Merge tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "We've had quite busy weeks in this cycle.  Looking at ALSA core, the
  significant changes are a few fixes wrt timer and sequencer ioctls
  that have been revealed by fuzzer recently.  Other than that, ASoC
  core got a few updates about DAI link handling, but these are rather
  straightforward refactoring.

  In drivers scene, ASoC received quite lots of new drivers in addition
  to bunch of updates for still ongoing Intel Skylake support and
  topology API.  HD-audio gained a new HDMI/DP hotplug notification via
  component.  FireWire got a pile of code refactoring/updates with
  SCS.1x driver integration.

  More highlights are shown below.

  [ NOTE: this contains also many commits for DRM.  This is due to the
    pull of drm stable branch into sound tree, as the base of i915 audio
    component work for HD-audio.  The highlights below don't contain
    these DRM changes, as these are supposed to be pulled via drm tree
    in anyway sooner or later.  ]

  Core:
   - Handful fixes to harden ALSA timer and sequencer ioctls against
     races reported by syzkaller fuzzer
   - Irq description string can be unique to each card; only for
     HD-audio for now

  ASoC:
   - Conversion of the array of DAI links to a list for supporting
     dynamically adding and removing DAI links
   - Topology API enhancements to make everything more component based
     and being able to specify PCM links via topology
   - 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
   - Lots of new features and cleanups for the Renesas drivers
   - ANC support for WM5110
   - New drivers: Imagination Technologies IPs, Atmel class D speaker,
     Cirrus CS47L24 and WM1831, Dialog DA7128, Realtek RT5659 and
     RT56156, Rockchip RK3036, TI PC3168A, and AMD ACP
   - Rename PCM1792a driver to be generic pcm179x

  HD-Audio:
   - Use audio component for i915 HDMI/DP hotplug handling
   - On-demand binding with i915 driver
   - bdl_pos_adj parameter adjustment for Baytrail controllers
   - Enable power_save_node for CX20722; this shouldn't lead to
     regression, hopefully
   - Kabylake HDMI/DP codec support
   - Quirks for Lenovo E50-80, Dell Latitude E-series, and other Dell
     machines
   - A few code refactoring

  FireWire:
   - Lots of code cleanup and refactoring
   - Integrate the support of SCS.1x devices into snd-oxfw driver;
     snd-scs1x driver is obsoleted

  USB-audio:
   - Fix possible NULL dereference at disconnection
   - A regression fix for Native Instruments devices

  Misc:
   - A few code cleanups of fm801 driver"

* tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (722 commits)
  ALSA: timer: Code cleanup
  ALSA: timer: Harden slave timer list handling
  ALSA: hda - Add fixup for Dell Latitidue E6540
  ALSA: timer: Fix race among timer ioctls
  ALSA: hda - add codec support for Kabylake display audio codec
  ALSA: timer: Fix double unlink of active_list
  ALSA: usb-audio: Fix mixer ctl regression of Native Instrument devices
  ALSA: hda - fix the headset mic detection problem for a Dell laptop
  ALSA: hda - Fix white noise on Dell Latitude E5550
  ALSA: hda_intel: add card number to irq description
  ALSA: seq: Fix race at timer setup and close
  ALSA: seq: Fix missing NULL check at remove_events ioctl
  ALSA: usb-audio: Avoid calling usb_autopm_put_interface() at disconnect
  ASoC: hdac_hdmi: remove unused hdac_hdmi_query_pin_connlist
  ASoC: AMD: Add missing include file
  ALSA: hda - Fixup inverted internal mic for Lenovo E50-80
  ALSA: usb: Add native DSD support for Oppo HA-1
  ASoC: Make aux_dev more like a generic component
  ASoC: bcm2835: cleanup includes by ordering them alphabetically
  ASoC: AMD: Manage ACP 2.x SRAM banks power
  ...

16 files changed, 277 insertions, 24 deletions

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/compress_driver.h b/include/sound/compress_driver.h
index fa1d055..c0abcdc 100644
--- a/include/sound/compress_driver.h
+++ b/include/sound/compress_driver.h
@@ -152,13 +152,18 @@ struct snd_compr {
 	unsigned int direction;
 	struct mutex lock;
 	int device;
+#ifdef CONFIG_SND_VERBOSE_PROCFS
+	char id[64];
+	struct snd_info_entry *proc_root;
+	struct snd_info_entry *proc_info_entry;
+#endif
 };
 
 /* compress device register APIs */
 int snd_compress_register(struct snd_compr *device);
 int snd_compress_deregister(struct snd_compr *device);
 int snd_compress_new(struct snd_card *card, int device,
-			int type, struct snd_compr *compr);
+			int type, const char *id, struct snd_compr *compr);
 
 /* dsp driver callback apis
  * For playback: driver should call snd_compress_fragment_elapsed() to let the
diff --git a/include/sound/core.h b/include/sound/core.h
index cdfecaf..31079ea 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -99,6 +99,7 @@ struct snd_card {
 	char driver[16];		/* driver name */
 	char shortname[32];		/* short name of this soundcard */
 	char longname[80];		/* name of this soundcard */
+	char irq_descr[32];		/* Interrupt description */
 	char mixername[80];		/* mixer name */
 	char components[128];		/* card components delimited with
 								space */
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..5681855 100644
--- a/include/sound/designware_i2s.h
+++ b/include/sound/designware_i2s.h
@@ -45,6 +45,12 @@ struct i2s_platform_data {
 	u32 snd_fmts;
 	u32 snd_rates;
 
+	#define DW_I2S_QUIRK_COMP_REG_OFFSET	(1 << 0)
+	#define DW_I2S_QUIRK_COMP_PARAM1	(1 << 1)
+	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/hda_i915.h b/include/sound/hda_i915.h
index 930b41e..fa341fc 100644
--- a/include/sound/hda_i915.h
+++ b/include/sound/hda_i915.h
@@ -10,6 +10,9 @@
 int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable);
 int snd_hdac_display_power(struct hdac_bus *bus, bool enable);
 int snd_hdac_get_display_clk(struct hdac_bus *bus);
+int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate);
+int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid,
+			   bool *audio_enabled, char *buffer, int max_bytes);
 int snd_hdac_i915_init(struct hdac_bus *bus);
 int snd_hdac_i915_exit(struct hdac_bus *bus);
 int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *);
@@ -26,6 +29,17 @@ static inline int snd_hdac_get_display_clk(struct hdac_bus *bus)
 {
 	return 0;
 }
+static inline int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid,
+					   int rate)
+{
+	return 0;
+}
+static inline int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid,
+					 bool *audio_enabled, char *buffer,
+					 int max_bytes)
+{
+	return -ENODEV;
+}
 static inline int snd_hdac_i915_init(struct hdac_bus *bus)
 {
 	return -ENODEV;
diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h
index 94dc6a9..ff1aecf 100644
--- a/include/sound/hda_register.h
+++ b/include/sound/hda_register.h
@@ -233,6 +233,15 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
 #define AZX_MLCTL_SPA			(1<<16)
 #define AZX_MLCTL_CPA			23
 
+
+/* registers for DMA Resume Capability Structure */
+#define AZX_DRSM_CAP_ID			0x5
+#define AZX_REG_DRSM_CTL		0x4
+/* Base used to calculate the iterating register offset */
+#define AZX_DRSM_BASE			0x08
+/* Interval used to calculate the iterating register offset */
+#define AZX_DRSM_INTERVAL		0x08
+
 /*
  * helpers to read the stream position
  */
diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h
index a4cadd9..07fa592 100644
--- a/include/sound/hdaudio_ext.h
+++ b/include/sound/hdaudio_ext.h
@@ -12,6 +12,7 @@
  * @spbcap: SPIB capabilities pointer
  * @mlcap: MultiLink capabilities pointer
  * @gtscap: gts capabilities pointer
+ * @drsmcap: dma resume capabilities pointer
  * @hlink_list: link list of HDA links
  */
 struct hdac_ext_bus {
@@ -23,6 +24,7 @@ struct hdac_ext_bus {
 	void __iomem *spbcap;
 	void __iomem *mlcap;
 	void __iomem *gtscap;
+	void __iomem *drsmcap;
 
 	struct list_head hlink_list;
 };
@@ -72,6 +74,9 @@ enum hdac_ext_stream_type {
  * @pplc_addr: processing pipe link stream pointer
  * @spib_addr: software position in buffers stream pointer
  * @fifo_addr: software position Max fifos stream pointer
+ * @dpibr_addr: DMA position in buffer resume pointer
+ * @dpib: DMA position in buffer
+ * @lpib: Linear position in buffer
  * @decoupled: stream host and link is decoupled
  * @link_locked: link is locked
  * @link_prepared: link is prepared
@@ -86,6 +91,10 @@ struct hdac_ext_stream {
 	void __iomem *spib_addr;
 	void __iomem *fifo_addr;
 
+	void __iomem *dpibr_addr;
+
+	u32 dpib;
+	u32 lpib;
 	bool decoupled:1;
 	bool link_locked:1;
 	bool link_prepared;
@@ -116,6 +125,11 @@ int snd_hdac_ext_stream_set_spib(struct hdac_ext_bus *ebus,
 				 struct hdac_ext_stream *stream, u32 value);
 int snd_hdac_ext_stream_get_spbmaxfifo(struct hdac_ext_bus *ebus,
 				 struct hdac_ext_stream *stream);
+void snd_hdac_ext_stream_drsm_enable(struct hdac_ext_bus *ebus,
+				bool enable, int index);
+int snd_hdac_ext_stream_set_dpibr(struct hdac_ext_bus *ebus,
+				struct hdac_ext_stream *stream, u32 value);
+int snd_hdac_ext_stream_set_lpib(struct hdac_ext_stream *stream, u32 value);
 
 void snd_hdac_ext_link_stream_start(struct hdac_ext_stream *hstream);
 void snd_hdac_ext_link_stream_clear(struct hdac_ext_stream *hstream);
@@ -133,6 +147,7 @@ struct hdac_ext_link {
 
 int snd_hdac_ext_bus_link_power_up(struct hdac_ext_link *link);
 int snd_hdac_ext_bus_link_power_down(struct hdac_ext_link *link);
+int snd_hdac_ext_bus_link_power_up_all(struct hdac_ext_bus *ebus);
 int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus);
 void snd_hdac_ext_link_set_stream_id(struct hdac_ext_link *link,
 				 int stream);
@@ -186,9 +201,15 @@ 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/i2c.h b/include/sound/i2c.h
index d125ff8..835254d 100644
--- a/include/sound/i2c.h
+++ b/include/sound/i2c.h
@@ -66,7 +66,7 @@ struct snd_i2c_bus {
 		struct snd_i2c_bit_ops *bit;
 		void *ops;
 	} hw_ops;		/* lowlevel operations */
-	struct snd_i2c_ops *ops;	/* midlevel operations */
+	const struct snd_i2c_ops *ops;	/* midlevel operations */
 
 	unsigned long private_value;
 	void *private_data;
diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h
index f6cbef7..fdabbb4 100644
--- a/include/sound/rawmidi.h
+++ b/include/sound/rawmidi.h
@@ -130,7 +130,7 @@ struct snd_rawmidi {
 	int ossreg;
 #endif
 
-	struct snd_rawmidi_global_ops *ops;
+	const struct snd_rawmidi_global_ops *ops;
 
 	struct snd_rawmidi_str streams[2];
 
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-dai.h b/include/sound/soc-dai.h
index 212eaaf..964b7de 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -222,6 +222,7 @@ struct snd_soc_dai_driver {
 	const char *name;
 	unsigned int id;
 	unsigned int base;
+	struct snd_soc_dobj dobj;
 
 	/* DAI driver callbacks */
 	int (*probe)(struct snd_soc_dai *dai);
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 fb955e6..7afb72c 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, \
@@ -787,6 +798,7 @@ struct snd_soc_component {
 	unsigned int registered_as_component:1;
 
 	struct list_head list;
+	struct list_head list_aux; /* for auxiliary component of the card */
 
 	struct snd_soc_dai_driver *dai_drv;
 	int num_dai;
@@ -830,6 +842,9 @@ struct snd_soc_component {
 	int (*probe)(struct snd_soc_component *);
 	void (*remove)(struct snd_soc_component *);
 
+	/* machine specific init */
+	int (*init)(struct snd_soc_component *component);
+
 #ifdef CONFIG_DEBUG_FS
 	void (*init_debugfs)(struct snd_soc_component *component);
 	const char *debugfs_prefix;
@@ -1037,6 +1052,9 @@ 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 +1119,20 @@ 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 */
@@ -1119,8 +1145,7 @@ struct snd_soc_card {
 	 */
 	struct snd_soc_aux_dev *aux_dev;
 	int num_aux_devs;
-	struct snd_soc_pcm_runtime *rtd_aux;
-	int num_aux_rtd;
+	struct list_head aux_comp_list;
 
 	const struct snd_kcontrol_new *controls;
 	int num_controls;
@@ -1201,6 +1226,9 @@ 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 +1253,10 @@ 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 */
@@ -1523,6 +1553,7 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
 	INIT_LIST_HEAD(&card->widgets);
 	INIT_LIST_HEAD(&card->paths);
 	INIT_LIST_HEAD(&card->dapm_list);
+	INIT_LIST_HEAD(&card->aux_comp_list);
 }
 
 static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc)
@@ -1644,6 +1675,14 @@ 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);
+
+int snd_soc_register_dai(struct snd_soc_component *component,
+	struct snd_soc_dai_driver *dai_drv);
+
 #include <sound/soc-dai.h>
 
 #ifdef CONFIG_DEBUG_FS