Merge tag 'asoc-fix-v6.11-merge-window' of...

Merge tag 'asoc-fix-v6.11-merge-window' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v6.11

A selection of routine fixes and quirks that came in since the merge
window.  The fsl-asoc-card change is a fix for systems with multiple
cards where updating templates in place leaks data from one card to
another.

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

@@ -220,6 +220,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "21J6"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "21M5"),
+		}
+	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {

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);

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;

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

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>

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,
 };
 

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);

sound/soc/sof/intel/hda-loader.c

@@ -310,6 +310,7 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream
 		return ret;
 	}
 
+	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));
@@ -318,7 +319,10 @@ int hda_cl_copy_fw(struct snd_sof_dev *sdev, struct hdac_ext_stream *hext_stream
 			dev_err(sdev->dev, "Code loader DMA did not complete\n");
 			return -ETIMEDOUT;
 		}
-	dev_dbg(sdev->dev, "Code loader DMA done, waiting for FW_ENTERED status\n");
+		dev_dbg(sdev->dev, "Code loader DMA done\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,

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 &&

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) {
+		/*
+		 * 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;
 	}
 

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