Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Mostly many small changes spread as seen in diffstat in sound/*
  directory by this update.  A significant change in the subsystem level
  is the introduction of snd_soc_component, which will help more generic
  handling of SoC and off-SoC components.

  Also, snd_BUG_ON() macro is enabled unconditionally now due to its
  misuses, so people might hit kernel warnings (it's a good thing for
  us).

   - compress-offload: support for capture by Charles Keepax
   - HD-audio: codec delay support by Dylan Reid
   - HD-audio: improvements/fixes in generic parser: better headphone
     mic and headset mic support, jack_modes hint consolidation, proper
     beep attach/detachment, generalized power filter controls by David
     Henningsson, et al
   - HD-audio: Improved management of HDMI codec pins/converters
   - HD-audio: Better pin/DAC assignment for VIA codecs
   - HD-audio: Haswell HDMI workarounds
   - HD-audio: ALC268 codec support, a few new quirks for Chromebooks
   - USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
     fix by Clemens Ladisch
   - USB: support for DSD formats by Daniel Mack
   - USB: A few UAC2 device endian/cock fixes by Eldad Zack
   - USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
     Yamaha THRxx devices
   - HDSPM: updates for TCO controls by Adrian Knoth
   - ASoC: Add a snd_soc_component object type for generic handling of
     SoC and off-SoC components by Kuninori Morimoto,
   - dmaengine: a large set of cleanups and conversions by Lars-Peter
     Clausen
   - ASoC DAPM: performance optimizations from Ryo Tsutsui
   - ASoC DAPM: support for mixer control sharing by Stephen Warren
   - ASoC: multiplatform ARM cleanups from Arnd Bergmann
   - ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"

* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
  ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
  ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
  ALSA: pcm_format_to_bits strong-typed conversion
  ALSA: compress: fix the states to check for allowing read
  ALSA: hda - Move Thinkpad X220 to use auto parser
  ALSA: USB: adjust for changed 3.8 USB API
  ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
  sound: oss/dmabuf: use dma_map_single
  ALSA: ali5451: use mdelay instead of large udelay constants
  ALSA: hda - Add the support for ALC286 codec
  ALSA: usb-audio: USB quirk for Yamaha THR10C
  ALSA: usb-audio: USB quirk for Yamaha THR5A
  ALSA: usb-audio: USB quirk for Yamaha THR10
  ALSA: usb-audio: Fix autopm error during probing
  ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
  ALSA: sound kconfig typo
  ALSA: emu10k1: Fix dock firmware loading
  ASoC: ux500: forward declare msp_i2s_platform_data
  ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
  ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
  ...

15 files changed, 220 insertions, 75 deletions

diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h
index f43aa7c8d040..715b6ba3d52a 100644
--- a/include/linux/mfd/arizona/registers.h
+++ b/include/linux/mfd/arizona/registers.h
@@ -85,12 +85,14 @@
 #define ARIZONA_FLL1_CONTROL_6                   0x176
 #define ARIZONA_FLL1_LOOP_FILTER_TEST_1          0x177
 #define ARIZONA_FLL1_NCO_TEST_0                  0x178
+#define ARIZONA_FLL1_CONTROL_7                   0x179
 #define ARIZONA_FLL1_SYNCHRONISER_1              0x181
 #define ARIZONA_FLL1_SYNCHRONISER_2              0x182
 #define ARIZONA_FLL1_SYNCHRONISER_3              0x183
 #define ARIZONA_FLL1_SYNCHRONISER_4              0x184
 #define ARIZONA_FLL1_SYNCHRONISER_5              0x185
 #define ARIZONA_FLL1_SYNCHRONISER_6              0x186
+#define ARIZONA_FLL1_SYNCHRONISER_7              0x187
 #define ARIZONA_FLL1_SPREAD_SPECTRUM             0x189
 #define ARIZONA_FLL1_GPIO_CLOCK                  0x18A
 #define ARIZONA_FLL2_CONTROL_1                   0x191
@@ -101,12 +103,14 @@
 #define ARIZONA_FLL2_CONTROL_6                   0x196
 #define ARIZONA_FLL2_LOOP_FILTER_TEST_1          0x197
 #define ARIZONA_FLL2_NCO_TEST_0                  0x198
+#define ARIZONA_FLL2_CONTROL_7                   0x199
 #define ARIZONA_FLL2_SYNCHRONISER_1              0x1A1
 #define ARIZONA_FLL2_SYNCHRONISER_2              0x1A2
 #define ARIZONA_FLL2_SYNCHRONISER_3              0x1A3
 #define ARIZONA_FLL2_SYNCHRONISER_4              0x1A4
 #define ARIZONA_FLL2_SYNCHRONISER_5              0x1A5
 #define ARIZONA_FLL2_SYNCHRONISER_6              0x1A6
+#define ARIZONA_FLL2_SYNCHRONISER_7              0x1A7
 #define ARIZONA_FLL2_SPREAD_SPECTRUM             0x1A9
 #define ARIZONA_FLL2_GPIO_CLOCK                  0x1AA
 #define ARIZONA_MIC_CHARGE_PUMP_1                0x200
@@ -217,6 +221,8 @@
 #define ARIZONA_PDM_SPK1_CTRL_2                  0x491
 #define ARIZONA_PDM_SPK2_CTRL_1                  0x492
 #define ARIZONA_PDM_SPK2_CTRL_2                  0x493
+#define ARIZONA_SPK_CTRL_2                       0x4B5
+#define ARIZONA_SPK_CTRL_3                       0x4B6
 #define ARIZONA_DAC_COMP_1                       0x4DC
 #define ARIZONA_DAC_COMP_2                       0x4DD
 #define ARIZONA_DAC_COMP_3                       0x4DE
@@ -1682,6 +1688,13 @@
 #define ARIZONA_FLL1_FRC_INTEG_VAL_WIDTH             12  /* FLL1_FRC_INTEG_VAL - [11:0] */
 
 /*
+ * R377 (0x179) - FLL1 Control 7
+ */
+#define ARIZONA_FLL1_GAIN_MASK                   0x003c  /* FLL1_GAIN */
+#define ARIZONA_FLL1_GAIN_SHIFT                       2  /* FLL1_GAIN */
+#define ARIZONA_FLL1_GAIN_WIDTH                       4  /* FLL1_GAIN */
+
+/*
  * R385 (0x181) - FLL1 Synchroniser 1
  */
 #define ARIZONA_FLL1_SYNC_ENA                    0x0001  /* FLL1_SYNC_ENA */
@@ -1728,6 +1741,17 @@
 #define ARIZONA_FLL1_CLK_SYNC_SRC_WIDTH               4  /* FLL1_CLK_SYNC_SRC - [3:0] */
 
 /*
+ * R391 (0x187) - FLL1 Synchroniser 7
+ */
+#define ARIZONA_FLL1_SYNC_GAIN_MASK              0x003c  /* FLL1_SYNC_GAIN */
+#define ARIZONA_FLL1_SYNC_GAIN_SHIFT                  2  /* FLL1_SYNC_GAIN */
+#define ARIZONA_FLL1_SYNC_GAIN_WIDTH                  4  /* FLL1_SYNC_GAIN */
+#define ARIZONA_FLL1_SYNC_BW                     0x0001  /* FLL1_SYNC_BW */
+#define ARIZONA_FLL1_SYNC_BW_MASK                0x0001  /* FLL1_SYNC_BW */
+#define ARIZONA_FLL1_SYNC_BW_SHIFT                    0  /* FLL1_SYNC_BW */
+#define ARIZONA_FLL1_SYNC_BW_WIDTH                    1  /* FLL1_SYNC_BW */
+
+/*
  * R393 (0x189) - FLL1 Spread Spectrum
  */
 #define ARIZONA_FLL1_SS_AMPL_MASK                0x0030  /* FLL1_SS_AMPL - [5:4] */
@@ -1820,6 +1844,13 @@
 #define ARIZONA_FLL2_FRC_INTEG_VAL_WIDTH             12  /* FLL2_FRC_INTEG_VAL - [11:0] */
 
 /*
+ * R409 (0x199) - FLL2 Control 7
+ */
+#define ARIZONA_FLL2_GAIN_MASK                   0x003c  /* FLL2_GAIN */
+#define ARIZONA_FLL2_GAIN_SHIFT                       2  /* FLL2_GAIN */
+#define ARIZONA_FLL2_GAIN_WIDTH                       4  /* FLL2_GAIN */
+
+/*
  * R417 (0x1A1) - FLL2 Synchroniser 1
  */
 #define ARIZONA_FLL2_SYNC_ENA                    0x0001  /* FLL2_SYNC_ENA */
@@ -1866,6 +1897,17 @@
 #define ARIZONA_FLL2_CLK_SYNC_SRC_WIDTH               4  /* FLL2_CLK_SYNC_SRC - [3:0] */
 
 /*
+ * R423 (0x1A7) - FLL2 Synchroniser 7
+ */
+#define ARIZONA_FLL2_SYNC_GAIN_MASK              0x003c  /* FLL2_SYNC_GAIN */
+#define ARIZONA_FLL2_SYNC_GAIN_SHIFT                  2  /* FLL2_SYNC_GAIN */
+#define ARIZONA_FLL2_SYNC_GAIN_WIDTH                  4  /* FLL2_SYNC_GAIN */
+#define ARIZONA_FLL2_SYNC_BW_MASK                0x0001  /* FLL2_SYNC_BW */
+#define ARIZONA_FLL2_SYNC_BW_MASK                0x0001  /* FLL2_SYNC_BW */
+#define ARIZONA_FLL2_SYNC_BW_SHIFT                    0  /* FLL2_SYNC_BW */
+#define ARIZONA_FLL2_SYNC_BW_WIDTH                    1  /* FLL2_SYNC_BW */
+
+/*
  * R425 (0x1A9) - FLL2 Spread Spectrum
  */
 #define ARIZONA_FLL2_SS_AMPL_MASK                0x0030  /* FLL2_SS_AMPL - [5:4] */
diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h
index 8e21a094836d..68e776594889 100644
--- a/include/linux/mfd/wm8994/pdata.h
+++ b/include/linux/mfd/wm8994/pdata.h
@@ -17,6 +17,7 @@
 
 #define WM8994_NUM_LDO   2
 #define WM8994_NUM_GPIO 11
+#define WM8994_NUM_AIF   3
 
 struct wm8994_ldo_pdata {
 	/** GPIOs to enable regulator, 0 or less if not available */
@@ -215,6 +216,13 @@ struct wm8994_pdata {
 	 * system.
 	 */
 	bool spkmode_pu;
+
+	/**
+	 * Maximum number of channels clocks will be generated for,
+	 * useful for systems where and I2S bus with multiple data
+	 * lines is mastered.
+	 */
+	int max_channels_clocked[WM8994_NUM_AIF];
 };
 
 #endif
diff --git a/include/linux/usb/audio-v2.h b/include/linux/usb/audio-v2.h
index ed13053153f4..c5f2158ab00e 100644
--- a/include/linux/usb/audio-v2.h
+++ b/include/linux/usb/audio-v2.h
@@ -170,6 +170,8 @@ struct uac2_as_header_descriptor {
 	__u8 iChannelNames;
 } __attribute__((packed));
 
+#define UAC2_FORMAT_TYPE_I_RAW_DATA	(1 << 31)
+
 /* 4.10.1.2 Class-Specific AS Isochronous Audio Data Endpoint Descriptor */
 
 struct uac2_iso_endpoint_descriptor {
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
index ff6c74153fa1..9031a26249b5 100644
--- a/include/sound/compress_driver.h
+++ b/include/sound/compress_driver.h
@@ -56,8 +56,6 @@ struct snd_compr_runtime {
 	u64 buffer_size;
 	u32 fragment_size;
 	u32 fragments;
-	u64 hw_pointer;
-	u64 app_pointer;
 	u64 total_bytes_available;
 	u64 total_bytes_transferred;
 	wait_queue_head_t sleep;
@@ -121,7 +119,7 @@ struct snd_compr_ops {
 	int (*trigger)(struct snd_compr_stream *stream, int cmd);
 	int (*pointer)(struct snd_compr_stream *stream,
 			struct snd_compr_tstamp *tstamp);
-	int (*copy)(struct snd_compr_stream *stream, const char __user *buf,
+	int (*copy)(struct snd_compr_stream *stream, char __user *buf,
 		       size_t count);
 	int (*mmap)(struct snd_compr_stream *stream,
 			struct vm_area_struct *vma);
diff --git a/include/sound/control.h b/include/sound/control.h
index 8332e865c759..34bc93d80d55 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -189,7 +189,6 @@ int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
  *
  * Add a virtual slave control to the given master element created via
  * snd_ctl_create_virtual_master() beforehand.
- * Returns zero if successful or a negative error code.
  *
  * All slaves must be the same type (returning the same information
  * via info callback).  The function doesn't check it, so it's your
@@ -199,6 +198,8 @@ int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
  * at most two channels,
  * logarithmic volume control (dB level) thus no linear volume,
  * master can only attenuate the volume without gain
+ *
+ * Return: Zero if successful or a negative error code.
  */
 static inline int
 snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
@@ -219,6 +220,8 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
  * When the control peeks the hardware values directly and the value
  * can be changed by other means than the put callback of the element,
  * this function should be used to keep the value always up-to-date.
+ *
+ * Return: Zero if successful or a negative error code.
  */
 static inline int
 snd_ctl_add_slave_uncached(struct snd_kcontrol *master,
diff --git a/include/sound/core.h b/include/sound/core.h
index 7cede2d6aa86..5bfe5136441c 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -229,7 +229,7 @@ int snd_register_device_for_dev(int type, struct snd_card *card,
  * This function uses the card's device pointer to link to the
  * correct &struct device.
  *
- * Returns zero if successful, or a negative error code on failure.
+ * Return: Zero if successful, or a negative error code on failure.
  */
 static inline int snd_register_device(int type, struct snd_card *card, int dev,
 				      const struct file_operations *f_ops,
@@ -379,18 +379,10 @@ void __snd_printk(unsigned int level, const char *file, int line,
  * snd_BUG_ON - debugging check macro
  * @cond: condition to evaluate
  *
- * When CONFIG_SND_DEBUG is set, this macro evaluates the given condition,
- * and call WARN() and returns the value if it's non-zero.
- * 
- * When CONFIG_SND_DEBUG is not set, this just returns zero, and the given
- * condition is ignored.
- *
- * NOTE: the argument won't be evaluated at all when CONFIG_SND_DEBUG=n.
- * Thus, don't put any statement that influences on the code behavior,
- * such as pre/post increment, to the argument of this macro.
- * If you want to evaluate and give a warning, use standard WARN_ON().
+ * Has the same behavior as WARN_ON when CONFIG_SND_DEBUG is set,
+ * otherwise just evaluates the conditional and returns the value.
  */
-#define snd_BUG_ON(cond)	WARN((cond), "BUG? (%s)\n", __stringify(cond))
+#define snd_BUG_ON(cond)	WARN_ON((cond))
 
 #else /* !CONFIG_SND_DEBUG */
 
@@ -400,11 +392,11 @@ __printf(2, 3)
 static inline void _snd_printd(int level, const char *format, ...) {}
 
 #define snd_BUG()			do { } while (0)
-static inline int __snd_bug_on(int cond)
-{
-	return 0;
-}
-#define snd_BUG_ON(cond)	__snd_bug_on(0 && (cond))  /* always false */
+
+#define snd_BUG_ON(condition) ({ \
+	int __ret_warn_on = !!(condition); \
+	unlikely(__ret_warn_on); \
+})
 
 #endif /* CONFIG_SND_DEBUG */
 
diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index b877334bbb0f..f11c35cd5532 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -16,6 +16,7 @@
 #define __SOUND_DMAENGINE_PCM_H__
 
 #include <sound/pcm.h>
+#include <sound/soc.h>
 #include <linux/dmaengine.h>
 
 /**
@@ -32,9 +33,6 @@ snd_pcm_substream_to_dma_direction(const struct snd_pcm_substream *substream)
 		return DMA_DEV_TO_MEM;
 }
 
-void snd_dmaengine_pcm_set_data(struct snd_pcm_substream *substream, void *data);
-void *snd_dmaengine_pcm_get_data(struct snd_pcm_substream *substream);
-
 int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream,
 	const struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config);
 int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd);
@@ -42,9 +40,100 @@ snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream)
 snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream);
 
 int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream,
-	dma_filter_fn filter_fn, void *filter_data);
+	struct dma_chan *chan);
 int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream);
 
+int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream,
+	dma_filter_fn filter_fn, void *filter_data);
+int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream);
+
+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);
 
+/**
+ * struct snd_dmaengine_dai_dma_data - DAI DMA configuration data
+ * @addr: Address of the DAI data source or destination register.
+ * @addr_width: Width of the DAI data source or destination register.
+ * @maxburst: Maximum number of words(note: words, as in units of the
+ * src_addr_width member, not bytes) that can be send to or received from the
+ * DAI in one burst.
+ * @slave_id: Slave requester id for the DMA channel.
+ * @filter_data: Custom DMA channel filter data, this will usually be used when
+ * requesting the DMA channel.
+ */
+struct snd_dmaengine_dai_dma_data {
+	dma_addr_t addr;
+	enum dma_slave_buswidth addr_width;
+	u32 maxburst;
+	unsigned int slave_id;
+	void *filter_data;
+};
+
+void snd_dmaengine_pcm_set_config_from_dai_data(
+	const struct snd_pcm_substream *substream,
+	const struct snd_dmaengine_dai_dma_data *dma_data,
+	struct dma_slave_config *config);
+
+
+/*
+ * Try to request the DMA channel using compat_request_channel or
+ * compat_filter_fn if it couldn't be requested through devicetree.
+ */
+#define SND_DMAENGINE_PCM_FLAG_COMPAT BIT(0)
+/*
+ * Don't try to request the DMA channels through devicetree. This flag only
+ * 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.
+ */
+#define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3)
+
+/**
+ * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM
+ * @prepare_slave_config: Callback used to fill in the DMA slave_config for a
+ *   PCM substream. Will be called from the PCM drivers hwparams callback.
+ * @compat_request_channel: Callback to request a DMA channel for platforms
+ *   which do not use devicetree.
+ * @compat_filter_fn: Will be used as the filter function when requesting a
+ *  channel for platforms which do not use devicetree. The filter parameter
+ *  will be the DAI's DMA data.
+ * @pcm_hardware: snd_pcm_hardware struct to be used for the PCM.
+ * @prealloc_buffer_size: Size of the preallocated audio buffer.
+ *
+ * Note: If both compat_request_channel and compat_filter_fn are set
+ * compat_request_channel will be used to request the channel and
+ * compat_filter_fn will be ignored. Otherwise the channel will be requested
+ * using dma_request_channel with compat_filter_fn as the filter function.
+ */
+struct snd_dmaengine_pcm_config {
+	int (*prepare_slave_config)(struct snd_pcm_substream *substream,
+			struct snd_pcm_hw_params *params,
+			struct dma_slave_config *slave_config);
+	struct dma_chan *(*compat_request_channel)(
+			struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_substream *substream);
+	dma_filter_fn compat_filter_fn;
+
+	const struct snd_pcm_hardware *pcm_hardware;
+	unsigned int prealloc_buffer_size;
+};
+
+int snd_dmaengine_pcm_register(struct device *dev,
+	const struct snd_dmaengine_pcm_config *config,
+	unsigned int flags);
+void snd_dmaengine_pcm_unregister(struct device *dev);
+
+int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params,
+	struct dma_slave_config *slave_config);
+
 #endif
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index f841ba4bacb8..dfb42ca6d043 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -1787,6 +1787,7 @@ struct snd_emu10k1 {
 	unsigned int next_free_voice;
 
 	const struct firmware *firmware;
+	const struct firmware *dock_fw;
 
 #ifdef CONFIG_PM_SLEEP
 	unsigned int *saved_ptr;
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 5ec42dbd2308..b48792fe386b 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -181,6 +181,8 @@ struct snd_pcm_ops {
 #define SNDRV_PCM_FMTBIT_G723_24_1B	_SNDRV_PCM_FMTBIT(G723_24_1B)
 #define SNDRV_PCM_FMTBIT_G723_40	_SNDRV_PCM_FMTBIT(G723_40)
 #define SNDRV_PCM_FMTBIT_G723_40_1B	_SNDRV_PCM_FMTBIT(G723_40_1B)
+#define SNDRV_PCM_FMTBIT_DSD_U8		_SNDRV_PCM_FMTBIT(DSD_U8)
+#define SNDRV_PCM_FMTBIT_DSD_U16_LE	_SNDRV_PCM_FMTBIT(DSD_U16_LE)
 
 #ifdef SNDRV_LITTLE_ENDIAN
 #define SNDRV_PCM_FMTBIT_S16		SNDRV_PCM_FMTBIT_S16_LE
@@ -659,7 +661,7 @@ static inline snd_pcm_sframes_t snd_pcm_capture_hw_avail(struct snd_pcm_runtime
  *
  * Checks whether enough free space is available on the playback buffer.
  *
- * Returns non-zero if available, or zero if not.
+ * Return: Non-zero if available, or zero if not.
  */
 static inline int snd_pcm_playback_ready(struct snd_pcm_substream *substream)
 {
@@ -673,7 +675,7 @@ static inline int snd_pcm_playback_ready(struct snd_pcm_substream *substream)
  *
  * Checks whether enough capture data is available on the capture buffer.
  *
- * Returns non-zero if available, or zero if not.
+ * Return: Non-zero if available, or zero if not.
  */
 static inline int snd_pcm_capture_ready(struct snd_pcm_substream *substream)
 {
@@ -685,10 +687,10 @@ static inline int snd_pcm_capture_ready(struct snd_pcm_substream *substream)
  * snd_pcm_playback_data - check whether any data exists on the playback buffer
  * @substream: the pcm substream instance
  *
- * Checks whether any data exists on the playback buffer. If stop_threshold
- * is bigger or equal to boundary, then this function returns always non-zero.
+ * Checks whether any data exists on the playback buffer.
  *
- * Returns non-zero if exists, or zero if not.
+ * Return: Non-zero if any data exists, or zero if not. If stop_threshold
+ * is bigger or equal to boundary, then this function returns always non-zero.
  */
 static inline int snd_pcm_playback_data(struct snd_pcm_substream *substream)
 {
@@ -705,7 +707,7 @@ static inline int snd_pcm_playback_data(struct snd_pcm_substream *substream)
  *
  * Checks whether the playback buffer is empty.
  *
- * Returns non-zero if empty, or zero if not.
+ * Return: Non-zero if empty, or zero if not.
  */
 static inline int snd_pcm_playback_empty(struct snd_pcm_substream *substream)
 {
@@ -719,7 +721,7 @@ static inline int snd_pcm_playback_empty(struct snd_pcm_substream *substream)
  *
  * Checks whether the capture buffer is empty.
  *
- * Returns non-zero if empty, or zero if not.
+ * Return: Non-zero if empty, or zero if not.
  */
 static inline int snd_pcm_capture_empty(struct snd_pcm_substream *substream)
 {
@@ -852,7 +854,7 @@ int snd_pcm_format_big_endian(snd_pcm_format_t format);
  * snd_pcm_format_cpu_endian - Check the PCM format is CPU-endian
  * @format: the format to check
  *
- * Returns 1 if the given PCM format is CPU-endian, 0 if
+ * Return: 1 if the given PCM format is CPU-endian, 0 if
  * opposite, or a negative error code if endian not specified.
  */
 int snd_pcm_format_cpu_endian(snd_pcm_format_t format);
@@ -963,7 +965,7 @@ struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream,
  * contiguous in kernel virtual space, but not in physical memory.  Use this
  * if the buffer is accessed by kernel code but not by device DMA.
  *
- * Returns 1 if the buffer was changed, 0 if not changed, or a negative error
+ * Return: 1 if the buffer was changed, 0 if not changed, or a negative error
  * code.
  */
 static int snd_pcm_lib_alloc_vmalloc_buffer
@@ -975,6 +977,9 @@ static int snd_pcm_lib_alloc_vmalloc_buffer
  *
  * This function works like snd_pcm_lib_alloc_vmalloc_buffer(), but uses
  * vmalloc_32(), i.e., the pages are allocated from 32-bit-addressable memory.
+ *
+ * Return: 1 if the buffer was changed, 0 if not changed, or a negative error
+ * code.
  */
 static int snd_pcm_lib_alloc_vmalloc_32_buffer
 			(struct snd_pcm_substream *substream, size_t size);
@@ -1070,6 +1075,8 @@ const char *snd_pcm_format_name(snd_pcm_format_t format);
 /**
  * snd_pcm_stream_str - Get a string naming the direction of a stream
  * @substream: the pcm substream instance
+ *
+ * Return: A string naming the direction of the stream.
  */
 static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream)
 {
@@ -1126,4 +1133,10 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream,
 			   unsigned long private_value,
 			   struct snd_pcm_chmap **info_ret);
 
+/* Strong-typed conversion of pcm_format to bitwise */
+static inline u64 pcm_format_to_bits(snd_pcm_format_t pcm_format)
+{
+	return 1ULL << (__force int) pcm_format;
+}
+
 #endif /* __SOUND_PCM_H */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 3d84808952b9..ae9a227d35d3 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -95,14 +95,6 @@ struct snd_soc_dai_driver;
 struct snd_soc_dai;
 struct snd_ac97_bus_ops;
 
-/* Digital Audio Interface registration */
-int snd_soc_register_dai(struct device *dev,
-		struct snd_soc_dai_driver *dai_drv);
-void snd_soc_unregister_dai(struct device *dev);
-int snd_soc_register_dais(struct device *dev,
-		struct snd_soc_dai_driver *dai_drv, size_t count);
-void snd_soc_unregister_dais(struct device *dev, size_t count);
-
 /* Digital Audio Interface clocking API.*/
 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	unsigned int freq, int dir);
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 44a30b108683..d4609029f014 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -566,7 +566,6 @@ struct snd_soc_dapm_update {
 
 /* DAPM context */
 struct snd_soc_dapm_context {
-	int n_widgets; /* number of widgets in this context */
 	enum snd_soc_bias_level bias_level;
 	enum snd_soc_bias_level suspend_bias_level;
 	struct delayed_work delayed_work;
diff --git a/include/sound/soc.h b/include/sound/soc.h
index a6a059ca3874..85c15226103b 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -324,6 +324,8 @@ struct snd_soc_dai_link;
 struct snd_soc_platform_driver;
 struct snd_soc_codec;
 struct snd_soc_codec_driver;
+struct snd_soc_component;
+struct snd_soc_component_driver;
 struct soc_enum;
 struct snd_soc_jack;
 struct snd_soc_jack_zone;
@@ -371,12 +373,20 @@ int snd_soc_suspend(struct device *dev);
 int snd_soc_resume(struct device *dev);
 int snd_soc_poweroff(struct device *dev);
 int snd_soc_register_platform(struct device *dev,
-		struct snd_soc_platform_driver *platform_drv);
+		const struct snd_soc_platform_driver *platform_drv);
 void snd_soc_unregister_platform(struct device *dev);
+int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform,
+		const struct snd_soc_platform_driver *platform_drv);
+void snd_soc_remove_platform(struct snd_soc_platform *platform);
+struct snd_soc_platform *snd_soc_lookup_platform(struct device *dev);
 int snd_soc_register_codec(struct device *dev,
 		const struct snd_soc_codec_driver *codec_drv,
 		struct snd_soc_dai_driver *dai_drv, int num_dai);
 void snd_soc_unregister_codec(struct device *dev);
+int snd_soc_register_component(struct device *dev,
+			 const struct snd_soc_component_driver *cmpnt_drv,
+			 struct snd_soc_dai_driver *dai_drv, int num_dai);
+void snd_soc_unregister_component(struct device *dev);
 int snd_soc_codec_volatile_register(struct snd_soc_codec *codec,
 				    unsigned int reg);
 int snd_soc_codec_readable_register(struct snd_soc_codec *codec,
@@ -801,10 +811,10 @@ struct snd_soc_platform_driver {
 		struct snd_soc_dai *);
 
 	/* platform stream pcm ops */
-	struct snd_pcm_ops *ops;
+	const struct snd_pcm_ops *ops;
 
 	/* platform stream compress ops */
-	struct snd_compr_ops *compr_ops;
+	const struct snd_compr_ops *compr_ops;
 
 	/* platform stream completion event */
 	int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
@@ -823,7 +833,7 @@ struct snd_soc_platform {
 	const char *name;
 	int id;
 	struct device *dev;
-	struct snd_soc_platform_driver *driver;
+	const struct snd_soc_platform_driver *driver;
 	struct mutex mutex;
 
 	unsigned int suspended:1; /* platform is suspended */
@@ -841,6 +851,20 @@ struct snd_soc_platform {
 #endif
 };
 
+struct snd_soc_component_driver {
+	const char *name;
+};
+
+struct snd_soc_component {
+	const char *name;
+	int id;
+	int num_dai;
+	struct device *dev;
+	struct list_head list;
+
+	const struct snd_soc_component_driver *driver;
+};
+
 struct snd_soc_dai_link {
 	/* config - must be set by machine driver */
 	const char *name;			/* Codec name */
@@ -1086,7 +1110,6 @@ struct soc_enum {
 	unsigned int mask;
 	const char * const *texts;
 	const unsigned int *values;
-	void *dapm;
 };
 
 /* codec IO */
diff --git a/include/sound/tas5086.h b/include/sound/tas5086.h
new file mode 100644
index 000000000000..aac481b7db8f
--- /dev/null
+++ b/include/sound/tas5086.h
@@ -0,0 +1,7 @@
+#ifndef _SND_SOC_CODEC_TAS5086_H_
+#define _SND_SOC_CODEC_TAS5086_H_
+
+#define TAS5086_CLK_IDX_MCLK	0
+#define TAS5086_CLK_IDX_SCLK	1
+
+#endif /* _SND_SOC_CODEC_TAS5086_H_ */
diff --git a/include/sound/tegra_wm8903.h b/include/sound/tegra_wm8903.h
deleted file mode 100644
index 57b202ee97c3..000000000000
--- a/include/sound/tegra_wm8903.h
+++ /dev/null
@@ -1,26 +0,0 @@
-/*
- * Copyright 2011 NVIDIA, Inc.
- *
- * This software is licensed under the terms of the GNU General Public
- * License version 2, as published by the Free Software Foundation, and
- * may be copied, distributed, and modified under those terms.
- *
- * 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 __SOUND_TEGRA_WM38903_H
-#define __SOUND_TEGRA_WM38903_H
-
-struct tegra_wm8903_platform_data {
-	int gpio_spkr_en;
-	int gpio_hp_det;
-	int gpio_hp_mute;
-	int gpio_int_mic_en;
-	int gpio_ext_mic_en;
-};
-
-#endif
diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h
index 1774a5c3ef10..e3983d508272 100644
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@@ -214,7 +214,9 @@ typedef int __bitwise snd_pcm_format_t;
 #define	SNDRV_PCM_FORMAT_G723_24_1B	((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */
 #define	SNDRV_PCM_FORMAT_G723_40	((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */
 #define	SNDRV_PCM_FORMAT_G723_40_1B	((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */
-#define	SNDRV_PCM_FORMAT_LAST		SNDRV_PCM_FORMAT_G723_40_1B
+#define	SNDRV_PCM_FORMAT_DSD_U8		((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD (x8) */
+#define	SNDRV_PCM_FORMAT_DSD_U16_LE	((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD (x16), little endian */
+#define	SNDRV_PCM_FORMAT_LAST		SNDRV_PCM_FORMAT_DSD_U16_LE
 
 #ifdef SNDRV_LITTLE_ENDIAN
 #define	SNDRV_PCM_FORMAT_S16		SNDRV_PCM_FORMAT_S16_LE