Merge tag 'sound-5.11-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Here are some piled fixes, hopefully the last big one for 5.11.

  All changes are device-specific small fixes, and majority of commits
  are for ASoC while USB-audio got a bit large changes for addressing
  the regression for devices with quirks"

* tag 'sound-5.11-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (31 commits)
  ALSA: hda/hdmi - enable runtime pm for CI AMD display audio
  ALSA: firewire-tascam: Fix integer overflow in midi_port_work()
  ALSA: fireface: Fix integer overflow in transmit_midi_msg()
  ALSA: hda/tegra: fix tegra-hda on tegra30 soc
  clk: tegra30: Add hda clock default rates to clock driver
  ALSA: doc: Fix reference to mixart.rst
  ALSA: usb-audio: Fix implicit feedback sync setup for Pioneer devices
  ALSA: usb-audio: Annotate the endpoint index in audioformat
  ALSA: usb-audio: Avoid unnecessary interface re-setup
  ALSA: usb-audio: Choose audioformat of a counter-part substream
  ALSA: usb-audio: Fix the missing endpoints creations for quirks
  ALSA: hda/realtek: fix right sounds and mute/micmute LEDs for HP machines
  ASoC: AMD Renoir - add DMI entry for Lenovo ThinkPad X395
  ASoC: amd: Replacing MSI with Legacy IRQ model
  ASoC: AMD Renoir - add DMI entry for Lenovo ThinkPad E14 Gen 2
  ASoC: meson: axg-tdm-interface: fix loopback
  ASoC: meson: axg-tdmin: fix axg skew offset
  ASoC: max98373: don't access volatile registers in bias level off
  ASoC: rt711: mutex between calibration and power state changes
  ASoC: Intel: haswell: Add missing pm_ops
  ...

Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml

 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+# Copyright (C) 2020 Texas Instruments Incorporated
+# Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
 %YAML 1.2
 ---
 $id: http://devicetree.org/schemas/sound/ti,j721e-cpb-audio.yaml#
@@ -7,7 +9,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml#
 title: Texas Instruments J721e Common Processor Board Audio Support
 
 maintainers:
-  - Peter Ujfalusi <peter.ujfalusi@ti.com>
+  - Peter Ujfalusi <peter.ujfalusi@gmail.com>
 
 description: |
   The audio support on the board is using pcm3168a codec connected to McASP10

Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml

 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+# Copyright (C) 2020 Texas Instruments Incorporated
+# Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
 %YAML 1.2
 ---
 $id: http://devicetree.org/schemas/sound/ti,j721e-cpb-ivi-audio.yaml#
@@ -7,7 +9,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml#
 title: Texas Instruments J721e Common Processor Board Audio Support
 
 maintainers:
-  - Peter Ujfalusi <peter.ujfalusi@ti.com>
+  - Peter Ujfalusi <peter.ujfalusi@gmail.com>
 
 description: |
   The Infotainment board plugs into the Common Processor Board, the support of the

Documentation/sound/alsa-configuration.rst

@@ -1501,7 +1501,7 @@ Module for Digigram miXart8 sound cards.
 
 This module supports multiple cards.
 Note: One miXart8 board will be represented as 4 alsa cards.
-See MIXART.txt for details.
+See Documentation/sound/cards/mixart.rst for details.
 
 When the driver is compiled as a module and the hotplug firmware
 is supported, the firmware data is loaded via hotplug automatically.

MAINTAINERS

@@ -12848,7 +12848,7 @@ F:	include/misc/ocxl*
 F:	include/uapi/misc/ocxl.h
 
 OMAP AUDIO SUPPORT
-M:	Peter Ujfalusi <peter.ujfalusi@ti.com>
+M:	Peter Ujfalusi <peter.ujfalusi@gmail.com>
 M:	Jarkko Nikula <jarkko.nikula@bitmer.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 L:	linux-omap@vger.kernel.org
@@ -17541,7 +17541,7 @@ F:	arch/xtensa/
 F:	drivers/irqchip/irq-xtensa-*
 
 TEXAS INSTRUMENTS ASoC DRIVERS
-M:	Peter Ujfalusi <peter.ujfalusi@ti.com>
+M:	Peter Ujfalusi <peter.ujfalusi@gmail.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Maintained
 F:	sound/soc/ti/
@@ -17851,7 +17851,7 @@ F:	Documentation/devicetree/bindings/net/nfc/trf7970a.txt
 F:	drivers/nfc/trf7970a.c
 
 TI TWL4030 SERIES SOC CODEC DRIVER
-M:	Peter Ujfalusi <peter.ujfalusi@ti.com>
+M:	Peter Ujfalusi <peter.ujfalusi@gmail.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Maintained
 F:	sound/soc/codecs/twl4030*

drivers/clk/tegra/clk-tegra30.c

@@ -1256,6 +1256,8 @@ static struct tegra_clk_init_table init_table[] __initdata = {
 	{ TEGRA30_CLK_I2S3_SYNC, TEGRA30_CLK_CLK_MAX, 24000000, 0 },
 	{ TEGRA30_CLK_I2S4_SYNC, TEGRA30_CLK_CLK_MAX, 24000000, 0 },
 	{ TEGRA30_CLK_VIMCLK_SYNC, TEGRA30_CLK_CLK_MAX, 24000000, 0 },
+	{ TEGRA30_CLK_HDA, TEGRA30_CLK_PLL_P, 102000000, 0 },
+	{ TEGRA30_CLK_HDA2CODEC_2X, TEGRA30_CLK_PLL_P, 48000000, 0 },
 	/* must be the last entry */
 	{ TEGRA30_CLK_CLK_MAX, TEGRA30_CLK_CLK_MAX, 0, 0 },
 };

sound/firewire/fireface/ff-transaction.c

@@ -88,7 +88,7 @@ static void transmit_midi_msg(struct snd_ff *ff, unsigned int port)
 
 	/* Set interval to next transaction. */
 	ff->next_ktime[port] = ktime_add_ns(ktime_get(),
-				ff->rx_bytes[port] * 8 * NSEC_PER_SEC / 31250);
+			ff->rx_bytes[port] * 8 * (NSEC_PER_SEC / 31250));
 
 	if (quad_count == 1)
 		tcode = TCODE_WRITE_QUADLET_REQUEST;

sound/firewire/tascam/tascam-transaction.c

@@ -209,7 +209,7 @@ static void midi_port_work(struct work_struct *work)
 
 	/* Set interval to next transaction. */
 	port->next_ktime = ktime_add_ns(ktime_get(),
-				port->consume_bytes * 8 * NSEC_PER_SEC / 31250);
+			port->consume_bytes * 8 * (NSEC_PER_SEC / 31250));
 
 	/* Start this transaction. */
 	port->idling = false;

sound/pci/hda/hda_intel.c

@@ -2598,7 +2598,8 @@ static const struct pci_device_id azx_ids[] = {
 	  .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB },
 	/* ATI HDMI */
 	{ PCI_DEVICE(0x1002, 0x0002),
-	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
+	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS |
+	  AZX_DCAPS_PM_RUNTIME },
 	{ PCI_DEVICE(0x1002, 0x1308),
 	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
 	{ PCI_DEVICE(0x1002, 0x157a),
@@ -2660,9 +2661,11 @@ static const struct pci_device_id azx_ids[] = {
 	{ PCI_DEVICE(0x1002, 0xaab0),
 	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
 	{ PCI_DEVICE(0x1002, 0xaac0),
-	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
+	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS |
+	  AZX_DCAPS_PM_RUNTIME },
 	{ PCI_DEVICE(0x1002, 0xaac8),
-	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS },
+	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS |
+	  AZX_DCAPS_PM_RUNTIME },
 	{ PCI_DEVICE(0x1002, 0xaad8),
 	  .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS |
 	  AZX_DCAPS_PM_RUNTIME },

sound/pci/hda/hda_tegra.c

@@ -388,7 +388,7 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev)
 	 * in powers of 2, next available ratio is 16 which can be
 	 * used as a limiting factor here.
 	 */
-	if (of_device_is_compatible(np, "nvidia,tegra194-hda"))
+	if (of_device_is_compatible(np, "nvidia,tegra30-hda"))
 		chip->bus.core.sdo_limit = 16;
 
 	/* codec detection */

sound/pci/hda/patch_realtek.c

@@ -7970,6 +7970,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED),
+	SND_PCI_QUIRK(0x103c, 0x8780, "HP ZBook Fury 17 G7 Mobile Workstation",
+		      ALC285_FIXUP_HP_GPIO_AMP_INIT),
+	SND_PCI_QUIRK(0x103c, 0x8783, "HP ZBook Fury 15 G7 Mobile Workstation",
+		      ALC285_FIXUP_HP_GPIO_AMP_INIT),
 	SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED),

sound/soc/amd/raven/pci-acp3x.c

@@ -140,21 +140,14 @@ static int snd_acp3x_probe(struct pci_dev *pci,
 		goto release_regions;
 	}
 
-	/* check for msi interrupt support */
-	ret = pci_enable_msi(pci);
-	if (ret)
-		/* msi is not enabled */
-		irqflags = IRQF_SHARED;
-	else
-		/* msi is enabled */
-		irqflags = 0;
+	irqflags = IRQF_SHARED;
 
 	addr = pci_resource_start(pci, 0);
 	adata->acp3x_base = devm_ioremap(&pci->dev, addr,
 					pci_resource_len(pci, 0));
 	if (!adata->acp3x_base) {
 		ret = -ENOMEM;
-		goto disable_msi;
+		goto release_regions;
 	}
 	pci_set_master(pci);
 	pci_set_drvdata(pci, adata);
@@ -162,7 +155,7 @@ static int snd_acp3x_probe(struct pci_dev *pci,
 	adata->pme_en = rv_readl(adata->acp3x_base + mmACP_PME_EN);
 	ret = acp3x_init(adata);
 	if (ret)
-		goto disable_msi;
+		goto release_regions;
 
 	val = rv_readl(adata->acp3x_base + mmACP_I2S_PIN_CONFIG);
 	switch (val) {
@@ -251,8 +244,6 @@ static int snd_acp3x_probe(struct pci_dev *pci,
 de_init:
 	if (acp3x_deinit(adata->acp3x_base))
 		dev_err(&pci->dev, "ACP de-init failed\n");
-disable_msi:
-	pci_disable_msi(pci);
 release_regions:
 	pci_release_regions(pci);
 disable_pci:
@@ -311,7 +302,6 @@ static void snd_acp3x_remove(struct pci_dev *pci)
 		dev_err(&pci->dev, "ACP de-init failed\n");
 	pm_runtime_forbid(&pci->dev);
 	pm_runtime_get_noresume(&pci->dev);
-	pci_disable_msi(pci);
 	pci_release_regions(pci);
 	pci_disable_device(pci);
 }

sound/soc/amd/renoir/rn-pci-acp3x.c

@@ -171,6 +171,20 @@ static const struct dmi_system_id rn_acp_quirk_table[] = {
 			DMI_EXACT_MATCH(DMI_BOARD_NAME, "LNVNB161216"),
 		}
 	},
+	{
+		/* Lenovo ThinkPad E14 Gen 2 */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_EXACT_MATCH(DMI_BOARD_NAME, "20T6CTO1WW"),
+		}
+	},
+	{
+		/* Lenovo ThinkPad X395 */
+		.matches = {
+			DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"),
+		}
+	},
 	{}
 };
 

sound/soc/atmel/Kconfig

@@ -143,7 +143,7 @@ config SND_MCHP_SOC_SPDIFTX
 	  - sama7g5
 
 	  This S/PDIF TX driver is compliant with IEC-60958 standard and
-	  includes programable User Data and Channel Status fields.
+	  includes programmable User Data and Channel Status fields.
 
 config SND_MCHP_SOC_SPDIFRX
 	tristate "Microchip ASoC driver for boards using S/PDIF RX"
@@ -157,5 +157,5 @@ config SND_MCHP_SOC_SPDIFRX
 	  - sama7g5
 
 	  This S/PDIF RX driver is compliant with IEC-60958 standard and
-	  includes programable User Data and Channel Status fields.
+	  includes programmable User Data and Channel Status fields.
 endif

sound/soc/codecs/Kconfig

@@ -457,7 +457,7 @@ config SND_SOC_ADAU7118_HW
 	help
 	  Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM
 	  Converter. In this mode, the device works in standalone mode which
-	  means that there is no bus to comunicate with it. Stereo mode is not
+	  means that there is no bus to communicate with it. Stereo mode is not
 	  supported in this mode.
 
 	  To compile this driver as a module, choose M here: the module

sound/soc/codecs/max98373-i2c.c

@@ -19,6 +19,12 @@
 #include <sound/tlv.h>
 #include "max98373.h"
 
+static const u32 max98373_i2c_cache_reg[] = {
+	MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK,
+	MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK,
+	MAX98373_R20B6_BDE_CUR_STATE_READBACK,
+};
+
 static struct reg_default max98373_reg[] = {
 	{MAX98373_R2000_SW_RESET, 0x00},
 	{MAX98373_R2001_INT_RAW1, 0x00},
@@ -472,6 +478,11 @@ static struct snd_soc_dai_driver max98373_dai[] = {
 static int max98373_suspend(struct device *dev)
 {
 	struct max98373_priv *max98373 = dev_get_drvdata(dev);
+	int i;
+
+	/* cache feedback register values before suspend */
+	for (i = 0; i < max98373->cache_num; i++)
+		regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val);
 
 	regcache_cache_only(max98373->regmap, true);
 	regcache_mark_dirty(max98373->regmap);
@@ -509,6 +520,7 @@ static int max98373_i2c_probe(struct i2c_client *i2c,
 {
 	int ret = 0;
 	int reg = 0;
+	int i;
 	struct max98373_priv *max98373 = NULL;
 
 	max98373 = devm_kzalloc(&i2c->dev, sizeof(*max98373), GFP_KERNEL);
@@ -534,6 +546,14 @@ static int max98373_i2c_probe(struct i2c_client *i2c,
 		return ret;
 	}
 
+	max98373->cache_num = ARRAY_SIZE(max98373_i2c_cache_reg);
+	max98373->cache = devm_kcalloc(&i2c->dev, max98373->cache_num,
+				       sizeof(*max98373->cache),
+				       GFP_KERNEL);
+
+	for (i = 0; i < max98373->cache_num; i++)
+		max98373->cache[i].reg = max98373_i2c_cache_reg[i];
+
 	/* voltage/current slot & gpio configuration */
 	max98373_slot_config(&i2c->dev, max98373);
 

sound/soc/codecs/max98373-sdw.c

@@ -23,6 +23,12 @@ struct sdw_stream_data {
 	struct sdw_stream_runtime *sdw_stream;
 };
 
+static const u32 max98373_sdw_cache_reg[] = {
+	MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK,
+	MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK,
+	MAX98373_R20B6_BDE_CUR_STATE_READBACK,
+};
+
 static struct reg_default max98373_reg[] = {
 	{MAX98373_R0040_SCP_INIT_STAT_1, 0x00},
 	{MAX98373_R0041_SCP_INIT_MASK_1, 0x00},
@@ -245,6 +251,11 @@ static const struct regmap_config max98373_sdw_regmap = {
 static __maybe_unused int max98373_suspend(struct device *dev)
 {
 	struct max98373_priv *max98373 = dev_get_drvdata(dev);
+	int i;
+
+	/* cache feedback register values before suspend */
+	for (i = 0; i < max98373->cache_num; i++)
+		regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val);
 
 	regcache_cache_only(max98373->regmap, true);
 
@@ -757,6 +768,7 @@ static int max98373_init(struct sdw_slave *slave, struct regmap *regmap)
 {
 	struct max98373_priv *max98373;
 	int ret;
+	int i;
 	struct device *dev = &slave->dev;
 
 	/*  Allocate and assign private driver data structure  */
@@ -768,6 +780,14 @@ static int max98373_init(struct sdw_slave *slave, struct regmap *regmap)
 	max98373->regmap = regmap;
 	max98373->slave = slave;
 
+	max98373->cache_num = ARRAY_SIZE(max98373_sdw_cache_reg);
+	max98373->cache = devm_kcalloc(dev, max98373->cache_num,
+				       sizeof(*max98373->cache),
+				       GFP_KERNEL);
+
+	for (i = 0; i < max98373->cache_num; i++)
+		max98373->cache[i].reg = max98373_sdw_cache_reg[i];
+
 	/* Read voltage and slot configuration */
 	max98373_slot_config(dev, max98373);
 

sound/soc/codecs/max98373.c

@@ -168,6 +168,31 @@ static SOC_ENUM_SINGLE_DECL(max98373_adc_samplerate_enum,
 			    MAX98373_R2051_MEAS_ADC_SAMPLING_RATE, 0,
 			    max98373_ADC_samplerate_text);
 
+static int max98373_feedback_get(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component);
+	int i;
+
+	if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) {
+		/*
+		 * Register values will be cached before suspend. The cached value
+		 * will be a valid value and userspace will happy with that.
+		 */
+		for (i = 0; i < max98373->cache_num; i++) {
+			if (mc->reg == max98373->cache[i].reg) {
+				ucontrol->value.integer.value[0] = max98373->cache[i].val;
+				return 0;
+			}
+		}
+	}
+
+	return snd_soc_put_volsw(kcontrol, ucontrol);
+}
+
 static const struct snd_kcontrol_new max98373_snd_controls[] = {
 SOC_SINGLE("Digital Vol Sel Switch", MAX98373_R203F_AMP_DSP_CFG,
 	MAX98373_AMP_VOL_SEL_SHIFT, 1, 0),
@@ -209,8 +234,10 @@ SOC_SINGLE("ADC PVDD FLT Switch", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG,
 	MAX98373_FLT_EN_SHIFT, 1, 0),
 SOC_SINGLE("ADC TEMP FLT Switch", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG,
 	MAX98373_FLT_EN_SHIFT, 1, 0),
-SOC_SINGLE("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0),
-SOC_SINGLE("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0),
+SOC_SINGLE_EXT("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0,
+	max98373_feedback_get, NULL),
+SOC_SINGLE_EXT("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0,
+	max98373_feedback_get, NULL),
 SOC_SINGLE("ADC PVDD FLT Coeff", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG,
 	0, 0x3, 0),
 SOC_SINGLE("ADC TEMP FLT Coeff", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG,
@@ -226,7 +253,8 @@ SOC_SINGLE("BDE LVL1 Thresh", MAX98373_R2097_BDE_L1_THRESH, 0, 0xFF, 0),
 SOC_SINGLE("BDE LVL2 Thresh", MAX98373_R2098_BDE_L2_THRESH, 0, 0xFF, 0),
 SOC_SINGLE("BDE LVL3 Thresh", MAX98373_R2099_BDE_L3_THRESH, 0, 0xFF, 0),
 SOC_SINGLE("BDE LVL4 Thresh", MAX98373_R209A_BDE_L4_THRESH, 0, 0xFF, 0),
-SOC_SINGLE("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0),
+SOC_SINGLE_EXT("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0,
+	max98373_feedback_get, NULL),
 SOC_SINGLE("BDE Clip Mode Switch", MAX98373_R2092_BDE_CLIPPER_MODE, 0, 1, 0),
 SOC_SINGLE("BDE Thresh Hysteresis", MAX98373_R209B_BDE_THRESH_HYST, 0, 0xFF, 0),
 SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0),

sound/soc/codecs/max98373.h

@@ -203,6 +203,11 @@
 /* MAX98373_R2000_SW_RESET */
 #define MAX98373_SOFT_RESET (0x1 << 0)
 
+struct max98373_cache {
+	u32 reg;
+	u32 val;
+};
+
 struct max98373_priv {
 	struct regmap *regmap;
 	int reset_gpio;
@@ -212,6 +217,9 @@ struct max98373_priv {
 	bool interleave_mode;
 	unsigned int ch_size;
 	bool tdm_mode;
+	/* cache for reading a valid fake feedback value */
+	struct max98373_cache *cache;
+	int cache_num;
 	/* variables to support soundwire */
 	struct sdw_slave *slave;
 	bool hw_init;

sound/soc/codecs/rt711.c

@@ -462,6 +462,8 @@ static int rt711_set_amp_gain_put(struct snd_kcontrol *kcontrol,
 	unsigned int read_ll, read_rl;
 	int i;
 
+	mutex_lock(&rt711->calibrate_mutex);
+
 	/* Can't use update bit function, so read the original value first */
 	addr_h = mc->reg;
 	addr_l = mc->rreg;
@@ -547,6 +549,8 @@ static int rt711_set_amp_gain_put(struct snd_kcontrol *kcontrol,
 	if (dapm->bias_level <= SND_SOC_BIAS_STANDBY)
 		regmap_write(rt711->regmap,
 				RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D3);
+
+	mutex_unlock(&rt711->calibrate_mutex);
 	return 0;
 }
 
@@ -859,9 +863,11 @@ static int rt711_set_bias_level(struct snd_soc_component *component,
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
+		mutex_lock(&rt711->calibrate_mutex);
 		regmap_write(rt711->regmap,
 			RT711_SET_AUDIO_POWER_STATE,
 			AC_PWRST_D3);
+		mutex_unlock(&rt711->calibrate_mutex);
 		break;
 
 	default:

sound/soc/fsl/imx-hdmi.c

@@ -164,6 +164,7 @@ static int imx_hdmi_probe(struct platform_device *pdev)
 
 	if ((hdmi_out && hdmi_in) || (!hdmi_out && !hdmi_in)) {
 		dev_err(&pdev->dev, "Invalid HDMI DAI link\n");
+		ret = -EINVAL;
 		goto fail;
 	}
 

sound/soc/intel/boards/haswell.c

@@ -189,6 +189,7 @@ static struct platform_driver haswell_audio = {
 	.probe = haswell_audio_probe,
 	.driver = {
 		.name = "haswell-audio",
+		.pm = &snd_soc_pm_ops,
 	},
 };
 

sound/soc/intel/skylake/cnl-sst.c

@@ -224,6 +224,7 @@ static int cnl_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id)
 				"dsp boot timeout, status=%#x error=%#x\n",
 				sst_dsp_shim_read(ctx, CNL_ADSP_FW_STATUS),
 				sst_dsp_shim_read(ctx, CNL_ADSP_ERROR_CODE));
+			ret = -ETIMEDOUT;
 			goto err;
 		}
 	} else {

sound/soc/meson/axg-tdm-interface.c

@@ -467,8 +467,20 @@ static int axg_tdm_iface_set_bias_level(struct snd_soc_component *component,
 	return ret;
 }
 
+static const struct snd_soc_dapm_widget axg_tdm_iface_dapm_widgets[] = {
+	SND_SOC_DAPM_SIGGEN("Playback Signal"),
+};
+
+static const struct snd_soc_dapm_route axg_tdm_iface_dapm_routes[] = {
+	{ "Loopback", NULL, "Playback Signal" },
+};
+
 static const struct snd_soc_component_driver axg_tdm_iface_component_drv = {
-	.set_bias_level	= axg_tdm_iface_set_bias_level,
+	.dapm_widgets		= axg_tdm_iface_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(axg_tdm_iface_dapm_widgets),
+	.dapm_routes		= axg_tdm_iface_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(axg_tdm_iface_dapm_routes),
+	.set_bias_level		= axg_tdm_iface_set_bias_level,
 };
 
 static const struct of_device_id axg_tdm_iface_of_match[] = {

sound/soc/meson/axg-tdmin.c

@@ -224,15 +224,6 @@ static const struct axg_tdm_formatter_ops axg_tdmin_ops = {
 };
 
 static const struct axg_tdm_formatter_driver axg_tdmin_drv = {
-	.component_drv	= &axg_tdmin_component_drv,
-	.regmap_cfg	= &axg_tdmin_regmap_cfg,
-	.ops		= &axg_tdmin_ops,
-	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.skew_offset	= 2,
-	},
-};
-
-static const struct axg_tdm_formatter_driver g12a_tdmin_drv = {
 	.component_drv	= &axg_tdmin_component_drv,
 	.regmap_cfg	= &axg_tdmin_regmap_cfg,
 	.ops		= &axg_tdmin_ops,
@@ -247,10 +238,10 @@ static const struct of_device_id axg_tdmin_of_match[] = {
 		.data = &axg_tdmin_drv,
 	}, {
 		.compatible = "amlogic,g12a-tdmin",
-		.data = &g12a_tdmin_drv,
+		.data = &axg_tdmin_drv,
 	}, {
 		.compatible = "amlogic,sm1-tdmin",
-		.data = &g12a_tdmin_drv,
+		.data = &axg_tdmin_drv,
 	}, {}
 };
 MODULE_DEVICE_TABLE(of, axg_tdmin_of_match);

sound/soc/qcom/lpass-cpu.c

@@ -270,18 +270,6 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
 	struct lpaif_i2sctl *i2sctl = drvdata->i2sctl;
 	unsigned int id = dai->driver->id;
 	int ret = -EINVAL;
-	unsigned int val = 0;
-
-	ret = regmap_read(drvdata->lpaif_map,
-				LPAIF_I2SCTL_REG(drvdata->variant, dai->driver->id), &val);
-	if (ret) {
-		dev_err(dai->dev, "error reading from i2sctl reg: %d\n", ret);
-		return ret;
-	}
-	if (val == LPAIF_I2SCTL_RESET_STATE) {
-		dev_err(dai->dev, "error in i2sctl register state\n");
-		return -ENOTRECOVERABLE;
-	}
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -454,20 +442,16 @@ static bool lpass_cpu_regmap_volatile(struct device *dev, unsigned int reg)
 	struct lpass_variant *v = drvdata->variant;
 	int i;
 
-	for (i = 0; i < v->i2s_ports; ++i)
-		if (reg == LPAIF_I2SCTL_REG(v, i))
-			return true;
 	for (i = 0; i < v->irq_ports; ++i)
 		if (reg == LPAIF_IRQSTAT_REG(v, i))
 			return true;
 
 	for (i = 0; i < v->rdma_channels; ++i)
-		if (reg == LPAIF_RDMACURR_REG(v, i) || reg == LPAIF_RDMACTL_REG(v, i))
+		if (reg == LPAIF_RDMACURR_REG(v, i))
 			return true;
 
 	for (i = 0; i < v->wrdma_channels; ++i)
-		if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start) ||
-			reg == LPAIF_WRDMACTL_REG(v, i + v->wrdma_channel_start))
+		if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start))
 			return true;
 
 	return false;

sound/soc/qcom/lpass-platform.c

@@ -452,7 +452,6 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component,
 	unsigned int reg_irqclr = 0, val_irqclr = 0;
 	unsigned int  reg_irqen = 0, val_irqen = 0, val_mask = 0;
 	unsigned int dai_id = cpu_dai->driver->id;
-	unsigned int dma_ctrl_reg = 0;
 
 	ch = pcm_data->dma_ch;
 	if (dir ==  SNDRV_PCM_STREAM_PLAYBACK) {
@@ -469,17 +468,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component,
 		id = pcm_data->dma_ch - v->wrdma_channel_start;
 		map = drvdata->lpaif_map;
 	}
-	ret = regmap_read(map, LPAIF_DMACTL_REG(v, ch, dir, dai_id), &dma_ctrl_reg);
-	if (ret) {
-		dev_err(soc_runtime->dev, "error reading from rdmactl reg: %d\n", ret);
-		return ret;
-	}
 
-	if (dma_ctrl_reg == LPAIF_DMACTL_RESET_STATE ||
-		dma_ctrl_reg == LPAIF_DMACTL_RESET_STATE + 1) {
-		dev_err(soc_runtime->dev, "error in rdmactl register state\n");
-		return -ENOTRECOVERABLE;
-	}
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
@@ -500,7 +489,6 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component,
 					"error writing to rdmactl reg: %d\n", ret);
 				return ret;
 			}
-			map = drvdata->hdmiif_map;
 			reg_irqclr = LPASS_HDMITX_APP_IRQCLEAR_REG(v);
 			val_irqclr = (LPAIF_IRQ_ALL(ch) |
 					LPAIF_IRQ_HDMI_REQ_ON_PRELOAD(ch) |
@@ -519,7 +507,6 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component,
 			break;
 		case MI2S_PRIMARY:
 		case MI2S_SECONDARY:
-			map = drvdata->lpaif_map;
 			reg_irqclr = LPAIF_IRQCLEAR_REG(v, LPAIF_IRQ_PORT_HOST);
 			val_irqclr = LPAIF_IRQ_ALL(ch);
 
@@ -563,7 +550,6 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component,
 					"error writing to rdmactl reg: %d\n", ret);
 				return ret;
 			}
-			map = drvdata->hdmiif_map;
 			reg_irqen = LPASS_HDMITX_APP_IRQEN_REG(v);
 			val_mask = (LPAIF_IRQ_ALL(ch) |
 					LPAIF_IRQ_HDMI_REQ_ON_PRELOAD(ch) |
@@ -573,7 +559,6 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component,
 			break;
 		case MI2S_PRIMARY:
 		case MI2S_SECONDARY:
-			map = drvdata->lpaif_map;
 			reg_irqen = LPAIF_IRQEN_REG(v, LPAIF_IRQ_PORT_HOST);
 			val_mask = LPAIF_IRQ_ALL(ch);
 			val_irqen = 0;
@@ -838,6 +823,39 @@ static void lpass_platform_pcm_free(struct snd_soc_component *component,
 	}
 }
 
+static int lpass_platform_pcmops_suspend(struct snd_soc_component *component)
+{
+	struct lpass_data *drvdata = snd_soc_component_get_drvdata(component);
+	struct regmap *map;
+	unsigned int dai_id = component->id;
+
+	if (dai_id == LPASS_DP_RX)
+		map = drvdata->hdmiif_map;
+	else
+		map = drvdata->lpaif_map;
+
+	regcache_cache_only(map, true);
+	regcache_mark_dirty(map);
+
+	return 0;
+}
+
+static int lpass_platform_pcmops_resume(struct snd_soc_component *component)
+{
+	struct lpass_data *drvdata = snd_soc_component_get_drvdata(component);
+	struct regmap *map;
+	unsigned int dai_id = component->id;
+
+	if (dai_id == LPASS_DP_RX)
+		map = drvdata->hdmiif_map;
+	else
+		map = drvdata->lpaif_map;
+
+	regcache_cache_only(map, false);
+	return regcache_sync(map);
+}
+
+
 static const struct snd_soc_component_driver lpass_component_driver = {
 	.name		= DRV_NAME,
 	.open		= lpass_platform_pcmops_open,
@@ -850,6 +868,8 @@ static const struct snd_soc_component_driver lpass_component_driver = {
 	.mmap		= lpass_platform_pcmops_mmap,
 	.pcm_construct	= lpass_platform_pcm_new,
 	.pcm_destruct	= lpass_platform_pcm_free,
+	.suspend		= lpass_platform_pcmops_suspend,
+	.resume			= lpass_platform_pcmops_resume,
 
 };
 

sound/soc/sh/rcar/adg.c

@@ -366,25 +366,27 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable)
 	struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
 	struct device *dev = rsnd_priv_to_dev(priv);
 	struct clk *clk;
-	int i, ret;
+	int i;
 
 	for_each_rsnd_clk(clk, adg, i) {
-		ret = 0;
 		if (enable) {
-			ret = clk_prepare_enable(clk);
+			int ret = clk_prepare_enable(clk);
 
 			/*
 			 * We shouldn't use clk_get_rate() under
 			 * atomic context. Let's keep it when
 			 * rsnd_adg_clk_enable() was called
 			 */
-			adg->clk_rate[i] = clk_get_rate(adg->clk[i]);
+			adg->clk_rate[i] = 0;
+			if (ret < 0)
+				dev_warn(dev, "can't use clk %d\n", i);
+			else
+				adg->clk_rate[i] = clk_get_rate(clk);
 		} else {
-			clk_disable_unprepare(clk);
+			if (adg->clk_rate[i])
+				clk_disable_unprepare(clk);
+			adg->clk_rate[i] = 0;
 		}
-
-		if (ret < 0)
-			dev_warn(dev, "can't use clk %d\n", i);
 	}
 }
 

sound/soc/soc-dapm.c

@@ -2486,6 +2486,7 @@ void snd_soc_dapm_free_widget(struct snd_soc_dapm_widget *w)
 	enum snd_soc_dapm_direction dir;
 
 	list_del(&w->list);
+	list_del(&w->dirty);
 	/*
 	 * remove source and sink paths associated to this widget.
 	 * While removing the path, remove reference to it from both

sound/soc/sof/Kconfig

@@ -122,7 +122,7 @@ config SND_SOC_SOF_DEBUG_XRUN_STOP
 	bool "SOF stop on XRUN"
 	help
 	  This option forces PCMs to stop on any XRUN event. This is useful to
-	  preserve any trace data ond pipeline status prior to the XRUN.
+	  preserve any trace data and pipeline status prior to the XRUN.
 	  Say Y if you are debugging SOF FW pipeline XRUNs.
 	  If unsure select "N".
 

sound/usb/card.c

@@ -450,10 +450,8 @@ lookup_device_name(u32 id)
 static void snd_usb_audio_free(struct snd_card *card)
 {
 	struct snd_usb_audio *chip = card->private_data;
-	struct snd_usb_endpoint *ep, *n;
 
-	list_for_each_entry_safe(ep, n, &chip->ep_list, list)
-		snd_usb_endpoint_free(ep);
+	snd_usb_endpoint_free_all(chip);
 
 	mutex_destroy(&chip->mutex);
 	if (!atomic_read(&chip->shutdown))
@@ -611,6 +609,7 @@ static int snd_usb_audio_create(struct usb_interface *intf,
 	chip->usb_id = usb_id;
 	INIT_LIST_HEAD(&chip->pcm_list);
 	INIT_LIST_HEAD(&chip->ep_list);
+	INIT_LIST_HEAD(&chip->iface_ref_list);
 	INIT_LIST_HEAD(&chip->midi_list);
 	INIT_LIST_HEAD(&chip->mixer_list);
 

sound/usb/card.h

@@ -18,6 +18,7 @@ struct audioformat {
 	unsigned int frame_size;	/* samples per frame for non-audio */
 	unsigned char iface;		/* interface number */
 	unsigned char altsetting;	/* corresponding alternate setting */
+	unsigned char ep_idx;		/* endpoint array index */
 	unsigned char altset_idx;	/* array index of altenate setting */
 	unsigned char attributes;	/* corresponding attributes of cs endpoint */
 	unsigned char endpoint;		/* endpoint */
@@ -42,6 +43,7 @@ struct audioformat {
 };
 
 struct snd_usb_substream;
+struct snd_usb_iface_ref;
 struct snd_usb_endpoint;
 struct snd_usb_power_domain;
 
@@ -58,6 +60,7 @@ struct snd_urb_ctx {
 
 struct snd_usb_endpoint {
 	struct snd_usb_audio *chip;
+	struct snd_usb_iface_ref *iface_ref;
 
 	int opened;		/* open refcount; protect with chip->mutex */
 	atomic_t running;	/* running status */

sound/usb/endpoint.c

@@ -24,6 +24,14 @@
 #define EP_FLAG_RUNNING		1
 #define EP_FLAG_STOPPING	2
 
+/* interface refcounting */
+struct snd_usb_iface_ref {
+	unsigned char iface;
+	bool need_setup;
+	int opened;
+	struct list_head list;
+};
+
 /*
  * snd_usb_endpoint is a model that abstracts everything related to an
  * USB endpoint and its streaming.
@@ -488,6 +496,28 @@ static void snd_complete_urb(struct urb *urb)
 	clear_bit(ctx->index, &ep->active_mask);
 }
 
+/*
+ * Find or create a refcount object for the given interface
+ *
+ * The objects are released altogether in snd_usb_endpoint_free_all()
+ */
+static struct snd_usb_iface_ref *
+iface_ref_find(struct snd_usb_audio *chip, int iface)
+{
+	struct snd_usb_iface_ref *ip;
+
+	list_for_each_entry(ip, &chip->iface_ref_list, list)
+		if (ip->iface == iface)
+			return ip;
+
+	ip = kzalloc(sizeof(*ip), GFP_KERNEL);
+	if (!ip)
+		return NULL;
+	ip->iface = iface;
+	list_add_tail(&ip->list, &chip->iface_ref_list);
+	return ip;
+}
+
 /*
  * Get the existing endpoint object corresponding EP
  * Returns NULL if not present.
@@ -520,8 +550,8 @@ snd_usb_get_endpoint(struct snd_usb_audio *chip, int ep_num)
  *
  * Returns zero on success or a negative error code.
  *
- * New endpoints will be added to chip->ep_list and must be freed by
- * calling snd_usb_endpoint_free().
+ * New endpoints will be added to chip->ep_list and freed by
+ * calling snd_usb_endpoint_free_all().
  *
  * For SND_USB_ENDPOINT_TYPE_SYNC, the caller needs to guarantee that
  * bNumEndpoints > 1 beforehand.
@@ -653,11 +683,17 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip,
 		} else {
 			ep->iface = fp->iface;
 			ep->altsetting = fp->altsetting;
-			ep->ep_idx = 0;
+			ep->ep_idx = fp->ep_idx;
 		}
 		usb_audio_dbg(chip, "Open EP 0x%x, iface=%d:%d, idx=%d\n",
 			      ep_num, ep->iface, ep->altsetting, ep->ep_idx);
 
+		ep->iface_ref = iface_ref_find(chip, ep->iface);
+		if (!ep->iface_ref) {
+			ep = NULL;
+			goto unlock;
+		}
+
 		ep->cur_audiofmt = fp;
 		ep->cur_channels = fp->channels;
 		ep->cur_rate = params_rate(params);
@@ -681,6 +717,11 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip,
 			      ep->implicit_fb_sync);
 
 	} else {
+		if (WARN_ON(!ep->iface_ref)) {
+			ep = NULL;
+			goto unlock;
+		}
+
 		if (!endpoint_compatible(ep, fp, params)) {
 			usb_audio_err(chip, "Incompatible EP setup for 0x%x\n",
 				      ep_num);
@@ -692,6 +733,9 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip,
 			      ep_num, ep->opened);
 	}
 
+	if (!ep->iface_ref->opened++)
+		ep->iface_ref->need_setup = true;
+
 	ep->opened++;
 
  unlock:
@@ -760,12 +804,16 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip,
 	mutex_lock(&chip->mutex);
 	usb_audio_dbg(chip, "Closing EP 0x%x (count %d)\n",
 		      ep->ep_num, ep->opened);
-	if (!--ep->opened) {
+
+	if (!--ep->iface_ref->opened)
 		endpoint_set_interface(chip, ep, false);
+
+	if (!--ep->opened) {
 		ep->iface = 0;
 		ep->altsetting = 0;
 		ep->cur_audiofmt = NULL;
 		ep->cur_rate = 0;
+		ep->iface_ref = NULL;
 		usb_audio_dbg(chip, "EP 0x%x closed\n", ep->ep_num);
 	}
 	mutex_unlock(&chip->mutex);
@@ -775,6 +823,8 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip,
 void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep)
 {
 	ep->need_setup = true;
+	if (ep->iface_ref)
+		ep->iface_ref->need_setup = true;
 }
 
 /*
@@ -1195,11 +1245,13 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip,
 	int err = 0;
 
 	mutex_lock(&chip->mutex);
+	if (WARN_ON(!ep->iface_ref))
+		goto unlock;
 	if (!ep->need_setup)
 		goto unlock;
 
-	/* No need to (re-)configure the sync EP belonging to the same altset */
-	if (ep->ep_idx) {
+	/* If the interface has been already set up, just set EP parameters */
+	if (!ep->iface_ref->need_setup) {
 		err = snd_usb_endpoint_set_params(chip, ep);
 		if (err < 0)
 			goto unlock;
@@ -1242,6 +1294,8 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip,
 			goto unlock;
 	}
 
+	ep->iface_ref->need_setup = false;
+
  done:
 	ep->need_setup = false;
 	err = 1;
@@ -1387,15 +1441,21 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep)
 }
 
 /**
- * snd_usb_endpoint_free: Free the resources of an snd_usb_endpoint
+ * snd_usb_endpoint_free_all: Free the resources of an snd_usb_endpoint
+ * @card: The chip
  *
- * @ep: the endpoint to free
- *
- * This free all resources of the given ep.
+ * This free all endpoints and those resources
  */
-void snd_usb_endpoint_free(struct snd_usb_endpoint *ep)
+void snd_usb_endpoint_free_all(struct snd_usb_audio *chip)
 {
-	kfree(ep);
+	struct snd_usb_endpoint *ep, *en;
+	struct snd_usb_iface_ref *ip, *in;
+
+	list_for_each_entry_safe(ep, en, &chip->ep_list, list)
+		kfree(ep);
+
+	list_for_each_entry_safe(ip, in, &chip->iface_ref_list, list)
+		kfree(ip);
 }
 
 /*

sound/usb/endpoint.h

@@ -42,7 +42,7 @@ void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep);
 void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep);
 int  snd_usb_endpoint_activate(struct snd_usb_endpoint *ep);
 void snd_usb_endpoint_release(struct snd_usb_endpoint *ep);
-void snd_usb_endpoint_free(struct snd_usb_endpoint *ep);
+void snd_usb_endpoint_free_all(struct snd_usb_audio *chip);
 
 int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep);
 int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep,

sound/usb/implicit.c

@@ -58,8 +58,6 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = {
 	IMPLICIT_FB_FIXED_DEV(0x0499, 0x172f, 0x81, 2), /* Steinberg UR22C */
 	IMPLICIT_FB_FIXED_DEV(0x0d9a, 0x00df, 0x81, 2), /* RTX6001 */
 	IMPLICIT_FB_FIXED_DEV(0x22f0, 0x0006, 0x81, 3), /* Allen&Heath Qu-16 */
-	IMPLICIT_FB_FIXED_DEV(0x2b73, 0x000a, 0x82, 0), /* Pioneer DJ DJM-900NXS2 */
-	IMPLICIT_FB_FIXED_DEV(0x2b73, 0x0017, 0x82, 0), /* Pioneer DJ DJM-250MK2 */
 	IMPLICIT_FB_FIXED_DEV(0x1686, 0xf029, 0x82, 2), /* Zoom UAC-2 */
 	IMPLICIT_FB_FIXED_DEV(0x2466, 0x8003, 0x86, 2), /* Fractal Audio Axe-Fx II */
 	IMPLICIT_FB_FIXED_DEV(0x0499, 0x172a, 0x86, 2), /* Yamaha MODX */
@@ -100,7 +98,7 @@ static const struct snd_usb_implicit_fb_match capture_implicit_fb_quirks[] = {
 /* set up sync EP information on the audioformat */
 static int add_implicit_fb_sync_ep(struct snd_usb_audio *chip,
 				   struct audioformat *fmt,
-				   int ep, int ifnum,
+				   int ep, int ep_idx, int ifnum,
 				   const struct usb_host_interface *alts)
 {
 	struct usb_interface *iface;
@@ -115,7 +113,7 @@ static int add_implicit_fb_sync_ep(struct snd_usb_audio *chip,
 	fmt->sync_ep = ep;
 	fmt->sync_iface = ifnum;
 	fmt->sync_altsetting = alts->desc.bAlternateSetting;
-	fmt->sync_ep_idx = 0;
+	fmt->sync_ep_idx = ep_idx;
 	fmt->implicit_fb = 1;
 	usb_audio_dbg(chip,
 		      "%d:%d: added %s implicit_fb sync_ep %x, iface %d:%d\n",
@@ -147,7 +145,7 @@ static int add_generic_uac2_implicit_fb(struct snd_usb_audio *chip,
 	    (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) !=
 					USB_ENDPOINT_USAGE_IMPLICIT_FB)
 		return 0;
-	return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress,
+	return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 0,
 				       ifnum, alts);
 }
 
@@ -173,10 +171,32 @@ static int add_roland_implicit_fb(struct snd_usb_audio *chip,
 	    (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) !=
 					USB_ENDPOINT_USAGE_IMPLICIT_FB)
 		return 0;
-	return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress,
+	return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 0,
 				       ifnum, alts);
 }
 
+/* Pioneer devices: playback and capture streams sharing the same iface/altset
+ */
+static int add_pioneer_implicit_fb(struct snd_usb_audio *chip,
+				   struct audioformat *fmt,
+				   struct usb_host_interface *alts)
+{
+	struct usb_endpoint_descriptor *epd;
+
+	if (alts->desc.bNumEndpoints != 2)
+		return 0;
+
+	epd = get_endpoint(alts, 1);
+	if (!usb_endpoint_is_isoc_in(epd) ||
+	    (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC ||
+	    ((epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) !=
+	     USB_ENDPOINT_USAGE_DATA &&
+	     (epd->bmAttributes & USB_ENDPOINT_USAGE_MASK) !=
+	     USB_ENDPOINT_USAGE_IMPLICIT_FB))
+		return 0;
+	return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 1,
+				       alts->desc.bInterfaceNumber, alts);
+}
 
 static int __add_generic_implicit_fb(struct snd_usb_audio *chip,
 				     struct audioformat *fmt,
@@ -197,7 +217,7 @@ static int __add_generic_implicit_fb(struct snd_usb_audio *chip,
 	if (!usb_endpoint_is_isoc_in(epd) ||
 	    (epd->bmAttributes & USB_ENDPOINT_SYNCTYPE) != USB_ENDPOINT_SYNC_ASYNC)
 		return 0;
-	return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress,
+	return add_implicit_fb_sync_ep(chip, fmt, epd->bEndpointAddress, 0,
 				       iface, alts);
 }
 
@@ -250,7 +270,7 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip,
 		case IMPLICIT_FB_NONE:
 			return 0; /* No quirk */
 		case IMPLICIT_FB_FIXED:
-			return add_implicit_fb_sync_ep(chip, fmt, p->ep_num,
+			return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, 0,
 						       p->iface, NULL);
 		}
 	}
@@ -278,6 +298,14 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip,
 			return 1;
 	}
 
+	/* Pioneer devices implicit feedback with vendor spec class */
+	if (attr == USB_ENDPOINT_SYNC_ASYNC &&
+	    alts->desc.bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
+	    USB_ID_VENDOR(chip->usb_id) == 0x2b73 /* Pioneer */) {
+		if (add_pioneer_implicit_fb(chip, fmt, alts))
+			return 1;
+	}
+
 	/* Try the generic implicit fb if available */
 	if (chip->generic_implicit_fb)
 		return add_generic_implicit_fb(chip, fmt, alts);
@@ -295,8 +323,8 @@ static int audioformat_capture_quirk(struct snd_usb_audio *chip,
 
 	p = find_implicit_fb_entry(chip, capture_implicit_fb_quirks, alts);
 	if (p && p->type == IMPLICIT_FB_FIXED)
-		return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, p->iface,
-					       NULL);
+		return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, 0,
+					       p->iface, NULL);
 	return 0;
 }
 
@@ -378,20 +406,19 @@ snd_usb_find_implicit_fb_sync_format(struct snd_usb_audio *chip,
 				     int stream)
 {
 	struct snd_usb_substream *subs;
-	const struct audioformat *fp, *sync_fmt;
+	const struct audioformat *fp, *sync_fmt = NULL;
 	int score, high_score;
 
-	/* When sharing the same altset, use the original audioformat */
+	/* Use the original audioformat as fallback for the shared altset */
 	if (target->iface == target->sync_iface &&
 	    target->altsetting == target->sync_altsetting)
-		return target;
+		sync_fmt = target;
 
 	subs = find_matching_substream(chip, stream, target->sync_ep,
 				       target->fmt_type);
 	if (!subs)
-		return NULL;
+		return sync_fmt;
 
-	sync_fmt = NULL;
 	high_score = 0;
 	list_for_each_entry(fp, &subs->fmt_list, list) {
 		score = match_endpoint_audioformats(subs, fp,

sound/usb/quirks-table.h

@@ -3362,6 +3362,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 					.altsetting = 1,
 					.altset_idx = 1,
 					.endpoint = 0x86,
+					.ep_idx = 1,
 					.ep_attr = USB_ENDPOINT_XFER_ISOC|
 						 USB_ENDPOINT_SYNC_ASYNC|
 						 USB_ENDPOINT_USAGE_IMPLICIT_FB,
@@ -3450,6 +3451,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 					.altsetting = 1,
 					.altset_idx = 1,
 					.endpoint = 0x82,
+					.ep_idx = 1,
 					.ep_attr = USB_ENDPOINT_XFER_ISOC|
 						USB_ENDPOINT_SYNC_ASYNC|
 						USB_ENDPOINT_USAGE_IMPLICIT_FB,
@@ -3506,6 +3508,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 					.altsetting = 1,
 					.altset_idx = 1,
 					.endpoint = 0x82,
+					.ep_idx = 1,
 					.ep_attr = USB_ENDPOINT_XFER_ISOC|
 						 USB_ENDPOINT_SYNC_ASYNC|
 						 USB_ENDPOINT_USAGE_IMPLICIT_FB,
@@ -3562,6 +3565,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 					.altsetting = 1,
 					.altset_idx = 1,
 					.endpoint = 0x82,
+					.ep_idx = 1,
 					.ep_attr = USB_ENDPOINT_XFER_ISOC|
 						 USB_ENDPOINT_SYNC_ASYNC|
 						 USB_ENDPOINT_USAGE_IMPLICIT_FB,
@@ -3619,6 +3623,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 					.altsetting = 1,
 					.altset_idx = 1,
 					.endpoint = 0x82,
+					.ep_idx = 1,
 					.ep_attr = USB_ENDPOINT_XFER_ISOC|
 						USB_ENDPOINT_SYNC_ASYNC|
 					USB_ENDPOINT_USAGE_IMPLICIT_FB,
@@ -3679,6 +3684,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 					.altsetting = 1,
 					.altset_idx = 1,
 					.endpoint = 0x82,
+					.ep_idx = 1,
 					.ep_attr = USB_ENDPOINT_XFER_ISOC|
 					    USB_ENDPOINT_SYNC_ASYNC|
 					    USB_ENDPOINT_USAGE_IMPLICIT_FB,

sound/usb/quirks.c

@@ -120,6 +120,40 @@ static int create_standard_audio_quirk(struct snd_usb_audio *chip,
 	return 0;
 }
 
+/* create the audio stream and the corresponding endpoints from the fixed
+ * audioformat object; this is used for quirks with the fixed EPs
+ */
+static int add_audio_stream_from_fixed_fmt(struct snd_usb_audio *chip,
+					   struct audioformat *fp)
+{
+	int stream, err;
+
+	stream = (fp->endpoint & USB_DIR_IN) ?
+		SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+
+	snd_usb_audioformat_set_sync_ep(chip, fp);
+
+	err = snd_usb_add_audio_stream(chip, stream, fp);
+	if (err < 0)
+		return err;
+
+	err = snd_usb_add_endpoint(chip, fp->endpoint,
+				   SND_USB_ENDPOINT_TYPE_DATA);
+	if (err < 0)
+		return err;
+
+	if (fp->sync_ep) {
+		err = snd_usb_add_endpoint(chip, fp->sync_ep,
+					   fp->implicit_fb ?
+					   SND_USB_ENDPOINT_TYPE_DATA :
+					   SND_USB_ENDPOINT_TYPE_SYNC);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
 /*
  * create a stream for an endpoint/altsetting without proper descriptors
  */
@@ -131,8 +165,8 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
 	struct audioformat *fp;
 	struct usb_host_interface *alts;
 	struct usb_interface_descriptor *altsd;
-	int stream, err;
 	unsigned *rate_table = NULL;
+	int err;
 
 	fp = kmemdup(quirk->data, sizeof(*fp), GFP_KERNEL);
 	if (!fp)
@@ -153,11 +187,6 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
 		fp->rate_table = rate_table;
 	}
 
-	stream = (fp->endpoint & USB_DIR_IN)
-		? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
-	err = snd_usb_add_audio_stream(chip, stream, fp);
-	if (err < 0)
-		goto error;
 	if (fp->iface != get_iface_desc(&iface->altsetting[0])->bInterfaceNumber ||
 	    fp->altset_idx >= iface->num_altsetting) {
 		err = -EINVAL;
@@ -165,7 +194,7 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
 	}
 	alts = &iface->altsetting[fp->altset_idx];
 	altsd = get_iface_desc(alts);
-	if (altsd->bNumEndpoints < 1) {
+	if (altsd->bNumEndpoints <= fp->ep_idx) {
 		err = -EINVAL;
 		goto error;
 	}
@@ -175,7 +204,14 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip,
 	if (fp->datainterval == 0)
 		fp->datainterval = snd_usb_parse_datainterval(chip, alts);
 	if (fp->maxpacksize == 0)
-		fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize);
+		fp->maxpacksize = le16_to_cpu(get_endpoint(alts, fp->ep_idx)->wMaxPacketSize);
+	if (!fp->fmt_type)
+		fp->fmt_type = UAC_FORMAT_TYPE_I;
+
+	err = add_audio_stream_from_fixed_fmt(chip, fp);
+	if (err < 0)
+		goto error;
+
 	usb_set_interface(chip->dev, fp->iface, 0);
 	snd_usb_init_pitch(chip, fp);
 	snd_usb_init_sample_rate(chip, fp, fp->rate_max);
@@ -417,7 +453,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
 	struct usb_host_interface *alts;
 	struct usb_interface_descriptor *altsd;
 	struct audioformat *fp;
-	int stream, err;
+	int err;
 
 	/* both PCM and MIDI interfaces have 2 or more altsettings */
 	if (iface->num_altsetting < 2)
@@ -482,9 +518,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip,
 		return -ENXIO;
 	}
 
-	stream = (fp->endpoint & USB_DIR_IN)
-		? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
-	err = snd_usb_add_audio_stream(chip, stream, fp);
+	err = add_audio_stream_from_fixed_fmt(chip, fp);
 	if (err < 0) {
 		list_del(&fp->list); /* unlink for avoiding double-free */
 		kfree(fp);

sound/usb/usbaudio.h

@@ -44,6 +44,7 @@ struct snd_usb_audio {
 
 	struct list_head pcm_list;	/* list of pcm streams */
 	struct list_head ep_list;	/* list of audio-related endpoints */
+	struct list_head iface_ref_list; /* list of interface refcounts */
 	int pcm_devs;
 
 	struct list_head midi_list;	/* list of midi interfaces */