Merge tag 'v4.4.177' into toradex_vf_4.4-nextColibri-VF_LXDE-Image_2.8b6.183-20190331

This is the 4.4.177 stable release

31 files changed, 184 insertions, 51 deletions

diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 6163bf3e8177..2272aee12871 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -500,7 +500,8 @@ static int snd_compress_check_input(struct snd_compr_params *params)
 {
 	/* first let's check the buffer parameter's */
 	if (params->buffer.fragment_size == 0 ||
-	    params->buffer.fragments > INT_MAX / params->buffer.fragment_size)
+	    params->buffer.fragments > INT_MAX / params->buffer.fragment_size ||
+	    params->buffer.fragments == 0)
 		return -EINVAL;
 
 	/* now codec parameters */
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 6bda8f6c5f84..cdff5f976480 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -25,6 +25,7 @@
 #include <linux/time.h>
 #include <linux/mutex.h>
 #include <linux/device.h>
+#include <linux/nospec.h>
 #include <sound/core.h>
 #include <sound/minors.h>
 #include <sound/pcm.h>
@@ -125,6 +126,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card,
 				return -EFAULT;
 			if (stream < 0 || stream > 1)
 				return -EINVAL;
+			stream = array_index_nospec(stream, 2);
 			if (get_user(subdevice, &info->subdevice))
 				return -EFAULT;
 			mutex_lock(&register_mutex);
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 5bc7ddf8fc70..3ce2b8771762 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1849,8 +1849,6 @@ int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream,
 		      unsigned int cmd, void *arg)
 {
 	switch (cmd) {
-	case SNDRV_PCM_IOCTL1_INFO:
-		return 0;
 	case SNDRV_PCM_IOCTL1_RESET:
 		return snd_pcm_lib_ioctl_reset(substream, arg);
 	case SNDRV_PCM_IOCTL1_CHANNEL_INFO:
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 0ad194002c0c..9b6dcdea4431 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -214,11 +214,7 @@ int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info)
 	info->subdevices_avail = pstr->substream_count - pstr->substream_opened;
 	strlcpy(info->subname, substream->name, sizeof(info->subname));
 	runtime = substream->runtime;
-	/* AB: FIXME!!! This is definitely nonsense */
-	if (runtime) {
-		info->sync = runtime->sync;
-		substream->ops->ioctl(substream, SNDRV_PCM_IOCTL1_INFO, info);
-	}
+
 	return 0;
 }
 
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 091290d1f3ea..3a0361458597 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -382,7 +382,7 @@ static const struct ieee1394_device_id bebob_id_table[] = {
 	/* Apogee Electronics, DA/AD/DD-16X (X-FireWire card) */
 	SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00010048, &spec_normal),
 	/* Apogee Electronics, Ensemble */
-	SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x00001eee, &spec_normal),
+	SND_BEBOB_DEV_ENTRY(VEN_APOGEE, 0x01eeee, &spec_normal),
 	/* ESI, Quatafire610 */
 	SND_BEBOB_DEV_ENTRY(VEN_ESI, 0x00010064, &spec_normal),
 	/* AcousticReality, eARMasterOne */
@@ -422,7 +422,19 @@ static const struct ieee1394_device_id bebob_id_table[] = {
 	/* Focusrite, SaffirePro 26 I/O */
 	SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000003, &saffirepro_26_spec),
 	/* Focusrite, SaffirePro 10 I/O */
-	SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000006, &saffirepro_10_spec),
+	{
+		// The combination of vendor_id and model_id is the same as the
+		// same as the one of Liquid Saffire 56.
+		.match_flags	= IEEE1394_MATCH_VENDOR_ID |
+				  IEEE1394_MATCH_MODEL_ID |
+				  IEEE1394_MATCH_SPECIFIER_ID |
+				  IEEE1394_MATCH_VERSION,
+		.vendor_id	= VEN_FOCUSRITE,
+		.model_id	= 0x000006,
+		.specifier_id	= 0x00a02d,
+		.version	= 0x010001,
+		.driver_data	= (kernel_ulong_t)&saffirepro_10_spec,
+	},
 	/* Focusrite, Saffire(no label and LE) */
 	SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, MODEL_FOCUSRITE_SAFFIRE_BOTH,
 			    &saffire_spec),
diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c
index 69f76ff5693d..718d5e3b7806 100644
--- a/sound/isa/wavefront/wavefront_synth.c
+++ b/sound/isa/wavefront/wavefront_synth.c
@@ -785,6 +785,9 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header)
 	DPRINT (WF_DEBUG_LOAD_PATCH, "downloading patch %d\n",
 				      header->number);
 
+	if (header->number >= ARRAY_SIZE(dev->patch_status))
+		return -EINVAL;
+
 	dev->patch_status[header->number] |= WF_SLOT_FILLED;
 
 	bptr = buf;
@@ -809,6 +812,9 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header)
 	DPRINT (WF_DEBUG_LOAD_PATCH, "downloading program %d\n",
 		header->number);
 
+	if (header->number >= ARRAY_SIZE(dev->prog_status))
+		return -EINVAL;
+
 	dev->prog_status[header->number] = WF_SLOT_USED;
 
 	/* XXX need to zero existing SLOT_USED bit for program_status[i]
@@ -898,6 +904,9 @@ wavefront_send_sample (snd_wavefront_t *dev,
 		header->number = x;
 	}
 
+	if (header->number >= WF_MAX_SAMPLE)
+		return -EINVAL;
+
 	if (header->size) {
 
 		/* XXX it's a debatable point whether or not RDONLY semantics
diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c
index d2951ed4bf71..1984291ebd07 100644
--- a/sound/pci/cs46xx/dsp_spos.c
+++ b/sound/pci/cs46xx/dsp_spos.c
@@ -899,6 +899,9 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip)
 	struct dsp_spos_instance * ins = chip->dsp_spos_instance;
 	int i;
 
+	if (!ins)
+		return 0;
+
 	snd_info_free_entry(ins->proc_sym_info_entry);
 	ins->proc_sym_info_entry = NULL;
 
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index 50b216fc369f..5d422d65e62b 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -36,6 +36,7 @@
 #include <linux/init.h>
 #include <linux/mutex.h>
 #include <linux/moduleparam.h>
+#include <linux/nospec.h>
 
 #include <sound/core.h>
 #include <sound/tlv.h>
@@ -1000,6 +1001,8 @@ static int snd_emu10k1_ipcm_poke(struct snd_emu10k1 *emu,
 
 	if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT)
 		return -EINVAL;
+	ipcm->substream = array_index_nospec(ipcm->substream,
+					     EMU10K1_FX8010_PCM_COUNT);
 	if (ipcm->channels > 32)
 		return -EINVAL;
 	pcm = &emu->fx8010.pcm[ipcm->substream];
@@ -1046,6 +1049,8 @@ static int snd_emu10k1_ipcm_peek(struct snd_emu10k1 *emu,
 
 	if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT)
 		return -EINVAL;
+	ipcm->substream = array_index_nospec(ipcm->substream,
+					     EMU10K1_FX8010_PCM_COUNT);
 	pcm = &emu->fx8010.pcm[ipcm->substream];
 	mutex_lock(&emu->fx8010.lock);
 	spin_lock_irq(&emu->reg_lock);
diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index 6efadbfb3fe3..7ea201c05e5d 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -109,7 +109,8 @@ static int hda_codec_driver_probe(struct device *dev)
 	err = snd_hda_codec_build_controls(codec);
 	if (err < 0)
 		goto error_module;
-	if (codec->card->registered) {
+	/* only register after the bus probe finished; otherwise it's racy */
+	if (!codec->bus->bus_probing && codec->card->registered) {
 		err = snd_card_register(codec->card);
 		if (err < 0)
 			goto error_module;
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 776dffa88aee..171e11be938d 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -68,6 +68,7 @@ struct hda_bus {
 	unsigned int response_reset:1;	/* controller was reset */
 	unsigned int in_reset:1;	/* during reset operation */
 	unsigned int no_response_fallback:1; /* don't fallback at RIRB error */
+	unsigned int bus_probing :1;	/* during probing process */
 
 	int primary_dig_out_type;	/* primary digital out PCM type */
 	unsigned int mixer_assigned;	/* codec addr for mixer name */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index f964743b104c..74c9600876d6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2100,6 +2100,7 @@ static int azx_probe_continue(struct azx *chip)
 	int val;
 	int err;
 
+	to_hda_bus(bus)->bus_probing = 1;
 	hda->probe_continued = 1;
 
 	/* Request display power well for the HDA controller or codec. For
@@ -2200,6 +2201,7 @@ i915_power_fail:
 	if (err < 0)
 		hda->init_failed = 1;
 	complete_all(&hda->probe_wait);
+	to_hda_bus(bus)->bus_probing = 0;
 	return err;
 }
 
diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c
index 17fd81736d3d..039fbbb1e53c 100644
--- a/sound/pci/hda/hda_tegra.c
+++ b/sound/pci/hda/hda_tegra.c
@@ -249,10 +249,12 @@ static int hda_tegra_suspend(struct device *dev)
 	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;
 	struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip);
+	struct hdac_bus *bus = azx_bus(chip);
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
 
 	azx_stop_chip(chip);
+	synchronize_irq(bus->irq);
 	azx_enter_link_reset(chip);
 	hda_tegra_disable_clocks(hda);
 
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index aea3cc2abe3a..40dd46556452 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -853,6 +853,8 @@ static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK),
+	SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK),
+	SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK),
 	SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0467e5ba82e0..5d8ac2d798df 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4792,6 +4792,13 @@ static void alc280_fixup_hp_9480m(struct hda_codec *codec,
 	}
 }
 
+static void alc_fixup_disable_mic_vref(struct hda_codec *codec,
+				  const struct hda_fixup *fix, int action)
+{
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ);
+}
+
 /* for hda_fixup_thinkpad_acpi() */
 #include "thinkpad_helper.c"
 
@@ -4891,6 +4898,7 @@ enum {
 	ALC293_FIXUP_LENOVO_SPK_NOISE,
 	ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY,
 	ALC255_FIXUP_DELL_SPK_NOISE,
+	ALC225_FIXUP_DISABLE_MIC_VREF,
 	ALC225_FIXUP_DELL1_MIC_NO_PRESENCE,
 	ALC295_FIXUP_DISABLE_DAC3,
 	ALC280_FIXUP_HP_HEADSET_MIC,
@@ -5546,6 +5554,12 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
 	},
+	[ALC225_FIXUP_DISABLE_MIC_VREF] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc_fixup_disable_mic_vref,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE
+	},
 	[ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = {
 		.type = HDA_FIXUP_VERBS,
 		.v.verbs = (const struct hda_verb[]) {
@@ -5555,7 +5569,7 @@ static const struct hda_fixup alc269_fixups[] = {
 			{}
 		},
 		.chained = true,
-		.chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE
+		.chain_id = ALC225_FIXUP_DISABLE_MIC_VREF
 	},
 	[ALC280_FIXUP_HP_HEADSET_MIC] = {
 		.type = HDA_FIXUP_FUNC,
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 7c8941b8b2de..dd6c9e6a1d53 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -30,6 +30,7 @@
 #include <linux/math64.h>
 #include <linux/vmalloc.h>
 #include <linux/io.h>
+#include <linux/nospec.h>
 
 #include <sound/core.h>
 #include <sound/control.h>
@@ -4065,15 +4066,16 @@ static int snd_hdsp_channel_info(struct snd_pcm_substream *substream,
 				    struct snd_pcm_channel_info *info)
 {
 	struct hdsp *hdsp = snd_pcm_substream_chip(substream);
-	int mapped_channel;
+	unsigned int channel = info->channel;
 
-	if (snd_BUG_ON(info->channel >= hdsp->max_channels))
+	if (snd_BUG_ON(channel >= hdsp->max_channels))
 		return -EINVAL;
+	channel = array_index_nospec(channel, hdsp->max_channels);
 
-	if ((mapped_channel = hdsp->channel_map[info->channel]) < 0)
+	if (hdsp->channel_map[channel] < 0)
 		return -EINVAL;
 
-	info->offset = mapped_channel * HDSP_CHANNEL_BUFFER_BYTES;
+	info->offset = hdsp->channel_map[channel] * HDSP_CHANNEL_BUFFER_BYTES;
 	info->first = 0;
 	info->step = 32;
 	return 0;
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 7d60d5b03f63..fbb5b979f910 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -228,7 +228,7 @@ config SND_SOC_FSL_SAI_WM9712
 
 config SND_SOC_EUKREA_TLV320
 	tristate "Eukrea TLV320"
-	depends on ARCH_MXC && I2C
+	depends on ARCH_MXC && !ARM64 && I2C
 	select SND_SOC_TLV320AIC23_I2C
 	select SND_SOC_IMX_AUDMUX
 	select SND_SOC_IMX_SSI
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index e8adead8be00..a87836d4de15 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -394,7 +394,8 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		break;
 	case SND_SOC_DAIFMT_RIGHT_J:
 		/* Data on rising edge of bclk, frame high, right aligned */
-		xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCR_xWA;
+		xccr |= ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP;
+		xcr  |= ESAI_xCR_xWA;
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 		/* Data on rising edge of bclk, frame high, 1clk before data */
@@ -451,12 +452,12 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		return -EINVAL;
 	}
 
-	mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR;
+	mask = ESAI_xCR_xFSL | ESAI_xCR_xFSR | ESAI_xCR_xWA;
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, mask, xcr);
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, mask, xcr);
 
 	mask = ESAI_xCCR_xCKP | ESAI_xCCR_xHCKP | ESAI_xCCR_xFSP |
-		ESAI_xCCR_xFSD | ESAI_xCCR_xCKD | ESAI_xCR_xWA;
+		ESAI_xCCR_xFSD | ESAI_xCCR_xCKD;
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, mask, xccr);
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, mask, xccr);
 
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index fc57da341d61..136df38c4536 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -86,49 +86,49 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf,
 	if (!buf)
 		return -ENOMEM;
 
-	ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n",
+	ret = scnprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n",
 		       pdcr, ptcr);
 
 	if (ptcr & IMX_AUDMUX_V2_PTCR_TFSDIR)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxFS output from %s, ",
 				audmux_port_string((ptcr >> 27) & 0x7));
 	else
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxFS input, ");
 
 	if (ptcr & IMX_AUDMUX_V2_PTCR_TCLKDIR)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxClk output from %s",
 				audmux_port_string((ptcr >> 22) & 0x7));
 	else
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"TxClk input");
 
-	ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n");
+	ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n");
 
 	if (ptcr & IMX_AUDMUX_V2_PTCR_SYN) {
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				"Port is symmetric");
 	} else {
 		if (ptcr & IMX_AUDMUX_V2_PTCR_RFSDIR)
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxFS output from %s, ",
 					audmux_port_string((ptcr >> 17) & 0x7));
 		else
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxFS input, ");
 
 		if (ptcr & IMX_AUDMUX_V2_PTCR_RCLKDIR)
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxClk output from %s",
 					audmux_port_string((ptcr >> 12) & 0x7));
 		else
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					"RxClk input");
 	}
 
-	ret += snprintf(buf + ret, PAGE_SIZE - ret,
+	ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 			"\nData received from %s\n",
 			audmux_port_string((pdcr >> 13) & 0x7));
 
diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
index 2b96b11fbe71..1d9dfb92b3b4 100644
--- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c
+++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c
@@ -398,7 +398,13 @@ static int sst_media_hw_params(struct snd_pcm_substream *substream,
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
 {
-	snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	int ret;
+
+	ret =
+		snd_pcm_lib_malloc_pages(substream,
+				params_buffer_bytes(params));
+	if (ret)
+		return ret;
 	memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
 	return 0;
 }
diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c
index 33917146d9c4..054b1d514e8a 100644
--- a/sound/soc/intel/atom/sst/sst_loader.c
+++ b/sound/soc/intel/atom/sst/sst_loader.c
@@ -354,14 +354,14 @@ static int sst_request_fw(struct intel_sst_drv *sst)
 	const struct firmware *fw;
 
 	retval = request_firmware(&fw, sst->firmware_name, sst->dev);
-	if (fw == NULL) {
-		dev_err(sst->dev, "fw is returning as null\n");
-		return -EINVAL;
-	}
 	if (retval) {
 		dev_err(sst->dev, "request fw failed %d\n", retval);
 		return retval;
 	}
+	if (fw == NULL) {
+		dev_err(sst->dev, "fw is returning as null\n");
+		return -EINVAL;
+	}
 	mutex_lock(&sst->sst_lock);
 	retval = sst_cache_and_parse_fw(sst, fw);
 	mutex_unlock(&sst->sst_lock);
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
index 3f8a1e10bed0..e5ca41ffa890 100644
--- a/sound/soc/intel/boards/broadwell.c
+++ b/sound/soc/intel/boards/broadwell.c
@@ -191,7 +191,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
 		.stream_name = "Loopback",
 		.cpu_dai_name = "Loopback Pin",
 		.platform_name = "haswell-pcm-audio",
-		.dynamic = 0,
+		.dynamic = 1,
 		.codec_name = "snd-soc-dummy",
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
index 22558572cb9c..de955c2e8c4e 100644
--- a/sound/soc/intel/boards/haswell.c
+++ b/sound/soc/intel/boards/haswell.c
@@ -145,7 +145,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = {
 		.stream_name = "Loopback",
 		.cpu_dai_name = "Loopback Pin",
 		.platform_name = "haswell-pcm-audio",
-		.dynamic = 0,
+		.dynamic = 1,
 		.codec_name = "snd-soc-dummy",
 		.codec_dai_name = "snd-soc-dummy-dai",
 		.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c
index 09db2aec12a3..776e809a8aab 100644
--- a/sound/soc/omap/omap-dmic.c
+++ b/sound/soc/omap/omap-dmic.c
@@ -48,6 +48,8 @@ struct omap_dmic {
 	struct device *dev;
 	void __iomem *io_base;
 	struct clk *fclk;
+	struct pm_qos_request pm_qos_req;
+	int latency;
 	int fclk_freq;
 	int out_freq;
 	int clk_div;
@@ -124,6 +126,8 @@ static void omap_dmic_dai_shutdown(struct snd_pcm_substream *substream,
 
 	mutex_lock(&dmic->mutex);
 
+	pm_qos_remove_request(&dmic->pm_qos_req);
+
 	if (!dai->active)
 		dmic->active = 0;
 
@@ -226,6 +230,8 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
 	/* packet size is threshold * channels */
 	dma_data = snd_soc_dai_get_dma_data(dai, substream);
 	dma_data->maxburst = dmic->threshold * channels;
+	dmic->latency = (OMAP_DMIC_THRES_MAX - dmic->threshold) * USEC_PER_SEC /
+			params_rate(params);
 
 	return 0;
 }
@@ -236,6 +242,9 @@ static int omap_dmic_dai_prepare(struct snd_pcm_substream *substream,
 	struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
 	u32 ctrl;
 
+	if (pm_qos_request_active(&dmic->pm_qos_req))
+		pm_qos_update_request(&dmic->pm_qos_req, dmic->latency);
+
 	/* Configure uplink threshold */
 	omap_dmic_write(dmic, OMAP_DMIC_FIFO_CTRL_REG, dmic->threshold);
 
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 8d0d45d330e7..8eb2d12b6a34 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -54,6 +54,8 @@ struct omap_mcpdm {
 	unsigned long phys_base;
 	void __iomem *io_base;
 	int irq;
+	struct pm_qos_request pm_qos_req;
+	int latency[2];
 
 	struct mutex mutex;
 
@@ -273,6 +275,9 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
 				  struct snd_soc_dai *dai)
 {
 	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+	int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+	int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
 
 	mutex_lock(&mcpdm->mutex);
 
@@ -285,6 +290,14 @@ static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
 		}
 	}
 
+	if (mcpdm->latency[stream2])
+		pm_qos_update_request(&mcpdm->pm_qos_req,
+				      mcpdm->latency[stream2]);
+	else if (mcpdm->latency[stream1])
+		pm_qos_remove_request(&mcpdm->pm_qos_req);
+
+	mcpdm->latency[stream1] = 0;
+
 	mutex_unlock(&mcpdm->mutex);
 }
 
@@ -296,7 +309,7 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
 	int stream = substream->stream;
 	struct snd_dmaengine_dai_dma_data *dma_data;
 	u32 threshold;
-	int channels;
+	int channels, latency;
 	int link_mask = 0;
 
 	channels = params_channels(params);
@@ -336,14 +349,25 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
 
 		dma_data->maxburst =
 				(MCPDM_DN_THRES_MAX - threshold) * channels;
+		latency = threshold;
 	} else {
 		/* If playback is not running assume a stereo stream to come */
 		if (!mcpdm->config[!stream].link_mask)
 			mcpdm->config[!stream].link_mask = (0x3 << 3);
 
 		dma_data->maxburst = threshold * channels;
+		latency = (MCPDM_DN_THRES_MAX - threshold);
 	}
 
+	/*
+	 * The DMA must act to a DMA request within latency time (usec) to avoid
+	 * under/overflow
+	 */
+	mcpdm->latency[stream] = latency * USEC_PER_SEC / params_rate(params);
+
+	if (!mcpdm->latency[stream])
+		mcpdm->latency[stream] = 10;
+
 	/* Check if we need to restart McPDM with this stream */
 	if (mcpdm->config[stream].link_mask &&
 	    mcpdm->config[stream].link_mask != link_mask)
@@ -358,6 +382,20 @@ static int omap_mcpdm_prepare(struct snd_pcm_substream *substream,
 				  struct snd_soc_dai *dai)
 {
 	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+	struct pm_qos_request *pm_qos_req = &mcpdm->pm_qos_req;
+	int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+	int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+	int latency = mcpdm->latency[stream2];
+
+	/* Prevent omap hardware from hitting off between FIFO fills */
+	if (!latency || mcpdm->latency[stream1] < latency)
+		latency = mcpdm->latency[stream1];
+
+	if (pm_qos_request_active(pm_qos_req))
+		pm_qos_update_request(pm_qos_req, latency);
+	else if (latency)
+		pm_qos_add_request(pm_qos_req, PM_QOS_CPU_DMA_LATENCY, latency);
 
 	if (!omap_mcpdm_active(mcpdm)) {
 		omap_mcpdm_start(mcpdm);
@@ -419,6 +457,9 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai)
 	free_irq(mcpdm->irq, (void *)mcpdm);
 	pm_runtime_disable(mcpdm->dev);
 
+	if (pm_qos_request_active(&mcpdm->pm_qos_req))
+		pm_qos_remove_request(&mcpdm->pm_qos_req);
+
 	return 0;
 }
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index fa6b74a304a7..b927f9c81d92 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -1711,6 +1711,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 	}
 
 	card->instantiated = 1;
+	dapm_mark_endpoints_dirty(card);
 	snd_soc_dapm_sync(&card->dapm);
 	mutex_unlock(&card->mutex);
 	mutex_unlock(&client_mutex);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 0aefed8ab0cf..7e26d173da41 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1943,19 +1943,19 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 		out = is_connected_output_ep(w, NULL);
 	}
 
-	ret = snprintf(buf, PAGE_SIZE, "%s: %s%s  in %d out %d",
+	ret = scnprintf(buf, PAGE_SIZE, "%s: %s%s  in %d out %d",
 		       w->name, w->power ? "On" : "Off",
 		       w->force ? " (forced)" : "", in, out);
 
 	if (w->reg >= 0)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret,
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 				" - R%d(0x%x) mask 0x%x",
 				w->reg, w->reg, w->mask << w->shift);
 
-	ret += snprintf(buf + ret, PAGE_SIZE - ret, "\n");
+	ret += scnprintf(buf + ret, PAGE_SIZE - ret, "\n");
 
 	if (w->sname)
-		ret += snprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
+		ret += scnprintf(buf + ret, PAGE_SIZE - ret, " stream %s %s\n",
 				w->sname,
 				w->active ? "active" : "inactive");
 
@@ -1968,7 +1968,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file,
 			if (!p->connect)
 				continue;
 
-			ret += snprintf(buf + ret, PAGE_SIZE - ret,
+			ret += scnprintf(buf + ret, PAGE_SIZE - ret,
 					" %s  \"%s\" \"%s\"\n",
 					(rdir == SND_SOC_DAPM_DIR_IN) ? "in" : "out",
 					p->name ? p->name : "static",
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index c1e76feb3529..824f4d7fc41f 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1770,6 +1770,7 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp,
 	struct snd_soc_tplg_ops *ops, const struct firmware *fw, u32 id)
 {
 	struct soc_tplg tplg;
+	int ret;
 
 	/* setup parsing context */
 	memset(&tplg, 0, sizeof(tplg));
@@ -1783,7 +1784,12 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp,
 	tplg.bytes_ext_ops = ops->bytes_ext_ops;
 	tplg.bytes_ext_ops_count = ops->bytes_ext_ops_count;
 
-	return soc_tplg_load(&tplg);
+	ret = soc_tplg_load(&tplg);
+	/* free the created components if fail to load topology */
+	if (ret)
+		snd_soc_tplg_component_remove(comp, SND_SOC_TPLG_INDEX_ALL);
+
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_tplg_component_load);
 
diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c
index e557946718a9..d9fcae071b47 100644
--- a/sound/synth/emux/emux_hwdep.c
+++ b/sound/synth/emux/emux_hwdep.c
@@ -22,9 +22,9 @@
 #include <sound/core.h>
 #include <sound/hwdep.h>
 #include <linux/uaccess.h>
+#include <linux/nospec.h>
 #include "emux_voice.h"
 
-
 #define TMP_CLIENT_ID	0x1001
 
 /*
@@ -66,13 +66,16 @@ snd_emux_hwdep_misc_mode(struct snd_emux *emu, void __user *arg)
 		return -EFAULT;
 	if (info.mode < 0 || info.mode >= EMUX_MD_END)
 		return -EINVAL;
+	info.mode = array_index_nospec(info.mode, EMUX_MD_END);
 
 	if (info.port < 0) {
 		for (i = 0; i < emu->num_ports; i++)
 			emu->portptrs[i]->ctrls[info.mode] = info.value;
 	} else {
-		if (info.port < emu->num_ports)
+		if (info.port < emu->num_ports) {
+			info.port = array_index_nospec(info.port, emu->num_ports);
 			emu->portptrs[info.port]->ctrls[info.mode] = info.value;
+		}
 	}
 	return 0;
 }
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 97d6a18e6956..f7eb0d2f797b 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -1816,7 +1816,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 				char *name)
 {
 	struct uac_processing_unit_descriptor *desc = raw_desc;
-	int num_ins = desc->bNrInPins;
+	int num_ins;
 	struct usb_mixer_elem_info *cval;
 	struct snd_kcontrol *kctl;
 	int i, err, nameid, type, len;
@@ -1831,7 +1831,13 @@ static int build_audio_procunit(struct mixer_build *state, int unitid,
 		0, NULL, default_value_info
 	};
 
-	if (desc->bLength < 13 || desc->bLength < 13 + num_ins ||
+	if (desc->bLength < 13) {
+		usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid);
+		return -EINVAL;
+	}
+
+	num_ins = desc->bNrInPins;
+	if (desc->bLength < 13 + num_ins ||
 	    desc->bLength < num_ins + uac_processing_unit_bControlSize(desc, state->mixer->protocol)) {
 		usb_audio_err(state->chip, "invalid %s descriptor (id %d)\n", name, unitid);
 		return -EINVAL;
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index a9079654107c..1ea1384bc236 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -313,6 +313,9 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum,
 	return 0;
 }
 
+/* Setup an implicit feedback endpoint from a quirk. Returns 0 if no quirk
+ * applies. Returns 1 if a quirk was found.
+ */
 static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
 					 struct usb_device *dev,
 					 struct usb_interface_descriptor *altsd,
@@ -381,7 +384,7 @@ add_sync_ep:
 
 	subs->data_endpoint->sync_master = subs->sync_endpoint;
 
-	return 0;
+	return 1;
 }
 
 static int set_sync_endpoint(struct snd_usb_substream *subs,
@@ -420,6 +423,10 @@ static int set_sync_endpoint(struct snd_usb_substream *subs,
 	if (err < 0)
 		return err;
 
+	/* endpoint set by quirk */
+	if (err > 0)
+		return 0;
+
 	if (altsd->bNumEndpoints < 2)
 		return 0;
 
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 15cbe2565703..d32727c74a16 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3321,6 +3321,9 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 					}
 				}
 			},
+			{
+				.ifnum = -1
+			},
 		}
 	}
 },