Merge remote-tracking branch 'asoc/for-5.9' into asoc-next

11 files changed, 87 insertions, 73 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/hda_codec.h b/include/sound/hda_codec.h
index d16a4229209b..e378ed7f4824 100644
--- a/include/sound/hda_codec.h
+++ b/include/sound/hda_codec.h
@@ -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/hdmi-codec.h b/include/sound/hdmi-codec.h
index 83b17682e01c..17eebd34835a 100644
--- a/include/sound/hdmi-codec.h
+++ b/include/sound/hdmi-codec.h
@@ -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/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