Merge tag 'asoc-v4.5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next

ASoC: Updates for v4.5

This is quite a busy release on the driver front with a lot of new
drivers being added but comparatively quiet on the core side with only
one big change going in and that a fairly straightforward refactoring.

 - Conversion of the array of DAI links to a list by Mengdong Lin,
   supporting dynamically adding and removing DAI links.
 - Some more fixes for the topology code, though it is still not final
   and ready for enabling in production.  We really need to get to the
   point where that can be done.
 - A pile of changes for Intel SkyLake drivers which hopefully deliver
   some useful initial functionality for systems with this chipset,
   though there is more work still to come.
 - New drivers for a number of Imagination Technologies IPs.
 - Lots of new features and cleanups for the Renesas drivers.
 - ANC support for WM5110.
 - New driver for Atmel class D speaker drivers.
 - New drivers for Cirrus CS47L24 and WM1831.
 - New driver for Dialog DA7128.
 - New drivers for Realtek RT5659 and RT56156.
 - New driver for Rockchip RK3036.
 - New driver for TI PC3168A

12 files changed, 231 insertions, 22 deletions

diff --git a/include/linux/platform_data/asoc-s3c.h b/include/linux/platform_data/asoc-s3c.h
index 5e0bc77..15bf56e 100644
--- a/include/linux/platform_data/asoc-s3c.h
+++ b/include/linux/platform_data/asoc-s3c.h
@@ -13,6 +13,9 @@
  */
 #define S3C64XX_AC97_GPD  0
 #define S3C64XX_AC97_GPE  1
+
+#include <linux/dmaengine.h>
+
 extern void s3c64xx_ac97_setup_gpio(int);
 
 struct samsung_i2s {
@@ -39,6 +42,11 @@ struct samsung_i2s {
  */
 struct s3c_audio_pdata {
 	int (*cfg_gpio)(struct platform_device *);
+	dma_filter_fn dma_filter;
+	void *dma_playback;
+	void *dma_capture;
+	void *dma_play_sec;
+	void *dma_capture_mic;
 	union {
 		struct samsung_i2s i2s;
 	} type;
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 74bc8547..15aa5f0 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -417,11 +417,13 @@
 #define AC97_RATES_MIC_ADC	4
 #define AC97_RATES_SPDIF	5
 
+#define AC97_NUM_GPIOS		16
 /*
  *
  */
 
 struct snd_ac97;
+struct snd_ac97_gpio_priv;
 struct snd_pcm_chmap;
 
 struct snd_ac97_build_ops {
@@ -529,6 +531,7 @@ struct snd_ac97 {
 	struct delayed_work power_work;
 #endif
 	struct device dev;
+	struct snd_ac97_gpio_priv *gpio_priv;
 
 	struct snd_pcm_chmap *chmaps[2]; /* channel-maps (optional) */
 };
diff --git a/include/sound/da7218.h b/include/sound/da7218.h
new file mode 100644
index 0000000..0dbb818
--- /dev/null
+++ b/include/sound/da7218.h
@@ -0,0 +1,109 @@
+/*
+ * da7218.h - DA7218 ASoC Codec Driver Platform Data
+ *
+ * Copyright (c) 2015 Dialog Semiconductor
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.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.
+ */
+
+#ifndef _DA7218_PDATA_H
+#define _DA7218_PDATA_H
+
+/* Mic Bias */
+enum da7218_micbias_voltage {
+	DA7218_MICBIAS_1_2V = -1,
+	DA7218_MICBIAS_1_6V,
+	DA7218_MICBIAS_1_8V,
+	DA7218_MICBIAS_2_0V,
+	DA7218_MICBIAS_2_2V,
+	DA7218_MICBIAS_2_4V,
+	DA7218_MICBIAS_2_6V,
+	DA7218_MICBIAS_2_8V,
+	DA7218_MICBIAS_3_0V,
+};
+
+enum da7218_mic_amp_in_sel {
+	DA7218_MIC_AMP_IN_SEL_DIFF = 0,
+	DA7218_MIC_AMP_IN_SEL_SE_P,
+	DA7218_MIC_AMP_IN_SEL_SE_N,
+};
+
+/* DMIC */
+enum da7218_dmic_data_sel {
+	DA7218_DMIC_DATA_LRISE_RFALL = 0,
+	DA7218_DMIC_DATA_LFALL_RRISE,
+};
+
+enum da7218_dmic_samplephase {
+	DA7218_DMIC_SAMPLE_ON_CLKEDGE = 0,
+	DA7218_DMIC_SAMPLE_BETWEEN_CLKEDGE,
+};
+
+enum da7218_dmic_clk_rate {
+	DA7218_DMIC_CLK_3_0MHZ = 0,
+	DA7218_DMIC_CLK_1_5MHZ,
+};
+
+/* Headphone Detect */
+enum da7218_hpldet_jack_rate {
+	DA7218_HPLDET_JACK_RATE_5US = 0,
+	DA7218_HPLDET_JACK_RATE_10US,
+	DA7218_HPLDET_JACK_RATE_20US,
+	DA7218_HPLDET_JACK_RATE_40US,
+	DA7218_HPLDET_JACK_RATE_80US,
+	DA7218_HPLDET_JACK_RATE_160US,
+	DA7218_HPLDET_JACK_RATE_320US,
+	DA7218_HPLDET_JACK_RATE_640US,
+};
+
+enum da7218_hpldet_jack_debounce {
+	DA7218_HPLDET_JACK_DEBOUNCE_OFF = 0,
+	DA7218_HPLDET_JACK_DEBOUNCE_2,
+	DA7218_HPLDET_JACK_DEBOUNCE_3,
+	DA7218_HPLDET_JACK_DEBOUNCE_4,
+};
+
+enum da7218_hpldet_jack_thr {
+	DA7218_HPLDET_JACK_THR_84PCT = 0,
+	DA7218_HPLDET_JACK_THR_88PCT,
+	DA7218_HPLDET_JACK_THR_92PCT,
+	DA7218_HPLDET_JACK_THR_96PCT,
+};
+
+struct da7218_hpldet_pdata {
+	enum da7218_hpldet_jack_rate jack_rate;
+	enum da7218_hpldet_jack_debounce jack_debounce;
+	enum da7218_hpldet_jack_thr jack_thr;
+	bool comp_inv;
+	bool hyst;
+	bool discharge;
+};
+
+struct da7218_pdata {
+	/* Mic */
+	enum da7218_micbias_voltage micbias1_lvl;
+	enum da7218_micbias_voltage micbias2_lvl;
+	enum da7218_mic_amp_in_sel mic1_amp_in_sel;
+	enum da7218_mic_amp_in_sel mic2_amp_in_sel;
+
+	/* DMIC */
+	enum da7218_dmic_data_sel dmic1_data_sel;
+	enum da7218_dmic_data_sel dmic2_data_sel;
+	enum da7218_dmic_samplephase dmic1_samplephase;
+	enum da7218_dmic_samplephase dmic2_samplephase;
+	enum da7218_dmic_clk_rate dmic1_clk_rate;
+	enum da7218_dmic_clk_rate dmic2_clk_rate;
+
+	/* HP Diff Supply - DA7217 only */
+	bool hp_diff_single_supply;
+
+	/* HP Detect - DA7218 only */
+	struct da7218_hpldet_pdata *hpldet_pdata;
+};
+
+#endif /* _DA7218_PDATA_H */
diff --git a/include/sound/da7219.h b/include/sound/da7219.h
index 3f39e13..02876ac 100644
--- a/include/sound/da7219.h
+++ b/include/sound/da7219.h
@@ -14,17 +14,10 @@
 #ifndef __DA7219_PDATA_H
 #define __DA7219_PDATA_H
 
-/* LDO */
-enum da7219_ldo_lvl_sel {
-	DA7219_LDO_LVL_SEL_1_05V = 0,
-	DA7219_LDO_LVL_SEL_1_10V,
-	DA7219_LDO_LVL_SEL_1_20V,
-	DA7219_LDO_LVL_SEL_1_40V,
-};
-
 /* Mic Bias */
 enum da7219_micbias_voltage {
-	DA7219_MICBIAS_1_8V = 1,
+	DA7219_MICBIAS_1_6V = 0,
+	DA7219_MICBIAS_1_8V,
 	DA7219_MICBIAS_2_0V,
 	DA7219_MICBIAS_2_2V,
 	DA7219_MICBIAS_2_4V,
@@ -41,9 +34,6 @@ enum da7219_mic_amp_in_sel {
 struct da7219_aad_pdata;
 
 struct da7219_pdata {
-	/* Internal LDO */
-	enum da7219_ldo_lvl_sel ldo_lvl_sel;
-
 	/* Mic */
 	enum da7219_micbias_voltage micbias_lvl;
 	enum da7219_mic_amp_in_sel mic_amp_in_sel;
diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h
index 8966ba7..e0bb458 100644
--- a/include/sound/designware_i2s.h
+++ b/include/sound/designware_i2s.h
@@ -45,6 +45,11 @@ struct i2s_platform_data {
 	u32 snd_fmts;
 	u32 snd_rates;
 
+	#define DW_I2S_QUIRK_COMP_REG_OFFSET	(1 << 0)
+	unsigned int quirks;
+	unsigned int i2s_reg_comp1;
+	unsigned int i2s_reg_comp2;
+
 	void *play_dma_data;
 	void *capture_dma_data;
 	bool (*filter)(struct dma_chan *chan, void *slave);
diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h
index a4cadd9..425af06 100644
--- a/include/sound/hdaudio_ext.h
+++ b/include/sound/hdaudio_ext.h
@@ -186,9 +186,15 @@ struct hdac_ext_device {
 	/* codec ops */
 	struct hdac_ext_codec_ops ops;
 
+	struct snd_card *card;
+	void *scodec;
 	void *private_data;
 };
 
+struct hdac_ext_dma_params {
+	u32 format;
+	u8 stream_tag;
+};
 #define to_ehdac_device(dev) (container_of((dev), \
 				 struct hdac_ext_device, hdac))
 /*
diff --git a/include/sound/rt5659.h b/include/sound/rt5659.h
new file mode 100644
index 0000000..656c4d5
--- /dev/null
+++ b/include/sound/rt5659.h
@@ -0,0 +1,49 @@
+/*
+ * linux/sound/rt5659.h -- Platform data for RT5659
+ *
+ * Copyright 2013 Realtek Microelectronics
+ *
+ * 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 __LINUX_SND_RT5659_H
+#define __LINUX_SND_RT5659_H
+
+enum rt5659_dmic1_data_pin {
+	RT5659_DMIC1_NULL,
+	RT5659_DMIC1_DATA_IN2N,
+	RT5659_DMIC1_DATA_GPIO5,
+	RT5659_DMIC1_DATA_GPIO9,
+	RT5659_DMIC1_DATA_GPIO11,
+};
+
+enum rt5659_dmic2_data_pin {
+	RT5659_DMIC2_NULL,
+	RT5659_DMIC2_DATA_IN2P,
+	RT5659_DMIC2_DATA_GPIO6,
+	RT5659_DMIC2_DATA_GPIO10,
+	RT5659_DMIC2_DATA_GPIO12,
+};
+
+enum rt5659_jd_src {
+	RT5659_JD_NULL,
+	RT5659_JD3,
+};
+
+struct rt5659_platform_data {
+	bool in1_diff;
+	bool in3_diff;
+	bool in4_diff;
+
+	int ldo1_en; /* GPIO for LDO1_EN */
+	int reset; /* GPIO for RESET */
+
+	enum rt5659_dmic1_data_pin dmic1_data_pin;
+	enum rt5659_dmic2_data_pin dmic2_data_pin;
+	enum rt5659_jd_src jd_src;
+};
+
+#endif
+
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 95a937e..9706946 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -49,6 +49,9 @@ struct device;
 #define SND_SOC_DAPM_SIGGEN(wname) \
 {	.id = snd_soc_dapm_siggen, .name = wname, .kcontrol_news = NULL, \
 	.num_kcontrols = 0, .reg = SND_SOC_NOPM }
+#define SND_SOC_DAPM_SINK(wname) \
+{	.id = snd_soc_dapm_sink, .name = wname, .kcontrol_news = NULL, \
+	.num_kcontrols = 0, .reg = SND_SOC_NOPM }
 #define SND_SOC_DAPM_INPUT(wname) \
 {	.id = snd_soc_dapm_input, .name = wname, .kcontrol_news = NULL, \
 	.num_kcontrols = 0, .reg = SND_SOC_NOPM }
@@ -485,6 +488,7 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_aif_in,		/* audio interface input */
 	snd_soc_dapm_aif_out,		/* audio interface output */
 	snd_soc_dapm_siggen,		/* signal generator */
+	snd_soc_dapm_sink,
 	snd_soc_dapm_dai_in,		/* link to DAI structure */
 	snd_soc_dapm_dai_out,
 	snd_soc_dapm_dai_link,		/* link between two DAI structures */
diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h
index 086cd7f..5b68e3f 100644
--- a/include/sound/soc-topology.h
+++ b/include/sound/soc-topology.h
@@ -92,8 +92,10 @@ struct snd_soc_tplg_kcontrol_ops {
 /* Bytes ext operations, for TLV byte controls */
 struct snd_soc_tplg_bytes_ext_ops {
 	u32 id;
-	int (*get)(unsigned int __user *bytes, unsigned int size);
-	int (*put)(const unsigned int __user *bytes, unsigned int size);
+	int (*get)(struct snd_kcontrol *kcontrol, unsigned int __user *bytes,
+							unsigned int size);
+	int (*put)(struct snd_kcontrol *kcontrol,
+			const unsigned int __user *bytes, unsigned int size);
 };
 
 /*
diff --git a/include/sound/soc.h b/include/sound/soc.h
index a8b4b9c..a2deeb9 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -110,6 +110,14 @@
 	.put = snd_soc_put_volsw, \
 	.private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \
 					  max, invert, 0) }
+#define SOC_DOUBLE_STS(xname, reg, shift_left, shift_right, max, invert) \
+{									\
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),		\
+	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,		\
+	.access = SNDRV_CTL_ELEM_ACCESS_READ |				\
+		SNDRV_CTL_ELEM_ACCESS_VOLATILE,				\
+	.private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right,	\
+					  max, invert, 0) }
 #define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
 	.info = snd_soc_info_volsw, \
@@ -293,6 +301,9 @@
 		{.base = xbase, .num_regs = xregs,	      \
 		 .mask = xmask }) }
 
+/*
+ * SND_SOC_BYTES_EXT is deprecated, please USE SND_SOC_BYTES_TLV instead
+ */
 #define SND_SOC_BYTES_EXT(xname, xcount, xhandler_get, xhandler_put) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_bytes_info_ext, \
@@ -1037,6 +1048,9 @@ struct snd_soc_dai_link {
 
 	/* pmdown_time is ignored at stop */
 	unsigned int ignore_pmdown_time:1;
+
+	struct list_head list; /* DAI link list of the soc card */
+	struct snd_soc_dobj dobj; /* For topology */
 };
 
 struct snd_soc_codec_conf {
@@ -1101,12 +1115,20 @@ struct snd_soc_card {
 				   struct snd_soc_dapm_context *dapm,
 				   enum snd_soc_bias_level level);
 
+	int (*add_dai_link)(struct snd_soc_card *,
+			    struct snd_soc_dai_link *link);
+	void (*remove_dai_link)(struct snd_soc_card *,
+			    struct snd_soc_dai_link *link);
+
 	long pmdown_time;
 
 	/* CPU <--> Codec DAI links  */
-	struct snd_soc_dai_link *dai_link;
-	int num_links;
-	struct snd_soc_pcm_runtime *rtd;
+	struct snd_soc_dai_link *dai_link;  /* predefined links only */
+	int num_links;  /* predefined links only */
+	struct list_head dai_link_list; /* all links */
+	int num_dai_links;
+
+	struct list_head rtd_list;
 	int num_rtd;
 
 	/* optional codec specific configuration */
@@ -1201,6 +1223,9 @@ struct snd_soc_pcm_runtime {
 	struct dentry *debugfs_dpcm_root;
 	struct dentry *debugfs_dpcm_state;
 #endif
+
+	unsigned int num; /* 0-based and monotonic increasing */
+	struct list_head list; /* rtd list of the soc card */
 };
 
 /* mixer control */
@@ -1225,8 +1250,10 @@ struct soc_bytes_ext {
 	struct snd_soc_dobj dobj;
 
 	/* used for TLV byte control */
-	int (*get)(unsigned int __user *bytes, unsigned int size);
-	int (*put)(const unsigned int __user *bytes, unsigned int size);
+	int (*get)(struct snd_kcontrol *kcontrol, unsigned int __user *bytes,
+			unsigned int size);
+	int (*put)(struct snd_kcontrol *kcontrol, const unsigned int __user *bytes,
+			unsigned int size);
 };
 
 /* multi register control */
@@ -1644,6 +1671,11 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev,
 				   struct device_node *of_node,
 				   struct snd_soc_dai_link *dai_link);
 
+int snd_soc_add_dai_link(struct snd_soc_card *card,
+				struct snd_soc_dai_link *dai_link);
+void snd_soc_remove_dai_link(struct snd_soc_card *card,
+			     struct snd_soc_dai_link *dai_link);
+
 #include <sound/soc-dai.h>
 
 #ifdef CONFIG_DEBUG_FS
diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h
index 26539a7..c4cc1e4 100644
--- a/include/uapi/sound/asoc.h
+++ b/include/uapi/sound/asoc.h
@@ -243,7 +243,7 @@ struct snd_soc_tplg_manifest {
 	__le32 control_elems;	/* number of control elements */
 	__le32 widget_elems;	/* number of widget elements */
 	__le32 graph_elems;	/* number of graph elements */
-	__le32 dai_elems;	/* number of DAI elements */
+	__le32 pcm_elems;	/* number of PCM elements */
 	__le32 dai_link_elems;	/* number of DAI link elements */
 	struct snd_soc_tplg_private priv;
 } __attribute__((packed));
diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h
index d9bd9ca..9625484 100644
--- a/include/uapi/sound/compress_params.h
+++ b/include/uapi/sound/compress_params.h
@@ -73,7 +73,8 @@
 #define SND_AUDIOCODEC_IEC61937              ((__u32) 0x0000000B)
 #define SND_AUDIOCODEC_G723_1                ((__u32) 0x0000000C)
 #define SND_AUDIOCODEC_G729                  ((__u32) 0x0000000D)
-#define SND_AUDIOCODEC_MAX                   SND_AUDIOCODEC_G729
+#define SND_AUDIOCODEC_BESPOKE               ((__u32) 0x0000000E)
+#define SND_AUDIOCODEC_MAX                   SND_AUDIOCODEC_BESPOKE
 
 /*
  * Profile and modes are listed with bit masks. This allows for a
@@ -312,7 +313,7 @@ struct snd_enc_flac {
 
 struct snd_enc_generic {
 	__u32 bw;	/* encoder bandwidth */
-	__s32 reserved[15];
+	__s32 reserved[15];	/* Can be used for SND_AUDIOCODEC_BESPOKE */
 } __attribute__((packed, aligned(4)));
 
 union snd_codec_options {