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

63 files changed, 2170 insertions, 5013 deletions

diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
new file mode 100644
index 000000000000..7d13479f9c3c
--- /dev/null
+++ 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;
+};
diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt
new file mode 100644
index 000000000000..7e5fd37c1b3f
--- /dev/null
+++ 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";
+};
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 809d72b8eff1..a46ddb85e83a 100644
--- 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)
 
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index c3c912d023cc..42a0a39b77e6 100644
--- 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
diff --git a/MAINTAINERS b/MAINTAINERS
index b5e09128898f..a77b9440d87d 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -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>
diff --git a/arch/arm/plat-samsung/s3c-dma-ops.c b/arch/arm/plat-samsung/s3c-dma-ops.c
index 0cc40aea3f5a..98b10ba67dc7 100644
--- 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;
 }
diff --git a/include/sound/core.h b/include/sound/core.h
index c586617cfa0d..2a14f1f02d4f 100644
--- 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
@@ -376,6 +377,11 @@ 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
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index c728d28ae9a5..27a72d5d4b00 100644
--- 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);
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 8e2ad52078b6..d22cb0a06feb 100644
--- 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 {
diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h
index 1f59ea2a4a76..d956c3593f65 100644
--- 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
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 82bb029d4414..6e03b465e44e 100644
--- 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);
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 11048cc744d0..915b4d7fbb23 100644
--- 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;
diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c
index 2c6386503940..fe9e6e2f2c5b 100644
--- 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;
 }
 
diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c
index 9942691cc0ca..afef0d738078 100644
--- 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 */
diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c
index a59c88818f48..461d94cfecbe 100644
--- 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);
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 59c5e9c03d53..8de66ccd7279 100644
--- 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"
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 8a005f0e5ca4..5b6c4e3c92ca 100644
--- 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);
 }
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 701c2e069b10..7aa9870040c1 100644
--- 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);
 
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index e3c7ba8d7582..ac41e9cdc976 100644
--- 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(struc