Merge tag 'asoc-v5.9' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for v5.9

The biggest changes here one again come from Mormioto-san who has
continued his dilligent work cleaning up long standing issues in the
APIs, it's particularly nice to see the transition from digital_mute()
to mute_stream() finally completed. There's also been a lot of work on
the x86 code again, this time a big focus has been on cleaning up some
issues identified by various static tests, and on the Freescale systems.
Otherwise the biggest thing has been a lot of driver additions:

 - Convert users of digital_mute() to mute_stream().
 - Simplify I/O helper functions.
 - Add a helper for getting the RTD from a substream.
 - Many, many fixes and cleanups to the x86 code.
 - New drivers for Freescale MQS and i.MX6sx, Intel KeemBay I2S, Maxim
   MAX98360A and MAX98373 Soundwire, several Mediatek boards, nVidia
   Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries boards (some
   of the first phones I worked on!) and TI J721e EVM.

10 files changed, 83 insertions, 73 deletions

diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h
index f4cc364d837f..0fea49bfc5e8 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 cc0b29bbcde3..7754631a3102 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