Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus

2013-09-11 11:17:15 +01:00 · 2013-09-11 11:17:15 +01:00 · c34c0d7684
commit c34c0d7684
parent 29dc5dd229 4345adf92d
63 changed files with 2174 additions and 5017 deletions
--- a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
+++ b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
@ -0,0 +1,34 @@
+Freescale i.MX audio complex with S/PDIF transceiver
+
+Required properties:
+
+  - compatible : "fsl,imx-audio-spdif"
+
+  - model : The user-visible name of this sound complex
+
+  - spdif-controller : The phandle of the i.MX S/PDIF controller
+
+
+Optional properties:
+
+  - spdif-out : This is a boolean property. If present, the transmitting
+    function of S/PDIF will be enabled, indicating there's a physical
+    S/PDIF out connector/jack on the board or it's connecting to some
+    other IP block, such as an HDMI encoder/display-controller.
+
+  - spdif-in : This is a boolean property. If present, the receiving
+    function of S/PDIF will be enabled, indicating there's a physical
+    S/PDIF in connector/jack on the board.
+
+* Note: At least one of these two properties should be set in the DT binding.
+
+
+Example:
+
+sound-spdif {
+	compatible = "fsl,imx-audio-spdif";
+	model = "imx-spdif";
+	spdif-controller = <&spdif>;
+	spdif-out;
+	spdif-in;
+};
--- a/Documentation/devicetree/bindings/sound/mvebu-audio.txt
+++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt
@ -0,0 +1,29 @@
+* mvebu (Kirkwood, Dove, Armada 370) audio controller
+
+Required properties:
+
+- compatible: "marvell,mvebu-audio"
+
+- reg: physical base address of the controller and length of memory mapped
+  region.
+
+- interrupts: list of two irq numbers.
+  The first irq is used for data flow and the second one is used for errors.
+
+- clocks: one or two phandles.
+  The first one is mandatory and defines the internal clock.
+  The second one is optional and defines an external clock.
+
+- clock-names: names associated to the clocks:
+	"internal" for the internal clock
+	"extclk" for the external clock
+
+Example:
+
+i2s1: audio-controller@b4000 {
+	compatible = "marvell,mvebu-audio";
+	reg = <0xb4000 0x2210>;
+	interrupts = <21>, <22>;
+	clocks = <&gate_clk 13>;
+	clock-names = "internal";
+};
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@ -244,6 +244,7 @@ STAC9227/9228/9229/927x
  5stack-no-fp	D965 5stack without front panel
  dell-3stack	Dell Dimension E520
  dell-bios	Fixes with Dell BIOS setup
+  dell-bios-amic Fixes with Dell BIOS setup including analog mic
  volknob	Fixes with volume-knob widget 0x24
  auto		BIOS setup (default)

--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@ -454,6 +454,8 @@ The generic parser supports the following hints:
 - need_dac_fix (bool): limits the DACs depending on the channel count
 - primary_hp (bool): probe headphone jacks as the primary outputs;
  default true
+- multi_io (bool): try probing multi-I/O config (e.g. shared
+  line-in/surround, mic/clfe jacks)
 - multi_cap_vol (bool): provide multiple capture volumes
 - inv_dmic_split (bool): provide split internal mic volume/switch for
  phase-inverted digital mics
--- a/11
+++ b/11
@ -7676,6 +7676,17 @@ F:	include/sound/
 F:	include/uapi/sound/
 F:	sound/

+SOUND - COMPRESSED AUDIO
+M:	Vinod Koul <vinod.koul@intel.com>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+T:	git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
+S:	Supported
+F:	Documentation/sound/alsa/compress_offload.txt
+F:	include/sound/compress_driver.h
+F:	include/uapi/sound/compress_*
+F:	sound/core/compress_offload.c
+F:	sound/soc/soc-compress.c
+
 SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC)
 M:	Liam Girdwood <lgirdwood@gmail.com>
 M:	Mark Brown <broonie@kernel.org>
--- a/arch/arm/plat-samsung/s3c-dma-ops.c
+++ b/arch/arm/plat-samsung/s3c-dma-ops.c
@ -82,7 +82,8 @@ static int s3c_dma_config(unsigned ch, struct samsung_dma_config *param)
 static int s3c_dma_prepare(unsigned ch, struct samsung_dma_prep *param)
 {
 	struct cb_data *data;
-	int len = (param->cap == DMA_CYCLIC) ? param->period : param->len;
+	dma_addr_t pos = param->buf;
+	dma_addr_t end = param->buf + param->len;

 	list_for_each_entry(data, &dma_list, node)
 		if (data->ch == ch)
@ -94,7 +95,15 @@ static int s3c_dma_prepare(unsigned ch, struct samsung_dma_prep *param)
 		data->fp_param = param->fp_param;
 	}

-	s3c2410_dma_enqueue(ch, (void *)data, param->buf, len);
+	if (param->cap != DMA_CYCLIC) {
+		s3c2410_dma_enqueue(ch, (void *)data, param->buf, param->len);
+		return 0;
+	}
+
+	while (pos < end) {
+		s3c2410_dma_enqueue(ch, (void *)data, pos, param->period);
+		pos += param->period;
+	}

 	return 0;
 }
--- a/include/sound/core.h
+++ b/include/sound/core.h
@ -27,6 +27,7 @@
 #include <linux/rwsem.h>		/* struct rw_semaphore */
 #include <linux/pm.h>			/* pm_message_t */
 #include <linux/stringify.h>
+#include <linux/printk.h>

 /* number of supported soundcards */
 #ifdef CONFIG_SND_DYNAMIC_MINORS
@ -375,6 +376,11 @@ void __snd_printk(unsigned int level, const char *file, int line,
 */
 #define snd_BUG()		WARN(1, "BUG?\n")

+/**
+ * Suppress high rates of output when CONFIG_SND_DEBUG is enabled.
+ */
+#define snd_printd_ratelimit() printk_ratelimit()
+
 /**
 * snd_BUG_ON - debugging check macro
 * @cond: condition to evaluate
@ -398,6 +404,8 @@ static inline void _snd_printd(int level, const char *format, ...) {}
 	unlikely(__ret_warn_on); \
 })

+static inline bool snd_printd_ratelimit(void) { return false; }
+
 #endif /* CONFIG_SND_DEBUG */

 #ifdef CONFIG_SND_DEBUG_VERBOSE
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@ -413,7 +413,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
 			 struct snd_soc_dapm_widget *sink);

 /* dapm path setup */
-int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
+int snd_soc_dapm_new_widgets(struct snd_soc_card *card);
 void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
 int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
 			    const struct snd_soc_dapm_route *route, int num);
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@ -697,7 +697,6 @@ struct snd_soc_codec {
 	unsigned int probed:1; /* Codec has been probed */
 	unsigned int ac97_registered:1; /* Codec has been AC97 registered */
 	unsigned int ac97_created:1; /* Codec has been created by SoC */
-	unsigned int sysfs_registered:1; /* codec has been sysfs registered */
 	unsigned int cache_init:1; /* codec cache has been initialized */
 	unsigned int using_regmap:1; /* using regmap access */
 	u32 cache_only;  /* Suppress writes to hardware */
@ -705,7 +704,6 @@ struct snd_soc_codec {

 	/* codec IO */
 	void *control_data; /* codec control (i2c/3wire) data */
-	enum snd_soc_control_type control_type;
 	hw_write_t hw_write;
 	unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int);
 	unsigned int (*read)(struct snd_soc_codec *, unsigned int);
@ -724,7 +722,6 @@ struct snd_soc_codec {
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_codec_root;
 	struct dentry *debugfs_reg;
-	struct dentry *debugfs_dapm;
 #endif
 };

@ -849,7 +846,6 @@ struct snd_soc_platform {

 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_platform_root;
-	struct dentry *debugfs_dapm;
 #endif
 };

@ -934,6 +930,10 @@ struct snd_soc_dai_link {
 	/* machine stream operations */
 	const struct snd_soc_ops *ops;
 	const struct snd_soc_compr_ops *compr_ops;
+
+	/* For unidirectional dai links */
+	bool playback_only;
+	bool capture_only;
 };

 struct snd_soc_codec_conf {
--- a/include/uapi/sound/hdspm.h
+++ b/include/uapi/sound/hdspm.h
@ -111,7 +111,7 @@ struct hdspm_ltc {
 	enum hdspm_ltc_input_format input_format;
 };

-#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_mixer_ioctl)
+#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_ltc)

 /**
 * The status data reflects the device's current state
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@ -184,7 +184,7 @@ static void xrun(struct snd_pcm_substream *substream)
 	do {								\
 		if (xrun_debug(substream, XRUN_DEBUG_BASIC)) {		\
 			xrun_log_show(substream);			\
-			if (printk_ratelimit()) {			\
+			if (snd_printd_ratelimit()) {			\
 				snd_printd("PCM: " fmt, ##args);	\
 			}						\
 			dump_stack_on_xrun(substream);			\
@ -342,7 +342,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 		return -EPIPE;
 	}
 	if (pos >= runtime->buffer_size) {
-		if (printk_ratelimit()) {
+		if (snd_printd_ratelimit()) {
 			char name[16];
 			snd_pcm_debug_name(substream, name, sizeof(name));
 			xrun_log_show(substream);
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@ -1022,7 +1022,7 @@ static void dummy_proc_write(struct snd_info_entry *entry,
 		if (i >= ARRAY_SIZE(fields))
 			continue;
 		snd_info_get_str(item, ptr, sizeof(item));
-		if (strict_strtoull(item, 0, &val))
+		if (kstrtoull(item, 0, &val))
 			continue;
 		if (fields[i].size == sizeof(int))
 			*get_dummy_int_ptr(dummy, fields[i].offset) = val;
--- a/sound/firewire/speakers.c
+++ b/sound/firewire/speakers.c
@ -49,7 +49,6 @@ struct fwspk {
 	struct snd_card *card;
 	struct fw_unit *unit;
 	const struct device_info *device_info;
-	struct snd_pcm_substream *pcm;
 	struct mutex mutex;
 	struct cmp_connection connection;
 	struct amdtp_out_stream stream;
@ -363,8 +362,7 @@ static int fwspk_create_pcm(struct fwspk *fwspk)
 		return err;
 	pcm->private_data = fwspk;
 	strcpy(pcm->name, fwspk->device_info->short_name);
-	fwspk->pcm = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
-	fwspk->pcm->ops = &ops;
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops);
 	return 0;
 }

--- a/sound/isa/gus/interwave.c
+++ b/sound/isa/gus/interwave.c
@ -443,8 +443,7 @@ static void snd_interwave_detect_memory(struct snd_gus_card *gus)
 		for (i = 0; i < 8; ++i)
 			iwave[i] = snd_gf1_peek(gus, bank_pos + i);
 #ifdef CONFIG_SND_DEBUG_ROM
-		printk(KERN_DEBUG "ROM at 0x%06x = %*phC\n", bank_pos,
-				  8, iwave);
+		printk(KERN_DEBUG "ROM at 0x%06x = %8phC\n", bank_pos, iwave);
 #endif
 		if (strncmp(iwave, "INTRWAVE", 8))
 			continue;	/* first check */
--- a/sound/oss/dmabuf.c
+++ b/sound/oss/dmabuf.c
@ -557,7 +557,6 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock)
 	unsigned long flags;
 	int err = 0, n = 0;
 	struct dma_buffparms *dmap = adev->dmap_in;
-	int go;

 	if (!(adev->open_mode & OPEN_READ))
 		return -EIO;
@ -584,7 +583,7 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock)
 			spin_unlock_irqrestore(&dmap->lock,flags);
 			return -EAGAIN;
 		}
-		if ((go = adev->go))
+		if (adev->go)
 			timeout = dmabuf_timeout(dmap);

 		spin_unlock_irqrestore(&dmap->lock,flags);
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@ -152,14 +152,9 @@ config SND_HDA_CODEC_HDMI
 	  This module is automatically loaded at probing.

 config SND_HDA_I915
-	bool "Build Display HD-audio controller/codec power well support for i915 cards"
+	bool
+	default y
 	depends on DRM_I915
-	help
-	  Say Y here to include full HDMI and DisplayPort HD-audio controller/codec
-	  power-well support for Intel Haswell graphics cards based on the i915 driver.
-
-	  Note that this option must be enabled for Intel Haswell C+ stepping machines, otherwise
-	  the GPU audio controller/codecs will not be initialized or damaged when exit from S3 mode.

 config SND_HDA_CODEC_CIRRUS
 	bool "Build Cirrus Logic codec support"
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@ -666,6 +666,64 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
 }
 EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);

+
+/* return DEVLIST_LEN parameter of the given widget */
+static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid)
+{
+	unsigned int wcaps = get_wcaps(codec, nid);
+	unsigned int parm;
+
+	if (!codec->dp_mst || !(wcaps & AC_WCAP_DIGITAL) ||
+	    get_wcaps_type(wcaps) != AC_WID_PIN)
+		return 0;
+
+	parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN);
+	if (parm == -1 && codec->bus->rirb_error)
+		parm = 0;
+	return parm & AC_DEV_LIST_LEN_MASK;
+}
+
+/**
+ * snd_hda_get_devices - copy device list without cache
+ * @codec: the HDA codec
+ * @nid: NID of the pin to parse
+ * @dev_list: device list array
+ * @max_devices: max. number of devices to store
+ *
+ * Copy the device list. This info is dynamic and so not cached.
+ * Currently called only from hda_proc.c, so not exported.
+ */
+int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid,
+			u8 *dev_list, int max_devices)
+{
+	unsigned int parm;
+	int i, dev_len, devices;
+
+	parm = get_num_devices(codec, nid);
+	if (!parm)	/* not multi-stream capable */
+		return 0;
+
+	dev_len = parm + 1;
+	dev_len = dev_len < max_devices ? dev_len : max_devices;
+
+	devices = 0;
+	while (devices < dev_len) {
+		parm = snd_hda_codec_read(codec, nid, 0,
+					  AC_VERB_GET_DEVICE_LIST, devices);
+		if (parm == -1 && codec->bus->rirb_error)
+			break;
+
+		for (i = 0; i < 8; i++) {
+			dev_list[devices] = (u8)parm;
+			parm >>= 4;
+			devices++;
+			if (devices >= dev_len)
+				break;
+		}
+	}
+	return devices;
+}
+
 /**
 * snd_hda_queue_unsol_event - add an unsolicited event to queue
 * @bus: the BUS
@ -1216,11 +1274,13 @@ static void hda_jackpoll_work(struct work_struct *work)
 {
 	struct hda_codec *codec =
 		container_of(work, struct hda_codec, jackpoll_work.work);
-	if (!codec->jackpoll_interval)
-		return;

 	snd_hda_jack_set_dirty_all(codec);
 	snd_hda_jack_poll_all(codec);
+
+	if (!codec->jackpoll_interval)
+		return;
+
 	queue_delayed_work(codec->bus->workq, &codec->jackpoll_work,
 			   codec->jackpoll_interval);
 }
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@ -94,6 +94,8 @@ enum {
 #define AC_VERB_GET_HDMI_DIP_XMIT		0x0f32
 #define AC_VERB_GET_HDMI_CP_CTRL		0x0f33
 #define AC_VERB_GET_HDMI_CHAN_SLOT		0x0f34
+#define AC_VERB_GET_DEVICE_SEL			0xf35
+#define AC_VERB_GET_DEVICE_LIST			0xf36

 /*
 * SET verbs
@ -133,6 +135,7 @@ enum {
 #define AC_VERB_SET_HDMI_DIP_XMIT		0x732
 #define AC_VERB_SET_HDMI_CP_CTRL		0x733
 #define AC_VERB_SET_HDMI_CHAN_SLOT		0x734
+#define AC_VERB_SET_DEVICE_SEL			0x735

 /*
 * Parameter IDs
@ -154,6 +157,7 @@ enum {
 #define AC_PAR_GPIO_CAP			0x11
 #define AC_PAR_AMP_OUT_CAP		0x12
 #define AC_PAR_VOL_KNB_CAP		0x13
+#define AC_PAR_DEVLIST_LEN		0x15
 #define AC_PAR_HDMI_LPCM_CAP		0x20

 /*
@ -251,6 +255,11 @@ enum {
 #define AC_UNSOL_RES_TAG_SHIFT		26
 #define AC_UNSOL_RES_SUBTAG		(0x1f<<21)
 #define AC_UNSOL_RES_SUBTAG_SHIFT	21
+#define AC_UNSOL_RES_DE			(0x3f<<15)  /* Device Entry
+						     * (for DP1.2 MST)
+						     */
+#define AC_UNSOL_RES_DE_SHIFT		15
+#define AC_UNSOL_RES_IA			(1<<2)	/* Inactive (for DP1.2 MST) */
 #define AC_UNSOL_RES_ELDV		(1<<1)	/* ELD Data valid (for HDMI) */
 #define AC_UNSOL_RES_PD			(1<<0)	/* pinsense detect */
 #define AC_UNSOL_RES_CP_STATE		(1<<1)	/* content protection */
@ -352,6 +361,10 @@ enum {
 #define AC_LPCMCAP_44K			(1<<30)	/* 44.1kHz support */
 #define AC_LPCMCAP_44K_MS		(1<<31)	/* 44.1kHz-multiplies support */

+/* Display pin's device list length */
+#define AC_DEV_LIST_LEN_MASK		0x3f
+#define AC_MAX_DEV_LIST_LEN		64
+
 /*
 * Control Parameters
 */
@ -460,6 +473,11 @@ enum {
 #define AC_DEFCFG_PORT_CONN		(0x3<<30)
 #define AC_DEFCFG_PORT_CONN_SHIFT	30

+/* Display pin's device list entry */
+#define AC_DE_PD			(1<<0)
+#define AC_DE_ELDV			(1<<1)
+#define AC_DE_IA			(1<<2)
+
 /* device device types (0x0-0xf) */
 enum {
 	AC_JACK_LINE_OUT,
@ -885,6 +903,7 @@ struct hda_codec {
 	unsigned int pcm_format_first:1; /* PCM format must be set first */
 	unsigned int epss:1;		/* supporting EPSS? */
 	unsigned int cached_write:1;	/* write only to caches */
+	unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */
 #ifdef CONFIG_PM
 	unsigned int power_on :1;	/* current (global) power-state */
 	unsigned int d3_stop_clk:1;	/* support D3 operation without BCLK */
@ -972,6 +991,8 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums,
 			  const hda_nid_t *list);
 int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
 			   hda_nid_t nid, int recursive);
+int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid,
+			u8 *dev_list, int max_devices);
 int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 				u32 *ratesp, u64 *formatsp, unsigned int *bpsp);

--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@ -142,6 +142,9 @@ static void parse_user_hints(struct hda_codec *codec)
 	val = snd_hda_get_bool_hint(codec, "primary_hp");
 	if (val >= 0)
 		spec->no_primary_hp = !val;
+	val = snd_hda_get_bool_hint(codec, "multi_io");
+	if (val >= 0)
+		spec->no_multi_io = !val;
 	val = snd_hda_get_bool_hint(codec, "multi_cap_vol");
 	if (val >= 0)
 		spec->multi_cap_vol = !!val;
@ -813,6 +816,8 @@ static void resume_path_from_idx(struct hda_codec *codec, int path_idx)

 static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_value *ucontrol);
+static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol);

 enum {
 	HDA_CTL_WIDGET_VOL,
@ -830,7 +835,13 @@ static const struct snd_kcontrol_new control_templates[] = {
 		.put = hda_gen_mixer_mute_put, /* replaced */
 		.private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0),
 	},
-	HDA_BIND_MUTE(NULL, 0, 0, 0),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.info = snd_hda_mixer_amp_switch_info,
+		.get = snd_hda_mixer_bind_switch_get,
+		.put = hda_gen_bind_mute_put, /* replaced */
+		.private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0),
+	},
 };

 /* add dynamic controls from template */
@ -937,8 +948,8 @@ static int add_stereo_sw(struct hda_codec *codec, const char *pfx,
 }

 /* playback mute control with the software mute bit check */
-static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
-				  struct snd_ctl_elem_value *ucontrol)
+static void sync_auto_mute_bits(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct hda_gen_spec *spec = codec->spec;
@ -949,10 +960,22 @@ static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
 		ucontrol->value.integer.value[0] &= enabled;
 		ucontrol->value.integer.value[1] &= enabled;
 	}
+}

+static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	sync_auto_mute_bits(kcontrol, ucontrol);
 	return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
 }

+static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	sync_auto_mute_bits(kcontrol, ucontrol);
+	return snd_hda_mixer_bind_switch_put(kcontrol, ucontrol);
+}
+
 /* any ctl assigned to the path with the given index? */
 static bool path_has_mixer(struct hda_codec *codec, int path_idx, int ctl_type)
 {
@ -1541,7 +1564,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 					      cfg->speaker_pins,
 					      spec->multiout.extra_out_nid,
 					      spec->speaker_paths);
-			if (fill_mio_first && cfg->line_outs == 1 &&
+			if (!spec->no_multi_io &&
+			    fill_mio_first && cfg->line_outs == 1 &&
 			    cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
 				err = fill_multi_ios(codec, cfg->line_out_pins[0], true);
 				if (!err)
@ -1554,7 +1578,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 				   spec->private_dac_nids, spec->out_paths,
 				   spec->main_out_badness);

-	if (fill_mio_first &&
+	if (!spec->no_multi_io && fill_mio_first &&
 	    cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
 		/* try to fill multi-io first */
 		err = fill_multi_ios(codec, cfg->line_out_pins[0], false);
@ -1582,7 +1606,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 			return err;
 		badness += err;
 	}
-	if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
+	if (!spec->no_multi_io &&
+	    cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
 		err = fill_multi_ios(codec, cfg->line_out_pins[0], false);
 		if (err < 0)
 			return err;
@ -1600,7 +1625,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 				check_aamix_out_path(codec, spec->speaker_paths[0]);
 	}

-	if (cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+	if (!spec->no_multi_io &&
+	    cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
 		if (count_multiio_pins(codec, cfg->hp_pins[0]) >= 2)
 			spec->multi_ios = 1; /* give badness */

@ -3724,7 +3750,8 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx,
 /* check each pin in the given array; returns true if any of them is plugged */
 static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
 {
-	int i, present = 0;
+	int i;
+	bool present = false;

 	for (i = 0; i < num_pins; i++) {
 		hda_nid_t nid = pins[i];
@ -3733,14 +3760,15 @@ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
 		/* don't detect pins retasked as inputs */
 		if (snd_hda_codec_get_pin_target(codec, nid) & AC_PINCTL_IN_EN)
 			continue;
-		present |= snd_hda_jack_detect(codec, nid);
+		if (snd_hda_jack_detect_state(codec, nid) == HDA_JACK_PRESENT)
+			present = true;
 	}
 	return present;
 }

 /* standard HP/line-out auto-mute helper */
 static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
-			bool mute)
+			int *paths, bool mute)
 {
 	struct hda_gen_spec *spec = codec->spec;
 	int i;
@ -3752,10 +3780,19 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
 			break;

 		if (spec->auto_mute_via_amp) {
+			struct nid_path *path;
+			hda_nid_t mute_nid;
+
+			path = snd_hda_get_path_from_idx(codec, paths[i]);
+			if (!path)
+				continue;
+			mute_nid = get_amp_nid_(path->ctls[NID_PATH_MUTE_CTL]);
+			if (!mute_nid)
+				continue;
 			if (mute)
-				spec->mute_bits |= (1ULL << nid);
+				spec->mute_bits |= (1ULL << mute_nid);
 			else
-				spec->mute_bits &= ~(1ULL << nid);
+				spec->mute_bits &= ~(1ULL << mute_nid);
 			set_pin_eapd(codec, nid, !mute);
 			continue;
 		}
@ -3786,14 +3823,19 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
 void snd_hda_gen_update_outputs(struct hda_codec *codec)
 {
 	struct hda_gen_spec *spec = codec->spec;
+	int *paths;
 	int on;

 	/* Control HP pins/amps depending on master_mute state;
 	 * in general, HP pins/amps control should be enabled in all cases,
 	 * but currently set only for master_mute, just to be safe
 	 */
+	if (spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)
+		paths = spec->out_paths;
+	else
+		paths = spec->hp_paths;
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
-		    spec->autocfg.hp_pins, spec->master_mute);
+		    spec->autocfg.hp_pins, paths, spec->master_mute);

 	if (!spec->automute_speaker)
 		on = 0;
@ -3801,8 +3843,12 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec)
 		on = spec->hp_jack_present | spec->line_jack_present;
 	on |= spec->master_mute;
 	spec->speaker_muted = on;
+	if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
+		paths = spec->out_paths;
+	else
+		paths = spec->speaker_paths;
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins),
-		    spec->autocfg.speaker_pins, on);
+		    spec->autocfg.speaker_pins, paths, on);

 	/* toggle line-out mutes if needed, too */
 	/* if LO is a copy of either HP or Speaker, don't need to handle it */
@ -3815,8 +3861,9 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec)
 		on = spec->hp_jack_present;
 	on |= spec->master_mute;
 	spec->line_out_muted = on;
+	paths = spec->out_paths;
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
-		    spec->autocfg.line_out_pins, on);
+		    spec->autocfg.line_out_pins, paths, on);
 }
 EXPORT_SYMBOL_HDA(snd_hda_gen_update_outputs);

@ -3887,7 +3934,7 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja
 		/* don't detect pins retasked as outputs */
 		if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN)
 			continue;
-		if (snd_hda_jack_detect(codec, pin)) {
+		if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) {
 			mux_select(codec, 0, spec->am_entry[i].idx);
 			return;
 		}
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@ -220,6 +220,7 @@ struct hda_gen_spec {
 	unsigned int hp_mic:1; /* Allow HP as a mic-in */
 	unsigned int suppress_hp_mic_detect:1; /* Don't detect HP/mic */
 	unsigned int no_primary_hp:1; /* Don't prefer HP pins to speaker pins */
+	unsigned int no_multi_io:1; /* Don't try multi I/O config */
 	unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */
 	unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */
 	unsigned int own_eapd_ctl:1; /* set EAPD by own function */
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@ -295,7 +295,7 @@ static ssize_t type##_store(struct device *dev,			\
 	struct snd_hwdep *hwdep = dev_get_drvdata(dev);		\
 	struct hda_codec *codec = hwdep->private_data;		\
 	unsigned long val;					\
-	int err = strict_strtoul(buf, 0, &val);			\
+	int err = kstrtoul(buf, 0, &val);			\
 	if (err < 0)						\
 		return err;					\
 	codec->type = val;					\
@ -654,7 +654,7 @@ int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp)
 	p = snd_hda_get_hint(codec, key);
 	if (!p)
 		ret = -ENOENT;
-	else if (strict_strtoul(p, 0, &val))
+	else if (kstrtoul(p, 0, &val))
 		ret = -EINVAL;
 	else {
 		*valp = val;
@ -751,7 +751,7 @@ static void parse_##name##_mode(char *buf, struct hda_bus *bus, \
 				 struct hda_codec **codecp) \
 { \
 	unsigned long val; \
-	if (!strict_strtoul(buf, 0, &val)) \
+	if (!kstrtoul(buf, 0, &val)) \
 		(*codecp)->name = val; \
 }

--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@ -1160,7 +1160,7 @@ static int azx_reset(struct azx *chip, int full_reset)
 		goto __skip;

 	/* clear STATESTS */
-	azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+	azx_writew(chip, STATESTS, STATESTS_INT_MASK);

 	/* reset controller */
 	azx_enter_link_reset(chip);
@ -1242,7 +1242,7 @@ static void azx_int_clear(struct azx *chip)
 	}

 	/* clear STATESTS */
-	azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+	azx_writew(chip, STATESTS, STATESTS_INT_MASK);

 	/* clear rirb status */
 	azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
@ -1451,8 +1451,8 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)

 #if 0
 	/* clear state status int */
-	if (azx_readb(chip, STATESTS) & 0x04)
-		azx_writeb(chip, STATESTS, 0x04);
+	if (azx_readw(chip, STATESTS) & 0x04)
+		azx_writew(chip, STATESTS, 0x04);
 #endif
 	spin_unlock(&chip->reg_lock);
 	
@ -2971,6 +2971,10 @@ static int azx_runtime_suspend(struct device *dev)
 	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;

+	/* enable controller wake up event */
+	azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
+		  STATESTS_INT_MASK);
+
 	azx_stop_chip(chip);
 	azx_enter_link_reset(chip);
 	azx_clear_irq_pending(chip);
@ -2983,11 +2987,31 @@ static int azx_runtime_resume(struct device *dev)
 {
 	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;
+	struct hda_bus *bus;
+	struct hda_codec *codec;
+	int status;

 	if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
 		hda_display_power(true);
+
+	/* Read STATESTS before controller reset */
+	status = azx_readw(chip, STATESTS);
+
 	azx_init_pci(chip);
 	azx_init_chip(chip, 1);
+
+	bus = chip->bus;
+	if (status && bus) {
+		list_for_each_entry(codec, &bus->codec_list, list)
+			if (status & (1 << codec->addr))
+				queue_delayed_work(codec->bus->workq,
+						   &codec->jackpoll_work, codec->jackpoll_interval);
+	}
+
+	/* disable controller Wake Up event*/
+	azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
+			~STATESTS_INT_MASK);
+
 	return 0;
 }

@ -3831,11 +3855,13 @@ static int azx_probe_continue(struct azx *chip)

 	/* Request power well for Haswell HDA controller and codec */
 	if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) {
+#ifdef CONFIG_SND_HDA_I915
 		err = hda_i915_init();
 		if (err < 0) {
 			snd_printk(KERN_ERR SFX "Error request power-well from i915\n");
 			goto out_free;
 		}
+#endif
 		hda_display_power(true);
 	}

--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@ -194,18 +194,24 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid)
 EXPORT_SYMBOL_HDA(snd_hda_pin_sense);

 /**
- * snd_hda_jack_detect - query pin Presence Detect status
+ * snd_hda_jack_detect_state - query pin Presence Detect status
 * @codec: the CODEC to sense
 * @nid: the pin NID to sense
 *
- * Query and return the pin's Presence Detect status.
+ * Query and return the pin's Presence Detect status, as either
+ * HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT or HDA_JACK_PHANTOM.
 */
-int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
+int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid)
 {
-	u32 sense = snd_hda_pin_sense(codec, nid);
-	return get_jack_plug_state(sense);
+	struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid);
+	if (jack && jack->phantom_jack)
+		return HDA_JACK_PHANTOM;
+	else if (snd_hda_pin_sense(codec, nid) & AC_PINSENSE_PRESENCE)
+		return HDA_JACK_PRESENT;
+	else
+		return HDA_JACK_NOT_PRESENT;
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_detect);
+EXPORT_SYMBOL_HDA(snd_hda_jack_detect_state);

 /**
 * snd_hda_jack_detect_enable - enable the jack-detection
@ -247,8 +253,8 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable);
 int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid,
 				 hda_nid_t gating_nid)
 {
-	struct hda_jack_tbl *gated = snd_hda_jack_tbl_get(codec, gated_nid);
-	struct hda_jack_tbl *gating = snd_hda_jack_tbl_get(codec, gating_nid);
+	struct hda_jack_tbl *gated = snd_hda_jack_tbl_new(codec, gated_nid);
+	struct hda_jack_tbl *gating = snd_hda_jack_tbl_new(codec, gating_nid);

 	if (!gated || !gating)
 		return -EINVAL;
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@ -75,7 +75,18 @@ int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid,
 				 hda_nid_t gating_nid);

 u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid);
-int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid);
+
+/* the jack state returned from snd_hda_jack_detect_state() */
+enum {
+	HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT, HDA_JACK_PHANTOM,
+};
+
+int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid);
+
+static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
+{
+	return snd_hda_jack_detect_state(codec, nid) != HDA_JACK_NOT_PRESENT;
+}

 bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid);

--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@ -582,6 +582,36 @@ static void print_gpio(struct snd_info_buffer *buffer,
 	print_nid_array(buffer, codec, nid, &codec->nids);
 }

+static void print_device_list(struct snd_info_buffer *buffer,
+			    struct hda_codec *codec, hda_nid_t nid)
+{
+	int i, curr = -1;
+	u8 dev_list[AC_MAX_DEV_LIST_LEN];
+	int devlist_len;
+
+	devlist_len = snd_hda_get_devices(codec, nid, dev_list,
+					AC_MAX_DEV_LIST_LEN);
+	snd_iprintf(buffer, "  Devices: %d\n", devlist_len);
+	if (devlist_len <= 0)
+		return;
+
+	curr = snd_hda_codec_read(codec, nid, 0,
+				AC_VERB_GET_DEVICE_SEL, 0);
+
+	for (i = 0; i < devlist_len; i++) {
+		if (i == curr)
+			snd_iprintf(buffer, "    *");
+		else
+			snd_iprintf(buffer, "     ");
+
+		snd_iprintf(buffer,
+			"Dev %02d: PD = %d, ELDV = %d, IA = %d\n", i,
+			!!(dev_list[i] & AC_DE_PD),
+			!!(dev_list[i] & AC_DE_ELDV),
+			!!(dev_list[i] & AC_DE_IA));
+	}
+}
+
 static void print_codec_info(struct snd_info_entry *entry,
 			     struct snd_info_buffer *buffer)
 {
@ -751,6 +781,9 @@ static void print_codec_info(struct snd_info_entry *entry,
 				    (wid_caps & AC_WCAP_DELAY) >>
 				    AC_WCAP_DELAY_SHIFT);

+		if (wid_type == AC_WID_PIN && codec->dp_mst)
+			print_device_list(buffer, codec, nid);
+
 		if (wid_caps & AC_WCAP_CONN_LIST)
 			print_conn_list(buffer, codec, nid, wid_type,
 					conn, conn_len);
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@ -66,6 +66,8 @@ struct conexant_spec {
 	hda_nid_t eapds[4];
 	bool dynamic_eapd;

+	unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */
+
 #ifdef ENABLE_CXT_STATIC_QUIRKS
 	const struct snd_kcontrol_new *mixers[5];
 	int num_mixers;
@ -3200,6 +3202,9 @@ static int cx_auto_init(struct hda_codec *codec)
 	snd_hda_gen_init(codec);
 	if (!spec->dynamic_eapd)
 		cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
+
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT);
+
 	return 0;
 }

@ -3224,6 +3229,8 @@ enum {
 	CXT_PINCFG_LEMOTE_A1205,
 	CXT_FIXUP_STEREO_DMIC,
 	CXT_FIXUP_INC_MIC_BOOST,
+	CXT_FIXUP_HEADPHONE_MIC_PIN,
+	CXT_FIXUP_HEADPHONE_MIC,
 };

 static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
@ -3246,6 +3253,59 @@ static void cxt5066_increase_mic_boost(struct hda_codec *codec,
 				  (0 << AC_AMPCAP_MUTE_SHIFT));
 }

+static void cxt_update_headset_mode(struct hda_codec *codec)
+{
+	/* The verbs used in this function were tested on a Conexant CX20751/2 codec. */
+	int i;
+	bool mic_mode = false;
+	struct conexant_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+
+	hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]];
+
+	for (i = 0; i < cfg->num_inputs; i++)
+		if (cfg->inputs[i].pin == mux_pin) {
+			mic_mode = !!cfg->inputs[i].is_headphone_mic;
+			break;
+		}
+
+	if (mic_mode) {
+		snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x7c); /* enable merged mode for analog int-mic */
+		spec->gen.hp_jack_present = false;
+	} else {
+		snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x54); /* disable merged mode for analog int-mic */
+		spec->gen.hp_jack_present = snd_hda_jack_detect(codec, spec->gen.autocfg.hp_pins[0]);
+	}
+
+	snd_hda_gen_update_outputs(codec);
+}
+
+static void cxt_update_headset_mode_hook(struct hda_codec *codec,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	cxt_update_headset_mode(codec);
+}
+
+static void cxt_fixup_headphone_mic(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	struct conexant_spec *spec = codec->spec;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		spec->parse_flags |= HDA_PINCFG_HEADPHONE_MIC;
+		break;
+	case HDA_FIXUP_ACT_PROBE:
+		spec->gen.cap_sync_hook = cxt_update_headset_mode_hook;
+		spec->gen.automute_hook = cxt_update_headset_mode;
+		break;
+	case HDA_FIXUP_ACT_INIT:
+		cxt_update_headset_mode(codec);
+		break;
+	}
+}
+
+
 /* ThinkPad X200 & co with cxt5051 */
 static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = {
 	{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
@ -3302,6 +3362,19 @@ static const struct hda_fixup cxt_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cxt5066_increase_mic_boost,
 	},
+	[CXT_FIXUP_HEADPHONE_MIC_PIN] = {
+		.type = HDA_FIXUP_PINS,
+		.chained = true,
+		.chain_id = CXT_FIXUP_HEADPHONE_MIC,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, 0x03a1913d }, /* use as headphone mic, without its own jack detect */
+			{ }
+		}
+	},
+	[CXT_FIXUP_HEADPHONE_MIC] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = cxt_fixup_headphone_mic,
+	},
 };

 static const struct snd_pci_quirk cxt5051_fixups[] = {
@ -3311,6 +3384,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = {

 static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
@ -3395,7 +3469,8 @@ static int patch_conexant_auto(struct hda_codec *codec)

 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);

-	err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0);
+	err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL,
+				       spec->parse_flags);
 	if (err < 0)
 		goto error;

@ -3416,6 +3491,8 @@ static int patch_conexant_auto(struct hda_codec *codec)
 		codec->bus->allow_bus_reset = 1;
 	}

+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
+
 	return 0;

 error:
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@ -959,6 +959,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	int pin_nid;
 	int pin_idx;
 	struct hda_jack_tbl *jack;
+	int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT;

 	jack = snd_hda_jack_tbl_get_from_tag(codec, tag);
 	if (!jack)
@ -967,8 +968,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	jack->jack_dirty = 1;

 	_snd_printd(SND_PR_VERBOSE,
-		"HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
-		codec->addr, pin_nid,
+		"HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n",
+		codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA),
 		!!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV));

 	pin_idx = pin_nid_to_pin_index(spec, pin_nid);
@ -1992,8 +1993,10 @@ static int patch_generic_hdmi(struct hda_codec *codec)
 		return -EINVAL;
 	}
 	codec->patch_ops = generic_hdmi_patch_ops;
-	if (codec->vendor_id == 0x80862807)
+	if (codec->vendor_id == 0x80862807) {
 		codec->patch_ops.set_power_state = haswell_set_power_state;
+		codec->dp_mst = true;
+	}

 	generic_hdmi_init_per_pins(codec);

--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@ -282,6 +282,7 @@ static void alc_eapd_shutup(struct hda_codec *codec)
 {
 	alc_auto_setup_eapd(codec, false);
 	msleep(200);
+	snd_hda_shutup_pins(codec);
 }

 /* generic EAPD initialization */
@ -826,7 +827,8 @@ static inline void alc_shutup(struct hda_codec *codec)

 	if (spec && spec->shutup)
 		spec->shutup(codec);
-	snd_hda_shutup_pins(codec);
+	else
+		snd_hda_shutup_pins(codec);
 }

 #define alc_free	snd_hda_gen_free
@ -1853,8 +1855,10 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec,
 				       const struct hda_fixup *fix, int action)
 {
 	struct alc_spec *spec = codec->spec;
-	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->gen.no_primary_hp = 1;
+		spec->gen.no_multi_io = 1;
+	}
 }

 static const struct hda_fixup alc882_fixups[] = {
@ -2533,6 +2537,7 @@ enum {
 	ALC269_TYPE_ALC269VD,
 	ALC269_TYPE_ALC280,
 	ALC269_TYPE_ALC282,
+	ALC269_TYPE_ALC283,
 	ALC269_TYPE_ALC284,
 	ALC269_TYPE_ALC286,
 };
@ -2558,6 +2563,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	case ALC269_TYPE_ALC269VB:
 	case ALC269_TYPE_ALC269VD:
 	case ALC269_TYPE_ALC282:
+	case ALC269_TYPE_ALC283:
 	case ALC269_TYPE_ALC286:
 		ssids = alc269_ssids;
 		break;
@ -2583,15 +2589,81 @@ static void alc269_shutup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;

-	if (spec->codec_variant != ALC269_TYPE_ALC269VB)
-		return;
-
 	if (spec->codec_variant == ALC269_TYPE_ALC269VB)
 		alc269vb_toggle_power_output(codec, 0);
 	if (spec->codec_variant == ALC269_TYPE_ALC269VB &&
 			(alc_get_coef0(codec) & 0x00ff) == 0x018) {
 		msleep(150);
 	}
+	snd_hda_shutup_pins(codec);
+}
+
+static void alc283_init(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0];
+	bool hp_pin_sense;
+	int val;
+
+	if (!hp_pin)
+		return;
+	hp_pin_sense = snd_hda_jack_detect(codec, hp_pin);
+
+	/* Index 0x43 Direct Drive HP AMP LPM Control 1 */
+	/* Headphone capless set to high power mode */
+	alc_write_coef_idx(codec, 0x43, 0x9004);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+	if (hp_pin_sense)
+		msleep(85);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+
+	if (hp_pin_sense)
+		msleep(85);
+	/* Index 0x46 Combo jack auto switch control 2 */
+	/* 3k pull low control for Headset jack. */
+	val = alc_read_coef_idx(codec, 0x46);
+	alc_write_coef_idx(codec, 0x46, val & ~(3 << 12));
+	/* Headphone capless set to normal mode */
+	alc_write_coef_idx(codec, 0x43, 0x9614);
+}
+
+static void alc283_shutup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0];
+	bool hp_pin_sense;
+	int val;
+
+	if (!hp_pin) {
+		alc269_shutup(codec);
+		return;
+	}
+
+	hp_pin_sense = snd_hda_jack_detect(codec, hp_pin);
+
+	alc_write_coef_idx(codec, 0x43, 0x9004);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+	if (hp_pin_sense)
+		msleep(85);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0);
+
+	val = alc_read_coef_idx(codec, 0x46);
+	alc_write_coef_idx(codec, 0x46, val | (3 << 12));
+
+	if (hp_pin_sense)
+		msleep(85);
+	snd_hda_shutup_pins(codec);
+	alc_write_coef_idx(codec, 0x43, 0x9614);
 }

 static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg,
@ -2722,6 +2794,7 @@ static int alc269_resume(struct hda_codec *codec)
 	hda_call_check_power_status(codec, 0x01);
 	if (spec->has_alc5505_dsp)
 		alc5505_dsp_resume(codec);
+
 	return 0;
 }
 #endif /* CONFIG_PM */
@ -3261,6 +3334,28 @@ static void alc_fixup_headset_mode_alc668(struct hda_codec *codec,
 	alc_fixup_headset_mode(codec, fix, action);
 }

+/* Returns the nid of the external mic input pin, or 0 if it cannot be found. */
+static int find_ext_mic_pin(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+	hda_nid_t nid;
+	unsigned int defcfg;
+	int i;
+
+	for (i = 0; i < cfg->num_inputs; i++) {
+		if (cfg->inputs[i].type != AUTO_PIN_MIC)
+			continue;
+		nid = cfg->inputs[i].pin;
+		defcfg = snd_hda_codec_get_pincfg(codec, nid);
+		if (snd_hda_get_input_pin_attr(defcfg) == INPUT_PIN_ATTR_INT)
+			continue;
+		return nid;
+	}
+
+	return 0;
+}
+
 static void alc271_hp_gate_mic_jack(struct hda_codec *codec,
 				    const struct hda_fixup *fix,
 				    int action)
@ -3268,11 +3363,12 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec,
 	struct alc_spec *spec = codec->spec;

 	if (action == HDA_FIXUP_ACT_PROBE) {
-		if (snd_BUG_ON(!spec->gen.am_entry[1].pin ||
-			       !spec->gen.autocfg.hp_pins[0]))
+		int mic_pin = find_ext_mic_pin(codec);
+		int hp_pin = spec->gen.autocfg.hp_pins[0];
+
+		if (snd_BUG_ON(!mic_pin || !hp_pin))
 			return;
-		snd_hda_jack_set_gating_jack(codec, spec->gen.am_entry[1].pin,
-					     spec->gen.autocfg.hp_pins[0]);
+		snd_hda_jack_set_gating_jack(codec, mic_pin, hp_pin);
 	}
 }

@ -3308,6 +3404,45 @@ static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec,
 	}
 }

+static void alc283_hp_automute_hook(struct hda_codec *codec,
+				    struct hda_jack_tbl *jack)
+{
+	struct alc_spec *spec = codec->spec;
+	int vref;
+
+	msleep(200);
+	snd_hda_gen_hp_automute(codec, jack);
+
+	vref = spec->gen.hp_jack_present ? PIN_VREF80 : 0;
+
+	msleep(600);
+	snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			    vref);
+}
+
+static void alc283_chromebook_caps(struct hda_codec *codec)
+{
+	snd_hda_override_wcaps(codec, 0x03, 0);
+}
+
+static void alc283_fixup_chromebook(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+	int val;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		alc283_chromebook_caps(codec);
+		spec->gen.hp_automute_hook = alc283_hp_automute_hook;
+		/* MIC2-VREF control */
+		/* Set to manual mode */
+		val = alc_read_coef_idx(codec, 0x06);
+		alc_write_coef_idx(codec, 0x06, val & ~0x000c);
+		break;
+	}
+}
+
 enum {
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
@ -3344,6 +3479,7 @@ enum {
 	ALC269_FIXUP_ACER_AC700,
 	ALC269_FIXUP_LIMIT_INT_MIC_BOOST,
 	ALC269VB_FIXUP_ORDISSIMO_EVE2,
+	ALC283_FIXUP_CHROME_BOOK,
 };

 static const struct hda_fixup alc269_fixups[] = {
@ -3595,11 +3731,20 @@ static const struct hda_fixup alc269_fixups[] = {
 			{ }
 		},
 	},
+	[ALC283_FIXUP_CHROME_BOOK] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc283_fixup_chromebook,
+	},
 };

 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC),
+	SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),
+	SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
+	SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
+	SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
+	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
 	SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05c4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
@ -3637,6 +3782,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+	SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK),
 	SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@ -3655,11 +3801,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
-	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
-	SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),
-	SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
-	SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
-	SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
 	SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
@ -3670,8 +3811,16 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
-	SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
 	SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
@ -3840,11 +3989,15 @@ static int patch_alc269(struct hda_codec *codec)
 	case 0x10ec0290:
 		spec->codec_variant = ALC269_TYPE_ALC280;
 		break;
-	case 0x10ec0233:
 	case 0x10ec0282:
-	case 0x10ec0283:
 		spec->codec_variant = ALC269_TYPE_ALC282;
 		break;
+	case 0x10ec0233:
+	case 0x10ec0283:
+		spec->codec_variant = ALC269_TYPE_ALC283;
+		spec->shutup = alc283_shutup;
+		spec->init_hook = alc283_init;
+		break;
 	case 0x10ec0284:
 	case 0x10ec0292:
 		spec->codec_variant = ALC269_TYPE_ALC284;
@ -3872,7 +4025,8 @@ static int patch_alc269(struct hda_codec *codec)
 	codec->patch_ops.suspend = alc269_suspend;
 	codec->patch_ops.resume = alc269_resume;
 #endif
-	spec->shutup = alc269_shutup;
+	if (!spec->shutup)
+		spec->shutup = alc269_shutup;

 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);

--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@ -158,6 +158,7 @@ enum {
 	STAC_D965_VERBS,
 	STAC_DELL_3ST,
 	STAC_DELL_BIOS,
+	STAC_DELL_BIOS_AMIC,
 	STAC_DELL_BIOS_SPDIF,
 	STAC_927X_DELL_DMIC,
 	STAC_927X_VOLKNOB,
@ -3231,8 +3232,6 @@ static const struct hda_fixup stac927x_fixups[] = {
 	[STAC_DELL_BIOS] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
-			/* configure the analog microphone on some laptops */
-			{ 0x0c, 0x90a79130 },
 			/* correct the front output jack as a hp out */
 			{ 0x0f, 0x0221101f },
 			/* correct the front input jack as a mic */
@ -3242,6 +3241,16 @@ static const struct hda_fixup stac927x_fixups[] = {
 		.chained = true,
 		.chain_id = STAC_927X_DELL_DMIC,
 	},
+	[STAC_DELL_BIOS_AMIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			/* configure the analog microphone on some laptops */
+			{ 0x0c, 0x90a79130 },
+			{}
+		},
+		.chained = true,
+		.chain_id = STAC_DELL_BIOS,
+	},
 	[STAC_DELL_BIOS_SPDIF] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@ -3270,6 +3279,7 @@ static const struct hda_model_fixup stac927x_models[] = {
 	{ .id = STAC_D965_5ST_NO_FP, .name = "5stack-no-fp" },
 	{ .id = STAC_DELL_3ST, .name = "dell-3stack" },
 	{ .id = STAC_DELL_BIOS, .name = "dell-bios" },
+	{ .id = STAC_DELL_BIOS_AMIC, .name = "dell-bios-amic" },
 	{ .id = STAC_927X_VOLKNOB, .name = "volknob" },
 	{}
 };
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@ -207,9 +207,9 @@ static void vt1708_stop_hp_work(struct hda_codec *codec)
 		return;
 	if (spec->hp_work_active) {
 		snd_hda_codec_write(codec, 0x1, 0, 0xf81, 1);
+		codec->jackpoll_interval = 0;
 		cancel_delayed_work_sync(&codec->jackpoll_work);
 		spec->hp_work_active = false;
-		codec->jackpoll_interval = 0;
 	}
 }

--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@ -28,6 +28,7 @@
 #include <linux/interrupt.h>
 #include <linux/pci.h>
 #include <linux/module.h>
+#include <linux/vmalloc.h>

 #include <sound/core.h>
 #include <sound/info.h>
@ -198,6 +199,31 @@ MODULE_PARM_DESC(enable, "Enable RME Digi96 soundcard.");
 #define RME96_AD1852_VOL_BITS 14
 #define RME96_AD1855_VOL_BITS 10

+/* Defines for snd_rme96_trigger */
+#define RME96_TB_START_PLAYBACK 1
+#define RME96_TB_START_CAPTURE 2
+#define RME96_TB_STOP_PLAYBACK 4
+#define RME96_TB_STOP_CAPTURE 8
+#define RME96_TB_RESET_PLAYPOS 16
+#define RME96_TB_RESET_CAPTUREPOS 32
+#define RME96_TB_CLEAR_PLAYBACK_IRQ 64
+#define RME96_TB_CLEAR_CAPTURE_IRQ 128
+#define RME96_RESUME_PLAYBACK	(RME96_TB_START_PLAYBACK)
+#define RME96_RESUME_CAPTURE	(RME96_TB_START_CAPTURE)
+#define RME96_RESUME_BOTH	(RME96_RESUME_PLAYBACK \
+				| RME96_RESUME_CAPTURE)
+#define RME96_START_PLAYBACK	(RME96_TB_START_PLAYBACK \
+				| RME96_TB_RESET_PLAYPOS)
+#define RME96_START_CAPTURE	(RME96_TB_START_CAPTURE \
+				| RME96_TB_RESET_CAPTUREPOS)
+#define RME96_START_BOTH	(RME96_START_PLAYBACK \
+				| RME96_START_CAPTURE)
+#define RME96_STOP_PLAYBACK	(RME96_TB_STOP_PLAYBACK \
+				| RME96_TB_CLEAR_PLAYBACK_IRQ)
+#define RME96_STOP_CAPTURE	(RME96_TB_STOP_CAPTURE \
+				| RME96_TB_CLEAR_CAPTURE_IRQ)
+#define RME96_STOP_BOTH		(RME96_STOP_PLAYBACK \
+				| RME96_STOP_CAPTURE)

 struct rme96 {
 	spinlock_t    lock;
@ -214,6 +240,13 @@ struct rme96 {

 	u8 rev; /* card revision number */

+#ifdef CONFIG_PM
+	u32 playback_pointer;
+	u32 capture_pointer;
+	void *playback_suspend_buffer;
+	void *capture_suspend_buffer;
+#endif
+
 	struct snd_pcm_substream *playback_substream;
 	struct snd_pcm_substream *capture_substream;

@ -344,6 +377,8 @@ static struct snd_pcm_hardware snd_rme96_playback_spdif_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@ -373,6 +408,8 @@ static struct snd_pcm_hardware snd_rme96_capture_spdif_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@ -402,6 +439,8 @@ static struct snd_pcm_hardware snd_rme96_playback_adat_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@ -427,6 +466,8 @@ static struct snd_pcm_hardware snd_rme96_capture_adat_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@ -1045,54 +1086,35 @@ snd_rme96_capture_hw_params(struct snd_pcm_substream *substream,
 }

 static void
-snd_rme96_playback_start(struct rme96 *rme96,
-			 int from_pause)
+snd_rme96_trigger(struct rme96 *rme96,
+		  int op)
 {
-	if (!from_pause) {
+	if (op & RME96_TB_RESET_PLAYPOS)
 		writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS);
-	}
-
-	rme96->wcreg |= RME96_WCR_START;
-	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
-
-static void
-snd_rme96_capture_start(struct rme96 *rme96,
-			int from_pause)
-{
-	if (!from_pause) {
+	if (op & RME96_TB_RESET_CAPTUREPOS)
 		writel(0, rme96->iobase + RME96_IO_RESET_REC_POS);
+	if (op & RME96_TB_CLEAR_PLAYBACK_IRQ) {
+		rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+		if (rme96->rcreg & RME96_RCR_IRQ)
+			writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ);
 	}
-
-	rme96->wcreg |= RME96_WCR_START_2;
+	if (op & RME96_TB_CLEAR_CAPTURE_IRQ) {
+		rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+		if (rme96->rcreg & RME96_RCR_IRQ_2)
+			writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ);
+	}
+	if (op & RME96_TB_START_PLAYBACK)
+		rme96->wcreg |= RME96_WCR_START;
+	if (op & RME96_TB_STOP_PLAYBACK)
+		rme96->wcreg &= ~RME96_WCR_START;
+	if (op & RME96_TB_START_CAPTURE)
+		rme96->wcreg |= RME96_WCR_START_2;
+	if (op & RME96_TB_STOP_CAPTURE)
+		rme96->wcreg &= ~RME96_WCR_START_2;
 	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
 }

-static void
-snd_rme96_playback_stop(struct rme96 *rme96)
-{
-	/*
-	 * Check if there is an unconfirmed IRQ, if so confirm it, or else
-	 * the hardware will not stop generating interrupts
-	 */
-	rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
-	if (rme96->rcreg & RME96_RCR_IRQ) {
-		writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ);
-	}	
-	rme96->wcreg &= ~RME96_WCR_START;
-	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}

-static void
-snd_rme96_capture_stop(struct rme96 *rme96)
-{
-	rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
-	if (rme96->rcreg & RME96_RCR_IRQ_2) {
-		writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ);
-	}	
-	rme96->wcreg &= ~RME96_WCR_START_2;
-	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}

 static irqreturn_t
 snd_rme96_interrupt(int irq,
@ -1155,6 +1177,7 @@ snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;

+	snd_pcm_set_sync(substream);
 	spin_lock_irq(&rme96->lock);	
        if (rme96->playback_substream != NULL) {
 		spin_unlock_irq(&rme96->lock);
@ -1191,6 +1214,7 @@ snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;

+	snd_pcm_set_sync(substream);
 	runtime->hw = snd_rme96_capture_spdif_info;
        if (snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG &&
            (rate = snd_rme96_capture_getrate(rme96, &isadat)) > 0)
@ -1222,6 +1246,7 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;        
 	
+	snd_pcm_set_sync(substream);
 	spin_lock_irq(&rme96->lock);	
        if (rme96->playback_substream != NULL) {
 		spin_unlock_irq(&rme96->lock);
@ -1253,6 +1278,7 @@ snd_rme96_capture_adat_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;

+	snd_pcm_set_sync(substream);
 	runtime->hw = snd_rme96_capture_adat_info;
        if (snd_rme96_getinputtype(rme96) == RME96_INPUT_ANALOG) {
                /* makes no sense to use analog input. Note that analog
@ -1288,7 +1314,7 @@ snd_rme96_playback_close(struct snd_pcm_substream *substream)

 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISPLAYING(rme96)) {
-		snd_rme96_playback_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK);
 	}
 	rme96->playback_substream = NULL;
 	rme96->playback_periodsize = 0;
@ -1309,7 +1335,7 @@ snd_rme96_capture_close(struct snd_pcm_substream *substream)
 	
 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISRECORDING(rme96)) {
-		snd_rme96_capture_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_CAPTURE);
 	}
 	rme96->capture_substream = NULL;
 	rme96->capture_periodsize = 0;
@ -1324,7 +1350,7 @@ snd_rme96_playback_prepare(struct snd_pcm_substream *substream)
 	
 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISPLAYING(rme96)) {
-		snd_rme96_playback_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK);
 	}
 	writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS);
 	spin_unlock_irq(&rme96->lock);
@ -1338,7 +1364,7 @@ snd_rme96_capture_prepare(struct snd_pcm_substream *substream)
 	
 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISRECORDING(rme96)) {
-		snd_rme96_capture_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_CAPTURE);
 	}
 	writel(0, rme96->iobase + RME96_IO_RESET_REC_POS);
 	spin_unlock_irq(&rme96->lock);
@ -1350,41 +1376,55 @@ snd_rme96_playback_trigger(struct snd_pcm_substream *substream,
 			   int cmd)
 {
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+	struct snd_pcm_substream *s;
+	bool sync;
+
+	snd_pcm_group_for_each_entry(s, substream) {
+		if (snd_pcm_substream_chip(s) == rme96)
+			snd_pcm_trigger_done(s, substream);
+	}
+
+	sync = (rme96->playback_substream && rme96->capture_substream) &&
+	       (rme96->playback_substream->group ==
+		rme96->capture_substream->group);

 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		if (!RME96_ISPLAYING(rme96)) {
-			if (substream != rme96->playback_substream) {
+			if (substream != rme96->playback_substream)
 				return -EBUSY;
-			}
-			snd_rme96_playback_start(rme96, 0);
+			snd_rme96_trigger(rme96, sync ? RME96_START_BOTH
+						 : RME96_START_PLAYBACK);
 		}
 		break;

+	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		if (RME96_ISPLAYING(rme96)) {
-			if (substream != rme96->playback_substream) {
+			if (substream != rme96->playback_substream)
 				return -EBUSY;
-			}
-			snd_rme96_playback_stop(rme96);
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 :  RME96_STOP_PLAYBACK);
 		}
 		break;

 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		if (RME96_ISPLAYING(rme96)) {
-			snd_rme96_playback_stop(rme96);
-		}
+		if (RME96_ISPLAYING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 : RME96_STOP_PLAYBACK);
 		break;

+	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (!RME96_ISPLAYING(rme96)) {
-			snd_rme96_playback_start(rme96, 1);
-		}
+		if (!RME96_ISPLAYING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH
+						 : RME96_RESUME_PLAYBACK);
 		break;
-		
+
 	default:
 		return -EINVAL;
 	}
+
 	return 0;
 }

@ -1393,38 +1433,51 @@ snd_rme96_capture_trigger(struct snd_pcm_substream *substream,
 			  int cmd)
 {
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+	struct snd_pcm_substream *s;
+	bool sync;
+
+	snd_pcm_group_for_each_entry(s, substream) {
+		if (snd_pcm_substream_chip(s) == rme96)
+			snd_pcm_trigger_done(s, substream);
+	}
+
+	sync = (rme96->playback_substream && rme96->capture_substream) &&
+	       (rme96->playback_substream->group ==
+		rme96->capture_substream->group);

 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		if (!RME96_ISRECORDING(rme96)) {
-			if (substream != rme96->capture_substream) {
+			if (substream != rme96->capture_substream)
 				return -EBUSY;
-			}
-			snd_rme96_capture_start(rme96, 0);
+			snd_rme96_trigger(rme96, sync ? RME96_START_BOTH
+						 : RME96_START_CAPTURE);
 		}
 		break;

+	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		if (RME96_ISRECORDING(rme96)) {
-			if (substream != rme96->capture_substream) {
+			if (substream != rme96->capture_substream)
 				return -EBUSY;
-			}
-			snd_rme96_capture_stop(rme96);
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 : RME96_STOP_CAPTURE);
 		}
 		break;

 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		if (RME96_ISRECORDING(rme96)) {
-			snd_rme96_capture_stop(rme96);
-		}
+		if (RME96_ISRECORDING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 : RME96_STOP_CAPTURE);
 		break;

+	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (!RME96_ISRECORDING(rme96)) {
-			snd_rme96_capture_start(rme96, 1);
-		}
+		if (!RME96_ISRECORDING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH
+						 : RME96_RESUME_CAPTURE);
 		break;
-		
+
 	default:
 		return -EINVAL;
 	}
@ -1505,8 +1558,7 @@ snd_rme96_free(void *private_data)
 	        return;
 	}
 	if (rme96->irq >= 0) {
-		snd_rme96_playback_stop(rme96);
-		snd_rme96_capture_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_BOTH);
 		rme96->areg &= ~RME96_AR_DAC_EN;
 		writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
 		free_irq(rme96->irq, (void *)rme96);
@ -1520,6 +1572,10 @@ snd_rme96_free(void *private_data)
 		pci_release_regions(rme96->pci);
 		rme96->port = 0;
 	}
+#ifdef CONFIG_PM
+	vfree(rme96->playback_suspend_buffer);
+	vfree(rme96->capture_suspend_buffer);
+#endif
 	pci_disable_device(rme96->pci);
 }

@ -1606,8 +1662,7 @@ snd_rme96_create(struct rme96 *rme96)
 	rme96->capture_periodsize = 0;
 	
 	/* make sure playback/capture is stopped, if by some reason active */
-	snd_rme96_playback_stop(rme96);
-	snd_rme96_capture_stop(rme96);
+	snd_rme96_trigger(rme96, RME96_STOP_BOTH);
 	
 	/* set default values in registers */
 	rme96->wcreg =
@ -2319,6 +2374,87 @@ snd_rme96_create_switches(struct snd_card *card,
 * Card initialisation
 */

+#ifdef CONFIG_PM
+
+static int
+snd_rme96_suspend(struct pci_dev *pci,
+		  pm_message_t state)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct rme96 *rme96 = card->private_data;
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+	snd_pcm_suspend(rme96->playback_substream);
+	snd_pcm_suspend(rme96->capture_substream);
+
+	/* save capture & playback pointers */
+	rme96->playback_pointer = readl(rme96->iobase + RME96_IO_GET_PLAY_POS)
+				  & RME96_RCR_AUDIO_ADDR_MASK;
+	rme96->capture_pointer = readl(rme96->iobase + RME96_IO_GET_REC_POS)
+				 & RME96_RCR_AUDIO_ADDR_MASK;
+
+	/* save playback and capture buffers */
+	memcpy_fromio(rme96->playback_suspend_buffer,
+		      rme96->iobase + RME96_IO_PLAY_BUFFER, RME96_BUFFER_SIZE);
+	memcpy_fromio(rme96->capture_suspend_buffer,
+		      rme96->iobase + RME96_IO_REC_BUFFER, RME96_BUFFER_SIZE);
+
+	/* disable the DAC  */
+	rme96->areg &= ~RME96_AR_DAC_EN;
+	writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+
+	pci_disable_device(pci);
+	pci_save_state(pci);
+
+	return 0;
+}
+
+static int
+snd_rme96_resume(struct pci_dev *pci)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct rme96 *rme96 = card->private_data;
+
+	pci_restore_state(pci);
+	if (pci_enable_device(pci) < 0) {
+		printk(KERN_ERR "rme96: pci_enable_device failed, disabling device\n");
+		snd_card_disconnect(card);
+		return -EIO;
+	}
+
+	/* reset playback and record buffer pointers */
+	writel(0, rme96->iobase + RME96_IO_SET_PLAY_POS
+		  + rme96->playback_pointer);
+	writel(0, rme96->iobase + RME96_IO_SET_REC_POS
+		  + rme96->capture_pointer);
+
+	/* restore playback and capture buffers */
+	memcpy_toio(rme96->iobase + RME96_IO_PLAY_BUFFER,
+		    rme96->playback_suspend_buffer, RME96_BUFFER_SIZE);
+	memcpy_toio(rme96->iobase + RME96_IO_REC_BUFFER,
+		    rme96->capture_suspend_buffer, RME96_BUFFER_SIZE);
+
+	/* reset the ADC */
+	writel(rme96->areg | RME96_AR_PD2,
+	       rme96->iobase + RME96_IO_ADDITIONAL_REG);
+	writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+
+	/* reset and enable DAC, restore analog volume */
+	snd_rme96_reset_dac(rme96);
+	rme96->areg |= RME96_AR_DAC_EN;
+	writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+	if (RME96_HAS_ANALOG_OUT(rme96)) {
+		usleep_range(3000, 10000);
+		snd_rme96_apply_dac_volume(rme96);
+	}
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+
+	return 0;
+}
+
+#endif
+
 static void snd_rme96_card_free(struct snd_card *card)
 {
 	snd_rme96_free(card->private_data);
@ -2355,6 +2491,23 @@ snd_rme96_probe(struct pci_dev *pci,
 		return err;
 	}
 	
+#ifdef CONFIG_PM
+	rme96->playback_suspend_buffer = vmalloc(RME96_BUFFER_SIZE);
+	if (!rme96->playback_suspend_buffer) {
+		snd_printk(KERN_ERR
+			   "Failed to allocate playback suspend buffer!\n");
+		snd_card_free(card);
+		return -ENOMEM;
+	}
+	rme96->capture_suspend_buffer = vmalloc(RME96_BUFFER_SIZE);
+	if (!rme96->capture_suspend_buffer) {
+		snd_printk(KERN_ERR
+			   "Failed to allocate capture suspend buffer!\n");
+		snd_card_free(card);
+		return -ENOMEM;
+	}
+#endif
+
 	strcpy(card->driver, "Digi96");
 	switch (rme96->pci->device) {
 	case PCI_DEVICE_ID_RME_DIGI96:
@ -2397,6 +2550,10 @@ static struct pci_driver rme96_driver = {
 	.id_table = snd_rme96_ids,
 	.probe = snd_rme96_probe,
 	.remove = snd_rme96_remove,
+#ifdef CONFIG_PM
+	.suspend = snd_rme96_suspend,
+	.resume = snd_rme96_resume,
+#endif
 };

 module_pci_driver(rme96_driver);
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
--- a/sound/soc/cirrus/ep93xx-i2s.c
+++ b/sound/soc/cirrus/ep93xx-i2s.c
@ -408,7 +408,6 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
 	return 0;

 fail_put_lrclk:
-	dev_set_drvdata(&pdev->dev, NULL);
 	clk_put(info->lrclk);
 fail_put_sclk:
 	clk_put(info->sclk);
@ -423,7 +422,6 @@ static int ep93xx_i2s_remove(struct platform_device *pdev)
 	struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev);

 	snd_soc_unregister_component(&pdev->dev);
-	dev_set_drvdata(&pdev->dev, NULL);
 	clk_put(info->lrclk);
 	clk_put(info->sclk);
 	clk_put(info->mclk);
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@ -50,20 +50,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"DMIC AIF", NULL, "DMic"},
 };

-static int dmic_probe(struct snd_soc_codec *codec)
-{
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
-				  ARRAY_SIZE(dmic_dapm_widgets));
-        snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-	snd_soc_dapm_new_widgets(dapm);
-
-	return 0;
-}
-
 static struct snd_soc_codec_driver soc_dmic = {
-	.probe	= dmic_probe,
+	.dapm_widgets = dmic_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets),
+	.dapm_routes = intercon,
+	.num_dapm_routes = ARRAY_SIZE(intercon),
 };

 static int dmic_dev_probe(struct platform_device *pdev)
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@ -50,8 +50,6 @@ static const struct regmap_range_cfg rt5640_ranges[] = {

 static struct reg_default init_list[] = {
 	{RT5640_PR_BASE + 0x3d,	0x3600},
-	{RT5640_PR_BASE + 0x1c,	0x0D21},
-	{RT5640_PR_BASE + 0x1b,	0x0000},
 	{RT5640_PR_BASE + 0x12,	0x0aa8},
 	{RT5640_PR_BASE + 0x14,	0x0aaa},
 	{RT5640_PR_BASE + 0x20,	0x6110},
@ -384,15 +382,11 @@ static const SOC_ENUM_SINGLE_DECL(

 static const struct snd_kcontrol_new rt5640_snd_controls[] = {
 	/* Speaker Output Volume */
-	SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL,
-		RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1),
 	SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL,
 		RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1),
 	SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL,
 		RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv),
 	/* Headphone Output Volume */
-	SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL,
-		RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1),
 	SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL,
 		RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1),
 	SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL,
@ -737,6 +731,22 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = {
 			RT5640_M_BST1_MM_SFT, 1, 1),
 };

+static const struct snd_kcontrol_new spk_l_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL,
+		RT5640_L_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new spk_r_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL,
+		RT5640_R_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new hp_l_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL,
+		RT5640_L_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new hp_r_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL,
+		RT5640_R_MUTE_SFT, 1, 1);
+
 /* Stereo ADC source */
 static const char * const rt5640_stereo_adc1_src[] = {
 	"DIG MIX", "ADC"
@ -868,33 +878,6 @@ static const SOC_ENUM_SINGLE_DECL(
 static const struct snd_kcontrol_new rt5640_sdi_mux =
 	SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum);

-static int spk_event(struct snd_soc_dapm_widget *w,
-	struct snd_kcontrol *kcontrol, int event)
-{
-	struct snd_soc_codec *codec = w->codec;
-	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
-
-	switch (event) {
-	case SND_SOC_DAPM_POST_PMU:
-		regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1,
-					0x0001, 0x0001);
-		regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c,
-					0xf000, 0xf000);
-		break;
-
-	case SND_SOC_DAPM_PRE_PMD:
-		regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c,
-					0xf000, 0x0000);
-		regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1,
-					0x0001, 0x0000);
-		break;
-
-	default:
-		return 0;
-	}
-	return 0;
-}
-
 static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@ -943,6 +926,117 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }

+void hp_amp_power_on(struct snd_soc_codec *codec)
+{
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	/* depop parameters */
+	regmap_update_bits(rt5640->regmap, RT5640_PR_BASE +
+		RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200);
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2,
+		RT5640_DEPOP_MASK, RT5640_DEPOP_MAN);
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1,
+		RT5640_HP_CP_MASK | RT5640_HP_SG_MASK | RT5640_HP_CB_MASK,
+		RT5640_HP_CP_PU | RT5640_HP_SG_DIS | RT5640_HP_CB_PU);
+	regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1,
+			   0x9f00);
+	/* headphone amp power on */
+	regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1,
+		RT5640_PWR_FV1 | RT5640_PWR_FV2, 0);
+	regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1,
+		RT5640_PWR_HA,
+		RT5640_PWR_HA);
+	usleep_range(10000, 15000);
+	regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1,
+		RT5640_PWR_FV1 | RT5640_PWR_FV2 ,
+		RT5640_PWR_FV1 | RT5640_PWR_FV2);
+}
+
+static void rt5640_pmu_depop(struct snd_soc_codec *codec)
+{
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2,
+		RT5640_DEPOP_MASK | RT5640_DIG_DP_MASK,
+		RT5640_DEPOP_AUTO | RT5640_DIG_DP_EN);
+	regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP,
+		RT5640_PM_HP_MASK, RT5640_PM_HP_HV);
+
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3,
+		RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | RT5640_CP_FQ3_MASK,
+		(RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) |
+		(RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) |
+		(RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT));
+
+	regmap_write(rt5640->regmap, RT5640_PR_BASE +
+		RT5640_MAMP_INT_REG2, 0x1c00);
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1,
+		RT5640_HP_CP_MASK | RT5640_HP_SG_MASK,
+		RT5640_HP_CP_PD | RT5640_HP_SG_EN);
+	regmap_update_bits(rt5640->regmap, RT5640_PR_BASE +
+		RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400);
+}
+
+static int rt5640_hp_event(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		rt5640_pmu_depop(codec);
+		rt5640->hp_mute = 0;
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		rt5640->hp_mute = 1;
+		usleep_range(70000, 75000);
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
+static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		hp_amp_power_on(codec);
+		break;
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
+static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		if (!rt5640->hp_mute)
+			usleep_range(80000, 85000);
+
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2,
 			RT5640_PWR_PLL_BIT, 0, NULL, 0),
@ -1132,15 +1226,28 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
 		rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)),
 	SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1,
 		RT5640_PWR_MA_BIT, 0, NULL, 0),
-	SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1,
-		SND_SOC_NOPM, 0, NULL, 0),
-	SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1,
+	SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM,
+		0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0,
+		rt5640_hp_event,
+		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1,
 		RT5640_PWR_HP_L_BIT, 0, NULL, 0),
-	SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1,
+	SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1,
 		RT5640_PWR_HP_R_BIT, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1,
-		SND_SOC_NOPM, 0, spk_event,
-		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		RT5640_PWR_CLS_D_BIT, 0, NULL, 0),
+
+	/* Output Switch */
+	SND_SOC_DAPM_SWITCH("Speaker L Playback", SND_SOC_NOPM, 0, 0,
+			&spk_l_enable_control),
+	SND_SOC_DAPM_SWITCH("Speaker R Playback", SND_SOC_NOPM, 0, 0,
+			&spk_r_enable_control),
+	SND_SOC_DAPM_SWITCH("HP L Playback", SND_SOC_NOPM, 0, 0,
+			&hp_l_enable_control),
+	SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0,
+			&hp_r_enable_control),
+	SND_SOC_DAPM_POST("HP Post", rt5640_hp_post_event),
 	/* Output Lines */
 	SND_SOC_DAPM_OUTPUT("SPOLP"),
 	SND_SOC_DAPM_OUTPUT("SPOLN"),
@ -1381,9 +1488,11 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
 	{"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"},
 	{"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"},
 	{"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"},
+	{"HPO MIX L", NULL, "HP L Amp"},
 	{"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"},
 	{"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"},
 	{"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"},
+	{"HPO MIX R", NULL, "HP R Amp"},

 	{"LOUT MIX", "DAC L1 Switch", "DAC L1"},
 	{"LOUT MIX", "DAC R1 Switch", "DAC R1"},
@ -1396,13 +1505,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
 	{"Mono MIX", "OUTVOL L Switch", "OUTVOL L"},
 	{"Mono MIX", "BST1 Switch", "BST1"},

-	{"HP L Amp", NULL, "HPO MIX L"},
-	{"HP R Amp", NULL, "HPO MIX R"},
+	{"HP Amp", NULL, "HPO MIX L"},
+	{"HP Amp", NULL, "HPO MIX R"},

-	{"SPOLP", NULL, "SPOL MIX"},
-	{"SPOLN", NULL, "SPOL MIX"},
-	{"SPORP", NULL, "SPOR MIX"},
-	{"SPORN", NULL, "SPOR MIX"},
+	{"Speaker L Playback", "Switch", "SPOL MIX"},
+	{"Speaker R Playback", "Switch", "SPOR MIX"},
+	{"SPOLP", NULL, "Speaker L Playback"},
+	{"SPOLN", NULL, "Speaker L Playback"},
+	{"SPORP", NULL, "Speaker R Playback"},
+	{"SPORN", NULL, "Speaker R Playback"},

 	{"SPOLP", NULL, "Improve SPK Amp Drv"},
 	{"SPOLN", NULL, "Improve SPK Amp Drv"},
@ -1412,8 +1523,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
 	{"HPOL", NULL, "Improve HP Amp Drv"},
 	{"HPOR", NULL, "Improve HP Amp Drv"},

-	{"HPOL", NULL, "HP L Amp"},
-	{"HPOR", NULL, "HP R Amp"},
+	{"HP L Playback", "Switch", "HP Amp"},
+	{"HP R Playback", "Switch", "HP Amp"},
+	{"HPOL", NULL, "HP L Playback"},
+	{"HPOR", NULL, "HP R Playback"},
 	{"LOUTL", NULL, "LOUT MIX"},
 	{"LOUTR", NULL, "LOUT MIX"},
 	{"MONOP", NULL, "Mono MIX"},
@ -1792,17 +1905,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec,
 				RT5640_PWR_BG | RT5640_PWR_VREF2,
 				RT5640_PWR_VREF1 | RT5640_PWR_MB |
 				RT5640_PWR_BG | RT5640_PWR_VREF2);
-			mdelay(10);
+			usleep_range(10000, 15000);
 			snd_soc_update_bits(codec, RT5640_PWR_ANLG1,
 				RT5640_PWR_FV1 | RT5640_PWR_FV2,
 				RT5640_PWR_FV1 | RT5640_PWR_FV2);
 			regcache_sync(rt5640->regmap);
 			snd_soc_update_bits(codec, RT5640_DUMMY1,
 						0x0301, 0x0301);
-			snd_soc_update_bits(codec, RT5640_DEPOP_M1,
-						0x001d, 0x0019);
-			snd_soc_update_bits(codec, RT5640_DEPOP_M2,
-						0x2000, 0x2000);
 			snd_soc_update_bits(codec, RT5640_MICBIAS,
 						0x0030, 0x0030);
 		}
@ -1846,8 +1955,6 @@ static int rt5640_probe(struct snd_soc_codec *codec)
 	rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF);

 	snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301);
-	snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019);
-	snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000);
 	snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030);
 	snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00);

@ -2069,6 +2176,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c,
 		regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4,
 					RT5640_IN_DF2, RT5640_IN_DF2);

+	rt5640->hp_mute = 1;
+
 	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640,
 			rt5640_dai, ARRAY_SIZE(rt5640_dai));
 	if (ret < 0)
--- a/sound/soc/codecs/rt5640.h
+++ b/sound/soc/codecs/rt5640.h
@ -145,6 +145,8 @@


 /* Index of Codec Private Register definition */
+#define RT5640_CHPUMP_INT_REG1			0x24
+#define RT5640_MAMP_INT_REG2			0x37
 #define RT5640_3D_SPK				0x63
 #define RT5640_WND_1				0x6c
 #define RT5640_WND_2				0x6d
@ -153,6 +155,7 @@
 #define RT5640_WND_5				0x70
 #define RT5640_WND_8				0x73
 #define RT5640_DIP_SPK_INF			0x75
+#define RT5640_HP_DCC_INT1			0x77
 #define RT5640_EQ_BW_LOP			0xa0
 #define RT5640_EQ_GN_LOP			0xa1
 #define RT5640_EQ_FC_BP1			0xa2
@ -1201,6 +1204,14 @@
 #define RT5640_CP_FQ2_SFT			4
 #define RT5640_CP_FQ3_MASK			(0x7)
 #define RT5640_CP_FQ3_SFT			0
+#define RT5640_CP_FQ_1_5_KHZ			0
+#define RT5640_CP_FQ_3_KHZ			1
+#define RT5640_CP_FQ_6_KHZ			2
+#define RT5640_CP_FQ_12_KHZ			3
+#define RT5640_CP_FQ_24_KHZ			4
+#define RT5640_CP_FQ_48_KHZ			5
+#define RT5640_CP_FQ_96_KHZ			6
+#define RT5640_CP_FQ_192_KHZ			7

 /* HPOUT charge pump (0x91) */
 #define RT5640_OSW_L_MASK			(0x1 << 11)
@ -2087,6 +2098,7 @@ struct rt5640_priv {
 	int pll_out;

 	int dmic_en;
+	bool hp_mute;
 };

 #endif
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@ -561,8 +561,9 @@ static int ssm2602_suspend(struct snd_soc_codec *codec)

 static int ssm2602_resume(struct snd_soc_codec *codec)
 {
-	snd_soc_cache_sync(codec);
+	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);

+	regcache_sync(ssm2602->regmap);
 	ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

 	return 0;
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@ -338,18 +338,6 @@ static inline int aic32x4_get_divs(int mclk, int rate)
 	return -EINVAL;
 }

-static int aic32x4_add_widgets(struct snd_soc_codec *codec)
-{
-	snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets,
-				  ARRAY_SIZE(aic32x4_dapm_widgets));
-
-	snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes,
-				ARRAY_SIZE(aic32x4_dapm_routes));
-
-	snd_soc_dapm_new_widgets(&codec->dapm);
-	return 0;
-}
-
 static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				  int clk_id, unsigned int freq, int dir)
 {
@ -683,9 +671,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
 	}

 	aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	snd_soc_add_codec_controls(codec, aic32x4_snd_controls,
-			     ARRAY_SIZE(aic32x4_snd_controls));
-	aic32x4_add_widgets(codec);

 	/*
 	 * Workaround: for an unknown reason, the ADC needs to be powered up
@ -714,6 +699,13 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
 	.suspend = aic32x4_suspend,
 	.resume = aic32x4_resume,
 	.set_bias_level = aic32x4_set_bias_level,
+
+	.controls = aic32x4_snd_controls,
+	.num_controls = ARRAY_SIZE(aic32x4_snd_controls),
+	.dapm_widgets = aic32x4_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets),
+	.dapm_routes = aic32x4_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes),
 };

 static int aic32x4_i2c_probe(struct i2c_client *i2c,
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@ -1202,7 +1202,6 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec)
 		break;
 	}

-	snd_soc_dapm_new_widgets(dapm);
 	return 0;
 }

--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@ -3174,7 +3174,7 @@ static ssize_t wm8962_beep_set(struct device *dev,
 	long int time;
 	int ret;

-	ret = strict_strtol(buf, 10, &time);
+	ret = kstrtol(buf, 10, &time);
 	if (ret != 0)
 		return ret;

--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@ -421,13 +421,11 @@ static int dw_i2s_probe(struct platform_device *pdev)
 					 dw_i2s_dai, 1);
 	if (ret != 0) {
 		dev_err(&pdev->dev, "not able to register dai\n");
-		goto err_set_drvdata;
+		goto err_clk_disable;
 	}

 	return 0;

-err_set_drvdata:
-	dev_set_drvdata(&pdev->dev, NULL);
 err_clk_disable:
 	clk_disable(dev->clk);
 err_clk_put:
@ -440,7 +438,6 @@ static int dw_i2s_remove(struct platform_device *pdev)
 	struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev);

 	snd_soc_unregister_component(&pdev->dev);
-	dev_set_drvdata(&pdev->dev, NULL);

 	clk_put(dev->clk);

--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@ -193,6 +193,17 @@ config SND_SOC_IMX_SGTL5000
 	  Say Y if you want to add support for SoC audio on an i.MX board with
 	  a sgtl5000 codec.

+config SND_SOC_IMX_SPDIF
+	tristate "SoC Audio support for i.MX boards with S/PDIF"
+	select SND_SOC_IMX_PCM_DMA
+	select SND_SOC_FSL_SPDIF
+	select SND_SOC_SPDIF
+	select REGMAP_MMIO
+	help
+	  SoC Audio support for i.MX boards with S/PDIF
+	  Say Y if you want to add support for SoC audio on an i.MX board with
+	  a S/DPDIF.
+
 config SND_SOC_IMX_MC13783
 	tristate "SoC Audio support for I.MX boards with mc13783"
 	depends on MFD_MC13783 && ARM
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@ -45,6 +45,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
 snd-soc-wm1133-ev1-objs := wm1133-ev1.o
 snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
 snd-soc-imx-wm8962-objs := imx-wm8962.o
+snd-soc-imx-spdif-objs := imx-spdif.o
 snd-soc-imx-mc13783-objs := imx-mc13783.o

 obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
@ -53,4 +54,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
 obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
 obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
 obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
+obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
 obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@ -411,8 +411,8 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
 	return 0;
 }

-int fsl_spdif_startup(struct snd_pcm_substream *substream,
-			struct snd_soc_dai *cpu_dai)
+static int fsl_spdif_startup(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *cpu_dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
@ -546,7 +546,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream,
 	return 0;
 }

-struct snd_soc_dai_ops fsl_spdif_dai_ops = {
+static struct snd_soc_dai_ops fsl_spdif_dai_ops = {
 	.startup = fsl_spdif_startup,
 	.hw_params = fsl_spdif_hw_params,
 	.trigger = fsl_spdif_trigger,
@ -555,7 +555,6 @@ struct snd_soc_dai_ops fsl_spdif_dai_ops = {


 /*
- * ============================================
 * FSL SPDIF IEC958 controller(mixer) functions
 *
 *	Channel status get/put control
@ -563,7 +562,6 @@ struct snd_soc_dai_ops fsl_spdif_dai_ops = {
 *	Valid bit value get control
 *	DPLL lock status get control
 *	User bit sync mode selection control
- * ============================================
 */

 static int fsl_spdif_info(struct snd_kcontrol *kcontrol,
@ -921,7 +919,7 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai)
 	return 0;
 }

-struct snd_soc_dai_driver fsl_spdif_dai = {
+static struct snd_soc_dai_driver fsl_spdif_dai = {
 	.probe = &fsl_spdif_dai_probe,
 	.playback = {
 		.channels_min = 2,
@ -942,11 +940,7 @@ static const struct snd_soc_component_driver fsl_spdif_component = {
 	.name		= "fsl-spdif",
 };

-/*
- * ================
- * FSL SPDIF REGMAP
- * ================
- */
+/* FSL SPDIF REGMAP */

 static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg)
 {
@ -1077,9 +1071,9 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
 			break;
 	}

-	dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate",
+	dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate\n",
 			spdif_priv->txclk_src[index], rate[index]);
-	dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate",
+	dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate\n",
 			spdif_priv->txclk_div[index], rate[index]);

 	return 0;
@ -1119,10 +1113,8 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	}

 	regs = devm_ioremap_resource(&pdev->dev, res);
-	if (IS_ERR(regs)) {
-		dev_err(&pdev->dev, "could not map device resources\n");
+	if (IS_ERR(regs))
 		return PTR_ERR(regs);
-	}

 	spdif_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev,
 			"core", regs, &fsl_spdif_regmap_config);
@ -1184,7 +1176,7 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 					 &spdif_priv->cpu_dai_drv, 1);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
-		goto error_dev;
+		return ret;
 	}

 	ret = imx_pcm_dma_init(pdev);
@ -1197,8 +1189,6 @@ static int fsl_spdif_probe(struct platform_device *pdev)

 error_component:
 	snd_soc_unregister_component(&pdev->dev);
-error_dev:
-	dev_set_drvdata(&pdev->dev, NULL);

 	return ret;
 }
@ -1207,7 +1197,6 @@ static int fsl_spdif_remove(struct platform_device *pdev)
 {
 	imx_pcm_dma_exit(pdev);
 	snd_soc_unregister_component(&pdev->dev);
-	dev_set_drvdata(&pdev->dev, NULL);

 	return 0;
 }
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@ -1114,7 +1114,6 @@ static int fsl_ssi_probe(struct platform_device *pdev)
 	snd_soc_unregister_component(&pdev->dev);

 error_dev:
-	dev_set_drvdata(&pdev->dev, NULL);
 	device_remove_file(&pdev->dev, dev_attr);

 error_clk:
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@ -335,7 +335,8 @@ static int imx_audmux_probe(struct platform_device *pdev)
 	if (audmux_type == IMX31_AUDMUX)
 		audmux_debugfs_init();

-	imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node);
+	if (of_id)
+		imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node);

 	return 0;
 }
--- a/sound/soc/fsl/imx-spdif.c
+++ b/sound/soc/fsl/imx-spdif.c
@ -0,0 +1,148 @@
+/*
+ * Copyright (C) 2013 Freescale Semiconductor, Inc.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+
+struct imx_spdif_data {
+	struct snd_soc_dai_link dai[2];
+	struct snd_soc_card card;
+	struct platform_device *txdev;
+	struct platform_device *rxdev;
+};
+
+static int imx_spdif_audio_probe(struct platform_device *pdev)
+{
+	struct device_node *spdif_np, *np = pdev->dev.of_node;
+	struct imx_spdif_data *data;
+	int ret = 0, num_links = 0;
+
+	spdif_np = of_parse_phandle(np, "spdif-controller", 0);
+	if (!spdif_np) {
+		dev_err(&pdev->dev, "failed to find spdif-controller\n");
+		ret = -EINVAL;
+		goto end;
+	}
+
+	data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+	if (!data) {
+		dev_err(&pdev->dev, "failed to allocate memory\n");
+		ret = -ENOMEM;
+		goto end;
+	}
+
+	if (of_property_read_bool(np, "spdif-out")) {
+		data->dai[num_links].name = "S/PDIF TX";
+		data->dai[num_links].stream_name = "S/PDIF PCM Playback";
+		data->dai[num_links].codec_dai_name = "dit-hifi";
+		data->dai[num_links].codec_name = "spdif-dit";
+		data->dai[num_links].cpu_of_node = spdif_np;
+		data->dai[num_links].platform_of_node = spdif_np;
+		num_links++;
+
+		data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0);
+		if (IS_ERR(data->txdev)) {
+			ret = PTR_ERR(data->txdev);
+			dev_err(&pdev->dev, "register dit failed: %d\n", ret);
+			goto end;
+		}
+	}
+
+	if (of_property_read_bool(np, "spdif-in")) {
+		data->dai[num_links].name = "S/PDIF RX";
+		data->dai[num_links].stream_name = "S/PDIF PCM Capture";
+		data->dai[num_links].codec_dai_name = "dir-hifi";
+		data->dai[num_links].codec_name = "spdif-dir";
+		data->dai[num_links].cpu_of_node = spdif_np;
+		data->dai[num_links].platform_of_node = spdif_np;
+		num_links++;
+
+		data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0);
+		if (IS_ERR(data->rxdev)) {
+			ret = PTR_ERR(data->rxdev);
+			dev_err(&pdev->dev, "register dir failed: %d\n", ret);
+			goto error_dit;
+		}
+	}
+
+	if (!num_links) {
+		dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n");
+		goto error_dir;
+	}
+
+	data->card.dev = &pdev->dev;
+	data->card.num_links = num_links;
+	data->card.dai_link = data->dai;
+
+	ret = snd_soc_of_parse_card_name(&data->card, "model");
+	if (ret)
+		goto error_dir;
+
+	ret = snd_soc_register_card(&data->card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret);
+		goto error_dir;
+	}
+
+	platform_set_drvdata(pdev, data);
+
+	goto end;
+
+error_dir:
+	if (data->rxdev)
+		platform_device_unregister(data->rxdev);
+error_dit:
+	if (data->txdev)
+		platform_device_unregister(data->txdev);
+end:
+	if (spdif_np)
+		of_node_put(spdif_np);
+
+	return ret;
+}
+
+static int imx_spdif_audio_remove(struct platform_device *pdev)
+{
+	struct imx_spdif_data *data = platform_get_drvdata(pdev);
+
+	if (data->rxdev)
+		platform_device_unregister(data->rxdev);
+	if (data->txdev)
+		platform_device_unregister(data->txdev);
+
+	snd_soc_unregister_card(&data->card);
+
+	return 0;
+}
+
+static const struct of_device_id imx_spdif_dt_ids[] = {
+	{ .compatible = "fsl,imx-audio-spdif", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids);
+
+static struct platform_driver imx_spdif_driver = {
+	.driver = {
+		.name = "imx-spdif",
+		.owner = THIS_MODULE,
+		.of_match_table = imx_spdif_dt_ids,
+	},
+	.probe = imx_spdif_audio_probe,
+	.remove = imx_spdif_audio_remove,
+};
+
+module_platform_driver(imx_spdif_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-spdif");
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@ -105,6 +105,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev)
 static struct platform_driver asoc_simple_card = {
 	.driver = {
 		.name	= "asoc-simple-card",
+		.owner = THIS_MODULE,
 	},
 	.probe		= asoc_simple_card_probe,
 	.remove		= asoc_simple_card_remove,
@ -112,6 +113,7 @@ static struct platform_driver asoc_simple_card = {

 module_platform_driver(asoc_simple_card);

+MODULE_ALIAS("platform:asoc-simple-card");
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("ASoC Simple Sound Card");
 MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@ -1,6 +1,6 @@
 config SND_KIRKWOOD_SOC
-	tristate "SoC Audio for the Marvell Kirkwood chip"
-	depends on ARCH_KIRKWOOD || COMPILE_TEST
+	tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
+	depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Kirkwood I2S interface. You will also need to select the
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@ -22,6 +22,8 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <linux/platform_data/asoc-kirkwood.h>
+#include <linux/of.h>
+
 #include "kirkwood.h"

 #define DRV_NAME	"mvebu-audio"
@ -453,6 +455,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 	struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai;
 	struct kirkwood_dma_data *priv;
 	struct resource *mem;
+	struct device_node *np = pdev->dev.of_node;
 	int err;

 	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
@ -473,14 +476,16 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 		return -ENXIO;
 	}

-	if (!data) {
-		dev_err(&pdev->dev, "no platform data ?!\n");
+	if (np) {
+		priv->burst = 128;		/* might be 32 or 128 */
+	} else if (data) {
+		priv->burst = data->burst;
+	} else {
+		dev_err(&pdev->dev, "no DT nor platform data ?!\n");
 		return -EINVAL;
 	}

-	priv->burst = data->burst;
-
-	priv->clk = devm_clk_get(&pdev->dev, NULL);
+	priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL);
 	if (IS_ERR(priv->clk)) {
 		dev_err(&pdev->dev, "no clock\n");
 		return PTR_ERR(priv->clk);
@ -507,7 +512,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 	priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24;

 	/* Select the burst size */
-	if (data->burst == 32) {
+	if (priv->burst == 32) {
 		priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32;
 		priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32;
 	} else {
@ -552,12 +557,21 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev)
 	return 0;
 }

+#ifdef CONFIG_OF
+static struct of_device_id mvebu_audio_of_match[] = {
+	{ .compatible = "marvell,mvebu-audio" },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, mvebu_audio_of_match);
+#endif
+
 static struct platform_driver kirkwood_i2s_driver = {
 	.probe  = kirkwood_i2s_dev_probe,
 	.remove = kirkwood_i2s_dev_remove,
 	.driver = {
 		.name = DRV_NAME,
 		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(mvebu_audio_of_match),
 	},
 };

--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@ -105,11 +105,13 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
 		.stream_name	= "HiFi Playback",
 		.codec_dai_name	= "sgtl5000",
 		.ops		= &mxs_sgtl5000_hifi_ops,
+		.playback_only	= true,
 	}, {
 		.name		= "HiFi Rx",
 		.stream_name	= "HiFi Capture",
 		.codec_dai_name	= "sgtl5000",
 		.ops		= &mxs_sgtl5000_hifi_ops,
+		.capture_only	= true,
 	},
 };

--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@ -781,7 +781,7 @@ static ssize_t prop##_store(struct device *dev,				\
 	unsigned long val;						\
 	int status;							\
 									\
-	status = strict_strtoul(buf, 0, &val);				\
+	status = kstrtoul(buf, 0, &val);				\
 	if (status)							\
 		return status;						\
 									\
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@ -90,6 +90,13 @@ static void dma_enqueue(struct snd_pcm_substream *substream)
 	dma_info.period = prtd->dma_period;
 	dma_info.len = prtd->dma_period*limit;

+	if (dma_info.cap == DMA_CYCLIC) {
+		dma_info.buf = pos;
+		prtd->params->ops->prepare(prtd->params->ch, &dma_info);
+		prtd->dma_loaded += limit;
+		return;
+	}
+
 	while (prtd->dma_loaded < limit) {
 		pr_debug("dma_loaded: %d\n", prtd->dma_loaded);

--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@ -235,6 +235,8 @@ struct fsi_stream {
 	struct sh_dmae_slave	slave; /* see fsi_handler_init() */
 	struct work_struct	work;
 	dma_addr_t		dma;
+	int			loop_cnt;
+	int			additional_pos;
 };

 struct fsi_clk {
@ -1289,6 +1291,8 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io)
 	io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
 			 BUSOP_SET(16, PACKAGE_16BITBUS_STREAM);

+	io->loop_cnt = 2; /* push 1st, 2nd period first, then 3rd, 4th... */
+	io->additional_pos = 0;
 	io->dma = dma_map_single(dai->dev, runtime->dma_area,
 				 snd_pcm_lib_buffer_bytes(io->substream), dir);
 	return 0;
@ -1305,11 +1309,15 @@ static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io)
 	return 0;
 }

-static dma_addr_t fsi_dma_get_area(struct fsi_stream *io)
+static dma_addr_t fsi_dma_get_area(struct fsi_stream *io, int additional)
 {
 	struct snd_pcm_runtime *runtime = io->substream->runtime;
+	int period = io->period_pos + additional;

-	return io->dma + samples_to_bytes(runtime, io->buff_sample_pos);
+	if (period >= runtime->periods)
+		period = 0;
+
+	return io->dma + samples_to_bytes(runtime, period * io->period_samples);
 }

 static void fsi_dma_complete(void *data)
@ -1321,7 +1329,7 @@ static void fsi_dma_complete(void *data)
 	enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ?
 		DMA_TO_DEVICE : DMA_FROM_DEVICE;

-	dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io),
+	dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io, 0),
 			samples_to_bytes(runtime, io->period_samples), dir);

 	io->buff_sample_pos += io->period_samples;
@ -1347,7 +1355,7 @@ static void fsi_dma_do_work(struct work_struct *work)
 	struct snd_pcm_runtime *runtime;
 	enum dma_data_direction dir;
 	int is_play = fsi_stream_is_play(fsi, io);
-	int len;
+	int len, i;
 	dma_addr_t buf;

 	if (!fsi_stream_is_working(fsi, io))
@ -1357,26 +1365,33 @@ static void fsi_dma_do_work(struct work_struct *work)
 	runtime	= io->substream->runtime;
 	dir	= is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
 	len	= samples_to_bytes(runtime, io->period_samples);
-	buf	= fsi_dma_get_area(io);

-	dma_sync_single_for_device(dai->dev, buf, len, dir);
+	for (i = 0; i < io->loop_cnt; i++) {
+		buf	= fsi_dma_get_area(io, io->additional_pos);

-	desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
-					   DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
-	if (!desc) {
-		dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
-		return;
+		dma_sync_single_for_device(dai->dev, buf, len, dir);
+
+		desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
+					DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+		if (!desc) {
+			dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
+			return;
+		}
+
+		desc->callback		= fsi_dma_complete;
+		desc->callback_param	= io;
+
+		if (dmaengine_submit(desc) < 0) {
+			dev_err(dai->dev, "tx_submit() fail\n");
+			return;
+		}
+
+		dma_async_issue_pending(io->chan);
+
+		io->additional_pos = 1;
 	}

-	desc->callback		= fsi_dma_complete;
-	desc->callback_param	= io;
-
-	if (dmaengine_submit(desc) < 0) {
-		dev_err(dai->dev, "tx_submit() fail\n");
-		return;
-	}
-
-	dma_async_issue_pending(io->chan);
+	io->loop_cnt = 1;

 	/*
 	 * FIXME
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@ -203,7 +203,7 @@ static ssize_t pmdown_time_set(struct device *dev,
 	struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
 	int ret;

-	ret = strict_strtol(buf, 10, &rtd->pmdown_time);
+	ret = kstrtol(buf, 10, &rtd->pmdown_time);
 	if (ret)
 		return ret;

@ -248,6 +248,7 @@ static ssize_t codec_reg_write_file(struct file *file,
 	char *start = buf;
 	unsigned long reg, value;
 	struct snd_soc_codec *codec = file->private_data;
+	int ret;

 	buf_size = min(count, (sizeof(buf)-1));
 	if (copy_from_user(buf, user_buf, buf_size))
@ -259,8 +260,9 @@ static ssize_t codec_reg_write_file(struct file *file,
 	reg = simple_strtoul(start, &start, 16);
 	while (*start == ' ')
 		start++;
-	if (strict_strtoul(start, 16, &value))
-		return -EINVAL;
+	ret = kstrtoul(start, 16, &value);
+	if (ret)
+		return ret;

 	/* Userspace has been fiddling around behind the kernel's back */
 	add_taint(TAINT_USER, LOCKDEP_NOW_UNRELIABLE);
@ -1243,9 +1245,6 @@ static int soc_post_component_init(struct snd_soc_card *card,
 	}
 	rtd->card = card;

-	/* Make sure all DAPM widgets are instantiated */
-	snd_soc_dapm_new_widgets(&codec->dapm);
-
 	/* machine controls, routes and widgets are not prefixed */
 	temp = codec->name_prefix;
 	codec->name_prefix = NULL;
@ -1741,8 +1740,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 		snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
 					card->num_dapm_routes);

-	snd_soc_dapm_new_widgets(&card->dapm);
-
 	for (i = 0; i < card->num_links; i++) {
 		dai_link = &card->dai_link[i];
 		dai_fmt = dai_link->dai_fmt;
@ -1821,12 +1818,12 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 		}
 	}

-	snd_soc_dapm_new_widgets(&card->dapm);
-
 	if (card->fully_routed)
 		list_for_each_entry(codec, &card->codec_dev_list, card_list)
 			snd_soc_dapm_auto_nc_codec_pins(codec);

+	snd_soc_dapm_new_widgets(card);
+
 	ret = snd_card_register(card->snd_card);
 	if (ret < 0) {
 		dev_err(card->dev, "ASoC: failed to register soundcard %d\n",
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@ -229,6 +229,8 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 			template.id = snd_soc_dapm_kcontrol;
 			template.name = kcontrol->id.name;

+			data->value = template.on_val;
+
 			data->widget = snd_soc_dapm_new_control(widget->dapm,
 				&template);
 			if (!data->widget) {
@ -2374,6 +2376,9 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
 			wsource->ext = 1;
 	}

+	dapm_mark_dirty(wsource, "Route added");
+	dapm_mark_dirty(wsink, "Route added");
+
 	/* connect static paths */
 	if (control == NULL) {
 		list_add(&path->list, &dapm->card->paths);
@ -2436,9 +2441,6 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
 		return 0;
 	}

-	dapm_mark_dirty(wsource, "Route added");
-	dapm_mark_dirty(wsink, "Route added");
-
 	return 0;
 err:
 	kfree(path);
@ -2712,9 +2714,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes);
 *
 * Returns 0 for success.
 */
-int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
+int snd_soc_dapm_new_widgets(struct snd_soc_card *card)
 {
-	struct snd_soc_card *card = dapm->card;
 	struct snd_soc_dapm_widget *w;
 	unsigned int val;

--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@ -183,8 +183,6 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
 		list_add(&(pins[i].list), &jack->pins);
 	}

-	snd_soc_dapm_new_widgets(&jack->codec->card->dapm);
-
 	/* Update to reflect the last reported status; canned jack
 	 * implementations are likely to set their state before the
 	 * card has an opportunity to associate pins.
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@ -2020,6 +2020,16 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 			capture = 1;
 	}

+	if (rtd->dai_link->playback_only) {
+		playback = 1;
+		capture = 0;
+	}
+
+	if (rtd->dai_link->capture_only) {
+		playback = 0;
+		capture = 1;
+	}
+
 	/* create the PCM */
 	if (rtd->dai_link->no_pcm) {
 		snprintf(new_name, sizeof(new_name), "(%s)",
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@ -346,10 +346,10 @@ static int usb6fire_fw_check(u8 *version)
 		if (!memcmp(version, known_fw_versions + i, 2))
 			return 0;

-	snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. "
+	snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %4ph. "
 			"please reconnect to power. if this failure "
 			"still happens, check your firmware installation.",
-			4, version);
+			version);
 	return -EINVAL;
 }

--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@ -418,6 +418,9 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
 	struct snd_usb_endpoint *ep;
 	int is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK;

+	if (WARN_ON(!alts))
+		return NULL;
+
 	mutex_lock(&chip->mutex);

 	list_for_each_entry(ep, &chip->ep_list, list) {
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@ -327,6 +327,137 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum,
 	return 0;
 }

+static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
+					 struct usb_device *dev,
+					 struct usb_interface_descriptor *altsd,
+					 unsigned int attr)
+{
+	struct usb_host_interface *alts;
+	struct usb_interface *iface;
+	unsigned int ep;
+
+	/* Implicit feedback sync EPs consumers are always playback EPs */
+	if (subs->direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
+	switch (subs->stream->chip->usb_id) {
+	case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
+	case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
+		ep = 0x81;
+		iface = usb_ifnum_to_if(dev, 3);
+
+		if (!iface || iface->num_altsetting == 0)
+			return -EINVAL;
+
+		alts = &iface->altsetting[1];
+		goto add_sync_ep;
+		break;
+	case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
+	case USB_ID(0x0763, 0x2081):
+		ep = 0x81;
+		iface = usb_ifnum_to_if(dev, 2);
+
+		if (!iface || iface->num_altsetting == 0)
+			return -EINVAL;
+
+		alts = &iface->altsetting[1];
+		goto add_sync_ep;
+	}
+	if (attr == USB_ENDPOINT_SYNC_ASYNC &&
+	    altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
+	    altsd->bInterfaceProtocol == 2 &&
+	    altsd->bNumEndpoints == 1 &&
+	    USB_ID_VENDOR(subs->stream->chip->usb_id) == 0x0582 /* Roland */ &&
+	    search_roland_implicit_fb(dev, altsd->bInterfaceNumber + 1,
+				      altsd->bAlternateSetting,
+				      &alts, &ep) >= 0) {
+		goto add_sync_ep;
+	}
+
+	/* No quirk */
+	return 0;
+
+add_sync_ep:
+	subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
+						   alts, ep, !subs->direction,
+						   SND_USB_ENDPOINT_TYPE_DATA);
+	if (!subs->sync_endpoint)
+		return -EINVAL;
+
+	subs->data_endpoint->sync_master = subs->sync_endpoint;
+
+	return 0;
+}
+
+static int set_sync_endpoint(struct snd_usb_substream *subs,
+			     struct audioformat *fmt,
+			     struct usb_device *dev,
+			     struct usb_host_interface *alts,
+			     struct usb_interface_descriptor *altsd)
+{
+	int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
+	unsigned int ep, attr;
+	bool implicit_fb;
+	int err;
+
+	/* we need a sync pipe in async OUT or adaptive IN mode */
+	/* check the number of EP, since some devices have broken
+	 * descriptors which fool us.  if it has only one EP,
+	 * assume it as adaptive-out or sync-in.
+	 */
+	attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE;
+
+	err = set_sync_ep_implicit_fb_quirk(subs, dev, altsd, attr);
+	if (err < 0)
+		return err;
+
+	if (altsd->bNumEndpoints < 2)
+		return 0;
+
+	if ((is_playback && attr != USB_ENDPOINT_SYNC_ASYNC) ||
+	    (!is_playback && attr != USB_ENDPOINT_SYNC_ADAPTIVE))
+		return 0;
+
+	/* check sync-pipe endpoint */
+	/* ... and check descriptor size before accessing bSynchAddress
+	   because there is a version of the SB Audigy 2 NX firmware lacking
+	   the audio fields in the endpoint descriptors */
+	if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC ||
+	    (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+	     get_endpoint(alts, 1)->bSynchAddress != 0)) {
+		snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n",
+			   dev->devnum, fmt->iface, fmt->altsetting,
+			   get_endpoint(alts, 1)->bmAttributes,
+			   get_endpoint(alts, 1)->bLength,
+			   get_endpoint(alts, 1)->bSynchAddress);
+		return -EINVAL;
+	}
+	ep = get_endpoint(alts, 1)->bEndpointAddress;
+	if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+	    ((is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
+	     (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
+		snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n",
+			   dev->devnum, fmt->iface, fmt->altsetting,
+			   is_playback, ep, get_endpoint(alts, 0)->bSynchAddress);
+		return -EINVAL;
+	}
+
+	implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK)
+			== USB_ENDPOINT_USAGE_IMPLICIT_FB;
+
+	subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
+						   alts, ep, !subs->direction,
+						   implicit_fb ?
+							SND_USB_ENDPOINT_TYPE_DATA :
+							SND_USB_ENDPOINT_TYPE_SYNC);
+	if (!subs->sync_endpoint)
+		return -EINVAL;
+
+	subs->data_endpoint->sync_master = subs->sync_endpoint;
+
+	return 0;
+}
+
 /*
 * find a matching format and set up the interface
 */
@ -336,9 +467,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 	struct usb_host_interface *alts;
 	struct usb_interface_descriptor *altsd;
 	struct usb_interface *iface;
-	unsigned int ep, attr;
-	int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
-	int err, implicit_fb = 0;
+	int err;

 	iface = usb_ifnum_to_if(dev, fmt->iface);
 	if (WARN_ON(!iface))
@ -383,118 +512,22 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 	subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip,
 						   alts, fmt->endpoint, subs->direction,
 						   SND_USB_ENDPOINT_TYPE_DATA);
+
 	if (!subs->data_endpoint)
 		return -EINVAL;

-	/* we need a sync pipe in async OUT or adaptive IN mode */
-	/* check the number of EP, since some devices have broken
-	 * descriptors which fool us.  if it has only one EP,
-	 * assume it as adaptive-out or sync-in.
-	 */
-	attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE;
+	err = set_sync_endpoint(subs, fmt, dev, alts, altsd);
+	if (err < 0)
+		return err;

-	switch (subs->stream->chip->usb_id) {
-	case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
-	case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
-		if (is_playback) {
-			implicit_fb = 1;
-			ep = 0x81;
-			iface = usb_ifnum_to_if(dev, 3);
-
-			if (!iface || iface->num_altsetting == 0)
-				return -EINVAL;
-
-			alts = &iface->altsetting[1];
-			goto add_sync_ep;
-		}
-		break;
-	case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
-	case USB_ID(0x0763, 0x2081):
-		if (is_playback) {
-			implicit_fb = 1;
-			ep = 0x81;
-			iface = usb_ifnum_to_if(dev, 2);
-
-			if (!iface || iface->num_altsetting == 0)
-				return -EINVAL;
-
-			alts = &iface->altsetting[1];
-			goto add_sync_ep;
-		}
-	}
-	if (is_playback &&
-	    attr == USB_ENDPOINT_SYNC_ASYNC &&
-	    altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
-	    altsd->bInterfaceProtocol == 2 &&
-	    altsd->bNumEndpoints == 1 &&
-	    USB_ID_VENDOR(subs->stream->chip->usb_id) == 0x0582 /* Roland */ &&
-	    search_roland_implicit_fb(dev, altsd->bInterfaceNumber + 1,
-				      altsd->bAlternateSetting,
-				      &alts, &ep) >= 0) {
-		implicit_fb = 1;
-		goto add_sync_ep;
-	}
-
-	if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) ||
-	     (!is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) &&
-	    altsd->bNumEndpoints >= 2) {
-		/* check sync-pipe endpoint */
-		/* ... and check descriptor size before accessing bSynchAddress
-		   because there is a version of the SB Audigy 2 NX firmware lacking
-		   the audio fields in the endpoint descriptors */
-		if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC ||
-		    (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
-		     get_endpoint(alts, 1)->bSynchAddress != 0 &&
-		     !implicit_fb)) {
-			snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n",
-				   dev->devnum, fmt->iface, fmt->altsetting,
-				   get_endpoint(alts, 1)->bmAttributes,
-				   get_endpoint(alts, 1)->bLength,
-				   get_endpoint(alts, 1)->bSynchAddress);
-			return -EINVAL;
-		}
-		ep = get_endpoint(alts, 1)->bEndpointAddress;
-		if (!implicit_fb &&
-		    get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
-		    (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
-		     (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
-			snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n",
-				   dev->devnum, fmt->iface, fmt->altsetting,
-				   is_playback, ep, get_endpoint(alts, 0)->bSynchAddress);
-			return -EINVAL;
-		}
-
-		implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK)
-				== USB_ENDPOINT_USAGE_IMPLICIT_FB;
-
-add_sync_ep:
-		subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
-							   alts, ep, !subs->direction,
-							   implicit_fb ?
-								SND_USB_ENDPOINT_TYPE_DATA :
-								SND_USB_ENDPOINT_TYPE_SYNC);
-		if (!subs->sync_endpoint)
-			return -EINVAL;
-
-		subs->data_endpoint->sync_master = subs->sync_endpoint;
-	}
-
-	if ((err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt)) < 0)
+	err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt);
+	if (err < 0)
 		return err;

 	subs->cur_audiofmt = fmt;

 	snd_usb_set_format_quirk(subs, fmt);

-#if 0
-	printk(KERN_DEBUG
-	       "setting done: format = %d, rate = %d..%d, channels = %d\n",
-	       fmt->format, fmt->rate_min, fmt->rate_max, fmt->channels);
-	printk(KERN_DEBUG
-	       "  datapipe = 0x%0x, syncpipe = 0x%0x\n",
-	       subs->datapipe, subs->syncpipe);
-#endif
-
 	return 0;
 }

--- a/sound/usb/usx2y/usbusx2y.c
+++ b/sound/usb/usx2y/usbusx2y.c
@ -305,11 +305,9 @@ static void usX2Y_unlinkSeq(struct snd_usX2Y_AsyncSeq *S)
 {
 	int	i;
 	for (i = 0; i < URBS_AsyncSeq; ++i) {
-		if (S[i].urb) {
-			usb_kill_urb(S->urb[i]);
-			usb_free_urb(S->urb[i]);
-			S->urb[i] = NULL;
-		}
+		usb_kill_urb(S->urb[i]);
+		usb_free_urb(S->urb[i]);
+		S->urb[i] = NULL;
 	}
 	kfree(S->buffer);
 }