Merge tag 'asoc-fix-v6.14-merge-window' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v6.14

A bunch of fixes that came in during the merge window, plus a few new
device IDs.  The i.MX changes are a little large since they add some new
quirk data as well as device IDs, and the audio graph card change for
picking the correct endpoint for links is large due to updating a number
of call sites.

22 files changed, 205 insertions, 68 deletions

diff --git a/Documentation/devicetree/bindings/arm/fsl.yaml b/Documentation/devicetree/bindings/arm/fsl.yaml
index 6e0dcf4307f1..df26b6b26e8c 100644
--- a/Documentation/devicetree/bindings/arm/fsl.yaml
+++ b/Documentation/devicetree/bindings/arm/fsl.yaml
@@ -1091,6 +1091,7 @@ properties:
               - dmo,imx8mp-data-modul-edm-sbc # i.MX8MP eDM SBC
               - emcraft,imx8mp-navqp      # i.MX8MP Emcraft Systems NavQ+ Kit
               - fsl,imx8mp-evk            # i.MX8MP EVK Board
+              - fsl,imx8mp-evk-revb4      # i.MX8MP EVK Rev B4 Board
               - gateworks,imx8mp-gw71xx-2x # i.MX8MP Gateworks Board
               - gateworks,imx8mp-gw72xx-2x # i.MX8MP Gateworks Board
               - gateworks,imx8mp-gw73xx-2x # i.MX8MP Gateworks Board
@@ -1262,6 +1263,7 @@ properties:
         items:
           - enum:
               - fsl,imx8qm-mek           # i.MX8QM MEK Board
+              - fsl,imx8qm-mek-revd      # i.MX8QM MEK Rev D Board
               - toradex,apalis-imx8      # Apalis iMX8 Modules
               - toradex,apalis-imx8-v1.1 # Apalis iMX8 V1.1 Modules
           - const: fsl,imx8qm
@@ -1290,6 +1292,7 @@ properties:
           - enum:
               - einfochips,imx8qxp-ai_ml  # i.MX8QXP AI_ML Board
               - fsl,imx8qxp-mek           # i.MX8QXP MEK Board
+              - fsl,imx8qxp-mek-wcpu      # i.MX8QXP MEK WCPU Board
           - const: fsl,imx8qxp
 
       - description: i.MX8DXL based Boards
diff --git a/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml
index 5aa00617291c..1f0e6787a746 100644
--- a/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml
+++ b/Documentation/devicetree/bindings/sound/ti,pcm1681.yaml
@@ -4,7 +4,7 @@
 $id: http://devicetree.org/schemas/sound/ti,pcm1681.yaml#
 $schema: http://devicetree.org/meta-schemas/core.yaml#
 
-title: Texas Instruments PCM1681 8-channel PWM Processor
+title: Texas Instruments PCM1681 8-channel Digital-to-Analog Converter
 
 maintainers:
   - Shenghao Ding <shenghao-ding@ti.com>
diff --git a/drivers/firmware/cirrus/Kconfig b/drivers/firmware/cirrus/Kconfig
index ee09269c63b5..0a883091259a 100644
--- a/drivers/firmware/cirrus/Kconfig
+++ b/drivers/firmware/cirrus/Kconfig
@@ -6,15 +6,13 @@ config FW_CS_DSP
 
 config FW_CS_DSP_KUNIT_TEST_UTILS
 	tristate
-	depends on KUNIT
-	select REGMAP
+	depends on KUNIT && REGMAP
 	select FW_CS_DSP
 
 config FW_CS_DSP_KUNIT_TEST
 	tristate "KUnit tests for Cirrus Logic cs_dsp" if !KUNIT_ALL_TESTS
-	depends on KUNIT
+	depends on KUNIT && REGMAP
 	default KUNIT_ALL_TESTS
-	select REGMAP
 	select FW_CS_DSP
 	select FW_CS_DSP_KUNIT_TEST_UTILS
 	help
diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c
index 1f59ee248771..89e99ed4275a 100644
--- a/sound/soc/amd/acp/acp-i2s.c
+++ b/sound/soc/amd/acp/acp-i2s.c
@@ -181,6 +181,7 @@ static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mas
 			break;
 		default:
 			dev_err(dev, "Unknown chip revision %d\n", chip->acp_rev);
+			spin_unlock_irq(&adata->acp_lock);
 			return -EINVAL;
 		}
 	}
diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c
index ecf57a6cb7c3..b16587d8f97a 100644
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -307,6 +307,34 @@ 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, "83L3"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "83N6"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "83Q2"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "83Q3"),
+		}
+	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
 			DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."),
 			DMI_MATCH(DMI_PRODUCT_NAME, "UM5302TA"),
 		}
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index ca4cc954efa8..eb97ac73ec06 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -2203,6 +2203,8 @@ static int da7213_i2c_probe(struct i2c_client *i2c)
 		return ret;
 	}
 
+	mutex_init(&da7213->ctrl_lock);
+
 	pm_runtime_set_autosuspend_delay(&i2c->dev, 100);
 	pm_runtime_use_autosuspend(&i2c->dev);
 	pm_runtime_set_active(&i2c->dev);
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index f508df01145b..e7bd561a8f40 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -101,7 +101,7 @@ static const struct snd_kcontrol_new es8316_snd_controls[] = {
 	SOC_DOUBLE_R_TLV("DAC Playback Volume", ES8316_DAC_VOLL,
 			 ES8316_DAC_VOLR, 0, 0xc0, 1, dac_vol_tlv),
 	SOC_SINGLE("DAC Soft Ramp Switch", ES8316_DAC_SET1, 4, 1, 1),
-	SOC_SINGLE("DAC Soft Ramp Rate", ES8316_DAC_SET1, 2, 4, 0),
+	SOC_SINGLE("DAC Soft Ramp Rate", ES8316_DAC_SET1, 2, 3, 0),
 	SOC_SINGLE("DAC Notch Filter Switch", ES8316_DAC_SET2, 6, 1, 0),
 	SOC_SINGLE("DAC Double Fs Switch", ES8316_DAC_SET2, 7, 1, 0),
 	SOC_SINGLE("DAC Stereo Enhancement", ES8316_DAC_SET3, 0, 7, 0),
diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c
index b06eead7e0f6..066d92b54312 100644
--- a/sound/soc/codecs/es8326.c
+++ b/sound/soc/codecs/es8326.c
@@ -911,7 +911,7 @@ static void es8326_jack_detect_handler(struct work_struct *work)
 			regmap_write(es8326->regmap, ES8326_INT_SOURCE,
 					(ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON));
 			regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x1f);
-			regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x08);
+			regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x0d);
 			queue_delayed_work(system_wq, &es8326->jack_detect_work,
 					msecs_to_jiffies(400));
 			es8326->hp = 1;
@@ -1023,7 +1023,7 @@ static void es8326_init(struct snd_soc_component *component)
 	struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component);
 
 	regmap_write(es8326->regmap, ES8326_RESET, 0x1f);
-	regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x0E);
+	regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x3E);
 	regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0);
 	usleep_range(10000, 15000);
 	regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xd9);
diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c
index 2b3c0f9e178c..9cb74962161a 100644
--- a/sound/soc/codecs/rt5514.c
+++ b/sound/soc/codecs/rt5514.c
@@ -1091,8 +1091,7 @@ static int rt5514_set_bias_level(struct snd_soc_component *component,
 static int rt5514_probe(struct snd_soc_component *component)
 {
 	struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component);
-	struct platform_device *pdev = container_of(component->dev,
-						   struct platform_device, dev);
+	struct platform_device *pdev = to_platform_device(component->dev);
 
 	rt5514->mclk = devm_clk_get_optional(component->dev, "mclk");
 	if (IS_ERR(rt5514->mclk))
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index e5fbf5305ea2..c4cf3cff58de 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -6,6 +6,7 @@ comment "Common SoC Audio options for Freescale CPUs:"
 config SND_SOC_FSL_ASRC
 	tristate "Asynchronous Sample Rate Converter (ASRC) module support"
 	depends on HAS_DMA
+	select DMA_SHARED_BUFFER
 	select REGMAP_MMIO
 	select SND_SOC_GENERIC_DMAENGINE_PCM
 	select SND_COMPRESS_ACCEL
diff --git a/sound/soc/fsl/fsl_asrc_m2m.c b/sound/soc/fsl/fsl_asrc_m2m.c
index 4906843e2a8f..f46881f71e43 100644
--- a/sound/soc/fsl/fsl_asrc_m2m.c
+++ b/sound/soc/fsl/fsl_asrc_m2m.c
@@ -183,7 +183,7 @@ static int asrc_dmaconfig(struct fsl_asrc_pair *pair,
 }
 
 /* main function of converter */
-static void asrc_m2m_device_run(struct fsl_asrc_pair *pair, struct snd_compr_task_runtime *task)
+static int asrc_m2m_device_run(struct fsl_asrc_pair *pair, struct snd_compr_task_runtime *task)
 {
 	struct fsl_asrc *asrc = pair->asrc;
 	struct device *dev = &asrc->pdev->dev;
@@ -193,7 +193,7 @@ static void asrc_m2m_device_run(struct fsl_asrc_pair *pair, struct snd_compr_tas
 	unsigned int out_dma_len;
 	unsigned int width;
 	u32 fifo_addr;
-	int ret;
+	int ret = 0;
 
 	/* set ratio mod */
 	if (asrc->m2m_set_ratio_mod) {
@@ -215,6 +215,7 @@ static void asrc_m2m_device_run(struct fsl_asrc_pair *pair, struct snd_compr_tas
 	    in_buf_len > ASRC_M2M_BUFFER_SIZE ||
 	    in_buf_len % (width * pair->channels / 8)) {
 		dev_err(dev, "out buffer size is error: [%d]\n", in_buf_len);
+		ret = -EINVAL;
 		goto end;
 	}
 
@@ -245,6 +246,7 @@ static void asrc_m2m_device_run(struct fsl_asrc_pair *pair, struct snd_compr_tas
 		}
 	} else if (out_dma_len > ASRC_M2M_BUFFER_SIZE) {
 		dev_err(dev, "cap buffer size error\n");
+		ret = -EINVAL;
 		goto end;
 	}
 
@@ -263,12 +265,14 @@ static void asrc_m2m_device_run(struct fsl_asrc_pair *pair, struct snd_compr_tas
 
 	if (!wait_for_completion_interruptible_timeout(&pair->complete[IN], 10 * HZ)) {
 		dev_err(dev, "out DMA task timeout\n");
+		ret = -ETIMEDOUT;
 		goto end;
 	}
 
 	if (out_dma_len > 0) {
 		if (!wait_for_completion_interruptible_timeout(&pair->complete[OUT], 10 * HZ)) {
 			dev_err(dev, "cap DMA task timeout\n");
+			ret = -ETIMEDOUT;
 			goto end;
 		}
 	}
@@ -278,7 +282,7 @@ static void asrc_m2m_device_run(struct fsl_asrc_pair *pair, struct snd_compr_tas
 	/* update payload length for capture */
 	task->output_size = out_dma_len;
 end:
-	return;
+	return ret;
 }
 
 static int fsl_asrc_m2m_comp_open(struct snd_compr_stream *stream)
@@ -525,9 +529,7 @@ static int fsl_asrc_m2m_comp_task_start(struct snd_compr_stream *stream,
 	struct snd_compr_runtime *runtime = stream->runtime;
 	struct fsl_asrc_pair *pair = runtime->private_data;
 
-	asrc_m2m_device_run(pair, task);
-
-	return 0;
+	return asrc_m2m_device_run(pair, task);
 }
 
 static int fsl_asrc_m2m_comp_task_stop(struct snd_compr_stream *stream,
@@ -633,7 +635,7 @@ int fsl_asrc_m2m_suspend(struct fsl_asrc *asrc)
 
 	for (i = 0; i < PAIR_CTX_NUM; i++) {
 		pair = asrc->pair[i];
-		if (!pair)
+		if (!pair || !pair->dma_buffer[IN].area || !pair->dma_buffer[OUT].area)
 			continue;
 		if (!completion_done(&pair->complete[IN])) {
 			if (pair->dma_chan[IN])
diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c
index c36b1a2ac949..ee94b256b770 100644
--- a/sound/soc/generic/audio-graph-card2.c
+++ b/sound/soc/generic/audio-graph-card2.c
@@ -648,23 +648,23 @@ multi_err:
 
 static int graph_parse_node_single(struct simple_util_priv *priv,
 				   enum graph_type gtype,
-				   struct device_node *port,
+				   struct device_node *ep,
 				   struct link_info *li, int is_cpu)
 {
-	struct device_node *ep __free(device_node) = of_graph_get_next_port_endpoint(port, NULL);
-
 	return __graph_parse_node(priv, gtype, ep, li, is_cpu, 0);
 }
 
 static int graph_parse_node(struct simple_util_priv *priv,
 			    enum graph_type gtype,
-			    struct device_node *port,
+			    struct device_node *ep,
 			    struct link_info *li, int is_cpu)
 {
+	struct device_node *port __free(device_node) = ep_to_port(ep);
+
 	if (graph_lnk_is_multi(port))
 		return graph_parse_node_multi(priv, gtype, port, li, is_cpu);
 	else
-		return graph_parse_node_single(priv, gtype, port, li, is_cpu);
+		return graph_parse_node_single(priv, gtype, ep, li, is_cpu);
 }
 
 static void graph_parse_daifmt(struct device_node *node, unsigned int *daifmt)
@@ -722,14 +722,15 @@ static unsigned int graph_parse_bitframe(struct device_node *ep)
 
 static void graph_link_init(struct simple_util_priv *priv,
 			    struct device_node *lnk,
-			    struct device_node *port_cpu,
-			    struct device_node *port_codec,
+			    struct device_node *ep_cpu,
+			    struct device_node *ep_codec,
 			    struct link_info *li,
 			    int is_cpu_node)
 {
 	struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link);
 	struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link);
-	struct device_node *ep_cpu, *ep_codec;
+	struct device_node *port_cpu = ep_to_port(ep_cpu);
+	struct device_node *port_codec = ep_to_port(ep_codec);
 	struct device_node *multi_cpu_port = NULL, *multi_codec_port = NULL;
 	struct snd_soc_dai_link_component *dlc;
 	unsigned int daifmt = 0;
@@ -739,25 +740,23 @@ static void graph_link_init(struct simple_util_priv *priv,
 	int multi_cpu_port_idx = 1, multi_codec_port_idx = 1;
 	int i;
 
-	of_node_get(port_cpu);
 	if (graph_lnk_is_multi(port_cpu)) {
 		multi_cpu_port = port_cpu;
 		ep_cpu = graph_get_next_multi_ep(&multi_cpu_port, multi_cpu_port_idx++);
 		of_node_put(port_cpu);
 		port_cpu = ep_to_port(ep_cpu);
 	} else {
-		ep_cpu = of_graph_get_next_port_endpoint(port_cpu, NULL);
+		of_node_get(ep_cpu);
 	}
 	struct device_node *ports_cpu __free(device_node) = port_to_ports(port_cpu);
 
-	of_node_get(port_codec);
 	if (graph_lnk_is_multi(port_codec)) {
 		multi_codec_port = port_codec;
 		ep_codec = graph_get_next_multi_ep(&multi_codec_port, multi_codec_port_idx++);
 		of_node_put(port_codec);
 		port_codec = ep_to_port(ep_codec);
 	} else {
-		ep_codec = of_graph_get_next_port_endpoint(port_codec, NULL);
+		of_node_get(ep_codec);
 	}
 	struct device_node *ports_codec __free(device_node) = port_to_ports(port_codec);
 
@@ -833,7 +832,7 @@ int audio_graph2_link_normal(struct simple_util_priv *priv,
 {
 	struct device_node *cpu_port = lnk;
 	struct device_node *cpu_ep	__free(device_node) = of_graph_get_next_port_endpoint(cpu_port, NULL);
-	struct device_node *codec_port	__free(device_node) = of_graph_get_remote_port(cpu_ep);
+	struct device_node *codec_ep	__free(device_node) = of_graph_get_remote_endpoint(cpu_ep);
 	int ret;
 
 	/*
@@ -841,18 +840,18 @@ int audio_graph2_link_normal(struct simple_util_priv *priv,
 	 * see
 	 *	__graph_parse_node() :: DAI Naming
 	 */
-	ret = graph_parse_node(priv, GRAPH_NORMAL, codec_port, li, 0);
+	ret = graph_parse_node(priv, GRAPH_NORMAL, codec_ep, li, 0);
 	if (ret < 0)
 		return ret;
 
 	/*
 	 * call CPU, and set DAI Name
 	 */
-	ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_port, li, 1);
+	ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_ep, li, 1);
 	if (ret < 0)
 		return ret;
 
-	graph_link_init(priv, lnk, cpu_port, codec_port, li, 1);
+	graph_link_init(priv, lnk, cpu_ep, codec_ep, li, 1);
 
 	return ret;
 }
@@ -864,15 +863,15 @@ int audio_graph2_link_dpcm(struct simple_util_priv *priv,
 {
 	struct device_node *ep	__free(device_node) = of_graph_get_next_port_endpoint(lnk, NULL);
 	struct device_node *rep	__free(device_node) = of_graph_get_remote_endpoint(ep);
-	struct device_node *cpu_port = NULL;
-	struct device_node *codec_port = NULL;
+	struct device_node *cpu_ep = NULL;
+	struct device_node *codec_ep = NULL;
 	struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link);
 	struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link);
 	int is_cpu = graph_util_is_ports0(lnk);
 	int ret;
 
 	if (is_cpu) {
-		cpu_port = of_graph_get_remote_port(ep); /* rport */
+		cpu_ep = rep;
 
 		/*
 		 * dpcm {
@@ -901,12 +900,12 @@ int audio_graph2_link_dpcm(struct simple_util_priv *priv,
 		dai_link->dynamic		= 1;
 		dai_link->dpcm_merged_format	= 1;
 
-		ret = graph_parse_node(priv, GRAPH_DPCM, cpu_port, li, 1);
+		ret = graph_parse_node(priv, GRAPH_DPCM, cpu_ep, li, 1);
 		if (ret)
-			goto err;
+			return ret;
 
 	} else {
-		codec_port = of_graph_get_remote_port(ep); /* rport */
+		codec_ep = rep;
 
 		/*
 		 * dpcm {
@@ -937,18 +936,15 @@ int audio_graph2_link_dpcm(struct simple_util_priv *priv,
 		dai_link->no_pcm		= 1;
 		dai_link->be_hw_params_fixup	= simple_util_be_hw_params_fixup;
 
-		ret = graph_parse_node(priv, GRAPH_DPCM, codec_port, li, 0);
+		ret = graph_parse_node(priv, GRAPH_DPCM, codec_ep, li, 0);
 		if (ret < 0)
-			goto err;
+			return ret;
 	}
 
 	graph_parse_convert(ep,  dai_props); /* at node of <dpcm> */
 	graph_parse_convert(rep, dai_props); /* at node of <CPU/Codec> */
 
-	graph_link_init(priv, lnk, cpu_port, codec_port, li, is_cpu);
-err:
-	of_node_put(cpu_port);
-	of_node_put(codec_port);
+	graph_link_init(priv, lnk, cpu_ep, codec_ep, li, is_cpu);
 
 	return ret;
 }
@@ -1013,26 +1009,26 @@ int audio_graph2_link_c2c(struct simple_util_priv *priv,
 	struct device_node *ep0 __free(device_node) = of_graph_get_next_port_endpoint(port0, NULL);
 	struct device_node *ep1 __free(device_node) = of_graph_get_next_port_endpoint(port1, NULL);
 
-	struct device_node *codec0_port __free(device_node) = of_graph_get_remote_port(ep0);
-	struct device_node *codec1_port __free(device_node) = of_graph_get_remote_port(ep1);
+	struct device_node *codec0_ep __free(device_node) = of_graph_get_remote_endpoint(ep0);
+	struct device_node *codec1_ep __free(device_node) = of_graph_get_remote_endpoint(ep1);
 
 	/*
 	 * call Codec first.
 	 * see
 	 *	__graph_parse_node() :: DAI Naming
 	 */
-	ret = graph_parse_node(priv, GRAPH_C2C, codec1_port, li, 0);
+	ret = graph_parse_node(priv, GRAPH_C2C, codec1_ep, li, 0);
 	if (ret < 0)
 		return ret;
 
 	/*
 	 * call CPU, and set DAI Name
 	 */
-	ret = graph_parse_node(priv, GRAPH_C2C, codec0_port, li, 1);
+	ret = graph_parse_node(priv, GRAPH_C2C, codec0_ep, li, 1);
 	if (ret < 0)
 		return ret;
 
-	graph_link_init(priv, lnk, codec0_port, codec1_port, li, 1);
+	graph_link_init(priv, lnk, codec0_ep, codec1_ep, li, 1);
 
 	return ret;
 }
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index 9caa4407c1ca..6446cda0f857 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -1132,7 +1132,22 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_SSP0_AIF2 |
 					BYT_RT5640_MCLK_EN),
 	},
-	{	/* Vexia Edu Atla 10 tablet */
+	{
+		/* Vexia Edu Atla 10 tablet 5V version */
+		.matches = {
+			/* Having all 3 of these not set is somewhat unique */
+			DMI_MATCH(DMI_SYS_VENDOR, "To be filled by O.E.M."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "To be filled by O.E.M."),
+			DMI_MATCH(DMI_BOARD_NAME, "To be filled by O.E.M."),
+			/* Above strings are too generic, also match on BIOS date */
+			DMI_MATCH(DMI_BIOS_DATE, "05/14/2015"),
+		},
+		.driver_data = (void *)(BYTCR_INPUT_DEFAULTS |
+					BYT_RT5640_JD_NOT_INV |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
+	},
+	{	/* Vexia Edu Atla 10 tablet 9V version */
 		.matches = {
 			DMI_MATCH(DMI_BOARD_VENDOR, "AMI Corporation"),
 			DMI_MATCH(DMI_BOARD_NAME, "Aptio CRB"),
diff --git a/sound/soc/renesas/Kconfig b/sound/soc/renesas/Kconfig
index 426632996a0a..cb01fb36355f 100644
--- a/sound/soc/renesas/Kconfig
+++ b/sound/soc/renesas/Kconfig
@@ -67,7 +67,7 @@ config SND_SH7760_AC97
 
 config SND_SIU_MIGOR
 	tristate "SIU sound support on Migo-R"
-	depends on SH_MIGOR && I2C
+	depends on SH_MIGOR && I2C && DMADEVICES
 	select SND_SOC_SH4_SIU
 	select SND_SOC_WM8978
 	help
diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c
index bd0dc586e24a..7f5fcaecee4b 100644
--- a/sound/soc/rockchip/rockchip_i2s_tdm.c
+++ b/sound/soc/rockchip/rockchip_i2s_tdm.c
@@ -22,7 +22,6 @@
 
 #define DRV_NAME "rockchip-i2s-tdm"
 
-#define DEFAULT_MCLK_FS				256
 #define CH_GRP_MAX				4  /* The max channel 8 / 2 */
 #define MULTIPLEX_CH_MAX			10
 
@@ -70,6 +69,8 @@ struct rk_i2s_tdm_dev {
 	bool has_playback;
 	bool has_capture;
 	struct snd_soc_dai_driver *dai;
+	unsigned int mclk_rx_freq;
+	unsigned int mclk_tx_freq;
 };
 
 static int to_ch_num(unsigned int val)
@@ -617,6 +618,27 @@ static int rockchip_i2s_trcm_mode(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+static int rockchip_i2s_tdm_set_sysclk(struct snd_soc_dai *cpu_dai, int stream,
+				       unsigned int freq, int dir)
+{
+	struct rk_i2s_tdm_dev *i2s_tdm = to_info(cpu_dai);
+
+	if (i2s_tdm->clk_trcm) {
+		i2s_tdm->mclk_tx_freq = freq;
+		i2s_tdm->mclk_rx_freq = freq;
+	} else {
+		if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+			i2s_tdm->mclk_tx_freq = freq;
+		else
+			i2s_tdm->mclk_rx_freq = freq;
+	}
+
+	dev_dbg(i2s_tdm->dev, "The target mclk_%s freq is: %d\n",
+		stream ? "rx" : "tx", freq);
+
+	return 0;
+}
+
 static int rockchip_i2s_tdm_hw_params(struct snd_pcm_substream *substream,
 				      struct snd_pcm_hw_params *params,
 				      struct snd_soc_dai *dai)
@@ -631,15 +653,19 @@ static int rockchip_i2s_tdm_hw_params(struct snd_pcm_substream *substream,
 
 		if (i2s_tdm->clk_trcm == TRCM_TX) {
 			mclk = i2s_tdm->mclk_tx;
+			mclk_rate = i2s_tdm->mclk_tx_freq;
 		} else if (i2s_tdm->clk_trcm == TRCM_RX) {
 			mclk = i2s_tdm->mclk_rx;
+			mclk_rate = i2s_tdm->mclk_rx_freq;
 		} else if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 			mclk = i2s_tdm->mclk_tx;
+			mclk_rate = i2s_tdm->mclk_tx_freq;
 		} else {
 			mclk = i2s_tdm->mclk_rx;
+			mclk_rate = i2s_tdm->mclk_rx_freq;
 		}
 
-		err = clk_set_rate(mclk, DEFAULT_MCLK_FS * params_rate(params));
+		err = clk_set_rate(mclk, mclk_rate);
 		if (err)
 			return err;
 
@@ -799,6 +825,7 @@ static const struct snd_soc_dai_ops rockchip_i2s_tdm_dai_ops = {
 	.hw_params = rockchip_i2s_tdm_hw_params,
 	.set_bclk_ratio	= rockchip_i2s_tdm_set_bclk_ratio,
 	.set_fmt = rockchip_i2s_tdm_set_fmt,
+	.set_sysclk = rockchip_i2s_tdm_set_sysclk,
 	.set_tdm_slot = rockchip_dai_tdm_slot,
 	.trigger = rockchip_i2s_tdm_trigger,
 };
diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index 0b85b29d1067..1e7bf00d7c46 100644
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -175,8 +175,7 @@ static int imx8_run(struct snd_sof_dev *sdev)
 
 static int imx8_probe(struct snd_sof_dev *sdev)
 {
-	struct platform_device *pdev =
-		container_of(sdev->dev, struct platform_device, dev);
+	struct platform_device *pdev = to_platform_device(sdev->dev);
 	struct device_node *np = pdev->dev.of_node;
 	struct device_node *res_node;
 	struct resource *mmio;
@@ -607,10 +606,31 @@ static struct snd_sof_of_mach sof_imx8_machs[] = {
 		.drv_name = "asoc-audio-graph-card2",
 	},
 	{
+		.compatible = "fsl,imx8qxp-mek-wcpu",
+		.sof_tplg_filename = "sof-imx8-wm8962.tplg",
+		.drv_name = "asoc-audio-graph-card2",
+	},
+	{
 		.compatible = "fsl,imx8qm-mek",
 		.sof_tplg_filename = "sof-imx8-wm8960.tplg",
 		.drv_name = "asoc-audio-graph-card2",
 	},
+	{
+		.compatible = "fsl,imx8qm-mek-revd",
+		.sof_tplg_filename = "sof-imx8-wm8962.tplg",
+		.drv_name = "asoc-audio-graph-card2",
+	},
+	{
+		.compatible = "fsl,imx8qxp-mek-bb",
+		.sof_tplg_filename = "sof-imx8-cs42888.tplg",
+		.drv_name = "asoc-audio-graph-card2",
+	},
+	{
+		.compatible = "fsl,imx8qm-mek-bb",
+		.sof_tplg_filename = "sof-imx8-cs42888.tplg",
+		.drv_name = "asoc-audio-graph-card2",
+	},
+
 	{}
 };
 
diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c
index ff42743efa79..3cabdebac558 100644
--- a/sound/soc/sof/imx/imx8m.c
+++ b/sound/soc/sof/imx/imx8m.c
@@ -144,8 +144,7 @@ static int imx8m_reset(struct snd_sof_dev *sdev)
 
 static int imx8m_probe(struct snd_sof_dev *sdev)
 {
-	struct platform_device *pdev =
-		container_of(sdev->dev, struct platform_device, dev);
+	struct platform_device *pdev = to_platform_device(sdev->dev);
 	struct device_node *np = pdev->dev.of_node;
 	struct device_node *res_node;
 	struct resource *mmio;
@@ -295,6 +294,17 @@ static struct snd_soc_dai_driver imx8m_dai[] = {
 	},
 },
 {
+	.name = "sai2",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 32,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 32,
+	},
+},
+{
 	.name = "sai3",
 	.playback = {
 		.channels_min = 1,
@@ -306,6 +316,39 @@ static struct snd_soc_dai_driver imx8m_dai[] = {
 	},
 },
 {
+	.name = "sai5",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 32,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 32,
+	},
+},
+{
+	.name = "sai6",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 32,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 32,
+	},
+},
+{
+	.name = "sai7",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 32,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 32,
+	},
+},
+{
 	.name = "micfil",
 	.capture = {
 		.channels_min = 1,
@@ -472,6 +515,11 @@ static const struct snd_sof_dsp_ops sof_imx8m_ops = {
 
 static struct snd_sof_of_mach sof_imx8mp_machs[] = {
 	{
+		.compatible = "fsl,imx8mp-evk-revb4",
+		.sof_tplg_filename = "sof-imx8mp-wm8962.tplg",
+		.drv_name = "asoc-audio-graph-card2",
+	},
+	{
 		.compatible = "fsl,imx8mp-evk",
 		.sof_tplg_filename = "sof-imx8mp-wm8960.tplg",
 		.drv_name = "asoc-audio-graph-card2",
diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c
index 6965791ab6ef..0704da27e69d 100644
--- a/sound/soc/sof/imx/imx8ulp.c
+++ b/sound/soc/sof/imx/imx8ulp.c
@@ -155,8 +155,7 @@ static int imx8ulp_reset(struct snd_sof_dev *sdev)
 
 static int imx8ulp_probe(struct snd_sof_dev *sdev)
 {
-	struct platform_device *pdev =
-		container_of(sdev->dev, struct platform_device, dev);
+	struct platform_device *pdev = to_platform_device(sdev->dev);
 	struct device_node *np = pdev->dev.of_node;
 	struct device_node *res_node;
 	struct resource *mmio;
diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c
index 5282c0071534..e1f0e38c2407 100644
--- a/sound/soc/sof/intel/bdw.c
+++ b/sound/soc/sof/intel/bdw.c
@@ -410,8 +410,7 @@ static int bdw_probe(struct snd_sof_dev *sdev)
 {
 	struct snd_sof_pdata *pdata = sdev->pdata;
 	const struct sof_dev_desc *desc = pdata->desc;
-	struct platform_device *pdev =
-		container_of(sdev->dev, struct platform_device, dev);
+	struct platform_device *pdev = to_platform_device(sdev->dev);
 	const struct sof_intel_dsp_desc *chip;
 	struct resource *mmio;
 	u32 base, size;
diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c
index 536d4c89d2f0..cae7dc0036c6 100644
--- a/sound/soc/sof/intel/byt.c
+++ b/sound/soc/sof/intel/byt.c
@@ -109,8 +109,7 @@ static int byt_acpi_probe(struct snd_sof_dev *sdev)
 {
 	struct snd_sof_pdata *pdata = sdev->pdata;
 	const struct sof_dev_desc *desc = pdata->desc;
-	struct platform_device *pdev =
-		container_of(sdev->dev, struct platform_device, dev);
+	struct platform_device *pdev = to_platform_device(sdev->dev);
 	const struct sof_intel_dsp_desc *chip;
 	struct resource *mmio;
 	u32 base, size;
diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c
index 9955dfa520ae..31437fdd4e92 100644
--- a/sound/soc/sof/mediatek/mt8186/mt8186.c
+++ b/sound/soc/sof/mediatek/mt8186/mt8186.c
@@ -238,7 +238,7 @@ static int mt8186_run(struct snd_sof_dev *sdev)
 
 static int mt8186_dsp_probe(struct snd_sof_dev *sdev)
 {
-	struct platform_device *pdev = container_of(sdev->dev, struct platform_device, dev);
+	struct platform_device *pdev = to_platform_device(sdev->dev);
 	struct adsp_priv *priv;
 	int ret;
 
diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c
index 6032b566c679..371563d7ce79 100644
--- a/sound/soc/sof/mediatek/mt8195/mt8195.c
+++ b/sound/soc/sof/mediatek/mt8195/mt8195.c
@@ -228,7 +228,7 @@ static int mt8195_run(struct snd_sof_dev *sdev)
 
 static int mt8195_dsp_probe(struct snd_sof_dev *sdev)
 {
-	struct platform_device *pdev = container_of(sdev->dev, struct platform_device, dev);
+	struct platform_device *pdev = to_platform_device(sdev->dev);
 	struct adsp_priv *priv;
 	int ret;
 
@@ -341,7 +341,7 @@ static int mt8195_dsp_shutdown(struct snd_sof_dev *sdev)
 
 static void mt8195_dsp_remove(struct snd_sof_dev *sdev)
 {
-	struct platform_device *pdev = container_of(sdev->dev, struct platform_device, dev);
+	struct platform_device *pdev = to_platform_device(sdev->dev);
 	struct adsp_priv *priv = sdev->pdata->hw_pdata;
 
 	platform_device_unregister(priv->ipc_dev);
@@ -351,7 +351,7 @@ static void mt8195_dsp_remove(struct snd_sof_dev *sdev)
 
 static int mt8195_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state)
 {
-	struct platform_device *pdev = container_of(sdev->dev, struct platform_device, dev);
+	struct platform_device *pdev = to_platform_device(sdev->dev);
 	int ret;
 	u32 reset_sw, dbg_pc;