Merge tag 'sound-fix-6.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A collection of fixes for 6.6-rc1. All small and easy ones.

   - The corrections of the previous PCM iov_iter transitions

   - Regression fixes in MIDI 2.0 / USB changes

   - Various ASoC codec fixes for Cirrus, Realtek, WCD

   - ASoC AMD quirks and ASoC Intel AVS driver workaround"

* tag 'sound-fix-6.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (21 commits)
  ALSA: hda/realtek - ALC287 I2S speaker platform support
  ASoC: amd: yc: Fix a non-functional mic on Lenovo 82TL
  ASoC: Intel: avs: Provide support for fallback topology
  ALSA: seq: Fix snd_seq_expand_var_event() call to user-space
  ALSA: usb-audio: Fix potential memory leaks at error path for UMP open
  ALSA: hda/cirrus: Fix broken audio on hardware with two CS42L42 codecs.
  ASoC: rt5645: NULL pointer access when removing jack
  ASoC: amd: yc: Add DMI entries to support Victus by HP Gaming Laptop 15-fb0xxx (8A3E)
  MAINTAINERS: Update the MAINTAINERS enties for TEXAS INSTRUMENTS ASoC DRIVERS
  ALSA: sb: Fix wrong argument in commented code
  ALSA: pcm: Fix error checks of default read/write copy ops
  ASoC: Name iov_iter argument as iterator instead of buffer
  ASoC: dmaengine: Drop unused iov_iter for process callback
  ALSA: hda/tas2781: Use standard clamp() macro
  ASoC: cs35l56: Waiting for firmware to boot must be tolerant of I/O errors
  ASoC: dt-bindings: fsl_easrc: Add support for imx8mp-easrc
  ASoC: cs42l43: Fix missing error code in cs42l43_codec_probe()
  ASoC: cs35l45: Rename DACPCM1 Source control
  ASoC: cs35l45: Fix "Dead assigment" warning
  ASoC: cs35l45: Add support for Chip ID 0x35A460
  ...

Documentation/devicetree/bindings/sound/fsl,easrc.yaml

@@ -14,7 +14,13 @@ properties:
     pattern: "^easrc@.*"
 
   compatible:
-    const: fsl,imx8mn-easrc
+    oneOf:
+      - enum:
+          - fsl,imx8mn-easrc
+      - items:
+          - enum:
+              - fsl,imx8mp-easrc
+          - const: fsl,imx8mn-easrc
 
   reg:
     maxItems: 1

MAINTAINERS

@@ -21244,7 +21244,7 @@ F:	sound/soc/ti/
 TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS
 M:	Shenghao Ding <shenghao-ding@ti.com>
 M:	Kevin Lu <kevin-lu@ti.com>
-M:	Baojun Xu <x1077012@ti.com>
+M:	Baojun Xu <baojun.xu@ti.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Maintained
 F:	Documentation/devicetree/bindings/sound/tas2552.txt

include/sound/dmaengine_pcm.h

@@ -142,7 +142,7 @@ struct snd_dmaengine_pcm_config {
 			struct snd_pcm_substream *substream);
 	int (*process)(struct snd_pcm_substream *substream,
 		       int channel, unsigned long hwoff,
-		       struct iov_iter *buf, unsigned long bytes);
+		       unsigned long bytes);
 	dma_filter_fn compat_filter_fn;
 	struct device *dma_dev;
 	const char *chan_names[SNDRV_PCM_STREAM_LAST + 1];

include/sound/soc-component.h

@@ -139,7 +139,7 @@ struct snd_soc_component_driver {
 		struct snd_pcm_audio_tstamp_report *audio_tstamp_report);
 	int (*copy)(struct snd_soc_component *component,
 		    struct snd_pcm_substream *substream, int channel,
-		    unsigned long pos, struct iov_iter *buf,
+		    unsigned long pos, struct iov_iter *iter,
 		    unsigned long bytes);
 	struct page *(*page)(struct snd_soc_component *component,
 			     struct snd_pcm_substream *substream,
@@ -511,7 +511,7 @@ int snd_soc_pcm_component_ioctl(struct snd_pcm_substream *substream,
 int snd_soc_pcm_component_sync_stop(struct snd_pcm_substream *substream);
 int snd_soc_pcm_component_copy(struct snd_pcm_substream *substream,
 			       int channel, unsigned long pos,
-			       struct iov_iter *buf, unsigned long bytes);
+			       struct iov_iter *iter, unsigned long bytes);
 struct page *snd_soc_pcm_component_page(struct snd_pcm_substream *substream,
 					unsigned long offset);
 int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream,

sound/core/pcm_lib.c

@@ -1992,8 +1992,8 @@ static int default_write_copy(struct snd_pcm_substream *substream,
 			      int channel, unsigned long hwoff,
 			      struct iov_iter *iter, unsigned long bytes)
 {
-	if (!copy_from_iter(get_dma_ptr(substream->runtime, channel, hwoff),
-			    bytes, iter))
+	if (copy_from_iter(get_dma_ptr(substream->runtime, channel, hwoff),
+			   bytes, iter) != bytes)
 		return -EFAULT;
 	return 0;
 }
@@ -2025,8 +2025,8 @@ static int default_read_copy(struct snd_pcm_substream *substream,
 			     int channel, unsigned long hwoff,
 			     struct iov_iter *iter, unsigned long bytes)
 {
-	if (!copy_to_iter(get_dma_ptr(substream->runtime, channel, hwoff),
-			  bytes, iter))
+	if (copy_to_iter(get_dma_ptr(substream->runtime, channel, hwoff),
+			 bytes, iter) != bytes)
 		return -EFAULT;
 	return 0;
 }

sound/core/seq/seq_memory.c

@@ -187,8 +187,13 @@ int snd_seq_expand_var_event(const struct snd_seq_event *event, int count, char
 	err = expand_var_event(event, 0, len, buf, in_kernel);
 	if (err < 0)
 		return err;
-	if (len != newlen)
+	if (len != newlen) {
+		if (in_kernel)
 			memset(buf + len, 0, newlen - len);
+		else if (clear_user((__force void __user *)buf + len,
+				    newlen - len))
+			return -EFAULT;
+	}
 	return newlen;
 }
 EXPORT_SYMBOL(snd_seq_expand_var_event);

sound/isa/sb/emu8000_pcm.c

@@ -456,7 +456,7 @@ static int emu8k_pcm_silence(struct snd_pcm_substream *subs,
 	/* convert to word unit */
 	pos = (pos << 1) + rec->loop_start[voice];
 	count <<= 1;
-	LOOP_WRITE(rec, pos, USER_SOCKPTR(NULL), count);
+	LOOP_WRITE(rec, pos, NULL, count);
 	return 0;
 }
 

sound/pci/hda/patch_cs8409.c

@@ -888,7 +888,7 @@ static void cs42l42_resume(struct sub_codec *cs42l42)
 
 	/* Initialize CS42L42 companion codec */
 	cs8409_i2c_bulk_write(cs42l42, cs42l42->init_seq, cs42l42->init_seq_num);
-	usleep_range(30000, 35000);
+	msleep(CS42L42_INIT_TIMEOUT_MS);
 
 	/* Clear interrupts, by reading interrupt status registers */
 	cs8409_i2c_bulk_read(cs42l42, irq_regs, ARRAY_SIZE(irq_regs));

sound/pci/hda/patch_cs8409.h

@@ -229,6 +229,7 @@ enum cs8409_coefficient_index_registers {
 #define CS42L42_I2C_SLEEP_US			(2000)
 #define CS42L42_PDN_TIMEOUT_US			(250000)
 #define CS42L42_PDN_SLEEP_US			(2000)
+#define CS42L42_INIT_TIMEOUT_MS			(45)
 #define CS42L42_FULL_SCALE_VOL_MASK		(2)
 #define CS42L42_FULL_SCALE_VOL_0DB		(1)
 #define CS42L42_FULL_SCALE_VOL_MINUS6DB		(0)

sound/pci/hda/patch_realtek.c

@@ -7057,6 +7057,27 @@ static void alc295_fixup_dell_inspiron_top_speakers(struct hda_codec *codec,
 	}
 }
 
+/* Forcibly assign NID 0x03 to HP while NID 0x02 to SPK */
+static void alc287_fixup_bind_dacs(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+	static const hda_nid_t conn[] = { 0x02, 0x03 }; /* exclude 0x06 */
+	static const hda_nid_t preferred_pairs[] = {
+		0x17, 0x02, 0x21, 0x03, 0
+	};
+
+	if (action != HDA_FIXUP_ACT_PRE_PROBE)
+		return;
+
+	snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn);
+	spec->gen.preferred_dacs = preferred_pairs;
+	spec->gen.auto_mute_via_amp = 1;
+	snd_hda_codec_write_cache(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			    0x0); /* Make sure 0x14 was disable */
+}
+
+
 enum {
 	ALC269_FIXUP_GPIO2,
 	ALC269_FIXUP_SONY_VAIO,
@@ -7319,6 +7340,7 @@ enum {
 	ALC287_FIXUP_TAS2781_I2C,
 	ALC245_FIXUP_HP_MUTE_LED_COEFBIT,
 	ALC245_FIXUP_HP_X360_MUTE_LEDS,
+	ALC287_FIXUP_THINKPAD_I2S_SPK,
 };
 
 /* A special fixup for Lenovo C940 and Yoga Duet 7;
@@ -9413,6 +9435,10 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC245_FIXUP_HP_GPIO_LED
 	},
+	[ALC287_FIXUP_THINKPAD_I2S_SPK] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc287_fixup_bind_dacs,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -10544,6 +10570,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x17, 0x90170111},
 		{0x19, 0x03a11030},
 		{0x21, 0x03211020}),
+	SND_HDA_PIN_QUIRK(0x10ec0287, 0x17aa, "Lenovo", ALC287_FIXUP_THINKPAD_I2S_SPK,
+		{0x17, 0x90170110},
+		{0x19, 0x03a11030},
+		{0x21, 0x03211020}),
 	SND_HDA_PIN_QUIRK(0x10ec0286, 0x1025, "Acer", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE,
 		{0x12, 0x90a60130},
 		{0x17, 0x90170110},

sound/pci/hda/tas2781_hda_i2c.c

@@ -173,16 +173,6 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static int tasdevice_hda_clamp(int val, int max)
-{
-	if (val > max)
-		val = max;
-
-	if (val < 0)
-		val = 0;
-	return val;
-}
-
 static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol,
 		struct snd_ctl_elem_value *ucontrol)
 {
@@ -191,7 +181,7 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol,
 	int max = tas_priv->rcabin.ncfgs - 1;
 	int val, ret = 0;
 
-	val = tasdevice_hda_clamp(nr_profile, max);
+	val = clamp(nr_profile, 0, max);
 
 	if (tas_priv->rcabin.profile_cfg_id != val) {
 		tas_priv->rcabin.profile_cfg_id = val;
@@ -248,7 +238,7 @@ static int tasdevice_program_put(struct snd_kcontrol *kcontrol,
 	int max = tas_fw->nr_programs - 1;
 	int val, ret = 0;
 
-	val = tasdevice_hda_clamp(nr_program, max);
+	val = clamp(nr_program, 0, max);
 
 	if (tas_priv->cur_prog != val) {
 		tas_priv->cur_prog = val;
@@ -277,7 +267,7 @@ static int tasdevice_config_put(struct snd_kcontrol *kcontrol,
 	int max = tas_fw->nr_configurations - 1;
 	int val, ret = 0;
 
-	val = tasdevice_hda_clamp(nr_config, max);
+	val = clamp(nr_config, 0, max);
 
 	if (tas_priv->cur_conf != val) {
 		tas_priv->cur_conf = val;

sound/soc/amd/yc/acp6x-mach.c

@@ -213,6 +213,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "21J6"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "82TL"),
+		}
+	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {
@@ -325,6 +332,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_BOARD_NAME, "8A22"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "HP"),
+			DMI_MATCH(DMI_BOARD_NAME, "8A3E"),
+		}
+	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {

sound/soc/atmel/mchp-pdmc.c

@@ -954,7 +954,7 @@ static int mchp_pdmc_dt_init(struct mchp_pdmc *dd)
 /* used to clean the channel index found on RHR's MSB */
 static int mchp_pdmc_process(struct snd_pcm_substream *substream,
 			     int channel, unsigned long hwoff,
-			     struct iov_iter *buf, unsigned long bytes)
+			     unsigned long bytes)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	u8 *dma_ptr = runtime->dma_area + hwoff +

sound/soc/codecs/Kconfig

@@ -1968,11 +1968,15 @@ config SND_SOC_UDA1380
 	tristate
 	depends on I2C
 
+config SND_SOC_WCD_CLASSH
+	tristate
+
 config SND_SOC_WCD9335
 	tristate "WCD9335 Codec"
 	depends on SLIMBUS
 	select REGMAP_SLIMBUS
 	select REGMAP_IRQ
+	select SND_SOC_WCD_CLASSH
 	help
 	  The WCD9335 is a standalone Hi-Fi audio CODEC IC, supports
 	  Qualcomm Technologies, Inc. (QTI) multimedia solutions,
@@ -1987,6 +1991,7 @@ config SND_SOC_WCD934X
 	depends on SLIMBUS
 	select REGMAP_IRQ
 	select REGMAP_SLIMBUS
+	select SND_SOC_WCD_CLASSH
 	select SND_SOC_WCD_MBHC
 	depends on MFD_WCD934X || COMPILE_TEST
 	help
@@ -1997,6 +2002,7 @@ config SND_SOC_WCD938X
 	depends on SND_SOC_WCD938X_SDW
 	tristate
 	depends on SOUNDWIRE || !SOUNDWIRE
+	select SND_SOC_WCD_CLASSH
 
 config SND_SOC_WCD938X_SDW
 	tristate "WCD9380/WCD9385 Codec - SDW"

sound/soc/codecs/Makefile

@@ -303,10 +303,11 @@ snd-soc-twl4030-objs := twl4030.o
 snd-soc-twl6040-objs := twl6040.o
 snd-soc-uda1334-objs := uda1334.o
 snd-soc-uda1380-objs := uda1380.o
+snd-soc-wcd-classh-objs := wcd-clsh-v2.o
 snd-soc-wcd-mbhc-objs := wcd-mbhc-v2.o
-snd-soc-wcd9335-objs := wcd-clsh-v2.o wcd9335.o
-snd-soc-wcd934x-objs := wcd-clsh-v2.o wcd934x.o
-snd-soc-wcd938x-objs := wcd938x.o wcd-clsh-v2.o
+snd-soc-wcd9335-objs := wcd9335.o
+snd-soc-wcd934x-objs := wcd934x.o
+snd-soc-wcd938x-objs := wcd938x.o
 snd-soc-wcd938x-sdw-objs := wcd938x-sdw.o
 snd-soc-wl1273-objs := wl1273.o
 snd-soc-wm-adsp-objs := wm_adsp.o
@@ -685,6 +686,7 @@ obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
 obj-$(CONFIG_SND_SOC_TWL6040)	+= snd-soc-twl6040.o
 obj-$(CONFIG_SND_SOC_UDA1334)	+= snd-soc-uda1334.o
 obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
+obj-$(CONFIG_SND_SOC_WCD_CLASSH)	+= snd-soc-wcd-classh.o
 obj-$(CONFIG_SND_SOC_WCD_MBHC)	+= snd-soc-wcd-mbhc.o
 obj-$(CONFIG_SND_SOC_WCD9335)	+= snd-soc-wcd9335.o
 obj-$(CONFIG_SND_SOC_WCD934X)	+= snd-soc-wcd934x.o

sound/soc/codecs/cs35l45.c

@@ -279,7 +279,7 @@ static const struct snd_kcontrol_new cs35l45_dsp_muxes[] = {
 };
 
 static const struct snd_kcontrol_new cs35l45_dac_muxes[] = {
-	SOC_DAPM_ENUM("DACPCM1 Source", cs35l45_dacpcm_enums[0]),
+	SOC_DAPM_ENUM("DACPCM Source", cs35l45_dacpcm_enums[0]),
 };
 
 static const struct snd_soc_dapm_widget cs35l45_dapm_widgets[] = {
@@ -333,7 +333,7 @@ static const struct snd_soc_dapm_widget cs35l45_dapm_widgets[] = {
 	SND_SOC_DAPM_MUX("DSP_RX7 Source", SND_SOC_NOPM, 0, 0, &cs35l45_dsp_muxes[6]),
 	SND_SOC_DAPM_MUX("DSP_RX8 Source", SND_SOC_NOPM, 0, 0, &cs35l45_dsp_muxes[7]),
 
-	SND_SOC_DAPM_MUX("DACPCM1 Source", SND_SOC_NOPM, 0, 0, &cs35l45_dac_muxes[0]),
+	SND_SOC_DAPM_MUX("DACPCM Source", SND_SOC_NOPM, 0, 0, &cs35l45_dac_muxes[0]),
 
 	SND_SOC_DAPM_OUT_DRV("AMP", SND_SOC_NOPM, 0, 0, NULL, 0),
 
@@ -403,7 +403,7 @@ static const struct snd_soc_dapm_route cs35l45_dapm_routes[] = {
 	{ "ASP_RX1", NULL, "ASP_EN" },
 	{ "ASP_RX2", NULL, "ASP_EN" },
 
-	{ "AMP", NULL, "DACPCM1 Source"},
+	{ "AMP", NULL, "DACPCM Source"},
 	{ "AMP", NULL, "GLOBAL_EN"},
 
 	CS35L45_DSP_MUX_ROUTE("DSP_RX1"),
@@ -427,7 +427,7 @@ static const struct snd_soc_dapm_route cs35l45_dapm_routes[] = {
 	{"DSP1 Preload", NULL, "DSP1 Preloader"},
 	{"DSP1", NULL, "DSP1 Preloader"},
 
-	CS35L45_DAC_MUX_ROUTE("DACPCM1"),
+	CS35L45_DAC_MUX_ROUTE("DACPCM"),
 
 	{ "SPK", NULL, "AMP"},
 };
@@ -969,7 +969,7 @@ static irqreturn_t cs35l45_dsp_virt2_mbox_cb(int irq, void *data)
 
 	ret = regmap_read(cs35l45->regmap, CS35L45_DSP_VIRT2_MBOX_3, &mbox_val);
 	if (!ret && mbox_val)
-		ret = cs35l45_dsp_virt2_mbox3_irq_handle(cs35l45, mbox_val & CS35L45_MBOX3_CMD_MASK,
+		cs35l45_dsp_virt2_mbox3_irq_handle(cs35l45, mbox_val & CS35L45_MBOX3_CMD_MASK,
 				(mbox_val & CS35L45_MBOX3_DATA_MASK) >> CS35L45_MBOX3_DATA_SHIFT);
 
 	/* Handle DSP trace log IRQ */
@@ -1078,6 +1078,7 @@ static int cs35l45_initialize(struct cs35l45_private *cs35l45)
 
 	switch (dev_id[0]) {
 	case 0x35A450:
+	case 0x35A460:
 		break;
 	default:
 		dev_err(cs35l45->dev, "Bad DEVID 0x%x\n", dev_id[0]);

sound/soc/codecs/cs35l56-shared.c

@@ -243,26 +243,27 @@ int cs35l56_wait_for_firmware_boot(struct cs35l56_base *cs35l56_base)
 {
 	unsigned int reg;
 	unsigned int val;
-	int ret;
+	int read_ret, poll_ret;
 
 	if (cs35l56_base->rev < CS35L56_REVID_B0)
 		reg = CS35L56_DSP1_HALO_STATE_A1;
 	else
 		reg = CS35L56_DSP1_HALO_STATE;
 
-	ret = regmap_read_poll_timeout(cs35l56_base->regmap, reg,
-				       val,
+	/*
+	 * This can't be a regmap_read_poll_timeout() because cs35l56 will NAK
+	 * I2C until it has booted which would terminate the poll
+	 */
+	poll_ret = read_poll_timeout(regmap_read, read_ret,
 				     (val < 0xFFFF) && (val >= CS35L56_HALO_STATE_BOOT_DONE),
 				     CS35L56_HALO_STATE_POLL_US,
-				       CS35L56_HALO_STATE_TIMEOUT_US);
-
-	if ((ret < 0) && (ret != -ETIMEDOUT)) {
-		dev_err(cs35l56_base->dev, "Failed to read HALO_STATE: %d\n", ret);
-		return ret;
-	}
+				     CS35L56_HALO_STATE_TIMEOUT_US,
+				     false,
+				     cs35l56_base->regmap, reg, &val);
 
-	if ((ret == -ETIMEDOUT) || (val != CS35L56_HALO_STATE_BOOT_DONE)) {
-		dev_err(cs35l56_base->dev, "Firmware boot fail: HALO_STATE=%#x\n", val);
+	if (poll_ret) {
+		dev_err(cs35l56_base->dev, "Firmware boot timed out(%d): HALO_STATE=%#x\n",
+			read_ret, val);
 		return -EIO;
 	}
 

sound/soc/codecs/cs42l43.c

@@ -2205,7 +2205,8 @@ static int cs42l43_codec_probe(struct platform_device *pdev)
 	// Don't use devm as we need to get against the MFD device
 	priv->mclk = clk_get_optional(cs42l43->dev, "mclk");
 	if (IS_ERR(priv->mclk)) {
-		dev_err_probe(priv->dev, PTR_ERR(priv->mclk), "Failed to get mclk\n");
+		ret = PTR_ERR(priv->mclk);
+		dev_err_probe(priv->dev, ret, "Failed to get mclk\n");
 		goto err_pm;
 	}
 

sound/soc/codecs/rt5645.c

@@ -3269,13 +3269,17 @@ static int rt5645_component_set_jack(struct snd_soc_component *component,
 {
 	struct snd_soc_jack *mic_jack = NULL;
 	struct snd_soc_jack *btn_jack = NULL;
-	int *type = (int *)data;
+	int type;
 
-	if (*type & SND_JACK_MICROPHONE)
+	if (hs_jack) {
+		type = *(int *)data;
+
+		if (type & SND_JACK_MICROPHONE)
 			mic_jack = hs_jack;
-	if (*type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 |
+		if (type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 |
 			SND_JACK_BTN_2 | SND_JACK_BTN_3))
 			btn_jack = hs_jack;
+	}
 
 	return rt5645_set_jack_detect(component, hs_jack, mic_jack, btn_jack);
 }

sound/soc/codecs/wcd-clsh-v2.c

@@ -355,6 +355,7 @@ void wcd_clsh_set_hph_mode(struct wcd_clsh_ctrl *ctrl, int mode)
 		wcd_clsh_v2_set_hph_mode(comp, mode);
 
 }
+EXPORT_SYMBOL_GPL(wcd_clsh_set_hph_mode);
 
 static void wcd_clsh_set_flyback_current(struct snd_soc_component *comp,
 					 int mode)
@@ -869,11 +870,13 @@ int wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl,
 
 	return 0;
 }
+EXPORT_SYMBOL_GPL(wcd_clsh_ctrl_set_state);
 
 int wcd_clsh_ctrl_get_state(struct wcd_clsh_ctrl *ctrl)
 {
 	return ctrl->state;
 }
+EXPORT_SYMBOL_GPL(wcd_clsh_ctrl_get_state);
 
 struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc(struct snd_soc_component *comp,
 					  int version)
@@ -890,8 +893,13 @@ struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc(struct snd_soc_component *comp,
 
 	return ctrl;
 }
+EXPORT_SYMBOL_GPL(wcd_clsh_ctrl_alloc);
 
 void wcd_clsh_ctrl_free(struct wcd_clsh_ctrl *ctrl)
 {
 	kfree(ctrl);
 }
+EXPORT_SYMBOL_GPL(wcd_clsh_ctrl_free);
+
+MODULE_DESCRIPTION("WCD93XX Class-H driver");
+MODULE_LICENSE("GPL");

sound/soc/intel/avs/pcm.c

@@ -796,6 +796,28 @@ static int avs_component_probe(struct snd_soc_component *component)
 
 	ret = avs_load_topology(component, filename);
 	kfree(filename);
+	if (ret == -ENOENT && !strncmp(mach->tplg_filename, "hda-", 4)) {
+		unsigned int vendor_id;
+
+		if (sscanf(mach->tplg_filename, "hda-%08x-tplg.bin", &vendor_id) != 1)
+			return ret;
+
+		if (((vendor_id >> 16) & 0xFFFF) == 0x8086)
+			mach->tplg_filename = devm_kasprintf(adev->dev, GFP_KERNEL,
+							     "hda-8086-generic-tplg.bin");
+		else
+			mach->tplg_filename = devm_kasprintf(adev->dev, GFP_KERNEL,
+							     "hda-generic-tplg.bin");
+
+		filename = kasprintf(GFP_KERNEL, "%s/%s", component->driver->topology_name_prefix,
+				     mach->tplg_filename);
+		if (!filename)
+			return -ENOMEM;
+
+		dev_info(card->dev, "trying to load fallback topology %s\n", mach->tplg_filename);
+		ret = avs_load_topology(component, filename);
+		kfree(filename);
+	}
 	if (ret < 0)
 		return ret;
 

sound/soc/soc-component.c

@@ -1054,7 +1054,7 @@ int snd_soc_pcm_component_sync_stop(struct snd_pcm_substream *substream)
 
 int snd_soc_pcm_component_copy(struct snd_pcm_substream *substream,
 			       int channel, unsigned long pos,
-			       struct iov_iter *buf, unsigned long bytes)
+			       struct iov_iter *iter, unsigned long bytes)
 {
 	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_component *component;
@@ -1065,7 +1065,7 @@ int snd_soc_pcm_component_copy(struct snd_pcm_substream *substream,
 		if (component->driver->copy)
 			return soc_component_ret(component,
 				component->driver->copy(component, substream,
-					channel, pos, buf, bytes));
+					channel, pos, iter, bytes));
 
 	return -EINVAL;
 }

sound/soc/soc-generic-dmaengine-pcm.c

@@ -290,29 +290,29 @@ static snd_pcm_uframes_t dmaengine_pcm_pointer(
 static int dmaengine_copy(struct snd_soc_component *component,
 			  struct snd_pcm_substream *substream,
 			  int channel, unsigned long hwoff,
-			  struct iov_iter *buf, unsigned long bytes)
+			  struct iov_iter *iter, unsigned long bytes)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct dmaengine_pcm *pcm = soc_component_to_pcm(component);
 	int (*process)(struct snd_pcm_substream *substream,
 		       int channel, unsigned long hwoff,
-		       struct iov_iter *buf, unsigned long bytes) = pcm->config->process;
+		       unsigned long bytes) = pcm->config->process;
 	bool is_playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	void *dma_ptr = runtime->dma_area + hwoff +
 			channel * (runtime->dma_bytes / runtime->channels);
 
 	if (is_playback)
-		if (copy_from_iter(dma_ptr, bytes, buf) != bytes)
+		if (copy_from_iter(dma_ptr, bytes, iter) != bytes)
 			return -EFAULT;
 
 	if (process) {
-		int ret = process(substream, channel, hwoff, buf, bytes);
+		int ret = process(substream, channel, hwoff, bytes);
 		if (ret < 0)
 			return ret;
 	}
 
 	if (!is_playback)
-		if (copy_to_iter(dma_ptr, bytes, buf) != bytes)
+		if (copy_to_iter(dma_ptr, bytes, iter) != bytes)
 			return -EFAULT;
 
 	return 0;

sound/soc/stm/stm32_sai_sub.c

@@ -1246,7 +1246,7 @@ static const struct snd_soc_dai_ops stm32_sai_pcm_dai_ops2 = {
 
 static int stm32_sai_pcm_process_spdif(struct snd_pcm_substream *substream,
 				       int channel, unsigned long hwoff,
-				       struct iov_iter *buf, unsigned long bytes)
+				       unsigned long bytes)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);

sound/usb/midi2.c

@@ -265,7 +265,7 @@ static void free_midi_urbs(struct snd_usb_midi2_endpoint *ep)
 
 	if (!ep)
 		return;
-	for (i = 0; i < ep->num_urbs; ++i) {
+	for (i = 0; i < NUM_URBS; ++i) {
 		ctx = &ep->urbs[i];
 		if (!ctx->urb)
 			break;
@@ -279,6 +279,7 @@ static void free_midi_urbs(struct snd_usb_midi2_endpoint *ep)
 }
 
 /* allocate URBs for an EP */
+/* the callers should handle allocation errors via free_midi_urbs() */
 static int alloc_midi_urbs(struct snd_usb_midi2_endpoint *ep)
 {
 	struct snd_usb_midi2_urb *ctx;
@@ -351,9 +352,11 @@ static int snd_usb_midi_v2_open(struct snd_ump_endpoint *ump, int dir)
 		return -EIO;
 	if (ep->direction == STR_OUT) {
 		err = alloc_midi_urbs(ep);
-		if (err)
+		if (err) {
+			free_midi_urbs(ep);
 			return err;
 		}
+	}
 	return 0;
 }