Merge tag 'sound-6.2-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This became a slightly big update, but it's more or less expected, as
  the first batch after holidays.

  All changes (but for the last two last-minute fixes) have been stewed
  in linux-next long enough, so it's fairly safe to take:

   - PCM UAF fix in 32bit compat layer

   - ASoC board-specific fixes for Intel, AMD, Medathek, Qualcomm

   - SOF power management fixes

   - ASoC Intel link failure fixes

   - A series of fixes for USB-audio regressions

   - CS35L41 HD-audio codec regression fixes

   - HD-audio device-specific fixes / quirks

  Note that one SPI patch has been taken in ASoC subtree mistakenly, and
  the same fix is found in spi tree, but it should be OK to apply"

* tag 'sound-6.2-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (39 commits)
  ALSA: pcm: Move rwsem lock inside snd_ctl_elem_read to prevent UAF
  ALSA: usb-audio: Fix possible NULL pointer dereference in snd_usb_pcm_has_fixed_rate()
  ALSA: hda/realtek: Enable mute/micmute LEDs on HP Spectre x360 13-aw0xxx
  ASoC: fsl-asoc-card: Fix naming of AC'97 CODEC widgets
  ASoC: fsl_ssi: Rename AC'97 streams to avoid collisions with AC'97 CODEC
  ALSA: hda/hdmi: Add a HP device 0x8715 to force connect list
  ALSA: control-led: use strscpy in set_led_id()
  ALSA: usb-audio: Always initialize fixed_rate in snd_usb_find_implicit_fb_sync_format()
  ASoC: dt-bindings: qcom,lpass-tx-macro: correct clocks on SC7280
  ASoC: dt-bindings: qcom,lpass-wsa-macro: correct clocks on SM8250
  ASoC: qcom: Fix building APQ8016 machine driver without SOUNDWIRE
  ALSA: hda: cs35l41: Check runtime suspend capability at runtime_idle
  ALSA: hda: cs35l41: Don't return -EINVAL from system suspend/resume
  ASoC: fsl_micfil: Correct the number of steps on SX controls
  ALSA: hda/realtek: fix mute/micmute LEDs don't work for a HP platform
  Revert "ALSA: usb-audio: Drop superfluous interface setup at parsing"
  ALSA: usb-audio: More refactoring of hw constraint rules
  ALSA: usb-audio: Relax hw constraints for implicit fb sync
  ALSA: usb-audio: Make sure to stop endpoints before closing EPs
  ALSA: hda - Enable headset mic on another Dell laptop with ALC3254
  ...

Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml

@@ -16,6 +16,7 @@ properties:
   compatible:
     enum:
       - mediatek,mt8186-mt6366-rt1019-rt5682s-sound
+      - mediatek,mt8186-mt6366-rt5682s-max98360-sound
 
   mediatek,platform:
     $ref: "/schemas/types.yaml#/definitions/phandle"

Documentation/devicetree/bindings/sound/qcom,lpass-tx-macro.yaml

@@ -30,7 +30,9 @@ properties:
     const: 0
 
   clocks:
-    maxItems: 5
+    oneOf:
+      - maxItems: 3
+      - maxItems: 5
 
   clock-names:
     oneOf:

Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml

@@ -9,9 +9,6 @@ title: LPASS(Low Power Audio Subsystem) VA Macro audio codec
 maintainers:
   - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
 
-allOf:
-  - $ref: dai-common.yaml#
-
 properties:
   compatible:
     enum:
@@ -30,15 +27,12 @@ properties:
     const: 0
 
   clocks:
-    maxItems: 5
+    minItems: 5
+    maxItems: 6
 
   clock-names:
-    items:
-      - const: mclk
-      - const: npl
-      - const: macro
-      - const: dcodec
-      - const: fsgen
+    minItems: 5
+    maxItems: 6
 
   clock-output-names:
     maxItems: 1
@@ -55,10 +49,51 @@ required:
   - reg
   - "#sound-dai-cells"
 
+allOf:
+  - $ref: dai-common.yaml#
+
+  - if:
+      properties:
+        compatible:
+          enum:
+            - qcom,sc7280-lpass-wsa-macro
+            - qcom,sm8450-lpass-wsa-macro
+            - qcom,sc8280xp-lpass-wsa-macro
+    then:
+      properties:
+        clocks:
+          maxItems: 5
+        clock-names:
+          items:
+            - const: mclk
+            - const: npl
+            - const: macro
+            - const: dcodec
+            - const: fsgen
+
+  - if:
+      properties:
+        compatible:
+          enum:
+            - qcom,sm8250-lpass-wsa-macro
+    then:
+      properties:
+        clocks:
+          minItems: 6
+        clock-names:
+          items:
+            - const: mclk
+            - const: npl
+            - const: macro
+            - const: dcodec
+            - const: va
+            - const: fsgen
+
 unevaluatedProperties: false
 
 examples:
   - |
+    #include <dt-bindings/clock/qcom,sm8250-lpass-aoncc.h>
     #include <dt-bindings/sound/qcom,q6afe.h>
     codec@3240000 {
       compatible = "qcom,sm8250-lpass-wsa-macro";
@@ -69,7 +104,8 @@ examples:
                <&audiocc 0>,
                <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
                <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>,
+               <&aoncc LPASS_CDC_VA_MCLK>,
                <&vamacro>;
-      clock-names = "mclk", "npl", "macro", "dcodec", "fsgen";
+      clock-names = "mclk", "npl", "macro", "dcodec", "va", "fsgen";
       clock-output-names = "mclk";
     };

sound/core/control.c

@@ -1203,14 +1203,19 @@ static int snd_ctl_elem_read(struct snd_card *card,
 	const u32 pattern = 0xdeadbeef;
 	int ret;
 
+	down_read(&card->controls_rwsem);
 	kctl = snd_ctl_find_id(card, &control->id);
-	if (kctl == NULL)
-		return -ENOENT;
+	if (kctl == NULL) {
+		ret = -ENOENT;
+		goto unlock;
+	}
 
 	index_offset = snd_ctl_get_ioff(kctl, &control->id);
 	vd = &kctl->vd[index_offset];
-	if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_READ) || kctl->get == NULL)
-		return -EPERM;
+	if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_READ) || kctl->get == NULL) {
+		ret = -EPERM;
+		goto unlock;
+	}
 
 	snd_ctl_build_ioff(&control->id, kctl, index_offset);
 
@@ -1220,7 +1225,7 @@ static int snd_ctl_elem_read(struct snd_card *card,
 	info.id = control->id;
 	ret = __snd_ctl_elem_info(card, kctl, &info, NULL);
 	if (ret < 0)
-		return ret;
+		goto unlock;
 #endif
 
 	if (!snd_ctl_skip_validation(&info))
@@ -1230,7 +1235,7 @@ static int snd_ctl_elem_read(struct snd_card *card,
 		ret = kctl->get(kctl, control);
 	snd_power_unref(card);
 	if (ret < 0)
-		return ret;
+		goto unlock;
 	if (!snd_ctl_skip_validation(&info) &&
 	    sanity_check_elem_value(card, control, &info, pattern) < 0) {
 		dev_err(card->dev,
@@ -1238,8 +1243,11 @@ static int snd_ctl_elem_read(struct snd_card *card,
 			control->id.iface, control->id.device,
 			control->id.subdevice, control->id.name,
 			control->id.index);
-		return -EINVAL;
+		ret = -EINVAL;
+		goto unlock;
 	}
+unlock:
+	up_read(&card->controls_rwsem);
 	return ret;
 }
 
@@ -1253,9 +1261,7 @@ static int snd_ctl_elem_read_user(struct snd_card *card,
 	if (IS_ERR(control))
 		return PTR_ERR(control);
 
-	down_read(&card->controls_rwsem);
 	result = snd_ctl_elem_read(card, control);
-	up_read(&card->controls_rwsem);
 	if (result < 0)
 		goto error;
 

sound/core/control_led.c

@@ -530,12 +530,11 @@ static ssize_t set_led_id(struct snd_ctl_led_card *led_card, const char *buf, si
 			  bool attach)
 {
 	char buf2[256], *s, *os;
-	size_t len = max(sizeof(s) - 1, count);
 	struct snd_ctl_elem_id id;
 	int err;
 
-	strncpy(buf2, buf, len);
-	buf2[len] = '\0';
+	if (strscpy(buf2, buf, sizeof(buf2)) < 0)
+		return -E2BIG;
 	memset(&id, 0, sizeof(id));
 	id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
 	s = buf2;

sound/pci/hda/cs35l41_hda.c

@@ -598,8 +598,8 @@ static int cs35l41_system_suspend(struct device *dev)
 	dev_dbg(cs35l41->dev, "System Suspend\n");
 
 	if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST_NO_VSPK_SWITCH) {
-		dev_err(cs35l41->dev, "System Suspend not supported\n");
-		return -EINVAL;
+		dev_err_once(cs35l41->dev, "System Suspend not supported\n");
+		return 0; /* don't block the whole system suspend */
 	}
 
 	ret = pm_runtime_force_suspend(dev);
@@ -624,8 +624,8 @@ static int cs35l41_system_resume(struct device *dev)
 	dev_dbg(cs35l41->dev, "System Resume\n");
 
 	if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST_NO_VSPK_SWITCH) {
-		dev_err(cs35l41->dev, "System Resume not supported\n");
-		return -EINVAL;
+		dev_err_once(cs35l41->dev, "System Resume not supported\n");
+		return 0; /* don't block the whole system resume */
 	}
 
 	if (cs35l41->reset_gpio) {
@@ -647,6 +647,15 @@ static int cs35l41_system_resume(struct device *dev)
 	return ret;
 }
 
+static int cs35l41_runtime_idle(struct device *dev)
+{
+	struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev);
+
+	if (cs35l41->hw_cfg.bst_type == CS35L41_EXT_BOOST_NO_VSPK_SWITCH)
+		return -EBUSY; /* suspend not supported yet on this model */
+	return 0;
+}
+
 static int cs35l41_runtime_suspend(struct device *dev)
 {
 	struct cs35l41_hda *cs35l41 = dev_get_drvdata(dev);
@@ -1536,7 +1545,8 @@ void cs35l41_hda_remove(struct device *dev)
 EXPORT_SYMBOL_NS_GPL(cs35l41_hda_remove, SND_HDA_SCODEC_CS35L41);
 
 const struct dev_pm_ops cs35l41_hda_pm_ops = {
-	RUNTIME_PM_OPS(cs35l41_runtime_suspend, cs35l41_runtime_resume, NULL)
+	RUNTIME_PM_OPS(cs35l41_runtime_suspend, cs35l41_runtime_resume,
+		       cs35l41_runtime_idle)
 	SYSTEM_SLEEP_PM_OPS(cs35l41_system_suspend, cs35l41_system_resume)
 };
 EXPORT_SYMBOL_NS_GPL(cs35l41_hda_pm_ops, SND_HDA_SCODEC_CS35L41);

sound/pci/hda/patch_hdmi.c

@@ -1981,6 +1981,7 @@ static const struct snd_pci_quirk force_connect_list[] = {
 	SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1),
 	SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1),
 	SND_PCI_QUIRK(0x103c, 0x8711, "HP", 1),
+	SND_PCI_QUIRK(0x103c, 0x8715, "HP", 1),
 	SND_PCI_QUIRK(0x1462, 0xec94, "MS-7C94", 1),
 	SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", 1),
 	{}

sound/pci/hda/patch_realtek.c

@@ -3564,6 +3564,15 @@ static void alc256_init(struct hda_codec *codec)
 	hda_nid_t hp_pin = alc_get_hp_pin(spec);
 	bool hp_pin_sense;
 
+	if (spec->ultra_low_power) {
+		alc_update_coef_idx(codec, 0x03, 1<<1, 1<<1);
+		alc_update_coef_idx(codec, 0x08, 3<<2, 3<<2);
+		alc_update_coef_idx(codec, 0x08, 7<<4, 0);
+		alc_update_coef_idx(codec, 0x3b, 1<<15, 0);
+		alc_update_coef_idx(codec, 0x0e, 7<<6, 7<<6);
+		msleep(30);
+	}
+
 	if (!hp_pin)
 		hp_pin = 0x21;
 
@@ -3575,14 +3584,6 @@ static void alc256_init(struct hda_codec *codec)
 		msleep(2);
 
 	alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x1); /* Low power */
-	if (spec->ultra_low_power) {
-		alc_update_coef_idx(codec, 0x03, 1<<1, 1<<1);
-		alc_update_coef_idx(codec, 0x08, 3<<2, 3<<2);
-		alc_update_coef_idx(codec, 0x08, 7<<4, 0);
-		alc_update_coef_idx(codec, 0x3b, 1<<15, 0);
-		alc_update_coef_idx(codec, 0x0e, 7<<6, 7<<6);
-		msleep(30);
-	}
 
 	snd_hda_codec_write(codec, hp_pin, 0,
 			    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
@@ -3713,6 +3714,13 @@ static void alc225_init(struct hda_codec *codec)
 	hda_nid_t hp_pin = alc_get_hp_pin(spec);
 	bool hp1_pin_sense, hp2_pin_sense;
 
+	if (spec->ultra_low_power) {
+		alc_update_coef_idx(codec, 0x08, 0x0f << 2, 3<<2);
+		alc_update_coef_idx(codec, 0x0e, 7<<6, 7<<6);
+		alc_update_coef_idx(codec, 0x33, 1<<11, 0);
+		msleep(30);
+	}
+
 	if (spec->codec_variant != ALC269_TYPE_ALC287 &&
 		spec->codec_variant != ALC269_TYPE_ALC245)
 		/* required only at boot or S3 and S4 resume time */
@@ -3734,12 +3742,6 @@ static void alc225_init(struct hda_codec *codec)
 		msleep(2);
 
 	alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x1); /* Low power */
-	if (spec->ultra_low_power) {
-		alc_update_coef_idx(codec, 0x08, 0x0f << 2, 3<<2);
-		alc_update_coef_idx(codec, 0x0e, 7<<6, 7<<6);
-		alc_update_coef_idx(codec, 0x33, 1<<11, 0);
-		msleep(30);
-	}
 
 	if (hp1_pin_sense || spec->ultra_low_power)
 		snd_hda_codec_write(codec, hp_pin, 0,
@@ -4644,6 +4646,16 @@ static void alc285_fixup_hp_coef_micmute_led(struct hda_codec *codec,
 	}
 }
 
+static void alc285_fixup_hp_gpio_micmute_led(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+		spec->micmute_led_polarity = 1;
+	alc_fixup_hp_gpio_led(codec, action, 0, 0x04);
+}
+
 static void alc236_fixup_hp_coef_micmute_led(struct hda_codec *codec,
 				const struct hda_fixup *fix, int action)
 {
@@ -4665,6 +4677,13 @@ static void alc285_fixup_hp_mute_led(struct hda_codec *codec,
 	alc285_fixup_hp_coef_micmute_led(codec, fix, action);
 }
 
+static void alc285_fixup_hp_spectre_x360_mute_led(struct hda_codec *codec,
+				const struct hda_fixup *fix, int action)
+{
+	alc285_fixup_hp_mute_led_coefbit(codec, fix, action);
+	alc285_fixup_hp_gpio_micmute_led(codec, fix, action);
+}
+
 static void alc236_fixup_hp_mute_led(struct hda_codec *codec,
 				const struct hda_fixup *fix, int action)
 {
@@ -7106,6 +7125,7 @@ enum {
 	ALC285_FIXUP_ASUS_G533Z_PINS,
 	ALC285_FIXUP_HP_GPIO_LED,
 	ALC285_FIXUP_HP_MUTE_LED,
+	ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED,
 	ALC236_FIXUP_HP_GPIO_LED,
 	ALC236_FIXUP_HP_MUTE_LED,
 	ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF,
@@ -8486,6 +8506,10 @@ static const struct hda_fixup alc269_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc285_fixup_hp_mute_led,
 	},
+	[ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc285_fixup_hp_spectre_x360_mute_led,
+	},
 	[ALC236_FIXUP_HP_GPIO_LED] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = alc236_fixup_hp_gpio_led,
@@ -9239,6 +9263,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK),
 	SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS),
 	SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS),
+	SND_PCI_QUIRK(0x1028, 0x0c03, "Dell Precision 5340", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x0c19, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS),
 	SND_PCI_QUIRK(0x1028, 0x0c1a, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS),
 	SND_PCI_QUIRK(0x1028, 0x0c1b, "Dell Precision 3440", ALC236_FIXUP_DELL_DUAL_CODECS),
@@ -9327,6 +9352,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO),
 	SND_PCI_QUIRK(0x103c, 0x86e7, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1),
 	SND_PCI_QUIRK(0x103c, 0x86e8, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1),
+	SND_PCI_QUIRK(0x103c, 0x86f9, "HP Spectre x360 13-aw0xxx", ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x8720, "HP EliteBook x360 1040 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED),
@@ -9406,6 +9432,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8b5d, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x8b5e, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
+	SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
 	SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),

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

@@ -206,6 +206,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "UM5302TA"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "M5402RA"),
+		}
+	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {
@@ -220,6 +227,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "Redmi Book Pro 14 2022"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "Razer"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "Blade 14 (2022) - RZ09-0427"),
+		}
+	},
 	{}
 };
 

sound/soc/codecs/rt9120.c

@@ -177,8 +177,20 @@ static int rt9120_codec_probe(struct snd_soc_component *comp)
 	return 0;
 }
 
+static int rt9120_codec_suspend(struct snd_soc_component *comp)
+{
+	return pm_runtime_force_suspend(comp->dev);
+}
+
+static int rt9120_codec_resume(struct snd_soc_component *comp)
+{
+	return pm_runtime_force_resume(comp->dev);
+}
+
 static const struct snd_soc_component_driver rt9120_component_driver = {
 	.probe = rt9120_codec_probe,
+	.suspend = rt9120_codec_suspend,
+	.resume = rt9120_codec_resume,
 	.controls = rt9120_snd_controls,
 	.num_controls = ARRAY_SIZE(rt9120_snd_controls),
 	.dapm_widgets = rt9120_dapm_widgets,

sound/soc/codecs/wm8904.c

@@ -697,6 +697,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w,
 	int dcs_mask;
 	int dcs_l, dcs_r;
 	int dcs_l_reg, dcs_r_reg;
+	int an_out_reg;
 	int timeout;
 	int pwr_reg;
 
@@ -712,6 +713,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w,
 		dcs_mask = WM8904_DCS_ENA_CHAN_0 | WM8904_DCS_ENA_CHAN_1;
 		dcs_r_reg = WM8904_DC_SERVO_8;
 		dcs_l_reg = WM8904_DC_SERVO_9;
+		an_out_reg = WM8904_ANALOGUE_OUT1_LEFT;
 		dcs_l = 0;
 		dcs_r = 1;
 		break;
@@ -720,6 +722,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w,
 		dcs_mask = WM8904_DCS_ENA_CHAN_2 | WM8904_DCS_ENA_CHAN_3;
 		dcs_r_reg = WM8904_DC_SERVO_6;
 		dcs_l_reg = WM8904_DC_SERVO_7;
+		an_out_reg = WM8904_ANALOGUE_OUT2_LEFT;
 		dcs_l = 2;
 		dcs_r = 3;
 		break;
@@ -792,6 +795,10 @@ static int out_pga_event(struct snd_soc_dapm_widget *w,
 		snd_soc_component_update_bits(component, reg,
 				    WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP,
 				    WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP);
+
+		/* Update volume, requires PGA to be powered */
+		val = snd_soc_component_read(component, an_out_reg);
+		snd_soc_component_write(component, an_out_reg, val);
 		break;
 
 	case SND_SOC_DAPM_POST_PMU:

sound/soc/fsl/fsl-asoc-card.c

@@ -121,11 +121,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
 
 static const struct snd_soc_dapm_route audio_map_ac97[] = {
 	/* 1st half -- Normal DAPM routes */
-	{"Playback",  NULL, "AC97 Playback"},
-	{"AC97 Capture",  NULL, "Capture"},
+	{"AC97 Playback",  NULL, "CPU AC97 Playback"},
+	{"CPU AC97 Capture",  NULL, "AC97 Capture"},
 	/* 2nd half -- ASRC DAPM routes */
-	{"AC97 Playback",  NULL, "ASRC-Playback"},
-	{"ASRC-Capture",  NULL, "AC97 Capture"},
+	{"CPU AC97 Playback",  NULL, "ASRC-Playback"},
+	{"ASRC-Capture",  NULL, "CPU AC97 Capture"},
 };
 
 static const struct snd_soc_dapm_route audio_map_tx[] = {

sound/soc/fsl/fsl_micfil.c

@@ -315,21 +315,21 @@ static int hwvad_detected(struct snd_kcontrol *kcontrol,
 
 static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = {
 	SOC_SINGLE_SX_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL,
-			  MICFIL_OUTGAIN_CHX_SHIFT(0), 0xF, 0x7, gain_tlv),
+			  MICFIL_OUTGAIN_CHX_SHIFT(0), 0x8, 0xF, gain_tlv),
 	SOC_SINGLE_SX_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL,
-			  MICFIL_OUTGAIN_CHX_SHIFT(1), 0xF, 0x7, gain_tlv),
+			  MICFIL_OUTGAIN_CHX_SHIFT(1), 0x8, 0xF, gain_tlv),
 	SOC_SINGLE_SX_TLV("CH2 Volume", REG_MICFIL_OUT_CTRL,
-			  MICFIL_OUTGAIN_CHX_SHIFT(2), 0xF, 0x7, gain_tlv),
+			  MICFIL_OUTGAIN_CHX_SHIFT(2), 0x8, 0xF, gain_tlv),
 	SOC_SINGLE_SX_TLV("CH3 Volume", REG_MICFIL_OUT_CTRL,
-			  MICFIL_OUTGAIN_CHX_SHIFT(3), 0xF, 0x7, gain_tlv),
+			  MICFIL_OUTGAIN_CHX_SHIFT(3), 0x8, 0xF, gain_tlv),
 	SOC_SINGLE_SX_TLV("CH4 Volume", REG_MICFIL_OUT_CTRL,
-			  MICFIL_OUTGAIN_CHX_SHIFT(4), 0xF, 0x7, gain_tlv),
+			  MICFIL_OUTGAIN_CHX_SHIFT(4), 0x8, 0xF, gain_tlv),
 	SOC_SINGLE_SX_TLV("CH5 Volume", REG_MICFIL_OUT_CTRL,
-			  MICFIL_OUTGAIN_CHX_SHIFT(5), 0xF, 0x7, gain_tlv),
+			  MICFIL_OUTGAIN_CHX_SHIFT(5), 0x8, 0xF, gain_tlv),
 	SOC_SINGLE_SX_TLV("CH6 Volume", REG_MICFIL_OUT_CTRL,
-			  MICFIL_OUTGAIN_CHX_SHIFT(6), 0xF, 0x7, gain_tlv),
+			  MICFIL_OUTGAIN_CHX_SHIFT(6), 0x8, 0xF, gain_tlv),
 	SOC_SINGLE_SX_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL,
-			  MICFIL_OUTGAIN_CHX_SHIFT(7), 0xF, 0x7, gain_tlv),
+			  MICFIL_OUTGAIN_CHX_SHIFT(7), 0x8, 0xF, gain_tlv),
 	SOC_ENUM_EXT("MICFIL Quality Select",
 		     fsl_micfil_quality_enum,
 		     micfil_quality_get, micfil_quality_set),

sound/soc/fsl/fsl_ssi.c

@@ -1189,14 +1189,14 @@ static struct snd_soc_dai_driver fsl_ssi_ac97_dai = {
 	.symmetric_channels = 1,
 	.probe = fsl_ssi_dai_probe,
 	.playback = {
-		.stream_name = "AC97 Playback",
+		.stream_name = "CPU AC97 Playback",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_48000,
 		.formats = SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S20,
 	},
 	.capture = {
-		.stream_name = "AC97 Capture",
+		.stream_name = "CPU AC97 Capture",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_48000,

sound/soc/intel/boards/Kconfig

@@ -554,10 +554,12 @@ config SND_SOC_INTEL_SOF_NAU8825_MACH
 	select SND_SOC_RT1015P
 	select SND_SOC_MAX98373_I2C
 	select SND_SOC_MAX98357A
+	select SND_SOC_NAU8315
 	select SND_SOC_DMIC
 	select SND_SOC_HDAC_HDMI
 	select SND_SOC_INTEL_HDA_DSP_COMMON
 	select SND_SOC_INTEL_SOF_MAXIM_COMMON
+	select SND_SOC_INTEL_SOF_REALTEK_COMMON
 	help
 	   This adds support for ASoC machine driver for SOF platforms
 	   with nau8825 codec.

sound/soc/intel/boards/sof_nau8825.c

@@ -48,6 +48,7 @@
 #define SOF_MAX98373_SPEAKER_AMP_PRESENT	BIT(15)
 #define SOF_MAX98360A_SPEAKER_AMP_PRESENT	BIT(16)
 #define SOF_RT1015P_SPEAKER_AMP_PRESENT	BIT(17)
+#define SOF_NAU8318_SPEAKER_AMP_PRESENT	BIT(18)
 
 static unsigned long sof_nau8825_quirk = SOF_NAU8825_SSP_CODEC(0);
 
@@ -338,6 +339,13 @@ static struct snd_soc_dai_link_component rt1019p_component[] = {
 	}
 };
 
+static struct snd_soc_dai_link_component nau8318_components[] = {
+	{
+		.name = "NVTN2012:00",
+		.dai_name = "nau8315-hifi",
+	}
+};
+
 static struct snd_soc_dai_link_component dummy_component[] = {
 	{
 		.name = "snd-soc-dummy",
@@ -486,6 +494,11 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
 			max_98360a_dai_link(&links[id]);
 		} else if (sof_nau8825_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) {
 			sof_rt1015p_dai_link(&links[id]);
+		} else if (sof_nau8825_quirk &
+				SOF_NAU8318_SPEAKER_AMP_PRESENT) {
+			links[id].codecs = nau8318_components;
+			links[id].num_codecs = ARRAY_SIZE(nau8318_components);
+			links[id].init = speaker_codec_init;
 		} else {
 			goto devm_err;
 		}
@@ -618,7 +631,7 @@ static const struct platform_device_id board_ids[] = {
 
 	},
 	{
-		.name = "adl_rt1019p_nau8825",
+		.name = "adl_rt1019p_8825",
 		.driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) |
 					SOF_SPEAKER_AMP_PRESENT |
 					SOF_RT1019P_SPEAKER_AMP_PRESENT |
@@ -626,7 +639,7 @@ static const struct platform_device_id board_ids[] = {
 					SOF_NAU8825_NUM_HDMIDEV(4)),
 	},
 	{
-		.name = "adl_max98373_nau8825",
+		.name = "adl_max98373_8825",
 		.driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) |
 					SOF_SPEAKER_AMP_PRESENT |
 					SOF_MAX98373_SPEAKER_AMP_PRESENT |
@@ -637,7 +650,7 @@ static const struct platform_device_id board_ids[] = {
 	},
 	{
 		/* The limitation of length of char array, shorten the name */
-		.name = "adl_mx98360a_nau8825",
+		.name = "adl_mx98360a_8825",
 		.driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) |
 					SOF_SPEAKER_AMP_PRESENT |
 					SOF_MAX98360A_SPEAKER_AMP_PRESENT |
@@ -648,7 +661,7 @@ static const struct platform_device_id board_ids[] = {
 
 	},
 	{
-		.name = "adl_rt1015p_nau8825",
+		.name = "adl_rt1015p_8825",
 		.driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) |
 					SOF_SPEAKER_AMP_PRESENT |
 					SOF_RT1015P_SPEAKER_AMP_PRESENT |
@@ -657,6 +670,16 @@ static const struct platform_device_id board_ids[] = {
 					SOF_BT_OFFLOAD_SSP(2) |
 					SOF_SSP_BT_OFFLOAD_PRESENT),
 	},
+	{
+		.name = "adl_nau8318_8825",
+		.driver_data = (kernel_ulong_t)(SOF_NAU8825_SSP_CODEC(0) |
+					SOF_SPEAKER_AMP_PRESENT |
+					SOF_NAU8318_SPEAKER_AMP_PRESENT |
+					SOF_NAU8825_SSP_AMP(1) |
+					SOF_NAU8825_NUM_HDMIDEV(4) |
+					SOF_BT_OFFLOAD_SSP(2) |
+					SOF_SSP_BT_OFFLOAD_PRESENT),
+	},
 	{ }
 };
 MODULE_DEVICE_TABLE(platform, board_ids);

sound/soc/intel/common/soc-acpi-intel-adl-match.c

@@ -450,6 +450,11 @@ static const struct snd_soc_acpi_codecs adl_lt6911_hdmi = {
 	.codecs = {"INTC10B0"}
 };
 
+static const struct snd_soc_acpi_codecs adl_nau8318_amp = {
+	.num_codecs = 1,
+	.codecs = {"NVTN2012"}
+};
+
 struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
 	{
 		.comp_ids = &adl_rt5682_rt5682s_hp,
@@ -474,21 +479,21 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
 	},
 	{
 		.id = "10508825",
-		.drv_name = "adl_rt1019p_nau8825",
+		.drv_name = "adl_rt1019p_8825",
 		.machine_quirk = snd_soc_acpi_codec_list,
 		.quirk_data = &adl_rt1019p_amp,
 		.sof_tplg_filename = "sof-adl-rt1019-nau8825.tplg",
 	},
 	{
 		.id = "10508825",
-		.drv_name = "adl_max98373_nau8825",
+		.drv_name = "adl_max98373_8825",
 		.machine_quirk = snd_soc_acpi_codec_list,
 		.quirk_data = &adl_max98373_amp,
 		.sof_tplg_filename = "sof-adl-max98373-nau8825.tplg",
 	},
 	{
 		.id = "10508825",
-		.drv_name = "adl_mx98360a_nau8825",
+		.drv_name = "adl_mx98360a_8825",
 		.machine_quirk = snd_soc_acpi_codec_list,
 		.quirk_data = &adl_max98360a_amp,
 		.sof_tplg_filename = "sof-adl-max98360a-nau8825.tplg",
@@ -502,11 +507,18 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
 	},
 	{
 		.id = "10508825",
-		.drv_name = "adl_rt1015p_nau8825",
+		.drv_name = "adl_rt1015p_8825",
 		.machine_quirk = snd_soc_acpi_codec_list,
 		.quirk_data = &adl_rt1015p_amp,
 		.sof_tplg_filename = "sof-adl-rt1015-nau8825.tplg",
 	},
+	{
+		.id = "10508825",
+		.drv_name = "adl_nau8318_8825",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &adl_nau8318_amp,
+		.sof_tplg_filename = "sof-adl-nau8318-nau8825.tplg",
+	},
 	{
 		.id = "10508825",
 		.drv_name = "sof_nau8825",

sound/soc/intel/common/soc-acpi-intel-rpl-match.c

@@ -203,6 +203,25 @@ static const struct snd_soc_acpi_link_adr rpl_sdw_rt711_link2_rt1316_link01_rt71
 	{}
 };
 
+static const struct snd_soc_acpi_link_adr rpl_sdw_rt711_link2_rt1316_link01[] = {
+	{
+		.mask = BIT(2),
+		.num_adr = ARRAY_SIZE(rt711_sdca_2_adr),
+		.adr_d = rt711_sdca_2_adr,
+	},
+	{
+		.mask = BIT(0),
+		.num_adr = ARRAY_SIZE(rt1316_0_group2_adr),
+		.adr_d = rt1316_0_group2_adr,
+	},
+	{
+		.mask = BIT(1),
+		.num_adr = ARRAY_SIZE(rt1316_1_group2_adr),
+		.adr_d = rt1316_1_group2_adr,
+	},
+	{}
+};
+
 static const struct snd_soc_acpi_link_adr rpl_sdw_rt711_link0_rt1318_link12_rt714_link3[] = {
 	{
 		.mask = BIT(0),
@@ -227,6 +246,25 @@ static const struct snd_soc_acpi_link_adr rpl_sdw_rt711_link0_rt1318_link12_rt71
 	{}
 };
 
+static const struct snd_soc_acpi_link_adr rpl_sdw_rt711_link0_rt1318_link12[] = {
+	{
+		.mask = BIT(0),
+		.num_adr = ARRAY_SIZE(rt711_sdca_0_adr),
+		.adr_d = rt711_sdca_0_adr,
+	},
+	{
+		.mask = BIT(1),
+		.num_adr = ARRAY_SIZE(rt1318_1_group1_adr),
+		.adr_d = rt1318_1_group1_adr,
+	},
+	{
+		.mask = BIT(2),
+		.num_adr = ARRAY_SIZE(rt1318_2_group1_adr),
+		.adr_d = rt1318_2_group1_adr,
+	},
+	{}
+};
+
 static const struct snd_soc_acpi_link_adr rpl_sdw_rt1316_link12_rt714_link0[] = {
 	{
 		.mask = BIT(1),
@@ -271,12 +309,24 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_sdw_machines[] = {
 		.drv_name = "sof_sdw",
 		.sof_tplg_filename = "sof-rpl-rt711-l0-rt1318-l12-rt714-l3.tplg",
 	},
+	{
+		.link_mask = 0x7, /* rt711 on link0 & two rt1318s on link1 and link2 */
+		.links = rpl_sdw_rt711_link0_rt1318_link12,
+		.drv_name = "sof_sdw",
+		.sof_tplg_filename = "sof-rpl-rt711-l0-rt1318-l12.tplg",
+	},
 	{
 		.link_mask = 0x7, /* rt714 on link0 & two rt1316s on link1 and link2 */
 		.links = rpl_sdw_rt1316_link12_rt714_link0,
 		.drv_name = "sof_sdw",
 		.sof_tplg_filename = "sof-rpl-rt1316-l12-rt714-l0.tplg",
 	},
+	{
+		.link_mask = 0x7, /* rt711 on link2 & two rt1316s on link0 and link1 */
+		.links = rpl_sdw_rt711_link2_rt1316_link01,
+		.drv_name = "sof_sdw",
+		.sof_tplg_filename = "sof-rpl-rt711-l2-rt1316-l01.tplg",
+	},
 	{
 		.link_mask = 0x1, /* link0 required */
 		.links = rpl_rvp,

sound/soc/mediatek/Kconfig

@@ -182,10 +182,12 @@ config SND_SOC_MT8186_MT6366_DA7219_MAX98357
 	  If unsure select "N".
 
 config SND_SOC_MT8186_MT6366_RT1019_RT5682S
-	tristate "ASoC Audio driver for MT8186 with RT1019 RT5682S codec"
+	tristate "ASoC Audio driver for MT8186 with RT1019 RT5682S MAX98357A/MAX98360 codec"
 	depends on I2C && GPIOLIB
 	depends on SND_SOC_MT8186 && MTK_PMIC_WRAP
+	select SND_SOC_MAX98357A
 	select SND_SOC_MT6358
+	select SND_SOC_MAX98357A
 	select SND_SOC_RT1015P
 	select SND_SOC_RT5682S
 	select SND_SOC_BT_SCO

sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c

@@ -1083,6 +1083,21 @@ static struct snd_soc_card mt8186_mt6366_rt1019_rt5682s_soc_card = {
 	.num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf),
 };
 
+static struct snd_soc_card mt8186_mt6366_rt5682s_max98360_soc_card = {
+	.name = "mt8186_rt5682s_max98360",
+	.owner = THIS_MODULE,
+	.dai_link = mt8186_mt6366_rt1019_rt5682s_dai_links,
+	.num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links),
+	.controls = mt8186_mt6366_rt1019_rt5682s_controls,
+	.num_controls = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_controls),
+	.dapm_widgets = mt8186_mt6366_rt1019_rt5682s_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_widgets),
+	.dapm_routes = mt8186_mt6366_rt1019_rt5682s_routes,
+	.num_dapm_routes = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_routes),
+	.codec_conf = mt8186_mt6366_rt1019_rt5682s_codec_conf,
+	.num_configs = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_codec_conf),
+};
+
 static int mt8186_mt6366_rt1019_rt5682s_dev_probe(struct platform_device *pdev)
 {
 	struct snd_soc_card *card;
@@ -1232,9 +1247,14 @@ static int mt8186_mt6366_rt1019_rt5682s_dev_probe(struct platform_device *pdev)
 
 #if IS_ENABLED(CONFIG_OF)
 static const struct of_device_id mt8186_mt6366_rt1019_rt5682s_dt_match[] = {
-	{	.compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound",
+	{
+		.compatible = "mediatek,mt8186-mt6366-rt1019-rt5682s-sound",
 		.data = &mt8186_mt6366_rt1019_rt5682s_soc_card,
 	},
+	{
+		.compatible = "mediatek,mt8186-mt6366-rt5682s-max98360-sound",
+		.data = &mt8186_mt6366_rt5682s_max98360_soc_card,
+	},
 	{}
 };
 MODULE_DEVICE_TABLE(of, mt8186_mt6366_rt1019_rt5682s_dt_match);

sound/soc/qcom/Kconfig

@@ -2,7 +2,6 @@
 menuconfig SND_SOC_QCOM
 	tristate "ASoC support for QCOM platforms"
 	depends on ARCH_QCOM || COMPILE_TEST
-	imply SND_SOC_QCOM_COMMON
 	help
 	  Say Y or M if you want to add support to use audio devices
 	  in Qualcomm Technologies SOC-based platforms.
@@ -60,14 +59,16 @@ config SND_SOC_STORM
 config SND_SOC_APQ8016_SBC
 	tristate "SoC Audio support for APQ8016 SBC platforms"
 	select SND_SOC_LPASS_APQ8016
-	depends on SND_SOC_QCOM_COMMON
+	select SND_SOC_QCOM_COMMON
 	help
 	  Support for Qualcomm Technologies LPASS audio block in
 	  APQ8016 SOC-based systems.
 	  Say Y if you want to use audio devices on MI2S.
 
 config SND_SOC_QCOM_COMMON
-	depends on SOUNDWIRE
+	tristate
+
+config SND_SOC_QCOM_SDW
 	tristate
 
 config SND_SOC_QDSP6_COMMON
@@ -144,7 +145,7 @@ config SND_SOC_MSM8996
 	depends on QCOM_APR
 	depends on COMMON_CLK
 	select SND_SOC_QDSP6
-	depends on SND_SOC_QCOM_COMMON
+	select SND_SOC_QCOM_COMMON
 	help
 	  Support for Qualcomm Technologies LPASS audio block in
 	  APQ8096 SoC-based systems.
@@ -155,7 +156,7 @@ config SND_SOC_SDM845
 	depends on QCOM_APR && I2C && SOUNDWIRE
 	depends on COMMON_CLK
 	select SND_SOC_QDSP6
-	depends on SND_SOC_QCOM_COMMON
+	select SND_SOC_QCOM_COMMON
 	select SND_SOC_RT5663
 	select SND_SOC_MAX98927
 	imply SND_SOC_CROS_EC_CODEC
@@ -169,7 +170,8 @@ config SND_SOC_SM8250
 	depends on QCOM_APR && SOUNDWIRE
 	depends on COMMON_CLK
 	select SND_SOC_QDSP6
-	depends on SND_SOC_QCOM_COMMON
+	select SND_SOC_QCOM_COMMON
+	select SND_SOC_QCOM_SDW
 	help
 	  To add support for audio on Qualcomm Technologies Inc.
 	  SM8250 SoC-based systems.
@@ -180,7 +182,8 @@ config SND_SOC_SC8280XP
 	depends on QCOM_APR && SOUNDWIRE
 	depends on COMMON_CLK
 	select SND_SOC_QDSP6
-	depends on SND_SOC_QCOM_COMMON
+	select SND_SOC_QCOM_COMMON
+	select SND_SOC_QCOM_SDW
 	help
 	  To add support for audio on Qualcomm Technologies Inc.
 	  SC8280XP SoC-based systems.
@@ -190,7 +193,7 @@ config SND_SOC_SC7180
 	tristate "SoC Machine driver for SC7180 boards"
 	depends on I2C && GPIOLIB
 	depends on SOUNDWIRE || SOUNDWIRE=n
-	depends on SND_SOC_QCOM_COMMON
+	select SND_SOC_QCOM_COMMON
 	select SND_SOC_LPASS_SC7180
 	select SND_SOC_MAX98357A
 	select SND_SOC_RT5682_I2C
@@ -204,7 +207,7 @@ config SND_SOC_SC7180
 config SND_SOC_SC7280
 	tristate "SoC Machine driver for SC7280 boards"
 	depends on I2C && SOUNDWIRE
-	depends on SND_SOC_QCOM_COMMON
+	select SND_SOC_QCOM_COMMON
 	select SND_SOC_LPASS_SC7280
 	select SND_SOC_MAX98357A
 	select SND_SOC_WCD938X_SDW

sound/soc/qcom/Makefile

@@ -28,6 +28,7 @@ snd-soc-sdm845-objs := sdm845.o
 snd-soc-sm8250-objs := sm8250.o
 snd-soc-sc8280xp-objs := sc8280xp.o
 snd-soc-qcom-common-objs := common.o
+snd-soc-qcom-sdw-objs := sdw.o
 
 obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o
 obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o
@@ -38,6 +39,7 @@ obj-$(CONFIG_SND_SOC_SC8280XP) += snd-soc-sc8280xp.o
 obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o
 obj-$(CONFIG_SND_SOC_SM8250) += snd-soc-sm8250.o
 obj-$(CONFIG_SND_SOC_QCOM_COMMON) += snd-soc-qcom-common.o
+obj-$(CONFIG_SND_SOC_QCOM_SDW) += snd-soc-qcom-sdw.o
 
 #DSP lib
 obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/

sound/soc/qcom/common.c

@@ -180,120 +180,6 @@ int qcom_snd_parse_of(struct snd_soc_card *card)
 }
 EXPORT_SYMBOL_GPL(qcom_snd_parse_of);
 
-int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream,
-			 struct sdw_stream_runtime *sruntime,
-			 bool *stream_prepared)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
-	int ret;
-
-	if (!sruntime)
-		return 0;
-
-	switch (cpu_dai->id) {
-	case WSA_CODEC_DMA_RX_0:
-	case WSA_CODEC_DMA_RX_1:
-	case RX_CODEC_DMA_RX_0:
-	case RX_CODEC_DMA_RX_1:
-	case TX_CODEC_DMA_TX_0:
-	case TX_CODEC_DMA_TX_1:
-	case TX_CODEC_DMA_TX_2:
-	case TX_CODEC_DMA_TX_3:
-		break;
-	default:
-		return 0;
-	}
-
-	if (*stream_prepared) {
-		sdw_disable_stream(sruntime);
-		sdw_deprepare_stream(sruntime);
-		*stream_prepared = false;
-	}
-
-	ret = sdw_prepare_stream(sruntime);
-	if (ret)
-		return ret;
-
-	/**
-	 * NOTE: there is a strict hw requirement about the ordering of port
-	 * enables and actual WSA881x PA enable. PA enable should only happen
-	 * after soundwire ports are enabled if not DC on the line is
-	 * accumulated resulting in Click/Pop Noise
-	 * PA enable/mute are handled as part of codec DAPM and digital mute.
-	 */
-
-	ret = sdw_enable_stream(sruntime);
-	if (ret) {
-		sdw_deprepare_stream(sruntime);
-		return ret;
-	}
-	*stream_prepared  = true;
-
-	return ret;
-}
-EXPORT_SYMBOL_GPL(qcom_snd_sdw_prepare);
-
-int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream,
-			   struct snd_pcm_hw_params *params,
-			   struct sdw_stream_runtime **psruntime)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai;
-	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
-	struct sdw_stream_runtime *sruntime;
-	int i;
-
-	switch (cpu_dai->id) {
-	case WSA_CODEC_DMA_RX_0:
-	case RX_CODEC_DMA_RX_0:
-	case RX_CODEC_DMA_RX_1:
-	case TX_CODEC_DMA_TX_0:
-	case TX_CODEC_DMA_TX_1:
-	case TX_CODEC_DMA_TX_2:
-	case TX_CODEC_DMA_TX_3:
-		for_each_rtd_codec_dais(rtd, i, codec_dai) {
-			sruntime = snd_soc_dai_get_stream(codec_dai, substream->stream);
-			if (sruntime != ERR_PTR(-ENOTSUPP))
-				*psruntime = sruntime;
-		}
-		break;
-	}
-
-	return 0;
-
-}
-EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_params);
-
-int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream,
-			 struct sdw_stream_runtime *sruntime, bool *stream_prepared)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
-
-	switch (cpu_dai->id) {
-	case WSA_CODEC_DMA_RX_0:
-	case WSA_CODEC_DMA_RX_1:
-	case RX_CODEC_DMA_RX_0:
-	case RX_CODEC_DMA_RX_1:
-	case TX_CODEC_DMA_TX_0:
-	case TX_CODEC_DMA_TX_1:
-	case TX_CODEC_DMA_TX_2:
-	case TX_CODEC_DMA_TX_3:
-		if (sruntime && *stream_prepared) {
-			sdw_disable_stream(sruntime);
-			sdw_deprepare_stream(sruntime);
-			*stream_prepared = false;
-		}
-		break;
-	default:
-		break;
-	}
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_free);
-
 int qcom_snd_wcd_jack_setup(struct snd_soc_pcm_runtime *rtd,
 			    struct snd_soc_jack *jack, bool *jack_setup)
 {

sound/soc/qcom/common.h

@@ -5,19 +5,9 @@
 #define __QCOM_SND_COMMON_H__
 
 #include <sound/soc.h>
-#include <linux/soundwire/sdw.h>
 
 int qcom_snd_parse_of(struct snd_soc_card *card);
 int qcom_snd_wcd_jack_setup(struct snd_soc_pcm_runtime *rtd,
 			    struct snd_soc_jack *jack, bool *jack_setup);
 
-int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream,
-			 struct sdw_stream_runtime *runtime,
-			 bool *stream_prepared);
-int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream,
-			   struct snd_pcm_hw_params *params,
-			   struct sdw_stream_runtime **psruntime);
-int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream,
-			 struct sdw_stream_runtime *sruntime,
-			 bool *stream_prepared);
 #endif

sound/soc/qcom/lpass-cpu.c

@@ -1037,10 +1037,11 @@ static void of_lpass_cpu_parse_dai_data(struct device *dev,
 					struct lpass_data *data)
 {
 	struct device_node *node;
-	int ret, id;
+	int ret, i, id;
 
 	/* Allow all channels by default for backwards compatibility */
-	for (id = 0; id < data->variant->num_dai; id++) {
+	for (i = 0; i < data->variant->num_dai; i++) {
+		id = data->variant->dai_driver[i].id;
 		data->mi2s_playback_sd_mode[id] = LPAIF_I2SCTL_MODE_8CH;
 		data->mi2s_capture_sd_mode[id] = LPAIF_I2SCTL_MODE_8CH;
 	}

sound/soc/qcom/sc8280xp.c

@@ -12,6 +12,7 @@
 #include <linux/input-event-codes.h>
 #include "qdsp6/q6afe.h"
 #include "common.h"
+#include "sdw.h"
 
 #define DRIVER_NAME		"sc8280xp"
 

sound/soc/qcom/sdw.c

+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2018, Linaro Limited.
+// Copyright (c) 2018, The Linux Foundation. All rights reserved.
+
+#include <linux/module.h>
+#include <sound/soc.h>
+#include "qdsp6/q6afe.h"
+#include "sdw.h"
+
+int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream,
+			 struct sdw_stream_runtime *sruntime,
+			 bool *stream_prepared)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	int ret;
+
+	if (!sruntime)
+		return 0;
+
+	switch (cpu_dai->id) {
+	case WSA_CODEC_DMA_RX_0:
+	case WSA_CODEC_DMA_RX_1:
+	case RX_CODEC_DMA_RX_0:
+	case RX_CODEC_DMA_RX_1:
+	case TX_CODEC_DMA_TX_0:
+	case TX_CODEC_DMA_TX_1:
+	case TX_CODEC_DMA_TX_2:
+	case TX_CODEC_DMA_TX_3:
+		break;
+	default:
+		return 0;
+	}
+
+	if (*stream_prepared) {
+		sdw_disable_stream(sruntime);
+		sdw_deprepare_stream(sruntime);
+		*stream_prepared = false;
+	}
+
+	ret = sdw_prepare_stream(sruntime);
+	if (ret)
+		return ret;
+
+	/**
+	 * NOTE: there is a strict hw requirement about the ordering of port
+	 * enables and actual WSA881x PA enable. PA enable should only happen
+	 * after soundwire ports are enabled if not DC on the line is
+	 * accumulated resulting in Click/Pop Noise
+	 * PA enable/mute are handled as part of codec DAPM and digital mute.
+	 */
+
+	ret = sdw_enable_stream(sruntime);
+	if (ret) {
+		sdw_deprepare_stream(sruntime);
+		return ret;
+	}
+	*stream_prepared  = true;
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(qcom_snd_sdw_prepare);
+
+int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream,
+			   struct snd_pcm_hw_params *params,
+			   struct sdw_stream_runtime **psruntime)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct sdw_stream_runtime *sruntime;
+	int i;
+
+	switch (cpu_dai->id) {
+	case WSA_CODEC_DMA_RX_0:
+	case RX_CODEC_DMA_RX_0:
+	case RX_CODEC_DMA_RX_1:
+	case TX_CODEC_DMA_TX_0:
+	case TX_CODEC_DMA_TX_1:
+	case TX_CODEC_DMA_TX_2:
+	case TX_CODEC_DMA_TX_3:
+		for_each_rtd_codec_dais(rtd, i, codec_dai) {
+			sruntime = snd_soc_dai_get_stream(codec_dai, substream->stream);
+			if (sruntime != ERR_PTR(-ENOTSUPP))
+				*psruntime = sruntime;
+		}
+		break;
+	}
+
+	return 0;
+
+}
+EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_params);
+
+int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream,
+			 struct sdw_stream_runtime *sruntime, bool *stream_prepared)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+
+	switch (cpu_dai->id) {
+	case WSA_CODEC_DMA_RX_0:
+	case WSA_CODEC_DMA_RX_1:
+	case RX_CODEC_DMA_RX_0:
+	case RX_CODEC_DMA_RX_1:
+	case TX_CODEC_DMA_TX_0:
+	case TX_CODEC_DMA_TX_1:
+	case TX_CODEC_DMA_TX_2:
+	case TX_CODEC_DMA_TX_3:
+		if (sruntime && *stream_prepared) {
+			sdw_disable_stream(sruntime);
+			sdw_deprepare_stream(sruntime);
+			*stream_prepared = false;
+		}
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_free);
+MODULE_LICENSE("GPL v2");

sound/soc/qcom/sdw.h

+/* SPDX-License-Identifier: GPL-2.0 */
+// Copyright (c) 2018, The Linux Foundation. All rights reserved.
+
+#ifndef __QCOM_SND_SDW_H__
+#define __QCOM_SND_SDW_H__
+
+#include <linux/soundwire/sdw.h>
+
+int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream,
+			 struct sdw_stream_runtime *runtime,
+			 bool *stream_prepared);
+int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream,
+			   struct snd_pcm_hw_params *params,
+			   struct sdw_stream_runtime **psruntime);
+int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream,
+			 struct sdw_stream_runtime *sruntime,
+			 bool *stream_prepared);
+#endif

sound/soc/qcom/sm8250.c

@@ -12,6 +12,7 @@
 #include <linux/input-event-codes.h>
 #include "qdsp6/q6afe.h"
 #include "common.h"
+#include "sdw.h"
 
 #define DRIVER_NAME		"sm8250"
 #define MI2S_BCLK_RATE		1536000

sound/soc/sof/debug.c

@@ -353,7 +353,9 @@ int snd_sof_dbg_init(struct snd_sof_dev *sdev)
 			return err;
 	}
 
-	return 0;
+	return snd_sof_debugfs_buf_item(sdev, &sdev->fw_state,
+					sizeof(sdev->fw_state),
+					"fw_state", 0444);
 }
 EXPORT_SYMBOL_GPL(snd_sof_dbg_init);
 

sound/soc/sof/pm.c

@@ -182,7 +182,7 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
 	const struct sof_ipc_pm_ops *pm_ops = sdev->ipc->ops->pm;
 	const struct sof_ipc_tplg_ops *tplg_ops = sdev->ipc->ops->tplg;
 	pm_message_t pm_state;
-	u32 target_state = 0;
+	u32 target_state = snd_sof_dsp_power_target(sdev);
 	int ret;
 
 	/* do nothing if dsp suspend callback is not set */
@@ -192,6 +192,9 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
 	if (runtime_suspend && !sof_ops(sdev)->runtime_suspend)
 		return 0;
 
+	if (tplg_ops && tplg_ops->tear_down_all_pipelines)
+		tplg_ops->tear_down_all_pipelines(sdev, false);
+
 	if (sdev->fw_state != SOF_FW_BOOT_COMPLETE)
 		goto suspend;
 
@@ -206,7 +209,6 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
 		}
 	}
 
-	target_state = snd_sof_dsp_power_target(sdev);
 	pm_state.event = target_state;
 
 	/* Skip to platform-specific suspend if DSP is entering D0 */
@@ -217,9 +219,6 @@ static int sof_suspend(struct device *dev, bool runtime_suspend)
 		goto suspend;
 	}
 
-	if (tplg_ops->tear_down_all_pipelines)
-		tplg_ops->tear_down_all_pipelines(sdev, false);
-
 	/* suspend DMA trace */
 	sof_fw_trace_suspend(sdev, pm_state);
 

sound/usb/implicit.c

@@ -471,7 +471,7 @@ snd_usb_find_implicit_fb_sync_format(struct snd_usb_audio *chip,
 	subs = find_matching_substream(chip, stream, target->sync_ep,
 				       target->fmt_type);
 	if (!subs)
-		return sync_fmt;
+		goto end;
 
 	high_score = 0;
 	list_for_each_entry(fp, &subs->fmt_list, list) {
@@ -485,6 +485,7 @@ snd_usb_find_implicit_fb_sync_format(struct snd_usb_audio *chip,
 		}
 	}
 
+ end:
 	if (fixed_rate)
 		*fixed_rate = snd_usb_pcm_has_fixed_rate(subs);
 	return sync_fmt;

sound/usb/pcm.c

@@ -160,9 +160,12 @@ find_substream_format(struct snd_usb_substream *subs,
 bool snd_usb_pcm_has_fixed_rate(struct snd_usb_substream *subs)
 {
 	const struct audioformat *fp;
-	struct snd_usb_audio *chip = subs->stream->chip;
+	struct snd_usb_audio *chip;
 	int rate = -1;
 
+	if (!subs)
+		return false;
+	chip = subs->stream->chip;
 	if (!(chip->quirk_flags & QUIRK_FLAG_FIXED_RATE))
 		return false;
 	list_for_each_entry(fp, &subs->fmt_list, list) {
@@ -525,6 +528,8 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
 		if (snd_usb_endpoint_compatible(chip, subs->data_endpoint,
 						fmt, hw_params))
 			goto unlock;
+		if (stop_endpoints(subs, false))
+			sync_pending_stops(subs);
 		close_endpoints(chip, subs);
 	}
 
@@ -787,11 +792,27 @@ static int apply_hw_params_minmax(struct snd_interval *it, unsigned int rmin,
 	return changed;
 }
 
+/* get the specified endpoint object that is being used by other streams
+ * (i.e. the parameter is locked)
+ */
+static const struct snd_usb_endpoint *
+get_endpoint_in_use(struct snd_usb_audio *chip, int endpoint,
+		    const struct snd_usb_endpoint *ref_ep)
+{
+	const struct snd_usb_endpoint *ep;
+
+	ep = snd_usb_get_endpoint(chip, endpoint);
+	if (ep && ep->cur_audiofmt && (ep != ref_ep || ep->opened > 1))
+		return ep;
+	return NULL;
+}
+
 static int hw_rule_rate(struct snd_pcm_hw_params *params,
 			struct snd_pcm_hw_rule *rule)
 {
 	struct snd_usb_substream *subs = rule->private;
 	struct snd_usb_audio *chip = subs->stream->chip;
+	const struct snd_usb_endpoint *ep;
 	const struct audioformat *fp;
 	struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
 	unsigned int rmin, rmax, r;
@@ -803,6 +824,29 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params,
 	list_for_each_entry(fp, &subs->fmt_list, list) {
 		if (!hw_check_valid_format(subs, params, fp))
 			continue;
+
+		ep = get_endpoint_in_use(chip, fp->endpoint,
+					 subs->data_endpoint);
+		if (ep) {
+			hwc_debug("rate limit %d for ep#%x\n",
+				  ep->cur_rate, fp->endpoint);
+			rmin = min(rmin, ep->cur_rate);
+			rmax = max(rmax, ep->cur_rate);
+			continue;
+		}
+
+		if (fp->implicit_fb) {
+			ep = get_endpoint_in_use(chip, fp->sync_ep,
+						 subs->sync_endpoint);
+			if (ep) {
+				hwc_debug("rate limit %d for sync_ep#%x\n",
+					  ep->cur_rate, fp->sync_ep);
+				rmin = min(rmin, ep->cur_rate);
+				rmax = max(rmax, ep->cur_rate);
+				continue;
+			}
+		}
+
 		r = snd_usb_endpoint_get_clock_rate(chip, fp->clock);
 		if (r > 0) {
 			if (!snd_interval_test(it, r))
@@ -872,6 +916,8 @@ static int hw_rule_format(struct snd_pcm_hw_params *params,
 			  struct snd_pcm_hw_rule *rule)
 {
 	struct snd_usb_substream *subs = rule->private;
+	struct snd_usb_audio *chip = subs->stream->chip;
+	const struct snd_usb_endpoint *ep;
 	const struct audioformat *fp;
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 	u64 fbits;
@@ -881,6 +927,27 @@ static int hw_rule_format(struct snd_pcm_hw_params *params,
 	list_for_each_entry(fp, &subs->fmt_list, list) {
 		if (!hw_check_valid_format(subs, params, fp))
 			continue;
+
+		ep = get_endpoint_in_use(chip, fp->endpoint,
+					 subs->data_endpoint);
+		if (ep) {
+			hwc_debug("format limit %d for ep#%x\n",
+				  ep->cur_format, fp->endpoint);
+			fbits |= pcm_format_to_bits(ep->cur_format);
+			continue;
+		}
+
+		if (fp->implicit_fb) {
+			ep = get_endpoint_in_use(chip, fp->sync_ep,
+						 subs->sync_endpoint);
+			if (ep) {
+				hwc_debug("format limit %d for sync_ep#%x\n",
+					  ep->cur_format, fp->sync_ep);
+				fbits |= pcm_format_to_bits(ep->cur_format);
+				continue;
+			}
+		}
+
 		fbits |= fp->formats;
 	}
 	return apply_hw_params_format_bits(fmt, fbits);
@@ -913,98 +980,95 @@ static int hw_rule_period_time(struct snd_pcm_hw_params *params,
 	return apply_hw_params_minmax(it, pmin, UINT_MAX);
 }
 
-/* get the EP or the sync EP for implicit fb when it's already set up */
-static const struct snd_usb_endpoint *
-get_sync_ep_from_substream(struct snd_usb_substream *subs)
-{
-	struct snd_usb_audio *chip = subs->stream->chip;
-	const struct audioformat *fp;
-	const struct snd_usb_endpoint *ep;
-
-	list_for_each_entry(fp, &subs->fmt_list, list) {
-		ep = snd_usb_get_endpoint(chip, fp->endpoint);
-		if (ep && ep->cur_audiofmt) {
-			/* if EP is already opened solely for this substream,
-			 * we still allow us to change the parameter; otherwise
-			 * this substream has to follow the existing parameter
-			 */
-			if (ep->cur_audiofmt != subs->cur_audiofmt || ep->opened > 1)
-				return ep;
-		}
-		if (!fp->implicit_fb)
-			continue;
-		/* for the implicit fb, check the sync ep as well */
-		ep = snd_usb_get_endpoint(chip, fp->sync_ep);
-		if (ep && ep->cur_audiofmt)
-			return ep;
-	}
-	return NULL;
-}
-
 /* additional hw constraints for implicit feedback mode */
-static int hw_rule_format_implicit_fb(struct snd_pcm_hw_params *params,
-				      struct snd_pcm_hw_rule *rule)
-{
-	struct snd_usb_substream *subs = rule->private;
-	const struct snd_usb_endpoint *ep;
-	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
-
-	ep = get_sync_ep_from_substream(subs);
-	if (!ep)
-		return 0;
-
-	hwc_debug("applying %s\n", __func__);
-	return apply_hw_params_format_bits(fmt, pcm_format_to_bits(ep->cur_format));
-}
-
-static int hw_rule_rate_implicit_fb(struct snd_pcm_hw_params *params,
-				    struct snd_pcm_hw_rule *rule)
-{
-	struct snd_usb_substream *subs = rule->private;
-	const struct snd_usb_endpoint *ep;
-	struct snd_interval *it;
-
-	ep = get_sync_ep_from_substream(subs);
-	if (!ep)
-		return 0;
-
-	hwc_debug("applying %s\n", __func__);
-	it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
-	return apply_hw_params_minmax(it, ep->cur_rate, ep->cur_rate);
-}
-
 static int hw_rule_period_size_implicit_fb(struct snd_pcm_hw_params *params,
 					   struct snd_pcm_hw_rule *rule)
 {
 	struct snd_usb_substream *subs = rule->private;
+	struct snd_usb_audio *chip = subs->stream->chip;
+	const struct audioformat *fp;
 	const struct snd_usb_endpoint *ep;
 	struct snd_interval *it;
+	unsigned int rmin, rmax;
 
-	ep = get_sync_ep_from_substream(subs);
-	if (!ep)
-		return 0;
-
-	hwc_debug("applying %s\n", __func__);
 	it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
-	return apply_hw_params_minmax(it, ep->cur_period_frames,
-				      ep->cur_period_frames);
+	hwc_debug("hw_rule_period_size: (%u,%u)\n", it->min, it->max);
+	rmin = UINT_MAX;
+	rmax = 0;
+	list_for_each_entry(fp, &subs->fmt_list, list) {
+		if (!hw_check_valid_format(subs, params, fp))
+			continue;
+		ep = get_endpoint_in_use(chip, fp->endpoint,
+					 subs->data_endpoint);
+		if (ep) {
+			hwc_debug("period size limit %d for ep#%x\n",
+				  ep->cur_period_frames, fp->endpoint);
+			rmin = min(rmin, ep->cur_period_frames);
+			rmax = max(rmax, ep->cur_period_frames);
+			continue;
+		}
+
+		if (fp->implicit_fb) {
+			ep = get_endpoint_in_use(chip, fp->sync_ep,
+						 subs->sync_endpoint);
+			if (ep) {
+				hwc_debug("period size limit %d for sync_ep#%x\n",
+					  ep->cur_period_frames, fp->sync_ep);
+				rmin = min(rmin, ep->cur_period_frames);
+				rmax = max(rmax, ep->cur_period_frames);
+				continue;
+			}
+		}
+	}
+
+	if (!rmax)
+		return 0; /* no limit by implicit fb */
+	return apply_hw_params_minmax(it, rmin, rmax);
 }
 
 static int hw_rule_periods_implicit_fb(struct snd_pcm_hw_params *params,
 				       struct snd_pcm_hw_rule *rule)
 {
 	struct snd_usb_substream *subs = rule->private;
+	struct snd_usb_audio *chip = subs->stream->chip;
+	const struct audioformat *fp;
 	const struct snd_usb_endpoint *ep;
 	struct snd_interval *it;
+	unsigned int rmin, rmax;
 
-	ep = get_sync_ep_from_substream(subs);
-	if (!ep)
-		return 0;
-
-	hwc_debug("applying %s\n", __func__);
 	it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIODS);
-	return apply_hw_params_minmax(it, ep->cur_buffer_periods,
-				      ep->cur_buffer_periods);
+	hwc_debug("hw_rule_periods: (%u,%u)\n", it->min, it->max);
+	rmin = UINT_MAX;
+	rmax = 0;
+	list_for_each_entry(fp, &subs->fmt_list, list) {
+		if (!hw_check_valid_format(subs, params, fp))
+			continue;
+		ep = get_endpoint_in_use(chip, fp->endpoint,
+					 subs->data_endpoint);
+		if (ep) {
+			hwc_debug("periods limit %d for ep#%x\n",
+				  ep->cur_buffer_periods, fp->endpoint);
+			rmin = min(rmin, ep->cur_buffer_periods);
+			rmax = max(rmax, ep->cur_buffer_periods);
+			continue;
+		}
+
+		if (fp->implicit_fb) {
+			ep = get_endpoint_in_use(chip, fp->sync_ep,
+						 subs->sync_endpoint);
+			if (ep) {
+				hwc_debug("periods limit %d for sync_ep#%x\n",
+					  ep->cur_buffer_periods, fp->sync_ep);
+				rmin = min(rmin, ep->cur_buffer_periods);
+				rmax = max(rmax, ep->cur_buffer_periods);
+				continue;
+			}
+		}
+	}
+
+	if (!rmax)
+		return 0; /* no limit by implicit fb */
+	return apply_hw_params_minmax(it, rmin, rmax);
 }
 
 /*
@@ -1113,16 +1177,6 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre
 		return err;
 
 	/* additional hw constraints for implicit fb */
-	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
-				  hw_rule_format_implicit_fb, subs,
-				  SNDRV_PCM_HW_PARAM_FORMAT, -1);
-	if (err < 0)
-		return err;
-	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
-				  hw_rule_rate_implicit_fb, subs,
-				  SNDRV_PCM_HW_PARAM_RATE, -1);
-	if (err < 0)
-		return err;
 	err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
 				  hw_rule_period_size_implicit_fb, subs,
 				  SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1);

sound/usb/stream.c

@@ -1222,6 +1222,12 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip,
 			if (err < 0)
 				return err;
 		}
+
+		/* try to set the interface... */
+		usb_set_interface(chip->dev, iface_no, 0);
+		snd_usb_init_pitch(chip, fp);
+		snd_usb_init_sample_rate(chip, fp, fp->rate_max);
+		usb_set_interface(chip->dev, iface_no, altno);
 	}
 	return 0;
 }