Merge tag 'asoc-fix-v6.19-rc1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v6.19

We've been quite busy with fixes since the merge window, though not in
any particularly exciting ways - the standout thing is the fix for _SX
controls which were broken by a change to how we do clamping, otherwise
it's all fairly run of the mill fixes and quirks.

sound/firewire/tascam/tascam-stream.c

@@ -282,20 +282,22 @@ static int keep_resources(struct snd_tscm *tscm, unsigned int rate,
 			  struct amdtp_stream *stream)
 {
 	struct fw_iso_resources *resources;
+	int speed;
 	int err;
 
-	if (stream == &tscm->tx_stream)
+	if (stream == &tscm->tx_stream) {
 		resources = &tscm->tx_resources;
-	else
+		speed = fw_parent_device(tscm->unit)->max_speed;
+	} else {
 		resources = &tscm->rx_resources;
+		speed = SCODE_400;
+	}
 
 	err = amdtp_tscm_set_parameters(stream, rate);
 	if (err < 0)
 		return err;
 
-	return fw_iso_resources_allocate(resources,
-				amdtp_stream_get_max_payload(stream),
-				fw_parent_device(tscm->unit)->max_speed);
+	return fw_iso_resources_allocate(resources, amdtp_stream_get_max_payload(stream), speed);
 }
 
 static int init_stream(struct snd_tscm *tscm, struct amdtp_stream *s)
@@ -455,7 +457,6 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate)
 	}
 
 	if (!amdtp_stream_running(&tscm->rx_stream)) {
-		int spd = fw_parent_device(tscm->unit)->max_speed;
 		unsigned int tx_init_skip_cycles;
 
 		err = set_stream_formats(tscm, rate);
@@ -466,13 +467,13 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate)
 		if (err < 0)
 			goto error;
 
-		err = amdtp_domain_add_stream(&tscm->domain, &tscm->rx_stream,
-					      tscm->rx_resources.channel, spd);
+		err = amdtp_domain_add_stream(&tscm->domain, &tscm->rx_stream, tscm->rx_resources.channel,
+					      fw_parent_device(tscm->unit)->max_speed);
 		if (err < 0)
 			goto error;
 
-		err = amdtp_domain_add_stream(&tscm->domain, &tscm->tx_stream,
-					      tscm->tx_resources.channel, spd);
+		err = amdtp_domain_add_stream(&tscm->domain, &tscm->tx_stream, tscm->tx_resources.channel,
+					      SCODE_400);
 		if (err < 0)
 			goto error;
 

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

@@ -661,6 +661,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "Bravo 15 C7UCX"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "HONOR"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "GOH-X"),
+		}
+	},
 	{}
 };
 

sound/soc/codecs/ak4458.c

@@ -783,16 +783,12 @@ static int ak4458_i2c_probe(struct i2c_client *i2c)
 
 	pm_runtime_enable(&i2c->dev);
 	regcache_cache_only(ak4458->regmap, true);
-	ak4458_reset(ak4458, false);
 
 	return 0;
 }
 
 static void ak4458_i2c_remove(struct i2c_client *i2c)
 {
-	struct ak4458_priv *ak4458 = i2c_get_clientdata(i2c);
-
-	ak4458_reset(ak4458, true);
 	pm_runtime_disable(&i2c->dev);
 }
 

sound/soc/codecs/rt1320-sdw.c

@@ -115,7 +115,8 @@ static const struct reg_sequence rt1320_blind_write[] = {
 static const struct reg_sequence rt1320_vc_blind_write[] = {
 	{ 0xc003, 0xe0 },
 	{ 0xe80a, 0x01 },
-	{ 0xc5c3, 0xf3 },
+	{ 0xc5c3, 0xf2 },
+	{ 0xc5c8, 0x03 },
 	{ 0xc057, 0x51 },
 	{ 0xc054, 0x35 },
 	{ 0xca05, 0xd6 },
@@ -126,8 +127,6 @@ static const struct reg_sequence rt1320_vc_blind_write[] = {
 	{ 0xc609, 0x40 },
 	{ 0xc046, 0xff },
 	{ 0xc045, 0xff },
-	{ 0xda81, 0x14 },
-	{ 0xda8d, 0x14 },
 	{ 0xc044, 0xff },
 	{ 0xc043, 0xff },
 	{ 0xc042, 0xff },
@@ -136,8 +135,8 @@ static const struct reg_sequence rt1320_vc_blind_write[] = {
 	{ 0xcc10, 0x01 },
 	{ 0xc700, 0xf0 },
 	{ 0xc701, 0x13 },
-	{ 0xc901, 0x09 },
-	{ 0xc900, 0xd0 },
+	{ 0xc901, 0x04 },
+	{ 0xc900, 0x73 },
 	{ 0xde03, 0x05 },
 	{ 0xdd0b, 0x0d },
 	{ 0xdd0a, 0xff },
@@ -153,6 +152,7 @@ static const struct reg_sequence rt1320_vc_blind_write[] = {
 	{ 0xf082, 0xff },
 	{ 0xf081, 0xff },
 	{ 0xf080, 0xff },
+	{ 0xe801, 0x01 },
 	{ 0xe802, 0xf8 },
 	{ 0xe803, 0xbe },
 	{ 0xc003, 0xc0 },
@@ -202,7 +202,7 @@ static const struct reg_sequence rt1320_vc_blind_write[] = {
 	{ 0x3fc2bfc3, 0x00 },
 	{ 0x3fc2bfc2, 0x00 },
 	{ 0x3fc2bfc1, 0x00 },
-	{ 0x3fc2bfc0, 0x03 },
+	{ 0x3fc2bfc0, 0x07 },
 	{ 0x0000d486, 0x43 },
 	{ SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, RT1320_SDCA_CTL_REQ_POWER_STATE, 0), 0x00 },
 	{ 0x1000db00, 0x07 },
@@ -241,9 +241,7 @@ static const struct reg_sequence rt1320_vc_blind_write[] = {
 	{ 0x1000db21, 0x00 },
 	{ 0x1000db22, 0x00 },
 	{ 0x1000db23, 0x00 },
-	{ 0x0000d540, 0x01 },
-	{ 0x0000c081, 0xfc },
-	{ 0x0000f01e, 0x80 },
+	{ 0x0000d540, 0x21 },
 	{ 0xc01b, 0xfc },
 	{ 0xc5d1, 0x89 },
 	{ 0xc5d8, 0x0a },

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

@@ -1045,8 +1045,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	 * The notifier is initialized in snd_soc_card_jack_new(), then
 	 * snd_soc_jack_notifier_register can be called.
 	 */
-	if (of_property_read_bool(np, "hp-det-gpios") ||
-	    of_property_read_bool(np, "hp-det-gpio") /* deprecated */) {
+	if (of_property_present(np, "hp-det-gpios") ||
+	    of_property_present(np, "hp-det-gpio") /* deprecated */) {
 		ret = simple_util_init_jack(&priv->card, &priv->hp_jack,
 					    1, NULL, "Headphone Jack");
 		if (ret)
@@ -1055,8 +1055,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 		snd_soc_jack_notifier_register(&priv->hp_jack.jack, &hp_jack_nb);
 	}
 
-	if (of_property_read_bool(np, "mic-det-gpios") ||
-	    of_property_read_bool(np, "mic-det-gpio") /* deprecated */) {
+	if (of_property_present(np, "mic-det-gpios") ||
+	    of_property_present(np, "mic-det-gpio") /* deprecated */) {
 		ret = simple_util_init_jack(&priv->card, &priv->mic_jack,
 					    0, NULL, "Mic Jack");
 		if (ret)

sound/soc/fsl/fsl_asrc_dma.c

@@ -473,5 +473,8 @@ struct snd_soc_component_driver fsl_asrc_component = {
 	.pointer	= fsl_asrc_dma_pcm_pointer,
 	.pcm_construct	= fsl_asrc_dma_pcm_new,
 	.legacy_dai_naming = 1,
+#ifdef CONFIG_DEBUG_FS
+	.debugfs_prefix	= "asrc",
+#endif
 };
 EXPORT_SYMBOL_GPL(fsl_asrc_component);

sound/soc/fsl/fsl_easrc.c

@@ -1577,6 +1577,9 @@ static const struct snd_soc_component_driver fsl_easrc_component = {
 	.controls		= fsl_easrc_snd_controls,
 	.num_controls		= ARRAY_SIZE(fsl_easrc_snd_controls),
 	.legacy_dai_naming	= 1,
+#ifdef CONFIG_DEBUG_FS
+	.debugfs_prefix		= "easrc",
+#endif
 };
 
 static const struct reg_default fsl_easrc_reg_defaults[] = {

sound/soc/fsl/fsl_sai.c

@@ -917,8 +917,14 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream,
 					   tx ? sai->dma_params_tx.maxburst :
 					   sai->dma_params_rx.maxburst);
 
-	ret = snd_pcm_hw_constraint_list(substream->runtime, 0,
-					 SNDRV_PCM_HW_PARAM_RATE, &sai->constraint_rates);
+	if (sai->is_consumer_mode[tx])
+		ret = snd_pcm_hw_constraint_list(substream->runtime, 0,
+						 SNDRV_PCM_HW_PARAM_RATE,
+						 &fsl_sai_rate_constraints);
+	else
+		ret = snd_pcm_hw_constraint_list(substream->runtime, 0,
+						 SNDRV_PCM_HW_PARAM_RATE,
+						 &sai->constraint_rates);
 
 	return ret;
 }
@@ -1075,6 +1081,7 @@ static const struct reg_default fsl_sai_reg_defaults_ofs0[] = {
 	{FSL_SAI_TDR6, 0},
 	{FSL_SAI_TDR7, 0},
 	{FSL_SAI_TMR, 0},
+	{FSL_SAI_TTCTL, 0},
 	{FSL_SAI_RCR1(0), 0},
 	{FSL_SAI_RCR2(0), 0},
 	{FSL_SAI_RCR3(0), 0},
@@ -1098,12 +1105,14 @@ static const struct reg_default fsl_sai_reg_defaults_ofs8[] = {
 	{FSL_SAI_TDR6, 0},
 	{FSL_SAI_TDR7, 0},
 	{FSL_SAI_TMR, 0},
+	{FSL_SAI_TTCTL, 0},
 	{FSL_SAI_RCR1(8), 0},
 	{FSL_SAI_RCR2(8), 0},
 	{FSL_SAI_RCR3(8), 0},
 	{FSL_SAI_RCR4(8), 0},
 	{FSL_SAI_RCR5(8), 0},
 	{FSL_SAI_RMR, 0},
+	{FSL_SAI_RTCTL, 0},
 	{FSL_SAI_MCTL, 0},
 	{FSL_SAI_MDIV, 0},
 };

sound/soc/fsl/fsl_xcvr.c

@@ -1323,6 +1323,7 @@ static const struct reg_default fsl_xcvr_phy_reg_defaults[] = {
 };
 
 static const struct regmap_config fsl_xcvr_regmap_phy_cfg = {
+	.name = "phy",
 	.reg_bits = 8,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -1335,6 +1336,7 @@ static const struct regmap_config fsl_xcvr_regmap_phy_cfg = {
 };
 
 static const struct regmap_config fsl_xcvr_regmap_pllv0_cfg = {
+	.name = "pllv0",
 	.reg_bits = 8,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -1345,6 +1347,7 @@ static const struct regmap_config fsl_xcvr_regmap_pllv0_cfg = {
 };
 
 static const struct regmap_config fsl_xcvr_regmap_pllv1_cfg = {
+	.name = "pllv1",
 	.reg_bits = 8,
 	.reg_stride = 4,
 	.val_bits = 32,

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

@@ -699,6 +699,69 @@ static const struct snd_soc_acpi_adr_device cs35l56_1_fb_adr[] = {
 	},
 };
 
+static const struct snd_soc_acpi_adr_device cs35l56_6amp_1_fb_adr[] = {
+	{
+		.adr = 0x00013701FA355601ull,
+		.num_endpoints = ARRAY_SIZE(cs35l56_r_fb_endpoints),
+		.endpoints = cs35l56_r_fb_endpoints,
+		.name_prefix = "AMP6"
+	},
+	{
+		.adr = 0x00013601FA355601ull,
+		.num_endpoints = ARRAY_SIZE(cs35l56_3_fb_endpoints),
+		.endpoints = cs35l56_3_fb_endpoints,
+		.name_prefix = "AMP5"
+	},
+	{
+		.adr = 0x00013501FA355601ull,
+		.num_endpoints = ARRAY_SIZE(cs35l56_5_fb_endpoints),
+		.endpoints = cs35l56_5_fb_endpoints,
+		.name_prefix = "AMP4"
+	},
+};
+
+static const struct snd_soc_acpi_adr_device cs35l63_6amp_3_fb_adr[] = {
+	{
+		.adr = 0x00033001FA356301ull,
+		.num_endpoints = ARRAY_SIZE(cs35l56_l_fb_endpoints),
+		.endpoints = cs35l56_l_fb_endpoints,
+		.name_prefix = "AMP1"
+	},
+	{
+		.adr = 0x00033201FA356301ull,
+		.num_endpoints = ARRAY_SIZE(cs35l56_2_fb_endpoints),
+		.endpoints = cs35l56_2_fb_endpoints,
+		.name_prefix = "AMP3"
+	},
+	{
+		.adr = 0x00033401FA356301ull,
+		.num_endpoints = ARRAY_SIZE(cs35l56_4_fb_endpoints),
+		.endpoints = cs35l56_4_fb_endpoints,
+		.name_prefix = "AMP5"
+	},
+};
+
+static const struct snd_soc_acpi_adr_device cs35l63_6amp_2_fb_adr[] = {
+	{
+		.adr = 0x00023101FA356301ull,
+		.num_endpoints = ARRAY_SIZE(cs35l56_r_fb_endpoints),
+		.endpoints = cs35l56_r_fb_endpoints,
+		.name_prefix = "AMP2"
+	},
+	{
+		.adr = 0x00023301FA356301ull,
+		.num_endpoints = ARRAY_SIZE(cs35l56_3_fb_endpoints),
+		.endpoints = cs35l56_3_fb_endpoints,
+		.name_prefix = "AMP4"
+	},
+	{
+		.adr = 0x00023501FA356301ull,
+		.num_endpoints = ARRAY_SIZE(cs35l56_5_fb_endpoints),
+		.endpoints = cs35l56_5_fb_endpoints,
+		.name_prefix = "AMP6"
+	},
+};
+
 static const struct snd_soc_acpi_adr_device cs35l56_2_r_adr[] = {
 	{
 		.adr = 0x00023201FA355601ull,
@@ -1069,6 +1132,35 @@ static const struct snd_soc_acpi_link_adr mtl_cs35l56_x8_link0_link1_fb[] = {
 	{}
 };
 
+static const struct snd_soc_acpi_link_adr mtl_cs35l56_x6_link0_link1_fb[] = {
+	{
+		.mask = BIT(1),
+		.num_adr = ARRAY_SIZE(cs35l56_6amp_1_fb_adr),
+		.adr_d = cs35l56_6amp_1_fb_adr,
+	},
+	{
+		.mask = BIT(0),
+		/* First 3 amps in cs35l56_0_fb_adr */
+		.num_adr = 3,
+		.adr_d = cs35l56_0_fb_adr,
+	},
+	{}
+};
+
+static const struct snd_soc_acpi_link_adr mtl_cs35l63_x6_link2_link3_fb[] = {
+	{
+		.mask = BIT(3),
+		.num_adr = ARRAY_SIZE(cs35l63_6amp_3_fb_adr),
+		.adr_d = cs35l63_6amp_3_fb_adr,
+	},
+	{
+		.mask = BIT(2),
+		.num_adr = ARRAY_SIZE(cs35l63_6amp_2_fb_adr),
+		.adr_d = cs35l63_6amp_2_fb_adr,
+	},
+	{}
+};
+
 static const struct snd_soc_acpi_link_adr mtl_cs35l63_x2_link1_link3_fb[] = {
 	{
 		.mask = BIT(3),
@@ -1189,6 +1281,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = {
 		.sof_tplg_filename = "sof-mtl-cs35l56-l01-fb8.tplg",
 		.get_function_tplg_files = sof_sdw_get_tplg_files,
 	},
+	{
+		.link_mask = BIT(0) | BIT(1),
+		.links = mtl_cs35l56_x6_link0_link1_fb,
+		.drv_name = "sof_sdw",
+		.sof_tplg_filename = "sof-mtl-cs35l56-l01-fb6.tplg"
+	},
 	{
 		.link_mask = BIT(0),
 		.links = mtl_cs42l43_l0,
@@ -1202,6 +1300,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[] = {
 		.drv_name = "sof_sdw",
 		.sof_tplg_filename = "sof-mtl-cs35l56-l01-fb8.tplg",
 	},
+	{
+		.link_mask = BIT(2) | BIT(3),
+		.links = mtl_cs35l63_x6_link2_link3_fb,
+		.drv_name = "sof_sdw",
+		.sof_tplg_filename = "sof-mtl-cs35l56-l01-fb6.tplg",
+	},
 	{
 		.link_mask = GENMASK(3, 0),
 		.links = mtl_3_in_1_sdca,

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

@@ -15,49 +15,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_nvl_machines[] = {
 };
 EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_nvl_machines);
 
-/*
- * Multi-function codecs with three endpoints created for
- * headset, amp and dmic functions.
- */
-static const struct snd_soc_acpi_endpoint rt_mf_endpoints[] = {
-	{
-		.num = 0,
-		.aggregated = 0,
-		.group_position = 0,
-		.group_id = 0,
-	},
-	{
-		.num = 1,
-		.aggregated = 0,
-		.group_position = 0,
-		.group_id = 0,
-	},
-	{
-		.num = 2,
-		.aggregated = 0,
-		.group_position = 0,
-		.group_id = 0,
-	},
-};
-
-static const struct snd_soc_acpi_adr_device rt722_3_single_adr[] = {
-	{
-		.adr = 0x000330025d072201ull,
-		.num_endpoints = ARRAY_SIZE(rt_mf_endpoints),
-		.endpoints = rt_mf_endpoints,
-		.name_prefix = "rt722"
-	}
-};
-
-static const struct snd_soc_acpi_link_adr nvl_rt722_l3[] = {
-	{
-		.mask = BIT(3),
-		.num_adr = ARRAY_SIZE(rt722_3_single_adr),
-		.adr_d = rt722_3_single_adr,
-	},
-	{}
-};
-
 /* this table is used when there is no I2S codec present */
 struct snd_soc_acpi_mach snd_soc_acpi_intel_nvl_sdw_machines[] = {
 	/* mockup tests need to be first */
@@ -79,12 +36,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_nvl_sdw_machines[] = {
 		.drv_name = "sof_sdw",
 		.sof_tplg_filename = "sof-nvl-rt715-rt711-rt1308-mono.tplg",
 	},
-	{
-		.link_mask = BIT(3),
-		.links = nvl_rt722_l3,
-		.drv_name = "sof_sdw",
-		.sof_tplg_filename = "sof-nvl-rt722.tplg",
-	},
 	{},
 };
 EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_nvl_sdw_machines);

sound/soc/intel/common/sof-function-topology-lib.c

@@ -28,7 +28,7 @@ enum tplg_device_id {
 #define SOF_INTEL_PLATFORM_NAME_MAX 4
 
 int sof_sdw_get_tplg_files(struct snd_soc_card *card, const struct snd_soc_acpi_mach *mach,
-			   const char *prefix, const char ***tplg_files)
+			   const char *prefix, const char ***tplg_files, bool best_effort)
 {
 	struct snd_soc_acpi_mach_params mach_params = mach->mach_params;
 	struct snd_soc_dai_link *dai_link;
@@ -87,6 +87,9 @@ int sof_sdw_get_tplg_files(struct snd_soc_card *card, const struct snd_soc_acpi_
 			dev_dbg(card->dev,
 				"dai_link %s is not supported by separated tplg yet\n",
 				dai_link->name);
+			if (best_effort)
+				continue;
+
 			return 0;
 		}
 		if (tplg_mask & BIT(tplg_dev))

sound/soc/intel/common/sof-function-topology-lib.h

@@ -10,6 +10,6 @@
 #define _SND_SOC_ACPI_INTEL_GET_TPLG_H
 
 int sof_sdw_get_tplg_files(struct snd_soc_card *card, const struct snd_soc_acpi_mach *mach,
-			   const char *prefix, const char ***tplg_files);
+			   const char *prefix, const char ***tplg_files, bool best_effort);
 
 #endif

sound/soc/qcom/sdm845.c

@@ -365,10 +365,12 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream)
 		snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt);
 		break;
 	case QUATERNARY_MI2S_RX:
+		codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_I2S;
 		snd_soc_dai_set_sysclk(cpu_dai,
 			Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT,
 			MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK);
 		snd_soc_dai_set_fmt(cpu_dai, fmt);
+		snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt);
 		break;
 
 	case QUATERNARY_TDM_RX_0:

sound/soc/sdw_utils/soc_sdw_utils.c

@@ -1534,8 +1534,10 @@ int asoc_sdw_parse_sdw_endpoints(struct snd_soc_card *card,
 					 * endpoint check is not necessary
 					 */
 					if (dai_info->quirk &&
-					    !(dai_info->quirk_exclude ^ !!(dai_info->quirk & ctx->mc_quirk)))
+					    !(dai_info->quirk_exclude ^ !!(dai_info->quirk & ctx->mc_quirk))) {
+						(*num_devs)--;
 						continue;
+					}
 				} else {
 					/* Check SDCA codec endpoint if there is no matching quirk */
 					ret = is_sdca_endpoint_present(dev, codec_info, adr_link, i, j);
@@ -1543,8 +1545,10 @@ int asoc_sdw_parse_sdw_endpoints(struct snd_soc_card *card,
 						return ret;
 
 					/* The endpoint is not present, skip */
-					if (!ret)
+					if (!ret) {
+						(*num_devs)--;
 						continue;
+					}
 				}
 
 				dev_dbg(dev,

sound/soc/soc-ops.c

@@ -111,7 +111,8 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol,
 EXPORT_SYMBOL_GPL(snd_soc_put_enum_double);
 
 static int sdca_soc_q78_reg_to_ctl(struct soc_mixer_control *mc, unsigned int reg_val,
-				unsigned int mask, unsigned int shift, int max)
+				   unsigned int mask, unsigned int shift, int max,
+				   bool sx)
 {
 	int val = reg_val;
 
@@ -141,20 +142,26 @@ static unsigned int sdca_soc_q78_ctl_to_reg(struct soc_mixer_control *mc, int va
 }
 
 static int soc_mixer_reg_to_ctl(struct soc_mixer_control *mc, unsigned int reg_val,
-				unsigned int mask, unsigned int shift, int max)
+				unsigned int mask, unsigned int shift, int max,
+				bool sx)
 {
 	int val = (reg_val >> shift) & mask;
 
 	if (mc->sign_bit)
 		val = sign_extend32(val, mc->sign_bit);
 
-	val = clamp(val, mc->min, mc->max);
-	val -= mc->min;
+	if (sx) {
+		val -= mc->min; // SX controls intentionally can overflow here
+		val = min_t(unsigned int, val & mask, max);
+	} else {
+		val = clamp(val, mc->min, mc->max);
+		val -= mc->min;
+	}
 
 	if (mc->invert)
 		val = max - val;
 
-	return val & mask;
+	return val;
 }
 
 static unsigned int soc_mixer_ctl_to_reg(struct soc_mixer_control *mc, int val,
@@ -280,9 +287,10 @@ static int soc_put_volsw(struct snd_kcontrol *kcontrol,
 
 static int soc_get_volsw(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol,
-			 struct soc_mixer_control *mc, int mask, int max)
+			 struct soc_mixer_control *mc, int mask, int max, bool sx)
 {
-	int (*reg_to_ctl)(struct soc_mixer_control *, unsigned int, unsigned int, unsigned int, int);
+	int (*reg_to_ctl)(struct soc_mixer_control *, unsigned int, unsigned int,
+			  unsigned int, int, bool);
 	struct snd_soc_component *component = snd_kcontrol_chip(kcontrol);
 	unsigned int reg_val;
 	int val;
@@ -293,16 +301,16 @@ static int soc_get_volsw(struct snd_kcontrol *kcontrol,
 		reg_to_ctl = soc_mixer_reg_to_ctl;
 
 	reg_val = snd_soc_component_read(component, mc->reg);
-	val = reg_to_ctl(mc, reg_val, mask, mc->shift, max);
+	val = reg_to_ctl(mc, reg_val, mask, mc->shift, max, sx);
 
 	ucontrol->value.integer.value[0] = val;
 
 	if (snd_soc_volsw_is_stereo(mc)) {
 		if (mc->reg == mc->rreg) {
-			val = reg_to_ctl(mc, reg_val, mask, mc->rshift, max);
+			val = reg_to_ctl(mc, reg_val, mask, mc->rshift, max, sx);
 		} else {
 			reg_val = snd_soc_component_read(component, mc->rreg);
-			val = reg_to_ctl(mc, reg_val, mask, mc->shift, max);
+			val = reg_to_ctl(mc, reg_val, mask, mc->shift, max, sx);
 		}
 
 		ucontrol->value.integer.value[1] = val;
@@ -371,7 +379,7 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
 		(struct soc_mixer_control *)kcontrol->private_value;
 	unsigned int mask = soc_mixer_mask(mc);
 
-	return soc_get_volsw(kcontrol, ucontrol, mc, mask, mc->max - mc->min);
+	return soc_get_volsw(kcontrol, ucontrol, mc, mask, mc->max - mc->min, false);
 }
 EXPORT_SYMBOL_GPL(snd_soc_get_volsw);
 
@@ -413,7 +421,7 @@ int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol,
 		(struct soc_mixer_control *)kcontrol->private_value;
 	unsigned int mask = soc_mixer_sx_mask(mc);
 
-	return soc_get_volsw(kcontrol, ucontrol, mc, mask, mc->max);
+	return soc_get_volsw(kcontrol, ucontrol, mc, mask, mc->max, true);
 }
 EXPORT_SYMBOL_GPL(snd_soc_get_volsw_sx);
 

sound/soc/sof/intel/hda-sdw-bpt.c

@@ -10,6 +10,7 @@
  * Hardware interface for SoundWire BPT support with HDA DMA
  */
 
+#include <linux/lcm.h>
 #include <sound/hdaudio_ext.h>
 #include <sound/hda-mlink.h>
 #include <sound/hda-sdw-bpt.h>
@@ -236,6 +237,18 @@ static int hda_sdw_bpt_dma_disable(struct device *dev, struct hdac_ext_stream *s
 	return ret;
 }
 
+#define FIFO_ALIGNMENT	64
+
+unsigned int hda_sdw_bpt_get_buf_size_alignment(unsigned int dma_bandwidth)
+{
+	unsigned int num_channels = DIV_ROUND_UP(dma_bandwidth, BPT_FREQUENCY * 32);
+	unsigned int data_block = num_channels * 4;
+	unsigned int alignment = lcm(data_block, FIFO_ALIGNMENT);
+
+	return alignment;
+}
+EXPORT_SYMBOL_NS(hda_sdw_bpt_get_buf_size_alignment, "SND_SOC_SOF_INTEL_HDA_SDW_BPT");
+
 int hda_sdw_bpt_open(struct device *dev, int link_id, struct hdac_ext_stream **bpt_tx_stream,
 		     struct snd_dma_buffer *dmab_tx_bdl, u32 bpt_tx_num_bytes,
 		     u32 tx_dma_bandwidth, struct hdac_ext_stream **bpt_rx_stream,

sound/soc/sof/intel/pci-mtl.c

@@ -47,7 +47,7 @@ static const struct sof_dev_desc mtl_desc = {
 		[SOF_IPC_TYPE_4] = "intel/sof-ipc4-lib/mtl",
 	},
 	.default_tplg_path = {
-		[SOF_IPC_TYPE_4] = "intel/sof-ace-tplg",
+		[SOF_IPC_TYPE_4] = "intel/sof-ipc4-tplg",
 	},
 	.default_fw_filename = {
 		[SOF_IPC_TYPE_4] = "sof-mtl.ri",
@@ -77,7 +77,7 @@ static const struct sof_dev_desc arl_desc = {
 		[SOF_IPC_TYPE_4] = "intel/sof-ipc4-lib/arl",
 	},
 	.default_tplg_path = {
-		[SOF_IPC_TYPE_4] = "intel/sof-ace-tplg",
+		[SOF_IPC_TYPE_4] = "intel/sof-ipc4-tplg",
 	},
 	.default_fw_filename = {
 		[SOF_IPC_TYPE_4] = "sof-arl.ri",
@@ -107,7 +107,7 @@ static const struct sof_dev_desc arl_s_desc = {
 		[SOF_IPC_TYPE_4] = "intel/sof-ipc4-lib/arl-s",
 	},
 	.default_tplg_path = {
-		[SOF_IPC_TYPE_4] = "intel/sof-ace-tplg",
+		[SOF_IPC_TYPE_4] = "intel/sof-ipc4-tplg",
 	},
 	.default_fw_filename = {
 		[SOF_IPC_TYPE_4] = "sof-arl-s.ri",

sound/soc/sof/ipc4-topology.c

@@ -1752,11 +1752,9 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai
 		channel_count = params_channels(params);
 		sample_rate = params_rate(params);
 		bit_depth = params_width(params);
-		/*
-		 * Look for 32-bit blob first instead of 16-bit if copier
-		 * supports multiple formats
-		 */
-		if (bit_depth == 16 && !single_bitdepth) {
+
+		/* Prefer 32-bit blob if copier supports multiple formats */
+		if (bit_depth <= 16 && !single_bitdepth) {
 			dev_dbg(sdev->dev, "Looking for 32-bit blob first for DMIC\n");
 			format_change = true;
 			bit_depth = 32;
@@ -1799,10 +1797,18 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai
 		if (format_change) {
 			/*
 			 * The 32-bit blob was not found in NHLT table, try to
-			 * look for one based on the params
+			 * look for 16-bit for DMIC or based on the params for
+			 * SSP
 			 */
-			bit_depth = params_width(params);
-			format_change = false;
+			if (linktype == SOF_DAI_INTEL_DMIC) {
+				bit_depth = 16;
+				if (params_width(params) == 16)
+					format_change = false;
+			} else {
+				bit_depth = params_width(params);
+				format_change = false;
+			}
+
 			get_new_blob = true;
 		} else if (linktype == SOF_DAI_INTEL_DMIC && !single_bitdepth) {
 			/*
@@ -1837,7 +1843,7 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai
 	*len = cfg->size >> 2;
 	*dst = (u32 *)cfg->caps;
 
-	if (format_change) {
+	if (format_change || params_format(params) == SNDRV_PCM_FORMAT_FLOAT_LE) {
 		/*
 		 * Update the params to reflect that different blob was loaded
 		 * instead of the requested bit depth (16 -> 32 or 32 -> 16).
@@ -2280,8 +2286,19 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
 				ch_map >>= 4;
 			}
 
-			step = ch_count / blob->alh_cfg.device_count;
-			mask =  GENMASK(step - 1, 0);
+			if (swidget->id == snd_soc_dapm_dai_in && ch_count == out_ref_channels) {
+				/*
+				 * For playback DAI widgets where the channel number is equal to
+				 * the output reference channels, set the step = 0 to ensure all
+				 * the ch_mask is applied to all alh mappings.
+				 */
+				mask = ch_mask;
+				step = 0;
+			} else {
+				step = ch_count / blob->alh_cfg.device_count;
+				mask =  GENMASK(step - 1, 0);
+			}
+
 			/*
 			 * Set each gtw_cfg.node_id to blob->alh_cfg.mapping[]
 			 * for all widgets with the same stream name
@@ -2316,8 +2333,9 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
 				}
 
 				/*
-				 * Set the same channel mask for playback as the audio data is
-				 * duplicated for all speakers. For capture, split the channels
+				 * Set the same channel mask if the widget channel count is the same
+				 * as the FE channels for playback as the audio data is duplicated
+				 * for all speakers in this case. Otherwise, split the channels
 				 * among the aggregated DAIs. For example, with 4 channels on 2
 				 * aggregated DAIs, the channel_mask should be 0x3 and 0xc for the
 				 * two DAI's.
@@ -2326,10 +2344,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
 				 * the tables in soc_acpi files depending on the _ADR and devID
 				 * registers for each codec.
 				 */
-				if (w->id == snd_soc_dapm_dai_in)
-					blob->alh_cfg.mapping[i].channel_mask = ch_mask;
-				else
-					blob->alh_cfg.mapping[i].channel_mask = mask << (step * i);
+				blob->alh_cfg.mapping[i].channel_mask = mask << (step * i);
 
 				i++;
 			}

sound/soc/sof/topology.c

@@ -2106,8 +2106,8 @@ static int sof_route_load(struct snd_soc_component *scomp, int index,
 	/* source component */
 	source_swidget = snd_sof_find_swidget(scomp, (char *)route->source);
 	if (!source_swidget) {
-		dev_err(scomp->dev, "error: source %s not found\n",
-			route->source);
+		dev_err(scomp->dev, "source %s for sink %s is not found\n",
+			route->source, route->sink);
 		ret = -EINVAL;
 		goto err;
 	}
@@ -2125,8 +2125,8 @@ static int sof_route_load(struct snd_soc_component *scomp, int index,
 	/* sink component */
 	sink_swidget = snd_sof_find_swidget(scomp, (char *)route->sink);
 	if (!sink_swidget) {
-		dev_err(scomp->dev, "error: sink %s not found\n",
-			route->sink);
+		dev_err(scomp->dev, "sink %s for source %s is not found\n",
+			route->sink, route->source);
 		ret = -EINVAL;
 		goto err;
 	}
@@ -2506,12 +2506,28 @@ int snd_sof_load_topology(struct snd_soc_component *scomp, const char *file)
 	if (!tplg_files)
 		return -ENOMEM;
 
+	/* Try to use function topologies if possible */
 	if (!sof_pdata->disable_function_topology && !disable_function_topology &&
 	    sof_pdata->machine && sof_pdata->machine->get_function_tplg_files) {
+		/*
+		 * When the topology name contains 'dummy' word, it means that
+		 * there is no fallback option to monolithic topology in case
+		 * any of the function topologies might be missing.
+		 * In this case we should use best effort to form the card,
+		 * ignoring functionalities that we are missing a fragment for.
+		 *
+		 * Note: monolithic topologies also ignore these possibly
+		 * missing functions, so the functionality of the card would be
+		 * identical to the case if there would be a fallback monolithic
+		 * topology created for the configuration.
+		 */
+		bool no_fallback = strstr(file, "dummy");
+
 		tplg_cnt = sof_pdata->machine->get_function_tplg_files(scomp->card,
 								       sof_pdata->machine,
 								       tplg_filename_prefix,
-								       &tplg_files);
+								       &tplg_files,
+								       no_fallback);
 		if (tplg_cnt < 0) {
 			kfree(tplg_files);
 			return tplg_cnt;

sound/soc/tegra/tegra210_ahub.c

@@ -2077,7 +2077,7 @@ static const struct regmap_config tegra210_ahub_regmap_config = {
 	.val_bits		= 32,
 	.reg_stride		= 4,
 	.max_register		= TEGRA210_MAX_REGISTER_ADDR,
-	.cache_type		= REGCACHE_FLAT,
+	.cache_type		= REGCACHE_FLAT_S,
 };
 
 static const struct regmap_config tegra186_ahub_regmap_config = {
@@ -2085,7 +2085,7 @@ static const struct regmap_config tegra186_ahub_regmap_config = {
 	.val_bits		= 32,
 	.reg_stride		= 4,
 	.max_register		= TEGRA186_MAX_REGISTER_ADDR,
-	.cache_type		= REGCACHE_FLAT,
+	.cache_type		= REGCACHE_FLAT_S,
 };
 
 static const struct regmap_config tegra264_ahub_regmap_config = {
@@ -2094,7 +2094,7 @@ static const struct regmap_config tegra264_ahub_regmap_config = {
 	.reg_stride		= 4,
 	.writeable_reg		= tegra264_ahub_wr_reg,
 	.max_register		= TEGRA264_MAX_REGISTER_ADDR,
-	.cache_type		= REGCACHE_FLAT,
+	.cache_type		= REGCACHE_FLAT_S,
 };
 
 static const struct tegra_ahub_soc_data soc_data_tegra210 = {

sound/usb/mixer_quirks.c

@@ -3417,10 +3417,6 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer)
 #define RME_DIGIFACE_REGISTER(reg, mask) (((reg) << 16) | (mask))
 #define RME_DIGIFACE_INVERT BIT(31)
 
-/* Nonconst helpers */
-#define field_get(_mask, _reg) (((_reg) & (_mask)) >> (ffs(_mask) - 1))
-#define field_prep(_mask, _val) (((_val) << (ffs(_mask) - 1)) & (_mask))
-
 static int snd_rme_digiface_write_reg(struct snd_kcontrol *kcontrol, int item, u16 mask, u16 val)
 {
 	struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol);