Merge branch 'asoc-5.4' into asoc-5.5

20 files changed, 243 insertions, 69 deletions

diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index e609abcf3220..eb709d528259 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -901,16 +901,20 @@ static void max98373_slot_config(struct i2c_client *i2c,
 		max98373->i_slot = value & 0xF;
 	else
 		max98373->i_slot = 1;
-
-	max98373->reset_gpio = of_get_named_gpio(dev->of_node,
+	if (dev->of_node) {
+		max98373->reset_gpio = of_get_named_gpio(dev->of_node,
 						"maxim,reset-gpio", 0);
-	if (!gpio_is_valid(max98373->reset_gpio)) {
-		dev_err(dev, "Looking up %s property in node %s failed %d\n",
-			"maxim,reset-gpio", dev->of_node->full_name,
-			max98373->reset_gpio);
+		if (!gpio_is_valid(max98373->reset_gpio)) {
+			dev_err(dev, "Looking up %s property in node %s failed %d\n",
+				"maxim,reset-gpio", dev->of_node->full_name,
+				max98373->reset_gpio);
+		} else {
+			dev_dbg(dev, "maxim,reset-gpio=%d",
+				max98373->reset_gpio);
+		}
 	} else {
-		dev_dbg(dev, "maxim,reset-gpio=%d",
-			max98373->reset_gpio);
+		/* this makes reset_gpio as invalid */
+		max98373->reset_gpio = -1;
 	}
 
 	if (!device_property_read_u32(dev, "maxim,spkfb-slot-no", &value))
diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c
index 762595de956c..c506c9305043 100644
--- a/sound/soc/codecs/rt5651.c
+++ b/sound/soc/codecs/rt5651.c
@@ -1770,6 +1770,9 @@ static int rt5651_detect_headset(struct snd_soc_component *component)
 
 static bool rt5651_support_button_press(struct rt5651_priv *rt5651)
 {
+	if (!rt5651->hp_jack)
+		return false;
+
 	/* Button press support only works with internal jack-detection */
 	return (rt5651->hp_jack->status & SND_JACK_MICROPHONE) &&
 		rt5651->gpiod_hp_det == NULL;
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index 1ef470700ed5..c50b75ce82e0 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -995,6 +995,16 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component,
 {
 	struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component);
 
+	rt5682->hs_jack = hs_jack;
+
+	if (!hs_jack) {
+		regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2,
+				   RT5682_JD1_EN_MASK, RT5682_JD1_DIS);
+		regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL,
+				   RT5682_POW_JDH | RT5682_POW_JDL, 0);
+		return 0;
+	}
+
 	switch (rt5682->pdata.jd_src) {
 	case RT5682_JD1:
 		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_2,
@@ -1032,8 +1042,6 @@ static int rt5682_set_jack_detect(struct snd_soc_component *component,
 		break;
 	}
 
-	rt5682->hs_jack = hs_jack;
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index b633093afb9d..15ce64a48a87 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -533,13 +533,10 @@ static SOC_ENUM_SINGLE_DECL(dac_osr,
 static SOC_ENUM_SINGLE_DECL(adc_osr,
 			    WM8994_OVERSAMPLING, 1, osr_text);
 
-static const struct snd_kcontrol_new wm8994_snd_controls[] = {
+static const struct snd_kcontrol_new wm8994_common_snd_controls[] = {
 SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME,
 		 WM8994_AIF1_ADC1_RIGHT_VOLUME,
 		 1, 119, 0, digital_tlv),
-SOC_DOUBLE_R_TLV("AIF1ADC2 Volume", WM8994_AIF1_ADC2_LEFT_VOLUME,
-		 WM8994_AIF1_ADC2_RIGHT_VOLUME,
-		 1, 119, 0, digital_tlv),
 SOC_DOUBLE_R_TLV("AIF2ADC Volume", WM8994_AIF2_ADC_LEFT_VOLUME,
 		 WM8994_AIF2_ADC_RIGHT_VOLUME,
 		 1, 119, 0, digital_tlv),
@@ -556,8 +553,6 @@ SOC_ENUM("AIF2DACR Source", aif2dacr_src),
 
 SOC_DOUBLE_R_TLV("AIF1DAC1 Volume", WM8994_AIF1_DAC1_LEFT_VOLUME,
 		 WM8994_AIF1_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
-SOC_DOUBLE_R_TLV("AIF1DAC2 Volume", WM8994_AIF1_DAC2_LEFT_VOLUME,
-		 WM8994_AIF1_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
 SOC_DOUBLE_R_TLV("AIF2DAC Volume", WM8994_AIF2_DAC_LEFT_VOLUME,
 		 WM8994_AIF2_DAC_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
 
@@ -565,17 +560,12 @@ SOC_SINGLE_TLV("AIF1 Boost Volume", WM8994_AIF1_CONTROL_2, 10, 3, 0, aif_tlv),
 SOC_SINGLE_TLV("AIF2 Boost Volume", WM8994_AIF2_CONTROL_2, 10, 3, 0, aif_tlv),
 
 SOC_SINGLE("AIF1DAC1 EQ Switch", WM8994_AIF1_DAC1_EQ_GAINS_1, 0, 1, 0),
-SOC_SINGLE("AIF1DAC2 EQ Switch", WM8994_AIF1_DAC2_EQ_GAINS_1, 0, 1, 0),
 SOC_SINGLE("AIF2 EQ Switch", WM8994_AIF2_EQ_GAINS_1, 0, 1, 0),
 
 WM8994_DRC_SWITCH("AIF1DAC1 DRC Switch", WM8994_AIF1_DRC1_1, 2),
 WM8994_DRC_SWITCH("AIF1ADC1L DRC Switch", WM8994_AIF1_DRC1_1, 1),
 WM8994_DRC_SWITCH("AIF1ADC1R DRC Switch", WM8994_AIF1_DRC1_1, 0),
 
-WM8994_DRC_SWITCH("AIF1DAC2 DRC Switch", WM8994_AIF1_DRC2_1, 2),
-WM8994_DRC_SWITCH("AIF1ADC2L DRC Switch", WM8994_AIF1_DRC2_1, 1),
-WM8994_DRC_SWITCH("AIF1ADC2R DRC Switch", WM8994_AIF1_DRC2_1, 0),
-
 WM8994_DRC_SWITCH("AIF2DAC DRC Switch", WM8994_AIF2_DRC_1, 2),
 WM8994_DRC_SWITCH("AIF2ADCL DRC Switch", WM8994_AIF2_DRC_1, 1),
 WM8994_DRC_SWITCH("AIF2ADCR DRC Switch", WM8994_AIF2_DRC_1, 0),
@@ -594,9 +584,6 @@ SOC_SINGLE("Sidetone HPF Switch", WM8994_SIDETONE, 6, 1, 0),
 SOC_ENUM("AIF1ADC1 HPF Mode", aif1adc1_hpf),
 SOC_DOUBLE("AIF1ADC1 HPF Switch", WM8994_AIF1_ADC1_FILTERS, 12, 11, 1, 0),
 
-SOC_ENUM("AIF1ADC2 HPF Mode", aif1adc2_hpf),
-SOC_DOUBLE("AIF1ADC2 HPF Switch", WM8994_AIF1_ADC2_FILTERS, 12, 11, 1, 0),
-
 SOC_ENUM("AIF2ADC HPF Mode", aif2adc_hpf),
 SOC_DOUBLE("AIF2ADC HPF Switch", WM8994_AIF2_ADC_FILTERS, 12, 11, 1, 0),
 
@@ -637,6 +624,24 @@ SOC_SINGLE("AIF2DAC 3D Stereo Switch", WM8994_AIF2_DAC_FILTERS_2,
 	   8, 1, 0),
 };
 
+/* Controls not available on WM1811 */
+static const struct snd_kcontrol_new wm8994_snd_controls[] = {
+SOC_DOUBLE_R_TLV("AIF1ADC2 Volume", WM8994_AIF1_ADC2_LEFT_VOLUME,
+		 WM8994_AIF1_ADC2_RIGHT_VOLUME,
+		 1, 119, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("AIF1DAC2 Volume", WM8994_AIF1_DAC2_LEFT_VOLUME,
+		 WM8994_AIF1_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
+
+SOC_SINGLE("AIF1DAC2 EQ Switch", WM8994_AIF1_DAC2_EQ_GAINS_1, 0, 1, 0),
+
+WM8994_DRC_SWITCH("AIF1DAC2 DRC Switch", WM8994_AIF1_DRC2_1, 2),
+WM8994_DRC_SWITCH("AIF1ADC2L DRC Switch", WM8994_AIF1_DRC2_1, 1),
+WM8994_DRC_SWITCH("AIF1ADC2R DRC Switch", WM8994_AIF1_DRC2_1, 0),
+
+SOC_ENUM("AIF1ADC2 HPF Mode", aif1adc2_hpf),
+SOC_DOUBLE("AIF1ADC2 HPF Switch", WM8994_AIF1_ADC2_FILTERS, 12, 11, 1, 0),
+};
+
 static const struct snd_kcontrol_new wm8994_eq_controls[] = {
 SOC_SINGLE_TLV("AIF1DAC1 EQ1 Volume", WM8994_AIF1_DAC1_EQ_GAINS_1, 11, 31, 0,
 	       eq_tlv),
@@ -4385,13 +4390,15 @@ static int wm8994_component_probe(struct snd_soc_component *component)
 	wm8994_handle_pdata(wm8994);
 
 	wm_hubs_add_analogue_controls(component);
-	snd_soc_add_component_controls(component, wm8994_snd_controls,
-			     ARRAY_SIZE(wm8994_snd_controls));
+	snd_soc_add_component_controls(component, wm8994_common_snd_controls,
+				       ARRAY_SIZE(wm8994_common_snd_controls));
 	snd_soc_dapm_new_controls(dapm, wm8994_dapm_widgets,
 				  ARRAY_SIZE(wm8994_dapm_widgets));
 
 	switch (control->type) {
 	case WM8994:
+		snd_soc_add_component_controls(component, wm8994_snd_controls,
+					       ARRAY_SIZE(wm8994_snd_controls));
 		snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets,
 					  ARRAY_SIZE(wm8994_specific_dapm_widgets));
 		if (control->revision < 4) {
@@ -4411,8 +4418,10 @@ static int wm8994_component_probe(struct snd_soc_component *component)
 		}
 		break;
 	case WM8958:
+		snd_soc_add_component_controls(component, wm8994_snd_controls,
+					       ARRAY_SIZE(wm8994_snd_controls));
 		snd_soc_add_component_controls(component, wm8958_snd_controls,
-				     ARRAY_SIZE(wm8958_snd_controls));
+					       ARRAY_SIZE(wm8958_snd_controls));
 		snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets,
 					  ARRAY_SIZE(wm8958_dapm_widgets));
 		if (control->revision < 1) {
diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c
index ae28d9907c30..85396d920e0a 100644
--- a/sound/soc/codecs/wm_adsp.c
+++ b/sound/soc/codecs/wm_adsp.c
@@ -3697,11 +3697,16 @@ static int wm_adsp_buffer_parse_legacy(struct wm_adsp *dsp)
 	u32 xmalg, addr, magic;
 	int i, ret;
 
+	alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id);
+	if (!alg_region) {
+		adsp_err(dsp, "No algorithm region found\n");
+		return -EINVAL;
+	}
+
 	buf = wm_adsp_buffer_alloc(dsp);
 	if (!buf)
 		return -ENOMEM;
 
-	alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id);
 	xmalg = dsp->ops->sys_config_size / sizeof(__be32);
 
 	addr = alg_region->base + xmalg + ALG_XM_FIELD(magic);
diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c
index a437567b8cee..5ce643d62faf 100644
--- a/sound/soc/intel/boards/sof_rt5682.c
+++ b/sound/soc/intel/boards/sof_rt5682.c
@@ -308,6 +308,9 @@ static const struct snd_soc_dapm_widget sof_widgets[] = {
 	SND_SOC_DAPM_HP("Headphone Jack", NULL),
 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
 	SND_SOC_DAPM_SPK("Spk", NULL),
+};
+
+static const struct snd_soc_dapm_widget dmic_widgets[] = {
 	SND_SOC_DAPM_MIC("SoC DMIC", NULL),
 };
 
@@ -318,10 +321,6 @@ static const struct snd_soc_dapm_route sof_map[] = {
 
 	/* other jacks */
 	{ "IN1P", NULL, "Headset Mic" },
-
-	/* digital mics */
-	{"DMic", NULL, "SoC DMIC"},
-
 };
 
 static const struct snd_soc_dapm_route speaker_map[] = {
@@ -329,6 +328,11 @@ static const struct snd_soc_dapm_route speaker_map[] = {
 	{ "Spk", NULL, "Speaker" },
 };
 
+static const struct snd_soc_dapm_route dmic_map[] = {
+	/* digital mics */
+	{"DMic", NULL, "SoC DMIC"},
+};
+
 static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_card *card = rtd->card;
@@ -342,6 +346,28 @@ static int speaker_codec_init(struct snd_soc_pcm_runtime *rtd)
 	return ret;
 }
 
+static int dmic_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(&card->dapm, dmic_widgets,
+					ARRAY_SIZE(dmic_widgets));
+	if (ret) {
+		dev_err(card->dev, "DMic widget addition failed: %d\n", ret);
+		/* Don't need to add routes if widget addition failed */
+		return ret;
+	}
+
+	ret = snd_soc_dapm_add_routes(&card->dapm, dmic_map,
+				      ARRAY_SIZE(dmic_map));
+
+	if (ret)
+		dev_err(card->dev, "DMic map addition failed: %d\n", ret);
+
+	return ret;
+}
+
 /* sof audio machine driver for rt5682 codec */
 static struct snd_soc_card sof_audio_card_rt5682 = {
 	.name = "sof_rt5682",
@@ -445,6 +471,7 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev,
 		links[id].name = "dmic01";
 		links[id].cpus = &cpus[id];
 		links[id].cpus->dai_name = "DMIC01 Pin";
+		links[id].init = dmic_init;
 		if (dmic_be_num > 1) {
 			/* set up 2 BE links at most */
 			links[id + 1].name = "dmic16k";
@@ -621,8 +648,24 @@ static int sof_audio_probe(struct platform_device *pdev)
 					  &sof_audio_card_rt5682);
 }
 
+static int sof_rt5682_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct snd_soc_component *component = NULL;
+
+	for_each_card_components(card, component) {
+		if (!strcmp(component->name, rt5682_component[0].name)) {
+			snd_soc_component_set_jack(component, NULL, NULL);
+			break;
+		}
+	}
+
+	return 0;
+}
+
 static struct platform_driver sof_audio = {
 	.probe = sof_audio_probe,
+	.remove = sof_rt5682_remove,
 	.driver = {
 		.name = "sof_rt5682",
 		.pm = &snd_soc_pm_ops,
diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c
index c213913eb984..fd8c6642fb0d 100644
--- a/sound/soc/samsung/arndale_rt5631.c
+++ b/sound/soc/samsung/arndale_rt5631.c
@@ -5,6 +5,7 @@
 //  Author: Claude <claude@insginal.co.kr>
 
 #include <linux/module.h>
+#include <linux/of_device.h>
 #include <linux/platform_device.h>
 #include <linux/clk.h>
 
@@ -74,6 +75,17 @@ static struct snd_soc_card arndale_rt5631 = {
 	.num_links = ARRAY_SIZE(arndale_rt5631_dai),
 };
 
+static void arndale_put_of_nodes(struct snd_soc_card *card)
+{
+	struct snd_soc_dai_link *dai_link;
+	int i;
+
+	for_each_card_prelinks(card, i, dai_link) {
+		of_node_put(dai_link->cpus->of_node);
+		of_node_put(dai_link->codecs->of_node);
+	}
+}
+
 static int arndale_audio_probe(struct platform_device *pdev)
 {
 	int n, ret;
@@ -103,18 +115,31 @@ static int arndale_audio_probe(struct platform_device *pdev)
 		if (!arndale_rt5631_dai[0].codecs->of_node) {
 			dev_err(&pdev->dev,
 			"Property 'samsung,audio-codec' missing or invalid\n");
-			return -EINVAL;
+			ret = -EINVAL;
+			goto err_put_of_nodes;
 		}
 	}
 
 	ret = devm_snd_soc_register_card(card->dev, card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret);
+		goto err_put_of_nodes;
+	}
+	return 0;
 
-	if (ret)
-		dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret);
-
+err_put_of_nodes:
+	arndale_put_of_nodes(card);
 	return ret;
 }
 
+static int arndale_audio_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	arndale_put_of_nodes(card);
+	return 0;
+}
+
 static const struct of_device_id samsung_arndale_rt5631_of_match[] __maybe_unused = {
 	{ .compatible = "samsung,arndale-rt5631", },
 	{ .compatible = "samsung,arndale-alc5631", },
@@ -129,6 +154,7 @@ static struct platform_driver arndale_audio_driver = {
 		.of_match_table = of_match_ptr(samsung_arndale_rt5631_of_match),
 	},
 	.probe = arndale_audio_probe,
+	.remove = arndale_audio_remove,
 };
 
 module_platform_driver(arndale_audio_driver);
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index e7a04c892817..66910500e3b6 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1070,7 +1070,7 @@ static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd)
 			return ret;
 	}
 
-	snd_soc_dai_trigger(cpu_dai, substream, cmd);
+	ret = snd_soc_dai_trigger(cpu_dai, substream, cmd);
 	if (ret < 0)
 		return ret;
 
@@ -1155,7 +1155,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream,
 			return ret;
 	}
 
-	snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd);
+	ret = snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd);
 	if (ret < 0)
 		return ret;
 
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index aa9a1fca46fa..0fd032914a31 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1582,7 +1582,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg,
 
 	/* map user to kernel widget ID */
 	template.id = get_widget_id(le32_to_cpu(w->id));
-	if (template.id < 0)
+	if ((int)template.id < 0)
 		return template.id;
 
 	/* strings are allocated here, but used and freed by the widget */
diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig
index 479ba249e219..d62f51d33be1 100644
--- a/sound/soc/sof/intel/Kconfig
+++ b/sound/soc/sof/intel/Kconfig
@@ -273,6 +273,16 @@ config SND_SOC_SOF_HDA_AUDIO_CODEC
 	  Say Y if you want to enable HDAudio codecs with SOF.
 	  If unsure select "N".
 
+config SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1
+	bool "SOF enable DMI Link L1"
+	help
+	  This option enables DMI L1 for both playback and capture
+	  and disables known workarounds for specific HDaudio platforms.
+	  Only use to look into power optimizations on platforms not
+	  affected by DMI L1 issues. This option is not recommended.
+	  Say Y if you want to enable DMI Link L1
+	  If unsure, select "N".
+
 endif ## SND_SOC_SOF_HDA_COMMON
 
 config SND_SOC_SOF_HDA_LINK_BASELINE
diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c
index 170c74dc13b1..f395d0638876 100644
--- a/sound/soc/sof/intel/bdw.c
+++ b/sound/soc/sof/intel/bdw.c
@@ -37,6 +37,7 @@
 #define MBOX_SIZE       0x1000
 #define MBOX_DUMP_SIZE 0x30
 #define EXCEPT_OFFSET	0x800
+#define EXCEPT_MAX_HDR_SIZE	0x400
 
 /* DSP peripherals */
 #define DMAC0_OFFSET    0xFE000
@@ -228,6 +229,11 @@ static void bdw_get_registers(struct snd_sof_dev *sdev,
 	/* note: variable AR register array is not read */
 
 	/* then get panic info */
+	if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) {
+		dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n",
+			xoops->arch_hdr.totalsize);
+		return;
+	}
 	offset += xoops->arch_hdr.totalsize;
 	sof_mailbox_read(sdev, offset, panic_info, sizeof(*panic_info));
 
@@ -471,6 +477,7 @@ static int bdw_probe(struct snd_sof_dev *sdev)
 	/* TODO: add offsets */
 	sdev->mmio_bar = BDW_DSP_BAR;
 	sdev->mailbox_bar = BDW_DSP_BAR;
+	sdev->dsp_oops_offset = MBOX_OFFSET;
 
 	/* PCI base */
 	mmio = platform_get_resource(pdev, IORESOURCE_MEM,
diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c
index ecbc9c79bcad..b2597ecfdc1c 100644
--- a/sound/soc/sof/intel/byt.c
+++ b/sound/soc/sof/intel/byt.c
@@ -28,6 +28,7 @@
 #define MBOX_OFFSET		0x144000
 #define MBOX_SIZE		0x1000
 #define EXCEPT_OFFSET		0x800
+#define EXCEPT_MAX_HDR_SIZE	0x400
 
 /* DSP peripherals */
 #define DMAC0_OFFSET		0x098000
@@ -126,6 +127,11 @@ static void byt_get_registers(struct snd_sof_dev *sdev,
 	/* note: variable AR register array is not read */
 
 	/* then get panic info */
+	if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) {
+		dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n",
+			xoops->arch_hdr.totalsize);
+		return;
+	}
 	offset += xoops->arch_hdr.totalsize;
 	sof_mailbox_read(sdev, offset, panic_info, sizeof(*panic_info));
 
diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index bc41028a7a01..df1909e1d950 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -139,20 +139,16 @@ void hda_dsp_ctrl_misc_clock_gating(struct snd_sof_dev *sdev, bool enable)
  */
 int hda_dsp_ctrl_clock_power_gating(struct snd_sof_dev *sdev, bool enable)
 {
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
-	struct hdac_bus *bus = sof_to_bus(sdev);
-#endif
 	u32 val;
 
 	/* enable/disable audio dsp clock gating */
 	val = enable ? PCI_CGCTL_ADSPDCGE : 0;
 	snd_sof_pci_update_bits(sdev, PCI_CGCTL, PCI_CGCTL_ADSPDCGE, val);
 
-#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA)
-	/* enable/disable L1 support */
-	val = enable ? SOF_HDA_VS_EM2_L1SEN : 0;
-	snd_hdac_chip_updatel(bus, VS_EM2, SOF_HDA_VS_EM2_L1SEN, val);
-#endif
+	/* enable/disable DMI Link L1 support */
+	val = enable ? HDA_VS_INTEL_EM2_L1SEN : 0;
+	snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, HDA_VS_INTEL_EM2,
+				HDA_VS_INTEL_EM2_L1SEN, val);
 
 	/* enable/disable audio dsp power gating */
 	val = enable ? 0 : PCI_PGCTL_ADSPPGD;
diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c
index 6427f0b3a2f1..65c2af3fcaab 100644
--- a/sound/soc/sof/intel/hda-loader.c
+++ b/sound/soc/sof/intel/hda-loader.c
@@ -44,6 +44,7 @@ static int cl_stream_prepare(struct snd_sof_dev *sdev, unsigned int format,
 		return -ENODEV;
 	}
 	hstream = &dsp_stream->hstream;
+	hstream->substream = NULL;
 
 	/* allocate DMA buffer */
 	ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, &pci->dev, size, dmab);
diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c
index ad8d41f22e92..2c7447188402 100644
--- a/sound/soc/sof/intel/hda-stream.c
+++ b/sound/soc/sof/intel/hda-stream.c
@@ -185,6 +185,17 @@ hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction)
 			direction == SNDRV_PCM_STREAM_PLAYBACK ?
 			"playback" : "capture");
 
+	/*
+	 * Disable DMI Link L1 entry when capture stream is opened.
+	 * Workaround to address a known issue with host DMA that results
+	 * in xruns during pause/release in capture scenarios.
+	 */
+	if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1))
+		if (stream && direction == SNDRV_PCM_STREAM_CAPTURE)
+			snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR,
+						HDA_VS_INTEL_EM2,
+						HDA_VS_INTEL_EM2_L1SEN, 0);
+
 	return stream;
 }
 
@@ -193,23 +204,43 @@ int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag)
 {
 	struct hdac_bus *bus = sof_to_bus(sdev);
 	struct hdac_stream *s;
+	bool active_capture_stream = false;
+	bool found = false;
 
 	spin_lock_irq(&bus->reg_lock);
 
-	/* find used stream */
+	/*
+	 * close stream matching the stream tag
+	 * and check if there are any open capture streams.
+	 */
 	list_for_each_entry(s, &bus->stream_list, list) {
-		if (s->direction == direction &&
-		    s->opened && s->stream_tag == stream_tag) {
+		if (!s->opened)
+			continue;
+
+		if (s->direction == direction && s->stream_tag == stream_tag) {
 			s->opened = false;
-			spin_unlock_irq(&bus->reg_lock);
-			return 0;
+			found = true;
+		} else if (s->direction == SNDRV_PCM_STREAM_CAPTURE) {
+			active_capture_stream = true;
 		}
 	}
 
 	spin_unlock_irq(&bus->reg_lock);
 
-	dev_dbg(sdev->dev, "stream_tag %d not opened!\n", stream_tag);
-	return -ENODEV;
+	/* Enable DMI L1 entry if there are no capture streams open */
+	if (!IS_ENABLED(SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1))
+		if (!active_capture_stream)
+			snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR,
+						HDA_VS_INTEL_EM2,
+						HDA_VS_INTEL_EM2_L1SEN,
+						HDA_VS_INTEL_EM2_L1SEN);
+
+	if (!found) {
+		dev_dbg(sdev->dev, "stream_tag %d not opened!\n", stream_tag);
+		return -ENODEV;
+	}
+
+	return 0;
 }
 
 int hda_dsp_stream_trigger(struct snd_sof_dev *sdev,
diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c
index c72e9a09eee1..06e84679087b 100644
--- a/sound/soc/sof/intel/hda.c
+++ b/sound/soc/sof/intel/hda.c
@@ -35,6 +35,8 @@
 #define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348)
 #define IS_CNL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9dc8)
 
+#define EXCEPT_MAX_HDR_SIZE	0x400
+
 /*
  * Debug
  */
@@ -131,6 +133,11 @@ static void hda_dsp_get_registers(struct snd_sof_dev *sdev,
 	/* note: variable AR register array is not read */
 
 	/* then get panic info */
+	if (xoops->arch_hdr.totalsize > EXCEPT_MAX_HDR_SIZE) {
+		dev_err(sdev->dev, "invalid header size 0x%x. FW oops is bogus\n",
+			xoops->arch_hdr.totalsize);
+		return;
+	}
 	offset += xoops->arch_hdr.totalsize;
 	sof_block_read(sdev, sdev->mmio_bar, offset,
 		       panic_info, sizeof(*panic_info));
diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h
index 5591841a1b6f..23e430d3e056 100644
--- a/sound/soc/sof/intel/hda.h
+++ b/sound/soc/sof/intel/hda.h
@@ -39,7 +39,6 @@
 #define SOF_HDA_WAKESTS			0x0E
 #define SOF_HDA_WAKESTS_INT_MASK	((1 << 8) - 1)
 #define SOF_HDA_RIRBSTS			0x5d
-#define SOF_HDA_VS_EM2_L1SEN            BIT(13)
 
 /* SOF_HDA_GCTL register bist */
 #define SOF_HDA_GCTL_RESET		BIT(0)
@@ -228,6 +227,10 @@
 #define HDA_DSP_REG_HIPCIE		(HDA_DSP_IPC_BASE + 0x0C)
 #define HDA_DSP_REG_HIPCCTL		(HDA_DSP_IPC_BASE + 0x10)
 
+/* Intel Vendor Specific Registers */
+#define HDA_VS_INTEL_EM2		0x1030
+#define HDA_VS_INTEL_EM2_L1SEN		BIT(13)
+
 /*  HIPCI */
 #define HDA_DSP_REG_HIPCI_BUSY		BIT(31)
 #define HDA_DSP_REG_HIPCI_MSG_MASK	0x7FFFFFFF
diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index d7f32745fefe..9a9a381a908d 100644
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -546,10 +546,10 @@ int snd_sof_run_firmware(struct snd_sof_dev *sdev)
 				 msecs_to_jiffies(sdev->boot_timeout));
 	if (ret == 0) {
 		dev_err(sdev->dev, "error: firmware boot failure\n");
-		/* after this point FW_READY msg should be ignored */
-		sdev->boot_complete = true;
 		snd_sof_dsp_dbg_dump(sdev, SOF_DBG_REGS | SOF_DBG_MBOX |
 			SOF_DBG_TEXT | SOF_DBG_PCI);
+		/* after this point FW_READY msg should be ignored */
+		sdev->boot_complete = true;
 		return -EIO;
 	}
 
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c
index e3f6a6dc0f36..2b876d497447 100644
--- a/sound/soc/sof/pcm.c
+++ b/sound/soc/sof/pcm.c
@@ -244,7 +244,7 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream)
 		snd_soc_rtdcom_lookup(rtd, DRV_NAME);
 	struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component);
 	struct snd_sof_pcm *spcm;
-	int ret;
+	int ret, err = 0;
 
 	/* nothing to do for BE */
 	if (rtd->dai_link->no_pcm)
@@ -254,26 +254,26 @@ static int sof_pcm_hw_free(struct snd_pcm_substream *substream)
 	if (!spcm)
 		return -EINVAL;
 
-	if (!spcm->prepared[substream->stream])
-		return 0;
-
 	dev_dbg(sdev->dev, "pcm: free stream %d dir %d\n", spcm->pcm.pcm_id,
 		substream->stream);
 
-	ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm);
+	if (spcm->prepared[substream->stream]) {
+		ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm);
+		if (ret < 0)
+			err = ret;
+	}
 
 	snd_pcm_lib_free_pages(substream);
 
 	cancel_work_sync(&spcm->stream[substream->stream].period_elapsed_work);
 
-	if (ret < 0)
-		return ret;
-
 	ret = snd_sof_pcm_platform_hw_free(sdev, substream);
-	if (ret < 0)
+	if (ret < 0) {
 		dev_err(sdev->dev, "error: platform hw free failed\n");
+		err = ret;
+	}
 
-	return ret;
+	return err;
 }
 
 static int sof_pcm_prepare(struct snd_pcm_substream *substream)
@@ -323,6 +323,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct sof_ipc_stream stream;
 	struct sof_ipc_reply reply;
 	bool reset_hw_params = false;
+	bool ipc_first = false;
 	int ret;
 
 	/* nothing to do for BE */
@@ -343,6 +344,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_PAUSE;
+		ipc_first = true;
 		break;
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_RELEASE;
@@ -363,6 +365,7 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP;
+		ipc_first = true;
 		reset_hw_params = true;
 		break;
 	default:
@@ -370,12 +373,22 @@ static int sof_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 		return -EINVAL;
 	}
 
-	snd_sof_pcm_platform_trigger(sdev, substream, cmd);
+	/*
+	 * DMA and IPC sequence is different for start and stop. Need to send
+	 * STOP IPC before stop DMA
+	 */
+	if (!ipc_first)
+		snd_sof_pcm_platform_trigger(sdev, substream, cmd);
 
 	/* send IPC to the DSP */
 	ret = sof_ipc_tx_message(sdev->ipc, stream.hdr.cmd, &stream,
 				 sizeof(stream), &reply, sizeof(reply));
 
+	/* need to STOP DMA even if STOP IPC failed */
+	if (ipc_first)
+		snd_sof_pcm_platform_trigger(sdev, substream, cmd);
+
+	/* free PCM if reset_hw_params is set and the STOP IPC is successful */
 	if (!ret && reset_hw_params)
 		ret = sof_pcm_dsp_pcm_free(substream, sdev, spcm);
 
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index fc85efbad378..0aabb3190ddc 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -920,7 +920,9 @@ static void sof_parse_word_tokens(struct snd_soc_component *scomp,
 		for (j = 0; j < count; j++) {
 			/* match token type */
 			if (!(tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_WORD ||
-			      tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_SHORT))
+			      tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_SHORT ||
+			      tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_BYTE ||
+			      tokens[j].type == SND_SOC_TPLG_TUPLE_TYPE_BOOL))
 				continue;
 
 			/* match token id */