Merge tag 'sound-fix-6.11-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A collection of fixes gathered since the previous pull.

  We see a bit large LOCs at a HD-audio quirk, but that's only bulk COEF
  data, hence it's safe to take. In addition to that, there were two
  minor fixes for MIDI 2.0 handling for ALSA core, and the rest are all
  rather random small and device-specific fixes"

* tag 'sound-fix-6.11-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ASoC: fsl-asoc-card: Dynamically allocate memory for snd_soc_dai_link_components
  ASoC: amd: yc: Support mic on Lenovo Thinkpad E16 Gen 2
  ALSA: hda/realtek: Implement sound init sequence for Samsung Galaxy Book3 Pro 360
  ALSA: hda/realtek: cs35l41: Fixup remaining asus strix models
  ASoC: SOF: ipc4-topology: Preserve the DMA Link ID for ChainDMA on unprepare
  ASoC: SOF: ipc4-topology: Only handle dai_config with HW_PARAMS for ChainDMA
  ALSA: ump: Force 1 Group for MIDI1 FBs
  ALSA: ump: Don't update FB name for static blocks
  ALSA: usb-audio: Add a quirk for Sonix HD USB Camera
  ASoC: TAS2781: Fix tasdev_load_calibrated_data()
  ASoC: tegra: select CONFIG_SND_SIMPLE_CARD_UTILS
  ASoC: Intel: use soc_intel_is_byt_cr() only when IOSF_MBI is reachable
  ALSA: usb-audio: Move HD Webcam quirk to the right place
  ALSA: hda: tas2781: mark const variables as __maybe_unused
  ALSA: usb-audio: Fix microphone sound on HD webcam.
  ASoC: sof: amd: fix for firmware reload failure in Vangogh platform
  ASoC: Intel: Fix RT5650 SSP lookup
  ASOC: SOF: Intel: hda-loader: only wait for HDaudio IOC for IPC4 devices
  ASoC: SOF: imx8m: Fix DSP control regmap retrieval

11 files changed, 85 insertions, 40 deletions

diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index f54466ed8e3e..1769e07e83dc 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -224,6 +224,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 		.driver_data = &acp6x_card,
 		.matches = {
 			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "21M5"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
 			DMI_MATCH(DMI_PRODUCT_NAME, "82QF"),
 		}
 	},
diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c
index 63626b982d04..8f9a3ae7153e 100644
--- a/sound/soc/codecs/tas2781-fmwlib.c
+++ b/sound/soc/codecs/tas2781-fmwlib.c
@@ -2162,7 +2162,7 @@ static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i)
 		return;
 
 	cal = cal_fmw->calibrations;
-	if (cal)
+	if (!cal)
 		return;
 
 	load_calib_data(priv, &cal->dev_data);
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 82df887b3af5..f6c3aeff0d8e 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -306,27 +306,12 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 	return 0;
 }
 
-SND_SOC_DAILINK_DEFS(hifi,
-	DAILINK_COMP_ARRAY(COMP_EMPTY()),
-	DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY()),
-	DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(hifi_fe,
-	DAILINK_COMP_ARRAY(COMP_EMPTY()),
-	DAILINK_COMP_ARRAY(COMP_DUMMY()),
-	DAILINK_COMP_ARRAY(COMP_EMPTY()));
-
-SND_SOC_DAILINK_DEFS(hifi_be,
-	DAILINK_COMP_ARRAY(COMP_EMPTY()),
-	DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY()));
-
 static const struct snd_soc_dai_link fsl_asoc_card_dai[] = {
 	/* Default ASoC DAI Link*/
 	{
 		.name = "HiFi",
 		.stream_name = "HiFi",
 		.ops = &fsl_asoc_card_ops,
-		SND_SOC_DAILINK_REG(hifi),
 	},
 	/* DPCM Link between Front-End and Back-End (Optional) */
 	{
@@ -335,7 +320,6 @@ static const struct snd_soc_dai_link fsl_asoc_card_dai[] = {
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
 		.dynamic = 1,
-		SND_SOC_DAILINK_REG(hifi_fe),
 	},
 	{
 		.name = "HiFi-ASRC-BE",
@@ -345,7 +329,6 @@ static const struct snd_soc_dai_link fsl_asoc_card_dai[] = {
 		.dpcm_playback = 1,
 		.dpcm_capture = 1,
 		.no_pcm = 1,
-		SND_SOC_DAILINK_REG(hifi_be),
 	},
 };
 
@@ -637,6 +620,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	struct platform_device *cpu_pdev;
 	struct fsl_asoc_card_priv *priv;
 	struct device *codec_dev[2] = { NULL, NULL };
+	struct snd_soc_dai_link_component *dlc;
 	const char *codec_dai_name[2];
 	const char *codec_dev_name[2];
 	u32 asrc_fmt = 0;
@@ -717,7 +701,35 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 
 	memcpy(priv->dai_link, fsl_asoc_card_dai,
 	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
+	/*
+	 * "Default ASoC DAI Link": 1 cpus, 2 codecs, 1 platforms
+	 * "DPCM Link Front-End":  1 cpus, 1 codecs (dummy), 1 platforms
+	 * "DPCM Link Back-End": 1 cpus, 2 codecs
+	 * totally 10 components
+	 */
+	dlc = devm_kcalloc(&pdev->dev, 10, sizeof(*dlc), GFP_KERNEL);
+	if (!dlc) {
+		ret = -ENOMEM;
+		goto asrc_fail;
+	}
+
+	priv->dai_link[0].cpus = &dlc[0];
+	priv->dai_link[0].num_cpus = 1;
+	priv->dai_link[0].codecs = &dlc[1];
 	priv->dai_link[0].num_codecs = 1;
+	priv->dai_link[0].platforms = &dlc[3];
+	priv->dai_link[0].num_platforms = 1;
+
+	priv->dai_link[1].cpus = &dlc[4];
+	priv->dai_link[1].num_cpus = 1;
+	priv->dai_link[1].codecs = &dlc[5];
+	priv->dai_link[1].num_codecs = 0; /* dummy */
+	priv->dai_link[1].platforms = &dlc[6];
+	priv->dai_link[1].num_platforms = 1;
+
+	priv->dai_link[2].cpus = &dlc[7];
+	priv->dai_link[2].num_cpus = 1;
+	priv->dai_link[2].codecs = &dlc[8];
 	priv->dai_link[2].num_codecs = 1;
 
 	priv->card.dapm_routes = audio_map;
diff --git a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
index 75d0b931d895..de7a3f7f47f1 100644
--- a/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
+++ b/sound/soc/intel/common/soc-acpi-intel-ssp-common.c
@@ -64,6 +64,15 @@ static const struct codec_map amps[] = {
 	CODEC_MAP_ENTRY("RT1015P", "rt1015", RT1015P_ACPI_HID, CODEC_RT1015P),
 	CODEC_MAP_ENTRY("RT1019P", "rt1019", RT1019P_ACPI_HID, CODEC_RT1019P),
 	CODEC_MAP_ENTRY("RT1308", "rt1308", RT1308_ACPI_HID, CODEC_RT1308),
+
+	/*
+	 * Monolithic components
+	 *
+	 * Only put components that can serve as both the amp and the codec below this line.
+	 * This will ensure that if the part is used just as a codec and there is an amp as well
+	 * then the amp will be selected properly.
+	 */
+	CODEC_MAP_ENTRY("RT5650", "rt5650", RT5650_ACPI_HID, CODEC_RT5650),
 };
 
 enum snd_soc_acpi_intel_codec
diff --git a/sound/soc/intel/common/soc-intel-quirks.h b/sound/soc/intel/common/soc-intel-quirks.h
index de4e550c5b34..42bd51456b94 100644
--- a/sound/soc/intel/common/soc-intel-quirks.h
+++ b/sound/soc/intel/common/soc-intel-quirks.h
@@ -11,7 +11,7 @@
 
 #include <linux/platform_data/x86/soc.h>
 
-#if IS_ENABLED(CONFIG_X86)
+#if IS_REACHABLE(CONFIG_IOSF_MBI)
 
 #include <linux/dmi.h>
 #include <asm/iosf_mbi.h>
diff --git a/sound/soc/sof/amd/pci-vangogh.c b/sound/soc/sof/amd/pci-vangogh.c
index 16eb2994fbab..eba580840100 100644
--- a/sound/soc/sof/amd/pci-vangogh.c
+++ b/sound/soc/sof/amd/pci-vangogh.c
@@ -34,7 +34,6 @@ static const struct sof_amd_acp_desc vangogh_chip_info = {
 	.dsp_intr_base	= ACP5X_DSP_SW_INTR_BASE,
 	.sram_pte_offset = ACP5X_SRAM_PTE_OFFSET,
 	.hw_semaphore_offset = ACP5X_AXI2DAGB_SEM_0,
-	.acp_clkmux_sel = ACP5X_CLKMUX_SEL,
 	.probe_reg_offset = ACP5X_FUTURE_REG_ACLK_0,
 };
 
diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c
index 1c7019c3cbd3..cdd1e79ef9f6 100644
--- a/sound/soc/sof/imx/imx8m.c
+++ b/sound/soc/sof/imx/imx8m.c
@@ -234,7 +234,7 @@ static int imx8m_probe(struct snd_sof_dev *sdev)
 	/* set default mailbox offset for FW ready message */
 	sdev->dsp_box.offset = MBOX_OFFSET;
 
-	priv->regmap = syscon_regmap_lookup_by_compatible("fsl,dsp-ctrl");
+	priv->regmap = syscon_regmap_lookup_by_phandle(np, "fsl,dsp-ctrl");
 	if (IS_ERR(priv->regmap)) {
 		dev_err(sdev->dev, "cannot find dsp-ctrl registers");
 		ret = PTR_ERR(priv->regmap);
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index b8b914eaf7e0..75f6240cf3e1 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -310,15 +310,19 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream
 		return ret;
 	}
 
-	/* Wait for completion of transfer */
-	time_left = wait_for_completion_timeout(&hda_stream->ioc,
-						msecs_to_jiffies(HDA_CL_DMA_IOC_TIMEOUT_MS));
-
-	if (!time_left) {
-		dev_err(sdev->dev, "Code loader DMA did not complete\n");
-		return -ETIMEDOUT;
+	if (sdev->pdata->ipc_type == SOF_IPC_TYPE_4) {
+		/* Wait for completion of transfer */
+		time_left = wait_for_completion_timeout(&hda_stream->ioc,
+							msecs_to_jiffies(HDA_CL_DMA_IOC_TIMEOUT_MS));
+
+		if (!time_left) {
+			dev_err(sdev->dev, "Code loader DMA did not complete\n");
+			return -ETIMEDOUT;
+		}
+		dev_dbg(sdev->dev, "Code loader DMA done\n");
 	}
-	dev_dbg(sdev->dev, "Code loader DMA done, waiting for FW_ENTERED status\n");
+
+	dev_dbg(sdev->dev, "waiting for FW_ENTERED status\n");
 
 	status = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR,
 					chip->rom_status_reg, reg,
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index daf364f773dd..5a40b8fbbbd3 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -1307,9 +1307,10 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev)
 	const struct sof_dev_desc *desc = sof_pdata->desc;
 	struct hdac_bus *bus = sof_to_bus(sdev);
 	struct snd_soc_acpi_mach *mach = NULL;
-	enum snd_soc_acpi_intel_codec codec_type;
+	enum snd_soc_acpi_intel_codec codec_type, amp_type;
 	const char *tplg_filename;
 	const char *tplg_suffix;
+	bool amp_name_valid;
 
 	/* Try I2S or DMIC if it is supported */
 	if (interface_mask & (BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC)))
@@ -1413,15 +1414,16 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev)
 			}
 		}
 
-		codec_type = snd_soc_acpi_intel_detect_amp_type(sdev->dev);
+		amp_type = snd_soc_acpi_intel_detect_amp_type(sdev->dev);
+		codec_type = snd_soc_acpi_intel_detect_codec_type(sdev->dev);
+		amp_name_valid = amp_type != CODEC_NONE && amp_type != codec_type;
 
-		if (tplg_fixup &&
-		    mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_AMP_NAME &&
-		    codec_type != CODEC_NONE) {
-			tplg_suffix = snd_soc_acpi_intel_get_amp_tplg_suffix(codec_type);
+		if (tplg_fixup && amp_name_valid &&
+		    mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_AMP_NAME) {
+			tplg_suffix = snd_soc_acpi_intel_get_amp_tplg_suffix(amp_type);
 			if (!tplg_suffix) {
 				dev_err(sdev->dev, "no tplg suffix found, amp %d\n",
-					codec_type);
+					amp_type);
 				return NULL;
 			}
 
@@ -1436,7 +1438,6 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev)
 			add_extension = true;
 		}
 
-		codec_type = snd_soc_acpi_intel_detect_codec_type(sdev->dev);
 
 		if (tplg_fixup &&
 		    mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_CODEC_NAME &&
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index 90f6856ee80c..87be7f16e8c2 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -1358,7 +1358,13 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget)
 		ipc4_copier = dai->private;
 
 		if (pipeline->use_chain_dma) {
-			pipeline->msg.primary = 0;
+			/*
+			 * Preserve the DMA Link ID and clear other bits since
+			 * the DMA Link ID is only configured once during
+			 * dai_config, other fields are expected to be 0 for
+			 * re-configuration
+			 */
+			pipeline->msg.primary &= SOF_IPC4_GLB_CHAIN_DMA_LINK_ID_MASK;
 			pipeline->msg.extension = 0;
 		}
 
@@ -3095,8 +3101,14 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
 		return 0;
 
 	if (pipeline->use_chain_dma) {
-		pipeline->msg.primary &= ~SOF_IPC4_GLB_CHAIN_DMA_LINK_ID_MASK;
-		pipeline->msg.primary |= SOF_IPC4_GLB_CHAIN_DMA_LINK_ID(data->dai_data);
+		/*
+		 * Only configure the DMA Link ID for ChainDMA when this op is
+		 * invoked with SOF_DAI_CONFIG_FLAGS_HW_PARAMS
+		 */
+		if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) {
+			pipeline->msg.primary &= ~SOF_IPC4_GLB_CHAIN_DMA_LINK_ID_MASK;
+			pipeline->msg.primary |= SOF_IPC4_GLB_CHAIN_DMA_LINK_ID(data->dai_data);
+		}
 		return 0;
 	}
 
diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index 74effc57a7a0..2463c22e9cf6 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -78,6 +78,7 @@ config SND_SOC_TEGRA210_DMIC
 
 config SND_SOC_TEGRA210_I2S
 	tristate "Tegra210 I2S module"
+	select SND_SIMPLE_CARD_UTILS
 	help
 	  Config to enable the Inter-IC Sound (I2S) Controller which
 	  implements full-duplex and bidirectional and single direction