sound updates for 4.7-rc1

This time was again a relatively calm development cycle; most of updates are about drivers, and no radical changes are seen in any core code. Here are some highlights: ALSA core: - Continued hardening of ALSA hrtimer - A few leak fixes in timer interface - Fix poll error handling in PCM and compress - Add error propagation in compress API - Removal of dead rtctimer driver HD-audio: - Native ELD notify support for i915 HDMI - Realtek ALC234 & co support - Code refactoring to standardize chmap support - Continued development for SKL HDMI core support Firewire: - Apply delayed card registration to all drivers - Improved / stabilized the handling of PCM stream start / stop - Add tracepoints to dump a part of isochronous packet data - Fixed incoming/outgoing packet parameter usages - Add support for M-Audio profire series USB-audio: - Fixes for UAC2 clock source - SS+ support - Workaround for oft-seen repeated sample rate read errors ASoC: - Further slow progress on the topology code - Substantial updates and improvements for the da7219, es8328, fsl-ssi, Intel and rcar drivers. - Compress error handling in WM ADSP driver -----BEGIN PGP SIGNATURE----- Version: GnuPG v2 iQIcBAABCAAGBQJXPYgvAAoJEGwxgFQ9KSmka3IQAJfXxKYyL0mqOgUpFav2QprE j4nQFSQf2KMAHgod1iF4Pv5glRZ3T8CbWllu/+GT87ny4wwJH76D07VCZSnrA+cv NMxRMN8QiGWS+eNPDNqRbcpzQvgwRK17VAmvpIfZtdntq3IryPLyCnY+FJ6Xt5v7 CjgGjlKJQ8i6AJVtoKVlrCOTBPS8YezQ7o67v8+BNrHDyOr0pwLERhvqJBRjaCbj fKj+JNDsWyu4kX0nInKNGah+5Qiib68+UNK5M+/PnoWv9tEOBPNXeWqRkcRpwnrF t1BQLnKGdlcSIufXcvxHDdxLftJZ38w+EbnQ/2r+SYHYIwPqTWdvVeXZUiq70wW/ WBUEOHybaHTNc52nMpjo/PU72CHa29zvKq+QHMXMRmFfVrLepIgEpBRBUjENtCjM 3OUn1IhYiNI4FOfgLm5duuYSBVdS4C2qstBDMtGpP64l7AmBZMFtbGUP8pKhvpzF FR2VoQpBFLPo805lQBKYbxdpzUGqfR7M/O73WRMzB/ZPZa95VNCDoRDQBbYF4Wzy SByVcE56znxoS9AmbhU6LzCXxdyVp6YAXZNR0pHp+8QdrRoFQZwRhfNVN3FIeNub COV+0pCQ2GTYvVdfLjdh6VT4shXeg5ZrUVnE3akL+8OzXow9lKyhknvLHn71aTZi HT0vSirSdrEYf4zg6wtB =QsAc -----END PGP SIGNATURE----- Merge tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound updates from Takashi Iwai: "This time was again a relatively calm development cycle; most of updates are about drivers, and no radical changes are seen in any core code. Here are some highlights: ALSA core: - Continued hardening of ALSA hrtimer - A few leak fixes in timer interface - Fix poll error handling in PCM and compress - Add error propagation in compress API - Removal of dead rtctimer driver HD-audio: - Native ELD notify support for i915 HDMI - Realtek ALC234 & co support - Code refactoring to standardize chmap support - Continued development for SKL HDMI core support Firewire: - Apply delayed card registration to all drivers - Improved / stabilized the handling of PCM stream start / stop - Add tracepoints to dump a part of isochronous packet data - Fixed incoming/outgoing packet parameter usages - Add support for M-Audio profire series USB-audio: - Fixes for UAC2 clock source - SS+ support - Workaround for oft-seen repeated sample rate read errors ASoC: - Further slow progress on the topology code - Substantial updates and improvements for the da7219, es8328, fsl-ssi, Intel and rcar drivers. - Compress error handling in WM ADSP driver" * tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (177 commits) ALSA: firewire-lib: change a member of event structure to suppress sparse wanings to bool type sound: oss: Use setup_timer and mod_timer. ASoC: hdac_hdmi: Remove the unused 'timeout' variable ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix samples being dropped at Playback startup ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: pcm5102a: Add support for PCM5102A codec ASoC: hdac_hdmi: add link management ASoC: Intel: Skylake: add link management ALSA: hdac: add link pm and ref counting ALSA: au88x0: Fix zero clear of stream->resources ASoC: rt298: Add DMI match for Broxton-P reference platform ASoC: rt298: fix null deref on acpi driver data ASoC: dapm: deprecate MICBIAS widget type ALSA: firewire-lib: drop skip argument from helper functions to queue a packet ALSA: firewire-lib: add context information to tracepoints ALSA: firewire-lib: permit to flush queued packets only in process context for better PCM period granularity ...
2026-10-18 21:58:11 +07:00 · 2016-05-19 13:41:32 -07:00 · 2016-05-19 13:41:32 -07:00 · f4c80d5a16
commit f4c80d5a16
parent 7afd16f882 17e1717c11
144 changed files with 4737 additions and 1525 deletions
--- a/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt
+++ 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";
+};
--- 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
--- a/Documentation/devicetree/bindings/sound/pcm5102a.txt
+++ 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";
+	};
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@ -655,17 +655,6 @@ development branches in general while the development for the current
 and next kernels are found in for-linus and for-next branches,
 respectively.

-If you are using the latest Linus tree, it'd be better to pull the
-above GIT tree onto it.  If you are using the older kernels, an easy
-way to try the latest ALSA code is to build from the snapshot
-tarball.  There are daily tarballs and the latest snapshot tarball.
-All can be built just like normal alsa-driver release packages, that
-is, installed via the usual spells: configure, make and make
-install(-modules).  See INSTALL in the package.  The snapshot tarballs
-are found at:
-
- ftp://ftp.suse.com/pub/people/tiwai/snapshot/
-

 Sending a Bug Report
 ~~~~~~~~~~~~~~~~~~~~
@ -699,7 +688,12 @@ problems.
 alsa-info
 ~~~~~~~~~
 The script `alsa-info.sh` is a very useful tool to gather the audio
-device information.  You can fetch the latest version from:
+device information.  It's included in alsa-utils package.  The latest
+version can be found on git repository:
+
+- git://git.alsa-project.org/alsa-utils.git
+
+The script can be fetched directly from the following URL, too:

 - http://www.alsa-project.org/alsa-info.sh

@ -836,15 +830,11 @@ can get a proc-file dump at the current state, get a list of control
 (mixer) elements, set/get the control element value, simulate the PCM
 operation, the jack plugging simulation, etc.

-The package is found in:
-
- ftp://ftp.suse.com/pub/people/tiwai/misc/
-
-A git repository is available:
+The program is found in the git repository below:

 - git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git

-See README file in the tarball for more details about hda-emu
+See README file in the repository for more details about hda-emu
 program.


--- a/Documentation/sound/alsa/compress_offload.txt
+++ b/Documentation/sound/alsa/compress_offload.txt
@ -149,7 +149,7 @@ Gapless Playback
 ================
 When playing thru an album, the decoders have the ability to skip the encoder
 delay and padding and directly move from one track content to another. The end
-user can perceive this as gapless playback as we dont have silence while
+user can perceive this as gapless playback as we don't have silence while
 switching from one track to another

 Also, there might be low-intensity noises due to encoding. Perfect gapless is
@ -184,7 +184,7 @@ Sequence flow for gapless would be:
 - Fill data of the first track
 - Trigger start
 - User-space finished sending all,
- Indicaite next track data by sending set_next_track
+- Indicate next track data by sending set_next_track
 - Set metadata of the next track
 - then call partial_drain to flush most of buffer in DSP
 - Fill data of the next track
--- a/Documentation/sound/alsa/soc/dapm.txt
+++ b/Documentation/sound/alsa/soc/dapm.txt
@ -132,7 +132,7 @@ SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0),
 SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls,
 	ARRAY_SIZE(wm8731_output_mixer_controls)),

-If you dont want the mixer elements prefixed with the name of the mixer widget,
+If you don't want the mixer elements prefixed with the name of the mixer widget,
 you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same
 as for SND_SOC_DAPM_MIXER.

--- a/Documentation/sound/alsa/soc/overview.txt
+++ b/Documentation/sound/alsa/soc/overview.txt
@ -63,7 +63,7 @@ multiple re-usable component drivers :-
    and any audio DSP drivers for that platform.

  * Machine class driver: The machine driver class acts as the glue that
-    decribes and binds the other component drivers together to form an ALSA
+    describes and binds the other component drivers together to form an ALSA
    "sound card device". It handles any machine specific controls and
    machine level audio events (e.g. turning on an amp at start of playback).

--- a/Documentation/sound/alsa/timestamping.txt
+++ b/Documentation/sound/alsa/timestamping.txt
@ -129,7 +129,7 @@ will be required to issue multiple queries and perform an
 interpolation of the results

 In some hardware-specific configuration, the system timestamp is
-latched by a low-level audio subsytem, and the information provided
+latched by a low-level audio subsystem, and the information provided
 back to the driver. Due to potential delays in the communication with
 the hardware, there is a risk of misalignment with the avail and delay
 information. To make sure applications are not confused, a
--- a/1
+++ b/1
@ -4732,6 +4732,7 @@ 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
--- a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h
+++ b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h
@ -448,5 +448,11 @@
 #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 */
--- 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(
--- a/include/sound/hda_chmap.h
+++ b/include/sound/hda_chmap.h
@ -36,6 +36,8 @@ struct hdac_chmap_ops {
 	int (*chmap_validate)(struct hdac_chmap *hchmap, int ca,
 			int channels, unsigned char *chmap);

+	int (*get_spk_alloc)(struct hdac_device *hdac, int pcm_idx);
+
 	void (*get_chmap)(struct hdac_device *hdac, int pcm_idx,
 					unsigned char *chmap);
 	void (*set_chmap)(struct hdac_device *hdac, int pcm_idx,
--- a/include/sound/hda_i915.h
+++ b/include/sound/hda_i915.h
@ -10,8 +10,8 @@
 int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable);
 int snd_hdac_display_power(struct hdac_bus *bus, bool enable);
 void snd_hdac_i915_set_bclk(struct hdac_bus *bus);
-int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate);
-int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid,
+int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate);
+int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid,
 			   bool *audio_enabled, char *buffer, int max_bytes);
 int snd_hdac_i915_init(struct hdac_bus *bus);
 int snd_hdac_i915_exit(struct hdac_bus *bus);
@ -28,12 +28,12 @@ static inline int snd_hdac_display_power(struct hdac_bus *bus, bool enable)
 static inline void snd_hdac_i915_set_bclk(struct hdac_bus *bus)
 {
 }
-static inline int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid,
-					   int rate)
+static inline int snd_hdac_sync_audio_rate(struct hdac_device *codec,
+					   hda_nid_t nid, int rate)
 {
 	return 0;
 }
-static inline int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid,
+static inline int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid,
 					 bool *audio_enabled, char *buffer,
 					 int max_bytes)
 {
--- 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)), \
--- a/include/sound/hdmi-codec.h
+++ 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__ */
--- 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
--- 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 */
--- 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(
+	const struct snd_soc_dai_link_component *dlc);
+
 #include <sound/soc-dai.h>

 #ifdef CONFIG_DEBUG_FS
--- a/include/uapi/sound/asound.h
+++ b/include/uapi/sound/asound.h
@ -672,7 +672,7 @@ enum {

 /* global timers (device member) */
 #define SNDRV_TIMER_GLOBAL_SYSTEM	0
-#define SNDRV_TIMER_GLOBAL_RTC		1
+#define SNDRV_TIMER_GLOBAL_RTC		1	/* unused */
 #define SNDRV_TIMER_GLOBAL_HPET		2
 #define SNDRV_TIMER_GLOBAL_HRTIMER	3

--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@ -141,35 +141,6 @@ config SND_SEQ_HRTIMER_DEFAULT
 	  Say Y here to use the HR-timer backend as the default sequencer
 	  timer.

-config SND_RTCTIMER
-	tristate "RTC Timer support"
-	depends on RTC
-	select SND_TIMER
-	help
-	  Say Y here to enable RTC timer support for ALSA.  ALSA uses
-	  the RTC timer as a precise timing source and maps the RTC
-	  timer to ALSA's timer interface.  The ALSA sequencer code also
-	  can use this timing source.
-
-	  To compile this driver as a module, choose M here: the module
-	  will be called snd-rtctimer.
-
-	  Note that this option is exclusive with the new RTC drivers
-	  (CONFIG_RTC_CLASS) since this requires the old API.
-
-config SND_SEQ_RTCTIMER_DEFAULT
-	bool "Use RTC as default sequencer timer"
-	depends on SND_RTCTIMER && SND_SEQUENCER
-	depends on !SND_SEQ_HRTIMER_DEFAULT
-	default y
-	help
-	  Say Y here to use the RTC timer as the default sequencer
-	  timer.  This is strongly recommended because it ensures
-	  precise MIDI timing even when the system timer runs at less
-	  than 1000 Hz.
-
-	  If in doubt, say Y.
-
 config SND_DYNAMIC_MINORS
 	bool "Dynamic device file minor numbers"
 	help
--- a/Show more
+++ b/Show more