Merge tag 'sound-5.2-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "It might feel like deja vu to receive a bulk of changes at rc5, and it
  happens again; we've got a collection of fixes for ASoC. Most of fixes
  are targeted for the newly merged SOF (Sound Open Firmware) stuff and
  the relevant fixes for Intel platforms.

  Other than that, there are a few regression fixes for the recent ASoC
  core changes and HD-audio quirk, as well as a couple of FireWire fixes
  and for other ASoC codecs"

* tag 'sound-5.2-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (54 commits)
  Revert "ALSA: hda/realtek - Improve the headset mic for Acer Aspire laptops"
  ALSA: ice1712: Check correct return value to snd_i2c_sendbytes (EWS/DMX 6Fire)
  ALSA: oxfw: allow PCM capture for Stanton SCS.1m
  ALSA: firewire-motu: fix destruction of data for isochronous resources
  ASoC: Intel: sst: fix kmalloc call with wrong flags
  ASoC: core: Fix deadlock in snd_soc_instantiate_card()
  SoC: rt274: Fix internal jack assignment in set_jack callback
  ALSA: hdac: fix memory release for SST and SOF drivers
  ASoC: SOF: Intel: hda: use the defined ppcap functions
  ASoC: core: move DAI pre-links initiation to snd_soc_instantiate_card
  ASoC: Intel: cht_bsw_rt5672: fix kernel oops with platform_name override
  ASoC: Intel: cht_bsw_nau8824: fix kernel oops with platform_name override
  ASoC: Intel: bytcht_es8316: fix kernel oops with platform_name override
  ASoC: Intel: cht_bsw_max98090: fix kernel oops with platform_name override
  ASoC: sun4i-i2s: Add offset to RX channel select
  ASoC: sun4i-i2s: Fix sun8i tx channel offset mask
  ASoC: max98090: remove 24-bit format support if RJ is 0
  ASoC: da7219: Fix build error without CONFIG_I2C
  ASoC: SOF: Intel: hda: Fix COMPILE_TEST build error
  ASoC: SOF: fix DSP oops definitions in FW ABI
  ...

46 files changed, 465 insertions, 259 deletions

diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h
index 3b67c93ff101..3d174e20aa53 100644
--- a/include/sound/sof/dai.h
+++ b/include/sound/sof/dai.h
@@ -49,6 +49,7 @@ enum sof_ipc_dai_type {
 	SOF_DAI_INTEL_SSP,		/**< Intel SSP */
 	SOF_DAI_INTEL_DMIC,		/**< Intel DMIC */
 	SOF_DAI_INTEL_HDA,		/**< Intel HD/A */
+	SOF_DAI_INTEL_SOUNDWIRE,	/**< Intel SoundWire */
 };
 
 /* general purpose DAI configuration */
diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h
index ccb6a004b37b..1efcf7b18ec2 100644
--- a/include/sound/sof/header.h
+++ b/include/sound/sof/header.h
@@ -48,6 +48,7 @@
 #define SOF_IPC_FW_READY			SOF_GLB_TYPE(0x7U)
 #define SOF_IPC_GLB_DAI_MSG			SOF_GLB_TYPE(0x8U)
 #define SOF_IPC_GLB_TRACE_MSG			SOF_GLB_TYPE(0x9U)
+#define SOF_IPC_GLB_GDB_DEBUG                   SOF_GLB_TYPE(0xAU)
 
 /*
  * DSP Command Message Types
@@ -78,6 +79,7 @@
 #define SOF_IPC_COMP_GET_VALUE			SOF_CMD_TYPE(0x002)
 #define SOF_IPC_COMP_SET_DATA			SOF_CMD_TYPE(0x003)
 #define SOF_IPC_COMP_GET_DATA			SOF_CMD_TYPE(0x004)
+#define SOF_IPC_COMP_NOTIFICATION		SOF_CMD_TYPE(0x005)
 
 /* DAI messages */
 #define SOF_IPC_DAI_CONFIG			SOF_CMD_TYPE(0x001)
@@ -153,6 +155,27 @@ struct sof_ipc_compound_hdr {
 	uint32_t count;		/**< count of 0 means end of compound sequence */
 }  __packed;
 
+/**
+ * OOPS header architecture specific data.
+ */
+struct sof_ipc_dsp_oops_arch_hdr {
+	uint32_t arch;		/* Identifier of architecture */
+	uint32_t totalsize;	/* Total size of oops message */
+}  __packed;
+
+/**
+ * OOPS header platform specific data.
+ */
+struct sof_ipc_dsp_oops_plat_hdr {
+	uint32_t configidhi;	/* ConfigID hi 32bits */
+	uint32_t configidlo;	/* ConfigID lo 32bits */
+	uint32_t numaregs;	/* Special regs num */
+	uint32_t stackoffset;	/* Offset to stack pointer from beginning of
+				 * oops message
+				 */
+	uint32_t stackptr;	/* Stack ptr */
+}  __packed;
+
 /** @}*/
 
 #endif
diff --git a/include/sound/sof/info.h b/include/sound/sof/info.h
index 21dae04d8183..16528d2b4a50 100644
--- a/include/sound/sof/info.h
+++ b/include/sound/sof/info.h
@@ -18,6 +18,14 @@
 
 #define SOF_IPC_MAX_ELEMS	16
 
+/*
+ * Firmware boot info flag bits (64-bit)
+ */
+#define SOF_IPC_INFO_BUILD		BIT(0)
+#define SOF_IPC_INFO_LOCKS		BIT(1)
+#define SOF_IPC_INFO_LOCKSV		BIT(2)
+#define SOF_IPC_INFO_GDB		BIT(3)
+
 /* extended data types that can be appended onto end of sof_ipc_fw_ready */
 enum sof_ipc_ext_data {
 	SOF_IPC_EXT_DMA_BUFFER = 0,
@@ -49,16 +57,8 @@ struct sof_ipc_fw_ready {
 	uint32_t hostbox_size;
 	struct sof_ipc_fw_version version;
 
-	/* Miscellaneous debug flags showing build/debug features enabled */
-	union {
-		uint64_t reserved;
-		struct {
-			uint64_t build:1;
-			uint64_t locks:1;
-			uint64_t locks_verbose:1;
-			uint64_t gdb:1;
-		} bits;
-	} debug;
+	/* Miscellaneous flags */
+	uint64_t flags;
 
 	/* reserved for future use */
 	uint32_t reserved[4];
diff --git a/include/sound/sof/xtensa.h b/include/sound/sof/xtensa.h
index a7189984000d..d25c764b10e8 100644
--- a/include/sound/sof/xtensa.h
+++ b/include/sound/sof/xtensa.h
@@ -17,7 +17,8 @@
 
 /* Xtensa Firmware Oops data */
 struct sof_ipc_dsp_oops_xtensa {
-	struct sof_ipc_hdr hdr;
+	struct sof_ipc_dsp_oops_arch_hdr arch_hdr;
+	struct sof_ipc_dsp_oops_plat_hdr plat_hdr;
 	uint32_t exccause;
 	uint32_t excvaddr;
 	uint32_t ps;
@@ -38,7 +39,11 @@ struct sof_ipc_dsp_oops_xtensa {
 	uint32_t intenable;
 	uint32_t interrupt;
 	uint32_t sar;
-	uint32_t stack;
+	uint32_t debugcause;
+	uint32_t windowbase;
+	uint32_t windowstart;
+	uint32_t excsave1;
+	uint32_t ar[];
 }  __packed;
 
 #endif
diff --git a/include/uapi/sound/sof/abi.h b/include/uapi/sound/sof/abi.h
index 37e0a90dc9e6..0868eb47acf7 100644
--- a/include/uapi/sound/sof/abi.h
+++ b/include/uapi/sound/sof/abi.h
@@ -26,7 +26,7 @@
 
 /* SOF ABI version major, minor and patch numbers */
 #define SOF_ABI_MAJOR 3
-#define SOF_ABI_MINOR 4
+#define SOF_ABI_MINOR 6
 #define SOF_ABI_PATCH 0
 
 /* SOF ABI version number. Format within 32bit word is MMmmmppp */
diff --git a/sound/firewire/motu/motu-stream.c b/sound/firewire/motu/motu-stream.c
index 37e47fa7b0e3..81f7edc560d0 100644
--- a/sound/firewire/motu/motu-stream.c
+++ b/sound/firewire/motu/motu-stream.c
@@ -344,7 +344,7 @@ static void destroy_stream(struct snd_motu *motu,
 	}
 
 	amdtp_stream_destroy(stream);
-	fw_iso_resources_free(resources);
+	fw_iso_resources_destroy(resources);
 }
 
 int snd_motu_stream_init_duplex(struct snd_motu *motu)
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 5e31c460a90f..9fd145cc4b07 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -148,9 +148,6 @@ static int detect_quirks(struct snd_oxfw *oxfw)
 		oxfw->midi_input_ports = 0;
 		oxfw->midi_output_ports = 0;
 
-		/* Output stream exists but no data channels are useful. */
-		oxfw->has_output = false;
-
 		return snd_oxfw_scs1x_add(oxfw);
 	}
 
diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c
index ad1b55a1f045..a3a113ef5d56 100644
--- a/sound/hda/ext/hdac_ext_bus.c
+++ b/sound/hda/ext/hdac_ext_bus.c
@@ -162,7 +162,6 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_init);
 void snd_hdac_ext_bus_device_exit(struct hdac_device *hdev)
 {
 	snd_hdac_device_exit(hdev);
-	kfree(hdev);
 }
 EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_device_exit);
 
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index c5e46df9c548..6c51b8363f8b 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -826,7 +826,14 @@ static int snd_hda_codec_dev_free(struct snd_device *device)
 	if (codec->core.type == HDA_DEV_LEGACY)
 		snd_hdac_device_unregister(&codec->core);
 	codec_display_power(codec, false);
-	put_device(hda_codec_dev(codec));
+
+	/*
+	 * In the case of ASoC HD-audio bus, the device refcount is released in
+	 * snd_hdac_ext_bus_device_remove() explicitly.
+	 */
+	if (codec->core.type == HDA_DEV_LEGACY)
+		put_device(hda_codec_dev(codec));
+
 	return 0;
 }
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 974244978509..5b3c26991f26 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4120,18 +4120,19 @@ static struct coef_fw alc225_pre_hsmode[] = {
 static void alc_headset_mode_unplugged(struct hda_codec *codec)
 {
 	static struct coef_fw coef0255[] = {
+		WRITE_COEF(0x1b, 0x0c0b), /* LDO and MISC control */
 		WRITE_COEF(0x45, 0xd089), /* UAJ function set to menual mode */
 		UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), /* Direct Drive HP Amp control(Set to verb control)*/
 		WRITE_COEF(0x06, 0x6104), /* Set MIC2 Vref gate with HP */
 		WRITE_COEFEX(0x57, 0x03, 0x8aa6), /* Direct Drive HP Amp control */
 		{}
 	};
-	static struct coef_fw coef0255_1[] = {
-		WRITE_COEF(0x1b, 0x0c0b), /* LDO and MISC control */
-		{}
-	};
 	static struct coef_fw coef0256[] = {
 		WRITE_COEF(0x1b, 0x0c4b), /* LDO and MISC control */
+		WRITE_COEF(0x45, 0xd089), /* UAJ function set to menual mode */
+		WRITE_COEF(0x06, 0x6104), /* Set MIC2 Vref gate with HP */
+		WRITE_COEFEX(0x57, 0x03, 0x09a3), /* Direct Drive HP Amp control */
+		UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), /* Direct Drive HP Amp control(Set to verb control)*/
 		{}
 	};
 	static struct coef_fw coef0233[] = {
@@ -4194,13 +4195,11 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec)
 
 	switch (codec->core.vendor_id) {
 	case 0x10ec0255:
-		alc_process_coef_fw(codec, coef0255_1);
 		alc_process_coef_fw(codec, coef0255);
 		break;
 	case 0x10ec0236:
 	case 0x10ec0256:
 		alc_process_coef_fw(codec, coef0256);
-		alc_process_coef_fw(codec, coef0255);
 		break;
 	case 0x10ec0234:
 	case 0x10ec0274:
@@ -4253,6 +4252,12 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
 		WRITE_COEF(0x06, 0x6100), /* Set MIC2 Vref gate to normal */
 		{}
 	};
+	static struct coef_fw coef0256[] = {
+		UPDATE_COEFEX(0x57, 0x05, 1<<14, 1<<14), /* Direct Drive HP Amp control(Set to verb control)*/
+		WRITE_COEFEX(0x57, 0x03, 0x09a3),
+		WRITE_COEF(0x06, 0x6100), /* Set MIC2 Vref gate to normal */
+		{}
+	};
 	static struct coef_fw coef0233[] = {
 		UPDATE_COEF(0x35, 0, 1<<14),
 		WRITE_COEF(0x06, 0x2100),
@@ -4300,14 +4305,19 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
 	};
 
 	switch (codec->core.vendor_id) {
-	case 0x10ec0236:
 	case 0x10ec0255:
-	case 0x10ec0256:
 		alc_write_coef_idx(codec, 0x45, 0xc489);
 		snd_hda_set_pin_ctl_cache(codec, hp_pin, 0);
 		alc_process_coef_fw(codec, coef0255);
 		snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50);
 		break;
+	case 0x10ec0236:
+	case 0x10ec0256:
+		alc_write_coef_idx(codec, 0x45, 0xc489);
+		snd_hda_set_pin_ctl_cache(codec, hp_pin, 0);
+		alc_process_coef_fw(codec, coef0256);
+		snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50);
+		break;
 	case 0x10ec0234:
 	case 0x10ec0274:
 	case 0x10ec0294:
@@ -4389,6 +4399,14 @@ static void alc_headset_mode_default(struct hda_codec *codec)
 		WRITE_COEF(0x49, 0x0049),
 		{}
 	};
+	static struct coef_fw coef0256[] = {
+		WRITE_COEF(0x45, 0xc489),
+		WRITE_COEFEX(0x57, 0x03, 0x0da3),
+		WRITE_COEF(0x49, 0x0049),
+		UPDATE_COEFEX(0x57, 0x05, 1<<14, 0), /* Direct Drive HP Amp control(Set to verb control)*/
+		WRITE_COEF(0x06, 0x6100),
+		{}
+	};
 	static struct coef_fw coef0233[] = {
 		WRITE_COEF(0x06, 0x2100),
 		WRITE_COEF(0x32, 0x4ea3),
@@ -4439,11 +4457,16 @@ static void alc_headset_mode_default(struct hda_codec *codec)
 		alc_process_coef_fw(codec, alc225_pre_hsmode);
 		alc_process_coef_fw(codec, coef0225);
 		break;
-	case 0x10ec0236:
 	case 0x10ec0255:
-	case 0x10ec0256:
 		alc_process_coef_fw(codec, coef0255);
 		break;
+	case 0x10ec0236:
+	case 0x10ec0256:
+		alc_write_coef_idx(codec, 0x1b, 0x0e4b);
+		alc_write_coef_idx(codec, 0x45, 0xc089);
+		msleep(50);
+		alc_process_coef_fw(codec, coef0256);
+		break;
 	case 0x10ec0234:
 	case 0x10ec0274:
 	case 0x10ec0294:
@@ -4487,8 +4510,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec)
 	};
 	static struct coef_fw coef0256[] = {
 		WRITE_COEF(0x45, 0xd489), /* Set to CTIA type */
-		WRITE_COEF(0x1b, 0x0c6b),
-		WRITE_COEFEX(0x57, 0x03, 0x8ea6),
+		WRITE_COEF(0x1b, 0x0e6b),
 		{}
 	};
 	static struct coef_fw coef0233[] = {
@@ -4606,8 +4628,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec)
 	};
 	static struct coef_fw coef0256[] = {
 		WRITE_COEF(0x45, 0xe489), /* Set to OMTP Type */
-		WRITE_COEF(0x1b, 0x0c6b),
-		WRITE_COEFEX(0x57, 0x03, 0x8ea6),
+		WRITE_COEF(0x1b, 0x0e6b),
 		{}
 	};
 	static struct coef_fw coef0233[] = {
@@ -4739,13 +4760,37 @@ static void alc_determine_headset_type(struct hda_codec *codec)
 	};
 
 	switch (codec->core.vendor_id) {
-	case 0x10ec0236:
 	case 0x10ec0255:
+		alc_process_coef_fw(codec, coef0255);
+		msleep(300);
+		val = alc_read_coef_idx(codec, 0x46);
+		is_ctia = (val & 0x0070) == 0x0070;
+		break;
+	case 0x10ec0236:
 	case 0x10ec0256:
+		alc_write_coef_idx(codec, 0x1b, 0x0e4b);
+		alc_write_coef_idx(codec, 0x06, 0x6104);
+		alc_write_coefex_idx(codec, 0x57, 0x3, 0x09a3);
+
+		snd_hda_codec_write(codec, 0x21, 0,
+			    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+		msleep(80);
+		snd_hda_codec_write(codec, 0x21, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0);
+
 		alc_process_coef_fw(codec, coef0255);
 		msleep(300);
 		val = alc_read_coef_idx(codec, 0x46);
 		is_ctia = (val & 0x0070) == 0x0070;
+
+		alc_write_coefex_idx(codec, 0x57, 0x3, 0x0da3);
+		alc_update_coefex_idx(codec, 0x57, 0x5, 1<<14, 0);
+
+		snd_hda_codec_write(codec, 0x21, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+		msleep(80);
+		snd_hda_codec_write(codec, 0x21, 0,
+			    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE);
 		break;
 	case 0x10ec0234:
 	case 0x10ec0274:
@@ -6210,15 +6255,13 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chain_id = ALC269_FIXUP_THINKPAD_ACPI,
 	},
 	[ALC255_FIXUP_ACER_MIC_NO_PRESENCE] = {
-		.type = HDA_FIXUP_VERBS,
-		.v.verbs = (const struct hda_verb[]) {
-			/* Enable the Mic */
-			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x45 },
-			{ 0x20, AC_VERB_SET_PROC_COEF, 0x5089 },
-			{}
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+			{ }
 		},
 		.chained = true,
-		.chain_id = ALC269_FIXUP_LIFEBOOK_EXTMIC
+		.chain_id = ALC255_FIXUP_HEADSET_MODE
 	},
 	[ALC255_FIXUP_ASUS_MIC_NO_PRESENCE] = {
 		.type = HDA_FIXUP_PINS,
@@ -7263,10 +7306,6 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x19, 0x0181303F},
 		{0x21, 0x0221102f}),
 	SND_HDA_PIN_QUIRK(0x10ec0255, 0x1025, "Acer", ALC255_FIXUP_ACER_MIC_NO_PRESENCE,
-		{0x12, 0x90a60140},
-		{0x14, 0x90170120},
-		{0x21, 0x02211030}),
-	SND_HDA_PIN_QUIRK(0x10ec0255, 0x1025, "Acer", ALC255_FIXUP_ACER_MIC_NO_PRESENCE,
 		{0x12, 0x90a601c0},
 		{0x14, 0x90171120},
 		{0x21, 0x02211030}),
diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c
index 3729b132ad85..fe08dd9f1564 100644
--- a/sound/pci/ice1712/ews.c
+++ b/sound/pci/ice1712/ews.c
@@ -812,7 +812,7 @@ static int snd_ice1712_6fire_read_pca(struct snd_ice1712 *ice, unsigned char reg
 
 	snd_i2c_lock(ice->i2c);
 	byte = reg;
-	if (snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_6FIRE], &byte, 1)) {
+	if (snd_i2c_sendbytes(spec->i2cdevs[EWS_I2C_6FIRE], &byte, 1) != 1) {
 		snd_i2c_unlock(ice->i2c);
 		dev_err(ice->card->dev, "cannot send pca\n");
 		return -EIO;
diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c
index eab7c76cfcd9..71562154c0b1 100644
--- a/sound/soc/codecs/ak4458.c
+++ b/sound/soc/codecs/ak4458.c
@@ -304,7 +304,10 @@ static int ak4458_rstn_control(struct snd_soc_component *component, int bit)
 					  AK4458_00_CONTROL1,
 					  AK4458_RSTN_MASK,
 					  0x0);
-	return ret;
+	if (ret < 0)
+		return ret;
+
+	return 0;
 }
 
 static int ak4458_hw_params(struct snd_pcm_substream *substream,
@@ -536,9 +539,10 @@ static void ak4458_power_on(struct ak4458_priv *ak4458)
 	}
 }
 
-static void ak4458_init(struct snd_soc_component *component)
+static int ak4458_init(struct snd_soc_component *component)
 {
 	struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component);
+	int ret;
 
 	/* External Mute ON */
 	if (ak4458->mute_gpiod)
@@ -546,21 +550,21 @@ static void ak4458_init(struct snd_soc_component *component)
 
 	ak4458_power_on(ak4458);
 
-	snd_soc_component_update_bits(component, AK4458_00_CONTROL1,
+	ret = snd_soc_component_update_bits(component, AK4458_00_CONTROL1,
 			    0x80, 0x80);   /* ACKS bit = 1; 10000000 */
+	if (ret < 0)
+		return ret;
 
-	ak4458_rstn_control(component, 1);
+	return ak4458_rstn_control(component, 1);
 }
 
 static int ak4458_probe(struct snd_soc_component *component)
 {
 	struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component);
 
-	ak4458_init(component);
-
 	ak4458->fs = 48000;
 
-	return 0;
+	return ak4458_init(component);
 }
 
 static void ak4458_remove(struct snd_soc_component *component)
diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c
index ab27d2b94d02..c0190ec59e74 100644
--- a/sound/soc/codecs/cs4265.c
+++ b/sound/soc/codecs/cs4265.c
@@ -60,7 +60,7 @@ static const struct reg_default cs4265_reg_defaults[] = {
 static bool cs4265_readable_register(struct device *dev, unsigned int reg)
 {
 	switch (reg) {
-	case CS4265_CHIP_ID ... CS4265_SPDIF_CTL2:
+	case CS4265_CHIP_ID ... CS4265_MAX_REGISTER:
 		return true;
 	default:
 		return false;
diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c
index ebb9e0cf8364..28a4ac36c4f8 100644
--- a/sound/soc/codecs/cs42xx8.c
+++ b/sound/soc/codecs/cs42xx8.c
@@ -558,6 +558,7 @@ static int cs42xx8_runtime_resume(struct device *dev)
 	msleep(5);
 
 	regcache_cache_only(cs42xx8->regmap, false);
+	regcache_mark_dirty(cs42xx8->regmap);
 
 	ret = regcache_sync(cs42xx8->regmap);
 	if (ret) {
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index 7619ea31ab50..ada8c25e643d 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -1909,6 +1909,21 @@ static int max98090_configure_dmic(struct max98090_priv *max98090,
 	return 0;
 }
 
+static int max98090_dai_startup(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_component *component = dai->component;
+	struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component);
+	unsigned int fmt = max98090->dai_fmt;
+
+	/* Remove 24-bit format support if it is not in right justified mode. */
+	if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J) {
+		substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE;
+		snd_pcm_hw_constraint_msbits(substream->runtime, 0, 16, 16);
+	}
+	return 0;
+}
+
 static int max98090_dai_hw_params(struct snd_pcm_substream *substream,
 				   struct snd_pcm_hw_params *params,
 				   struct snd_soc_dai *dai)
@@ -2316,6 +2331,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect);
 #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
 
 static const struct snd_soc_dai_ops max98090_dai_ops = {
+	.startup = max98090_dai_startup,
 	.set_sysclk = max98090_dai_set_sysclk,
 	.set_fmt = max98090_dai_set_fmt,
 	.set_tdm_slot = max98090_set_tdm_slot,
diff --git a/sound/soc/codecs/rt274.c b/sound/soc/codecs/rt274.c
index adf59039a3b6..cdd312db3e78 100644
--- a/sound/soc/codecs/rt274.c
+++ b/sound/soc/codecs/rt274.c
@@ -405,6 +405,8 @@ static int rt274_mic_detect(struct snd_soc_component *component,
 {
 	struct rt274_priv *rt274 = snd_soc_component_get_drvdata(component);
 
+	rt274->jack = jack;
+
 	if (jack == NULL) {
 		/* Disable jack detection */
 		regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL,
@@ -412,7 +414,6 @@ static int rt274_mic_detect(struct snd_soc_component *component,
 
 		return 0;
 	}
-	rt274->jack = jack;
 
 	regmap_update_bits(rt274->regmap, RT274_EAPD_GPIO_IRQ_CTRL,
 				RT274_IRQ_EN, RT274_IRQ_EN);
diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c
index 9a037108b1ae..a746e11ccfe3 100644
--- a/sound/soc/codecs/rt5670.c
+++ b/sound/soc/codecs/rt5670.c
@@ -2882,6 +2882,18 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = {
 						 RT5670_DEV_GPIO |
 						 RT5670_JD_MODE3),
 	},
+	{
+		.callback = rt5670_quirk_cb,
+		.ident = "Aegex 10 tablet (RU2)",
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "AEGEX"),
+			DMI_MATCH(DMI_PRODUCT_VERSION, "RU2"),
+		},
+		.driver_data = (unsigned long *)(RT5670_DMIC_EN |
+						 RT5670_DMIC2_INR |
+						 RT5670_DEV_GPIO |
+						 RT5670_JD_MODE3),
+	},
 	{}
 };
 
diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c
index 84b6bd8b50e1..a4dfa0345c6e 100644
--- a/sound/soc/codecs/rt5677-spi.c
+++ b/sound/soc/codecs/rt5677-spi.c
@@ -101,7 +101,7 @@ static void rt5677_spi_reverse(u8 *dst, u32 dstlen, const u8 *src, u32 srclen)
 	u32 word_size = min_t(u32, dstlen, 8);
 
 	for (w = 0; w < dstlen; w += word_size) {
-		for (i = 0; i < word_size; i++) {
+		for (i = 0; i < word_size && i + w < dstlen; i++) {
 			si = w + word_size - i - 1;
 			dst[w + i] = si < srclen ? src[si] : 0;
 		}
@@ -152,8 +152,9 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len)
 		status |= spi_sync(g_spi, &m);
 		mutex_unlock(&spi_mutex);
 
+
 		/* Copy data back to caller buffer */
-		rt5677_spi_reverse(cb + offset, t[1].len, body, t[1].len);
+		rt5677_spi_reverse(cb + offset, len - offset, body, t[1].len);
 	}
 	return status;
 }
diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index 0b937924d2e4..ea035c12a325 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -282,8 +282,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair)
 		return -EINVAL;
 	}
 
-	if ((outrate > 8000 && outrate < 30000) &&
-	    (outrate/inrate > 24 || inrate/outrate > 8)) {
+	if ((outrate >= 8000 && outrate <= 30000) &&
+	    (outrate > 24 * inrate || inrate > 8 * outrate)) {
 		pair_err("exceed supported ratio range [1/24, 8] for \
 				inrate/outrate: %d/%d\n", inrate, outrate);
 		return -EINVAL;
diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c
index 1ec298dd0e4f..13db2854db3e 100644
--- a/sound/soc/intel/atom/sst/sst_pvt.c
+++ b/sound/soc/intel/atom/sst/sst_pvt.c
@@ -158,11 +158,11 @@ int sst_create_ipc_msg(struct ipc_post **arg, bool large)
 {
 	struct ipc_post *msg;
 
-	msg = kzalloc(sizeof(*msg), GFP_KERNEL);
+	msg = kzalloc(sizeof(*msg), GFP_ATOMIC);
 	if (!msg)
 		return -ENOMEM;
 	if (large) {
-		msg->mailbox_data = kzalloc(SST_MAILBOX_SIZE, GFP_KERNEL);
+		msg->mailbox_data = kzalloc(SST_MAILBOX_SIZE, GFP_ATOMIC);
 		if (!msg->mailbox_data) {
 			kfree(msg);
 			return -ENOMEM;
diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index 1f00d473793f..2fe1ce879123 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -487,6 +487,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev)
 	}
 
 	/* override plaform name, if required */
+	byt_cht_es8316_card.dev = dev;
 	platform_name = mach->mach_params.platform;
 
 	ret = snd_soc_fixup_dai_links_platform_name(&byt_cht_es8316_card,
@@ -567,7 +568,6 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev)
 		 (quirk & BYT_CHT_ES8316_MONO_SPEAKER) ? "mono" : "stereo",
 		 mic_name[BYT_CHT_ES8316_MAP(quirk)]);
 	byt_cht_es8316_card.long_name = long_name;
-	byt_cht_es8316_card.dev = dev;
 	sn