Merge tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kerne…

…l/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
  ...

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";
+};

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

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";
+	};

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.
 
 

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

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.
 

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).
 

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

MAINTAINERS

@@ -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

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 */

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(

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,

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)
 {

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)), \