Merge branch 'for-next' into for-linus

Merged 4.8 changes.

Documentation/sound/alsa/timestamping.txt

@@ -14,7 +14,7 @@ provides a refined estimate with a delay.
 event or application query.
 The difference (tstamp - trigger_tstamp) defines the elapsed time.
 
-The ALSA API provides reports two basic pieces of information, avail
+The ALSA API provides two basic pieces of information, avail
 and delay, which combined with the trigger and current system
 timestamps allow for applications to keep track of the 'fullness' of
 the ring buffer and the amount of queued samples.
@@ -53,21 +53,21 @@ case):
 The analog time is taken at the last stage of the playback, as close
 as possible to the actual transducer
 
-The link time is taken at the output of the SOC/chipset as the samples
+The link time is taken at the output of the SoC/chipset as the samples
 are pushed on a link. The link time can be directly measured if
 supported in hardware by sample counters or wallclocks (e.g. with
 HDAudio 24MHz or PTP clock for networked solutions) or indirectly
 estimated (e.g. with the frame counter in USB).
 
 The DMA time is measured using counters - typically the least reliable
-of all measurements due to the bursty natured of DMA transfers.
+of all measurements due to the bursty nature of DMA transfers.
 
 The app time corresponds to the time tracked by an application after
 writing in the ring buffer.
 
-The application can query what the hardware supports, define which
+The application can query the hardware capabilities, define which
 audio time it wants reported by selecting the relevant settings in
-audio_tstamp_config fields, get an estimate of the timestamp
+audio_tstamp_config fields, thus get an estimate of the timestamp
 accuracy. It can also request the delay-to-analog be included in the
 measurement. Direct access to the link time is very interesting on
 platforms that provide an embedded DSP; measuring directly the link
@@ -169,7 +169,7 @@ playback: systime: 938107562 nsec, audio time 938112708 nsec, 	systime delta -51
 Example 1 shows that the timestamp at the DMA level is close to 1ms
 ahead of the actual playback time (as a side time this sort of
 measurement can help define rewind safeguards). Compensating for the
-DMA-link delay in example 2 helps remove the hardware buffering abut
+DMA-link delay in example 2 helps remove the hardware buffering but
 the information is still very jittery, with up to one sample of
 error. In example 3 where the timestamps are measured with the link
 wallclock, the timestamps show a monotonic behavior and a lower

sound/core/control.c

@@ -807,6 +807,36 @@ static int snd_ctl_elem_list(struct snd_card *card,
 	return 0;
 }
 
+static bool validate_element_member_dimension(struct snd_ctl_elem_info *info)
+{
+	unsigned int members;
+	unsigned int i;
+
+	if (info->dimen.d[0] == 0)
+		return true;
+
+	members = 1;
+	for (i = 0; i < ARRAY_SIZE(info->dimen.d); ++i) {
+		if (info->dimen.d[i] == 0)
+			break;
+		members *= info->dimen.d[i];
+
+		/*
+		 * info->count should be validated in advance, to guarantee
+		 * calculation soundness.
+		 */
+		if (members > info->count)
+			return false;
+	}
+
+	for (++i; i < ARRAY_SIZE(info->dimen.d); ++i) {
+		if (info->dimen.d[i] > 0)
+			return false;
+	}
+
+	return members == info->count;
+}
+
 static int snd_ctl_elem_info(struct snd_ctl_file *ctl,
 			     struct snd_ctl_elem_info *info)
 {
@@ -1274,6 +1304,8 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file,
 	if (info->count < 1 ||
 	    info->count > max_value_counts[info->type])
 		return -EINVAL;
+	if (!validate_element_member_dimension(info))
+		return -EINVAL;
 	private_size = value_sizes[info->type] * info->count;
 
 	/*

sound/core/seq/oss/seq_oss_synth.c

@@ -70,11 +70,11 @@ struct seq_oss_synth {
 static int max_synth_devs;
 static struct seq_oss_synth *synth_devs[SNDRV_SEQ_OSS_MAX_SYNTH_DEVS];
 static struct seq_oss_synth midi_synth_dev = {
-	-1, /* seq_device */
-	SYNTH_TYPE_MIDI, /* synth_type */
-	0, /* synth_subtype */
-	16, /* nr_voices */
-	"MIDI", /* name */
+	.seq_device = -1,
+	.synth_type = SYNTH_TYPE_MIDI,
+	.synth_subtype = 0,
+	.nr_voices = 16,
+	.name = "MIDI",
 };
 
 static DEFINE_SPINLOCK(register_lock);

sound/core/seq/seq_timer.c

@@ -165,7 +165,7 @@ static void snd_seq_timer_interrupt(struct snd_timer_instance *timeri,
 	snd_seq_timer_update_tick(&tmr->tick, resolution);
 
 	/* register actual time of this timer update */
-	do_gettimeofday(&tmr->last_update);
+	ktime_get_ts64(&tmr->last_update);
 
 	spin_unlock_irqrestore(&tmr->lock, flags);
 
@@ -392,7 +392,7 @@ static int seq_timer_start(struct snd_seq_timer *tmr)
 		return -EINVAL;
 	snd_timer_start(tmr->timeri, tmr->ticks);
 	tmr->running = 1;
-	do_gettimeofday(&tmr->last_update);
+	ktime_get_ts64(&tmr->last_update);
 	return 0;
 }
 
@@ -420,7 +420,7 @@ static int seq_timer_continue(struct snd_seq_timer *tmr)
 	}
 	snd_timer_start(tmr->timeri, tmr->ticks);
 	tmr->running = 1;
-	do_gettimeofday(&tmr->last_update);
+	ktime_get_ts64(&tmr->last_update);
 	return 0;
 }
 
@@ -444,17 +444,12 @@ snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr)
 	spin_lock_irqsave(&tmr->lock, flags);
 	cur_time = tmr->cur_time;
 	if (tmr->running) { 
-		struct timeval tm;
-		int usec;
-		do_gettimeofday(&tm);
-		usec = (int)(tm.tv_usec - tmr->last_update.tv_usec);
-		if (usec < 0) {
-			cur_time.tv_nsec += (1000000 + usec) * 1000;
-			cur_time.tv_sec += tm.tv_sec - tmr->last_update.tv_sec - 1;
-		} else {
-			cur_time.tv_nsec += usec * 1000;
-			cur_time.tv_sec += tm.tv_sec - tmr->last_update.tv_sec;
-		}
+		struct timespec64 tm;
+
+		ktime_get_ts64(&tm);
+		tm = timespec64_sub(tm, tmr->last_update);
+		cur_time.tv_nsec = tm.tv_nsec;
+		cur_time.tv_sec = tm.tv_sec;
 		snd_seq_sanity_real_time(&cur_time);
 	}
 	spin_unlock_irqrestore(&tmr->lock, flags);

sound/core/seq/seq_timer.h

@@ -52,7 +52,7 @@ struct snd_seq_timer {
 	unsigned int skew;
 	unsigned int skew_base;
 
-	struct timeval 		last_update;	 /* time of last clock update, used for interpolation */
+	struct timespec64	last_update;	 /* time of last clock update, used for interpolation */
 
 	spinlock_t lock;
 };

sound/hda/hdmi_chmap.c

@@ -353,7 +353,8 @@ static void hdmi_std_setup_channel_mapping(struct hdac_chmap *chmap,
 		int hdmi_slot = 0;
 		/* fill actual channel mappings in ALSA channel (i) order */
 		for (i = 0; i < ch_alloc->channels; i++) {
-			while (!ch_alloc->speakers[7 - hdmi_slot] && !WARN_ON(hdmi_slot >= 8))
+			while (!WARN_ON(hdmi_slot >= 8) &&
+			       !ch_alloc->speakers[7 - hdmi_slot])
 				hdmi_slot++; /* skip zero slots */
 
 			hdmi_channel_mapping[ca][i] = (i << 4) | hdmi_slot++;
@@ -430,6 +431,12 @@ static int to_cea_slot(int ordered_ca, unsigned char pos)
 	int mask = snd_hdac_chmap_to_spk_mask(pos);
 	int i;
 
+	/* Add sanity check to pass klockwork check.
+	 * This should never happen.
+	 */
+	if (ordered_ca >= ARRAY_SIZE(channel_allocations))
+		return -1;
+
 	if (mask) {
 		for (i = 0; i < 8; i++) {
 			if (channel_allocations[ordered_ca].speakers[7 - i] == mask)
@@ -456,7 +463,15 @@ EXPORT_SYMBOL_GPL(snd_hdac_spk_to_chmap);
 /* from CEA slot to ALSA API channel position */
 static int from_cea_slot(int ordered_ca, unsigned char slot)
 {
-	int mask = channel_allocations[ordered_ca].speakers[7 - slot];
+	int mask;
+
+	/* Add sanity check to pass klockwork check.
+	 * This should never happen.
+	 */
+	if (slot >= 8)
+		return 0;
+
+	mask = channel_allocations[ordered_ca].speakers[7 - slot];
 
 	return snd_hdac_spk_to_chmap(mask);
 }
@@ -523,7 +538,8 @@ static void hdmi_setup_fake_chmap(unsigned char *map, int ca)
 	int ordered_ca = get_channel_allocation_order(ca);
 
 	for (i = 0; i < 8; i++) {
-		if (i < channel_allocations[ordered_ca].channels)
+		if (ordered_ca < ARRAY_SIZE(channel_allocations) &&
+		    i < channel_allocations[ordered_ca].channels)
 			map[i] = from_cea_slot(ordered_ca, hdmi_channel_mapping[ca][i] & 0x0f);
 		else
 			map[i] = 0;
@@ -551,6 +567,12 @@ int snd_hdac_get_active_channels(int ca)
 {
 	int ordered_ca = get_channel_allocation_order(ca);
 
+	/* Add sanity check to pass klockwork check.
+	 * This should never happen.
+	 */
+	if (ordered_ca >= ARRAY_SIZE(channel_allocations))
+		ordered_ca = 0;
+
 	return channel_allocations[ordered_ca].channels;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_get_active_channels);

sound/i2c/other/ak4114.c

@@ -121,7 +121,7 @@ int snd_ak4114_create(struct snd_card *card,
 
       __fail:
 	snd_ak4114_free(chip);
-	return err < 0 ? err : -EIO;
+	return err;
 }
 EXPORT_SYMBOL(snd_ak4114_create);
 

sound/i2c/other/ak4117.c

@@ -110,7 +110,7 @@ int snd_ak4117_create(struct snd_card *card, ak4117_read_t *read, ak4117_write_t
 
       __fail:
 	snd_ak4117_free(chip);
-	return err < 0 ? err : -EIO;
+	return err;
 }
 
 void snd_ak4117_reg_write(struct ak4117 *chip, unsigned char reg, unsigned char mask, unsigned char val)

sound/isa/ad1848/ad1848.c

@@ -170,15 +170,4 @@ static struct isa_driver snd_ad1848_driver = {
 	}
 };
 
-static int __init alsa_card_ad1848_init(void)
-{
-	return isa_register_driver(&snd_ad1848_driver, SNDRV_CARDS);
-}
-
-static void __exit alsa_card_ad1848_exit(void)
-{
-	isa_unregister_driver(&snd_ad1848_driver);
-}
-
-module_init(alsa_card_ad1848_init);
-module_exit(alsa_card_ad1848_exit);
+module_isa_driver(snd_ad1848_driver, SNDRV_CARDS);

sound/isa/adlib.c

@@ -112,15 +112,4 @@ static struct isa_driver snd_adlib_driver = {
 	}
 };
 
-static int __init alsa_card_adlib_init(void)
-{
-	return isa_register_driver(&snd_adlib_driver, SNDRV_CARDS);
-}
-
-static void __exit alsa_card_adlib_exit(void)
-{
-	isa_unregister_driver(&snd_adlib_driver);
-}
-
-module_init(alsa_card_adlib_init);
-module_exit(alsa_card_adlib_exit);
+module_isa_driver(snd_adlib_driver, SNDRV_CARDS);

sound/isa/cmi8328.c

@@ -469,15 +469,4 @@ static struct isa_driver snd_cmi8328_driver = {
 	},
 };
 
-static int __init alsa_card_cmi8328_init(void)
-{
-	return isa_register_driver(&snd_cmi8328_driver, CMI8328_MAX);
-}
-
-static void __exit alsa_card_cmi8328_exit(void)
-{
-	isa_unregister_driver(&snd_cmi8328_driver);
-}
-
-module_init(alsa_card_cmi8328_init)
-module_exit(alsa_card_cmi8328_exit)
+module_isa_driver(snd_cmi8328_driver, CMI8328_MAX);

sound/isa/cs423x/cs4231.c

@@ -186,15 +186,4 @@ static struct isa_driver snd_cs4231_driver = {
 	}
 };
 
-static int __init alsa_card_cs4231_init(void)
-{
-	return isa_register_driver(&snd_cs4231_driver, SNDRV_CARDS);
-}
-
-static void __exit alsa_card_cs4231_exit(void)
-{
-	isa_unregister_driver(&snd_cs4231_driver);
-}
-
-module_init(alsa_card_cs4231_init);
-module_exit(alsa_card_cs4231_exit);
+module_isa_driver(snd_cs4231_driver, SNDRV_CARDS);

sound/isa/galaxy/galaxy.c

@@ -634,15 +634,4 @@ static struct isa_driver snd_galaxy_driver = {
 	}
 };
 
-static int __init alsa_card_galaxy_init(void)
-{
-	return isa_register_driver(&snd_galaxy_driver, SNDRV_CARDS);
-}
-
-static void __exit alsa_card_galaxy_exit(void)
-{
-	isa_unregister_driver(&snd_galaxy_driver);
-}
-
-module_init(alsa_card_galaxy_init);
-module_exit(alsa_card_galaxy_exit);
+module_isa_driver(snd_galaxy_driver, SNDRV_CARDS);

sound/isa/gus/gusclassic.c

@@ -229,15 +229,4 @@ static struct isa_driver snd_gusclassic_driver = {
 	}
 };
 
-static int __init alsa_card_gusclassic_init(void)
-{
-	return isa_register_driver(&snd_gusclassic_driver, SNDRV_CARDS);
-}
-
-static void __exit alsa_card_gusclassic_exit(void)
-{
-	isa_unregister_driver(&snd_gusclassic_driver);
-}
-
-module_init(alsa_card_gusclassic_init);
-module_exit(alsa_card_gusclassic_exit);
+module_isa_driver(snd_gusclassic_driver, SNDRV_CARDS);

sound/isa/gus/gusextreme.c

@@ -358,15 +358,4 @@ static struct isa_driver snd_gusextreme_driver = {
 	}
 };
 
-static int __init alsa_card_gusextreme_init(void)
-{
-	return isa_register_driver(&snd_gusextreme_driver, SNDRV_CARDS);
-}
-
-static void __exit alsa_card_gusextreme_exit(void)
-{
-	isa_unregister_driver(&snd_gusextreme_driver);
-}
-
-module_init(alsa_card_gusextreme_init);
-module_exit(alsa_card_gusextreme_exit);
+module_isa_driver(snd_gusextreme_driver, SNDRV_CARDS);

sound/isa/gus/gusmax.c

@@ -370,15 +370,4 @@ static struct isa_driver snd_gusmax_driver = {
 	},
 };
 
-static int __init alsa_card_gusmax_init(void)
-{
-	return isa_register_driver(&snd_gusmax_driver, SNDRV_CARDS);
-}
-
-static void __exit alsa_card_gusmax_exit(void)
-{
-	isa_unregister_driver(&snd_gusmax_driver);
-}
-
-module_init(alsa_card_gusmax_init)
-module_exit(alsa_card_gusmax_exit)
+module_isa_driver(snd_gusmax_driver, SNDRV_CARDS);

sound/isa/sb/jazz16.c

@@ -387,15 +387,4 @@ static struct isa_driver snd_jazz16_driver = {
 	},
 };
 
-static int __init alsa_card_jazz16_init(void)
-{
-	return isa_register_driver(&snd_jazz16_driver, SNDRV_CARDS);
-}
-
-static void __exit alsa_card_jazz16_exit(void)
-{
-	isa_unregister_driver(&snd_jazz16_driver);
-}
-
-module_init(alsa_card_jazz16_init)
-module_exit(alsa_card_jazz16_exit)
+module_isa_driver(snd_jazz16_driver, SNDRV_CARDS);

sound/isa/sb/sb8.c

@@ -251,15 +251,4 @@ static struct isa_driver snd_sb8_driver = {
 	},
 };
 
-static int __init alsa_card_sb8_init(void)
-{
-	return isa_register_driver(&snd_sb8_driver, SNDRV_CARDS);
-}
-
-static void __exit alsa_card_sb8_exit(void)
-{
-	isa_unregister_driver(&snd_sb8_driver);
-}
-
-module_init(alsa_card_sb8_init)
-module_exit(alsa_card_sb8_exit)
+module_isa_driver(snd_sb8_driver, SNDRV_CARDS);

sound/isa/sc6000.c

@@ -711,15 +711,4 @@ static struct isa_driver snd_sc6000_driver = {
 };
 
 
-static int __init alsa_card_sc6000_init(void)
-{
-	return isa_register_driver(&snd_sc6000_driver, SNDRV_CARDS);
-}
-
-static void __exit alsa_card_sc6000_exit(void)
-{
-	isa_unregister_driver(&snd_sc6000_driver);
-}
-
-module_init(alsa_card_sc6000_init)
-module_exit(alsa_card_sc6000_exit)
+module_isa_driver(snd_sc6000_driver, SNDRV_CARDS);

sound/oss/ad1848.c

@@ -254,7 +254,7 @@ static void ad_write(ad1848_info * devc, int reg, int data)
 
 static void wait_for_calibration(ad1848_info * devc)
 {
-	int timeout = 0;
+	int timeout;
 
 	/*
 	 * Wait until the auto calibration process has finished.

sound/oss/aedsp16.c

@@ -482,13 +482,13 @@ static struct orVals orDMA[] __initdata = {
 };
 
 static struct aedsp16_info ae_config = {
-	DEF_AEDSP16_IOB,
-	DEF_AEDSP16_IRQ,
-	DEF_AEDSP16_MRQ,
-	DEF_AEDSP16_DMA,
-	-1,
-	-1,
-	INIT_NONE
+	.base_io = DEF_AEDSP16_IOB,
+	.irq = DEF_AEDSP16_IRQ,
+	.mpu_irq = DEF_AEDSP16_MRQ,
+	.dma = DEF_AEDSP16_DMA,
+	.mss_base = -1,
+	.mpu_base = -1,
+	.init = INIT_NONE
 };
 
 /*

sound/oss/sound_timer.c

@@ -17,7 +17,7 @@
 #include "sound_config.h"
 
 static volatile int initialized, opened, tmr_running;
-static volatile time_t tmr_offs, tmr_ctr;
+static volatile unsigned int tmr_offs, tmr_ctr;
 static volatile unsigned long ticks_offs;
 static volatile int curr_tempo, curr_timebase;
 static volatile unsigned long curr_ticks;

sound/oss/sys_timer.c

@@ -19,7 +19,7 @@
 #include "sound_config.h"
 
 static volatile int opened, tmr_running;
-static volatile time_t tmr_offs, tmr_ctr;
+static volatile unsigned int tmr_offs, tmr_ctr;
 static volatile unsigned long ticks_offs;
 static volatile int curr_tempo, curr_timebase;
 static volatile unsigned long curr_ticks;

sound/pci/ctxfi/cthw20k2.c

@@ -1615,23 +1615,23 @@ static int hw_dac_init(struct hw *hw, const struct dac_conf *info)
 	int i;
 	struct regs_cs4382 cs_read = {0};
 	struct regs_cs4382 cs_def = {
-				   0x00000001,  /* Mode Control 1 */
-				   0x00000000,  /* Mode Control 2 */
-				   0x00000084,  /* Mode Control 3 */
-				   0x00000000,  /* Filter Control */
-				   0x00000000,  /* Invert Control */
-				   0x00000024,  /* Mixing Control Pair 1 */
-				   0x00000000,  /* Vol Control A1 */
-				   0x00000000,  /* Vol Control B1 */
-				   0x00000024,  /* Mixing Control Pair 2 */
-				   0x00000000,  /* Vol Control A2 */
-				   0x00000000,  /* Vol Control B2 */
-				   0x00000024,  /* Mixing Control Pair 3 */
-				   0x00000000,  /* Vol Control A3 */
-				   0x00000000,  /* Vol Control B3 */
-				   0x00000024,  /* Mixing Control Pair 4 */
-				   0x00000000,  /* Vol Control A4 */
-				   0x00000000   /* Vol Control B4 */
+		.mode_control_1 = 0x00000001, /* Mode Control 1 */
+		.mode_control_2 = 0x00000000, /* Mode Control 2 */
+		.mode_control_3 = 0x00000084, /* Mode Control 3 */
+		.filter_control = 0x00000000, /* Filter Control */
+		.invert_control = 0x00000000, /* Invert Control */
+		.mix_control_P1 = 0x00000024, /* Mixing Control Pair 1 */
+		.vol_control_A1 = 0x00000000, /* Vol Control A1 */
+		.vol_control_B1 = 0x00000000, /* Vol Control B1 */
+		.mix_control_P2 = 0x00000024, /* Mixing Control Pair 2 */
+		.vol_control_A2 = 0x00000000, /* Vol Control A2 */
+		.vol_control_B2 = 0x00000000, /* Vol Control B2 */
+		.mix_control_P3 = 0x00000024, /* Mixing Control Pair 3 */
+		.vol_control_A3 = 0x00000000, /* Vol Control A3 */
+		.vol_control_B3 = 0x00000000, /* Vol Control B3 */
+		.mix_control_P4 = 0x00000024, /* Mixing Control Pair 4 */
+		.vol_control_A4 = 0x00000000, /* Vol Control A4 */
+		.vol_control_B4 = 0x00000000  /* Vol Control B4 */
 				 };
 
 	if (hw->model == CTSB1270) {

sound/pci/echoaudio/echoaudio.c

@@ -1272,11 +1272,11 @@ static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol,
 
 	chip = snd_kcontrol_chip(kcontrol);
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
-	uinfo->count = 1;
 	uinfo->value.integer.min = ECHOGAIN_MINOUT;
 	uinfo->value.integer.max = ECHOGAIN_MAXOUT;
 	uinfo->dimen.d[0] = num_busses_out(chip);
 	uinfo->dimen.d[1] = num_busses_in(chip);
+	uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1];
 	return 0;
 }
 
@@ -1344,11 +1344,11 @@ static int snd_echo_vmixer_info(struct snd_kcontrol *kcontrol,
 
 	chip = snd_kcontrol_chip(kcontrol);
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
-	uinfo->count = 1;
 	uinfo->value.integer.min = ECHOGAIN_MINOUT;
 	uinfo->value.integer.max = ECHOGAIN_MAXOUT;
 	uinfo->dimen.d[0] = num_busses_out(chip);
 	uinfo->dimen.d[1] = num_pipes_out(chip);
+	uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1];
 	return 0;
 }
 
@@ -1728,7 +1728,6 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_info *uinfo)
 {
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
-	uinfo->count = 96;
 	uinfo->value.integer.min = ECHOGAIN_MINOUT;
 	uinfo->value.integer.max = 0;
 #ifdef ECHOCARD_HAS_VMIXER
@@ -1738,6 +1737,7 @@ static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol,
 #endif
 	uinfo->dimen.d[1] = 16;	/* 16 channels */
 	uinfo->dimen.d[2] = 2;	/* 0=level, 1=peak */
+	uinfo->count = uinfo->dimen.d[0] * uinfo->dimen.d[1] * uinfo->dimen.d[2];
 	return 0;
 }
 

sound/pci/hda/hda_codec.c

@@ -3584,6 +3584,12 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid,
 	bool reset;
 
 	spdif = snd_hda_spdif_out_of_nid(codec, nid);
+	/* Add sanity check to pass klockwork check.
+	 * This should never happen.
+	 */
+	if (WARN_ON(spdif == NULL))
+		return;
+
 	curr_fmt = snd_hda_codec_read(codec, nid, 0,
 				      AC_VERB_GET_STREAM_FORMAT, 0);
 	reset = codec->spdif_status_reset &&
@@ -3768,7 +3774,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec,
 	spdif = snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid);
 	if (mout->dig_out_nid && mout->share_spdif &&
 	    mout->dig_out_used != HDA_DIG_EXCLUSIVE) {
-		if (chs == 2 &&
+		if (chs == 2 && spdif != NULL &&
 		    snd_hda_is_supported_format(codec, mout->dig_out_nid,
 						format) &&
 		    !(spdif->status & IEC958_AES0_NONAUDIO)) {

sound/pci/hda/hda_generic.c

@@ -2492,10 +2492,6 @@ static int create_loopback_mixing_ctl(struct hda_codec *codec)
 	if (!snd_hda_gen_add_kctl(spec, NULL, &loopback_mixing_enum))
 		return -ENOMEM;
 	spec->have_aamix_ctl = 1;
-	/* if no explicit aamix path is present (e.g. for Realtek codecs),
-	 * enable aamix as default -- just for compatibility
-	 */
-	spec->aamix_mode = !has_aamix_out_paths(spec);
 	return 0;
 }
 

sound/pci/hda/patch_hdmi.c

@@ -1680,6 +1680,11 @@ static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
 
 	mutex_lock(&codec->spdif_mutex);
 	spdif = snd_hda_spdif_out_of_nid(codec, cvt_nid);
+	/* Add sanity check to pass klockwork check.
+	 * This should never happen.
+	 */
+	if (WARN_ON(spdif == NULL))
+		return true;
 	non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO);
 	mutex_unlock(&codec->spdif_mutex);
 	return non_pcm;

sound/pci/hda/patch_realtek.c

@@ -3718,6 +3718,9 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec)
 	case 0x10ec0295:
 		alc_process_coef_fw(codec, coef0225);
 		break;
+	case 0x10ec0867:
+		alc_update_coefex_idx(codec, 0x57, 0x5, 1<<14, 0);
+		break;
 	}
 	codec_dbg(codec, "Headset jack set to unplugged mode.\n");
 }
@@ -3805,6 +3808,9 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
 		alc_process_coef_fw(codec, coef0293);
 		snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50);
 		break;
+	case 0x10ec0867:
+		alc_update_coefex_idx(codec, 0x57, 0x5, 0, 1<<14);
+		/* fallthru */
 	case 0x10ec0662:
 		snd_hda_set_pin_ctl_cache(codec, hp_pin, 0);
 		snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50);
@@ -3899,6 +3905,9 @@ static void alc_headset_mode_default(struct hda_codec *codec)
 	case 0x10ec0668:
 		alc_process_coef_fw(codec, coef0688);
 		break;
+	case 0x10ec0867:
+		alc_update_coefex_idx(codec, 0x57, 0x5, 1<<14, 0);
+		break;
 	}
 	codec_dbg(codec, "Headset jack set to headphone (default) mode.\n");
 }
@@ -3989,6 +3998,9 @@ static void alc_headset_mode_ctia(struct hda_codec *codec)
 	case 0x10ec0295:
 		alc_process_coef_fw(codec, coef0225);
 		break;
+	case 0x10ec0867:
+		alc_update_coefex_idx(codec, 0x57, 0x5, 1<<14, 0);
+		break;
 	}
 	codec_dbg(codec, "Headset jack set to iPhone-style headset mode.\n");
 }
@@ -4166,6 +4178,9 @@ static void alc_determine_headset_type(struct hda_codec *codec)
 		val = alc_read_coef_idx(codec, 0x46);
 		is_ctia = (val & 0x00f0) == 0x00f0;
 		break;
+	case 0x10ec0867:
+		is_ctia = true;
+		break;
 	}
 
 	codec_dbg(codec, "Headset jack detected iPhone-style headset: %s\n",
@@ -6532,6 +6547,8 @@ enum {
 	ALC668_FIXUP_DELL_XPS13,
 	ALC662_FIXUP_ASUS_Nx50,
 	ALC668_FIXUP_ASUS_Nx51,
+	ALC891_FIXUP_HEADSET_MODE,
+	ALC891_FIXUP_DELL_MIC_NO_PRESENCE,
 };
 
 static const struct hda_fixup alc662_fixups[] = {
@@ -6787,6 +6804,20 @@ static const struct hda_fixup alc662_fixups[] = {
 		.chained = true,
 		.chain_id = ALC662_FIXUP_BASS_CHMAP,
 	},
+	[ALC891_FIXUP_HEADSET_MODE] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_headset_mode,
+	},
+	[ALC891_FIXUP_DELL_MIC_NO_PRESENCE] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x03a1913d }, /* use as headphone mic, without its own jack detect */
+			{ 0x1b, 0x03a1113c }, /* use as headset mic, without its own jack detect */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC891_FIXUP_HEADSET_MODE
+	},
 };
 
 static const struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -6903,6 +6934,11 @@ static const struct hda_model_fixup alc662_fixup_models[] = {
 };
 
 static const struct snd_hda_pin_quirk alc662_pin_fixup_tbl[] = {
+	SND_HDA_PIN_QUIRK(0x10ec0867, 0x1028, "Dell", ALC891_FIXUP_DELL_MIC_NO_PRESENCE,
+		{0x17, 0x02211010},
+		{0x18, 0x01a19030},
+		{0x1a, 0x01813040},
+		{0x21, 0x01014020}),
 	SND_HDA_PIN_QUIRK(0x10ec0662, 0x1028, "Dell", ALC662_FIXUP_DELL_MIC_NO_PRESENCE,
 		{0x14, 0x01014010},
 		{0x18, 0x01a19020},
@@ -7091,7 +7127,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
 	HDA_CODEC_ENTRY(0x10ec0700, "ALC700", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0701, "ALC701", patch_alc269),
 	HDA_CODEC_ENTRY(0x10ec0703, "ALC703", patch_alc269),
-	HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc882),
+	HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc662),
 	HDA_CODEC_ENTRY(0x10ec0880, "ALC880", patch_alc880),
 	HDA_CODEC_ENTRY(0x10ec0882, "ALC882", patch_alc882),
 	HDA_CODEC_ENTRY(0x10ec0883, "ALC883", patch_alc882),

sound/pci/mixart/mixart_mixer.c

@@ -965,7 +965,7 @@ static int mixart_update_monitoring(struct snd_mixart* chip, int channel)
 	int err;
 	struct mixart_msg request;
 	struct mixart_set_out_audio_level audio_level;
-	u32 resp;
+	u32 resp = 0;
 
 	if(chip->pipe_out_ana.status == PIPE_UNDEFINED)
 		return -EINVAL; /* no pipe defined */

sound/pci/riptide/riptide.c

@@ -1496,7 +1496,7 @@ static int snd_riptide_prepare(struct snd_pcm_substream *substream)
 		f = PAGE_SIZE;
 		while ((size + (f >> 1) - 1) <= (f << 7) && (f << 1) > period)
 			f = f >> 1;
-		pages = (size + f - 1) / f;
+		pages = DIV_ROUND_UP(size, f);
 		data->size = size;
 		data->pages = pages;
 		snd_printdd

sound/ppc/awacs.c

@@ -991,6 +991,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
 		if (err < 0)
 			return err;
 	}
+	master_vol = NULL;
 	if (pm7500)
 		err = build_mixers(chip,
 				   ARRAY_SIZE(snd_pmac_awacs_mixers_pmac7500),

sound/sh/aica.c

@@ -63,9 +63,6 @@ MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard.");
 module_param(enable, bool, 0644);
 MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard.");
 
-/* Use workqueue */
-static struct workqueue_struct *aica_queue;
-
 /* Simple platform device */
 static struct platform_device *pd;
 static struct resource aica_memory_space[2] = {
@@ -327,7 +324,7 @@ static void aica_period_elapsed(unsigned long timer_var)
 		dreamcastcard->current_period = play_period;
 	if (unlikely(dreamcastcard->dma_check == 0))
 		dreamcastcard->dma_check = 1;
-	queue_work(aica_queue, &(dreamcastcard->spu_dma_work));
+	schedule_work(&(dreamcastcard->spu_dma_work));
 }
 
 static void spu_begin_dma(struct snd_pcm_substream *substream)
@@ -337,7 +334,7 @@ static void spu_begin_dma(struct snd_pcm_substream *substream)
 	runtime = substream->runtime;
 	dreamcastcard = substream->pcm->private_data;
 	/*get the queue to do the work */
-	queue_work(aica_queue, &(dreamcastcard->spu_dma_work));
+	schedule_work(&(dreamcastcard->spu_dma_work));
 	/* Timer may already be running */
 	if (unlikely(dreamcastcard->timer.data)) {
 		mod_timer(&dreamcastcard->timer, jiffies + 4);
@@ -381,7 +378,7 @@ static int snd_aicapcm_pcm_close(struct snd_pcm_substream
 				 *substream)
 {
 	struct snd_card_aica *dreamcastcard = substream->pcm->private_data;
-	flush_workqueue(aica_queue);
+	flush_work(&(dreamcastcard->spu_dma_work));
 	if (dreamcastcard->timer.data)
 		del_timer(&dreamcastcard->timer);
 	kfree(dreamcastcard->channel);
@@ -633,9 +630,6 @@ static int snd_aica_probe(struct platform_device *devptr)
 	if (unlikely(err < 0))
 		goto freedreamcast;
 	platform_set_drvdata(devptr, dreamcastcard);
-	aica_queue = create_workqueue(CARD_NAME);
-	if (unlikely(!aica_queue))
-		goto freedreamcast;
 	snd_printk
 	    ("ALSA Driver for Yamaha AICA Super Intelligent Sound Processor\n");
 	return 0;
@@ -671,10 +665,6 @@ static int __init aica_init(void)
 
 static void __exit aica_exit(void)
 {
-	/* Destroy the aica kernel thread            *
-	 * being extra cautious to check if it exists*/
-	if (likely(aica_queue))
-		destroy_workqueue(aica_queue);
 	platform_device_unregister(pd);
 	platform_driver_unregister(&snd_aica_driver);
 	/* Kill any sound still playing and reset ARM7 to safe state */

sound/usb/mixer_maps.c

@@ -107,8 +107,10 @@ static struct usbmix_name_map extigy_map[] = {
  * e.g. no Master and fake PCM volume
  *			Pavel Mihaylov <bin@bash.info>
  */
-static struct usbmix_dB_map mp3plus_dB_1 = {-4781, 0};	/* just guess */
-static struct usbmix_dB_map mp3plus_dB_2 = {-1781, 618}; /* just guess */
+static struct usbmix_dB_map mp3plus_dB_1 = {.min = -4781, .max = 0};
+						/* just guess */
+static struct usbmix_dB_map mp3plus_dB_2 = {.min = -1781, .max = 618};
+						/* just guess */
 
 static struct usbmix_name_map mp3plus_map[] = {
 	/* 1: IT pcm */