sound fixes for 4.17-rc3

A significant amount of fixes have been piled up at this time.
 
 - Possible Spectre v1 coverage in OSS sequencer API, control API,
   HD-audio hwdep ioctl, ASIHPI hwdep ioctl, OPL3, and HDSPM/RME
   channel_info API.
 
 - A regression fix in PCM delay reporting that happened at the code
   refactoring for the set_fs() removal
 
 - The long-standing bug in PCM sync_ptr ioctl that missed the audio
   timestamp field
 
 - USB-audio regression fixes due to the recent UAC2 jack support
 
 - vm_fault_t conversions in a couple of places
 
 - ASoC topology API fixes
 
 - Assorted driver fixes:
   * ASoC rsnd, FSL, Intel SST, DMIC, AMD, ADAU17x1, Realtek codec
   * FireWire typo fix
   * HD-audio quirks and USB-audio Dell fixup
   * USB-audio UAC3 corrections
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Merge tag 'sound-4.17-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A significant amount of fixes have been piled up at this time.

   - Possible Spectre v1 coverage in OSS sequencer API, control API,
     HD-audio hwdep ioctl, ASIHPI hwdep ioctl, OPL3, and HDSPM/RME
     channel_info API.

   - A regression fix in PCM delay reporting that happened at the code
     refactoring for the set_fs() removal

   - The long-standing bug in PCM sync_ptr ioctl that missed the audio
     timestamp field

   - USB-audio regression fixes due to the recent UAC2 jack support

   - vm_fault_t conversions in a couple of places

   - ASoC topology API fixes

   - Assorted driver fixes:
      * ASoC rsnd, FSL, Intel SST, DMIC, AMD, ADAU17x1, Realtek codec
      * FireWire typo fix
      * HD-audio quirks and USB-audio Dell fixup
      * USB-audio UAC3 corrections"

* tag 'sound-4.17-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (35 commits)
  ALSA: dice: fix error path to destroy initialized stream data
  ALSA: hda - Skip jack and others for non-existing PCM streams
  ALSA: hda/realtek - change the location for one of two front mics
  ALSA: rme9652: Hardening for potential Spectre v1
  ALSA: hdspm: Hardening for potential Spectre v1
  ALSA: asihpi: Hardening for potential Spectre v1
  ALSA: opl3: Hardening for potential Spectre v1
  ALSA: hda: Hardening for potential Spectre v1
  ALSA: control: Hardening for potential Spectre v1
  ALSA: seq: oss: Hardening for potential Spectre v1
  ALSA: seq: oss: Fix unbalanced use lock for synth MIDI device
  ALSA: hda/realtek - Update ALC255 depop optimize
  ALSA: hda/realtek - Add some fixes for ALC233
  ALSA: pcm: Change return type to vm_fault_t
  ALSA: usx2y: Change return type to vm_fault_t
  ALSA: usb-audio: ADC3: Fix channel mapping conversion for ADC3.
  ALSA: dice: fix OUI for TC group
  ALSA: usb-audio: Skip broken EU on Dell dock USB-audio
  ALSA: usb-audio: Fix missing endian conversion
  ALSA: usb-audio: Fix forgotten conversion of control query functions
  ...

include/sound/control.h

@@ -23,6 +23,7 @@
  */
 
 #include <linux/wait.h>
+#include <linux/nospec.h>
 #include <sound/asound.h>
 
 #define snd_kcontrol_chip(kcontrol) ((kcontrol)->private_data)
@@ -148,12 +149,14 @@ int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type);
 
 static inline unsigned int snd_ctl_get_ioffnum(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id)
 {
-	return id->numid - kctl->id.numid;
+	unsigned int ioff = id->numid - kctl->id.numid;
+	return array_index_nospec(ioff, kctl->count);
 }
 
 static inline unsigned int snd_ctl_get_ioffidx(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id)
 {
-	return id->index - kctl->id.index;
+	unsigned int ioff = id->index - kctl->id.index;
+	return array_index_nospec(ioff, kctl->count);
 }
 
 static inline unsigned int snd_ctl_get_ioff(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id)

sound/core/control.c

@@ -1492,7 +1492,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file,
                              int op_flag)
 {
 	struct snd_ctl_tlv header;
-	unsigned int *container;
+	unsigned int __user *container;
 	unsigned int container_size;
 	struct snd_kcontrol *kctl;
 	struct snd_ctl_elem_id id;

sound/core/pcm_compat.c

@@ -27,10 +27,11 @@ static int snd_pcm_ioctl_delay_compat(struct snd_pcm_substream *substream,
 				      s32 __user *src)
 {
 	snd_pcm_sframes_t delay;
+	int err;
 
-	delay = snd_pcm_delay(substream);
+	err = snd_pcm_delay(substream, &delay);
-	if (delay < 0)
+	if (err)
-		return delay;
+		return err;
 	if (put_user(delay, src))
 		return -EFAULT;
 	return 0;

sound/core/pcm_native.c

@@ -2692,7 +2692,8 @@ static int snd_pcm_hwsync(struct snd_pcm_substream *substream)
 	return err;
 }
 		
-static snd_pcm_sframes_t snd_pcm_delay(struct snd_pcm_substream *substream)
+static int snd_pcm_delay(struct snd_pcm_substream *substream,
+			 snd_pcm_sframes_t *delay)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	int err;
@@ -2708,7 +2709,9 @@ static snd_pcm_sframes_t snd_pcm_delay(struct snd_pcm_substream *substream)
 		n += runtime->delay;
 	}
 	snd_pcm_stream_unlock_irq(substream);
-	return err < 0 ? err : n;
+	if (!err)
+		*delay = n;
+	return err;
 }
 		
 static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream,
@@ -2751,6 +2754,7 @@ static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream,
 	sync_ptr.s.status.hw_ptr = status->hw_ptr;
 	sync_ptr.s.status.tstamp = status->tstamp;
 	sync_ptr.s.status.suspended_state = status->suspended_state;
+	sync_ptr.s.status.audio_tstamp = status->audio_tstamp;
 	snd_pcm_stream_unlock_irq(substream);
 	if (copy_to_user(_sync_ptr, &sync_ptr, sizeof(sync_ptr)))
 		return -EFAULT;
@@ -2916,11 +2920,13 @@ static int snd_pcm_common_ioctl(struct file *file,
 		return snd_pcm_hwsync(substream);
 	case SNDRV_PCM_IOCTL_DELAY:
 	{
-		snd_pcm_sframes_t delay = snd_pcm_delay(substream);
+		snd_pcm_sframes_t delay;
 		snd_pcm_sframes_t __user *res = arg;
+		int err;
 
-		if (delay < 0)
+		err = snd_pcm_delay(substream, &delay);
-			return delay;
+		if (err)
+			return err;
 		if (put_user(delay, res))
 			return -EFAULT;
 		return 0;
@@ -3008,13 +3014,7 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream,
 	case SNDRV_PCM_IOCTL_DROP:
 		return snd_pcm_drop(substream);
 	case SNDRV_PCM_IOCTL_DELAY:
-	{
+		return snd_pcm_delay(substream, frames);
-		result = snd_pcm_delay(substream);
-		if (result < 0)
-			return result;
-		*frames = result;
-		return 0;
-	}
 	default:
 		return -EINVAL;
 	}
@@ -3234,7 +3234,7 @@ static __poll_t snd_pcm_capture_poll(struct file *file, poll_table * wait)
 /*
  * mmap status record
  */
-static int snd_pcm_mmap_status_fault(struct vm_fault *vmf)
+static vm_fault_t snd_pcm_mmap_status_fault(struct vm_fault *vmf)
 {
 	struct snd_pcm_substream *substream = vmf->vma->vm_private_data;
 	struct snd_pcm_runtime *runtime;
@@ -3270,7 +3270,7 @@ static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file
 /*
  * mmap control record
  */
-static int snd_pcm_mmap_control_fault(struct vm_fault *vmf)
+static vm_fault_t snd_pcm_mmap_control_fault(struct vm_fault *vmf)
 {
 	struct snd_pcm_substream *substream = vmf->vma->vm_private_data;
 	struct snd_pcm_runtime *runtime;
@@ -3359,7 +3359,7 @@ snd_pcm_default_page_ops(struct snd_pcm_substream *substream, unsigned long ofs)
 /*
  * fault callback for mmapping a RAM page
  */
-static int snd_pcm_mmap_data_fault(struct vm_fault *vmf)
+static vm_fault_t snd_pcm_mmap_data_fault(struct vm_fault *vmf)
 {
 	struct snd_pcm_substream *substream = vmf->vma->vm_private_data;
 	struct snd_pcm_runtime *runtime;

sound/core/seq/oss/seq_oss_event.c

@@ -26,6 +26,7 @@
 #include <sound/seq_oss_legacy.h>
 #include "seq_oss_readq.h"
 #include "seq_oss_writeq.h"
+#include <linux/nospec.h>
 
 
 /*
@@ -287,10 +288,10 @@ note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, st
 {
 	struct seq_oss_synthinfo *info;
 
-	if (!snd_seq_oss_synth_is_valid(dp, dev))
+	info = snd_seq_oss_synth_info(dp, dev);
+	if (!info)
 		return -ENXIO;
 
-	info = &dp->synths[dev];
 	switch (info->arg.event_passing) {
 	case SNDRV_SEQ_OSS_PROCESS_EVENTS:
 		if (! info->ch || ch < 0 || ch >= info->nr_voices) {
@@ -298,6 +299,7 @@ note_on_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, st
 			return set_note_event(dp, dev, SNDRV_SEQ_EVENT_NOTEON, ch, note, vel, ev);
 		}
 
+		ch = array_index_nospec(ch, info->nr_voices);
 		if (note == 255 && info->ch[ch].note >= 0) {
 			/* volume control */
 			int type;
@@ -347,10 +349,10 @@ note_off_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, s
 {
 	struct seq_oss_synthinfo *info;
 
-	if (!snd_seq_oss_synth_is_valid(dp, dev))
+	info = snd_seq_oss_synth_info(dp, dev);
+	if (!info)
 		return -ENXIO;
 
-	info = &dp->synths[dev];
 	switch (info->arg.event_passing) {
 	case SNDRV_SEQ_OSS_PROCESS_EVENTS:
 		if (! info->ch || ch < 0 || ch >= info->nr_voices) {
@@ -358,6 +360,7 @@ note_off_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, s
 			return set_note_event(dp, dev, SNDRV_SEQ_EVENT_NOTEON, ch, note, vel, ev);
 		}
 
+		ch = array_index_nospec(ch, info->nr_voices);
 		if (info->ch[ch].note >= 0) {
 			note = info->ch[ch].note;
 			info->ch[ch].vel = 0;
@@ -381,7 +384,7 @@ note_off_event(struct seq_oss_devinfo *dp, int dev, int ch, int note, int vel, s
 static int
 set_note_event(struct seq_oss_devinfo *dp, int dev, int type, int ch, int note, int vel, struct snd_seq_event *ev)
 {
-	if (! snd_seq_oss_synth_is_valid(dp, dev))
+	if (!snd_seq_oss_synth_info(dp, dev))
 		return -ENXIO;
 	
 	ev->type = type;
@@ -399,7 +402,7 @@ set_note_event(struct seq_oss_devinfo *dp, int dev, int type, int ch, int note,
 static int
 set_control_event(struct seq_oss_devinfo *dp, int dev, int type, int ch, int param, int val, struct snd_seq_event *ev)
 {
-	if (! snd_seq_oss_synth_is_valid(dp, dev))
+	if (!snd_seq_oss_synth_info(dp, dev))
 		return -ENXIO;
 	
 	ev->type = type;

sound/core/seq/oss/seq_oss_midi.c

@@ -29,6 +29,7 @@
 #include "../seq_lock.h"
 #include <linux/init.h>
 #include <linux/slab.h>
+#include <linux/nospec.h>
 
 
 /*
@@ -315,6 +316,7 @@ get_mididev(struct seq_oss_devinfo *dp, int dev)
 {
 	if (dev < 0 || dev >= dp->max_mididev)
 		return NULL;
+	dev = array_index_nospec(dev, dp->max_mididev);
 	return get_mdev(dev);
 }
 

sound/core/seq/oss/seq_oss_synth.c

@@ -26,6 +26,7 @@
 #include <linux/init.h>
 #include <linux/module.h>
 #include <linux/slab.h>
+#include <linux/nospec.h>
 
 /*
  * constants
@@ -339,17 +340,13 @@ snd_seq_oss_synth_cleanup(struct seq_oss_devinfo *dp)
 	dp->max_synthdev = 0;
 }
 
-/*
+static struct seq_oss_synthinfo *
- * check if the specified device is MIDI mapped device
+get_synthinfo_nospec(struct seq_oss_devinfo *dp, int dev)
- */
-static int
-is_midi_dev(struct seq_oss_devinfo *dp, int dev)
 {
 	if (dev < 0 || dev >= dp->max_synthdev)
-		return 0;
+		return NULL;
-	if (dp->synths[dev].is_midi)
+	dev = array_index_nospec(dev, SNDRV_SEQ_OSS_MAX_SYNTH_DEVS);
-		return 1;
+	return &dp->synths[dev];
-	return 0;
 }
 
 /*
@@ -359,14 +356,20 @@ static struct seq_oss_synth *
 get_synthdev(struct seq_oss_devinfo *dp, int dev)
 {
 	struct seq_oss_synth *rec;
-	if (dev < 0 || dev >= dp->max_synthdev)
+	struct seq_oss_synthinfo *info = get_synthinfo_nospec(dp, dev);
+
+	if (!info)
 		return NULL;
-	if (! dp->synths[dev].opened)
+	if (!info->opened)
-		return NULL;
-	if (dp->synths[dev].is_midi)
-		return &midi_synth_dev;
-	if ((rec = get_sdev(dev)) == NULL)
 		return NULL;
+	if (info->is_midi) {
+		rec = &midi_synth_dev;
+		snd_use_lock_use(&rec->use_lock);
+	} else {
+		rec = get_sdev(dev);
+		if (!rec)
+			return NULL;
+	}
 	if (! rec->opened) {
 		snd_use_lock_free(&rec->use_lock);
 		return NULL;
@@ -402,10 +405,8 @@ snd_seq_oss_synth_reset(struct seq_oss_devinfo *dp, int dev)
 	struct seq_oss_synth *rec;
 	struct seq_oss_synthinfo *info;
 
-	if (snd_BUG_ON(dev < 0 || dev >= dp->max_synthdev))
+	info = get_synthinfo_nospec(dp, dev);
-		return;
+	if (!info || !info->opened)
-	info = &dp->synths[dev];
-	if (! info->opened)
 		return;
 	if (info->sysex)
 		info->sysex->len = 0; /* reset sysex */
@@ -454,12 +455,14 @@ snd_seq_oss_synth_load_patch(struct seq_oss_devinfo *dp, int dev, int fmt,
 			    const char __user *buf, int p, int c)
 {
 	struct seq_oss_synth *rec;
+	struct seq_oss_synthinfo *info;
 	int rc;
 
-	if (dev < 0 || dev >= dp->max_synthdev)
+	info = get_synthinfo_nospec(dp, dev);
+	if (!info)
 		return -ENXIO;
 
-	if (is_midi_dev(dp, dev))
+	if (info->is_midi)
 		return 0;
 	if ((rec = get_synthdev(dp, dev)) == NULL)
 		return -ENXIO;
@@ -467,24 +470,25 @@ snd_seq_oss_synth_load_patch(struct seq_oss_devinfo *dp, int dev, int fmt,
 	if (rec->oper.load_patch == NULL)
 		rc = -ENXIO;
 	else
-		rc = rec->oper.load_patch(&dp->synths[dev].arg, fmt, buf, p, c);
+		rc = rec->oper.load_patch(&info->arg, fmt, buf, p, c);
 	snd_use_lock_free(&rec->use_lock);
 	return rc;
 }
 
 /*
- * check if the device is valid synth device
+ * check if the device is valid synth device and return the synth info
  */
-int
+struct seq_oss_synthinfo *
-snd_seq_oss_synth_is_valid(struct seq_oss_devinfo *dp, int dev)
+snd_seq_oss_synth_info(struct seq_oss_devinfo *dp, int dev)
 {
 	struct seq_oss_synth *rec;
+
 	rec = get_synthdev(dp, dev);
 	if (rec) {
 		snd_use_lock_free(&rec->use_lock);
-		return 1;
+		return get_synthinfo_nospec(dp, dev);
 	}
-	return 0;
+	return NULL;
 }
 
 
@@ -499,16 +503,18 @@ snd_seq_oss_synth_sysex(struct seq_oss_devinfo *dp, int dev, unsigned char *buf,
 	int i, send;
 	unsigned char *dest;
 	struct seq_oss_synth_sysex *sysex;
+	struct seq_oss_synthinfo *info;
 
-	if (! snd_seq_oss_synth_is_valid(dp, dev))
+	info = snd_seq_oss_synth_info(dp, dev);
+	if (!info)
 		return -ENXIO;
 
-	sysex = dp->synths[dev].sysex;
+	sysex = info->sysex;
 	if (sysex == NULL) {
 		sysex = kzalloc(sizeof(*sysex), GFP_KERNEL);
 		if (sysex == NULL)
 			return -ENOMEM;
-		dp->synths[dev].sysex = sysex;
+		info->sysex = sysex;
 	}
 
 	send = 0;
@@ -553,10 +559,12 @@ snd_seq_oss_synth_sysex(struct seq_oss_devinfo *dp, int dev, unsigned char *buf,
 int
 snd_seq_oss_synth_addr(struct seq_oss_devinfo *dp, int dev, struct snd_seq_event *ev)
 {
-	if (! snd_seq_oss_synth_is_valid(dp, dev))
+	struct seq_oss_synthinfo *info = snd_seq_oss_synth_info(dp, dev);
+
+	if (!info)
 		return -EINVAL;
-	snd_seq_oss_fill_addr(dp, ev, dp->synths[dev].arg.addr.client,
+	snd_seq_oss_fill_addr(dp, ev, info->arg.addr.client,
-			      dp->synths[dev].arg.addr.port);
+			      info->arg.addr.port);
 	return 0;
 }
 
@@ -568,16 +576,18 @@ int
 snd_seq_oss_synth_ioctl(struct seq_oss_devinfo *dp, int dev, unsigned int cmd, unsigned long addr)
 {
 	struct seq_oss_synth *rec;
+	struct seq_oss_synthinfo *info;
 	int rc;
 
-	if (is_midi_dev(dp, dev))
+	info = get_synthinfo_nospec(dp, dev);
+	if (!info || info->is_midi)
 		return -ENXIO;
 	if ((rec = get_synthdev(dp, dev)) == NULL)
 		return -ENXIO;
 	if (rec->oper.ioctl == NULL)
 		rc = -ENXIO;
 	else
-		rc = rec->oper.ioctl(&dp->synths[dev].arg, cmd, addr);
+		rc = rec->oper.ioctl(&info->arg, cmd, addr);
 	snd_use_lock_free(&rec->use_lock);
 	return rc;
 }
@@ -589,7 +599,10 @@ snd_seq_oss_synth_ioctl(struct seq_oss_devinfo *dp, int dev, unsigned int cmd, u
 int
 snd_seq_oss_synth_raw_event(struct seq_oss_devinfo *dp, int dev, unsigned char *data, struct snd_seq_event *ev)
 {
-	if (! snd_seq_oss_synth_is_valid(dp, dev) || is_midi_dev(dp, dev))
+	struct seq_oss_synthinfo *info;
+
+	info = snd_seq_oss_synth_info(dp, dev);
+	if (!info || info->is_midi)
 		return -ENXIO;
 	ev->type = SNDRV_SEQ_EVENT_OSS;
 	memcpy(ev->data.raw8.d, data, 8);

sound/core/seq/oss/seq_oss_synth.h

@@ -37,7 +37,8 @@ void snd_seq_oss_synth_cleanup(struct seq_oss_devinfo *dp);
 void snd_seq_oss_synth_reset(struct seq_oss_devinfo *dp, int dev);
 int snd_seq_oss_synth_load_patch(struct seq_oss_devinfo *dp, int dev, int fmt,
 				 const char __user *buf, int p, int c);
-int snd_seq_oss_synth_is_valid(struct seq_oss_devinfo *dp, int dev);
+struct seq_oss_synthinfo *snd_seq_oss_synth_info(struct seq_oss_devinfo *dp,
+						 int dev);
 int snd_seq_oss_synth_sysex(struct seq_oss_devinfo *dp, int dev, unsigned char *buf,
 			    struct snd_seq_event *ev);
 int snd_seq_oss_synth_addr(struct seq_oss_devinfo *dp, int dev, struct snd_seq_event *ev);

sound/drivers/opl3/opl3_synth.c

@@ -21,6 +21,7 @@
 
 #include <linux/slab.h>
 #include <linux/export.h>
+#include <linux/nospec.h>
 #include <sound/opl3.h>
 #include <sound/asound_fm.h>
 
@@ -448,7 +449,7 @@ static int snd_opl3_set_voice(struct snd_opl3 * opl3, struct snd_dm_fm_voice * v
 {
 	unsigned short reg_side;
 	unsigned char op_offset;
-	unsigned char voice_offset;
+	unsigned char voice_offset, voice_op;
 
 	unsigned short opl3_reg;
 	unsigned char reg_val;
@@ -473,7 +474,9 @@ static int snd_opl3_set_voice(struct snd_opl3 * opl3, struct snd_dm_fm_voice * v
 		voice_offset = voice->voice - MAX_OPL2_VOICES;
 	}
 	/* Get register offset of operator */
-	op_offset = snd_opl3_regmap[voice_offset][voice->op];
+	voice_offset = array_index_nospec(voice_offset, MAX_OPL2_VOICES);
+	voice_op = array_index_nospec(voice->op, 4);
+	op_offset = snd_opl3_regmap[voice_offset][voice_op];
 
 	reg_val = 0x00;
 	/* Set amplitude modulation (tremolo) effect */

sound/firewire/dice/dice-stream.c

@@ -435,7 +435,7 @@ int snd_dice_stream_init_duplex(struct snd_dice *dice)
 		err = init_stream(dice, AMDTP_IN_STREAM, i);
 		if (err < 0) {
 			for (; i >= 0; i--)
-				destroy_stream(dice, AMDTP_OUT_STREAM, i);
+				destroy_stream(dice, AMDTP_IN_STREAM, i);
 			goto end;
 		}
 	}

sound/firewire/dice/dice.c

@@ -14,7 +14,7 @@ MODULE_LICENSE("GPL v2");
 #define OUI_WEISS		0x001c6a
 #define OUI_LOUD		0x000ff2
 #define OUI_FOCUSRITE		0x00130e
-#define OUI_TCELECTRONIC	0x001486
+#define OUI_TCELECTRONIC	0x000166
 
 #define DICE_CATEGORY_ID	0x04
 #define WEISS_CATEGORY_ID	0x00

sound/pci/asihpi/hpimsginit.c

@@ -23,6 +23,7 @@
 
 #include "hpi_internal.h"
 #include "hpimsginit.h"
+#include <linux/nospec.h>
 
 /* The actual message size for each object type */
 static u16 msg_size[HPI_OBJ_MAXINDEX + 1] = HPI_MESSAGE_SIZE_BY_OBJECT;
@@ -39,10 +40,12 @@ static void hpi_init_message(struct hpi_message *phm, u16 object,
 {
 	u16 size;
 
-	if ((object > 0) && (object <= HPI_OBJ_MAXINDEX))
+	if ((object > 0) && (object <= HPI_OBJ_MAXINDEX)) {
+		object = array_index_nospec(object, HPI_OBJ_MAXINDEX + 1);
 		size = msg_size[object];
-	else
+	} else {
 		size = sizeof(*phm);
+	}
 
 	memset(phm, 0, size);
 	phm->size = size;
@@ -66,10 +69,12 @@ void hpi_init_response(struct hpi_response *phr, u16 object, u16 function,
 {
 	u16 size;
 
-	if ((object > 0) && (object <= HPI_OBJ_MAXINDEX))
+	if ((object > 0) && (object <= HPI_OBJ_MAXINDEX)) {
+		object = array_index_nospec(object, HPI_OBJ_MAXINDEX + 1);
 		size = res_size[object];
-	else
+	} else {
 		size = sizeof(*phr);
+	}
 
 	memset(phr, 0, sizeof(*phr));
 	phr->size = size;

sound/pci/asihpi/hpioctl.c

@@ -33,6 +33,7 @@
 #include <linux/stringify.h>
 #include <linux/module.h>
 #include <linux/vmalloc.h>
+#include <linux/nospec.h>
 
 #ifdef MODULE_FIRMWARE
 MODULE_FIRMWARE("asihpi/dsp5000.bin");
@@ -186,7 +187,8 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
 		struct hpi_adapter *pa = NULL;
 
 		if (hm->h.adapter_index < ARRAY_SIZE(adapters))
-			pa = &adapters[hm->h.adapter_index];
+			pa = &adapters[array_index_nospec(hm->h.adapter_index,
+							  ARRAY_SIZE(adapters))];
 
 		if (!pa || !pa->adapter || !pa->adapter->type) {
 			hpi_init_response(&hr->r0, hm->h.object,

sound/pci/hda/hda_hwdep.c

@@ -21,6 +21,7 @@
 #include <linux/init.h>
 #include <linux/slab.h>
 #include <linux/compat.h>
+#include <linux/nospec.h>
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
@@ -51,7 +52,16 @@ static int get_wcap_ioctl(struct hda_codec *codec,
 	
 	if (get_user(verb, &arg->verb))
 		return -EFAULT;
-	res = get_wcaps(codec, verb >> 24);
+	/* open-code get_wcaps(verb>>24) with nospec */
+	verb >>= 24;
+	if (verb < codec->core.start_nid ||
+	    verb >= codec->core.start_nid + codec->core.num_nodes) {
+		res = 0;
+	} else {
+		verb -= codec->core.start_nid;
+		verb = array_index_nospec(verb, codec->core.num_nodes);
+		res = codec->wcaps[verb];
+	}
 	if (put_user(res, &arg->res))
 		return -EFAULT;
 	return 0;

sound/pci/hda/patch_hdmi.c

@@ -1383,6 +1383,8 @@ static void hdmi_pcm_setup_pin(struct hdmi_spec *spec,
 		pcm = get_pcm_rec(spec, per_pin->pcm_idx);
 	else
 		return;
+	if (!pcm->pcm)
+		return;
 	if (!test_bit(per_pin->pcm_idx, &spec->pcm_in_use))
 		return;
 
@@ -2151,8 +2153,13 @@ static int generic_hdmi_build_controls(struct hda_codec *codec)
 	int dev, err;
 	int pin_idx, pcm_idx;
 
-
 	for (pcm_idx = 0; pcm_idx < spec->pcm_used; pcm_idx++) {
+		if (!get_pcm_rec(spec, pcm_idx)->pcm) {
+			/* no PCM: mark this for skipping permanently */
+			set_bit(pcm_idx, &spec->pcm_bitmap);
+			continue;
+		}
+
 		err = generic_hdmi_build_jack(codec, pcm_idx);
 		if (err < 0)
 			return err;

sound/pci/hda/patch_realtek.c

@@ -331,6 +331,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
 		/* fallthrough */
 	case 0x10ec0215:
 	case 0x10ec0233:
+	case 0x10ec0235:
 	case 0x10ec0236:
 	case 0x10ec0255:
 	case 0x10ec0256:
@@ -6575,6 +6576,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
 	SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
 	SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION),
+	SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION),
 	SND_PCI_QUIRK(0x17aa, 0x3138, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION),
 	SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION),
 	SND_PCI_QUIRK(0x17aa, 0x3112, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY),
@@ -7160,8 +7162,11 @@ static int patch_alc269(struct hda_codec *codec)
 	case 0x10ec0298:
 		spec->codec_variant = ALC269_TYPE_ALC298;
 		break;
+	case 0x10ec0235:
 	case 0x10ec0255:
 		spec->codec_variant = ALC269_TYPE_ALC255;
+		spec->shutup = alc256_shutup;
+		spec->init_hook = alc256_init;
 		break;
 	case 0x10ec0236:
 	case 0x10ec0256:

sound/pci/rme9652/hdspm.c

@@ -137,6 +137,7 @@
 #include <linux/pci.h>
 #include <linux/math64.h>
 #include <linux/io.h>
+#include <linux/nospec.h>
 
 #include <sound/core.h>
 #include <sound/control.h>
@@ -5698,40 +5699,43 @@ static int snd_hdspm_channel_info(struct snd_pcm_substream *substream,
 		struct snd_pcm_channel_info *info)
 {
 	struct hdspm *hdspm = snd_pcm_substream_chip(substream);
+	unsigned int channel = info->channel;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		if (snd_BUG_ON(info->channel >= hdspm->max_channels_out)) {
+		if (snd_BUG_ON(channel >= hdspm->max_channels_out)) {
 			dev_info(hdspm->card->dev,
 				 "snd_hdspm_channel_info: output channel out of range (%d)\n",
-				 info->channel);
+				 channel);
 			return -EINVAL;
 		}
 
-		if (hdspm->channel_map_out[info->channel] < 0) {
+		channel = array_index_nospec(channel, hdspm->max_channels_out);
+		if (hdspm->channel_map_out[channel] < 0) {
 			dev_info(hdspm->card->dev,
 				 "snd_hdspm_channel_info: output channel %d mapped out\n",
-				 info->channel);
+				 channel);
 			return -EINVAL;
 		}
 
-		info->offset = hdspm->channel_map_out[info->channel] *
+		info->offset = hdspm->channel_map_out[channel] *
 			HDSPM_CHANNEL_BUFFER_BYTES;
 	} else {
-		if (snd_BUG_ON(info->channel >= hdspm->max_channels_in)) {
+		if (snd_BUG_ON(channel >= hdspm->max_channels_in)) {
 			dev_info(hdspm->card->dev,
 				 "snd_hdspm_channel_info: input channel out of range (%d)\n",
-				 info->channel);
+				 channel);
 			return -EINVAL;
 		}
 
-		if (hdspm->channel_map_in[info->channel] < 0) {
+		channel = array_index_nospec(channel, hdspm->max_channels_in);
+		if (hdspm->channel_map_in[channel] < 0) {
 			dev_info(hdspm->card->dev,
 				 "snd_hdspm_channel_info: input channel %d mapped out\n",
-				 info->channel);
+				 channel);
 			return -EINVAL;
 		}
 
-		info->offset = hdspm->channel_map_in[info->channel] *
+		info->offset = hdspm->channel_map_in[channel] *
 			HDSPM_CHANNEL_BUFFER_BYTES;
 	}
 

sound/pci/rme9652/rme9652.c

@@ -26,6 +26,7 @@
 #include <linux/pci.h>
 #include <linux/module.h>
 #include <linux/io.h>
+#include <linux/nospec.h>
 
 #include <sound/core.h>
 #include <sound/control.h>
@@ -2071,9 +2072,10 @@ static int snd_rme9652_channel_info(struct snd_pcm_substream *substream,
 	if (snd_BUG_ON(info->channel >= RME9652_NCHANNELS))
 		return -EINVAL;
 
-	if ((chn = rme9652->channel_map[info->channel]) < 0) {
+	chn = rme9652->channel_map[array_index_nospec(info->channel,
+						      RME9652_NCHANNELS)];
+	if (chn < 0)
 		return -EINVAL;
-	}
 
 	info->offset = chn * RME9652_CHANNEL_BUFFER_BYTES;
 	info->first = 0;

sound/soc/amd/acp-da7219-max98357a.c

@@ -43,7 +43,7 @@
 #define DUAL_CHANNEL		2
 
 static struct snd_soc_jack cz_jack;
-struct clk *da7219_dai_clk;
+static struct clk *da7219_dai_clk;
 
 static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd)
 {

sound/soc/codecs/adau17x1.c

@@ -502,7 +502,7 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	if (adau->sigmadsp) {
-		ret = adau17x1_setup_firmware(adau, params_rate(params));
+		ret = adau17x1_setup_firmware(component, params_rate(params));
 		if (ret < 0)
 			return ret;
 	}
@@ -835,26 +835,40 @@ bool adau17x1_volatile_register(struct device *dev, unsigned int reg)
 }
 EXPORT_SYMBOL_GPL(adau17x1_volatile_register);
 
-int adau17x1_setup_firmware(struct adau *adau, unsigned int rate)
+int adau17x1_setup_firmware(struct snd_soc_component *component,
+	unsigned int rate)
 {
 	int ret;
-	int dspsr;
+	int dspsr, dsp_run;
+	struct adau *adau = snd_soc_component_get_drvdata(component);
+	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+
+	snd_soc_dapm_mutex_lock(dapm);
 
 	ret = regmap_read(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, &dspsr);
 	if (ret)
-		return ret;
+		goto err;
+
+	ret = regmap_read(adau->regmap, ADAU17X1_DSP_RUN, &dsp_run);
+	if (ret)
+		goto err;
 
 	regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 1);
 	regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, 0xf);
+	regmap_write(adau->regmap, ADAU17X1_DSP_RUN, 0);
 
 	ret = sigmadsp_setup(adau->sigmadsp, rate);
 	if (ret) {
 		regmap_write(adau->regmap, ADAU17X1_DSP_ENABLE, 0);
-		return ret;
+		goto err;
 	}
 	regmap_write(adau->regmap, ADAU17X1_DSP_SAMPLING_RATE, dspsr);
+	regmap_write(adau->regmap, ADAU17X1_DSP_RUN, dsp_run);
 
-	return 0;
+err:
+	snd_soc_dapm_mutex_unlock(dapm);
+
+	return ret;
 }
 EXPORT_SYMBOL_GPL(adau17x1_setup_firmware);
 

sound/soc/codecs/adau17x1.h

@@ -68,7 +68,8 @@ int adau17x1_resume(struct snd_soc_component *component);
 
 extern const struct snd_soc_dai_ops adau17x1_dai_ops;
 
-int adau17x1_setup_firmware(struct adau *adau, unsigned int rate);
+int adau17x1_setup_firmware(struct snd_soc_component *component,
+	unsigned int rate);
 bool adau17x1_has_dsp(struct adau *adau);
 
 #define ADAU17X1_CLOCK_CONTROL			0x4000

sound/soc/codecs/msm8916-wcd-analog.c

@@ -1187,7 +1187,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev)
 		return irq;
 	}
 
-	ret = devm_request_irq(dev, irq, pm8916_mbhc_switch_irq_handler,
+	ret = devm_request_threaded_irq(dev, irq, NULL,
+			       pm8916_mbhc_switch_irq_handler,
 			       IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING |
 			       IRQF_ONESHOT,
 			       "mbhc switch irq", priv);
@@ -1201,7 +1202,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev)
 			return irq;
 		}
 
-		ret = devm_request_irq(dev, irq, mbhc_btn_press_irq_handler,
+		ret = devm_request_threaded_irq(dev, irq, NULL,
+				       mbhc_btn_press_irq_handler,
 				       IRQF_TRIGGER_RISING |
 				       IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
 				       "mbhc btn press irq", priv);
@@ -1214,7 +1216,8 @@ static int pm8916_wcd_analog_spmi_probe(struct platform_device *pdev)
 			return irq;
 		}
 
-		ret = devm_request_irq(dev, irq, mbhc_btn_release_irq_handler,
+		ret = devm_request_threaded_irq(dev, irq, NULL,
+				       mbhc_btn_release_irq_handler,
 				       IRQF_TRIGGER_RISING |
 				       IRQF_TRIGGER_FALLING | IRQF_ONESHOT,
 				       "mbhc btn release irq", priv);

sound/soc/codecs/rt5514.c

@@ -89,6 +89,7 @@ static const struct reg_default rt5514_reg[] = {
 	{RT5514_PLL3_CALIB_CTRL5,	0x40220012},
 	{RT5514_DELAY_BUF_CTRL1,	0x7fff006a},
 	{RT5514_DELAY_BUF_CTRL3,	0x00000000},
+	{RT5514_ASRC_IN_CTRL1,		0x00000003},
 	{RT5514_DOWNFILTER0_CTRL1,	0x00020c2f},
 	{RT5514_DOWNFILTER0_CTRL2,	0x00020c2f},
 	{RT5514_DOWNFILTER0_CTRL3,	0x10000362},
@@ -181,6 +182,7 @@ static bool rt5514_readable_register(struct device *dev, unsigned int reg)
 	case RT5514_PLL3_CALIB_CTRL5:
 	case RT5514_DELAY_BUF_CTRL1:
 	case RT5514_DELAY_BUF_CTRL3:
+	case RT5514_ASRC_IN_CTRL1:
 	case RT5514_DOWNFILTER0_CTRL1:
 	case RT5514_DOWNFILTER0_CTRL2:
 	case RT5514_DOWNFILTER0_CTRL3:
@@ -238,6 +240,7 @@ static bool rt5514_i2c_readable_register(struct device *dev,
 	case RT5514_DSP_MAPPING | RT5514_PLL3_CALIB_CTRL5:
 	case RT5514_DSP_MAPPING | RT5514_DELAY_BUF_CTRL1:
 	case RT5514_DSP_MAPPING | RT5514_DELAY_BUF_CTRL3:
+	case RT5514_DSP_MAPPING | RT5514_ASRC_IN_CTRL1:
 	case RT5514_DSP_MAPPING | RT5514_DOWNFILTER0_CTRL1:
 	case RT5514_DSP_MAPPING | RT5514_DOWNFILTER0_CTRL2:
 	case RT5514_DSP_MAPPING | RT5514_DOWNFILTER0_CTRL3:

sound/soc/fsl/fsl_esai.c

@@ -144,6 +144,13 @@ static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, bool tx, u32 ratio,
 
 	psr = ratio <= 256 * maxfp ? ESAI_xCCR_xPSR_BYPASS : ESAI_xCCR_xPSR_DIV8;
 
+	/* Do not loop-search if PM (1 ~ 256) alone can serve the ratio */
+	if (ratio <= 256) {
+		pm = ratio;
+		fp = 1;
+		goto out;
+	}
+
 	/* Set the max fluctuation -- 0.1% of the max devisor */
 	savesub = (psr ? 1 : 8)  * 256 * maxfp / 1000;
 

sound/soc/fsl/fsl_ssi.c

@@ -217,6 +217,7 @@ struct fsl_ssi_soc_data {
  * @dai_fmt: DAI configuration this device is currently used with
  * @streams: Mask of current active streams: BIT(TX) and BIT(RX)
  * @i2s_net: I2S and Network mode configurations of SCR register
+ *           (this is the initial settings based on the DAI format)
  * @synchronous: Use synchronous mode - both of TX and RX use STCK and SFCK
  * @use_dma: DMA is used or FIQ with stream filter
  * @use_dual_fifo: DMA with support for dual FIFO mode
@@ -829,16 +830,23 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	if (!fsl_ssi_is_ac97(ssi)) {
+		/*
+		 * Keep the ssi->i2s_net intact while having a local variable
+		 * to override settings for special use cases. Otherwise, the
+		 * ssi->i2s_net will lose the settings for regular use cases.
+		 */
+		u8 i2s_net = ssi->i2s_net;
+
 		/* Normal + Network mode to send 16-bit data in 32-bit frames */
 		if (fsl_ssi_is_i2s_cbm_cfs(ssi) && sample_size == 16)
-			ssi->i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET;
+			i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET;
 
 		/* Use Normal mode to send mono data at 1st slot of 2 slots */
 		if (channels == 1)
-			ssi->i2s_net = SSI_SCR_I2S_MODE_NORMAL;
+			i2s_net = SSI_SCR_I2S_MODE_NORMAL;
 
 		regmap_update_bits(regs, REG_SSI_SCR,
-				   SSI_SCR_I2S_NET_MASK, ssi->i2s_net);
+				   SSI_SCR_I2S_NET_MASK, i2s_net);
 	}
 
 	/* In synchronous mode, the SSI uses STCCR for capture */

sound/soc/intel/Kconfig

@@ -72,24 +72,28 @@ config SND_SOC_INTEL_BAYTRAIL
 	  for Baytrail Chromebooks but this option is now deprecated and is
 	  not recommended, use SND_SST_ATOM_HIFI2_PLATFORM instead.
 
+config SND_SST_ATOM_HIFI2_PLATFORM
+	tristate
+	select SND_SOC_COMPRESS
+
 config SND_SST_ATOM_HIFI2_PLATFORM_PCI
-	tristate "PCI HiFi2 (Medfield, Merrifield) Platforms"
+	tristate "PCI HiFi2 (Merrifield) Platforms"
 	depends on X86 && PCI
 	select SND_SST_IPC_PCI
-	select SND_SOC_COMPRESS
+	select SND_SST_ATOM_HIFI2_PLATFORM
 	help
-	  If you have a Intel Medfield or Merrifield/Edison platform, then
+	  If you have a Intel Merrifield/Edison platform, then
 	  enable this option by saying Y or m. Distros will typically not
-	  enable this option: Medfield devices are not available to
+	  enable this option: while Merrifield/Edison can run a mainline
-	  developers and while Merrifield/Edison can run a mainline kernel with
+	  kernel with limited functionality it will require a firmware file
-	  limited functionality it will require a firmware file which
+	  which is not in the standard firmware tree
-	  is not in the standard firmware tree
 
-config SND_SST_ATOM_HIFI2_PLATFORM
+config SND_SST_ATOM_HIFI2_PLATFORM_ACPI
 	tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms"
+	default ACPI
 	depends on X86 && ACPI
 	select SND_SST_IPC_ACPI
-	select SND_SOC_COMPRESS
+	select SND_SST_ATOM_HIFI2_PLATFORM
 	select SND_SOC_ACPI_INTEL_MATCH
 	select IOSF_MBI
 	help

sound/soc/omap/omap-dmic.c

@@ -281,7 +281,7 @@ static int omap_dmic_dai_trigger(struct snd_pcm_substream *substream,
 static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id,
 				 unsigned int freq)
 {
-	struct clk *parent_clk;
+	struct clk *parent_clk, *mux;
 	char *parent_clk_name;
 	int ret = 0;
 
@@ -329,14 +329,21 @@ static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id,
 		return -ENODEV;
 	}
 
+	mux = clk_get_parent(dmic->fclk);
+	if (IS_ERR(mux)) {
+		dev_err(dmic->dev, "can't get fck mux parent\n");
+		clk_put(parent_clk);
+		return -ENODEV;
+	}
+
 	mutex_lock(&dmic->mutex);
 	if (dmic->active) {
 		/* disable clock while reparenting */
 		pm_runtime_put_sync(dmic->dev);
-		ret = clk_set_parent(dmic->fclk, parent_clk);
+		ret = clk_set_parent(mux, parent_clk);
 		pm_runtime_get_sync(dmic->dev);
 	} else {
-		ret = clk_set_parent(dmic->fclk, parent_clk);
+		ret = clk_set_parent(mux, parent_clk);
 	}
 	mutex_unlock(&dmic->mutex);
 
@@ -349,6 +356,7 @@ static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id,
 	dmic->fclk_freq = freq;
 
 err_busy:
+	clk_put(mux);
 	clk_put(parent_clk);
 
 	return ret;

sound/soc/sh/rcar/core.c

@@ -1536,7 +1536,7 @@ static int rsnd_remove(struct platform_device *pdev)
 	return ret;
 }
 
-static int rsnd_suspend(struct device *dev)
+static int __maybe_unused rsnd_suspend(struct device *dev)
 {
 	struct rsnd_priv *priv = dev_get_drvdata(dev);
 
@@ -1545,7 +1545,7 @@ static int rsnd_suspend(struct device *dev)
 	return 0;
 }
 
-static int rsnd_resume(struct device *dev)
+static int __maybe_unused rsnd_resume(struct device *dev)
 {
 	struct rsnd_priv *priv = dev_get_drvdata(dev);
 

sound/soc/soc-topology.c

@@ -513,7 +513,7 @@ static void remove_widget(struct snd_soc_component *comp,
 	 */
 	if (dobj->widget.kcontrol_type == SND_SOC_TPLG_TYPE_ENUM) {
 		/* enumerated widget mixer */
-		for (i = 0; i < w->num_kcontrols; i++) {
+		for (i = 0; w->kcontrols != NULL && i < w->num_kcontrols; i++) {
 			struct snd_kcontrol *kcontrol = w->kcontrols[i];
 			struct soc_enum *se =
 				(struct soc_enum *)kcontrol->private_value;
@@ -530,7 +530,7 @@ static void remove_widget(struct snd_soc_component *comp,
 		}
 	} else {
 		/* volume mixer or bytes controls */
-		for (i = 0; i < w->num_kcontrols; i++) {
+		for (i = 0; w->kcontrols != NULL && i < w->num_kcontrols; i++) {
 			struct snd_kcontrol *kcontrol = w->kcontrols[i];
 
 			if (dobj->widget.kcontrol_type
@@ -1325,8 +1325,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create(
 			ec->hdr.name);
 
 		kc[i].name = kstrdup(ec->hdr.name, GFP_KERNEL);
-		if (kc[i].name == NULL)
+		if (kc[i].name == NULL) {
+			kfree(se);
 			goto err_se;
+		}
 		kc[i].private_value = (long)se;
 		kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER;
 		kc[i].access = ec->hdr.access;
@@ -1442,8 +1444,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create(
 			be->hdr.name, be->hdr.access);
 
 		kc[i].name = kstrdup(be->hdr.name, GFP_KERNEL);
-		if (kc[i].name == NULL)
+		if (kc[i].name == NULL) {
+			kfree(sbe);
 			goto err;
+		}
 		kc[i].private_value = (long)sbe;
 		kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER;
 		kc[i].access = be->hdr.access;
@@ -2576,7 +2580,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index)
 
 			/* match index */
 			if (dobj->index != index &&
-				dobj->index != SND_SOC_TPLG_INDEX_ALL)
+				index != SND_SOC_TPLG_INDEX_ALL)
 				continue;
 
 			switch (dobj->type) {

sound/usb/mixer.c

@@ -1776,7 +1776,8 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid,
 				build_feature_ctl(state, _ftr, ch_bits, control,
 						  &iterm, unitid, ch_read_only);
 			if (uac_v2v3_control_is_readable(master_bits, control))
-				build_feature_ctl(state, _ftr, 0, i, &iterm, unitid,
+				build_feature_ctl(state, _ftr, 0, control,
+						  &iterm, unitid,
 						  !uac_v2v3_control_is_writeable(master_bits,
 										 control));
 		}
@@ -1859,7 +1860,7 @@ static int parse_audio_input_terminal(struct mixer_build *state, int unitid,
 	check_input_term(state, d->bTerminalID, &iterm);
 	if (state->mixer->protocol == UAC_VERSION_2) {
 		/* Check for jack detection. */
-		if (uac_v2v3_control_is_readable(d->bmControls,
+		if (uac_v2v3_control_is_readable(le16_to_cpu(d->bmControls),
 						 UAC2_TE_CONNECTOR)) {
 			build_connector_control(state, &iterm, true);
 		}
@@ -2561,7 +2562,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 			if (err < 0 && err != -EINVAL)
 				return err;
 
-			if (uac_v2v3_control_is_readable(desc->bmControls,
+			if (uac_v2v3_control_is_readable(le16_to_cpu(desc->bmControls),
 							 UAC2_TE_CONNECTOR)) {
 				build_connector_control(&state, &state.oterm,
 							false);

sound/usb/mixer_maps.c

@@ -353,8 +353,11 @@ static struct usbmix_name_map bose_companion5_map[] = {
 /*
  * Dell usb dock with ALC4020 codec had a firmware problem where it got
  * screwed up when zero volume is passed; just skip it as a workaround
+ *
+ * Also the extension unit gives an access error, so skip it as well.
  */
 static const struct usbmix_name_map dell_alc4020_map[] = {
+	{ 4, NULL },	/* extension unit */
 	{ 16, NULL },
 	{ 19, NULL },
 	{ 0 }

sound/usb/stream.c

@@ -349,7 +349,7 @@ snd_pcm_chmap_elem *convert_chmap_v3(struct uac3_cluster_header_descriptor
 			 * TODO: this conversion is not complete, update it
 			 * after adding UAC3 values to asound.h
 			 */
-			switch (is->bChPurpose) {
+			switch (is->bChRelationship) {
 			case UAC3_CH_MONO:
 				map = SNDRV_CHMAP_MONO;
 				break;

sound/usb/usx2y/us122l.c

@@ -139,7 +139,7 @@ static void usb_stream_hwdep_vm_open(struct vm_area_struct *area)
 	snd_printdd(KERN_DEBUG "%i\n", atomic_read(&us122l->mmap_count));
 }
 
-static int usb_stream_hwdep_vm_fault(struct vm_fault *vmf)
+static vm_fault_t usb_stream_hwdep_vm_fault(struct vm_fault *vmf)
 {
 	unsigned long offset;
 	struct page *page;

sound/usb/usx2y/usX2Yhwdep.c

@@ -31,7 +31,7 @@
 #include "usbusx2y.h"
 #include "usX2Yhwdep.h"
 
-static int snd_us428ctls_vm_fault(struct vm_fault *vmf)
+static vm_fault_t snd_us428ctls_vm_fault(struct vm_fault *vmf)
 {
 	unsigned long offset;
 	struct page * page;

sound/usb/usx2y/usx2yhwdeppcm.c

@@ -652,7 +652,7 @@ static void snd_usX2Y_hwdep_pcm_vm_close(struct vm_area_struct *area)
 }
 
 
-static int snd_usX2Y_hwdep_pcm_vm_fault(struct vm_fault *vmf)
+static vm_fault_t snd_usX2Y_hwdep_pcm_vm_fault(struct vm_fault *vmf)
 {
 	unsigned long offset;
 	void *vaddr;