Merge tag 'cpuidle_psci-v5.5-rc4' of git://git.linaro.org/people/ulf.hansson/linux-pm into arm/drivers

Initial support for hierarchical CPU arrangement, managed by PSCI and its
corresponding cpuidle driver. This support is based upon using the generic
PM domain, which already supports devices belonging to CPUs.

Finally, these is a DTS patch that enables the hierarchical topology to be
used for the Qcom 410c Dragonboard, which supports the PSCI OS-initiated
mode.

* tag 'cpuidle_psci-v5.5-rc4' of git://git.linaro.org/people/ulf.hansson/linux-pm: (611 commits)
  arm64: dts: Convert to the hierarchical CPU topology layout for MSM8916
  cpuidle: psci: Add support for PM domains by using genpd
  PM / Domains: Introduce a genpd OF helper that removes a subdomain
  cpuidle: psci: Support CPU hotplug for the hierarchical model
  cpuidle: psci: Manage runtime PM in the idle path
  cpuidle: psci: Prepare to use OS initiated suspend mode via PM domains
  cpuidle: psci: Attach CPU devices to their PM domains
  cpuidle: psci: Add a helper to attach a CPU to its PM domain
  cpuidle: psci: Support hierarchical CPU idle states
  cpuidle: psci: Simplify OF parsing of CPU idle state nodes
  cpuidle: dt: Support hierarchical CPU idle states
  of: base: Add of_get_cpu_state_node() to get idle states for a CPU node
  firmware: psci: Export functions to manage the OSI mode
  dt: psci: Update DT bindings to support hierarchical PSCI states
  cpuidle: psci: Align psci_power_state count with idle state count
  Linux 5.5-rc4
  locks: print unsigned ino in /proc/locks
  riscv: export flush_icache_all to modules
  riscv: reject invalid syscalls below -1
  riscv: fix compile failure with EXPORT_SYMBOL() & !MMU
  ...

Link: https://lore.kernel.org/r/20200102160820.3572-1-ulf.hansson@linaro.org
Signed-off-by: Olof Johansson <olof@lixom.net>

23 files changed, 184 insertions, 105 deletions

diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 1fe581167b7b..d083225344a0 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -739,6 +739,10 @@ 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 2f3b7a35f2d9..ba56b59b3e17 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -883,7 +883,7 @@ 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/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);