Merge tag 'asoc-v4.2' into asoc-next

ASoC: Updates for v4.2

The big thing this release has been Liam's addition of topology support
to the core.  We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.

 - Support for loading ASoC topology maps from firmware, intended to be
   used to allow self-describing DSP firmware images to be built which
   can map controls added by the DSP to userspace without the kernel
   needing to know about individual DSP firmwares.
 - Lots of refactoring to avoid direct access to snd_soc_codec where
   it's not needed supporting future refactoring.
 - Big refactoring and cleanup serieses for the Wolfson ADSP and TI
   TAS2552 drivers.
 - Support for TI TAS571x power amplifiers.
 - Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
 - Support for x86 systems with RT5650 and Qualcomm Storm.

# gpg: Signature made Mon 08 Jun 2015 18:48:37 BST using RSA key ID 5D5487D0
# gpg: Oops: keyid_from_fingerprint: no pubkey
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"

Documentation/devicetree/bindings/sound/adi,adau1701.txt

@@ -20,6 +20,8 @@ Optional properties:
 			pin configurations as described in the datasheet,
 			table 53. Note that the value of this property has
 			to be prefixed with '/bits/ 8'.
+ - avdd-supply: 	Power supply for AVDD, providing 3.3V
+ - dvdd-supply: 	Power supply for DVDD, providing 3.3V
 
 Examples:
 
@@ -28,6 +30,8 @@ Examples:
 			compatible = "adi,adau1701";
 			reg = <0x34>;
 			reset-gpio = <&gpio 23 0>;
+			avdd-supply = <&vdd_3v3_reg>;
+			dvdd-supply = <&vdd_3v3_reg>;
 			adi,pll-mode-gpios = <&gpio 24 0 &gpio 25 0>;
 			adi,pin-config = /bits/ 8 <0x4 0x7 0x5 0x5 0x4 0x4
                                                    0x4 0x4 0x4 0x4 0x4 0x4>;

Documentation/devicetree/bindings/sound/bt-sco.txt

+Bluetooth-SCO audio CODEC
+
+This device support generic Bluetooth SCO link.
+
+Required properties:
+
+  - compatible : "delta,dfbmcs320"
+
+Example:
+
+codec: bt_sco {
+	compatible = "delta,dfbmcs320";
+};

Documentation/devicetree/bindings/sound/gtm601.txt

+GTM601 UMTS modem audio interface CODEC
+
+This device has no configuration interface. Sample rate is fixed - 8kHz.
+
+Required properties:
+
+  - compatible : "option,gtm601"
+
+Example:
+
+codec: gtm601_codec {
+	compatible = "option,gtm601";
+};

Documentation/devicetree/bindings/sound/max98090.txt

@@ -18,6 +18,12 @@ Optional properties:
 
 - maxim,dmic-freq: Frequency at which to clock DMIC
 
+- maxim,micbias: Micbias voltage applies to the analog mic, valid voltages value are:
+	0 - 2.2v
+	1 - 2.55v
+	2 - 2.4v
+	3 - 2.8v
+
 Pins on the device (for linking into audio routes):
 
   * MIC1

Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt

@@ -4,12 +4,21 @@ This node models the Qualcomm Technologies Low-Power Audio SubSystem (LPASS).
 
 Required properties:
 
-- compatible		: "qcom,lpass-cpu"
+- compatible		: "qcom,lpass-cpu" or "qcom,apq8016-lpass-cpu"
 - clocks		: Must contain an entry for each entry in clock-names.
 - clock-names		: A list which must include the following entries:
 				* "ahbix-clk"
 				* "mi2s-osr-clk"
 				* "mi2s-bit-clk"
+			: required clocks for "qcom,lpass-cpu-apq8016"
+				* "ahbix-clk"
+				* "mi2s-bit-clk0"
+				* "mi2s-bit-clk1"
+				* "mi2s-bit-clk2"
+				* "mi2s-bit-clk3"
+				* "pcnoc-mport-clk"
+				* "pcnoc-sway-clk"
+
 - interrupts		: Must contain an entry for each entry in
 			  interrupt-names.
 - interrupt-names	: A list which must include the following entries:
@@ -22,6 +31,8 @@ Required properties:
 - reg-names		: A list which must include the following entries:
 				* "lpass-lpaif"
 
+
+
 Optional properties:
 
 - qcom,adsp		: Phandle for the audio DSP node

Documentation/devicetree/bindings/sound/renesas,rsnd.txt

@@ -48,7 +48,7 @@ DAI subnode properties:
 
 Example:
 
-rcar_sound: rcar_sound@ec500000 {
+rcar_sound: sound@ec500000 {
 	#sound-dai-cells = <1>;
 	compatible = "renesas,rcar_sound-r8a7791", "renesas,rcar_sound-gen2";
 	reg =	<0 0xec500000 0 0x1000>, /* SCU */

Documentation/devicetree/bindings/sound/rt5677.txt

@@ -18,6 +18,7 @@ Required properties:
 Optional properties:
 
 - realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin.
+- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin.
 
 - realtek,in1-differential
 - realtek,in2-differential
@@ -70,6 +71,7 @@ rt5677 {
 
 	realtek,pow-ldo2-gpio =
 		<&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>;
+	realtek,reset-gpio = <&gpio TEGRA_GPIO(BB, 3) GPIO_ACTIVE_LOW>;
 	realtek,in1-differential = "true";
 	realtek,gpio-config = /bits/ 8  <0 0 0 0 0 2>;   /* pull up GPIO6 */
 	realtek,jd2-gpio = <3>;  /* Enables Jack detection for GPIO6 */

Documentation/devicetree/bindings/sound/tas571x.txt

+Texas Instruments TAS5711/TAS5717/TAS5719 stereo power amplifiers
+
+The codec is controlled through an I2C interface.  It also has two other
+signals that can be wired up to GPIOs: reset (strongly recommended), and
+powerdown (optional).
+
+Required properties:
+
+- compatible: "ti,tas5711", "ti,tas5717", or "ti,tas5719"
+- reg: The I2C address of the device
+- #sound-dai-cells: must be equal to 0
+
+Optional properties:
+
+- reset-gpios: GPIO specifier for the TAS571x's active low reset line
+- pdn-gpios: GPIO specifier for the TAS571x's active low powerdown line
+- clocks: clock phandle for the MCLK input
+- clock-names: should be "mclk"
+- AVDD-supply: regulator phandle for the AVDD supply (all chips)
+- DVDD-supply: regulator phandle for the DVDD supply (all chips)
+- HPVDD-supply: regulator phandle for the HPVDD supply (5717/5719)
+- PVDD_AB-supply: regulator phandle for the PVDD_AB supply (5717/5719)
+- PVDD_CD-supply: regulator phandle for the PVDD_CD supply (5717/5719)
+- PVDD_A-supply: regulator phandle for the PVDD_A supply (5711)
+- PVDD_B-supply: regulator phandle for the PVDD_B supply (5711)
+- PVDD_C-supply: regulator phandle for the PVDD_C supply (5711)
+- PVDD_D-supply: regulator phandle for the PVDD_D supply (5711)
+
+Example:
+
+	tas5717: audio-codec@2a {
+		compatible = "ti,tas5717";
+		reg = <0x2a>;
+		#sound-dai-cells = <0>;
+
+		reset-gpios = <&gpio5 1 GPIO_ACTIVE_LOW>;
+		pdn-gpios = <&gpio5 2 GPIO_ACTIVE_LOW>;
+
+		clocks = <&clk_core CLK_I2S>;
+		clock-names = "mclk";
+	};

Documentation/devicetree/bindings/sound/wm8741.txt

@@ -10,9 +10,20 @@ Required properties:
   - reg : the I2C address of the device for I2C, the chip select
           number for SPI.
 
+Optional properties:
+
+  - diff-mode: Differential output mode configuration. Default value for field
+    DIFF in register R8 (MODE_CONTROL_2). If absent, the default is 0, shall be:
+    0 = stereo
+    1 = mono left
+    2 = stereo reversed
+    3 = mono right
+
 Example:
 
 codec: wm8741@1a {
 	compatible = "wlf,wm8741";
 	reg = <0x1a>;
+
+	diff-mode = <3>;
 };

Documentation/devicetree/bindings/sound/zte,zx-i2s.txt

+ZTE ZX296702 I2S controller
+
+Required properties:
+ - compatible : Must be "zte,zx296702-i2s"
+ - reg : Must contain I2S core's registers location and length
+ - clocks : Pairs of phandle and specifier referencing the controller's clocks.
+ - clock-names: "tx" for the clock to the I2S interface.
+ - dmas: Pairs of phandle and specifier for the DMA channel that is used by
+   the core. The core expects two dma channels for transmit.
+ - dma-names : Must be "tx" and "rx"
+
+For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties
+please check:
+	* resource-names.txt
+	* clock/clock-bindings.txt
+	* dma/dma.txt
+
+Example:
+	i2s0: i2s0@0b005000 {
+		#sound-dai-cells = <0>;
+		compatible = "zte,zx296702-i2s";
+		reg = <0x0b005000 0x1000>;
+		clocks = <&lsp0clk ZX296702_I2S0_DIV>;
+		clock-names = "tx";
+		interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>;
+		dmas = <&dma 5>, <&dma 6>;
+		dma-names = "tx", "rx";
+		status = "okay";
+	};
+
+	sound {
+		compatible = "simple-audio-card";
+		simple-audio-card,name = "zx296702_snd";
+		simple-audio-card,format = "left_j";
+		simple-audio-card,bitclock-master = <&sndcodec>;
+		simple-audio-card,frame-master = <&sndcodec>;
+		sndcpu: simple-audio-card,cpu {
+			sound-dai = <&i2s0>;
+		};
+
+		sndcodec: simple-audio-card,codec {
+			sound-dai = <&acodec>;
+		};
+	};

Documentation/devicetree/bindings/sound/zte,zx-spdif.txt

+ZTE ZX296702 SPDIF controller
+
+Required properties:
+ - compatible : Must be "zte,zx296702-spdif"
+ - reg : Must contain SPDIF core's registers location and length
+ - clocks : Pairs of phandle and specifier referencing the controller's clocks.
+ - clock-names: "tx" for the clock to the SPDIF interface.
+ - dmas: Pairs of phandle and specifier for the DMA channel that is used by
+   the core. The core expects one dma channel for transmit.
+ - dma-names : Must be "tx"
+
+For more details on the 'dma', 'dma-names', 'clock' and 'clock-names' properties
+please check:
+	* resource-names.txt
+	* clock/clock-bindings.txt
+	* dma/dma.txt
+
+Example:
+	spdif0: spdif0@0b004000 {
+		compatible = "zte,zx296702-spdif";
+		reg = <0x0b004000 0x1000>;
+		clocks = <&lsp0clk ZX296702_SPDIF0_DIV>;
+		clock-names = "tx";
+		interrupts = <GIC_SPI 21 IRQ_TYPE_LEVEL_HIGH>;
+		dmas = <&dma 4>;
+		dma-names = "tx";
+		status = "okay";
+	};

Documentation/devicetree/bindings/vendor-prefixes.txt

@@ -54,6 +54,7 @@ cosmic	Cosmic Circuits
 crystalfontz	Crystalfontz America, Inc.
 dallas	Maxim Integrated Products (formerly Dallas Semiconductor)
 davicom	DAVICOM Semiconductor, Inc.
+delta	Delta Electronics, Inc.
 denx	Denx Software Engineering
 digi	Digi International Inc.
 digilent	Diglent, Inc.

MAINTAINERS

@@ -9924,6 +9924,12 @@ L:	netdev@vger.kernel.org
 S:	Maintained
 F:	drivers/net/ethernet/ti/netcp*
 
+TI TAS571X FAMILY ASoC CODEC DRIVER
+M:	Kevin Cernekee <cernekee@chromium.org>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+S:	Odd Fixes
+F:	sound/soc/codecs/tas571x*
+
 TI TWL4030 SERIES SOC CODEC DRIVER
 M:	Peter Ujfalusi <peter.ujfalusi@ti.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)

drivers/dma/sh/rcar-dmac.c

@@ -465,6 +465,7 @@ static dma_cookie_t rcar_dmac_tx_submit(struct dma_async_tx_descriptor *tx)
 static int rcar_dmac_desc_alloc(struct rcar_dmac_chan *chan, gfp_t gfp)
 {
 	struct rcar_dmac_desc_page *page;
+	unsigned long flags;
 	LIST_HEAD(list);
 	unsigned int i;
 
@@ -482,10 +483,10 @@ static int rcar_dmac_desc_alloc(struct rcar_dmac_chan *chan, gfp_t gfp)
 		list_add_tail(&desc->node, &list);
 	}
 
-	spin_lock_irq(&chan->lock);
+	spin_lock_irqsave(&chan->lock, flags);
 	list_splice_tail(&list, &chan->desc.free);
 	list_add_tail(&page->node, &chan->desc.pages);
-	spin_unlock_irq(&chan->lock);
+	spin_unlock_irqrestore(&chan->lock, flags);
 
 	return 0;
 }
@@ -516,6 +517,7 @@ static void rcar_dmac_desc_put(struct rcar_dmac_chan *chan,
 static void rcar_dmac_desc_recycle_acked(struct rcar_dmac_chan *chan)
 {
 	struct rcar_dmac_desc *desc, *_desc;
+	unsigned long flags;
 	LIST_HEAD(list);
 
 	/*
@@ -524,9 +526,9 @@ static void rcar_dmac_desc_recycle_acked(struct rcar_dmac_chan *chan)
 	 * list_for_each_entry_safe, isn't safe if we release the channel lock
 	 * around the rcar_dmac_desc_put() call.
 	 */
-	spin_lock_irq(&chan->lock);
+	spin_lock_irqsave(&chan->lock, flags);
 	list_splice_init(&chan->desc.wait, &list);
-	spin_unlock_irq(&chan->lock);
+	spin_unlock_irqrestore(&chan->lock, flags);
 
 	list_for_each_entry_safe(desc, _desc, &list, node) {
 		if (async_tx_test_ack(&desc->async_tx)) {
@@ -539,9 +541,9 @@ static void rcar_dmac_desc_recycle_acked(struct rcar_dmac_chan *chan)
 		return;
 
 	/* Put the remaining descriptors back in the wait list. */
-	spin_lock_irq(&chan->lock);
+	spin_lock_irqsave(&chan->lock, flags);
 	list_splice(&list, &chan->desc.wait);
-	spin_unlock_irq(&chan->lock);
+	spin_unlock_irqrestore(&chan->lock, flags);
 }
 
 /*
@@ -556,12 +558,13 @@ static void rcar_dmac_desc_recycle_acked(struct rcar_dmac_chan *chan)
 static struct rcar_dmac_desc *rcar_dmac_desc_get(struct rcar_dmac_chan *chan)
 {
 	struct rcar_dmac_desc *desc;
+	unsigned long flags;
 	int ret;
 
 	/* Recycle acked descriptors before attempting allocation. */
 	rcar_dmac_desc_recycle_acked(chan);
 
-	spin_lock_irq(&chan->lock);
+	spin_lock_irqsave(&chan->lock, flags);
 
 	while (list_empty(&chan->desc.free)) {
 		/*
@@ -570,17 +573,17 @@ static struct rcar_dmac_desc *rcar_dmac_desc_get(struct rcar_dmac_chan *chan)
 		 * allocated descriptors. If the allocation fails return an
 		 * error.
 		 */
-		spin_unlock_irq(&chan->lock);
+		spin_unlock_irqrestore(&chan->lock, flags);
 		ret = rcar_dmac_desc_alloc(chan, GFP_NOWAIT);
 		if (ret < 0)
 			return NULL;
-		spin_lock_irq(&chan->lock);
+		spin_lock_irqsave(&chan->lock, flags);
 	}
 
 	desc = list_first_entry(&chan->desc.free, struct rcar_dmac_desc, node);
 	list_del(&desc->node);
 
-	spin_unlock_irq(&chan->lock);
+	spin_unlock_irqrestore(&chan->lock, flags);
 
 	return desc;
 }
@@ -593,6 +596,7 @@ static struct rcar_dmac_desc *rcar_dmac_desc_get(struct rcar_dmac_chan *chan)
 static int rcar_dmac_xfer_chunk_alloc(struct rcar_dmac_chan *chan, gfp_t gfp)
 {
 	struct rcar_dmac_desc_page *page;
+	unsigned long flags;
 	LIST_HEAD(list);
 	unsigned int i;
 
@@ -606,10 +610,10 @@ static int rcar_dmac_xfer_chunk_alloc(struct rcar_dmac_chan *chan, gfp_t gfp)
 		list_add_tail(&chunk->node, &list);
 	}
 
-	spin_lock_irq(&chan->lock);
+	spin_lock_irqsave(&chan->lock, flags);
 	list_splice_tail(&list, &chan->desc.chunks_free);
 	list_add_tail(&page->node, &chan->desc.pages);
-	spin_unlock_irq(&chan->lock);
+	spin_unlock_irqrestore(&chan->lock, flags);
 
 	return 0;
 }
@@ -627,9 +631,10 @@ static struct rcar_dmac_xfer_chunk *
 rcar_dmac_xfer_chunk_get(struct rcar_dmac_chan *chan)
 {
 	struct rcar_dmac_xfer_chunk *chunk;
+	unsigned long flags;
 	int ret;
 
-	spin_lock_irq(&chan->lock);
+	spin_lock_irqsave(&chan->lock, flags);
 
 	while (list_empty(&chan->desc.chunks_free)) {
 		/*
@@ -638,18 +643,18 @@ rcar_dmac_xfer_chunk_get(struct rcar_dmac_chan *chan)
 		 * allocated descriptors. If the allocation fails return an
 		 * error.
 		 */
-		spin_unlock_irq(&chan->lock);
+		spin_unlock_irqrestore(&chan->lock, flags);
 		ret = rcar_dmac_xfer_chunk_alloc(chan, GFP_NOWAIT);
 		if (ret < 0)
 			return NULL;
-		spin_lock_irq(&chan->lock);
+		spin_lock_irqsave(&chan->lock, flags);
 	}
 
 	chunk = list_first_entry(&chan->desc.chunks_free,
 				 struct rcar_dmac_xfer_chunk, node);
 	list_del(&chunk->node);
 
-	spin_unlock_irq(&chan->lock);
+	spin_unlock_irqrestore(&chan->lock, flags);
 
 	return chunk;
 }

drivers/regulator/arizona-ldo1.c

@@ -78,11 +78,6 @@ static int arizona_ldo1_hc_set_voltage_sel(struct regulator_dev *rdev,
 	if (ret != 0)
 		return ret;
 
-	ret = regmap_update_bits(regmap, ARIZONA_DYNAMIC_FREQUENCY_SCALING_1,
-				 ARIZONA_SUBSYS_MAX_FREQ, val);
-	if (ret != 0)
-		return ret;
-
 	if (val)
 		return 0;
 

include/dt-bindings/sound/apq8016-lpass.h

+#ifndef __DT_APQ8016_LPASS_H
+#define __DT_APQ8016_LPASS_H
+
+#define MI2S_PRIMARY	0
+#define MI2S_SECONDARY	1
+#define MI2S_TERTIARY	2
+#define MI2S_QUATERNARY	3
+
+#endif /* __DT_APQ8016_LPASS_H */

include/dt-bindings/sound/audio-jack-events.h

+#ifndef __AUDIO_JACK_EVENTS_H
+#define __AUDIO_JACK_EVENTS_H
+
+#define JACK_HEADPHONE		1
+#define JACK_MICROPHONE		2
+#define JACK_LINEOUT		3
+#define JACK_LINEIN		4
+
+#endif /* __AUDIO_JACK_EVENTS_H */

include/dt-bindings/sound/tas2552.h

+#ifndef __DT_TAS2552_H
+#define __DT_TAS2552_H
+
+#define TAS2552_PLL_CLKIN		(0)
+#define TAS2552_PDM_CLK			(1)
+#define TAS2552_CLK_TARGET_MASK		(1)
+
+#define TAS2552_PLL_CLKIN_MCLK		((0 << 1) | TAS2552_PLL_CLKIN)
+#define TAS2552_PLL_CLKIN_BCLK		((1 << 1) | TAS2552_PLL_CLKIN)
+#define TAS2552_PLL_CLKIN_IVCLKIN	((2 << 1) | TAS2552_PLL_CLKIN)
+#define TAS2552_PLL_CLKIN_1_8_FIXED	((3 << 1) | TAS2552_PLL_CLKIN)
+
+#define TAS2552_PDM_CLK_PLL		((0 << 1) | TAS2552_PDM_CLK)
+#define TAS2552_PDM_CLK_IVCLKIN		((1 << 1) | TAS2552_PDM_CLK)
+#define TAS2552_PDM_CLK_BCLK		((2 << 1) | TAS2552_PDM_CLK)
+#define TAS2552_PDM_CLK_MCLK		((3 << 1) | TAS2552_PDM_CLK)
+
+#endif /* __DT_TAS2552_H */

include/sound/dmaengine_pcm.h

@@ -90,11 +90,6 @@ void snd_dmaengine_pcm_set_config_from_dai_data(
  * makes sense if SND_DMAENGINE_PCM_FLAG_COMPAT is set as well.
  */
 #define SND_DMAENGINE_PCM_FLAG_NO_DT BIT(1)
-/*
- * The platforms dmaengine driver does not support reporting the amount of
- * bytes that are still left to transfer.
- */
-#define SND_DMAENGINE_PCM_FLAG_NO_RESIDUE BIT(2)
 /*
  * The PCM is half duplex and the DMA channel is shared between capture and
  * playback.

include/sound/rt5645.h

@@ -15,7 +15,6 @@ struct rt5645_platform_data {
 	/* IN2 can optionally be differential */
 	bool in2_diff;
 
-	bool dmic_en;
 	unsigned int dmic1_data_pin;
 	/* 0 = IN2N; 1 = GPIO5; 2 = GPIO11 */
 	unsigned int dmic2_data_pin;
@@ -24,8 +23,6 @@ struct rt5645_platform_data {
 	unsigned int hp_det_gpio;
 	bool gpio_hp_det_active_high;
 
-	/* true if codec's jd function is used */
-	bool en_jd_func;
 	unsigned int jd_mode;
 };
 

include/sound/soc-dapm.h

@@ -15,6 +15,8 @@
 
 #include <linux/types.h>
 #include <sound/control.h>
+#include <sound/soc-topology.h>
+#include <sound/asoc.h>
 
 struct device;
 
@@ -107,6 +109,10 @@ struct device;
 {	.id = snd_soc_dapm_mux, .name = wname, \
 	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
 	.kcontrol_news = wcontrols, .num_kcontrols = 1}
+#define SND_SOC_DAPM_DEMUX(wname, wreg, wshift, winvert, wcontrols) \
+{	.id = snd_soc_dapm_demux, .name = wname, \
+	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.kcontrol_news = wcontrols, .num_kcontrols = 1}
 
 /* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
 #define SOC_PGA_ARRAY(wname, wreg, wshift, winvert,\
@@ -444,11 +450,15 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
 struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm(
 	struct snd_kcontrol *kcontrol);
 
+int snd_soc_dapm_force_bias_level(struct snd_soc_dapm_context *dapm,
+	enum snd_soc_bias_level level);
+
 /* dapm widget types */
 enum snd_soc_dapm_type {
 	snd_soc_dapm_input = 0,		/* input pin */
 	snd_soc_dapm_output,		/* output pin */
 	snd_soc_dapm_mux,			/* selects 1 analog signal from many inputs */
+	snd_soc_dapm_demux,			/* connects the input to one of multiple outputs */
 	snd_soc_dapm_mixer,			/* mixes several analog signals together */
 	snd_soc_dapm_mixer_named_ctl,		/* mixer with named controls */
 	snd_soc_dapm_pga,			/* programmable gain/attenuation (volume) */
@@ -563,6 +573,7 @@ struct snd_soc_dapm_widget {
 	int num_kcontrols;
 	const struct snd_kcontrol_new *kcontrol_news;
 	struct snd_kcontrol **kcontrols;
+	struct snd_soc_dobj dobj;
 
 	/* widget input and outputs */
 	struct list_head sources;
@@ -585,6 +596,10 @@ struct snd_soc_dapm_update {
 	int val;
 };
 
+struct snd_soc_dapm_wcache {
+	struct snd_soc_dapm_widget *widget;
+};
+
 /* DAPM context */
 struct snd_soc_dapm_context {
 	enum snd_soc_bias_level bias_level;
@@ -606,6 +621,9 @@ struct snd_soc_dapm_context {
 	int (*set_bias_level)(struct snd_soc_dapm_context *dapm,
 			      enum snd_soc_bias_level level);
 
+	struct snd_soc_dapm_wcache path_sink_cache;
+	struct snd_soc_dapm_wcache path_source_cache;
+
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_dapm;
 #endif
@@ -623,4 +641,35 @@ struct snd_soc_dapm_stats {
 	int neighbour_checks;
 };
 
+/**
+ * snd_soc_dapm_init_bias_level() - Initialize DAPM bias level
+ * @dapm: The DAPM context to initialize
+ * @level: The DAPM level to initialize to
+ *
+ * This function only sets the driver internal state of the DAPM level and will
+ * not modify the state of the device. Hence it should not be used during normal
+ * operation, but only to synchronize the internal state to the device state.
+ * E.g. during driver probe to set the DAPM level to the one corresponding with
+ * the power-on reset state of the device.
+ *
+ * To change the DAPM state of the device use snd_soc_dapm_set_bias_level().
+ */
+static inline void snd_soc_dapm_init_bias_level(
+	struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level)
+{
+	dapm->bias_level = level;
+}
+
+/**
+ * snd_soc_dapm_get_bias_level() - Get current DAPM bias level
+ * @dapm: The context for which to get the bias level
+ *
+ * Returns: The current bias level of the passed DAPM context.
+ */
+static inline enum snd_soc_bias_level snd_soc_dapm_get_bias_level(
+	struct snd_soc_dapm_context *dapm)
+{
+	return dapm->bias_level;
+}
+
 #endif

include/sound/soc-topology.h

+/*
+ * linux/sound/soc-topology.h -- ALSA SoC Firmware Controls and DAPM
+ *
+ * Copyright (C) 2012 Texas Instruments Inc.
+ * Copyright (C) 2015 Intel Corporation.
+ *
+ * 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.
+ *
+ * Simple file API to load FW that includes mixers, coefficients, DAPM graphs,
+ * algorithms, equalisers, DAIs, widgets, FE caps, BE caps, codec link caps etc.
+ */
+
+#ifndef __LINUX_SND_SOC_TPLG_H
+#define __LINUX_SND_SOC_TPLG_H
+
+#include <sound/asoc.h>
+#include <linux/list.h>
+
+struct firmware;
+struct snd_kcontrol;
+struct snd_soc_tplg_pcm_be;
+struct snd_ctl_elem_value;
+struct snd_ctl_elem_info;
+struct snd_soc_dapm_widget;
+struct snd_soc_component;
+struct snd_soc_tplg_pcm_fe;
+struct snd_soc_dapm_context;
+struct snd_soc_card;
+
+/* object scan be loaded and unloaded in groups with identfying indexes */
+#define SND_SOC_TPLG_INDEX_ALL	0	/* ID that matches all FW objects */
+
+/* dynamic object type */
+enum snd_soc_dobj_type {
+	SND_SOC_DOBJ_NONE		= 0,	/* object is not dynamic */
+	SND_SOC_DOBJ_MIXER,
+	SND_SOC_DOBJ_ENUM,
+	SND_SOC_DOBJ_BYTES,
+	SND_SOC_DOBJ_PCM,
+	SND_SOC_DOBJ_DAI_LINK,
+	SND_SOC_DOBJ_CODEC_LINK,
+	SND_SOC_DOBJ_WIDGET,
+};
+
+/* dynamic control object */
+struct snd_soc_dobj_control {
+	struct snd_kcontrol *kcontrol;
+	char **dtexts;
+	unsigned long *dvalues;
+};
+
+/* dynamic widget object */
+struct snd_soc_dobj_widget {
+	unsigned int kcontrol_enum:1;	/* this widget is an enum kcontrol */
+};
+
+/* dynamic PCM DAI object */
+struct snd_soc_dobj_pcm_dai {
+	struct snd_soc_tplg_pcm_dai *pd;
+	unsigned int count;
+};
+
+/* generic dynamic object - all dynamic objects belong to this struct */
+struct snd_soc_dobj {
+	enum snd_soc_dobj_type type;
+	unsigned int index;	/* objects can belong in different groups */
+	struct list_head list;
+	struct snd_soc_tplg_ops *ops;
+	union {
+		struct snd_soc_dobj_control control;
+		struct snd_soc_dobj_widget widget;
+		struct snd_soc_dobj_pcm_dai pcm_dai;
+	};
+	void *private; /* core does not touch this */
+};
+
+/*
+ * Kcontrol operations - used to map handlers onto firmware based controls.
+ */
+struct snd_soc_tplg_kcontrol_ops {
+	u32 id;
+	int (*get)(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol);
+	int (*put)(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol);
+	int (*info)(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_info *uinfo);
+};
+
+/*
+ * DAPM widget event handlers - used to map handlers onto widgets.
+ */
+struct snd_soc_tplg_widget_events {
+	u16 type;
+	int (*event_handler)(struct snd_soc_dapm_widget *w,
+			struct snd_kcontrol *k, int event);
+};
+
+/*
+ * Public API - Used by component drivers to load and unload dynamic objects
+ * and their resources.
+ */
+struct snd_soc_tplg_ops {
+
+	/* external kcontrol init - used for any driver specific init */
+	int (*control_load)(struct snd_soc_component *,
+		struct snd_kcontrol_new *, struct snd_soc_tplg_ctl_hdr *);
+	int (*control_unload)(struct snd_soc_component *,
+		struct snd_soc_dobj *);
+
+	/* external widget init - used for any driver specific init */
+	int (*widget_load)(struct snd_soc_component *,
+		struct snd_soc_dapm_widget *,
+		struct snd_soc_tplg_dapm_widget *);
+	int (*widget_unload)(struct snd_soc_component *,
+		struct snd_soc_dobj *);
+
+	/* FE - used for any driver specific init */
+	int (*pcm_dai_load)(struct snd_soc_component *,
+		struct snd_soc_tplg_pcm_dai *pcm_dai, int num_fe);
+	int (*pcm_dai_unload)(struct snd_soc_component *,
+		struct snd_soc_dobj *);
+
+	/* callback to handle vendor bespoke data */
+	int (*vendor_load)(struct snd_soc_component *,
+		struct snd_soc_tplg_hdr *);
+	int (*vendor_unload)(struct snd_soc_component *,
+		struct snd_soc_tplg_hdr *);
+
+	/* completion - called at completion of firmware loading */
+	void (*complete)(struct snd_soc_component *);
+
+	/* manifest - optional to inform component of manifest */
+	int (*manifest)(struct snd_soc_component *,
+		struct snd_soc_tplg_manifest *);
+
+	/* bespoke kcontrol handlers available for binding */
+	const struct snd_soc_tplg_kcontrol_ops *io_ops;
+	int io_ops_count;
+};
+
+/* gets a pointer to data from the firmware block header */
+static inline const void *snd_soc_tplg_get_data(struct snd_soc_tplg_hdr *hdr)
+{
+	const void *ptr = hdr;
+
+	return ptr + sizeof(*hdr);
+}
+
+/* Dynamic Object loading and removal for component drivers */
+int snd_soc_tplg_component_load(struct snd_soc_component *comp,
+	struct snd_soc_tplg_ops *ops, const struct firmware *fw,
+	u32 index);
+int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index);
+
+/* Widget removal - widgets also removed wth component API */
+void snd_soc_tplg_widget_remove(struct snd_soc_dapm_widget *w);
+void snd_soc_tplg_widget_remove_all(struct snd_soc_dapm_context *dapm,
+	u32 index);
+
+/* Binds event handlers to dynamic widgets */
+int snd_soc_tplg_widget_bind_event(struct snd_soc_dapm_widget *w,
+	const struct snd_soc_tplg_widget_events *events, int num_events,
+	u16 event_type);
+
+#endif

include/sound/soc.h

@@ -27,6 +27,7 @@
 #include <sound/compress_driver.h>
 #include <sound/control.h>
 #include <sound/ac97_codec.h>
+#include <sound/soc-topology.h>
 
 /*
  * Convenience kcontrol builders
@@ -190,8 +191,12 @@
 #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \
 {	.reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \
 	.mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues}
-#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \
-	SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues)
+#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xitems, xtexts, xvalues) \
+	SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xitems, xtexts, xvalues)
+#define SOC_VALUE_ENUM_SINGLE_AUTODISABLE(xreg, xshift, xmask, xitems, xtexts, xvalues) \
+{	.reg = xreg, .shift_l = xshift, .shift_r = xshift, \
+	.mask = xmask, .items = xitems, .texts = xtexts, \
+	.values = xvalues, .autodisable = 1}
 #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \
 	SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts)
 #define SOC_ENUM(xname, xenum) \
@@ -312,6 +317,11 @@
 							ARRAY_SIZE(xtexts), xtexts, xvalues)
 #define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \
 	SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
+
+#define SOC_VALUE_ENUM_SINGLE_AUTODISABLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \
+	const struct soc_enum name = SOC_VALUE_ENUM_SINGLE_AUTODISABLE(xreg, \
+		xshift, xmask, ARRAY_SIZE(xtexts), xtexts, xvalues)
+
 #define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \
 	const struct soc_enum name = SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(xtexts), xtexts)
 
@@ -767,6 +777,9 @@ struct snd_soc_component {
 
 	struct mutex io_mutex;
 
+	/* attached dynamic objects */
+	struct list_head dobj_list;
+
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_root;
 #endif
@@ -819,7 +832,7 @@ struct snd_soc_codec {
 	/* component */
 	struct snd_soc_component component;
 
-	/* dapm */
+	/* Don't access this directly, use snd_soc_codec_get_dapm() */
 	struct snd_soc_dapm_context dapm;
 
 #ifdef CONFIG_DEBUG_FS
@@ -961,6 +974,24 @@ struct snd_soc_dai_link {
 
 	enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */
 
+	/* codec/machine specific init - e.g. add machine controls */
+	int (*init)(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);
+
+	/* machine stream operations */
+	const struct snd_soc_ops *ops;
+	const struct snd_soc_compr_ops *compr_ops;
+
+	/* For unidirectional dai links */
+	bool playback_only;
+	bool capture_only;
+
+	/* Mark this pcm with non atomic ops */
+	bool nonatomic;
+
 	/* Keep DAI active over suspend */
 	unsigned int ignore_suspend:1;
 
@@ -969,9 +1000,6 @@ struct snd_soc_dai_link {
 	unsigned int symmetric_channels:1;
 	unsigned int symmetric_samplebits:1;
 
-	/* Mark this pcm with non atomic ops */
-	bool nonatomic;
-
 	/* Do not create a PCM for this DAI link (Backend link) */
 	unsigned int no_pcm:1;
 
@@ -982,23 +1010,11 @@ struct snd_soc_dai_link {
 	unsigned int dpcm_capture:1;
 	unsigned int dpcm_playback:1;
 
+	/* DPCM used FE & BE merged format */
+	unsigned int dpcm_merged_format:1;
+
 	/* pmdown_time is ignored at stop */
 	unsigned int ignore_pmdown_time:1;
-
-	/* codec/machine specific init - e.g. add machine controls */
-	int (*init)(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);
-
-	/* machine stream operations */
-	const struct snd_soc_ops *ops;
-	const struct snd_soc_compr_ops *compr_ops;
-
-	/* For unidirectional dai links */
-	bool playback_only;
-	bool capture_only;
 };
 
 struct snd_soc_codec_conf {
@@ -1111,6 +1127,9 @@ struct snd_soc_card {
 	struct list_head dapm_list;
 	struct list_head dapm_dirty;
 
+	/* attached dynamic objects */
+	struct list_head dobj_list;
+
 	/* Generic DAPM context for the card */
 	struct snd_soc_dapm_context dapm;
 	struct snd_soc_dapm_stats dapm_stats;
@@ -1170,6 +1189,7 @@ struct soc_mixer_control {
 	unsigned int sign_bit;
 	unsigned int invert:1;
 	unsigned int autodisable:1;
+	struct snd_soc_dobj dobj;
 };
 
 struct soc_bytes {
@@ -1180,6 +1200,8 @@ struct soc_bytes {
 
 struct soc_bytes_ext {
 	int max;
+	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);
@@ -1200,6 +1222,8 @@ struct soc_enum {
 	unsigned int mask;
 	const char * const *texts;
 	const unsigned int *values;
+	unsigned int autodisable:1;
+	struct snd_soc_dobj dobj;
 };
 
 /**
@@ -1281,6 +1305,58 @@ static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm(
 	return component->dapm_ptr;
 }
 
+/**
+ * snd_soc_codec_get_dapm() - Returns the DAPM context for the CODEC
+ * @codec: The CODEC for which to get the DAPM context
+ *
+ * Note: Use this function instead of directly accessing the CODEC's dapm field
+ */
+static inline struct snd_soc_dapm_context *snd_soc_codec_get_dapm(
+	struct snd_soc_codec *codec)
+{
+	return &codec->dapm;
+}
+
+/**
+ * snd_soc_dapm_init_bias_level() - Initialize CODEC DAPM bias level
+ * @dapm: The CODEC for which to initialize the DAPM bias level
+ * @level: The DAPM level to initialize to
+ *
+ * Initializes the CODEC DAPM bias level. See snd_soc_dapm_init_bias_level().
+ */
+static inline void snd_soc_codec_init_bias_level(struct snd_soc_codec *codec,
+	enum snd_soc_bias_level level)
+{
+	snd_soc_dapm_init_bias_level(snd_soc_codec_get_dapm(codec), level);
+}
+
+/**
+ * snd_soc_dapm_get_bias_level() - Get current CODEC DAPM bias level
+ * @codec: The CODEC for which to get the DAPM bias level
+ *
+ * Returns: The current DAPM bias level of the CODEC.
+ */
+static inline enum snd_soc_bias_level snd_soc_codec_get_bias_level(
+	struct snd_soc_codec *codec)
+{
+	return snd_soc_dapm_get_bias_level(snd_soc_codec_get_dapm(codec));
+}
+
+/**
+ * snd_soc_codec_force_bias_level() - Set the CODEC DAPM bias level
+ * @codec: The CODEC for which to set the level
+ * @level: The level to set to
+ *
+ * Forces the CODEC bias level to a specific state. See
+ * snd_soc_dapm_force_bias_level().
+ */
+static inline int snd_soc_codec_force_bias_level(struct snd_soc_codec *codec,
+	enum snd_soc_bias_level level)
+{
+	return snd_soc_dapm_force_bias_level(snd_soc_codec_get_dapm(codec),
+		level);
+}
+
 /**
  * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol
  * @kcontrol: The kcontrol

include/sound/tlv.h

@@ -31,12 +31,7 @@
  *                       ~(sizeof(unsigned int) - 1)) ....
  */
 
-#define SNDRV_CTL_TLVT_CONTAINER 0	/* one level down - group of TLVs */
-#define SNDRV_CTL_TLVT_DB_SCALE	1       /* dB scale */
-#define SNDRV_CTL_TLVT_DB_LINEAR 2	/* linear volume */
-#define SNDRV_CTL_TLVT_DB_RANGE 3	/* dB range container */
-#define SNDRV_CTL_TLVT_DB_MINMAX 4	/* dB scale with min/max */
-#define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5	/* dB scale with min/max with mute */
+#include <uapi/sound/tlv.h>
 
 #define TLV_ITEM(type, ...) \
 	(type), TLV_LENGTH(__VA_ARGS__), __VA_ARGS__
@@ -90,12 +85,4 @@
 
 #define TLV_DB_GAIN_MUTE	-9999999
 
-/*
- * channel-mapping TLV items
- *  TLV length must match with num_channels
- */
-#define SNDRV_CTL_TLVT_CHMAP_FIXED	0x101	/* fixed channel position */
-#define SNDRV_CTL_TLVT_CHMAP_VAR	0x102	/* channels freely swappable */
-#define SNDRV_CTL_TLVT_CHMAP_PAIRED	0x103	/* pair-wise swappable */
-
 #endif /* __SOUND_TLV_H */

include/uapi/sound/asoc.h

+/*
+ * uapi/sound/asoc.h -- ALSA SoC Firmware Controls and DAPM
+ *
+ * Copyright (C) 2012 Texas Instruments Inc.
+ * Copyright (C) 2015 Intel Corporation.
+ *
+ * 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.
+ *
+ * Simple file API to load FW that includes mixers, coefficients, DAPM graphs,
+ * algorithms, equalisers, DAIs, widgets etc.
+*/
+
+#ifndef __LINUX_UAPI_SND_ASOC_H
+#define __LINUX_UAPI_SND_ASOC_H
+
+#include <linux/types.h>
+#include <sound/asound.h>
+
+/*
+ * Maximum number of channels topology kcontrol can represent.
+ */
+#define SND_SOC_TPLG_MAX_CHAN		8
+
+/*
+ * Maximum number of PCM formats capability
+ */
+#define SND_SOC_TPLG_MAX_FORMATS	16
+
+/*
+ * Maximum number of PCM stream configs
+ */
+#define SND_SOC_TPLG_STREAM_CONFIG_MAX  8
+
+/* individual kcontrol info types - can be mixed with other types */
+#define SND_SOC_TPLG_CTL_VOLSW		1
+#define SND_SOC_TPLG_CTL_VOLSW_SX	2
+#define SND_SOC_TPLG_CTL_VOLSW_XR_SX	3
+#define SND_SOC_TPLG_CTL_ENUM		4
+#define SND_SOC_TPLG_CTL_BYTES		5
+#define SND_SOC_TPLG_CTL_ENUM_VALUE	6
+#define SND_SOC_TPLG_CTL_RANGE		7
+#define SND_SOC_TPLG_CTL_STROBE		8
+
+
+/* individual widget kcontrol info types - can be mixed with other types */
+#define SND_SOC_TPLG_DAPM_CTL_VOLSW		64
+#define SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE	65
+#define SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT		66
+#define SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE	67
+#define SND_SOC_TPLG_DAPM_CTL_PIN		68
+
+/* DAPM widget types - add new items to the end */
+#define SND_SOC_TPLG_DAPM_INPUT		0
+#define SND_SOC_TPLG_DAPM_OUTPUT	1
+#define SND_SOC_TPLG_DAPM_MUX		2
+#define SND_SOC_TPLG_DAPM_MIXER		3
+#define SND_SOC_TPLG_DAPM_PGA		4
+#define SND_SOC_TPLG_DAPM_OUT_DRV	5
+#define SND_SOC_TPLG_DAPM_ADC		6
+#define SND_SOC_TPLG_DAPM_DAC		7
+#define SND_SOC_TPLG_DAPM_SWITCH	8
+#define SND_SOC_TPLG_DAPM_PRE		9
+#define SND_SOC_TPLG_DAPM_POST		10
+#define SND_SOC_TPLG_DAPM_AIF_IN	11
+#define SND_SOC_TPLG_DAPM_AIF_OUT	12
+#define SND_SOC_TPLG_DAPM_DAI_IN	13
+#define SND_SOC_TPLG_DAPM_DAI_OUT	14
+#define SND_SOC_TPLG_DAPM_DAI_LINK	15
+#define SND_SOC_TPLG_DAPM_LAST		SND_SOC_TPLG_DAPM_DAI_LINK
+
+/* Header magic number and string sizes */
+#define SND_SOC_TPLG_MAGIC		0x41536F43 /* ASoC */
+
+/* string sizes */
+#define SND_SOC_TPLG_NUM_TEXTS		16
+
+/* ABI version */
+#define SND_SOC_TPLG_ABI_VERSION	0x2
+
+/* Max size of TLV data */
+#define SND_SOC_TPLG_TLV_SIZE		32
+
+/*
+ * File and Block header data types.
+ * Add new generic and vendor types to end of list.
+ * Generic types are handled by the core whilst vendors types are passed
+ * to the component drivers for handling.
+ */
+#define SND_SOC_TPLG_TYPE_MIXER		1
+#define SND_SOC_TPLG_TYPE_BYTES		2
+#define SND_SOC_TPLG_TYPE_ENUM		3
+#define SND_SOC_TPLG_TYPE_DAPM_GRAPH	4
+#define SND_SOC_TPLG_TYPE_DAPM_WIDGET	5
+#define SND_SOC_TPLG_TYPE_DAI_LINK	6
+#define SND_SOC_TPLG_TYPE_PCM		7
+#define SND_SOC_TPLG_TYPE_MANIFEST	8
+#define SND_SOC_TPLG_TYPE_CODEC_LINK	9
+#define SND_SOC_TPLG_TYPE_MAX	SND_SOC_TPLG_TYPE_CODEC_LINK
+
+/* vendor block IDs - please add new vendor types to end */
+#define SND_SOC_TPLG_TYPE_VENDOR_FW	1000
+#define SND_SOC_TPLG_TYPE_VENDOR_CONFIG	1001
+#define SND_SOC_TPLG_TYPE_VENDOR_COEFF	1002
+#define SND_SOC_TPLG_TYPEVENDOR_CODEC	1003
+
+#define SND_SOC_TPLG_STREAM_PLAYBACK	0
+#define SND_SOC_TPLG_STREAM_CAPTURE	1
+
+/*
+ * Block Header.
+ * This header preceeds all object and object arrays below.
+ */
+struct snd_soc_tplg_hdr {
+	__le32 magic;		/* magic number */
+	__le32 abi;		/* ABI version */
+	__le32 version;		/* optional vendor specific version details */
+	__le32 type;		/* SND_SOC_TPLG_TYPE_ */
+	__le32 size;		/* size of this structure */
+	__le32 vendor_type;	/* optional vendor specific type info */
+	__le32 payload_size;	/* data bytes, excluding this header */
+	__le32 index;		/* identifier for block */
+	__le32 count;		/* number of elements in block */
+} __attribute__((packed));
+
+/*
+ * Private data.
+ * All topology objects may have private data that can be used by the driver or
+ * firmware. Core will ignore this data.
+ */
+struct snd_soc_tplg_private {
+	__le32 size;	/* in bytes of private data */
+	char data[0];
+} __attribute__((packed));
+
+/*
+ * Kcontrol TLV data.
+ */
+struct snd_soc_tplg_ctl_tlv {
+	__le32 size;	/* in bytes aligned to 4 */
+	__le32 numid;	/* control element numeric identification */
+	__le32 count;	/* number of elem in data array */
+	__le32 data[SND_SOC_TPLG_TLV_SIZE];
+} __attribute__((packed));
+
+/*
+ * Kcontrol channel data
+ */
+struct snd_soc_tplg_channel {
+	__le32 size;	/* in bytes of this structure */
+	__le32 reg;
+	__le32 shift;
+	__le32 id;	/* ID maps to Left, Right, LFE etc */
+} __attribute__((packed));
+
+/*
+ * Kcontrol Operations IDs
+ */
+struct snd_soc_tplg_kcontrol_ops_id {
+	__le32 get;
+	__le32 put;
+	__le32 info;
+} __attribute__((packed));
+
+/*
+ * kcontrol header
+ */
+struct snd_soc_tplg_ctl_hdr {
+	__le32 size;	/* in bytes of this structure */
+	__le32 type;
+	char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+	__le32 access;
+	struct snd_soc_tplg_kcontrol_ops_id ops;
+	__le32 tlv_size;	/* non zero means control has TLV data */
+} __attribute__((packed));
+
+/*
+ * Stream Capabilities
+ */
+struct snd_soc_tplg_stream_caps {
+	__le32 size;		/* in bytes of this structure */
+	char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+	__le64 formats[SND_SOC_TPLG_MAX_FORMATS];	/* supported formats SNDRV_PCM_FMTBIT_* */
+	__le32 rates;		/* supported rates SNDRV_PCM_RATE_* */
+	__le32 rate_min;	/* min rate */
+	__le32 rate_max;	/* max rate */
+	__le32 channels_min;	/* min channels */
+	__le32 channels_max;	/* max channels */
+	__le32 periods_min;	/* min number of periods */
+	__le32 periods_max;	/* max number of periods */
+	__le32 period_size_min;	/* min period size bytes */
+	__le32 period_size_max;	/* max period size bytes */
+	__le32 buffer_size_min;	/* min buffer size bytes */
+	__le32 buffer_size_max;	/* max buffer size bytes */
+} __attribute__((packed));
+
+/*
+ * FE or BE Stream configuration supported by SW/FW
+ */
+struct snd_soc_tplg_stream {
+	__le32 size;		/* in bytes of this structure */
+	__le64 format;		/* SNDRV_PCM_FMTBIT_* */
+	__le32 rate;		/* SNDRV_PCM_RATE_* */
+	__le32 period_bytes;	/* size of period in bytes */
+	__le32 buffer_bytes;	/* size of buffer in bytes */
+	__le32 channels;	/* channels */
+	__le32 tdm_slot;	/* optional BE bitmask of supported TDM slots */
+	__le32 dai_fmt;		/* SND_SOC_DAIFMT_  */
+} __attribute__((packed));
+
+/*
+ * Duplex stream configuration supported by SW/FW.
+ */
+struct snd_soc_tplg_stream_config {
+	__le32 size;		/* in bytes of this structure */
+	char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+	struct snd_soc_tplg_stream playback;
+	struct snd_soc_tplg_stream capture;
+} __attribute__((packed));
+
+/*
+ * Manifest. List totals for each payload type. Not used in parsing, but will
+ * be passed to the component driver before any other objects in order for any
+ * global componnent resource allocations.
+ *
+ * File block representation for manifest :-
+ * +-----------------------------------+----+
+ * | struct snd_soc_tplg_hdr           |  1 |
+ * +-----------------------------------+----+
+ * | struct snd_soc_tplg_manifest      |  1 |
+ * +-----------------------------------+----+
+ */
+struct snd_soc_tplg_manifest {
+	__le32 size;		/* in bytes of this structure */
+	__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 dai_link_elems;	/* number of DAI link elements */
+} __attribute__((packed));
+
+/*
+ * Mixer kcontrol.
+ *
+ * File block representation for mixer kcontrol :-
+ * +-----------------------------------+----+
+ * | struct snd_soc_tplg_hdr           |  1 |
+ * +-----------------------------------+----+
+ * | struct snd_soc_tplg_mixer_control |  N |
+ * +-----------------------------------+----+
+ */
+struct snd_soc_tplg_mixer_control {
+	struct snd_soc_tplg_ctl_hdr hdr;
+	__le32 size;	/* in bytes of this structure */
+	__le32 min;
+	__le32 max;
+	__le32 platform_max;
+	__le32 invert;
+	__le32 num_channels;
+	struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN];
+	struct snd_soc_tplg_ctl_tlv tlv;
+	struct snd_soc_tplg_private priv;
+} __attribute__((packed));
+
+/*
+ * Enumerated kcontrol
+ *
+ * File block representation for enum kcontrol :-
+ * +-----------------------------------+----+
+ * | struct snd_soc_tplg_hdr           |  1 |
+ * +-----------------------------------+----+
+ * | struct snd_soc_tplg_enum_control  |  N |
+ * +-----------------------------------+----+
+ */
+struct snd_soc_tplg_enum_control {
+	struct snd_soc_tplg_ctl_hdr hdr;
+	__le32 size;	/* in bytes of this structure */
+	__le32 num_channels;
+	struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN];
+	__le32 items;
+	__le32 mask;
+	__le32 count;
+	char texts[SND_SOC_TPLG_NUM_TEXTS][SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+	__le32 values[SND_SOC_TPLG_NUM_TEXTS * SNDRV_CTL_ELEM_ID_NAME_MAXLEN / 4];
+	struct snd_soc_tplg_private priv;
+} __attribute__((packed));
+
+/*
+ * Bytes kcontrol
+ *
+ * File block representation for bytes kcontrol :-
+ * +-----------------------------------+----+
+ * | struct snd_soc_tplg_hdr           |  1 |
+ * +-----------------------------------+----+
+ * | struct snd_soc_tplg_bytes_control |  N |
+ * +-----------------------------------+----+
+ */
+struct snd_soc_tplg_bytes_control {
+	struct snd_soc_tplg_ctl_hdr hdr;
+	__le32 size;	/* in bytes of this structure */
+	__le32 max;
+	__le32 mask;
+	__le32 base;
+	__le32 num_regs;
+	struct snd_soc_tplg_private priv;
+} __attribute__((packed));
+
+/*
+ * DAPM Graph Element
+ *
+ * File block representation for DAPM graph elements :-
+ * +-------------------------------------+----+
+ * | struct snd_soc_tplg_hdr             |  1 |
+ * +-------------------------------------+----+
+ * | struct snd_soc_tplg_dapm_graph_elem |  N |
+ * +-------------------------------------+----+
+ */
+struct snd_soc_tplg_dapm_graph_elem {
+	char sink[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+	char control[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+	char source[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+} __attribute__((packed));
+
+/*
+ * DAPM Widget.
+ *
+ * File block representation for DAPM widget :-
+ * +-------------------------------------+-----+
+ * | struct snd_soc_tplg_hdr             |  1  |
+ * +-------------------------------------+-----+
+ * | struct snd_soc_tplg_dapm_widget     |  N  |
+ * +-------------------------------------+-----+
+ * |   struct snd_soc_tplg_enum_control  | 0|1 |
+ * |   struct snd_soc_tplg_mixer_control | 0|N |
+ * +-------------------------------------+-----+
+ *
+ * Optional enum or mixer control can be appended to the end of each widget
+ * in the block.
+ */
+struct snd_soc_tplg_dapm_widget {
+	__le32 size;		/* in bytes of this structure */
+	__le32 id;		/* SND_SOC_DAPM_CTL */
+	char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+	char sname[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+
+	__le32 reg;		/* negative reg = no direct dapm */
+	__le32 shift;		/* bits to shift */
+	__le32 mask;		/* non-shifted mask */
+	__u32 invert;		/* invert the power bit */
+	__u32 ignore_suspend;	/* kept enabled over suspend */
+	__u16 event_flags;
+	__u16 event_type;
+	__u16 num_kcontrols;
+	struct snd_soc_tplg_private priv;
+	/*
+	 * kcontrols that relate to this widget
+	 * follow here after widget private data
+	 */
+} __attribute__((packed));
+
+struct snd_soc_tplg_pcm_cfg_caps {
+	struct snd_soc_tplg_stream_caps caps;
+	struct snd_soc_tplg_stream_config configs[SND_SOC_TPLG_STREAM_CONFIG_MAX];
+	__le32 num_configs;	/* number of configs */
+} __attribute__((packed));
+
+/*
+ * Describes SW/FW specific features of PCM or DAI link.
+ *
+ * File block representation for PCM/DAI-Link :-
+ * +-----------------------------------+-----+
+ * | struct snd_soc_tplg_hdr           |  1  |
+ * +-----------------------------------+-----+
+ * | struct snd_soc_tplg_dapm_pcm_dai  |  N  |
+ * +-----------------------------------+-----+
+ */
+struct snd_soc_tplg_pcm_dai {
+	__le32 size;		/* in bytes of this structure */
+	char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
+	__le32 id;			/* unique ID - used to match */
+	__le32 playback;		/* supports playback mode */
+	__le32 capture;			/* supports capture mode */
+	__le32 compress;		/* 1 = compressed; 0 = PCM */
+	struct snd_soc_tplg_pcm_cfg_caps capconf[2];	/* capabilities and configs */
+} __attribute__((packed));
+
+#endif

include/uapi/sound/tlv.h

+/*
+ *   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.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ */
+
+#ifndef __UAPI_SOUND_TLV_H
+#define __UAPI_SOUND_TLV_H
+
+#define SNDRV_CTL_TLVT_CONTAINER 0	/* one level down - group of TLVs */
+#define SNDRV_CTL_TLVT_DB_SCALE	1       /* dB scale */
+#define SNDRV_CTL_TLVT_DB_LINEAR 2	/* linear volume */
+#define SNDRV_CTL_TLVT_DB_RANGE 3	/* dB range container */
+#define SNDRV_CTL_TLVT_DB_MINMAX 4	/* dB scale with min/max */
+#define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5	/* dB scale with min/max with mute */
+
+/*
+ * channel-mapping TLV items
+ *  TLV length must match with num_channels
+ */
+#define SNDRV_CTL_TLVT_CHMAP_FIXED	0x101	/* fixed channel position */
+#define SNDRV_CTL_TLVT_CHMAP_VAR	0x102	/* channels freely swappable */
+#define SNDRV_CTL_TLVT_CHMAP_PAIRED	0x103	/* pair-wise swappable */
+
+#endif

sound/soc/Kconfig

@@ -57,6 +57,7 @@ source "sound/soc/tegra/Kconfig"
 source "sound/soc/txx9/Kconfig"
 source "sound/soc/ux500/Kconfig"
 source "sound/soc/xtensa/Kconfig"
+source "sound/soc/zte/Kconfig"
 
 # Supported codecs
 source "sound/soc/codecs/Kconfig"

sound/soc/Makefile

 snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
 snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o
+snd-soc-core-objs += soc-topology.o
 
 ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),)
 snd-soc-core-objs += soc-generic-dmaengine-pcm.o
@@ -38,3 +39,4 @@ obj-$(CONFIG_SND_SOC)	+= tegra/
 obj-$(CONFIG_SND_SOC)	+= txx9/
 obj-$(CONFIG_SND_SOC)	+= ux500/
 obj-$(CONFIG_SND_SOC)	+= xtensa/
+obj-$(CONFIG_SND_SOC)	+= zte/

sound/soc/atmel/Kconfig

@@ -6,27 +6,22 @@ config SND_ATMEL_SOC
 	  the ATMEL SSC interface. You will also need
 	  to select the audio interfaces to support below.
 
+if SND_ATMEL_SOC
+
 config SND_ATMEL_SOC_PDC
-	tristate
-	depends on SND_ATMEL_SOC
+	bool
 
 config SND_ATMEL_SOC_DMA
-	tristate
-	depends on SND_ATMEL_SOC
+	bool
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 
 config SND_ATMEL_SOC_SSC
 	tristate
-	depends on SND_ATMEL_SOC
-	help
-	  Say Y or M if you want to add support for codecs the
-	  ATMEL SSC interface. You will also needs to select the individual
-	  machine drivers to support below.
 
 config SND_AT91_SOC_SAM9G20_WM8731
 	tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board"
 	depends on ARCH_AT91 || COMPILE_TEST
-	depends on ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI
+	depends on ATMEL_SSC && SND_SOC_I2C_AND_SPI
 	select SND_ATMEL_SOC_PDC
 	select SND_ATMEL_SOC_SSC
 	select SND_SOC_WM8731
@@ -37,7 +32,7 @@ config SND_AT91_SOC_SAM9G20_WM8731
 config SND_ATMEL_SOC_WM8904
 	tristate "Atmel ASoC driver for boards using WM8904 codec"
 	depends on ARCH_AT91 || COMPILE_TEST
-	depends on ATMEL_SSC && SND_ATMEL_SOC && I2C
+	depends on ATMEL_SSC && I2C
 	select SND_ATMEL_SOC_SSC
 	select SND_ATMEL_SOC_DMA
 	select SND_SOC_WM8904
@@ -48,10 +43,11 @@ config SND_ATMEL_SOC_WM8904
 config SND_AT91_SOC_SAM9X5_WM8731
 	tristate "SoC Audio support for WM8731-based at91sam9x5 board"
 	depends on ARCH_AT91 || COMPILE_TEST
-	depends on ATMEL_SSC && SND_ATMEL_SOC && SND_SOC_I2C_AND_SPI
+	depends on ATMEL_SSC && SND_SOC_I2C_AND_SPI
 	select SND_ATMEL_SOC_SSC
 	select SND_ATMEL_SOC_DMA
 	select SND_SOC_WM8731
 	help
 	  Say Y if you want to add support for audio SoC on an
 	  at91sam9x5 based board that is using WM8731 codec.
+endif

sound/soc/atmel/Makefile

 # AT91 Platform Support
-snd-soc-atmel-pcm-pdc-objs := atmel-pcm-pdc.o
-snd-soc-atmel-pcm-dma-objs := atmel-pcm-dma.o
-snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o
+snd-soc-atmel-pcm-$(CONFIG_SND_ATMEL_SOC_PDC) := atmel-pcm-pdc.o
+snd-soc-atmel-pcm-$(CONFIG_SND_ATMEL_SOC_DMA) += atmel-pcm-dma.o
+snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o $(snd-soc-atmel-pcm-y)
 
-obj-$(CONFIG_SND_ATMEL_SOC_PDC) += snd-soc-atmel-pcm-pdc.o
-obj-$(CONFIG_SND_ATMEL_SOC_DMA) += snd-soc-atmel-pcm-dma.o
 obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
 
 # AT91 Machine Support

sound/soc/atmel/atmel-pcm-dma.c

@@ -124,8 +124,7 @@ static const struct snd_dmaengine_pcm_config atmel_dmaengine_pcm_config = {
 
 int atmel_pcm_dma_platform_register(struct device *dev)
 {
-	return snd_dmaengine_pcm_register(dev, &atmel_dmaengine_pcm_config,
-			SND_DMAENGINE_PCM_FLAG_NO_RESIDUE);
+	return snd_dmaengine_pcm_register(dev, &atmel_dmaengine_pcm_config, 0);
 }
 EXPORT_SYMBOL(atmel_pcm_dma_platform_register);
 

sound/soc/atmel/sam9g20_wm8731.c

@@ -95,8 +95,9 @@ static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = {
 
 static const struct snd_soc_dapm_route intercon[] = {
 
-	/* speaker connected to LHPOUT */
+	/* speaker connected to LHPOUT/RHPOUT */
 	{"Ext Spk", NULL, "LHPOUT"},
+	{"Ext Spk", NULL, "RHPOUT"},
 
 	/* mic is connected to Mic Jack, with WM8731 Mic Bias */
 	{"MICIN", NULL, "Mic Bias"},
@@ -108,9 +109,7 @@ static const struct snd_soc_dapm_route intercon[] = {
  */
 static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 
 	printk(KERN_DEBUG
@@ -124,10 +123,6 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 		return ret;
 	}
 
-	/* not connected */
-	snd_soc_dapm_nc_pin(dapm, "RLINEIN");
-	snd_soc_dapm_nc_pin(dapm, "LLINEIN");
-
 #ifndef ENABLE_MIC_INPUT
 	snd_soc_dapm_nc_pin(&rtd->card->dapm, "Int Mic");
 #endif
@@ -158,6 +153,7 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = {
 	.num_dapm_widgets = ARRAY_SIZE(at91sam9g20ek_dapm_widgets),
 	.dapm_routes = intercon,
 	.num_dapm_routes = ARRAY_SIZE(intercon),
+	.fully_routed = true,
 };
 
 static int at91sam9g20ek_audio_probe(struct platform_device *pdev)

sound/soc/au1x/db1200.c

@@ -21,7 +21,7 @@
 #include "../codecs/wm8731.h"
 #include "psc.h"
 
-static struct platform_device_id db1200_pids[] = {
+static const struct platform_device_id db1200_pids[] = {
 	{
 		.name		= "db1200-ac97",
 		.driver_data	= 0,

sound/soc/cirrus/ep93xx-pcm.c

@@ -60,7 +60,6 @@ int devm_ep93xx_pcm_platform_register(struct device *dev)
 {
 	return devm_snd_dmaengine_pcm_register(dev,
 		&ep93xx_dmaengine_pcm_config,
-		SND_DMAENGINE_PCM_FLAG_NO_RESIDUE |
 		SND_DMAENGINE_PCM_FLAG_NO_DT |
 		SND_DMAENGINE_PCM_FLAG_COMPAT);
 }

sound/soc/codecs/88pm860x-codec.c

@@ -1140,7 +1140,7 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			/* Enable Audio PLL & Audio section */
 			data = AUDIO_PLL | AUDIO_SECTION_ON;
 			pm860x_reg_write(pm860x->i2c, REG_MISC2, data);
@@ -1156,7 +1156,6 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec,
 		pm860x_set_bits(pm860x->i2c, REG_MISC2, data, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/Kconfig

@@ -16,7 +16,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_88PM860X if MFD_88PM860X
 	select SND_SOC_L3
 	select SND_SOC_AB8500_CODEC if ABX500_CORE
-	select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
+	select SND_SOC_AC97_CODEC
 	select SND_SOC_AD1836 if SPI_MASTER
 	select SND_SOC_AD193X_SPI if SPI_MASTER
 	select SND_SOC_AD193X_I2C if I2C
@@ -54,7 +54,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_CS4271_SPI if SPI_MASTER
 	select SND_SOC_CS42XX8_I2C if I2C
 	select SND_SOC_CX20442 if TTY
-	select SND_SOC_DA7210 if I2C
+	select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_DA7213 if I2C
 	select SND_SOC_DA732X if I2C
 	select SND_SOC_DA9055 if I2C
@@ -104,6 +104,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
 	select SND_SOC_TAS2552 if I2C
 	select SND_SOC_TAS5086 if I2C
+	select SND_SOC_TAS571X if I2C
 	select SND_SOC_TFA9879 if I2C
 	select SND_SOC_TLV320AIC23_I2C if I2C
 	select SND_SOC_TLV320AIC23_SPI if SPI_MASTER
@@ -211,8 +212,9 @@ config SND_SOC_AB8500_CODEC
 	tristate
 
 config SND_SOC_AC97_CODEC
-	tristate
+	tristate "Build generic ASoC AC97 CODEC driver"
 	select SND_AC97_CODEC
+	select SND_SOC_AC97_BUS
 
 config SND_SOC_AD1836
 	tristate
@@ -611,6 +613,10 @@ config SND_SOC_TAS5086
 	tristate "Texas Instruments TAS5086 speaker amplifier"
 	depends on I2C
 
+config SND_SOC_TAS571X
+	tristate "Texas Instruments TAS5711/TAS5717/TAS5719 power amplifiers"
+	depends on I2C
+
 config SND_SOC_TFA9879
 	tristate "NXP Semiconductors TFA9879 amplifier"
 	depends on I2C

sound/soc/codecs/Makefile

@@ -106,6 +106,7 @@ snd-soc-sta350-objs := sta350.o
 snd-soc-sta529-objs := sta529.o
 snd-soc-stac9766-objs := stac9766.o
 snd-soc-tas5086-objs := tas5086.o
+snd-soc-tas571x-objs := tas571x.o
 snd-soc-tfa9879-objs := tfa9879.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o
@@ -288,6 +289,7 @@ obj-$(CONFIG_SND_SOC_STA529)   += snd-soc-sta529.o
 obj-$(CONFIG_SND_SOC_STAC9766)	+= snd-soc-stac9766.o
 obj-$(CONFIG_SND_SOC_TAS2552)	+= snd-soc-tas2552.o
 obj-$(CONFIG_SND_SOC_TAS5086)	+= snd-soc-tas5086.o
+obj-$(CONFIG_SND_SOC_TAS571X)	+= snd-soc-tas571x.o
 obj-$(CONFIG_SND_SOC_TFA9879)	+= snd-soc-tfa9879.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C)	+= snd-soc-tlv320aic23-i2c.o

sound/soc/codecs/ab8500-codec.c

@@ -1209,6 +1209,7 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
 				struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
 	struct device *dev = codec->dev;
 	bool apply_fir, apply_iir;
@@ -1234,15 +1235,14 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
 	apply_fir = req == ANC_APPLY_FIR || req == ANC_APPLY_FIR_IIR;
 	apply_iir = req == ANC_APPLY_IIR || req == ANC_APPLY_FIR_IIR;
 
-	status = snd_soc_dapm_force_enable_pin(&codec->dapm,
-					"ANC Configure Input");
+	status = snd_soc_dapm_force_enable_pin(dapm, "ANC Configure Input");
 	if (status < 0) {
 		dev_err(dev,
 			"%s: ERROR: Failed to enable power (status = %d)!\n",
 			__func__, status);
 		goto cleanup;
 	}
-	snd_soc_dapm_sync(&codec->dapm);
+	snd_soc_dapm_sync(dapm);
 
 	anc_configure(codec, apply_fir, apply_iir);
 
@@ -1259,8 +1259,8 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol,
 			drvdata->anc_status =  ANC_IIR_CONFIGURED;
 	}
 
-	status = snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input");
-	snd_soc_dapm_sync(&codec->dapm);
+	status = snd_soc_dapm_disable_pin(dapm, "ANC Configure Input");
+	snd_soc_dapm_sync(dapm);
 
 cleanup:
 	mutex_unlock(&drvdata->ctrl_lock);
@@ -1947,6 +1947,7 @@ static int ab8500_audio_init_audioblock(struct snd_soc_codec *codec)
 static int ab8500_audio_setup_mics(struct snd_soc_codec *codec,
 			struct amic_settings *amics)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	u8 value8;
 	unsigned int value;
 	int status;
@@ -1973,15 +1974,15 @@ static int ab8500_audio_setup_mics(struct snd_soc_codec *codec,
 	dev_dbg(codec->dev, "%s: Mic 1a regulator: %s\n", __func__,
 		amic_micbias_str(amics->mic1a_micbias));
 	route = &ab8500_dapm_routes_mic1a_vamicx[amics->mic1a_micbias];
-	status = snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+	status = snd_soc_dapm_add_routes(dapm, route, 1);
 	dev_dbg(codec->dev, "%s: Mic 1b regulator: %s\n", __func__,
 		amic_micbias_str(amics->mic1b_micbias));
 	route = &ab8500_dapm_routes_mic1b_vamicx[amics->mic1b_micbias];
-	status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+	status |= snd_soc_dapm_add_routes(dapm, route, 1);
 	dev_dbg(codec->dev, "%s: Mic 2 regulator: %s\n", __func__,
 		amic_micbias_str(amics->mic2_micbias));
 	route = &ab8500_dapm_routes_mic2_vamicx[amics->mic2_micbias];
-	status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+	status |= snd_soc_dapm_add_routes(dapm, route, 1);
 	if (status < 0) {
 		dev_err(codec->dev,
 			"%s: Failed to add AMic-regulator DAPM-routes (%d).\n",
@@ -2461,6 +2462,7 @@ static void ab8500_codec_of_probe(struct device *dev, struct device_node *np,
 
 static int ab8500_codec_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct device *dev = codec->dev;
 	struct device_node *np = dev->of_node;
 	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev);
@@ -2541,7 +2543,7 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec)
 		&ab8500_filter_controls[AB8500_FILTER_SID_FIR].private_value;
 	drvdata->sid_fir_values = (long *)fc->value;
 
-	(void)snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input");
+	snd_soc_dapm_disable_pin(dapm, "ANC Configure Input");
 
 	mutex_init(&drvdata->ctrl_lock);
 

sound/soc/codecs/ac97.c

@@ -44,10 +44,6 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 	return snd_ac97_set_rate(ac97, reg, substream->runtime->rate);
 }
 
-#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
-		SNDRV_PCM_RATE_48000)
-
 static const struct snd_soc_dai_ops ac97_dai_ops = {
 	.prepare	= ac97_prepare,
 };
@@ -58,13 +54,13 @@ static struct snd_soc_dai_driver ac97_dai = {
 		.stream_name = "AC97 Playback",
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = STD_AC97_RATES,
+		.rates = SNDRV_PCM_RATE_KNOT,
 		.formats = SND_SOC_STD_AC97_FMTS,},
 	.capture = {
 		.stream_name = "AC97 Capture",
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = STD_AC97_RATES,
+		.rates = SNDRV_PCM_RATE_KNOT,
 		.formats = SND_SOC_STD_AC97_FMTS,},
 	.ops = &ac97_dai_ops,
 };

sound/soc/codecs/ad1836.c

@@ -251,7 +251,7 @@ static int ad1836_resume(struct snd_soc_codec *codec)
 static int ad1836_probe(struct snd_soc_codec *codec)
 {
 	struct ad1836_priv *ad1836 = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	int num_dacs, num_adcs;
 	int ret = 0;
 	int i;

sound/soc/codecs/adau1373.c

@@ -1444,7 +1444,6 @@ static int adau1373_set_bias_level(struct snd_soc_codec *codec,
 			ADAU1373_PWDN_CTRL3_PWR_EN, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/adau1701.c

@@ -16,6 +16,7 @@
 #include <linux/of.h>
 #include <linux/of_gpio.h>
 #include <linux/of_device.h>
+#include <linux/regulator/consumer.h>
 #include <linux/regmap.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -101,6 +102,10 @@
 
 #define ADAU1701_FIRMWARE "adau1701.bin"
 
+static const char * const supply_names[] = {
+	"dvdd", "avdd"
+};
+
 struct adau1701 {
 	int gpio_nreset;
 	int gpio_pll_mode[2];
@@ -112,6 +117,7 @@ struct adau1701 {
 	u8 pin_config[12];
 
 	struct sigmadsp *sigmadsp;
+	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
 };
 
 static const struct snd_kcontrol_new adau1701_controls[] = {
@@ -565,7 +571,6 @@ static int adau1701_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -669,6 +674,13 @@ static int adau1701_probe(struct snd_soc_codec *codec)
 	if (ret)
 		return ret;
 
+	ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies),
+				    adau1701->supplies);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to enable regulators: %d\n", ret);
+		return ret;
+	}
+
 	/*
 	 * Let the pll_clkdiv variable default to something that won't happen
 	 * at runtime. That way, we can postpone the firmware download from
@@ -680,7 +692,7 @@ static int adau1701_probe(struct snd_soc_codec *codec)
 	/* initalize with pre-configured pll mode settings */
 	ret = adau1701_reset(codec, adau1701->pll_clkdiv, 0);
 	if (ret < 0)
-		return ret;
+		goto exit_regulators_disable;
 
 	/* set up pin config */
 	val = 0;
@@ -696,10 +708,60 @@ static int adau1701_probe(struct snd_soc_codec *codec)
 	regmap_write(adau1701->regmap, ADAU1701_PINCONF_1, val);
 
 	return 0;
+
+exit_regulators_disable:
+
+	regulator_bulk_disable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies);
+	return ret;
 }
 
+static int adau1701_remove(struct snd_soc_codec *codec)
+{
+	struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
+
+	if (gpio_is_valid(adau1701->gpio_nreset))
+		gpio_set_value_cansleep(adau1701->gpio_nreset, 0);
+
+	regulator_bulk_disable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int adau1701_suspend(struct snd_soc_codec *codec)
+{
+	struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
+
+	regulator_bulk_disable(ARRAY_SIZE(adau1701->supplies),
+			       adau1701->supplies);
+
+	return 0;
+}
+
+static int adau1701_resume(struct snd_soc_codec *codec)
+{
+	struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+        ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies),
+				    adau1701->supplies);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to enable regulators: %d\n", ret);
+		return ret;
+	}
+
+	return adau1701_reset(codec, adau1701->pll_clkdiv, 0);
+}
+#else
+#define adau1701_resume 	NULL
+#define adau1701_suspend 	NULL
+#endif /* CONFIG_PM */
+
 static struct snd_soc_codec_driver adau1701_codec_drv = {
 	.probe			= adau1701_probe,
+	.remove			= adau1701_remove,
+	.resume			= adau1701_resume,
+	.suspend		= adau1701_suspend,
 	.set_bias_level		= adau1701_set_bias_level,
 	.idle_bias_off		= true,
 
@@ -730,32 +792,58 @@ static int adau1701_i2c_probe(struct i2c_client *client,
 	struct device *dev = &client->dev;
 	int gpio_nreset = -EINVAL;
 	int gpio_pll_mode[2] = { -EINVAL, -EINVAL };
-	int ret;
+	int ret, i;
 
 	adau1701 = devm_kzalloc(dev, sizeof(*adau1701), GFP_KERNEL);
 	if (!adau1701)
 		return -ENOMEM;
 
+	for (i = 0; i < ARRAY_SIZE(supply_names); i++)
+		adau1701->supplies[i].supply = supply_names[i];
+
+	ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(adau1701->supplies),
+			adau1701->supplies);
+	if (ret < 0) {
+		dev_err(dev, "Failed to get regulators: %d\n", ret);
+		return ret;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies),
+			adau1701->supplies);
+	if (ret < 0) {
+		dev_err(dev, "Failed to enable regulators: %d\n", ret);
+		return ret;
+	}
+
 	adau1701->client = client;
 	adau1701->regmap = devm_regmap_init(dev, NULL, client,
 					    &adau1701_regmap);
-	if (IS_ERR(adau1701->regmap))
-		return PTR_ERR(adau1701->regmap);
+	if (IS_ERR(adau1701->regmap)) {
+		ret = PTR_ERR(adau1701->regmap);
+		goto exit_regulators_disable;
+	}
+
 
 	if (dev->of_node) {
 		gpio_nreset = of_get_named_gpio(dev->of_node, "reset-gpio", 0);
-		if (gpio_nreset < 0 && gpio_nreset != -ENOENT)
-			return gpio_nreset;
+		if (gpio_nreset < 0 && gpio_nreset != -ENOENT) {
+			ret = gpio_nreset;
+			goto exit_regulators_disable;
+		}
 
 		gpio_pll_mode[0] = of_get_named_gpio(dev->of_node,
 						   "adi,pll-mode-gpios", 0);
-		if (gpio_pll_mode[0] < 0 && gpio_pll_mode[0] != -ENOENT)
-			return gpio_pll_mode[0];
+		if (gpio_pll_mode[0] < 0 && gpio_pll_mode[0] != -ENOENT) {
+			ret = gpio_pll_mode[0];
+			goto exit_regulators_disable;
+		}
 
 		gpio_pll_mode[1] = of_get_named_gpio(dev->of_node,
 						   "adi,pll-mode-gpios", 1);
-		if (gpio_pll_mode[1] < 0 && gpio_pll_mode[1] != -ENOENT)
-			return gpio_pll_mode[1];
+		if (gpio_pll_mode[1] < 0 && gpio_pll_mode[1] != -ENOENT) {
+			ret = gpio_pll_mode[1];
+			goto exit_regulators_disable;
+		}
 
 		of_property_read_u32(dev->of_node, "adi,pll-clkdiv",
 				     &adau1701->pll_clkdiv);
@@ -769,7 +857,7 @@ static int adau1701_i2c_probe(struct i2c_client *client,
 		ret = devm_gpio_request_one(dev, gpio_nreset, GPIOF_OUT_INIT_LOW,
 					    "ADAU1701 Reset");
 		if (ret < 0)
-			return ret;
+			goto exit_regulators_disable;
 	}
 
 	if (gpio_is_valid(gpio_pll_mode[0]) &&
@@ -778,13 +866,13 @@ static int adau1701_i2c_probe(struct i2c_client *client,
 					    GPIOF_OUT_INIT_LOW,
 					    "ADAU1701 PLL mode 0");
 		if (ret < 0)
-			return ret;
+			goto exit_regulators_disable;
 
 		ret = devm_gpio_request_one(dev, gpio_pll_mode[1],
 					    GPIOF_OUT_INIT_LOW,
 					    "ADAU1701 PLL mode 1");
 		if (ret < 0)
-			return ret;
+			goto exit_regulators_disable;
 	}
 
 	adau1701->gpio_nreset = gpio_nreset;
@@ -795,11 +883,17 @@ static int adau1701_i2c_probe(struct i2c_client *client,
 
 	adau1701->sigmadsp = devm_sigmadsp_init_i2c(client,
 		&adau1701_sigmadsp_ops, ADAU1701_FIRMWARE);
-	if (IS_ERR(adau1701->sigmadsp))
-		return PTR_ERR(adau1701->sigmadsp);
+	if (IS_ERR(adau1701->sigmadsp)) {
+		ret = PTR_ERR(adau1701->sigmadsp);
+		goto exit_regulators_disable;
+	}
 
 	ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv,
 			&adau1701_dai, 1);
+
+exit_regulators_disable:
+
+	regulator_bulk_disable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies);
 	return ret;
 }
 

sound/soc/codecs/adau1761.c

@@ -466,7 +466,6 @@ static int adau1761_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -483,6 +482,7 @@ static enum adau1761_output_mode adau1761_get_lineout_mode(
 
 static int adau1761_setup_digmic_jackdetect(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau1761_platform_data *pdata = codec->dev->platform_data;
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
 	enum adau1761_digmic_jackdet_pin_mode mode;
@@ -515,21 +515,18 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_codec *codec)
 		if (ret)
 			return ret;
 	case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
-			adau1761_no_dmic_routes,
+		ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes,
 			ARRAY_SIZE(adau1761_no_dmic_routes));
 		if (ret)
 			return ret;
 		break;
 	case ADAU1761_DIGMIC_JACKDET_PIN_MODE_DIGMIC:
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
-			adau1761_dmic_widgets,
+		ret = snd_soc_dapm_new_controls(dapm, adau1761_dmic_widgets,
 			ARRAY_SIZE(adau1761_dmic_widgets));
 		if (ret)
 			return ret;
 
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
-			adau1761_dmic_routes,
+		ret = snd_soc_dapm_add_routes(dapm, adau1761_dmic_routes,
 			ARRAY_SIZE(adau1761_dmic_routes));
 		if (ret)
 			return ret;
@@ -547,6 +544,7 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_codec *codec)
 
 static int adau1761_setup_headphone_mode(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
 	struct adau1761_platform_data *pdata = codec->dev->platform_data;
 	enum adau1761_output_mode mode;
@@ -577,12 +575,12 @@ static int adau1761_setup_headphone_mode(struct snd_soc_codec *codec)
 	}
 
 	if (mode == ADAU1761_OUTPUT_MODE_HEADPHONE_CAPLESS) {
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
+		ret = snd_soc_dapm_new_controls(dapm,
 			adau1761_capless_dapm_widgets,
 			ARRAY_SIZE(adau1761_capless_dapm_widgets));
 		if (ret)
 			return ret;
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
+		ret = snd_soc_dapm_add_routes(dapm,
 			adau1761_capless_dapm_routes,
 			ARRAY_SIZE(adau1761_capless_dapm_routes));
 	} else {
@@ -590,12 +588,12 @@ static int adau1761_setup_headphone_mode(struct snd_soc_codec *codec)
 			ARRAY_SIZE(adau1761_mono_controls));
 		if (ret)
 			return ret;
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
+		ret = snd_soc_dapm_new_controls(dapm,
 			adau1761_mono_dapm_widgets,
 			ARRAY_SIZE(adau1761_mono_dapm_widgets));
 		if (ret)
 			return ret;
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
+		ret = snd_soc_dapm_add_routes(dapm,
 			adau1761_mono_dapm_routes,
 			ARRAY_SIZE(adau1761_mono_dapm_routes));
 	}
@@ -640,6 +638,7 @@ static bool adau1761_readable_register(struct device *dev, unsigned int reg)
 
 static int adau1761_codec_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau1761_platform_data *pdata = codec->dev->platform_data;
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
 	int ret;
@@ -692,14 +691,12 @@ static int adau1761_codec_probe(struct snd_soc_codec *codec)
 		return ret;
 
 	if (adau->type == ADAU1761) {
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
-			adau1761_dapm_widgets,
+		ret = snd_soc_dapm_new_controls(dapm, adau1761_dapm_widgets,
 			ARRAY_SIZE(adau1761_dapm_widgets));
 		if (ret)
 			return ret;
 
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
-			adau1761_dapm_routes,
+		ret = snd_soc_dapm_add_routes(dapm, adau1761_dapm_routes,
 			ARRAY_SIZE(adau1761_dapm_routes));
 		if (ret)
 			return ret;

sound/soc/codecs/adau1781.c

@@ -339,7 +339,6 @@ static int adau1781_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -383,6 +382,7 @@ static int adau1781_set_input_mode(struct adau *adau, unsigned int reg,
 
 static int adau1781_codec_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau1781_platform_data *pdata = dev_get_platdata(codec->dev);
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
 	int ret;
@@ -403,19 +403,17 @@ static int adau1781_codec_probe(struct snd_soc_codec *codec)
 	}
 
 	if (pdata && pdata->use_dmic) {
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
+		ret = snd_soc_dapm_new_controls(dapm,
 			adau1781_dmic_dapm_widgets,
 			ARRAY_SIZE(adau1781_dmic_dapm_widgets));
 		if (ret)
 			return ret;
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
-			adau1781_dmic_dapm_routes,
+		ret = snd_soc_dapm_add_routes(dapm, adau1781_dmic_dapm_routes,
 			ARRAY_SIZE(adau1781_dmic_dapm_routes));
 		if (ret)
 			return ret;
 	} else {
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
-			adau1781_adc_dapm_routes,
+		ret = snd_soc_dapm_add_routes(dapm, adau1781_adc_dapm_routes,
 			ARRAY_SIZE(adau1781_adc_dapm_routes));
 		if (ret)
 			return ret;

sound/soc/codecs/adau17x1.c

@@ -155,6 +155,7 @@ static int adau17x1_dsp_mux_enum_put(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
 	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
 	struct snd_soc_dapm_update update;
@@ -188,7 +189,7 @@ static int adau17x1_dsp_mux_enum_put(struct snd_kcontrol *kcontrol,
 		update.reg = reg;
 		update.val = val;
 
-		snd_soc_dapm_mux_update_power(&codec->dapm, kcontrol,
+		snd_soc_dapm_mux_update_power(dapm, kcontrol,
 				ucontrol->value.enumerated.item[0], e, &update);
 	}
 
@@ -444,8 +445,8 @@ static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id,
 static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai,
 		int clk_id, unsigned int freq, int dir)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec);
 	struct adau *adau = snd_soc_codec_get_drvdata(dai->codec);
-	struct snd_soc_dapm_context *dapm = &dai->codec->dapm;
 
 	switch (clk_id) {
 	case ADAU17X1_CLK_SRC_MCLK:
@@ -804,6 +805,7 @@ EXPORT_SYMBOL_GPL(adau17x1_setup_firmware);
 
 int adau17x1_add_widgets(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
@@ -811,14 +813,13 @@ int adau17x1_add_widgets(struct snd_soc_codec *codec)
 		ARRAY_SIZE(adau17x1_controls));
 	if (ret)
 		return ret;
-	ret = snd_soc_dapm_new_controls(&codec->dapm, adau17x1_dapm_widgets,
+	ret = snd_soc_dapm_new_controls(dapm, adau17x1_dapm_widgets,
 		ARRAY_SIZE(adau17x1_dapm_widgets));
 	if (ret)
 		return ret;
 
 	if (adau17x1_has_dsp(adau)) {
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
-			adau17x1_dsp_dapm_widgets,
+		ret = snd_soc_dapm_new_controls(dapm, adau17x1_dsp_dapm_widgets,
 			ARRAY_SIZE(adau17x1_dsp_dapm_widgets));
 		if (ret)
 			return ret;
@@ -840,21 +841,20 @@ EXPORT_SYMBOL_GPL(adau17x1_add_widgets);
 
 int adau17x1_add_routes(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau *adau = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	ret = snd_soc_dapm_add_routes(&codec->dapm, adau17x1_dapm_routes,
+	ret = snd_soc_dapm_add_routes(dapm, adau17x1_dapm_routes,
 		ARRAY_SIZE(adau17x1_dapm_routes));
 	if (ret)
 		return ret;
 
 	if (adau17x1_has_dsp(adau)) {
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
-			adau17x1_dsp_dapm_routes,
+		ret = snd_soc_dapm_add_routes(dapm, adau17x1_dsp_dapm_routes,
 			ARRAY_SIZE(adau17x1_dsp_dapm_routes));
 	} else {
-		ret = snd_soc_dapm_add_routes(&codec->dapm,
-			adau17x1_no_dsp_dapm_routes,
+		ret = snd_soc_dapm_add_routes(dapm, adau17x1_no_dsp_dapm_routes,
 			ARRAY_SIZE(adau17x1_no_dsp_dapm_routes));
 	}
 	return ret;

sound/soc/codecs/adau1977.c

@@ -202,7 +202,7 @@ static const struct snd_soc_dapm_route adau1977_dapm_routes[] = {
 		ADAU1977_REG_DC_HPF_CAL, (x) - 1, 1, 0)
 
 #define ADAU1977_DC_SUB_SWITCH(x) \
-	SOC_SINGLE("ADC" #x " DC Substraction Capture Switch", \
+	SOC_SINGLE("ADC" #x " DC Subtraction Capture Switch", \
 		ADAU1977_REG_DC_HPF_CAL, (x) + 3, 1, 0)
 
 static const struct snd_kcontrol_new adau1977_snd_controls[] = {
@@ -485,7 +485,7 @@ static int adau1977_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
 			ret = adau1977_power_enable(adau1977);
 		break;
 	case SND_SOC_BIAS_OFF:
@@ -493,12 +493,7 @@ static int adau1977_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	if (ret)
-		return ret;
-
-	codec->dapm.bias_level = level;
-
-	return 0;
+	return ret;
 }
 
 static int adau1977_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
@@ -853,12 +848,13 @@ static int adau1977_set_sysclk(struct snd_soc_codec *codec,
 
 static int adau1977_codec_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
 	switch (adau1977->type) {
 	case ADAU1977:
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
+		ret = snd_soc_dapm_new_controls(dapm,
 			adau1977_micbias_dapm_widgets,
 			ARRAY_SIZE(adau1977_micbias_dapm_widgets));
 		if (ret < 0)

sound/soc/codecs/adav80x.c

@@ -539,7 +539,7 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec,
 			      unsigned int freq, int dir)
 {
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	if (dir == SND_SOC_CLOCK_IN) {
 		switch (clk_id) {
@@ -622,6 +622,7 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec,
 static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id,
 		int source, unsigned int freq_in, unsigned int freq_out)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 	unsigned int pll_ctrl1 = 0;
 	unsigned int pll_ctrl2 = 0;
@@ -687,7 +688,7 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id,
 
 		adav80x->pll_src = source;
 
-		snd_soc_dapm_sync(&codec->dapm);
+		snd_soc_dapm_sync(dapm);
 	}
 
 	return 0;
@@ -714,7 +715,6 @@ static int adav80x_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -801,11 +801,12 @@ static struct snd_soc_dai_driver adav80x_dais[] = {
 
 static int adav80x_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
 
 	/* Force PLLs on for SYSCLK output */
-	snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1");
-	snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2");
+	snd_soc_dapm_force_enable_pin(dapm, "PLL1");
+	snd_soc_dapm_force_enable_pin(dapm, "PLL2");
 
 	/* Power down S/PDIF receiver, since it is currently not supported */
 	regmap_write(adav80x->regmap, ADAV80X_PLL_OUTE, 0x20);

sound/soc/codecs/ak4535.c

@@ -341,7 +341,6 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_update_bits(codec, AK4535_PM1, 0x80, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/ak4641.c

@@ -412,7 +412,7 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0x20);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			if (pdata && gpio_is_valid(pdata->gpio_power))
 				gpio_set_value(pdata->gpio_power, 1);
 			mdelay(1);
@@ -439,7 +439,6 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec,
 		regcache_mark_dirty(ak4641->regmap);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/ak4642.c

@@ -482,7 +482,6 @@ static int ak4642_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM);
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }

sound/soc/codecs/ak4671.c

@@ -577,7 +577,6 @@ static int ak4671_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/alc5623.c

@@ -826,7 +826,6 @@ static int alc5623_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -894,7 +893,7 @@ static int alc5623_resume(struct snd_soc_codec *codec)
 static int alc5623_probe(struct snd_soc_codec *codec)
 {
 	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	alc5623_reset(codec);
 

sound/soc/codecs/alc5632.c

@@ -1000,7 +1000,6 @@ static int alc5632_set_bias_level(struct snd_soc_codec *codec,
 				ALC5632_PWR_MANAG_ADD1_MASK, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/arizona.c

@@ -208,11 +208,12 @@ static const struct snd_soc_dapm_widget arizona_spkr =
 
 int arizona_init_spk(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
 	struct arizona *arizona = priv->arizona;
 	int ret;
 
-	ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkl, 1);
+	ret = snd_soc_dapm_new_controls(dapm, &arizona_spkl, 1);
 	if (ret != 0)
 		return ret;
 
@@ -220,8 +221,7 @@ int arizona_init_spk(struct snd_soc_codec *codec)
 	case WM8997:
 		break;
 	default:
-		ret = snd_soc_dapm_new_controls(&codec->dapm,
-						&arizona_spkr, 1);
+		ret = snd_soc_dapm_new_controls(dapm, &arizona_spkr, 1);
 		if (ret != 0)
 			return ret;
 		break;
@@ -258,13 +258,14 @@ static const struct snd_soc_dapm_route arizona_mono_routes[] = {
 
 int arizona_init_mono(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
 	struct arizona *arizona = priv->arizona;
 	int i;
 
 	for (i = 0; i < ARIZONA_MAX_OUTPUT; ++i) {
 		if (arizona->pdata.out_mono[i])
-			snd_soc_dapm_add_routes(&codec->dapm,
+			snd_soc_dapm_add_routes(dapm,
 						&arizona_mono_routes[i], 1);
 	}
 
@@ -274,6 +275,7 @@ EXPORT_SYMBOL_GPL(arizona_init_mono);
 
 int arizona_init_gpio(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
 	struct arizona *arizona = priv->arizona;
 	int i;
@@ -281,23 +283,21 @@ int arizona_init_gpio(struct snd_soc_codec *codec)
 	switch (arizona->type) {
 	case WM5110:
 	case WM8280:
-		snd_soc_dapm_disable_pin(&codec->dapm, "DRC2 Signal Activity");
+		snd_soc_dapm_disable_pin(dapm, "DRC2 Signal Activity");
 		break;
 	default:
 		break;
 	}
 
-	snd_soc_dapm_disable_pin(&codec->dapm, "DRC1 Signal Activity");
+	snd_soc_dapm_disable_pin(dapm, "DRC1 Signal Activity");
 
 	for (i = 0; i < ARRAY_SIZE(arizona->pdata.gpio_defaults); i++) {
 		switch (arizona->pdata.gpio_defaults[i] & ARIZONA_GPN_FN_MASK) {
 		case ARIZONA_GP_FN_DRC1_SIGNAL_DETECT:
-			snd_soc_dapm_enable_pin(&codec->dapm,
-						"DRC1 Signal Activity");
+			snd_soc_dapm_enable_pin(dapm, "DRC1 Signal Activity");
 			break;
 		case ARIZONA_GP_FN_DRC2_SIGNAL_DETECT:
-			snd_soc_dapm_enable_pin(&codec->dapm,
-						"DRC2 Signal Activity");
+			snd_soc_dapm_enable_pin(dapm, "DRC2 Signal Activity");
 			break;
 		default:
 			break;
@@ -851,6 +851,134 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w,
 }
 EXPORT_SYMBOL_GPL(arizona_hp_ev);
 
+static int arizona_dvfs_enable(struct snd_soc_codec *codec)
+{
+	const struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->arizona;
+	int ret;
+
+	ret = regulator_set_voltage(arizona->dcvdd, 1800000, 1800000);
+	if (ret) {
+		dev_err(codec->dev, "Failed to boost DCVDD: %d\n", ret);
+		return ret;
+	}
+
+	ret = regmap_update_bits(arizona->regmap,
+				 ARIZONA_DYNAMIC_FREQUENCY_SCALING_1,
+				 ARIZONA_SUBSYS_MAX_FREQ,
+				 ARIZONA_SUBSYS_MAX_FREQ);
+	if (ret) {
+		dev_err(codec->dev, "Failed to enable subsys max: %d\n", ret);
+		regulator_set_voltage(arizona->dcvdd, 1200000, 1800000);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int arizona_dvfs_disable(struct snd_soc_codec *codec)
+{
+	const struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->arizona;
+	int ret;
+
+	ret = regmap_update_bits(arizona->regmap,
+				 ARIZONA_DYNAMIC_FREQUENCY_SCALING_1,
+				 ARIZONA_SUBSYS_MAX_FREQ, 0);
+	if (ret) {
+		dev_err(codec->dev, "Failed to disable subsys max: %d\n", ret);
+		return ret;
+	}
+
+	ret = regulator_set_voltage(arizona->dcvdd, 1200000, 1800000);
+	if (ret) {
+		dev_err(codec->dev, "Failed to unboost DCVDD: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+int arizona_dvfs_up(struct snd_soc_codec *codec, unsigned int flags)
+{
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	mutex_lock(&priv->dvfs_lock);
+
+	if (!priv->dvfs_cached && !priv->dvfs_reqs) {
+		ret = arizona_dvfs_enable(codec);
+		if (ret)
+			goto err;
+	}
+
+	priv->dvfs_reqs |= flags;
+err:
+	mutex_unlock(&priv->dvfs_lock);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(arizona_dvfs_up);
+
+int arizona_dvfs_down(struct snd_soc_codec *codec, unsigned int flags)
+{
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	unsigned int old_reqs;
+	int ret = 0;
+
+	mutex_lock(&priv->dvfs_lock);
+
+	old_reqs = priv->dvfs_reqs;
+	priv->dvfs_reqs &= ~flags;
+
+	if (!priv->dvfs_cached && old_reqs && !priv->dvfs_reqs)
+		ret = arizona_dvfs_disable(codec);
+
+	mutex_unlock(&priv->dvfs_lock);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(arizona_dvfs_down);
+
+int arizona_dvfs_sysclk_ev(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	mutex_lock(&priv->dvfs_lock);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		if (priv->dvfs_reqs)
+			ret = arizona_dvfs_enable(codec);
+
+		priv->dvfs_cached = false;
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		/* We must ensure DVFS is disabled before the codec goes into
+		 * suspend so that we are never in an illegal state of DVFS
+		 * enabled without enough DCVDD
+		 */
+		priv->dvfs_cached = true;
+
+		if (priv->dvfs_reqs)
+			ret = arizona_dvfs_disable(codec);
+		break;
+	default:
+		break;
+	}
+
+	mutex_unlock(&priv->dvfs_lock);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(arizona_dvfs_sysclk_ev);
+
+void arizona_init_dvfs(struct arizona_priv *priv)
+{
+	mutex_init(&priv->dvfs_lock);
+}
+EXPORT_SYMBOL_GPL(arizona_init_dvfs);
+
 static unsigned int arizona_sysclk_48k_rates[] = {
 	6144000,
 	12288000,
@@ -1266,7 +1394,7 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream,
 	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
 	struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
 	int base = dai->driver->base;
-	int i, sr_val;
+	int i, sr_val, ret;
 
 	/*
 	 * We will need to be more flexible than this in future,
@@ -1282,6 +1410,23 @@ static int arizona_hw_params_rate(struct snd_pcm_substream *substream,
 	}
 	sr_val = i;
 
+	switch (priv->arizona->type) {
+	case WM5102:
+	case WM8997:
+		if (arizona_sr_vals[sr_val] >= 88200)
+			ret = arizona_dvfs_up(codec, ARIZONA_DVFS_SR1_RQ);
+		else
+			ret = arizona_dvfs_down(codec, ARIZONA_DVFS_SR1_RQ);
+
+		if (ret) {
+			arizona_aif_err(dai, "Failed to change DVFS %d\n", ret);
+			return ret;
+		}
+		break;
+	default:
+		break;
+	}
+
 	switch (dai_priv->clk) {
 	case ARIZONA_CLK_SYSCLK:
 		switch (priv->arizona->type) {
@@ -1474,6 +1619,7 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai,
 				  int clk_id, unsigned int freq, int dir)
 {
 	struct snd_soc_codec *codec = dai->codec;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
 	struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
 	struct snd_soc_dapm_route routes[2];
@@ -1504,15 +1650,15 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai,
 
 	routes[0].source = arizona_dai_clk_str(dai_priv->clk);
 	routes[1].source = arizona_dai_clk_str(dai_priv->clk);
-	snd_soc_dapm_del_routes(&codec->dapm, routes, ARRAY_SIZE(routes));
+	snd_soc_dapm_del_routes(dapm, routes, ARRAY_SIZE(routes));
 
 	routes[0].source = arizona_dai_clk_str(clk_id);
 	routes[1].source = arizona_dai_clk_str(clk_id);
-	snd_soc_dapm_add_routes(&codec->dapm, routes, ARRAY_SIZE(routes));
+	snd_soc_dapm_add_routes(dapm, routes, ARRAY_SIZE(routes));
 
 	dai_priv->clk = clk_id;
 
-	return snd_soc_dapm_sync(&codec->dapm);
+	return snd_soc_dapm_sync(dapm);
 }
 
 static int arizona_set_tristate(struct snd_soc_dai *dai, int tristate)

sound/soc/codecs/arizona.h

@@ -60,6 +60,9 @@
 #define ARIZONA_MAX_DAI  6
 #define ARIZONA_MAX_ADSP 4
 
+#define ARIZONA_DVFS_SR1_RQ	0x001
+#define ARIZONA_DVFS_ADSP1_RQ	0x100
+
 struct arizona;
 struct wm_adsp;
 
@@ -84,6 +87,10 @@ struct arizona_priv {
 
 	unsigned int spk_ena:2;
 	unsigned int spk_ena_pending:1;
+
+	unsigned int dvfs_reqs;
+	struct mutex dvfs_lock;
+	bool dvfs_cached;
 };
 
 #define ARIZONA_NUM_MIXER_INPUTS 103
@@ -107,8 +114,8 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
 			     arizona_mixer_tlv)
 
 #define ARIZONA_MUX_ENUM_DECL(name, reg) \
-	SOC_VALUE_ENUM_SINGLE_DECL(name, reg, 0, 0xff,			\
-				   arizona_mixer_texts, arizona_mixer_values)
+	SOC_VALUE_ENUM_SINGLE_AUTODISABLE_DECL( \
+		name, reg, 0, 0xff, arizona_mixer_texts, arizona_mixer_values)
 
 #define ARIZONA_MUX_CTL_DECL(name) \
 	const struct snd_kcontrol_new name##_mux =	\
@@ -245,6 +252,12 @@ struct arizona_fll {
 	char clock_ok_name[ARIZONA_FLL_NAME_LEN];
 };
 
+extern int arizona_dvfs_up(struct snd_soc_codec *codec, unsigned int flags);
+extern int arizona_dvfs_down(struct snd_soc_codec *codec, unsigned int flags);
+extern int arizona_dvfs_sysclk_ev(struct snd_soc_dapm_widget *w,
+				  struct snd_kcontrol *kcontrol, int event);
+extern void arizona_init_dvfs(struct arizona_priv *priv);
+
 extern int arizona_init_fll(struct arizona *arizona, int id, int base,
 			    int lock_irq, int ok_irq, struct arizona_fll *fll);
 extern int arizona_set_fll_refclk(struct arizona_fll *fll, int source,

sound/soc/codecs/bt-sco.c

@@ -63,7 +63,7 @@ static int bt_sco_remove(struct platform_device *pdev)
 	return 0;
 }
 
-static struct platform_device_id bt_sco_driver_ids[] = {
+static const struct platform_device_id bt_sco_driver_ids[] = {
 	{
 		.name		= "dfbmcs320",
 	},
@@ -74,9 +74,18 @@ static struct platform_device_id bt_sco_driver_ids[] = {
 };
 MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids);
 
+#if defined(CONFIG_OF)
+static const struct of_device_id bt_sco_codec_of_match[] = {
+	{ .compatible = "delta,dfbmcs320", },
+	{},
+};
+MODULE_DEVICE_TABLE(of, bt_sco_codec_of_match);
+#endif
+
 static struct platform_driver bt_sco_driver = {
 	.driver = {
 		.name = "bt-sco",
+		.of_match_table = of_match_ptr(bt_sco_codec_of_match),
 	},
 	.probe = bt_sco_probe,
 	.remove = bt_sco_remove,

sound/soc/codecs/cq93vc.c

@@ -92,7 +92,6 @@ static int cq93vc_set_bias_level(struct snd_soc_codec *codec,
 			     DAVINCI_VC_REG12_POWER_ALL_OFF);
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }

sound/soc/codecs/cs35l32.c

@@ -13,7 +13,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>

sound/soc/codecs/cs4265.c

@@ -503,7 +503,6 @@ static int cs4265_set_bias_level(struct snd_soc_codec *codec,
 			CS4265_PWRCTL_PDN);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/cs42l52.c

@@ -897,7 +897,7 @@ static int cs42l52_set_bias_level(struct snd_soc_codec *codec,
 				    CS42L52_PWRCTL1_PDN_CODEC, 0);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			regcache_cache_only(cs42l52->regmap, false);
 			regcache_sync(cs42l52->regmap);
 		}
@@ -908,7 +908,6 @@ static int cs42l52_set_bias_level(struct snd_soc_codec *codec,
 		regcache_cache_only(cs42l52->regmap, true);
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -956,7 +955,7 @@ static void cs42l52_beep_work(struct work_struct *work)
 	struct cs42l52_private *cs42l52 =
 		container_of(work, struct cs42l52_private, beep_work);
 	struct snd_soc_codec *codec = cs42l52->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	int i;
 	int val = 0;
 	int best = 0;

sound/soc/codecs/cs42l56.c

@@ -953,7 +953,7 @@ static int cs42l56_set_bias_level(struct snd_soc_codec *codec,
 				    CS42L56_PDN_ALL_MASK, 0);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			regcache_cache_only(cs42l56->regmap, false);
 			regcache_sync(cs42l56->regmap);
 			ret = regulator_bulk_enable(ARRAY_SIZE(cs42l56->supplies),
@@ -978,7 +978,6 @@ static int cs42l56_set_bias_level(struct snd_soc_codec *codec,
 						    cs42l56->supplies);
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -1026,7 +1025,7 @@ static void cs42l56_beep_work(struct work_struct *work)
 	struct cs42l56_private *cs42l56 =
 		container_of(work, struct cs42l56_private, beep_work);
 	struct snd_soc_codec *codec = cs42l56->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	int i;
 	int val = 0;
 	int best = 0;

sound/soc/codecs/cs42l73.c

@@ -1208,7 +1208,7 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			regcache_cache_only(cs42l73->regmap, false);
 			regcache_sync(cs42l73->regmap);
 		}
@@ -1228,7 +1228,6 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec,
 		snd_soc_update_bits(codec, CS42L73_DMMCC, CS42L73_MCLKDIS, 1);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/cs42xx8.c

@@ -380,7 +380,7 @@ EXPORT_SYMBOL_GPL(cs42xx8_regmap_config);
 static int cs42xx8_codec_probe(struct snd_soc_codec *codec)
 {
 	struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	switch (cs42xx8->drvdata->num_adcs) {
 	case 3:

sound/soc/codecs/cx20442.c

@@ -333,7 +333,7 @@ static int cx20442_set_bias_level(struct snd_soc_codec *codec,
 
 	switch (level) {
 	case SND_SOC_BIAS_PREPARE:
-		if (codec->dapm.bias_level != SND_SOC_BIAS_STANDBY)
+		if (snd_soc_codec_get_bias_level(codec) != SND_SOC_BIAS_STANDBY)
 			break;
 		if (IS_ERR(cx20442->por))
 			err = PTR_ERR(cx20442->por);
@@ -341,7 +341,7 @@ static int cx20442_set_bias_level(struct snd_soc_codec *codec,
 			err = regulator_enable(cx20442->por);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level != SND_SOC_BIAS_PREPARE)
+		if (snd_soc_codec_get_bias_level(codec) != SND_SOC_BIAS_PREPARE)
 			break;
 		if (IS_ERR(cx20442->por))
 			err = PTR_ERR(cx20442->por);
@@ -351,8 +351,6 @@ static int cx20442_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	if (!err)
-		codec->dapm.bias_level = level;
 
 	return err;
 }

sound/soc/codecs/da7213.c

@@ -1374,7 +1374,7 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			/* Enable VMID reference & master bias */
 			snd_soc_update_bits(codec, DA7213_REFERENCES,
 					    DA7213_VMID_EN | DA7213_BIAS_EN,
@@ -1387,7 +1387,6 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec,
 				    DA7213_VMID_EN | DA7213_BIAS_EN, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/da732x.c

@@ -1432,7 +1432,7 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			/* Init Codec */
 			snd_soc_write(codec, DA732X_REG_REF1,
 				      DA732X_VMID_FASTCHG);
@@ -1502,8 +1502,6 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/da9055.c

@@ -1364,7 +1364,7 @@ static int da9055_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			/* Enable VMID reference & master bias */
 			snd_soc_update_bits(codec, DA9055_REFERENCES,
 					    DA9055_VMID_EN | DA9055_BIAS_EN,
@@ -1377,7 +1377,6 @@ static int da9055_set_bias_level(struct snd_soc_codec *codec,
 				    DA9055_VMID_EN | DA9055_BIAS_EN, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/es8328.c

@@ -536,7 +536,7 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			snd_soc_update_bits(codec, ES8328_CONTROL1,
 					ES8328_CONTROL1_VMIDSEL_MASK |
 					ES8328_CONTROL1_ENREF,
@@ -566,7 +566,6 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec,
 				0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/isabelle.c

@@ -909,8 +909,6 @@ static int isabelle_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/jz4740.c

@@ -258,7 +258,7 @@ static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		/* The only way to clear the suspend flag is to reset the codec */
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
 			jz4740_codec_wakeup(regmap);
 
 		mask = JZ4740_CODEC_1_VREF_DISABLE |
@@ -281,8 +281,6 @@ static int jz4740_codec_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/lm4857.c

@@ -23,11 +23,6 @@
 #include <sound/soc.h>
 #include <sound/tlv.h>
 
-struct lm4857 {
-	struct regmap *regmap;
-	uint8_t mode;
-};
-
 static const struct reg_default lm4857_default_regs[] = {
 	{ 0x0, 0x00 },
 	{ 0x1, 0x00 },
@@ -46,66 +41,33 @@ static const struct reg_default lm4857_default_regs[] = {
 #define LM4857_WAKEUP 5
 #define LM4857_EPGAIN 4
 
-static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
-	struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
-
-	ucontrol->value.integer.value[0] = lm4857->mode;
-
-	return 0;
-}
-
-static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
-	struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
-	uint8_t value = ucontrol->value.integer.value[0];
-
-	lm4857->mode = value;
-
-	if (codec->dapm.bias_level == SND_SOC_BIAS_ON)
-		regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, value + 6);
-
-	return 1;
-}
-
-static int lm4857_set_bias_level(struct snd_soc_codec *codec,
-				 enum snd_soc_bias_level level)
-{
-	struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
-
-	switch (level) {
-	case SND_SOC_BIAS_ON:
-		regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F,
-			lm4857->mode + 6);
-		break;
-	case SND_SOC_BIAS_STANDBY:
-		regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, 0);
-		break;
-	default:
-		break;
-	}
-
-	codec->dapm.bias_level = level;
-
-	return 0;
-}
+static const unsigned int lm4857_mode_values[] = {
+	0,
+	6,
+	7,
+	8,
+	9,
+};
 
-static const char *lm4857_mode[] = {
+static const char * const lm4857_mode_texts[] = {
+	"Off",
 	"Earpiece",
 	"Loudspeaker",
 	"Loudspeaker + Headphone",
 	"Headphone",
 };
 
-static SOC_ENUM_SINGLE_EXT_DECL(lm4857_mode_enum, lm4857_mode);
+static SOC_VALUE_ENUM_SINGLE_AUTODISABLE_DECL(lm4857_mode_enum,
+	LM4857_CTRL, 0, 0xf, lm4857_mode_texts, lm4857_mode_values);
+
+static const struct snd_kcontrol_new lm4857_mode_ctrl =
+	SOC_DAPM_ENUM("Mode", lm4857_mode_enum);
 
 static const struct snd_soc_dapm_widget lm4857_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("IN"),
 
+	SND_SOC_DAPM_DEMUX("Mode", SND_SOC_NOPM, 0, 0, &lm4857_mode_ctrl),
+
 	SND_SOC_DAPM_OUTPUT("LS"),
 	SND_SOC_DAPM_OUTPUT("HP"),
 	SND_SOC_DAPM_OUTPUT("EP"),
@@ -127,24 +89,18 @@ static const struct snd_kcontrol_new lm4857_controls[] = {
 		LM4857_WAKEUP, 1, 0),
 	SOC_SINGLE("Earpiece 6dB Playback Switch", LM4857_CTRL,
 		LM4857_EPGAIN, 1, 0),
-
-	SOC_ENUM_EXT("Mode", lm4857_mode_enum,
-		lm4857_get_mode, lm4857_set_mode),
 };
 
-/* There is a demux between the input signal and the output signals.
- * Currently there is no easy way to model it in ASoC and since it does not make
- * much of a difference in practice simply connect the input direclty to the
- * outputs. */
 static const struct snd_soc_dapm_route lm4857_routes[] = {
-	{"LS", NULL, "IN"},
-	{"HP", NULL, "IN"},
-	{"EP", NULL, "IN"},
+	{ "Mode", NULL, "IN" },
+	{ "LS", "Loudspeaker", "Mode" },
+	{ "LS", "Loudspeaker + Headphone", "Mode" },
+	{ "HP", "Headphone", "Mode" },
+	{ "HP", "Loudspeaker + Headphone", "Mode" },
+	{ "EP", "Earpiece", "Mode" },
 };
 
-static struct snd_soc_codec_driver soc_codec_dev_lm4857 = {
-	.set_bias_level = lm4857_set_bias_level,
-
+static struct snd_soc_component_driver lm4857_component_driver = {
 	.controls = lm4857_controls,
 	.num_controls = ARRAY_SIZE(lm4857_controls),
 	.dapm_widgets = lm4857_dapm_widgets,
@@ -167,25 +123,14 @@ static const struct regmap_config lm4857_regmap_config = {
 static int lm4857_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-	struct lm4857 *lm4857;
-
-	lm4857 = devm_kzalloc(&i2c->dev, sizeof(*lm4857), GFP_KERNEL);
-	if (!lm4857)
-		return -ENOMEM;
-
-	i2c_set_clientdata(i2c, lm4857);
-
-	lm4857->regmap = devm_regmap_init_i2c(i2c, &lm4857_regmap_config);
-	if (IS_ERR(lm4857->regmap))
-		return PTR_ERR(lm4857->regmap);
+	struct regmap *regmap;
 
-	return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0);
-}
+	regmap = devm_regmap_init_i2c(i2c, &lm4857_regmap_config);
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
 
-static int lm4857_i2c_remove(struct i2c_client *i2c)
-{
-	snd_soc_unregister_codec(&i2c->dev);
-	return 0;
+	return devm_snd_soc_register_component(&i2c->dev,
+		&lm4857_component_driver, NULL, 0);
 }
 
 static const struct i2c_device_id lm4857_i2c_id[] = {
@@ -200,7 +145,6 @@ static struct i2c_driver lm4857_i2c_driver = {
 		.owner = THIS_MODULE,
 	},
 	.probe = lm4857_i2c_probe,
-	.remove = lm4857_i2c_remove,
 	.id_table = lm4857_i2c_id,
 };
 

sound/soc/codecs/lm49453.c

@@ -1271,7 +1271,7 @@ static int lm49453_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
 			regcache_sync(lm49453->regmap);
 
 		snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG,
@@ -1284,8 +1284,6 @@ static int lm49453_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/max98088.c

@@ -1571,7 +1571,7 @@ static int max98088_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
 			regcache_sync(max98088->regmap);
 
 		snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN,
@@ -1584,7 +1584,6 @@ static int max98088_set_bias_level(struct snd_soc_codec *codec,
 		regcache_mark_dirty(max98088->regmap);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/max98090.c

@@ -1500,7 +1500,7 @@ static const struct snd_soc_dapm_route max98091_dapm_routes[] = {
 static int max98090_add_widgets(struct snd_soc_codec *codec)
 {
 	struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	snd_soc_add_codec_controls(codec, max98090_snd_controls,
 		ARRAY_SIZE(max98090_snd_controls));
@@ -1798,16 +1798,17 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec,
 		 * away from ON. Disable the clock in that case, otherwise
 		 * enable it.
 		 */
-		if (!IS_ERR(max98090->mclk)) {
-			if (codec->dapm.bias_level == SND_SOC_BIAS_ON)
-				clk_disable_unprepare(max98090->mclk);
-			else
-				clk_prepare_enable(max98090->mclk);
-		}
+		if (IS_ERR(max98090->mclk))
+			break;
+
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON)
+			clk_disable_unprepare(max98090->mclk);
+		else
+			clk_prepare_enable(max98090->mclk);
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regcache_sync(max98090->regmap);
 			if (ret != 0) {
 				dev_err(codec->dev,
@@ -1824,7 +1825,6 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec,
 		regcache_mark_dirty(max98090->regmap);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -2187,7 +2187,6 @@ static void max98090_jack_work(struct work_struct *work)
 		struct max98090_priv,
 		jack_work.work);
 	struct snd_soc_codec *codec = max98090->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int status = 0;
 	int reg;
 
@@ -2266,8 +2265,6 @@ static void max98090_jack_work(struct work_struct *work)
 
 	snd_soc_jack_report(max98090->jack, status,
 			    SND_JACK_HEADSET | SND_JACK_BTN_0);
-
-	snd_soc_dapm_sync(dapm);
 }
 
 static irqreturn_t max98090_interrupt(int irq, void *data)
@@ -2422,6 +2419,8 @@ static int max98090_probe(struct snd_soc_codec *codec)
 	struct max98090_cdata *cdata;
 	enum max98090_type devtype;
 	int ret = 0;
+	int err;
+	unsigned int micbias;
 
 	dev_dbg(codec->dev, "max98090_probe\n");
 
@@ -2506,8 +2505,17 @@ static int max98090_probe(struct snd_soc_codec *codec)
 	snd_soc_write(codec, M98090_REG_BIAS_CONTROL,
 		M98090_VCM_MODE_MASK);
 
+	err = device_property_read_u32(codec->dev, "maxim,micbias", &micbias);
+	if (err) {
+		micbias = M98090_MBVSEL_2V8;
+		dev_info(codec->dev, "use default 2.8v micbias\n");
+	} else if (micbias < M98090_MBVSEL_2V2 || micbias > M98090_MBVSEL_2V8) {
+		dev_err(codec->dev, "micbias out of range 0x%x\n", micbias);
+		micbias = M98090_MBVSEL_2V8;
+	}
+
 	snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE,
-		M98090_MBVSEL_MASK, M98090_MBVSEL_2V8);
+		M98090_MBVSEL_MASK, micbias);
 
 	max98090_add_widgets(codec);
 

sound/soc/codecs/max98095.c

@@ -1650,16 +1650,17 @@ static int max98095_set_bias_level(struct snd_soc_codec *codec,
 		 * away from ON. Disable the clock in that case, otherwise
 		 * enable it.
 		 */
-		if (!IS_ERR(max98095->mclk)) {
-			if (codec->dapm.bias_level == SND_SOC_BIAS_ON)
-				clk_disable_unprepare(max98095->mclk);
-			else
-				clk_prepare_enable(max98095->mclk);
-		}
+		if (IS_ERR(max98095->mclk))
+			break;
+
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON)
+			clk_disable_unprepare(max98095->mclk);
+		else
+			clk_prepare_enable(max98095->mclk);
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regcache_sync(max98095->regmap);
 
 			if (ret != 0) {
@@ -1678,7 +1679,6 @@ static int max98095_set_bias_level(struct snd_soc_codec *codec,
 		regcache_mark_dirty(max98095->regmap);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 
@@ -2198,7 +2198,7 @@ static int max98095_suspend(struct snd_soc_codec *codec)
 	if (max98095->headphone_jack || max98095->mic_jack)
 		max98095_jack_detect_disable(codec);
 
-	max98095_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
 }
@@ -2208,7 +2208,7 @@ static int max98095_resume(struct snd_soc_codec *codec)
 	struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
 	struct i2c_client *client = to_i2c_client(codec->dev);
 
-	max98095_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	if (max98095->headphone_jack || max98095->mic_jack) {
 		max98095_jack_detect_enable(codec);
@@ -2301,8 +2301,8 @@ static int max98095_probe(struct snd_soc_codec *codec)
 		/* register an audio interrupt */
 		ret = request_threaded_irq(client->irq, NULL,
 			max98095_report_jack,
-			IRQF_TRIGGER_FALLING | IRQF_TRIGGER_RISING,
-			"max98095", codec);
+			IRQF_TRIGGER_FALLING | IRQF_TRIGGER_RISING |
+			IRQF_ONESHOT, "max98095", codec);
 		if (ret) {
 			dev_err(codec->dev, "Failed to request IRQ: %d\n", ret);
 			goto err_access;

sound/soc/codecs/max98357a.c

@@ -60,13 +60,12 @@ static int max98357a_codec_probe(struct snd_soc_codec *codec)
 {
 	struct gpio_desc *sdmode;
 
-	sdmode = devm_gpiod_get(codec->dev, "sdmode");
+	sdmode = devm_gpiod_get(codec->dev, "sdmode", GPIOD_OUT_LOW);
 	if (IS_ERR(sdmode)) {
 		dev_err(codec->dev, "%s() unable to get sdmode GPIO: %ld\n",
 				__func__, PTR_ERR(sdmode));
 		return PTR_ERR(sdmode);
 	}
-	gpiod_direction_output(sdmode, 0);
 	snd_soc_codec_set_drvdata(codec, sdmode);
 
 	return 0;

sound/soc/codecs/max9850.c

@@ -252,7 +252,7 @@ static int max9850_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			ret = regcache_sync(max9850->regmap);
 			if (ret) {
 				dev_err(codec->dev,
@@ -264,7 +264,6 @@ static int max9850_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_OFF:
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/ml26124.c

@@ -523,7 +523,7 @@ static int ml26124_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		/* VMID ON */
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG,
 					    ML26124_VMID, ML26124_VMID);
 			msleep(500);
@@ -536,7 +536,6 @@ static int ml26124_set_bias_level(struct snd_soc_codec *codec,
 				    ML26124_VMID, 0);
 		break;
 	}
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/pcm512x.c

@@ -242,7 +242,7 @@ static int pcm512x_overclock_pll_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
 
-	switch (codec->dapm.bias_level) {
+	switch (snd_soc_codec_get_bias_level(codec)) {
 	case SND_SOC_BIAS_OFF:
 	case SND_SOC_BIAS_STANDBY:
 		break;
@@ -270,7 +270,7 @@ static int pcm512x_overclock_dsp_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
 
-	switch (codec->dapm.bias_level) {
+	switch (snd_soc_codec_get_bias_level(codec)) {
 	case SND_SOC_BIAS_OFF:
 	case SND_SOC_BIAS_STANDBY:
 		break;
@@ -298,7 +298,7 @@ static int pcm512x_overclock_dac_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	struct pcm512x_priv *pcm512x = snd_soc_codec_get_drvdata(codec);
 
-	switch (codec->dapm.bias_level) {
+	switch (snd_soc_codec_get_bias_level(codec)) {
 	case SND_SOC_BIAS_OFF:
 	case SND_SOC_BIAS_STANDBY:
 		break;
@@ -641,8 +641,6 @@ static int pcm512x_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
-
 	return 0;
 }
 

sound/soc/codecs/rt286.c

@@ -301,6 +301,7 @@ static int rt286_support_power_controls[] = {
 
 static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
 {
+	struct snd_soc_dapm_context *dapm;
 	unsigned int val, buf;
 
 	*hp = false;
@@ -308,6 +309,9 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
 
 	if (!rt286->codec)
 		return -EINVAL;
+
+	dapm = snd_soc_codec_get_dapm(rt286->codec);
+
 	if (rt286->pdata.cbj_en) {
 		regmap_read(rt286->regmap, RT286_GET_HP_SENSE, &buf);
 		*hp = buf & 0x80000000;
@@ -316,14 +320,11 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
 			regmap_update_bits(rt286->regmap,
 				RT286_DC_GAIN, 0x200, 0x200);
 
-			snd_soc_dapm_force_enable_pin(&rt286->codec->dapm,
-							"HV");
-			snd_soc_dapm_force_enable_pin(&rt286->codec->dapm,
-							"VREF");
+			snd_soc_dapm_force_enable_pin(dapm, "HV");
+			snd_soc_dapm_force_enable_pin(dapm, "VREF");
 			/* power LDO1 */
-			snd_soc_dapm_force_enable_pin(&rt286->codec->dapm,
-							"LDO1");
-			snd_soc_dapm_sync(&rt286->codec->dapm);
+			snd_soc_dapm_force_enable_pin(dapm, "LDO1");
+			snd_soc_dapm_sync(dapm);
 
 			regmap_write(rt286->regmap, RT286_SET_MIC1, 0x24);
 			msleep(50);
@@ -360,11 +361,11 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
 		*mic = buf & 0x80000000;
 	}
 
-	snd_soc_dapm_disable_pin(&rt286->codec->dapm, "HV");
-	snd_soc_dapm_disable_pin(&rt286->codec->dapm, "VREF");
+	snd_soc_dapm_disable_pin(dapm, "HV");
+	snd_soc_dapm_disable_pin(dapm, "VREF");
 	if (!*hp)
-		snd_soc_dapm_disable_pin(&rt286->codec->dapm, "LDO1");
-	snd_soc_dapm_sync(&rt286->codec->dapm);
+		snd_soc_dapm_disable_pin(dapm, "LDO1");
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
@@ -391,6 +392,7 @@ static void rt286_jack_detect_work(struct work_struct *work)
 
 int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec);
 
 	rt286->jack = jack;
@@ -398,7 +400,7 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
 	if (jack) {
 		/* enable IRQ */
 		if (rt286->jack->status & SND_JACK_HEADPHONE)
-			snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO1");
+			snd_soc_dapm_force_enable_pin(dapm, "LDO1");
 		regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x2);
 		/* Send an initial empty report */
 		snd_soc_jack_report(rt286->jack, rt286->jack->status,
@@ -406,9 +408,9 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
 	} else {
 		/* disable IRQ */
 		regmap_update_bits(rt286->regmap, RT286_IRQ_CTRL, 0x2, 0x0);
-		snd_soc_dapm_disable_pin(&codec->dapm, "LDO1");
+		snd_soc_dapm_disable_pin(dapm, "LDO1");
 	}
-	snd_soc_dapm_sync(&codec->dapm);
+	snd_soc_dapm_sync(dapm);
 
 	return 0;
 }
@@ -985,7 +987,7 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec,
 {
 	switch (level) {
 	case SND_SOC_BIAS_PREPARE:
-		if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) {
+		if (SND_SOC_BIAS_STANDBY == snd_soc_codec_get_bias_level(codec)) {
 			snd_soc_write(codec,
 				RT286_SET_AUDIO_POWER, AC_PWRST_D0);
 			snd_soc_update_bits(codec,
@@ -1012,7 +1014,6 @@ static int rt286_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }

sound/soc/codecs/rt5631.c

@@ -1546,7 +1546,7 @@ static int rt5631_set_bias_level(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
 			snd_soc_update_bits(codec, RT5631_PWR_MANAG_ADD3,
 				RT5631_PWR_VREF | RT5631_PWR_MAIN_BIAS,
 				RT5631_PWR_VREF | RT5631_PWR_MAIN_BIAS);
@@ -1569,7 +1569,6 @@ static int rt5631_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -1615,7 +1614,7 @@ static int rt5631_probe(struct snd_soc_codec *codec)
 			RT5631_DMIC_R_CH_LATCH_RISING);
 	}
 
-	codec->dapm.bias_level = SND_SOC_BIAS_STANDBY;
+	snd_soc_codec_init_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
 }

sound/soc/codecs/rt5640.c

@@ -1870,7 +1870,7 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec,
 {
 	switch (level) {
 	case SND_SOC_BIAS_STANDBY:
-		if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) {
+		if (SND_SOC_BIAS_OFF == snd_soc_codec_get_bias_level(codec)) {
 			snd_soc_update_bits(codec, RT5640_PWR_ANLG1,
 				RT5640_PWR_VREF1 | RT5640_PWR_MB |
 				RT5640_PWR_BG | RT5640_PWR_VREF2,
@@ -1902,7 +1902,6 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -1935,11 +1934,12 @@ EXPORT_SYMBOL_GPL(rt5640_dmic_enable);
 
 static int rt5640_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
 
 	rt5640->codec = codec;
 
-	rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301);
 	snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030);
@@ -1951,18 +1951,18 @@ static int rt5640_probe(struct snd_soc_codec *codec)
 		snd_soc_add_codec_controls(codec,
 			rt5640_specific_snd_controls,
 			ARRAY_SIZE(rt5640_specific_snd_controls));
-		snd_soc_dapm_new_controls(&codec->dapm,
+		snd_soc_dapm_new_controls(dapm,
 			rt5640_specific_dapm_widgets,
 			ARRAY_SIZE(rt5640_specific_dapm_widgets));
-		snd_soc_dapm_add_routes(&codec->dapm,
+		snd_soc_dapm_add_routes(dapm,
 			rt5640_specific_dapm_routes,
 			ARRAY_SIZE(rt5640_specific_dapm_routes));
 		break;
 	case RT5640_ID_5639:
-		snd_soc_dapm_new_controls(&codec->dapm,
+		snd_soc_dapm_new_controls(dapm,
 			rt5639_specific_dapm_widgets,
 			ARRAY_SIZE(rt5639_specific_dapm_widgets));
-		snd_soc_dapm_add_routes(&codec->dapm,
+		snd_soc_dapm_add_routes(dapm,
 			rt5639_specific_dapm_routes,
 			ARRAY_SIZE(rt5639_specific_dapm_routes));
 		break;
@@ -1991,7 +1991,7 @@ static int rt5640_suspend(struct snd_soc_codec *codec)
 {
 	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
 
-	rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
 	rt5640_reset(codec);
 	regcache_cache_only(rt5640->regmap, true);
 	regcache_mark_dirty(rt5640->regmap);

sound/soc/codecs/rt5645.h

@@ -105,6 +105,7 @@
 #define RT5645_TDM_CTRL_1			0x77
 #define RT5645_TDM_CTRL_2			0x78
 #define RT5645_TDM_CTRL_3			0x79
+#define RT5650_TDM_CTRL_4			0x7a
 
 /* Function - Analog */
 #define RT5645_GLB_CLK				0x80
@@ -942,10 +943,6 @@
 #define RT5645_I2S2_SDI_I2S2			(0x1 << 6)
 
 /* ADC/DAC Clock Control 1 (0x73) */
-#define RT5645_I2S_BCLK_MS1_MASK		(0x1 << 15)
-#define RT5645_I2S_BCLK_MS1_SFT			15
-#define RT5645_I2S_BCLK_MS1_32			(0x0 << 15)
-#define RT5645_I2S_BCLK_MS1_64			(0x1 << 15)
 #define RT5645_I2S_PD1_MASK			(0x7 << 12)
 #define RT5645_I2S_PD1_SFT			12
 #define RT5645_I2S_PD1_1			(0x0 << 12)
@@ -1067,13 +1064,14 @@
 #define RT5645_SCLK_SRC_SFT			14
 #define RT5645_SCLK_SRC_MCLK			(0x0 << 14)
 #define RT5645_SCLK_SRC_PLL1			(0x1 << 14)
-#define RT5645_SCLK_SRC_RCCLK			(0x2 << 14) /* 15MHz */
-#define RT5645_PLL1_SRC_MASK			(0x3 << 12)
-#define RT5645_PLL1_SRC_SFT			12
-#define RT5645_PLL1_SRC_MCLK			(0x0 << 12)
-#define RT5645_PLL1_SRC_BCLK1			(0x1 << 12)
-#define RT5645_PLL1_SRC_BCLK2			(0x2 << 12)
-#define RT5645_PLL1_SRC_BCLK3			(0x3 << 12)
+#define RT5645_SCLK_SRC_RCCLK			(0x2 << 14)
+#define RT5645_PLL1_SRC_MASK			(0x7 << 11)
+#define RT5645_PLL1_SRC_SFT			11
+#define RT5645_PLL1_SRC_MCLK			(0x0 << 11)
+#define RT5645_PLL1_SRC_BCLK1			(0x1 << 11)
+#define RT5645_PLL1_SRC_BCLK2			(0x2 << 11)
+#define RT5645_PLL1_SRC_BCLK3			(0x3 << 11)
+#define RT5645_PLL1_SRC_RCCLK			(0x4 << 11)
 #define RT5645_PLL1_PD_MASK			(0x1 << 3)
 #define RT5645_PLL1_PD_SFT			3
 #define RT5645_PLL1_PD_1			(0x0 << 3)
@@ -2147,6 +2145,7 @@ enum {
 };
 
 enum {
+	RT5645_DMIC1_DISABLE,
 	RT5645_DMIC_DATA_IN2P,
 	RT5645_DMIC_DATA_GPIO6,
 	RT5645_DMIC_DATA_GPIO10,
@@ -2154,6 +2153,7 @@ enum {
 };
 
 enum {
+	RT5645_DMIC2_DISABLE,
 	RT5645_DMIC_DATA_IN2N,
 	RT5645_DMIC_DATA_GPIO5,
 	RT5645_DMIC_DATA_GPIO11,
@@ -2184,6 +2184,7 @@ struct rt5645_priv {
 	struct i2c_client *i2c;
 	struct snd_soc_jack *hp_jack;
 	struct snd_soc_jack *mic_jack;
+	struct snd_soc_jack *btn_jack;
 	struct delayed_work jack_detect_work;
 
 	int codec_type;
@@ -2196,9 +2197,12 @@ struct rt5645_priv {
 	int pll_src;
 	int pll_in;
 	int pll_out;
+
+	int jack_type;
+	bool en_button_func;
 };
 
 int rt5645_set_jack_detect(struct snd_soc_codec *codec,
-	struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack);
-
+	struct snd_soc_jack *hp_jack, struct snd_soc_jack *mic_jack,
+	struct snd_soc_jack *btn_jack);
 #endif /* __RT5645_H__ */

sound/soc/codecs/rt5651.c

@@ -1571,7 +1571,7 @@ static int rt5651_set_bias_level(struct snd_soc_codec *codec,
 {
 	switch (level) {
 	case SND_SOC_BIAS_PREPARE:
-		if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) {
+		if (SND_SOC_BIAS_STANDBY == snd_soc_codec_get_bias_level(codec)) {
 			snd_soc_update_bits(codec, RT5651_PWR_ANLG1,
 				RT5651_PWR_VREF1 | RT5651_PWR_MB |
 				RT5651_PWR_BG | RT5651_PWR_VREF2,
@@ -1604,7 +1604,6 @@ static int rt5651_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
@@ -1625,7 +1624,7 @@ static int rt5651_probe(struct snd_soc_codec *codec)
 		RT5651_PWR_FV1 | RT5651_PWR_FV2,
 		RT5651_PWR_FV1 | RT5651_PWR_FV2);
 
-	rt5651_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	return 0;
 }

sound/soc/codecs/rt5670.c

@@ -416,12 +416,12 @@ static bool rt5670_readable_register(struct device *dev, unsigned int reg)
 static int rt5670_headset_detect(struct snd_soc_codec *codec, int jack_insert)
 {
 	int val;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
 
 	if (jack_insert) {
-		snd_soc_dapm_force_enable_pin(&codec->dapm,
-						       "Mic Det Power");
-		snd_soc_dapm_sync(&codec->dapm);
+		snd_soc_dapm_force_enable_pin(dapm, "Mic Det Power");
+		snd_soc_dapm_sync(dapm);
 		snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x0);
 		snd_soc_update_bits(codec, RT5670_CJ_CTRL2,
 			RT5670_CBJ_DET_MODE | RT5670_CBJ_MN_JD,
@@ -447,15 +447,15 @@ static int rt5670_headset_detect(struct snd_soc_codec *codec, int jack_insert)
 		} else {
 			snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4);
 			rt5670->jack_type = SND_JACK_HEADPHONE;
-			snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power");
-			snd_soc_dapm_sync(&codec->dapm);
+			snd_soc_dapm_disable_pin(dapm, "Mic Det Power");
+			snd_soc_dapm_sync(dapm);
 		}
 	} else {
 		snd_soc_update_bits(codec, RT5670_INT_IRQ_ST, 0x8, 0x0);
 		snd_soc_update_bits(codec, RT5670_GEN_CTRL3, 0x4, 0x4);
 		rt5670->jack_type = 0;
-		snd_soc_dapm_disable_pin(&codec->dapm, "Mic Det Power");
-		snd_soc_dapm_sync(&codec->dapm);
+		snd_soc_dapm_disable_pin(dapm, "Mic Det Power");
+		snd_soc_dapm_sync(dapm);
 	}
 
 	return rt5670->jack_type;
@@ -2603,7 +2603,7 @@ static int rt5670_set_bias_level(struct snd_soc_codec *codec,
 
 	switch (level) {
 	case SND_SOC_BIAS_PREPARE:
-		if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) {
+		if (SND_SOC_BIAS_STANDBY == snd_soc_codec_get_bias_level(codec)) {
 			snd_soc_update_bits(codec, RT5670_PWR_ANLG1,
 				RT5670_PWR_VREF1 | RT5670_PWR_MB |
 				RT5670_PWR_BG | RT5670_PWR_VREF2,
@@ -2647,30 +2647,30 @@ static int rt5670_set_bias_level(struct snd_soc_codec *codec,
 	default:
 		break;
 	}
-	codec->dapm.bias_level = level;
 
 	return 0;
 }
 
 static int rt5670_probe(struct snd_soc_codec *codec)
 {
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 	struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec);
 
 	switch (snd_soc_read(codec, RT5670_RESET) & RT5670_ID_MASK) {
 	case RT5670_ID_5670:
 	case RT5670_ID_5671:
-		snd_soc_dapm_new_controls(&codec->dapm,
+		snd_soc_dapm_new_controls(dapm,
 			rt5670_specific_dapm_widgets,
 			ARRAY_SIZE(rt5670_specific_dapm_widgets));
-		snd_soc_dapm_add_routes(&codec->dapm,
+		snd_soc_dapm_add_routes(dapm,
 			rt5670_specific_dapm_routes,
 			ARRAY_SIZE(rt5670_specific_dapm_routes));
 		break;
 	case RT5670_ID_5672:
-		snd_soc_dapm_new_controls(&codec->dapm,
+		snd_soc_dapm_new_controls(dapm,
 			rt5672_specific_dapm_widgets,
 			ARRAY_SIZE(rt5672_specific_dapm_widgets));
-		snd_soc_dapm_add_routes(&codec->dapm,
+		snd_soc_dapm_add_routes(dapm,
 			rt5672_specific_dapm_routes,
 			ARRAY_SIZE(rt5672_specific_dapm_routes));
 		break;

sound/soc/codecs/rt5677.h

@@ -1446,6 +1446,16 @@
 #define RT5677_DSP_OB_4_7_CLK_SEL_MASK		(0xf << 8)
 #define RT5677_DSP_OB_4_7_CLK_SEL_SFT		8
 
+/* ASRC Control 8 (0x8a) */
+#define RT5677_I2S1_CLK_SEL_MASK		(0xf << 12)
+#define RT5677_I2S1_CLK_SEL_SFT			12
+#define RT5677_I2S2_CLK_SEL_MASK		(0xf << 8)
+#define RT5677_I2S2_CLK_SEL_SFT			8
+#define RT5677_I2S3_CLK_SEL_MASK		(0xf << 4)
+#define RT5677_I2S3_CLK_SEL_SFT			4
+#define RT5677_I2S4_CLK_SEL_MASK		(0xf)
+#define RT5677_I2S4_CLK_SEL_SFT			0
+
 /* VAD Function Control 4 (0x9f) */
 #define RT5677_VAD_SRC_MASK			(0x7 << 8)
 #define RT5677_VAD_SRC_SFT			8
@@ -1744,6 +1754,10 @@ enum {
 	RT5677_AD_MONO_R_FILTER = (0x1 << 12),
 	RT5677_DSP_OB_0_3_FILTER = (0x1 << 13),
 	RT5677_DSP_OB_4_7_FILTER = (0x1 << 14),
+	RT5677_I2S1_SOURCE = (0x1 << 15),
+	RT5677_I2S2_SOURCE = (0x1 << 16),
+	RT5677_I2S3_SOURCE = (0x1 << 17),
+	RT5677_I2S4_SOURCE = (0x1 << 18),
 };
 
 struct rt5677_priv {
@@ -1762,6 +1776,7 @@ struct rt5677_priv {
 	int pll_in;
 	int pll_out;
 	int pow_ldo2; /* POW_LDO2 pin */
+	int reset_pin; /* RESET pin */
 	enum rt5677_type type;
 #ifdef CONFIG_GPIOLIB
 	struct gpio_chip gpio_chip;

sound/soc/codecs/sirf-audio-codec.c

@@ -395,7 +395,7 @@ struct snd_soc_dai_driver sirf_audio_codec_dai = {
 
 static int sirf_audio_codec_probe(struct snd_soc_codec *codec)
 {
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
 
 	pm_runtime_enable(codec->dev);
 

sound/soc/codecs/sn95031.c

@@ -194,7 +194,7 @@ static int sn95031_set_vaud_bias(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_PREPARE:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) {
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) {
 			pr_debug("vaud_bias powering up pll\n");
 			/* power up the pll */
 			snd_soc_write(codec, SN95031_AUDPLLCTRL, BIT(5));
@@ -205,17 +205,22 @@ static int sn95031_set_vaud_bias(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+		switch (snd_soc_codec_get_bias_level(codec)) {
+		case SND_SOC_BIAS_OFF:
 			pr_debug("vaud_bias power up rail\n");
 			/* power up the rail */
 			snd_soc_write(codec, SN95031_VAUD,
 					BIT(2)|BIT(1)|BIT(0));
 			msleep(1);
-		} else if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) {
+			break;
+		case SND_SOC_BIAS_PREPARE:
 			/* turn off pcm */
 			pr_debug("vaud_bias power dn pcm\n");
 			snd_soc_update_bits(codec, SN95031_PCM2C2, BIT(0), 0);
 			snd_soc_write(codec, SN95031_AUDPLLCTRL, 0);
+			break;
+		default:
+			break;
 		}
 		break;
 
@@ -226,7 +231,6 @@ static int sn95031_set_vaud_bias(struct snd_soc_codec *codec,
 		break;
 	}
 
-	codec->dapm.bias_level = level;
 	return 0;
 }
 

sound/soc/codecs/ssm2518.c

@@ -510,7 +510,7 @@ static int ssm2518_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+		if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF)
 			ret = ssm2518_set_power(ssm2518, true);
 		break;
 	case SND_SOC_BIAS_OFF:
@@ -518,12 +518,7 @@ static int ssm2518_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
-	if (ret)
-		return ret;
-
-	codec->dapm.bias_level = level;
-
-	return 0;
+	return ret;
 }
 
 static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,