spdif_out;
unsigned int spdif_in_enable; /* SPDIF input enable? */
- const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
+ const hda_nid_t *follower_dig_outs; /* optional digital out follower widgets */
struct snd_array init_pins; /* initial (BIOS) pin configurations */
struct snd_array driver_pins; /* pin configs set by codec parser */
struct snd_array cvt_setups; /* audio convert setups */
@@ -415,6 +415,8 @@ __printf(2, 3)
struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec,
const char *fmt, ...);
+void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec);
+
static inline void snd_hda_codec_pcm_get(struct hda_pcm *pcm)
{
kref_get(&pcm->kref);
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index c1f78d9a6e47..6eed61e6cf8a 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -347,6 +347,9 @@ struct hdac_bus {
int bdl_pos_adj; /* BDL position adjustment */
+ /* delay time in us for dma stop */
+ unsigned int dma_stop_delay;
+
/* locks */
spinlock_t reg_lock;
struct mutex cmd_mutex;
diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h
index 83b17682e01c..7754631a3102 100644
--- a/include/sound/hdmi-codec.h
+++ b/include/sound/hdmi-codec.h
@@ -2,7 +2,7 @@
/*
* hdmi-codec.h - HDMI Codec driver API
*
- * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com
*
* Author: Jyri Sarha <jsarha@ti.com>
*/
@@ -76,7 +76,8 @@ struct hdmi_codec_ops {
* Mute/unmute HDMI audio stream.
* Optional
*/
- int (*digital_mute)(struct device *dev, void *data, bool enable);
+ int (*mute_stream)(struct device *dev, void *data,
+ bool enable, int direction);
/*
* Provides EDID-Like-Data from connected HDMI device.
@@ -99,6 +100,9 @@ struct hdmi_codec_ops {
int (*hook_plugged_cb)(struct device *dev, void *data,
hdmi_codec_plugged_cb fn,
struct device *codec_dev);
+
+ /* bit field */
+ unsigned int no_capture_mute:1;
};
/* HDMI codec initalization data */
diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h
index 3b47832b1c1f..5daa937684a4 100644
--- a/include/sound/memalloc.h
+++ b/include/sound/memalloc.h
@@ -94,7 +94,11 @@ static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
size_t offset)
{
struct snd_sg_buf *sgbuf = dmab->private_data;
- dma_addr_t addr = sgbuf->table[offset >> PAGE_SHIFT].addr;
+ dma_addr_t addr;
+
+ if (!sgbuf)
+ return dmab->addr + offset;
+ addr = sgbuf->table[offset >> PAGE_SHIFT].addr;
addr &= ~((dma_addr_t)PAGE_SIZE - 1);
return addr + offset % PAGE_SIZE;
}
@@ -106,6 +110,9 @@ static inline void *snd_sgbuf_get_ptr(struct snd_dma_buffer *dmab,
size_t offset)
{
struct snd_sg_buf *sgbuf = dmab->private_data;
+
+ if (!sgbuf)
+ return dmab->area + offset;
return sgbuf->table[offset >> PAGE_SHIFT].buf + offset % PAGE_SIZE;
}
diff --git a/include/sound/omap-hdmi-audio.h b/include/sound/omap-hdmi-audio.h
index 16c007b651f4..e5f82044a404 100644
--- a/include/sound/omap-hdmi-audio.h
+++ b/include/sound/omap-hdmi-audio.h
@@ -2,7 +2,7 @@
/*
* hdmi-audio.c -- OMAP4+ DSS HDMI audio support library
*
- * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com
*
* Author: Jyri Sarha <jsarha@ti.com>
*/
diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h
deleted file mode 100644
index 02e1d7778354..000000000000
--- a/include/sound/rt5670.h
+++ /dev/null
@@ -1,26 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0-only */
-/*
- * linux/sound/rt5670.h -- Platform data for RT5670
- *
- * Copyright 2014 Realtek Microelectronics
- */
-
-#ifndef __LINUX_SND_RT5670_H
-#define __LINUX_SND_RT5670_H
-
-struct rt5670_platform_data {
- int jd_mode;
- bool in2_diff;
- bool dev_gpio;
- bool gpio1_is_ext_spk_en;
-
- bool dmic_en;
- unsigned int dmic1_data_pin;
- /* 0 = GPIO6; 1 = IN2P; 3 = GPIO7*/
- unsigned int dmic2_data_pin;
- /* 0 = GPIO8; 1 = IN3N; */
- unsigned int dmic3_data_pin;
- /* 0 = GPIO9; 1 = GPIO10; 2 = GPIO5*/
-};
-
-#endif
diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h
index bbdd1542d6f1..86a1e956991e 100644
--- a/include/sound/simple_card_utils.h
+++ b/include/sound/simple_card_utils.h
@@ -12,9 +12,9 @@
#include <sound/soc.h>
#define asoc_simple_init_hp(card, sjack, prefix) \
- asoc_simple_init_jack(card, sjack, 1, prefix)
+ asoc_simple_init_jack(card, sjack, 1, prefix, NULL)
#define asoc_simple_init_mic(card, sjack, prefix) \
- asoc_simple_init_jack(card, sjack, 0, prefix)
+ asoc_simple_init_jack(card, sjack, 0, prefix, NULL)
struct asoc_simple_dai {
const char *name;
@@ -131,7 +131,7 @@ int asoc_simple_parse_pin_switches(struct snd_soc_card *card,
int asoc_simple_init_jack(struct snd_soc_card *card,
struct asoc_simple_jack *sjack,
- int is_hp, char *prefix);
+ int is_hp, char *prefix, char *pin);
int asoc_simple_init_priv(struct asoc_simple_priv *priv,
struct link_info *li);
diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 5663891148e3..089ea9441fd1 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -2,7 +2,8 @@
*
* soc-component.h
*
- * Copyright (c) 2019 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ * Copyright (C) 2019 Renesas Electronics Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
*/
#ifndef __SOC_COMPONENT_H
#define __SOC_COMPONENT_H
@@ -324,10 +325,12 @@ static inline int snd_soc_component_cache_sync(
return regcache_sync(component->regmap);
}
+void snd_soc_component_set_aux(struct snd_soc_component *component,
+ struct snd_soc_aux_dev *aux);
+int snd_soc_component_init(struct snd_soc_component *component);
+
/* component IO */
-int snd_soc_component_read(struct snd_soc_component *component,
- unsigned int reg, unsigned int *val);
-unsigned int snd_soc_component_read32(struct snd_soc_component *component,
+unsigned int snd_soc_component_read(struct snd_soc_component *component,
unsigned int reg);
int snd_soc_component_write(struct snd_soc_component *component,
unsigned int reg, unsigned int val);
@@ -359,6 +362,7 @@ int snd_soc_component_stream_event(struct snd_soc_component *component,
int snd_soc_component_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level);
+void snd_soc_component_setup_regmap(struct snd_soc_component *component);
#ifdef CONFIG_REGMAP
void snd_soc_component_init_regmap(struct snd_soc_component *component,
struct regmap *regmap);
@@ -421,16 +425,6 @@ int snd_soc_component_open(struct snd_soc_component *component,
struct snd_pcm_substream *substream);
int snd_soc_component_close(struct snd_soc_component *component,
struct snd_pcm_substream *substream);
-int snd_soc_component_prepare(struct snd_soc_component *component,
- struct snd_pcm_substream *substream);
-int snd_soc_component_hw_params(struct snd_soc_component *component,
- struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params);
-int snd_soc_component_hw_free(struct snd_soc_component *component,
- struct snd_pcm_substream *substream);
-int snd_soc_component_trigger(struct snd_soc_component *component,
- struct snd_pcm_substream *substream,
- int cmd);
void snd_soc_component_suspend(struct snd_soc_component *component);
void snd_soc_component_resume(struct snd_soc_component *component);
int snd_soc_component_is_suspended(struct snd_soc_component *component);
@@ -455,5 +449,13 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream,
struct vm_area_struct *vma);
int snd_soc_pcm_component_new(struct snd_soc_pcm_runtime *rtd);
void snd_soc_pcm_component_free(struct snd_soc_pcm_runtime *rtd);
+int snd_soc_pcm_component_prepare(struct snd_pcm_substream *substream);
+int snd_soc_pcm_component_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_component **last);
+void snd_soc_pcm_component_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_component *last);
+int snd_soc_pcm_component_trigger(struct snd_pcm_substream *substream,
+ int cmd);
#endif /* __SOC_COMPONENT_H */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 71e178c89793..776a60529e70 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -39,7 +39,7 @@ struct snd_compr_stream;
/*
* DAI Clock gating.
*
- * DAI bit clocks can be be gated (disabled) when the DAI is not
+ * DAI bit clocks can be gated (disabled) when the DAI is not
* sending or receiving PCM data in a frame. This can be used to save power.
*/
#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */
@@ -76,12 +76,12 @@ struct snd_compr_stream;
*
* This is wrt the codec, the inverse is true for the interface
* i.e. if the codec is clk and FRM master then the interface is
- * clk and frame slave.
+ * clk and frame secondary.
*/
#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */
-#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */
-#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */
-#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */
+#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk secondary & FRM master */
+#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame secondary */
+#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM secondary */
#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
@@ -247,7 +247,6 @@ struct snd_soc_dai_ops {
* DAI digital mute - optional.
* Called by soc-core to minimise any pops.
*/
- int (*digital_mute)(struct snd_soc_dai *dai, int mute);
int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
/*
@@ -281,6 +280,9 @@ struct snd_soc_dai_ops {
*/
snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
struct snd_soc_dai *);
+
+ /* bit field */
+ unsigned int no_capture_mute:1;
};
struct snd_soc_cdai_ops {
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index cc3dcb815282..c3039e97929a 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -16,6 +16,8 @@
#include <sound/asoc.h>
struct device;
+struct snd_soc_pcm_runtime;
+struct soc_enum;
/* widget has no PM register bit */
#define SND_SOC_NOPM -1
@@ -376,6 +378,24 @@ struct snd_soc_dapm_widget_list;
struct snd_soc_dapm_update;
enum snd_soc_dapm_direction;
+/*
+ * Bias levels
+ *
+ * @ON: Bias is fully on for audio playback and capture operations.
+ * @PREPARE: Prepare for audio operations. Called before DAPM switching for
+ * stream start and stop operations.
+ * @STANDBY: Low power standby state when no playback/capture operations are
+ * in progress. NOTE: The transition time between STANDBY and ON
+ * should be as fast as possible and no longer than 10ms.
+ * @OFF: Power Off. No restrictions on transition times.
+ */
+enum snd_soc_bias_level {
+ SND_SOC_BIAS_OFF = 0,
+ SND_SOC_BIAS_STANDBY = 1,
+ SND_SOC_BIAS_PREPARE = 2,
+ SND_SOC_BIAS_ON = 3,
+};
+
int dapm_regulator_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event);
int dapm_clock_event(struct snd_soc_dapm_widget *w,
diff --git a/include/sound/soc-link.h b/include/sound/soc-link.h
index 3dd6e33e94ec..337ac5666757 100644
--- a/include/sound/soc-link.h
+++ b/include/sound/soc-link.h
@@ -9,6 +9,7 @@
#define __SOC_LINK_H
int snd_soc_link_init(struct snd_soc_pcm_runtime *rtd);
+void snd_soc_link_exit(struct snd_soc_pcm_runtime *rtd);
int snd_soc_link_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params);
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 3ce7f0f5aa92..5e3919ffb00c 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -368,24 +368,6 @@
#define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \
const struct soc_enum name = SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(xtexts), xtexts)
-/*
- * Bias levels
- *
- * @ON: Bias is fully on for audio playback and capture operations.
- * @PREPARE: Prepare for audio operations. Called before DAPM switching for
- * stream start and stop operations.
- * @STANDBY: Low power standby state when no playback/capture operations are
- * in progress. NOTE: The transition time between STANDBY and ON
- * should be as fast as possible and no longer than 10ms.
- * @OFF: Power Off. No restrictions on transition times.
- */
-enum snd_soc_bias_level {
- SND_SOC_BIAS_OFF = 0,
- SND_SOC_BIAS_STANDBY = 1,
- SND_SOC_BIAS_PREPARE = 2,
- SND_SOC_BIAS_ON = 3,
-};
-
struct device_node;
struct snd_jack;
struct snd_soc_card;
@@ -432,11 +414,12 @@ static inline int snd_soc_resume(struct device *dev)
}
#endif
int snd_soc_poweroff(struct device *dev);
-int snd_soc_add_component(struct device *dev,
- struct snd_soc_component *component,
- const struct snd_soc_component_driver *component_driver,
- struct snd_soc_dai_driver *dai_drv,
- int num_dai);
+int snd_soc_component_initialize(struct snd_soc_component *component,
+ const struct snd_soc_component_driver *driver,
+ struct device *dev);
+int snd_soc_add_component(struct snd_soc_component *component,
+ struct snd_soc_dai_driver *dai_drv,
+ int num_dai);
int snd_soc_register_component(struct device *dev,
const struct snd_soc_component_driver *component_driver,
struct snd_soc_dai_driver *dai_drv, int num_dai);
@@ -801,6 +784,9 @@ struct snd_soc_dai_link {
/* codec/machine specific init - e.g. add machine controls */
int (*init)(struct snd_soc_pcm_runtime *rtd);
+ /* codec/machine specific exit - dual of init() */
+ void (*exit)(struct snd_soc_pcm_runtime *rtd);
+
/* optional hw_params re-writing for BE and FE sync */
int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params);
@@ -1183,6 +1169,8 @@ struct snd_soc_pcm_runtime {
/* see soc_new_pcm_runtime() */
#define asoc_rtd_to_cpu(rtd, n) (rtd)->dais[n]
#define asoc_rtd_to_codec(rtd, n) (rtd)->dais[n + (rtd)->num_cpus]
+#define asoc_substream_to_rtd(substream) \
+ (struct snd_soc_pcm_runtime *)snd_pcm_substream_chip(substream)
#define for_each_rtd_components(rtd, i, component) \
for ((i) = 0, component = NULL; \
diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h
index d22e84805025..275fd5b201ce 100644
--- a/include/sound/wm8960.h
+++ b/include/sound/wm8960.h
@@ -16,6 +16,23 @@ struct wm8960_data {
bool capless; /* Headphone outputs configured in capless mode */
bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */
+
+ /*
+ * Setup for headphone detection
+ *
+ * hp_cfg[0]: HPSEL[1:0] of R48 (Additional Control 4)
+ * hp_cfg[1]: {HPSWEN:HPSWPOL} of R24 (Additional Control 2).
+ * hp_cfg[2]: {TOCLKSEL:TOEN} of R23 (Additional Control 1).
+ */
+ u32 hp_cfg[3];
+
+ /*
+ * Setup for gpio configuration
+ *
+ * gpio_cfg[0]: ALRCGPIO of R9 (Audio interface)
+ * gpio_cfg[1]: {GPIOPOL:GPIOSEL[2:0]} of R48 (Additional Control 4).
+ */
+ u32 gpio_cfg[2];
};
#endif
generated by cgit v1.2.3 (git 2.39.0) at 2024-12-27 14:33:49 +0300