Merge tag 'sound-4.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This became a large update. The changes are scattered widely, and the
  majority of them are attributed to ASoC componentization. The gitk
  output made me dizzy, but it's slightly better than London tube.

  OK, below are some highlights:

   - Continued hardening works in ALSA PCM core; most of the existing
     syzkaller reports should have been covered.

   - USB-audio got the initial USB Audio Class 3 support, as well as
     UAC2 jack detection support and more DSD-device support.

   - ASoC componentization: finally each individual driver was converted
     to components framework, which is more future-proof for further
     works. Most of conversations were systematic.

   - Lots of fixes for Intel Baytrail / Cherrytrail devices with Realtek
     codecs, typically tablets and small PCs.

   - Fixes / cleanups for Samsung Odroid systems

   - Cleanups in Freescale SSI driver

   - New ASoC drivers:
      * AKM AK4458 and AK5558 codecs
      * A few AMD based machine drivers
      * Intel Kabylake machine drivers
      * Maxim MAX9759 codec
      * Motorola CPCAP codec
      * Socionext Uniphier SoCs
      * TI PCM1789 and TDA7419 codecs

   - Retirement of Blackfin drivers along with architecture removal"

* tag 'sound-4.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (497 commits)
  ALSA: pcm: Fix UAF at PCM release via PCM timer access
  ALSA: usb-audio: silence a static checker warning
  ASoC: tscs42xx: Remove owner assignment from i2c_driver
  ASoC: mediatek: remove "simple-mfd" in the example
  ASoC: cpcap: replace codec to component
  ASoC: Intel: bytcr_rt5651: don't use codec anymore
  ASoC: amd: don't use codec anymore
  ALSA: usb-audio: fix memory leak on cval
  ALSA: pcm: Fix mutex unbalance in OSS emulation ioctls
  ASoC: topology: Fix kcontrol name string handling
  ALSA: aloop: Mark paused device as inactive
  ALSA: usb-audio: update clock valid control
  ALSA: usb-audio: UAC2 jack detection
  ALSA: pcm: Return -EBUSY for OSS ioctls changing busy streams
  ALSA: pcm: Avoid potential races between OSS ioctls and read/write
  ALSA: usb-audio: Integrate native DSD support for ITF-USB based DACs.
  ALSA: usb-audio: FIX native DSD support for TEAC UD-501 DAC
  ALSA: usb-audio: Add native DSD support for Luxman DA-06
  ALSA: usb-audio: fix uac control query argument
  ASoC: nau8824: recover system clock when device changes
  ...

9 files changed, 43 insertions, 33 deletions

diff --git a/include/sound/da7219.h b/include/sound/da7219.h
index 409ef1397fd3..1bfcb16f2d10 100644
--- a/include/sound/da7219.h
+++ b/include/sound/da7219.h
@@ -36,6 +36,8 @@ struct da7219_aad_pdata;
 struct da7219_pdata {
 	bool wakeup_source;
 
+	const char *dai_clks_name;
+
 	/* Mic */
 	enum da7219_micbias_voltage micbias_lvl;
 	enum da7219_mic_amp_in_sel mic_amp_in_sel;
diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index 67be2445941a..e3481eebdd98 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -118,6 +118,8 @@ void snd_dmaengine_pcm_set_config_from_dai_data(
  *   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.
+ * @process: Callback used to apply processing on samples transferred from/to
+ *   user space.
  * @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.
@@ -140,6 +142,9 @@ struct snd_dmaengine_pcm_config {
 	struct dma_chan *(*compat_request_channel)(
 			struct snd_soc_pcm_runtime *rtd,
 			struct snd_pcm_substream *substream);
+	int (*process)(struct snd_pcm_substream *substream,
+		       int channel, unsigned long hwoff,
+		       void *buf, unsigned long bytes);
 	dma_filter_fn compat_filter_fn;
 	struct device *dma_dev;
 	const char *chan_names[SNDRV_PCM_STREAM_LAST + 1];
@@ -161,4 +166,6 @@ int snd_dmaengine_pcm_prepare_slave_config(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params,
 	struct dma_slave_config *slave_config);
 
+#define SND_DMAENGINE_PCM_DRV_NAME "snd_dmaengine_pcm"
+
 #endif
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 4f42affe777c..5ebcc51c0a6a 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -1710,6 +1710,7 @@ struct snd_emu10k1 {
 	unsigned int ecard_ctrl;		/* ecard control bits */
 	unsigned int address_mode;		/* address mode */
 	unsigned long dma_mask;			/* PCI DMA mask */
+	bool iommu_workaround;			/* IOMMU workaround needed */
 	unsigned int delay_pcm_irq;		/* in samples */
 	int max_cache_pages;			/* max memory size / PAGE_SIZE */
 	struct snd_dma_buffer silent_page;	/* silent page */
@@ -1718,7 +1719,6 @@ struct snd_emu10k1 {
 	struct snd_dma_buffer p16v_buffer;
 
 	struct snd_util_memhdr *memhdr;		/* page allocation list */
-	struct snd_emu10k1_memblk *reserved_page;	/* reserved page */
 
 	struct list_head mapped_link_head;
 	struct list_head mapped_order_link_head;
@@ -1878,6 +1878,8 @@ void snd_p16v_resume(struct snd_emu10k1 *emu);
 /* memory allocation */
 struct snd_util_memblk *snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *substream);
 int snd_emu10k1_free_pages(struct snd_emu10k1 *emu, struct snd_util_memblk *blk);
+int snd_emu10k1_alloc_pages_maybe_wider(struct snd_emu10k1 *emu, size_t size,
+					struct snd_dma_buffer *dmab);
 struct snd_util_memblk *snd_emu10k1_synth_alloc(struct snd_emu10k1 *emu, unsigned int size);
 int snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *blk);
 int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk, int offset, int size);
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index 5b2ed12f58ce..06536e01ed94 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -146,6 +146,8 @@ int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid,
 			int flags, unsigned int verb, unsigned int parm);
 bool snd_hdac_check_power_state(struct hdac_device *hdac,
 		hda_nid_t nid, unsigned int target_state);
+unsigned int snd_hdac_sync_power_state(struct hdac_device *hdac,
+		      hda_nid_t nid, unsigned int target_state);
 /**
  * snd_hdac_read_parm - read a codec parameter
  * @codec: the codec object
diff --git a/include/sound/pcm_oss.h b/include/sound/pcm_oss.h
index 760c969d885d..12bbf8c81112 100644
--- a/include/sound/pcm_oss.h
+++ b/include/sound/pcm_oss.h
@@ -57,6 +57,7 @@ struct snd_pcm_oss_runtime {
 	char *buffer;				/* vmallocated period */
 	size_t buffer_used;			/* used length from period buffer */
 	struct mutex params_lock;
+	atomic_t rw_ref;		/* concurrent read/write accesses */
 #ifdef CONFIG_SND_PCM_OSS_PLUGINS
 	struct snd_pcm_plugin *plugin_first;
 	struct snd_pcm_plugin *plugin_last;
diff --git a/include/sound/rt5651.h b/include/sound/rt5651.h
deleted file mode 100644
index 18b79a761f10..000000000000
--- a/include/sound/rt5651.h
+++ /dev/null
@@ -1,29 +0,0 @@
-/*
- * linux/sound/rt286.h -- Platform data for RT286
- *
- * Copyright 2013 Realtek Microelectronics
- *
- * 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.
- */
-
-#ifndef __LINUX_SND_RT5651_H
-#define __LINUX_SND_RT5651_H
-
-enum rt5651_jd_src {
-	RT5651_JD_NULL,
-	RT5651_JD1_1,
-	RT5651_JD1_2,
-	RT5651_JD2,
-};
-
-struct rt5651_platform_data {
-	/* IN2 can optionally be differential */
-	bool in2_diff;
-
-	bool dmic_en;
-	enum rt5651_jd_src jd_src;
-};
-
-#endif
diff --git a/include/sound/rt5659.h b/include/sound/rt5659.h
index 656c4d58948d..9012e2b25360 100644
--- a/include/sound/rt5659.h
+++ b/include/sound/rt5659.h
@@ -30,6 +30,7 @@ enum rt5659_dmic2_data_pin {
 enum rt5659_jd_src {
 	RT5659_JD_NULL,
 	RT5659_JD3,
+	RT5659_JD_HDA_HEADER,
 };
 
 struct rt5659_platform_data {
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 344b96c206a3..a6ce2de4e20a 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -269,6 +269,13 @@ struct device;
 	.reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \
 	.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \
 	.on_val = wflags}
+#define SND_SOC_DAPM_PINCTRL(wname, active, sleep) \
+{	.id = snd_soc_dapm_pinctrl, .name = wname, \
+	.priv = (&(struct snd_soc_dapm_pinctrl_priv) \
+		{ .active_state = active, .sleep_state = sleep,}), \
+	.reg = SND_SOC_NOPM, .event = dapm_pinctrl_event, \
+	.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD }
+
 
 
 /* dapm kcontrol types */
@@ -374,6 +381,8 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w,
 			 struct snd_kcontrol *kcontrol, int event);
 int dapm_clock_event(struct snd_soc_dapm_widget *w,
 			 struct snd_kcontrol *kcontrol, int event);
+int dapm_pinctrl_event(struct snd_soc_dapm_widget *w,
+			 struct snd_kcontrol *kcontrol, int event);
 
 /* dapm controls */
 int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
@@ -500,6 +509,7 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_pre,			/* machine specific pre widget - exec first */
 	snd_soc_dapm_post,			/* machine specific post widget - exec last */
 	snd_soc_dapm_supply,		/* power/clock supply */
+	snd_soc_dapm_pinctrl,		/* pinctrl */
 	snd_soc_dapm_regulator_supply,	/* external regulator */
 	snd_soc_dapm_clock_supply,	/* external clock */
 	snd_soc_dapm_aif_in,		/* audio interface input */
@@ -581,6 +591,7 @@ struct snd_soc_dapm_widget {
 
 	void *priv;				/* widget specific data */
 	struct regulator *regulator;		/* attached regulator */
+	struct pinctrl *pinctrl;		/* attached pinctrl */
 	const struct snd_soc_pcm_stream *params; /* params for dai links */
 	unsigned int num_params; /* number of params for dai links */
 	unsigned int params_select; /* currently selected param for dai link */
@@ -683,6 +694,11 @@ struct snd_soc_dapm_stats {
 	int neighbour_checks;
 };
 
+struct snd_soc_dapm_pinctrl_priv {
+	const char *active_state;
+	const char *sleep_state;
+};
+
 /**
  * snd_soc_dapm_init_bias_level() - Initialize DAPM bias level
  * @dapm: The DAPM context to initialize
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 747fd583b9dc..ad266d7e9553 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -586,10 +586,17 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg,
 				unsigned int mask, unsigned int value);
 
 #ifdef CONFIG_SND_SOC_AC97_BUS
-struct snd_ac97 *snd_soc_alloc_ac97_codec(struct snd_soc_codec *codec);
-struct snd_ac97 *snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
+#define snd_soc_alloc_ac97_codec(codec) \
+	snd_soc_alloc_ac97_component(&codec->component)
+#define snd_soc_new_ac97_codec(codec, id, id_mask) \
+	snd_soc_new_ac97_component(&codec->component, id, id_mask)
+#define snd_soc_free_ac97_codec(ac97) \
+	snd_soc_free_ac97_component(ac97)
+
+struct snd_ac97 *snd_soc_alloc_ac97_component(struct snd_soc_component *component);
+struct snd_ac97 *snd_soc_new_ac97_component(struct snd_soc_component *component,
 	unsigned int id, unsigned int id_mask);
-void snd_soc_free_ac97_codec(struct snd_ac97 *ac97);
+void snd_soc_free_ac97_component(struct snd_ac97 *ac97);
 
 int snd_soc_set_ac97_ops(struct snd_ac97_bus_ops *ops);
 int snd_soc_set_ac97_ops_of_reset(struct snd_ac97_bus_ops *ops,
@@ -1807,6 +1814,7 @@ int snd_soc_of_get_dai_name(struct device_node *of_node,
 int snd_soc_of_get_dai_link_codecs(struct device *dev,
 				   struct device_node *of_node,
 				   struct snd_soc_dai_link *dai_link);
+void snd_soc_of_put_dai_link_codecs(struct snd_soc_dai_link *dai_link);
 
 int snd_soc_add_dai_link(struct snd_soc_card *card,
 				struct snd_soc_dai_link *dai_link);