ASoC: Merge up fixes as a dependency for future SOF work

New SOF changes require the current set of fixes.

27 files changed, 228 insertions, 86 deletions

diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c
index 4a4f09f924bc..e3d398b8f54e 100644
--- a/sound/soc/codecs/da7219-aad.c
+++ b/sound/soc/codecs/da7219-aad.c
@@ -968,6 +968,8 @@ int da7219_aad_init(struct snd_soc_component *component)
 	INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work);
 	INIT_WORK(&da7219_aad->jack_det_work, da7219_aad_jack_det_work);
 
+	mutex_init(&da7219_aad->jack_det_mutex);
+
 	ret = request_threaded_irq(da7219_aad->irq, da7219_aad_pre_irq_thread,
 				   da7219_aad_irq_thread,
 				   IRQF_TRIGGER_LOW | IRQF_ONESHOT,
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c
index 01e8ffda2a4b..6d980fbc4207 100644
--- a/sound/soc/codecs/hdmi-codec.c
+++ b/sound/soc/codecs/hdmi-codec.c
@@ -428,8 +428,13 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream,
 {
 	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	bool has_capture = !hcp->hcd.no_i2s_capture;
+	bool has_playback = !hcp->hcd.no_i2s_playback;
 	int ret = 0;
 
+	if (!((has_playback && tx) || (has_capture && !tx)))
+		return 0;
+
 	mutex_lock(&hcp->lock);
 	if (hcp->busy) {
 		dev_err(dai->dev, "Only one simultaneous stream supported!\n");
@@ -468,6 +473,12 @@ static void hdmi_codec_shutdown(struct snd_pcm_substream *substream,
 				struct snd_soc_dai *dai)
 {
 	struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	bool has_capture = !hcp->hcd.no_i2s_capture;
+	bool has_playback = !hcp->hcd.no_i2s_playback;
+
+	if (!((has_playback && tx) || (has_capture && !tx)))
+		return;
 
 	hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN;
 	hcp->hcd.ops->audio_shutdown(dai->dev->parent, hcp->hcd.data);
diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c
index bf27bdd5be20..473d3cd39554 100644
--- a/sound/soc/codecs/lpass-tx-macro.c
+++ b/sound/soc/codecs/lpass-tx-macro.c
@@ -242,7 +242,7 @@ enum {
 
 struct tx_mute_work {
 	struct tx_macro *tx;
-	u32 decimator;
+	u8 decimator;
 	struct delayed_work dwork;
 };
 
@@ -635,7 +635,7 @@ exit:
 	return 0;
 }
 
-static bool is_amic_enabled(struct snd_soc_component *component, int decimator)
+static bool is_amic_enabled(struct snd_soc_component *component, u8 decimator)
 {
 	u16 adc_mux_reg, adc_reg, adc_n;
 
@@ -849,7 +849,7 @@ static int tx_macro_enable_dec(struct snd_soc_dapm_widget *w,
 			       struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
-	unsigned int decimator;
+	u8 decimator;
 	u16 tx_vol_ctl_reg, dec_cfg_reg, hpf_gate_reg, tx_gain_ctl_reg;
 	u8 hpf_cut_off_freq;
 	int hpf_delay = TX_MACRO_DMIC_HPF_DELAY_MS;
@@ -1064,7 +1064,8 @@ static int tx_macro_hw_params(struct snd_pcm_substream *substream,
 			      struct snd_soc_dai *dai)
 {
 	struct snd_soc_component *component = dai->component;
-	u32 decimator, sample_rate;
+	u32 sample_rate;
+	u8 decimator;
 	int tx_fs_rate;
 	struct tx_macro *tx = snd_soc_component_get_drvdata(component);
 
@@ -1128,7 +1129,7 @@ static int tx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream)
 {
 	struct snd_soc_component *component = dai->component;
 	struct tx_macro *tx = snd_soc_component_get_drvdata(component);
-	u16 decimator;
+	u8 decimator;
 
 	/* active decimator not set yet */
 	if (tx->active_decimator[dai->id] == -1)
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 17db29c25d96..725c530a3636 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -303,6 +303,10 @@ config SND_SOC_IMX_SGTL5000
 	  Say Y if you want to add support for SoC audio on an i.MX board with
 	  a sgtl5000 codec.
 
+	  Note that this is an old driver. Consider enabling
+	  SND_SOC_FSL_ASOC_CARD and SND_SOC_SGTL5000 to use the newer
+	  driver.
+
 config SND_SOC_IMX_SPDIF
 	tristate "SoC Audio support for i.MX boards with S/PDIF"
 	select SND_SOC_IMX_PCM_DMA
diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c
index acd43b6108e9..1a1d572cc1d0 100644
--- a/sound/soc/intel/avs/boards/da7219.c
+++ b/sound/soc/intel/avs/boards/da7219.c
@@ -117,6 +117,26 @@ static void avs_da7219_codec_exit(struct snd_soc_pcm_runtime *rtd)
 	snd_soc_component_set_jack(asoc_rtd_to_codec(rtd, 0)->component, NULL, NULL);
 }
 
+static int
+avs_da7219_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate, *channels;
+	struct snd_mask *fmt;
+
+	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	/* The ADSP will convert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP0 to 24 bit */
+	snd_mask_none(fmt);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
+	return 0;
+}
+
 static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port,
 			       struct snd_soc_dai_link **dai_link)
 {
@@ -148,6 +168,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
 	dl->num_platforms = 1;
 	dl->id = 0;
 	dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS;
+	dl->be_hw_params_fixup = avs_da7219_be_fixup;
 	dl->init = avs_da7219_codec_init;
 	dl->exit = avs_da7219_codec_exit;
 	dl->nonatomic = 1;
diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c
index 921f42caf7e0..183123d08c5a 100644
--- a/sound/soc/intel/avs/boards/max98357a.c
+++ b/sound/soc/intel/avs/boards/max98357a.c
@@ -8,6 +8,7 @@
 
 #include <linux/module.h>
 #include <linux/platform_device.h>
+#include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <sound/soc-acpi.h>
 #include <sound/soc-dapm.h>
@@ -24,6 +25,26 @@ static const struct snd_soc_dapm_route card_base_routes[] = {
 	{ "Spk", NULL, "Speaker" },
 };
 
+static int
+avs_max98357a_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate, *channels;
+	struct snd_mask *fmt;
+
+	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	/* The ADSP will convert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP0 to 16 bit */
+	snd_mask_none(fmt);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
+	return 0;
+}
+
 static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port,
 			       struct snd_soc_dai_link **dai_link)
 {
@@ -55,6 +76,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
 	dl->num_platforms = 1;
 	dl->id = 0;
 	dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS;
+	dl->be_hw_params_fixup = avs_max98357a_be_fixup;
 	dl->nonatomic = 1;
 	dl->no_pcm = 1;
 	dl->dpcm_playback = 1;
diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c
index b31fa931ba8b..b69fc5567135 100644
--- a/sound/soc/intel/avs/boards/nau8825.c
+++ b/sound/soc/intel/avs/boards/nau8825.c
@@ -33,15 +33,15 @@ avs_nau8825_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *co
 		return -EINVAL;
 	}
 
-	if (!SND_SOC_DAPM_EVENT_ON(event)) {
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000,
+					     SND_SOC_CLOCK_IN);
+	else
 		ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN);
-		if (ret < 0) {
-			dev_err(card->dev, "set sysclk err = %d\n", ret);
-			return ret;
-		}
-	}
+	if (ret < 0)
+		dev_err(card->dev, "Set sysclk failed: %d\n", ret);
 
-	return 0;
+	return ret;
 }
 
 static const struct snd_kcontrol_new card_controls[] = {
diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c
index 473e9fe5d0bf..b2c2ba93dcb5 100644
--- a/sound/soc/intel/avs/boards/rt5682.c
+++ b/sound/soc/intel/avs/boards/rt5682.c
@@ -169,6 +169,27 @@ static const struct snd_soc_ops avs_rt5682_ops = {
 	.hw_params = avs_rt5682_hw_params,
 };
 
+static int
+avs_rt5682_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params)
+{
+	struct snd_interval *rate, *channels;
+	struct snd_mask *fmt;
+
+	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+	fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+
+	/* The ADSP will convert the FE rate to 48k, stereo */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSPN to 24 bit */
+	snd_mask_none(fmt);
+	snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
+
+	return 0;
+}
+
 static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port,
 			       struct snd_soc_dai_link **dai_link)
 {
@@ -201,6 +222,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in
 	dl->id = 0;
 	dl->init = avs_rt5682_codec_init;
 	dl->exit = avs_rt5682_codec_exit;
+	dl->be_hw_params_fixup = avs_rt5682_be_fixup;
 	dl->ops = &avs_rt5682_ops;
 	dl->nonatomic = 1;
 	dl->no_pcm = 1;
diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c
index c5db69612762..2b7f5ad92aca 100644
--- a/sound/soc/intel/avs/boards/ssm4567.c
+++ b/sound/soc/intel/avs/boards/ssm4567.c
@@ -15,7 +15,6 @@
 #include <sound/soc-acpi.h>
 #include "../../../codecs/nau8825.h"
 
-#define SKL_NUVOTON_CODEC_DAI	"nau8825-hifi"
 #define SKL_SSM_CODEC_DAI	"ssm4567-hifi"
 
 static struct snd_soc_codec_conf card_codec_conf[] = {
@@ -34,41 +33,11 @@ static const struct snd_kcontrol_new card_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Right Speaker"),
 };
 
-static int
-platform_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event)
-{
-	struct snd_soc_dapm_context *dapm = w->dapm;
-	struct snd_soc_card *card = dapm->card;
-	struct snd_soc_dai *codec_dai;
-	int ret;
-
-	codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI);
-	if (!codec_dai) {
-		dev_err(card->dev, "Codec dai not found\n");
-		return -EINVAL;
-	}
-
-	if (SND_SOC_DAPM_EVENT_ON(event)) {
-		ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000,
-					     SND_SOC_CLOCK_IN);
-		if (ret < 0)
-			dev_err(card->dev, "set sysclk err = %d\n", ret);
-	} else {
-		ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN);
-		if (ret < 0)
-			dev_err(card->dev, "set sysclk err = %d\n", ret);
-	}
-
-	return ret;
-}
-
 static const struct snd_soc_dapm_widget card_widgets[] = {
 	SND_SOC_DAPM_SPK("Left Speaker", NULL),
 	SND_SOC_DAPM_SPK("Right Speaker", NULL),
 	SND_SOC_DAPM_SPK("DP1", NULL),
 	SND_SOC_DAPM_SPK("DP2", NULL),
-	SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control,
-			    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 };
 
 static const struct snd_soc_dapm_route card_base_routes[] = {
diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
index 56ee5fef66a8..28dd2046e4ac 100644
--- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c
@@ -559,7 +559,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = {
 	{
 		.comp_ids = &essx_83x6,
 		.drv_name = "sof-essx8336",
-		.sof_tplg_filename = "sof-adl-es83x6", /* the tplg suffix is added at run time */
+		.sof_tplg_filename = "sof-adl-es8336", /* the tplg suffix is added at run time */
 		.tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER |
 					SND_SOC_ACPI_TPLG_INTEL_SSP_MSB |
 					SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER,
diff --git a/sound/soc/qcom/qdsp6/q6prm.c b/sound/soc/qcom/qdsp6/q6prm.c
index 3aa63aac4a68..81554d202658 100644
--- a/sound/soc/qcom/qdsp6/q6prm.c
+++ b/sound/soc/qcom/qdsp6/q6prm.c
@@ -184,9 +184,9 @@ int q6prm_set_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_
 			  unsigned int freq)
 {
 	if (freq)
-		return q6prm_request_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq);
+		return q6prm_request_lpass_clock(dev, clk_id, clk_attr, clk_root, freq);
 
-	return q6prm_release_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq);
+	return q6prm_release_lpass_clock(dev, clk_id, clk_attr, clk_root, freq);
 }
 EXPORT_SYMBOL_GPL(q6prm_set_lpass_clock);
 
diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c
index a1037512da1f..e1dba6b79065 100644
--- a/sound/soc/sof/intel/hda-ctrl.c
+++ b/sound/soc/sof/intel/hda-ctrl.c
@@ -198,12 +198,15 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev)
 		goto err;
 	}
 
+	usleep_range(500, 1000);
+
 	/* exit HDA controller reset */
 	ret = hda_dsp_ctrl_link_reset(sdev, false);
 	if (ret < 0) {
 		dev_err(sdev->dev, "error: failed to exit HDA controller reset\n");
 		goto err;
 	}
+	usleep_range(1000, 1200);
 
 	hda_codec_detect_mask(sdev);
 
diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c
index e3b69dbc1308..c9231aeacc53 100644
--- a/sound/soc/sof/intel/hda-dsp.c
+++ b/sound/soc/sof/intel/hda-dsp.c
@@ -392,6 +392,12 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value)
 	snd_sof_dsp_update8(sdev, HDA_DSP_HDA_BAR, chip->d0i3_offset,
 			    SOF_HDA_VS_D0I3C_I3, value);
 
+	/*
+	 * The value written to the D0I3C::I3 bit may not be taken into account immediately.
+	 * A delay is recommended before checking if D0I3C::CIP is cleared
+	 */
+	usleep_range(30, 40);
+
 	/* Wait for cmd in progress to be cleared before exiting the function */
 	ret = hda_dsp_wait_d0i3c_done(sdev);
 	if (ret < 0) {
@@ -400,6 +406,12 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value)
 	}
 
 	reg = snd_sof_dsp_read8(sdev, HDA_DSP_HDA_BAR, chip->d0i3_offset);
+	/* Confirm d0i3 state changed with paranoia check */
+	if ((reg ^ value) & SOF_HDA_VS_D0I3C_I3) {
+		dev_err(sdev->dev, "failed to update D0I3C!\n");
+		return -EIO;
+	}
+
 	trace_sof_intel_D0I3C_updated(sdev, reg);
 
 	return 0;
diff --git a/sound/soc/sof/intel/pci-apl.c b/sound/soc/sof/intel/pci-apl.c
index 69279dcc92dc..aff6cb573c27 100644
--- a/sound/soc/sof/intel/pci-apl.c
+++ b/sound/soc/sof/intel/pci-apl.c
@@ -78,6 +78,7 @@ static const struct sof_dev_desc glk_desc = {
 	.nocodec_tplg_filename = "sof-glk-nocodec.tplg",
 	.ops = &sof_apl_ops,
 	.ops_init = sof_apl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 /* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-cnl.c b/sound/soc/sof/intel/pci-cnl.c
index 8db3f8d15b55..4c0c1c369dcd 100644
--- a/sound/soc/sof/intel/pci-cnl.c
+++ b/sound/soc/sof/intel/pci-cnl.c
@@ -48,6 +48,7 @@ static const struct sof_dev_desc cnl_desc = {
 	.nocodec_tplg_filename = "sof-cnl-nocodec.tplg",
 	.ops = &sof_cnl_ops,
 	.ops_init = sof_cnl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 static const struct sof_dev_desc cfl_desc = {
@@ -111,6 +112,7 @@ static const struct sof_dev_desc cml_desc = {
 	.nocodec_tplg_filename = "sof-cnl-nocodec.tplg",
 	.ops = &sof_cnl_ops,
 	.ops_init = sof_cnl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 /* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-icl.c b/sound/soc/sof/intel/pci-icl.c
index d6cf75e357db..6785669113b3 100644
--- a/sound/soc/sof/intel/pci-icl.c
+++ b/sound/soc/sof/intel/pci-icl.c
@@ -79,6 +79,7 @@ static const struct sof_dev_desc jsl_desc = {
 	.nocodec_tplg_filename = "sof-jsl-nocodec.tplg",
 	.ops = &sof_cnl_ops,
 	.ops_init = sof_cnl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 /* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-mtl.c b/sound/soc/sof/intel/pci-mtl.c
index 6e4e6d4ef5a5..b183dc0014b4 100644
--- a/sound/soc/sof/intel/pci-mtl.c
+++ b/sound/soc/sof/intel/pci-mtl.c
@@ -46,6 +46,7 @@ static const struct sof_dev_desc mtl_desc = {
 	.nocodec_tplg_filename = "sof-mtl-nocodec.tplg",
 	.ops = &sof_mtl_ops,
 	.ops_init = sof_mtl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 /* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-skl.c b/sound/soc/sof/intel/pci-skl.c
index 3a99dc444f92..5b4bccf81965 100644
--- a/sound/soc/sof/intel/pci-skl.c
+++ b/sound/soc/sof/intel/pci-skl.c
@@ -38,6 +38,7 @@ static struct sof_dev_desc skl_desc = {
 	.nocodec_tplg_filename = "sof-skl-nocodec.tplg",
 	.ops = &sof_skl_ops,
 	.ops_init = sof_skl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 static struct sof_dev_desc kbl_desc = {
@@ -61,6 +62,7 @@ static struct sof_dev_desc kbl_desc = {
 	.nocodec_tplg_filename = "sof-kbl-nocodec.tplg",
 	.ops = &sof_skl_ops,
 	.ops_init = sof_skl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 /* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c
index e80c4dfef85a..22e769e0831d 100644
--- a/sound/soc/sof/intel/pci-tgl.c
+++ b/sound/soc/sof/intel/pci-tgl.c
@@ -48,6 +48,7 @@ static const struct sof_dev_desc tgl_desc = {
 	.nocodec_tplg_filename = "sof-tgl-nocodec.tplg",
 	.ops = &sof_tgl_ops,
 	.ops_init = sof_tgl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 static const struct sof_dev_desc tglh_desc = {
@@ -110,6 +111,7 @@ static const struct sof_dev_desc ehl_desc = {
 	.nocodec_tplg_filename = "sof-ehl-nocodec.tplg",
 	.ops = &sof_tgl_ops,
 	.ops_init = sof_tgl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 static const struct sof_dev_desc adls_desc = {
@@ -141,6 +143,7 @@ static const struct sof_dev_desc adls_desc = {
 	.nocodec_tplg_filename = "sof-adl-nocodec.tplg",
 	.ops = &sof_tgl_ops,
 	.ops_init = sof_tgl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 static const struct sof_dev_desc adl_desc = {
@@ -172,6 +175,7 @@ static const struct sof_dev_desc adl_desc = {
 	.nocodec_tplg_filename = "sof-adl-nocodec.tplg",
 	.ops = &sof_tgl_ops,
 	.ops_init = sof_tgl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 static const struct sof_dev_desc adl_n_desc = {
@@ -203,6 +207,7 @@ static const struct sof_dev_desc adl_n_desc = {
 	.nocodec_tplg_filename = "sof-adl-nocodec.tplg",
 	.ops = &sof_tgl_ops,
 	.ops_init = sof_tgl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 static const struct sof_dev_desc rpls_desc = {
@@ -234,6 +239,7 @@ static const struct sof_dev_desc rpls_desc = {
 	.nocodec_tplg_filename = "sof-rpl-nocodec.tplg",
 	.ops = &sof_tgl_ops,
 	.ops_init = sof_tgl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 static const struct sof_dev_desc rpl_desc = {
@@ -265,6 +271,7 @@ static const struct sof_dev_desc rpl_desc = {
 	.nocodec_tplg_filename = "sof-rpl-nocodec.tplg",
 	.ops = &sof_tgl_ops,
 	.ops_init = sof_tgl_ops_init,
+	.ops_free = hda_ops_free,
 };
 
 /* PCI IDs */
diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c
index 5b2b409752c5..8c22a00266c0 100644
--- a/sound/soc/sof/intel/pci-tng.c
+++ b/sound/soc/sof/intel/pci-tng.c
@@ -75,11 +75,7 @@ static int tangier_pci_probe(struct snd_sof_dev *sdev)
 
 	/* LPE base */
 	base = pci_resource_start(pci, desc->resindex_lpe_base) - IRAM_OFFSET;
-	size = pci_resource_len(pci, desc->resindex_lpe_base);
-	if (size < PCI_BAR_SIZE) {
-		dev_err(sdev->dev, "error: I/O region is too small.\n");
-		return -ENODEV;
-	}
+	size = PCI_BAR_SIZE;
 
 	dev_dbg(sdev->dev, "LPE PHY base at 0x%x size 0x%x", base, size);
 	sdev->bar[DSP_BAR] = devm_ioremap(sdev->dev, base, size);
diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c
index dceb78bfe17c..b1f425b39db9 100644
--- a/sound/soc/sof/ipc3-topology.c
+++ b/sound/soc/sof/ipc3-topology.c
@@ -2081,7 +2081,9 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
 		break;
 	case SOF_DAI_INTEL_ALH:
 		if (data) {
-			config->dai_index = data->dai_index;
+			/* save the dai_index during hw_params and reuse it for hw_free */
+			if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS)
+				config->dai_index = data->dai_index;
 			config->alh.stream_id = data->dai_data;
 		}
 		break;
@@ -2089,7 +2091,30 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
 		break;
 	}
 
-	config->flags = flags;
+	/*
+	 * The dai_config op is invoked several times and the flags argument varies as below:
+	 * BE DAI hw_params: When the op is invoked during the BE DAI hw_params, flags contains
+	 * SOF_DAI_CONFIG_FLAGS_HW_PARAMS along with quirks
+	 * FE DAI hw_params: When invoked during FE DAI hw_params after the DAI widget has
+	 * just been set up in the DSP, flags is set to SOF_DAI_CONFIG_FLAGS_HW_PARAMS with no
+	 * quirks
+	 * BE DAI trigger: When invoked during the BE DAI trigger, flags is set to
+	 * SOF_DAI_CONFIG_FLAGS_PAUSE and contains no quirks
+	 * BE DAI hw_free: When invoked during the BE DAI hw_free, flags is set to
+	 * SOF_DAI_CONFIG_FLAGS_HW_FREE and contains no quirks
+	 * FE DAI hw_free: When invoked during the FE DAI hw_free, flags is set to
+	 * SOF_DAI_CONFIG_FLAGS_HW_FREE and contains no quirks
+	 *
+	 * The DAI_CONFIG IPC is sent to the DSP, only after the widget is set up during the FE
+	 * DAI hw_params. But since the BE DAI hw_params precedes the FE DAI hw_params, the quirks
+	 * need to be preserved when assigning the flags before sending the IPC.
+	 * For the case of PAUSE/HW_FREE, since there are no quirks, flags can be used as is.
+	 */
+
+	if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS)
+		config->flags |= flags;
+	else
+		config->flags = flags;
 
 	/* only send the IPC if the widget is set up in the DSP */
 	if (swidget->use_count > 0) {
@@ -2097,6 +2122,9 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
 					 &reply, sizeof(reply));
 		if (ret < 0)
 			dev_err(sdev->dev, "Failed to set dai config for %s\n", dai->name);
+
+		/* clear the flags once the IPC has been sent even if it fails */
+		config->flags = SOF_DAI_CONFIG_FLAGS_NONE;
 	}
 
 	return ret;
diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c
index 3de64ea2dc9a..4493bbd7faf1 100644
--- a/sound/soc/sof/ipc3.c
+++ b/sound/soc/sof/ipc3.c
@@ -970,8 +970,9 @@ static void sof_ipc3_rx_msg(struct snd_sof_dev *sdev)
 		return;
 	}
 
-	if (hdr.size < sizeof(hdr)) {
-		dev_err(sdev->dev, "The received message size is invalid\n");
+	if (hdr.size < sizeof(hdr) || hdr.size > SOF_IPC_MSG_MAX_SIZE) {
+		dev_err(sdev->dev, "The received message size is invalid: %u\n",
+			hdr.size);
 		return;
 	}
 
diff --git a/sound/soc/sof/ipc4-control.c b/sound/soc/sof/ipc4-control.c
index 67bd2233fd9a..9a71af1a613a 100644
--- a/sound/soc/sof/ipc4-control.c
+++ b/sound/soc/sof/ipc4-control.c
@@ -97,7 +97,8 @@ sof_ipc4_set_volume_data(struct snd_sof_dev *sdev, struct snd_sof_widget *swidge
 		}
 
 		/* set curve type and duration from topology */
-		data.curve_duration = gain->data.curve_duration;
+		data.curve_duration_l = gain->data.curve_duration_l;
+		data.curve_duration_h = gain->data.curve_duration_h;
 		data.curve_type = gain->data.curve_type;
 
 		msg->data_ptr = &data;
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index cb75f685e722..0bb16ed38e48 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -111,7 +111,7 @@ static const struct sof_topology_token gain_tokens[] = {
 		get_token_u32, offsetof(struct sof_ipc4_gain_data, curve_type)},
 	{SOF_TKN_GAIN_RAMP_DURATION,
 		SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32,
-		offsetof(struct sof_ipc4_gain_data, curve_duration)},
+		offsetof(struct sof_ipc4_gain_data, curve_duration_l)},
 	{SOF_TKN_GAIN_VAL, SND_SOC_TPLG_TUPLE_TYPE_WORD,
 		get_token_u32, offsetof(struct sof_ipc4_gain_data, init_val)},
 };
@@ -159,7 +159,7 @@ static void sof_ipc4_dbg_audio_format(struct device *dev,
 	for (i = 0; i < num_format; i++, ptr = (u8 *)ptr + object_size) {
 		fmt = ptr;
 		dev_dbg(dev,
-			" #%d: %uKHz, %ubit (ch_map %#x ch_cfg %u interleaving_style %u fmt_cfg %#x)\n",
+			" #%d: %uHz, %ubit (ch_map %#x ch_cfg %u interleaving_style %u fmt_cfg %#x)\n",
 			i, fmt->sampling_frequency, fmt->bit_depth, fmt->ch_map,
 			fmt->ch_cfg, fmt->interleaving_style, fmt->fmt_cfg);
 	}
@@ -715,7 +715,7 @@ static int sof_ipc4_widget_setup_comp_pga(struct snd_sof_widget *swidget)
 
 	dev_dbg(scomp->dev,
 		"pga widget %s: ramp type: %d, ramp duration %d, initial gain value: %#x, cpc %d\n",
-		swidget->widget->name, gain->data.curve_type, gain->data.curve_duration,
+		swidget->widget->name, gain->data.curve_type, gain->data.curve_duration_l,
 		gain->data.init_val, gain->base_config.cpc);
 
 	ret = sof_ipc4_widget_setup_msg(swidget, &gain->msg);
@@ -1003,6 +1003,7 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget)
 
 		ipc4_copier = dai->private;
 		if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) {
+			struct sof_ipc4_copier_data *copier_data = &ipc4_copier->data;
 			struct sof_ipc4_alh_configuration_blob *blob;
 			unsigned int group_id;
 
@@ -1012,6 +1013,9 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget)
 					   ALH_MULTI_GTW_BASE;
 				ida_free(&alh_group_ida, group_id);
 			}
+
+			/* clear the node ID */
+			copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK;
 		}
 	}
 
@@ -1963,8 +1967,15 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget *
 		pipeline->skip_during_fe_trigger = true;
 		fallthrough;
 	case SOF_DAI_INTEL_ALH:
-		copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK;
-		copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_data);
+		/*
+		 * Do not clear the node ID when this op is invoked with
+		 * SOF_DAI_CONFIG_FLAGS_HW_FREE. It is needed to free the group_ida during
+		 * unprepare.
+		 */
+		if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) {
+			copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK;
+			copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_data);
+		}
 		break;
 	case SOF_DAI_INTEL_DMIC:
 	case SOF_DAI_INTEL_SSP:
diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h
index 7877f0638011..22900b572766 100644
--- a/sound/soc/sof/ipc4-topology.h
+++ b/sound/soc/sof/ipc4-topology.h
@@ -46,7 +46,7 @@
 #define SOF_IPC4_NODE_INDEX_INTEL_SSP(x) (((x) & 0xf) << 4)
 
 /* Node ID for DMIC type DAI copiers */
-#define SOF_IPC4_NODE_INDEX_INTEL_DMIC(x) (((x) & 0x7) << 5)
+#define SOF_IPC4_NODE_INDEX_INTEL_DMIC(x) ((x) & 0x7)
 
 #define SOF_IPC4_GAIN_ALL_CHANNELS_MASK 0xffffffff
 #define SOF_IPC4_VOL_ZERO_DB	0x7fffffff
@@ -279,14 +279,16 @@ struct sof_ipc4_control_data {
  * @init_val: Initial value
  * @curve_type: Curve type
  * @reserved: reserved for future use
- * @curve_duration: Curve duration
+ * @curve_duration_l: Curve duration low part
+ * @curve_duration_h: Curve duration high part
  */
 struct sof_ipc4_gain_data {
 	uint32_t channels;
 	uint32_t init_val;
 	uint32_t curve_type;
 	uint32_t reserved;
-	uint32_t curve_duration;
+	uint32_t curve_duration_l;
+	uint32_t curve_duration_h;
 } __aligned(8);
 
 /**
diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c
index 760621bfc802..6de388a8d0b8 100644
--- a/sound/soc/sof/sof-audio.c
+++ b/sound/soc/sof/sof-audio.c
@@ -50,9 +50,27 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev,
 	/* reset route setup status for all routes that contain this widget */
 	sof_reset_route_setup_status(sdev, swidget);
 
+	/* free DAI config and continue to free widget even if it fails */
+	if (WIDGET_IS_DAI(swidget->id)) {
+		struct snd_sof_dai_config_data data;
+		unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_FREE;
+
+		data.dai_data = DMA_CHAN_INVALID;
+
+		if (tplg_ops && tplg_ops->dai_config) {
+			err = tplg_ops->dai_config(sdev, swidget, flags, &data);
+			if (err < 0)
+				dev_err(sdev->dev, "failed to free config for widget %s\n",
+					swidget->widget->name);
+		}
+	}
+
 	/* continue to disable core even if IPC fails */
-	if (tplg_ops && tplg_ops->widget_free)
-		err = tplg_ops->widget_free(sdev, swidget);
+	if (tplg_ops && tplg_ops->widget_free) {
+		ret = tplg_ops->widget_free(sdev, swidget);
+		if (ret < 0 && !err)
+			err = ret;
+	}
 
 	/*
 	 * disable widget core. continue to route setup status and complete flag
@@ -151,8 +169,12 @@ static int sof_widget_setup_unlocked(struct snd_sof_dev *sdev,
 
 	/* send config for DAI components */
 	if (WIDGET_IS_DAI(swidget->id)) {
-		unsigned int flags = SOF_DAI_CONFIG_FLAGS_NONE;
+		unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_PARAMS;
 
+		/*
+		 * The config flags saved during BE DAI hw_params will be used for IPC3. IPC4 does
+		 * not use the flags argument.
+		 */
 		if (tplg_ops && tplg_ops->dai_config) {
 			ret = tplg_ops->dai_config(sdev, swidget, flags, NULL);
 			if (ret < 0)
@@ -588,8 +610,8 @@ int sof_widget_list_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm,
 	ret = sof_walk_widgets_in_order(sdev, spcm, fe_params, platform_params,
 					dir, SOF_WIDGET_SETUP);
 	if (ret < 0) {
-		ret = sof_walk_widgets_in_order(sdev, spcm, fe_params, platform_params,
-						dir, SOF_WIDGET_UNPREPARE);
+		sof_walk_widgets_in_order(sdev, spcm, fe_params, platform_params,
+					  dir, SOF_WIDGET_UNPREPARE);
 		return ret;
 	}
 
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index 4a62ccc71fcb..9f3a038fe21a 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -1388,14 +1388,15 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
 	if (ret < 0) {
 		dev_err(scomp->dev, "failed to parse component pin tokens for %s\n",
 			w->name);
-		return ret;
+		goto widget_free;
 	}
 
 	if (swidget->num_sink_pins > SOF_WIDGET_MAX_NUM_PINS ||
 	    swidget->num_source_pins > SOF_WIDGET_MAX_NUM_PINS) {
 		dev_err(scomp->dev, "invalid pins for %s: [sink: %d, src: %d]\n",
 			swidget->widget->name, swidget->num_sink_pins, swidget->num_source_pins);
-		return -EINVAL;
+		ret = -EINVAL;
+		goto widget_free;
 	}
 
 	if (swidget->num_sink_pins > 1) {
@@ -1404,7 +1405,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
 		if (ret < 0) {
 			dev_err(scomp->dev, "failed to parse sink pin binding for %s\n",
 				w->name);
-			return ret;
+			goto widget_free;
 		}
 	}
 
@@ -1414,7 +1415,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
 		if (ret < 0) {
 			dev_err(scomp->dev, "failed to parse source pin binding for %s\n",
 				w->name);
-			return ret;
+			goto widget_free;
 		}
 	}
 
@@ -1436,9 +1437,8 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
 	case snd_soc_dapm_dai_out:
 		dai = kzalloc(sizeof(*dai), GFP_KERNEL);
 		if (!dai) {
-			kfree(swidget);
-			return -ENOMEM;
-
+			ret = -ENOMEM;
+			goto widget_free;
 		}
 
 		ret = sof_widget_parse_tokens(scomp, swidget, tw, token_list, token_list_size);
@@ -1496,8 +1496,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
 			tw->shift, swidget->id, tw->name,
 			strnlen(tw->sname, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) > 0
 				? tw->sname : "none");
-		kfree(swidget);
-		return ret;
+		goto widget_free;
 	}
 
 	if (sof_debug_check_flag(SOF_DBG_DISABLE_MULTICORE)) {
@@ -1518,10 +1517,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
 			if (ret) {
 				dev_err(scomp->dev, "widget event binding failed for %s\n",
 					swidget->widget->name);
-				kfree(swidget->private);
-				kfree(swidget->tuples);
-				kfree(swidget);
-				return ret;
+				goto free;
 			}
 		}
 	}
@@ -1532,10 +1528,8 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
 
 		spipe = kzalloc(sizeof(*spipe), GFP_KERNEL);
 		if (!spipe) {
-			kfree(swidget->private);
-			kfree(swidget->tuples);
-			kfree(swidget);
-			return -ENOMEM;
+			ret = -ENOMEM;
+			goto free;
 		}
 
 		spipe->pipe_widget = swidget;
@@ -1546,6 +1540,12 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index,
 	w->dobj.private = swidget;
 	list_add(&swidget->list, &sdev->widget_list);
 	return ret;
+free:
+	kfree(swidget->private);
+	kfree(swidget->tuples);
+widget_free:
+	kfree(swidget);
+	return ret;
 }
 
 static int sof_route_unload(struct snd_soc_component *scomp,