Merge branch 'for-linus' into for-next

Taking the 5.5 devel branch back into the main devel branch.
A USB-audio fix needs to be adjusted to adapt the changes that have
been formerly applied for stop_sync.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

30 files changed, 223 insertions, 114 deletions

diff --git a/include/sound/soc.h b/include/sound/soc.h
index c28a1ed5e8df..262896799826 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1150,6 +1150,7 @@ struct snd_soc_pcm_runtime {
 	unsigned int num_codecs;
 
 	struct delayed_work delayed_work;
+	void (*close_delayed_work_func)(struct snd_soc_pcm_runtime *rtd);
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_dpcm_root;
 #endif
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 5a1245509eac..c375c41496f8 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -740,6 +740,10 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
 	while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size)
 		runtime->boundary *= 2;
 
+	/* clear the buffer for avoiding possible kernel info leaks */
+	if (runtime->dma_area && !substream->ops->copy_user)
+		memset(runtime->dma_area, 0, runtime->dma_bytes);
+
 	snd_pcm_timer_resolution_change(substream);
 	snd_pcm_set_state(substream, SNDRV_PCM_STATE_SETUP);
 
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index f9707fb05efe..682ed39f79b0 100644
--- a/sound/hda/hdac_stream.c
+++ b/sound/hda/hdac_stream.c
@@ -120,10 +120,8 @@ void snd_hdac_stream_clear(struct hdac_stream *azx_dev)
 	snd_hdac_stream_updateb(azx_dev, SD_CTL,
 				SD_CTL_DMA_START | SD_INT_MASK, 0);
 	snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */
-	if (azx_dev->stripe) {
+	if (azx_dev->stripe)
 		snd_hdac_stream_updateb(azx_dev, SD_CTL_3B, SD_CTL_STRIPE_MASK, 0);
-		azx_dev->stripe = 0;
-	}
 	azx_dev->running = false;
 }
 EXPORT_SYMBOL_GPL(snd_hdac_stream_clear);
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 78005ddb7050..a74c85867eb3 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -835,7 +835,7 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr,
 		return -EAGAIN; /* give a chance to retry */
 	}
 
-	dev_WARN(chip->card->dev,
+	dev_err(chip->card->dev,
 		"azx_get_response timeout, switching to single_cmd mode: last cmd=0x%08x\n",
 		bus->last_cmd[addr]);
 	chip->single_cmd = 1;
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index b7a1abb3e231..32ed46464af7 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1809,13 +1809,14 @@ struct scp_msg {
 
 static void dspio_clear_response_queue(struct hda_codec *codec)
 {
+	unsigned long timeout = jiffies + msecs_to_jiffies(1000);
 	unsigned int dummy = 0;
-	int status = -1;
+	int status;
 
 	/* clear all from the response queue */
 	do {
 		status = dspio_read(codec, &dummy);
-	} while (status == 0);
+	} while (status == 0 && time_before(jiffies, timeout));
 }
 
 static int dspio_get_response_data(struct hda_codec *codec)
@@ -7588,12 +7589,14 @@ static void ca0132_process_dsp_response(struct hda_codec *codec,
 	struct ca0132_spec *spec = codec->spec;
 
 	codec_dbg(codec, "ca0132_process_dsp_response\n");
+	snd_hda_power_up_pm(codec);
 	if (spec->wait_scp) {
 		if (dspio_get_response_data(codec) >= 0)
 			spec->wait_scp = 0;
 	}
 
 	dspio_clear_response_queue(codec);
+	snd_hda_power_down_pm(codec);
 }
 
 static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb)
@@ -7604,11 +7607,10 @@ static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb)
 	/* Delay enabling the HP amp, to let the mic-detection
 	 * state machine run.
 	 */
-	cancel_delayed_work(&spec->unsol_hp_work);
-	schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500));
 	tbl = snd_hda_jack_tbl_get(codec, cb->nid);
 	if (tbl)
 		tbl->block_report = 1;
+	schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(500));
 }
 
 static void amic_callback(struct hda_codec *codec, struct hda_jack_callback *cb)
@@ -8454,12 +8456,25 @@ static void ca0132_reboot_notify(struct hda_codec *codec)
 	codec->patch_ops.free(codec);
 }
 
+#ifdef CONFIG_PM
+static int ca0132_suspend(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	cancel_delayed_work_sync(&spec->unsol_hp_work);
+	return 0;
+}
+#endif
+
 static const struct hda_codec_ops ca0132_patch_ops = {
 	.build_controls = ca0132_build_controls,
 	.build_pcms = ca0132_build_pcms,
 	.init = ca0132_init,
 	.free = ca0132_free,
 	.unsol_event = snd_hda_jack_unsol_event,
+#ifdef CONFIG_PM
+	.suspend = ca0132_suspend,
+#endif
 	.reboot_notify = ca0132_reboot_notify,
 };
 
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 78647ee02339..630b1f5c276d 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -2021,6 +2021,8 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo,
 		per_cvt->assigned = 0;
 		hinfo->nid = 0;
 
+		azx_stream(get_azx_dev(substream))->stripe = 0;
+
 		mutex_lock(&spec->pcm_lock);
 		snd_hda_spdif_ctls_unassign(codec, pcm_idx);
 		clear_bit(pcm_idx, &spec->pcm_in_use);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6d6e34b3b3aa..dbfafee97931 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7643,11 +7643,6 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
 		{0x1a, 0x90a70130},
 		{0x1b, 0x90170110},
 		{0x21, 0x03211020}),
-	SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB,
-		{0x12, 0xb7a60130},
-		{0x13, 0xb8a61140},
-		{0x16, 0x90170110},
-		{0x21, 0x04211020}),
 	SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
 		{0x12, 0x90a60130},
 		{0x14, 0x90170110},
@@ -7841,6 +7836,9 @@ static const struct snd_hda_pin_quirk alc269_fallback_pin_fixup_tbl[] = {
 	SND_HDA_PIN_QUIRK(0x10ec0236, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
 		{0x19, 0x40000000},
 		{0x1a, 0x40000000}),
+	SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB,
+		{0x19, 0x40000000},
+		{0x1a, 0x40000000}),
 	{}
 };
 
diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c
index f4ee6798154a..7a5621e5e233 100644
--- a/sound/soc/amd/acp-da7219-max98357a.c
+++ b/sound/soc/amd/acp-da7219-max98357a.c
@@ -96,14 +96,19 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd)
 	return 0;
 }
 
-static int da7219_clk_enable(struct snd_pcm_substream *substream,
-			     int wclk_rate, int bclk_rate)
+static int da7219_clk_enable(struct snd_pcm_substream *substream)
 {
 	int ret = 0;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 
-	clk_set_rate(da7219_dai_wclk, wclk_rate);
-	clk_set_rate(da7219_dai_bclk, bclk_rate);
+	/*
+	 * Set wclk to 48000 because the rate constraint of this driver is
+	 * 48000. ADAU7002 spec: "The ADAU7002 requires a BCLK rate that is
+	 * minimum of 64x the LRCLK sample rate." DA7219 is the only clk
+	 * source so for all codecs we have to limit bclk to 64X lrclk.
+	 */
+	clk_set_rate(da7219_dai_wclk, 48000);
+	clk_set_rate(da7219_dai_bclk, 48000 * 64);
 	ret = clk_prepare_enable(da7219_dai_bclk);
 	if (ret < 0) {
 		dev_err(rtd->dev, "can't enable master clock %d\n", ret);
@@ -156,7 +161,7 @@ static int cz_da7219_play_startup(struct snd_pcm_substream *substream)
 				   &constraints_rates);
 
 	machine->play_i2s_instance = I2S_SP_INSTANCE;
-	return 0;
+	return da7219_clk_enable(substream);
 }
 
 static int cz_da7219_cap_startup(struct snd_pcm_substream *substream)
@@ -178,7 +183,7 @@ static int cz_da7219_cap_startup(struct snd_pcm_substream *substream)
 
 	machine->cap_i2s_instance = I2S_SP_INSTANCE;
 	machine->capture_channel = CAP_CHANNEL1;
-	return 0;
+	return da7219_clk_enable(substream);
 }
 
 static int cz_max_startup(struct snd_pcm_substream *substream)
@@ -199,7 +204,7 @@ static int cz_max_startup(struct snd_pcm_substream *substream)
 				   &constraints_rates);
 
 	machine->play_i2s_instance = I2S_BT_INSTANCE;
-	return 0;
+	return da7219_clk_enable(substream);
 }
 
 static int cz_dmic0_startup(struct snd_pcm_substream *substream)
@@ -220,7 +225,7 @@ static int cz_dmic0_startup(struct snd_pcm_substream *substream)
 				   &constraints_rates);
 
 	machine->cap_i2s_instance = I2S_BT_INSTANCE;
-	return 0;
+	return da7219_clk_enable(substream);
 }
 
 static int cz_dmic1_startup(struct snd_pcm_substream *substream)
@@ -242,25 +247,7 @@ static int cz_dmic1_startup(struct snd_pcm_substream *substream)
 
 	machine->cap_i2s_instance = I2S_SP_INSTANCE;
 	machine->capture_channel = CAP_CHANNEL0;
-	return 0;
-}
-
-static int cz_da7219_params(struct snd_pcm_substream *substream,
-				      struct snd_pcm_hw_params *params)
-{
-	int wclk, bclk;
-
-	wclk = params_rate(params);
-	bclk = wclk * params_channels(params) *
-		snd_pcm_format_width(params_format(params));
-	/* ADAU7002 spec: "The ADAU7002 requires a BCLK rate
-	 * that is minimum of 64x the LRCLK sample rate."
-	 * DA7219 is the only clk source so for all codecs
-	 * we have to limit bclk to 64X lrclk.
-	 */
-	if (bclk < (wclk * 64))
-		bclk = wclk * 64;
-	return da7219_clk_enable(substream, wclk, bclk);
+	return da7219_clk_enable(substream);
 }
 
 static void cz_da7219_shutdown(struct snd_pcm_substream *substream)
@@ -271,31 +258,26 @@ static void cz_da7219_shutdown(struct snd_pcm_substream *substream)
 static const struct snd_soc_ops cz_da7219_play_ops = {
 	.startup = cz_da7219_play_startup,
 	.shutdown = cz_da7219_shutdown,
-	.hw_params = cz_da7219_params,
 };
 
 static const struct snd_soc_ops cz_da7219_cap_ops = {
 	.startup = cz_da7219_cap_startup,
 	.shutdown = cz_da7219_shutdown,
-	.hw_params = cz_da7219_params,
 };
 
 static const struct snd_soc_ops cz_max_play_ops = {
 	.startup = cz_max_startup,
 	.shutdown = cz_da7219_shutdown,
-	.hw_params = cz_da7219_params,
 };
 
 static const struct snd_soc_ops cz_dmic0_cap_ops = {
 	.startup = cz_dmic0_startup,
 	.shutdown = cz_da7219_shutdown,
-	.hw_params = cz_da7219_params,
 };
 
 static const struct snd_soc_ops cz_dmic1_cap_ops = {
 	.startup = cz_dmic1_startup,
 	.shutdown = cz_da7219_shutdown,
-	.hw_params = cz_da7219_params,
 };
 
 SND_SOC_DAILINK_DEF(designware1,
diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c
index f6bf4cfbea23..e46b6ada13b1 100644
--- a/sound/soc/codecs/max98090.c
+++ b/sound/soc/codecs/max98090.c
@@ -2103,26 +2103,40 @@ static void max98090_pll_det_disable_work(struct work_struct *work)
 			    M98090_IULK_MASK, 0);
 }
 
-static void max98090_pll_work(struct work_struct *work)
+static void max98090_pll_work(struct max98090_priv *max98090)
 {
-	struct max98090_priv *max98090 =
-		container_of(work, struct max98090_priv, pll_work);
 	struct snd_soc_component *component = max98090->component;
+	unsigned int pll;
+	int i;
 
 	if (!snd_soc_component_is_active(component))
 		return;
 
 	dev_info_ratelimited(component->dev, "PLL unlocked\n");
 
+	/*
+	 * As the datasheet suggested, the maximum PLL lock time should be
+	 * 7 msec.  The workaround resets the codec softly by toggling SHDN
+	 * off and on if PLL failed to lock for 10 msec.  Notably, there is
+	 * no suggested hold time for SHDN off.
+	 */
+
 	/* Toggle shutdown OFF then ON */
 	snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN,
 			    M98090_SHDNN_MASK, 0);
-	msleep(10);
 	snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN,
 			    M98090_SHDNN_MASK, M98090_SHDNN_MASK);
 
-	/* Give PLL time to lock */
-	msleep(10);
+	for (i = 0; i < 10; ++i) {
+		/* Give PLL time to lock */
+		usleep_range(1000, 1200);
+
+		/* Check lock status */
+		pll = snd_soc_component_read32(
+				component, M98090_REG_DEVICE_STATUS);
+		if (!(pll & M98090_ULK_MASK))
+			break;
+	}
 }
 
 static void max98090_jack_work(struct work_struct *work)
@@ -2259,7 +2273,7 @@ static irqreturn_t max98090_interrupt(int irq, void *data)
 
 	if (active & M98090_ULK_MASK) {
 		dev_dbg(component->dev, "M98090_ULK_MASK\n");
-		schedule_work(&max98090->pll_work);
+		max98090_pll_work(max98090);
 	}
 
 	if (active & M98090_JDET_MASK) {
@@ -2422,7 +2436,6 @@ static int max98090_probe(struct snd_soc_component *component)
 			  max98090_pll_det_enable_work);
 	INIT_WORK(&max98090->pll_det_disable_work,
 		  max98090_pll_det_disable_work);
-	INIT_WORK(&max98090->pll_work, max98090_pll_work);
 
 	/* Enable jack detection */
 	snd_soc_component_write(component, M98090_REG_JACK_DETECT,
@@ -2475,7 +2488,6 @@ static void max98090_remove(struct snd_soc_component *component)
 	cancel_delayed_work_sync(&max98090->jack_work);
 	cancel_delayed_work_sync(&max98090->pll_det_enable_work);
 	cancel_work_sync(&max98090->pll_det_disable_work);
-	cancel_work_sync(&max98090->pll_work);
 	max98090->component = NULL;
 }
 
diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h
index 57965cd678b4..a197114b0dad 100644
--- a/sound/soc/codecs/max98090.h
+++ b/sound/soc/codecs/max98090.h
@@ -1530,7 +1530,6 @@ struct max98090_priv {
 	struct delayed_work jack_work;
 	struct delayed_work pll_det_enable_work;
 	struct work_struct pll_det_disable_work;
-	struct work_struct pll_work;
 	struct snd_soc_jack *jack;
 	unsigned int dai_fmt;
 	int tdm_slots;
diff --git a/sound/soc/codecs/rt5677-spi.h b/sound/soc/codecs/rt5677-spi.h
index 3af36ec928e9..088b77931727 100644
--- a/sound/soc/codecs/rt5677-spi.h
+++ b/sound/soc/codecs/rt5677-spi.h
@@ -9,9 +9,25 @@
 #ifndef __RT5677_SPI_H__
 #define __RT5677_SPI_H__
 
+#if IS_ENABLED(CONFIG_SND_SOC_RT5677_SPI)
 int rt5677_spi_read(u32 addr, void *rxbuf, size_t len);
 int rt5677_spi_write(u32 addr, const void *txbuf, size_t len);
 int rt5677_spi_write_firmware(u32 addr, const struct firmware *fw);
 void rt5677_spi_hotword_detected(void);
+#else
+static inline int rt5677_spi_read(u32 addr, void *rxbuf, size_t len)
+{
+	return -EINVAL;
+}
+static inline int rt5677_spi_write(u32 addr, const void *txbuf, size_t len)
+{
+	return -EINVAL;
+}
+static inline int rt5677_spi_write_firmware(u32 addr, const struct firmware *fw)
+{
+	return -EINVAL;
+}
+static inline void rt5677_spi_hotword_detected(void){}
+#endif
 
 #endif /* __RT5677_SPI_H__ */
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index b1713fffa3eb..ae6f6121bc1b 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -73,6 +73,7 @@ struct rt5682_priv {
 static const struct reg_sequence patch_list[] = {
 	{RT5682_HP_IMP_SENS_CTRL_19, 0x1000},
 	{RT5682_DAC_ADC_DIG_VOL1, 0xa020},
+	{RT5682_I2C_CTRL, 0x000f},
 };
 
 static const struct reg_default rt5682_reg[] = {
@@ -2474,6 +2475,7 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682)
 	mutex_lock(&rt5682->calibrate_mutex);
 
 	rt5682_reset(rt5682->regmap);
+	regmap_write(rt5682->regmap, RT5682_I2C_CTRL, 0x000f);
 	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xa2af);
 	usleep_range(15000, 20000);
 	regmap_write(rt5682->regmap, RT5682_PWR_ANLG_1, 0xf2af);
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 7d7ea15d73e0..5ffbaddd6e49 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1806,6 +1806,12 @@ static int wm8904_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 
 	switch (clk_id) {
 	case WM8904_CLK_AUTO:
+		/* We don't have any rate constraints, so just ignore the
+		 * request to disable constraining.
+		 */
+		if (!freq)
+			return 0;
+
 		mclk_freq = clk_get_rate(priv->mclk);
 		/* enable FLL if a different sysclk is desired */
 		if (mclk_freq != freq) {
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 3e5c69fbc33a..d9d59f45833f 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2788,7 +2788,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
 
 	if (target % Fref == 0) {
 		fll_div->theta = 0;
-		fll_div->lambda = 0;
+		fll_div->lambda = 1;
 	} else {
 		gcd_fll = gcd(target, fratio * Fref);
 
@@ -2858,7 +2858,7 @@ static int wm8962_set_fll(struct snd_soc_component *component, int fll_id, int s
 		return -EINVAL;
 	}
 
-	if (fll_div.theta || fll_div.lambda)
+	if (fll_div.theta)
 		fll1 |= WM8962_FLL_FRAC;
 
 	/* Stop the FLL while we reconfigure */
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 10b82bf043d1..55e9f8800b3e 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -371,6 +371,7 @@ static int simple_for_each_link(struct asoc_simple_priv *priv,
 	do {
 		struct asoc_simple_data adata;
 		struct device_node *codec;
+		struct device_node *plat;
 		struct device_node *np;
 		int num = of_get_child_count(node);
 
@@ -381,6 +382,9 @@ static int simple_for_each_link(struct asoc_simple_priv *priv,
 			ret = -ENODEV;
 			goto error;
 		}
+		/* get platform */
+		plat = of_get_child_by_name(node, is_top ?
+					    PREFIX "plat" : "plat");
 
 		/* get convert-xxx property */
 		memset(&adata, 0, sizeof(adata));
@@ -389,6 +393,8 @@ static int simple_for_each_link(struct asoc_simple_priv *priv,
 
 		/* loop for all CPU/Codec node */
 		for_each_child_of_node(node, np) {
+			if (plat == np)
+				continue;
 			/*
 			 * It is DPCM
 			 * if it has many CPUs,
diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c
index fbecbb74350b..68bcec5241f7 100644
--- a/sound/soc/intel/atom/sst/sst.c
+++ b/sound/soc/intel/atom/sst/sst.c
@@ -14,6 +14,7 @@
 #include <linux/module.h>
 #include <linux/fs.h>
 #include <linux/interrupt.h>
+#include <linux/io.h>
 #include <linux/firmware.h>
 #include <linux/pm_runtime.h>
 #include <linux/pm_qos.h>
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index dd2b5ad08659..243f683bc02a 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -707,13 +707,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_MCLK_EN),
 	},
 	{
+		/* Teclast X89 */
 		.matches = {
 			DMI_MATCH(DMI_BOARD_VENDOR, "TECLAST"),
 			DMI_MATCH(DMI_BOARD_NAME, "tPAD"),
 		},
 		.driver_data = (void *)(BYT_RT5640_IN3_MAP |
-					BYT_RT5640_MCLK_EN |
-					BYT_RT5640_SSP0_AIF1),
+					BYT_RT5640_JD_SRC_JD1_IN4P |
+					BYT_RT5640_OVCD_TH_2000UA |
+					BYT_RT5640_OVCD_SF_1P0 |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
 	},
 	{	/* Toshiba Satellite Click Mini L9W-B */
 		.matches = {
diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
index 5d08ae066738..fb9ba8819706 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
@@ -9,45 +9,52 @@
 #include <sound/soc-acpi.h>
 #include <sound/soc-acpi-intel-match.h>
 
-static struct snd_soc_acpi_codecs cml_codecs = {
+static struct snd_soc_acpi_codecs rt1011_spk_codecs = {
 	.num_codecs = 1,
-	.codecs = {"10EC5682"}
+	.codecs = {"10EC1011"}
 };
 
-static struct snd_soc_acpi_codecs cml_spk_codecs = {
+static struct snd_soc_acpi_codecs max98357a_spk_codecs = {
 	.num_codecs = 1,
 	.codecs = {"MX98357A"}
 };
 
+/*
+ * The order of the three entries with .id = "10EC5682" matters
+ * here, because DSDT tables expose an ACPI HID for the MAX98357A
+ * speaker amplifier which is not populated on the board.
+ */
 struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = {
 	{
-		.id = "DLGS7219",
-		.drv_name = "cml_da7219_max98357a",
-		.quirk_data = &cml_spk_codecs,
+		.id = "10EC5682",
+		.drv_name = "cml_rt1011_rt5682",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &rt1011_spk_codecs,
 		.sof_fw_filename = "sof-cml.ri",
-		.sof_tplg_filename = "sof-cml-da7219-max98357a.tplg",
+		.sof_tplg_filename = "sof-cml-rt1011-rt5682.tplg",
 	},
 	{
-		.id = "MX98357A",
+		.id = "10EC5682",
 		.drv_name = "sof_rt5682",
-		.quirk_data = &cml_codecs,
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &max98357a_spk_codecs,
 		.sof_fw_filename = "sof-cml.ri",
 		.sof_tplg_filename = "sof-cml-rt5682-max98357a.tplg",
 	},
 	{
-		.id = "10EC1011",
-		.drv_name = "cml_rt1011_rt5682",
-		.quirk_data = &cml_codecs,
-		.sof_fw_filename = "sof-cml.ri",
-		.sof_tplg_filename = "sof-cml-rt1011-rt5682.tplg",
-	},
-	{
 		.id = "10EC5682",
 		.drv_name = "sof_rt5682",
 		.sof_fw_filename = "sof-cml.ri",
 		.sof_tplg_filename = "sof-cml-rt5682.tplg",
 	},
-
+	{
+		.id = "DLGS7219",
+		.drv_name = "cml_da7219_max98357a",
+		.machine_quirk = snd_soc_acpi_codec_list,
+		.quirk_data = &max98357a_spk_codecs,
+		.sof_fw_filename = "sof-cml.ri",
+		.sof_tplg_filename = "sof-cml-da7219-max98357a.tplg",
+	},
 	{},
 };
 EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cml_machines);
diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c
index 61f230324164..6615ef64c7f5 100644
--- a/sound/soc/soc-compress.c
+++ b/sound/soc/soc-compress.c
@@ -214,10 +214,8 @@ be_err:
  * This is to ensure there are no pops or clicks in between any music tracks
  * due to DAPM power cycling.
  */
-static void close_delayed_work(struct work_struct *work)
+static void close_delayed_work(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_pcm_runtime *rtd =
-			container_of(work, struct snd_soc_pcm_runtime, delayed_work.work);
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 
 	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
@@ -929,7 +927,7 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num)
 	}
 
 	/* DAPM dai link stream work */
-	INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
+	rtd->close_delayed_work_func = close_delayed_work;
 
 	rtd->compr = compr;
 	compr->private_data = rtd;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 062653ab03a3..1c84ff1a5bf9 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -419,7 +419,8 @@ static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd)
 
 	list_del(&rtd->list);
 
-	flush_delayed_work(&rtd->delayed_work);
+	if (delayed_work_pending(&rtd->delayed_work))
+		flush_delayed_work(&rtd->delayed_work);
 	snd_soc_pcm_component_free(rtd);
 
 	/*
@@ -435,6 +436,15 @@ static void soc_free_pcm_runtime(struct snd_soc_pcm_runtime *rtd)
 	device_unregister(rtd->dev);
 }
 
+static void close_delayed_work(struct work_struct *work) {
+	struct snd_soc_pcm_runtime *rtd =
+			container_of(work, struct snd_soc_pcm_runtime,
+				     delayed_work.work);
+
+	if (rtd->close_delayed_work_func)
+		rtd->close_delayed_work_func(rtd);
+}
+
 static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
 	struct snd_soc_card *card, struct snd_soc_dai_link *dai_link)
 {
@@ -470,6 +480,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime(
 
 	rtd->dev = dev;
 	dev_set_drvdata(dev, rtd);
+	INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
 
 	/*
 	 * for rtd->codec_dais
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 76b7ee637e86..01e7bc03d92f 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -637,10 +637,8 @@ out:
  * This is to ensure there are no pops or clicks in between any music tracks
  * due to DAPM power cycling.
  */
-static void close_delayed_work(struct work_struct *work)
+static void close_delayed_work(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_pcm_runtime *rtd =
-			container_of(work, struct snd_soc_pcm_runtime, delayed_work.work);
 	struct snd_soc_dai *codec_dai = rtd->codec_dais[0];
 
 	mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
@@ -660,7 +658,7 @@ static void close_delayed_work(struct work_struct *work)
 	mutex_unlock(&rtd->card->pcm_mutex);
 }
 
-static void codec2codec_close_delayed_work(struct work_struct *work)
+static void codec2codec_close_delayed_work(struct snd_soc_pcm_runtime *rtd)
 {
 	/*
 	 * Currently nothing to do for c2c links
@@ -2974,10 +2972,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 
 	/* DAPM dai link stream work */
 	if (rtd->dai_link->params)
-		INIT_DELAYED_WORK(&rtd->delayed_work,
-				  codec2codec_close_delayed_work);
+		rtd->close_delayed_work_func = codec2codec_close_delayed_work;
 	else
-		INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work);
+		rtd->close_delayed_work_func = close_delayed_work;
 
 	pcm->nonatomic = rtd->dai_link->nonatomic;
 	rtd->pcm = pcm;
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 81d2af000a5c..b28613149b0c 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -1933,11 +1933,13 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg,
 	ret = soc_tplg_dai_link_load(tplg, link, NULL);
 	if (ret < 0) {
 		dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n");
-		kfree(link->name);
-		kfree(link->stream_name);
-		kfree(link->cpus->dai_name);
-		kfree(link);
-		return ret;
+		goto err;
+	}
+
+	ret = snd_soc_add_dai_link(tplg->comp->card, link);
+	if (ret < 0) {
+		dev_err(tplg->comp->dev, "ASoC: adding FE link failed\n");
+		goto err;
 	}
 
 	link->dobj.index = tplg->index;
@@ -1945,8 +1947,13 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg,
 	link->dobj.type = SND_SOC_DOBJ_DAI_LINK;
 	list_add(&link->dobj.list, &tplg->comp->dobj_list);
 
-	snd_soc_add_dai_link(tplg->comp->card, link);
 	return 0;
+err:
+	kfree(link->name);
+	kfree(link->stream_name);
+	kfree(link->cpus->dai_name);
+	kfree(link);
+	return ret;
 }
 
 /* create a FE DAI and DAI link from the PCM object */
@@ -2039,6 +2046,7 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
 	int size;
 	int i;
 	bool abi_match;
+	int ret;
 
 	count = le32_to_cpu(hdr->count);
 
@@ -2080,7 +2088,12 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
 		}
 
 		/* create the FE DAIs and DAI links */
-		soc_tplg_pcm_create(tplg, _pcm);
+		ret = soc_tplg_pcm_create(tplg, _pcm);
+		if (ret < 0) {
+			if (!abi_match)
+				kfree(_pcm);
+			return ret;
+		}
 
 		/* offset by version-specific struct size and
 		 * real priv data size
diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c
index 2abf80b3eb52..92ef6a796fd5 100644
--- a/sound/soc/sof/intel/byt.c
+++ b/sound/soc/sof/intel/byt.c
@@ -24,7 +24,8 @@
 #define DRAM_OFFSET		0x100000
 #define DRAM_SIZE		(160 * 1024)
 #define SHIM_OFFSET		0x140000
-#define SHIM_SIZE		0x100
+#define SHIM_SIZE_BYT		0x100
+#define SHIM_SIZE_CHT		0x118
 #define MBOX_OFFSET		0x144000
 #define MBOX_SIZE		0x1000
 #define EXCEPT_OFFSET		0x800
@@ -75,7 +76,7 @@ static const struct snd_sof_debugfs_map byt_debugfs[] = {
 	 SOF_DEBUGFS_ACCESS_D0_ONLY},
 	{"dram", BYT_DSP_BAR, DRAM_OFFSET, DRAM_SIZE,
 	 SOF_DEBUGFS_ACCESS_D0_ONLY},
-	{"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE,
+	{"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE_BYT,
 	 SOF_DEBUGFS_ACCESS_ALWAYS},
 };
 
@@ -102,7 +103,7 @@ static const struct snd_sof_debugfs_map cht_debugfs[] = {
 	 SOF_DEBUGFS_ACCESS_D0_ONLY},
 	{"dram", BYT_DSP_BAR, DRAM_OFFSET, DRAM_SIZE,
 	 SOF_DEBUGFS_ACCESS_D0_ONLY},
-	{"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE,
+	{"shim", BYT_DSP_BAR, SHIM_OFFSET, SHIM_SIZE_CHT,
 	 SOF_DEBUGFS_ACCESS_ALWAYS},
 };
 
@@ -145,33 +146,33 @@ static void byt_dump(struct snd_sof_dev *sdev, u32 flags)
 	struct sof_ipc_dsp_oops_xtensa xoops;
 	struct sof_ipc_panic_info panic_info;
 	u32 stack[BYT_STACK_DUMP_SIZE];
-	u32 status, panic, imrd, imrx;
+	u64 status, panic, imrd, imrx;
 
 	/* now try generic SOF status messages */
-	status = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IPCD);
-	panic = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IPCX);
+	status = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IPCD);
+	panic = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IPCX);
 	byt_get_registers(sdev, &xoops, &panic_info, stack,
 			  BYT_STACK_DUMP_SIZE);
 	snd_sof_get_status(sdev, status, panic, &xoops, &panic_info, stack,
 			   BYT_STACK_DUMP_SIZE);
 
 	/* provide some context for firmware debug */
-	imrx = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IMRX);
-	imrd = snd_sof_dsp_read(sdev, BYT_DSP_BAR, SHIM_IMRD);
+	imrx = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IMRX);
+	imrd = snd_sof_dsp_read64(sdev, BYT_DSP_BAR, SHIM_IMRD);
 	dev_err(sdev->dev,
-		"error: ipc host -> DSP: pending %s complete %s raw 0x%8.8x\n",
+		"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);
 	dev_err(sdev->dev,
-		"error: mask host: pending %s complete %s raw 0x%8.8x\n",
+		"error: mask host: pending %s complete %s raw 0x%llx\n",
 		(imrx & SHIM_IMRX_BUSY) ? "yes" : "no",
 		(imrx & SHIM_IMRX_DONE) ? "yes" : "no", imrx);
 	dev_err(sdev->dev,
-		"error: ipc DSP -> host: pending %s complete %s raw 0x%8.8x\n",
+		"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);
 	dev_err(sdev->dev,
-		"error: mask DSP: pending %s complete %s raw 0x%8.8x\n",
+		"error: mask DSP: pending %s complete %s raw 0x%llx\n",
 		(imrd & SHIM_IMRD_BUSY) ? "yes" : "no",
 		(imrd & SHIM_IMRD_DONE) ? "yes" : "no", imrd);
 
diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index 9a9a381a908d..432d12bd4937 100644
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -50,8 +50,7 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset)
 
 	while (ext_hdr->hdr.cmd == SOF_IPC_FW_READY) {
 		/* read in ext structure */
-		offset += sizeof(*ext_hdr);
-		snd_sof_dsp_block_read(sdev, bar, offset,
+		snd_sof_dsp_block_read(sdev, bar, offset + sizeof(*ext_hdr),
 				   (void *)((u8 *)ext_data + sizeof(*ext_hdr)),
 				   ext_hdr->hdr.size - sizeof(*ext_hdr));
 
@@ -61,11 +60,15 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset)
 		/* process structure data */
 		switch (ext_hdr->type) {
 		case SOF_IPC_EXT_DMA_BUFFER:
+			ret = 0;
 			break;
 		case SOF_IPC_EXT_WINDOW:
 			ret = get_ext_windows(sdev, ext_hdr);
 			break;
 		default:
+			dev_warn(sdev->dev, "warning: unknown ext header type %d size 0x%x\n",
+				 ext_hdr->type, ext_hdr->hdr.size);
+			ret = 0;
 			break;
 		}
 
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index d82ab981e840..e20b806ec80f 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -3132,7 +3132,9 @@ found:
 	case SOF_DAI_INTEL_SSP:
 	case SOF_DAI_INTEL_DMIC:
 	case SOF_DAI_INTEL_ALH:
-		/* no resource needs to be released for SSP, DMIC and ALH */
+	case SOF_DAI_IMX_SAI:
+	case SOF_DAI_IMX_ESAI:
+		/* no resource needs to be released for all cases above */
 		break;
 	case SOF_DAI_INTEL_HDA:
 		ret = sof_link_hda_unload(sdev, link);
diff --git a/sound/usb/card.h b/sound/usb/card.h
index 2991b9986f66..395403a2d33f 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -145,6 +145,7 @@ struct snd_usb_substream {
 	struct snd_usb_endpoint *sync_endpoint;
 	unsigned long flags;
 	bool need_setup_ep;		/* (re)configure EP at prepare? */
+	bool need_setup_fmt;		/* (re)configure fmt after resume? */
 	unsigned int speed;		/* USB_SPEED_XXX */
 
 	u64 formats;			/* format bitmasks (all or'ed) */
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 8a52996041e9..0372f6212313 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -523,11 +523,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 	if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting))
 		return -EINVAL;
 
-	if (fmt == subs->cur_audiofmt)
+	if (fmt == subs->cur_audiofmt && !subs->need_setup_fmt)
 		return 0;
 
 	/* close the old interface */
-	if (subs->interface >= 0 && subs->interface != fmt->iface) {
+	if (subs->interface >= 0 && (subs->interface != fmt->iface || subs->need_setup_fmt)) {
 		if (!subs->stream->chip->keep_iface) {
 			err = usb_set_interface(subs->dev, subs->interface, 0);
 			if (err < 0) {
@@ -541,6 +541,9 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 		subs->altset_idx = 0;
 	}
 
+	if (subs->need_setup_fmt)
+		subs->need_setup_fmt = false;
+
 	/* set interface */
 	if (iface->cur_altsetting != alts) {
 		err = snd_usb_select_mode_quirk(subs, fmt);
@@ -1734,6 +1737,13 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea
 		subs->data_endpoint->retire_data_urb = retire_playback_urb;
 		subs->running = 0;
 		return 0;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		if (subs->stream->chip->setup_fmt_after_resume_quirk) {
+			stop_endpoints(subs);
+			subs->need_setup_fmt = true;
+			return 0;
+		}
+		break;
 	}
 
 	return -EINVAL;
@@ -1766,6 +1776,13 @@ static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream
 		subs->data_endpoint->retire_data_urb = retire_capture_urb;
 		subs->running = 1;
 		return 0;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		if (subs->stream->chip->setup_fmt_after_resume_quirk) {
+			stop_endpoints(subs);
+			subs->need_setup_fmt = true;
+			return 0;
+		}
+		break;
 	}
 
 	return -EINVAL;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 70c338f3ae24..d187aa6d50db 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -3466,7 +3466,8 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 		.vendor_name = "Dell",
 		.product_name = "WD19 Dock",
 		.profile_name = "Dell-WD15-Dock",
-		.ifnum = QUIRK_NO_INTERFACE
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_SETUP_FMT_AFTER_RESUME
 	}
 },
 /* MOTU Microbook II */
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 349e1e52996d..a81c2066499f 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -508,6 +508,16 @@ static int create_standard_mixer_quirk(struct snd_usb_audio *chip,
 	return snd_usb_create_mixer(chip, quirk->ifnum, 0);
 }
 
+
+static int setup_fmt_after_resume_quirk(struct snd_usb_audio *chip,
+				       struct usb_interface *iface,
+				       struct usb_driver *driver,
+				       const struct snd_usb_audio_quirk *quirk)
+{
+	chip->setup_fmt_after_resume_quirk = 1;
+	return 1;	/* Continue with creating streams and mixer */
+}
+
 /*
  * audio-interface quirks
  *
@@ -546,6 +556,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
 		[QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk,
 		[QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk,
 		[QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk,
+		[QUIRK_SETUP_FMT_AFTER_RESUME] = setup_fmt_after_resume_quirk,
 	};
 
 	if (quirk->type < QUIRK_TYPE_COUNT) {
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index ff3cbf653de8..6fe3ab582ec6 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -33,7 +33,7 @@ struct snd_usb_audio {
 	wait_queue_head_t shutdown_wait;
 	unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */
 	unsigned int tx_length_quirk:1; /* Put length specifier in transfers */
-	
+	unsigned int setup_fmt_after_resume_quirk:1; /* setup the format to interface after resume */
 	int num_interfaces;
 	int num_suspended_intf;
 	int sample_rate_read_error;
@@ -98,6 +98,7 @@ enum quirk_type {
 	QUIRK_AUDIO_EDIROL_UAXX,
 	QUIRK_AUDIO_ALIGN_TRANSFER,
 	QUIRK_AUDIO_STANDARD_MIXER,
+	QUIRK_SETUP_FMT_AFTER_RESUME,
 
 	QUIRK_TYPE_COUNT
 };