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

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;
+};

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";
+};

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)
 

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

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>

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;
 }

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

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);

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 {

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

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);

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;

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;
 }
 

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 */

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);

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"

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);
 }

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);
 

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;
 		}

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 */