Merge tag 'asoc-v4.7' into asoc-linus

ASoC: Updates for v4.7

The updates this time around are almost all driver code:

 - Further slow progress on the topology code.
 - Substantial updates and improvements for the da7219, es8328, fsl-ssi
   Intel and rcar drivers.

# gpg: Signature made Mon 16 May 2016 12:08:43 BST using RSA key ID 5D5487D0
# 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>"
# gpg: WARNING: This key is not certified with a trusted signature!
# gpg:          There is no indication that the signature belongs to the owner.
# Primary key fingerprint: 3F25 68AA C269 98F9 E813  A1C5 C3F4 36CA 30F5 D8EB
#      Subkey fingerprint: ADE6 68AA 6757 18B5 9FE2  9FEA 24D6 8B72 5D54 87D0

82 files changed, 3298 insertions, 485 deletions

diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt
new file mode 100644
index 000000000000..55b53e1fd72c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt
@@ -0,0 +1,51 @@
+Texas Instruments DaVinci McBSP module
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+This binding describes the "Multi-channel Buffered Serial Port" (McBSP)
+audio interface found in some TI DaVinci processors like the OMAP-L138 or AM180x.
+
+
+Required properties:
+~~~~~~~~~~~~~~~~~~~~
+- compatible :
+        "ti,da850-mcbsp" : for DA850, AM180x and OPAM-L138 platforms
+
+- reg : physical base address and length of the controller memory mapped
+        region(s).
+- reg-names : Should contain:
+        * "mpu" for the main registers (required).
+        * "dat" for the data FIFO (optional).
+
+- dmas: three element list of DMA controller phandles, DMA request line and
+	TC channel ordered triplets.
+- dma-names: identifier string for each DMA request line in the dmas property.
+	These strings correspond 1:1 with the ordered pairs in dmas. The dma
+	identifiers must be "rx" and "tx".
+
+Optional properties:
+~~~~~~~~~~~~~~~~~~~~
+- interrupts : Interrupt numbers for McBSP
+- interrupt-names : Known interrupt names are "rx" and "tx"
+
+- pinctrl-0: Should specify pin control group used for this controller.
+- pinctrl-names: Should contain only one value - "default", for more details
+        please refer to pinctrl-bindings.txt
+
+Example (AM1808):
+~~~~~~~~~~~~~~~~~
+
+mcbsp0: mcbsp@1d10000 {
+	compatible = "ti,da850-mcbsp";
+	pinctrl-names = "default";
+	pinctrl-0 = <&mcbsp0_pins>;
+
+	reg = 	<0x00110000 0x1000>,
+		<0x00310000 0x1000>;
+	reg-names = "mpu", "dat";
+	interrupts = <97 98>;
+	interrupts-names = "rx", "tx";
+	dmas = <&edma0 3 1
+		&edma0 2 1>;
+	dma-names = "tx", "rx";
+	status = "okay";
+};
diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt
index 044e5d76e2dd..740b467adf7d 100644
--- a/Documentation/devicetree/bindings/sound/fsl-sai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -7,8 +7,8 @@ codec/DSP interfaces.
 
 Required properties:
 
-  - compatible		: Compatible list, contains "fsl,vf610-sai" or
-			  "fsl,imx6sx-sai".
+  - compatible		: Compatible list, contains "fsl,vf610-sai",
+			  "fsl,imx6sx-sai" or "fsl,imx6ul-sai"
 
   - reg			: Offset and length of the register set for the device.
 
@@ -48,6 +48,11 @@ Required properties:
 			  receive data by following their own bit clocks and
 			  frame sync clocks separately.
 
+Optional properties (for mx6ul):
+
+  - fsl,sai-mclk-direction-output: This is a boolean property. If present,
+			 indicates that SAI will output the SAI MCLK clock.
+
 Note:
 - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the
   default synchronous mode (sync Rx with Tx) will be used, which means both
diff --git a/Documentation/devicetree/bindings/sound/pcm5102a.txt b/Documentation/devicetree/bindings/sound/pcm5102a.txt
new file mode 100644
index 000000000000..c63ab0b6ee19
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/pcm5102a.txt
@@ -0,0 +1,13 @@
+PCM5102a audio CODECs
+
+These devices does not use I2C or SPI.
+
+Required properties:
+
+  - compatible : set as "ti,pcm5102a"
+
+Examples:
+
+	pcm5102a: pcm5102a {
+		compatible = "ti,pcm5102a";
+	};
diff --git a/MAINTAINERS b/MAINTAINERS
index 9c567a431d8d..8e325edbf62c 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -4661,6 +4661,7 @@ FREESCALE SOC SOUND DRIVERS
 M:	Timur Tabi <timur@tabi.org>
 M:	Nicolin Chen <nicoleotsuka@gmail.com>
 M:	Xiubo Li <Xiubo.Lee@gmail.com>
+R:	Fabio Estevam <fabio.estevam@nxp.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 L:	linuxppc-dev@lists.ozlabs.org
 S:	Maintained
diff --git a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h
index 238c8db953eb..68353822afce 100644
--- a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h
+++ b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h
@@ -447,5 +447,11 @@
 #define IMX6UL_GPR1_ENET2_CLK_OUTPUT		(0x1 << 18)
 #define IMX6UL_GPR1_ENET_CLK_DIR		(0x3 << 17)
 #define IMX6UL_GPR1_ENET_CLK_OUTPUT		(0x3 << 17)
+#define IMX6UL_GPR1_SAI1_MCLK_DIR		(0x1 << 19)
+#define IMX6UL_GPR1_SAI2_MCLK_DIR		(0x1 << 20)
+#define IMX6UL_GPR1_SAI3_MCLK_DIR		(0x1 << 21)
+#define IMX6UL_GPR1_SAI_MCLK_MASK		(0x7 << 19)
+#define MCLK_DIR(x) (x == 1 ? IMX6UL_GPR1_SAI1_MCLK_DIR : x == 2 ? \
+		     IMX6UL_GPR1_SAI2_MCLK_DIR : IMX6UL_GPR1_SAI3_MCLK_DIR)
 
 #endif /* __LINUX_IMX6Q_IOMUXC_GPR_H */
diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index f86ef5ea9b01..67be2445941a 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -51,6 +51,16 @@ struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn,
 	void *filter_data);
 struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream);
 
+/*
+ * The DAI supports packed transfers, eg 2 16-bit samples in a 32-bit word.
+ * If this flag is set the dmaengine driver won't put any restriction on
+ * the supported sample formats and set the DMA transfer size to undefined.
+ * The DAI driver is responsible to disable any unsupported formats in it's
+ * configuration and catch corner cases that are not already handled in
+ * the ALSA core.
+ */
+#define SND_DMAENGINE_PCM_DAI_FLAG_PACK BIT(0)
+
 /**
  * struct snd_dmaengine_dai_dma_data - DAI DMA configuration data
  * @addr: Address of the DAI data source or destination register.
@@ -63,6 +73,7 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream)
  * requesting the DMA channel.
  * @chan_name: Custom channel name to use when requesting DMA channel.
  * @fifo_size: FIFO size of the DAI controller in bytes
+ * @flags: PCM_DAI flags, only SND_DMAENGINE_PCM_DAI_FLAG_PACK for now
  */
 struct snd_dmaengine_dai_dma_data {
 	dma_addr_t addr;
@@ -72,6 +83,7 @@ struct snd_dmaengine_dai_dma_data {
 	void *filter_data;
 	const char *chan_name;
 	unsigned int fifo_size;
+	unsigned int flags;
 };
 
 void snd_dmaengine_pcm_set_config_from_dai_data(
diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h
index 07fa59237feb..b9593b201599 100644
--- a/include/sound/hdaudio_ext.h
+++ b/include/sound/hdaudio_ext.h
@@ -14,6 +14,8 @@
  * @gtscap: gts capabilities pointer
  * @drsmcap: dma resume capabilities pointer
  * @hlink_list: link list of HDA links
+ * @lock: lock for link mgmt
+ * @cmd_dma_state: state of cmd DMAs: CORB and RIRB
  */
 struct hdac_ext_bus {
 	struct hdac_bus bus;
@@ -27,6 +29,9 @@ struct hdac_ext_bus {
 	void __iomem *drsmcap;
 
 	struct list_head hlink_list;
+
+	struct mutex lock;
+	bool cmd_dma_state;
 };
 
 int snd_hdac_ext_bus_init(struct hdac_ext_bus *sbus, struct device *dev,
@@ -142,6 +147,9 @@ struct hdac_ext_link {
 	void __iomem *ml_addr; /* link output stream reg pointer */
 	u32 lcaps;   /* link capablities */
 	u16 lsdiid;  /* link sdi identifier */
+
+	int ref_count;
+
 	struct list_head list;
 };
 
@@ -154,6 +162,11 @@ void snd_hdac_ext_link_set_stream_id(struct hdac_ext_link *link,
 void snd_hdac_ext_link_clear_stream_id(struct hdac_ext_link *link,
 				 int stream);
 
+int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus,
+				struct hdac_ext_link *link);
+int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus,
+				struct hdac_ext_link *link);
+
 /* update register macro */
 #define snd_hdac_updatel(addr, reg, mask, val)		\
 	writel(((readl(addr + reg) & ~(mask)) | (val)), \
diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h
new file mode 100644
index 000000000000..fc3a481ad91e
--- /dev/null
+++ b/include/sound/hdmi-codec.h
@@ -0,0 +1,100 @@
+/*
+ * hdmi-codec.h - HDMI Codec driver API
+ *
+ * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
+ *
+ * Author: Jyri Sarha <jsarha@ti.com>
+ *
+ * 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.
+ *
+ * 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 __HDMI_CODEC_H__
+#define __HDMI_CODEC_H__
+
+#include <linux/hdmi.h>
+#include <drm/drm_edid.h>
+#include <sound/asoundef.h>
+#include <uapi/sound/asound.h>
+
+/*
+ * Protocol between ASoC cpu-dai and HDMI-encoder
+ */
+struct hdmi_codec_daifmt {
+	enum {
+		HDMI_I2S,
+		HDMI_RIGHT_J,
+		HDMI_LEFT_J,
+		HDMI_DSP_A,
+		HDMI_DSP_B,
+		HDMI_AC97,
+		HDMI_SPDIF,
+	} fmt;
+	int bit_clk_inv:1;
+	int frame_clk_inv:1;
+	int bit_clk_master:1;
+	int frame_clk_master:1;
+};
+
+/*
+ * HDMI audio parameters
+ */
+struct hdmi_codec_params {
+	struct hdmi_audio_infoframe cea;
+	struct snd_aes_iec958 iec;
+	int sample_rate;
+	int sample_width;
+	int channels;
+};
+
+struct hdmi_codec_ops {
+	/*
+	 * Called when ASoC starts an audio stream setup.
+	 * Optional
+	 */
+	int (*audio_startup)(struct device *dev);
+
+	/*
+	 * Configures HDMI-encoder for audio stream.
+	 * Mandatory
+	 */
+	int (*hw_params)(struct device *dev,
+			 struct hdmi_codec_daifmt *fmt,
+			 struct hdmi_codec_params *hparms);
+
+	/*
+	 * Shuts down the audio stream.
+	 * Mandatory
+	 */
+	void (*audio_shutdown)(struct device *dev);
+
+	/*
+	 * Mute/unmute HDMI audio stream.
+	 * Optional
+	 */
+	int (*digital_mute)(struct device *dev, bool enable);
+
+	/*
+	 * Provides EDID-Like-Data from connected HDMI device.
+	 * Optional
+	 */
+	int (*get_eld)(struct device *dev, uint8_t *buf, size_t len);
+};
+
+/* HDMI codec initalization data */
+struct hdmi_codec_pdata {
+	const struct hdmi_codec_ops *ops;
+	uint i2s:1;
+	uint spdif:1;
+	int max_i2s_channels;
+};
+
+#define HDMI_CODEC_DRV_NAME "hdmi-audio-codec"
+
+#endif /* __HDMI_CODEC_H__ */
diff --git a/include/sound/pcm_iec958.h b/include/sound/pcm_iec958.h
index 0eed397aca8e..36f023acb201 100644
--- a/include/sound/pcm_iec958.h
+++ b/include/sound/pcm_iec958.h
@@ -6,4 +6,6 @@
 int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs,
 	size_t len);
 
+int snd_pcm_create_iec958_consumer_hw_params(struct snd_pcm_hw_params *params,
+					     u8 *cs, size_t len);
 #endif
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 97069466c38d..3101d53468aa 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -100,6 +100,7 @@ struct device;
 {       .id = snd_soc_dapm_mixer_named_ctl, .name = wname, \
 	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
 	.kcontrol_news = wcontrols, .num_kcontrols = wncontrols}
+/* DEPRECATED: use SND_SOC_DAPM_SUPPLY */
 #define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \
 {	.id = snd_soc_dapm_micbias, .name = wname, \
 	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
@@ -473,7 +474,7 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_out_drv,			/* output driver */
 	snd_soc_dapm_adc,			/* analog to digital converter */
 	snd_soc_dapm_dac,			/* digital to analog converter */
-	snd_soc_dapm_micbias,		/* microphone bias (power) */
+	snd_soc_dapm_micbias,		/* microphone bias (power) - DEPRECATED: use snd_soc_dapm_supply */
 	snd_soc_dapm_mic,			/* microphone */
 	snd_soc_dapm_hp,			/* headphones */
 	snd_soc_dapm_spk,			/* speaker */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 02b4a215fd75..fd7b58a58d6f 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1002,7 +1002,7 @@ struct snd_soc_dai_link {
 	 */
 	const char *platform_name;
 	struct device_node *platform_of_node;
-	int be_id;	/* optional ID for machine driver BE identification */
+	int id;	/* optional ID for machine driver link identification */
 
 	const struct snd_soc_pcm_stream *params;
 	unsigned int num_params;
@@ -1683,6 +1683,9 @@ void snd_soc_remove_dai_link(struct snd_soc_card *card,
 int snd_soc_register_dai(struct snd_soc_component *component,
 	struct snd_soc_dai_driver *dai_drv);
 
+struct snd_soc_dai *snd_soc_find_dai(