Merge tag 'v6.14-rc4' into next

Sync up with the mainline.

26 files changed, 220 insertions, 91 deletions

diff --git a/sound/soc/sof/amd/acp-ipc.c b/sound/soc/sof/amd/acp-ipc.c
index 5f371d9263f3..22d4b807e1bb 100644
--- a/sound/soc/sof/amd/acp-ipc.c
+++ b/sound/soc/sof/amd/acp-ipc.c
@@ -165,8 +165,9 @@ irqreturn_t acp_sof_ipc_irq_thread(int irq, void *context)
 	int dsp_msg, dsp_ack;
 	unsigned int status;
 
-	if (sdev->first_boot && sdev->fw_state != SOF_FW_BOOT_COMPLETE) {
+	if (unlikely(sdev->first_boot && sdev->fw_state != SOF_FW_BOOT_COMPLETE)) {
 		acp_mailbox_read(sdev, sdev->dsp_box.offset, &status, sizeof(status));
+
 		if ((status & SOF_IPC_PANIC_MAGIC_MASK) == SOF_IPC_PANIC_MAGIC) {
 			snd_sof_dsp_panic(sdev, sdev->dsp_box.offset + sizeof(status),
 					  true);
@@ -188,13 +189,21 @@ irqreturn_t acp_sof_ipc_irq_thread(int irq, void *context)
 
 	dsp_ack = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_SCRATCH_REG_0 + dsp_ack_write);
 	if (dsp_ack) {
-		spin_lock_irq(&sdev->ipc_lock);
-		/* handle immediate reply from DSP core */
-		acp_dsp_ipc_get_reply(sdev);
-		snd_sof_ipc_reply(sdev, 0);
-		/* set the done bit */
-		acp_dsp_ipc_dsp_done(sdev);
-		spin_unlock_irq(&sdev->ipc_lock);
+		if (likely(sdev->fw_state == SOF_FW_BOOT_COMPLETE)) {
+			spin_lock_irq(&sdev->ipc_lock);
+
+			/* handle immediate reply from DSP core */
+			acp_dsp_ipc_get_reply(sdev);
+			snd_sof_ipc_reply(sdev, 0);
+			/* set the done bit */
+			acp_dsp_ipc_dsp_done(sdev);
+
+			spin_unlock_irq(&sdev->ipc_lock);
+		} else {
+			dev_dbg_ratelimited(sdev->dev, "IPC reply before FW_BOOT_COMPLETE: %#x\n",
+					    dsp_ack);
+		}
+
 		ipc_irq = true;
 	}
 
diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c
index 33648ff8b833..9e13c96528be 100644
--- a/sound/soc/sof/amd/acp.c
+++ b/sound/soc/sof/amd/acp.c
@@ -27,6 +27,7 @@ MODULE_PARM_DESC(enable_fw_debug, "Enable Firmware debug");
 static struct acp_quirk_entry quirk_valve_galileo = {
 	.signed_fw_image = true,
 	.skip_iram_dram_size_mod = true,
+	.post_fw_run_delay = true,
 };
 
 const struct dmi_system_id acp_sof_quirk_table[] = {
diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h
index 800594440f73..2a19d82d6200 100644
--- a/sound/soc/sof/amd/acp.h
+++ b/sound/soc/sof/amd/acp.h
@@ -220,6 +220,7 @@ struct sof_amd_acp_desc {
 struct acp_quirk_entry {
 	bool signed_fw_image;
 	bool skip_iram_dram_size_mod;
+	bool post_fw_run_delay;
 };
 
 /* Common device data struct for ACP devices */
diff --git a/sound/soc/sof/amd/pci-vangogh.c b/sound/soc/sof/amd/pci-vangogh.c
index 53f64d6bc91b..28f2d4050a67 100644
--- a/sound/soc/sof/amd/pci-vangogh.c
+++ b/sound/soc/sof/amd/pci-vangogh.c
@@ -13,11 +13,9 @@
 
 #include <linux/module.h>
 #include <linux/pci.h>
-#include <linux/platform_device.h>
 #include <sound/sof.h>
 #include <sound/soc-acpi.h>
 
-#include "../ops.h"
 #include "../sof-pci-dev.h"
 #include "../../amd/mach-config.h"
 #include "acp.h"
diff --git a/sound/soc/sof/amd/vangogh.c b/sound/soc/sof/amd/vangogh.c
index 8e2672106ac6..6ed5f9aaa414 100644
--- a/sound/soc/sof/amd/vangogh.c
+++ b/sound/soc/sof/amd/vangogh.c
@@ -11,13 +11,10 @@
  * Hardware interface for Audio DSP on Vangogh platform
  */
 
-#include <linux/platform_device.h>
+#include <linux/delay.h>
 #include <linux/module.h>
 
-#include "../ops.h"
-#include "../sof-audio.h"
 #include "acp.h"
-#include "acp-dsp-offset.h"
 
 #define I2S_HS_INSTANCE		0
 #define I2S_BT_INSTANCE		1
@@ -136,6 +133,20 @@ static struct snd_soc_dai_driver vangogh_sof_dai[] = {
 	},
 };
 
+static int sof_vangogh_post_fw_run_delay(struct snd_sof_dev *sdev)
+{
+	/*
+	 * Resuming from suspend in some cases my cause the DSP firmware
+	 * to enter an unrecoverable faulty state.  Delaying a bit any host
+	 * to DSP transmission right after firmware boot completion seems
+	 * to resolve the issue.
+	 */
+	if (!sdev->first_boot)
+		usleep_range(100, 150);
+
+	return 0;
+}
+
 /* Vangogh ops */
 struct snd_sof_dsp_ops sof_vangogh_ops;
 EXPORT_SYMBOL_NS(sof_vangogh_ops, "SND_SOC_SOF_AMD_COMMON");
@@ -157,6 +168,9 @@ int sof_vangogh_ops_init(struct snd_sof_dev *sdev)
 
 		if (quirks->signed_fw_image)
 			sof_vangogh_ops.load_firmware = acp_sof_load_signed_firmware;
+
+		if (quirks->post_fw_run_delay)
+			sof_vangogh_ops.post_fw_run = sof_vangogh_post_fw_run_delay;
 	}
 
 	return 0;
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/atom.c b/sound/soc/sof/intel/atom.c
index 30e981c558c6..0d364bcdcfa9 100644
--- a/sound/soc/sof/intel/atom.c
+++ b/sound/soc/sof/intel/atom.c
@@ -78,20 +78,20 @@ void atom_dump(struct snd_sof_dev *sdev, u32 flags)
 	imrd = snd_sof_dsp_read64(sdev, DSP_BAR, SHIM_IMRD);
 	dev_err(sdev->dev,
 		"error: ipc host -> DSP: pending %s complete %s raw 0x%llx\n",
-		(panic & SHIM_IPCX_BUSY) ? "yes" : "no",
-		(panic & SHIM_IPCX_DONE) ? "yes" : "no", panic);
+		str_yes_no(panic & SHIM_IPCX_BUSY),
+		str_yes_no(panic & SHIM_IPCX_DONE), panic);
 	dev_err(sdev->dev,
 		"error: mask host: pending %s complete %s raw 0x%llx\n",
-		(imrx & SHIM_IMRX_BUSY) ? "yes" : "no",
-		(imrx & SHIM_IMRX_DONE) ? "yes" : "no", imrx);
+		str_yes_no(imrx & SHIM_IMRX_BUSY),
+		str_yes_no(imrx & SHIM_IMRX_DONE), imrx);
 	dev_err(sdev->dev,
 		"error: ipc DSP -> host: pending %s complete %s raw 0x%llx\n",
-		(status & SHIM_IPCD_BUSY) ? "yes" : "no",
-		(status & SHIM_IPCD_DONE) ? "yes" : "no", status);
+		str_yes_no(status & SHIM_IPCD_BUSY),
+		str_yes_no(status & SHIM_IPCD_DONE), status);
 	dev_err(sdev->dev,
 		"error: mask DSP: pending %s complete %s raw 0x%llx\n",
-		(imrd & SHIM_IMRD_BUSY) ? "yes" : "no",
-		(imrd & SHIM_IMRD_DONE) ? "yes" : "no", imrd);
+		str_yes_no(imrd & SHIM_IMRD_BUSY),
+		str_yes_no(imrd & SHIM_IMRD_DONE), imrd);
 
 }
 EXPORT_SYMBOL_NS(atom_dump, "SND_SOC_SOF_INTEL_ATOM_HIFI_EP");
diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c
index c4d92f3508b6..e1f0e38c2407 100644
--- a/sound/soc/sof/intel/bdw.c
+++ b/sound/soc/sof/intel/bdw.c
@@ -266,20 +266,20 @@ static void bdw_dump(struct snd_sof_dev *sdev, u32 flags)
 	imrd = snd_sof_dsp_read(sdev, BDW_DSP_BAR, SHIM_IMRD);
 	dev_err(sdev->dev,
 		"error: ipc host -> DSP: pending %s complete %s raw 0x%8.8x\n",
-		(panic & SHIM_IPCX_BUSY) ? "yes" : "no",
-		(panic & SHIM_IPCX_DONE) ? "yes" : "no", panic);
+		str_yes_no(panic & SHIM_IPCX_BUSY),
+		str_yes_no(panic & SHIM_IPCX_DONE), panic);
 	dev_err(sdev->dev,
 		"error: mask host: pending %s complete %s raw 0x%8.8x\n",
-		(imrx & SHIM_IMRX_BUSY) ? "yes" : "no",
-		(imrx & SHIM_IMRX_DONE) ? "yes" : "no", imrx);
+		str_yes_no(imrx & SHIM_IMRX_BUSY),
+		str_yes_no(imrx & SHIM_IMRX_DONE), imrx);
 	dev_err(sdev->dev,
 		"error: ipc DSP -> host: pending %s complete %s raw 0x%8.8x\n",
-		(status & SHIM_IPCD_BUSY) ? "yes" : "no",
-		(status & SHIM_IPCD_DONE) ? "yes" : "no", status);
+		str_yes_no(status & SHIM_IPCD_BUSY),
+		str_yes_no(status & SHIM_IPCD_DONE), status);
 	dev_err(sdev->dev,
 		"error: mask DSP: pending %s complete %s raw 0x%8.8x\n",
-		(imrd & SHIM_IMRD_BUSY) ? "yes" : "no",
-		(imrd & SHIM_IMRD_DONE) ? "yes" : "no", imrd);
+		str_yes_no(imrd & SHIM_IMRD_BUSY),
+		str_yes_no(imrd & SHIM_IMRD_DONE), imrd);
 }
 
 /*
@@ -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/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c
index 568f3dfe822f..2f9925830d1d 100644
--- a/sound/soc/sof/intel/hda-codec.c
+++ b/sound/soc/sof/intel/hda-codec.c
@@ -454,6 +454,7 @@ int hda_codec_i915_exit(struct snd_sof_dev *sdev)
 }
 EXPORT_SYMBOL_NS_GPL(hda_codec_i915_exit, "SND_SOC_SOF_HDA_AUDIO_CODEC_I915");
 
+MODULE_SOFTDEP("pre: snd-hda-codec-hdmi");
 #endif
 
 MODULE_LICENSE("Dual BSD/GPL");
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index c13f89b7065e..da12aabc1bb8 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/sound/soc/sof/intel/hda-dai.c
@@ -103,8 +103,10 @@ hda_dai_get_ops(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai
 	return sdai->platform_private;
 }
 
-int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream,
-			 struct snd_soc_dai *cpu_dai)
+static int
+hda_link_dma_cleanup(struct snd_pcm_substream *substream,
+		     struct hdac_ext_stream *hext_stream,
+		     struct snd_soc_dai *cpu_dai, bool release)
 {
 	const struct hda_dai_widget_dma_ops *ops = hda_dai_get_ops(substream, cpu_dai);
 	struct sof_intel_hda_stream *hda_stream;
@@ -128,6 +130,17 @@ int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_st
 		snd_hdac_ext_bus_link_clear_stream_id(hlink, stream_tag);
 	}
 
+	if (!release) {
+		/*
+		 * Force stream reconfiguration without releasing the channel on
+		 * subsequent stream restart (without free), including LinkDMA
+		 * reset.
+		 * The stream is released via hda_dai_hw_free()
+		 */
+		hext_stream->link_prepared = 0;
+		return 0;
+	}
+
 	if (ops->release_hext_stream)
 		ops->release_hext_stream(sdev, cpu_dai, substream);
 
@@ -211,7 +224,7 @@ static int __maybe_unused hda_dai_hw_free(struct snd_pcm_substream *substream,
 	if (!hext_stream)
 		return 0;
 
-	return hda_link_dma_cleanup(substream, hext_stream, cpu_dai);
+	return hda_link_dma_cleanup(substream, hext_stream, cpu_dai, true);
 }
 
 static int __maybe_unused hda_dai_hw_params_data(struct snd_pcm_substream *substream,
@@ -304,7 +317,8 @@ static int __maybe_unused hda_dai_trigger(struct snd_pcm_substream *substream, i
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
-		ret = hda_link_dma_cleanup(substream, hext_stream, dai);
+		ret = hda_link_dma_cleanup(substream, hext_stream, dai,
+					   cmd == SNDRV_PCM_TRIGGER_STOP ? false : true);
 		if (ret < 0) {
 			dev_err(sdev->dev, "%s: failed to clean up link DMA\n", __func__);
 			return ret;
@@ -489,6 +503,12 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
 	int ret;
 	int i;
 
+	if (!w) {
+		dev_err(cpu_dai->dev, "%s widget not found, check amp link num in the topology\n",
+			cpu_dai->name);
+		return -EINVAL;
+	}
+
 	ops = hda_dai_get_ops(substream, cpu_dai);
 	if (!ops) {
 		dev_err(cpu_dai->dev, "DAI widget ops not set\n");
@@ -568,6 +588,12 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
 	 */
 	for_each_rtd_cpu_dais(rtd, i, dai) {
 		w = snd_soc_dai_get_widget(dai, substream->stream);
+		if (!w) {
+			dev_err(cpu_dai->dev,
+				"%s widget not found, check amp link num in the topology\n",
+				dai->name);
+			return -EINVAL;
+		}
 		ipc4_copier = widget_to_copier(w);
 		memcpy(&ipc4_copier->dma_config_tlv[cpu_dai_id], dma_config_tlv,
 		       sizeof(*dma_config_tlv));
@@ -660,8 +686,7 @@ static int hda_dai_suspend(struct hdac_bus *bus)
 			}
 
 			ret = hda_link_dma_cleanup(hext_stream->link_substream,
-						   hext_stream,
-						   cpu_dai);
+						   hext_stream, cpu_dai, true);
 			if (ret < 0)
 				return ret;
 		}
diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c
index 5b5e484f9acf..1dd8d2092c3b 100644
--- a/sound/soc/sof/intel/hda-pcm.c
+++ b/sound/soc/sof/intel/hda-pcm.c
@@ -37,6 +37,11 @@ static bool hda_disable_rewinds;
 module_param_named(disable_rewinds, hda_disable_rewinds, bool, 0444);
 MODULE_PARM_DESC(disable_rewinds, "SOF HDA disable rewinds");
 
+static int hda_force_pause_support = -1;
+module_param_named(force_pause_support, hda_force_pause_support, int, 0444);
+MODULE_PARM_DESC(force_pause_support,
+		 "Pause support: -1: Use default, 0: Disable, 1: Enable (default -1)");
+
 u32 hda_dsp_get_mult_div(struct snd_sof_dev *sdev, int rate)
 {
 	switch (rate) {
@@ -240,6 +245,16 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev,
 	if (hda_always_enable_dmi_l1 && direction == SNDRV_PCM_STREAM_CAPTURE)
 		runtime->hw.info &= ~SNDRV_PCM_INFO_PAUSE;
 
+	/*
+	 * Do not advertise the PAUSE support if it is forced to be disabled via
+	 * module parameter or if the pause_supported is false for the PCM
+	 * device
+	 */
+	if (hda_force_pause_support == 0 ||
+	    (hda_force_pause_support == -1 &&
+	     !spcm->stream[substream->stream].pause_supported))
+		runtime->hw.info &= ~SNDRV_PCM_INFO_PAUSE;
+
 	if (hda_always_enable_dmi_l1 ||
 	    direction == SNDRV_PCM_STREAM_PLAYBACK ||
 	    spcm->stream[substream->stream].d0i3_compatible)
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index f991785f727e..be689f6e10c8 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -63,6 +63,11 @@ static int sdw_params_stream(struct device *dev,
 	struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(d, params_data->substream->stream);
 	struct snd_sof_dai_config_data data = { 0 };
 
+	if (!w) {
+		dev_err(dev, "%s widget not found, check amp link num in the topology\n",
+			d->name);
+		return -EINVAL;
+	}
 	data.dai_index = (params_data->link_id << 8) | d->id;
 	data.dai_data = params_data->alh_stream_id;
 	data.dai_node_id = data.dai_data;
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 22bd9c3c8216..ee4ccc1a5490 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -1038,8 +1038,6 @@ const struct hda_dai_widget_dma_ops *
 hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget);
 int hda_dai_config(struct snd_soc_dapm_widget *w, unsigned int flags,
 		   struct snd_sof_dai_config_data *data);
-int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream,
-			 struct snd_soc_dai *cpu_dai);
 
 static inline struct snd_sof_dev *widget_to_sdev(struct snd_soc_dapm_widget *w)
 {
diff --git a/sound/soc/sof/intel/pci-ptl.c b/sound/soc/sof/intel/pci-ptl.c
index 0aacdfac9fb4..c4fb6a2441b7 100644
--- a/sound/soc/sof/intel/pci-ptl.c
+++ b/sound/soc/sof/intel/pci-ptl.c
@@ -50,6 +50,7 @@ static const struct sof_dev_desc ptl_desc = {
 /* PCI IDs */
 static const struct pci_device_id sof_pci_ids[] = {
 	{ PCI_DEVICE_DATA(INTEL, HDA_PTL, &ptl_desc) }, /* PTL */
+	{ PCI_DEVICE_DATA(INTEL, HDA_PTL_H, &ptl_desc) }, /* PTL-H */
 	{ 0, }
 };
 MODULE_DEVICE_TABLE(pci, sof_pci_ids);
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index b55eb977e443..6d5cda813e48 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -765,10 +765,16 @@ static int sof_ipc4_widget_setup_comp_dai(struct snd_sof_widget *swidget)
 		}
 
 		list_for_each_entry(w, &sdev->widget_list, list) {
-			if (w->widget->sname &&
+			struct snd_sof_dai *alh_dai;
+
+			if (!WIDGET_IS_DAI(w->id) || !w->widget->sname ||
 			    strcmp(w->widget->sname, swidget->widget->sname))
 				continue;
 
+			alh_dai = w->private;
+			if (alh_dai->type != SOF_DAI_INTEL_ALH)
+				continue;
+
 			blob->alh_cfg.device_count++;
 		}
 
@@ -2061,11 +2067,13 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget,
 			list_for_each_entry(w, &sdev->widget_list, list) {
 				u32 node_type;
 
-				if (w->widget->sname &&
+				if (!WIDGET_IS_DAI(w->id) || !w->widget->sname ||
 				    strcmp(w->widget->sname, swidget->widget->sname))
 					continue;
 
 				dai = w->private;
+				if (dai->type != SOF_DAI_INTEL_ALH)
+					continue;
 				alh_copier = (struct sof_ipc4_copier *)dai->private;
 				alh_data = &alh_copier->data;
 				node_type = SOF_IPC4_GET_NODE_TYPE(alh_data->gtw_cfg.node_id);
@@ -2827,7 +2835,7 @@ static int sof_ipc4_widget_setup(struct snd_sof_dev *sdev, struct snd_sof_widget
 		msg->primary |= SOF_IPC4_MOD_INSTANCE(swidget->instance_id);
 
 		msg->extension &= ~SOF_IPC4_MOD_EXT_PARAM_SIZE_MASK;
-		msg->extension |= ipc_size >> 2;
+		msg->extension |= SOF_IPC4_MOD_EXT_PARAM_SIZE(ipc_size >> 2);
 
 		msg->extension &= ~SOF_IPC4_MOD_EXT_PPL_ID_MASK;
 		msg->extension |= SOF_IPC4_MOD_EXT_PPL_ID(pipe_widget->instance_id);
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;
 
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index 35a7462d8b69..c5c6353f18ce 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -511,6 +511,8 @@ static int sof_pcm_close(struct snd_soc_component *component,
 		 */
 	}
 
+	spcm->stream[substream->stream].substream = NULL;
+
 	return 0;
 }
 
diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h
index 01b819dd8498..62f3c11a9216 100644
--- a/sound/soc/sof/sof-audio.h
+++ b/sound/soc/sof/sof-audio.h
@@ -332,6 +332,7 @@ struct snd_sof_pcm_stream {
 	struct work_struct period_elapsed_work;
 	struct snd_soc_dapm_widget_list *list; /* list of connected DAPM widgets */
 	bool d0i3_compatible; /* DSP can be in D0I3 when this pcm is opened */
+	bool pause_supported; /* PCM device supports PAUSE operation */
 	unsigned int dsp_max_burst_size_in_ms; /* The maximum size of the host DMA burst in ms */
 	/*
 	 * flag to indicate that the DSP pipelines should be kept
diff --git a/sound/soc/sof/sof-client-ipc-flood-test.c b/sound/soc/sof/sof-client-ipc-flood-test.c
index b35c98896968..11b6f7da2882 100644
--- a/sound/soc/sof/sof-client-ipc-flood-test.c
+++ b/sound/soc/sof/sof-client-ipc-flood-test.c
@@ -158,7 +158,6 @@ static ssize_t sof_ipc_flood_dfs_write(struct file *file, const char __user *buf
 	unsigned long ipc_duration_ms = 0;
 	bool flood_duration_test = false;
 	unsigned long ipc_count = 0;
-	struct dentry *dentry;
 	int err;
 	char *string;
 	int ret;
@@ -182,14 +181,7 @@ static ssize_t sof_ipc_flood_dfs_write(struct file *file, const char __user *buf
 	 * ipc_duration_ms test floods the DSP for the time specified
 	 * in the debugfs entry.
 	 */
-	dentry = file->f_path.dentry;
-	if (strcmp(dentry->d_name.name, DEBUGFS_IPC_FLOOD_COUNT) &&
-	    strcmp(dentry->d_name.name, DEBUGFS_IPC_FLOOD_DURATION)) {
-		ret = -EINVAL;
-		goto out;
-	}
-
-	if (!strcmp(dentry->d_name.name, DEBUGFS_IPC_FLOOD_DURATION))
+	if (debugfs_get_aux_num(file))
 		flood_duration_test = true;
 
 	/* test completion criterion */
@@ -252,22 +244,15 @@ static ssize_t sof_ipc_flood_dfs_read(struct file *file, char __user *buffer,
 	struct sof_ipc_flood_priv *priv = cdev->data;
 	size_t size_ret;
 
-	struct dentry *dentry;
+	if (*ppos)
+		return 0;
 
-	dentry = file->f_path.dentry;
-	if (!strcmp(dentry->d_name.name, DEBUGFS_IPC_FLOOD_COUNT) ||
-	    !strcmp(dentry->d_name.name, DEBUGFS_IPC_FLOOD_DURATION)) {
-		if (*ppos)
-			return 0;
+	count = min_t(size_t, count, strlen(priv->buf));
+	size_ret = copy_to_user(buffer, priv->buf, count);
+	if (size_ret)
+		return -EFAULT;
 
-		count = min_t(size_t, count, strlen(priv->buf));
-		size_ret = copy_to_user(buffer, priv->buf, count);
-		if (size_ret)
-			return -EFAULT;
-
-		*ppos += count;
-		return count;
-	}
+	*ppos += count;
 	return count;
 }
 
@@ -320,12 +305,12 @@ static int sof_ipc_flood_probe(struct auxiliary_device *auxdev,
 	priv->dfs_root = debugfs_create_dir(dev_name(dev), debugfs_root);
 	if (!IS_ERR_OR_NULL(priv->dfs_root)) {
 		/* create read-write ipc_flood_count debugfs entry */
-		debugfs_create_file(DEBUGFS_IPC_FLOOD_COUNT, 0644, priv->dfs_root,
-				    cdev, &sof_ipc_flood_fops);
+		debugfs_create_file_aux_num(DEBUGFS_IPC_FLOOD_COUNT, 0644,
+				    priv->dfs_root, cdev, 0, &sof_ipc_flood_fops);
 
 		/* create read-write ipc_flood_duration_ms debugfs entry */
-		debugfs_create_file(DEBUGFS_IPC_FLOOD_DURATION, 0644,
-				    priv->dfs_root, cdev, &sof_ipc_flood_fops);
+		debugfs_create_file_aux_num(DEBUGFS_IPC_FLOOD_DURATION, 0644,
+				    priv->dfs_root, cdev, 1, &sof_ipc_flood_fops);
 
 		if (auxdev->id == 0) {
 			/*
diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h
index 843be3b6415d..abbb5ee7e08c 100644
--- a/sound/soc/sof/sof-priv.h
+++ b/sound/soc/sof/sof-priv.h
@@ -76,14 +76,6 @@ bool sof_debug_check_flag(int mask);
 #define SOF_IPC_DSP_REPLY		0
 #define SOF_IPC_HOST_REPLY		1
 
-/* convenience constructor for DAI driver streams */
-#define SOF_DAI_STREAM(sname, scmin, scmax, srates, sfmt) \
-	{.stream_name = sname, .channels_min = scmin, .channels_max = scmax, \
-	 .rates = srates, .formats = sfmt}
-
-#define SOF_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \
-	SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_FLOAT)
-
 /* So far the primary core on all DSPs has ID 0 */
 #define SOF_DSP_PRIMARY_CORE 0
 
diff --git a/sound/soc/sof/stream-ipc.c b/sound/soc/sof/stream-ipc.c
index 794c7bbccbaf..8262443ac89a 100644
--- a/sound/soc/sof/stream-ipc.c
+++ b/sound/soc/sof/stream-ipc.c
@@ -43,7 +43,7 @@ int sof_ipc_msg_data(struct snd_sof_dev *sdev,
 				return -ESTRPIPE;
 
 			posn_offset = stream->posn_offset;
-		} else {
+		} else if (sps->cstream) {
 
 			struct sof_compr_stream *sstream = sps->cstream->runtime->private_data;
 
@@ -51,6 +51,10 @@ int sof_ipc_msg_data(struct snd_sof_dev *sdev,
 				return -ESTRPIPE;
 
 			posn_offset = sstream->posn_offset;
+
+		} else {
+			dev_err(sdev->dev, "%s: No stream opened\n", __func__);
+			return -EINVAL;
 		}
 
 		snd_sof_dsp_mailbox_read(sdev, posn_offset, p, sz);
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index b3fca5fd87d6..688cc7ac1714 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -407,6 +407,10 @@ static const struct sof_topology_token stream_tokens[] = {
 		offsetof(struct snd_sof_pcm, stream[0].d0i3_compatible)},
 	{SOF_TKN_STREAM_CAPTURE_COMPATIBLE_D0I3, SND_SOC_TPLG_TUPLE_TYPE_BOOL, get_token_u16,
 		offsetof(struct snd_sof_pcm, stream[1].d0i3_compatible)},
+	{SOF_TKN_STREAM_PLAYBACK_PAUSE_SUPPORTED, SND_SOC_TPLG_TUPLE_TYPE_BOOL, get_token_u16,
+		offsetof(struct snd_sof_pcm, stream[0].pause_supported)},
+	{SOF_TKN_STREAM_CAPTURE_PAUSE_SUPPORTED, SND_SOC_TPLG_TUPLE_TYPE_BOOL, get_token_u16,
+		offsetof(struct snd_sof_pcm, stream[1].pause_supported)},
 };
 
 /* Leds */