Merge series "ASoC: Intel: machine driver updates for 5.9" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:

Small patchset to harden the SoundWire machine driver, change bad
HIDs, update PLL settings and avoid memory leaks. Given that the
SoundWire core parts are not upstream it's probably not necessary to
provide the patches to stable branches.

Bard Liao (1):
  ASoC: Intel: sof_sdw_rt711: remove hard-coded codec name

Kai Vehmanen (2):
  ASoC: Intel: sof_sdw: add support for systems without i915 audio
  ASoC: Intel: sof_sdw: avoid crash if invalid DSP topology loaded

Libin Yang (1):
  ASoC: Intel: common: change match table ehl-rt5660

Pierre-Louis Bossart (1):
  ASoC: Intel: sof_sdw_rt711: remove properties in card remove

Yong Zhi (1):
  ASoC: intel: board: sof_rt5682: Update rt1015 pll input clk freq

 sound/soc/intel/boards/sof_rt5682.c           |  9 +++++-
 sound/soc/intel/boards/sof_sdw.c              | 31 +++++++++++++------
 sound/soc/intel/boards/sof_sdw_common.h       |  2 ++
 sound/soc/intel/boards/sof_sdw_hdmi.c         |  6 ++++
 sound/soc/intel/boards/sof_sdw_rt711.c        | 17 +++++++++-
 .../intel/common/soc-acpi-intel-ehl-match.c   |  2 +-
 6 files changed, 54 insertions(+), 13 deletions(-)

base-commit: 22e9b54307987787efa0ee534aa9e31982ec1161
--
2.25.1

15 files changed, 98 insertions, 33 deletions

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 7ff659e28570..776a60529e70 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -161,6 +161,7 @@ void snd_soc_dai_resume(struct snd_soc_dai *dai);
 int snd_soc_dai_compress_new(struct snd_soc_dai *dai,
 			     struct snd_soc_pcm_runtime *rtd, int num);
 bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream);
+void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link);
 void snd_soc_dai_action(struct snd_soc_dai *dai,
 			int stream, int action);
 static inline void snd_soc_dai_activate(struct snd_soc_dai *dai,
diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c
index 67b5faa64ec3..929bb1798c43 100644
--- a/sound/soc/codecs/max98373.c
+++ b/sound/soc/codecs/max98373.c
@@ -331,13 +331,6 @@ static int max98373_probe(struct snd_soc_component *component)
 	regmap_write(max98373->regmap,
 		MAX98373_R202A_PCM_TO_SPK_MONO_MIX_2,
 		0x1);
-	/* Set inital volume (0dB) */
-	regmap_write(max98373->regmap,
-		MAX98373_R203D_AMP_DIG_VOL_CTRL,
-		0x00);
-	regmap_write(max98373->regmap,
-		MAX98373_R203E_AMP_PATH_GAIN,
-		0x00);
 	/* Enable DC blocker */
 	regmap_write(max98373->regmap,
 		MAX98373_R203F_AMP_DSP_CFG,
@@ -397,7 +390,6 @@ const struct snd_soc_component_driver soc_codec_dev_max98373 = {
 	.num_dapm_widgets	= ARRAY_SIZE(max98373_dapm_widgets),
 	.dapm_routes		= max98373_audio_map,
 	.num_dapm_routes	= ARRAY_SIZE(max98373_audio_map),
-	.idle_bias_on		= 1,
 	.use_pmdown_time	= 1,
 	.endianness		= 1,
 	.non_legacy_dai_naming	= 1,
diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c
index 89b1c8b68004..5fb9653d9131 100644
--- a/sound/soc/codecs/rt286.c
+++ b/sound/soc/codecs/rt286.c
@@ -272,13 +272,13 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic)
 		regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf);
 		*mic = buf & 0x80000000;
 	}
-	if (!*mic) {
+
+	if (!*hp) {
 		snd_soc_dapm_disable_pin(dapm, "HV");
 		snd_soc_dapm_disable_pin(dapm, "VREF");
-	}
-	if (!*hp)
 		snd_soc_dapm_disable_pin(dapm, "LDO1");
-	snd_soc_dapm_sync(dapm);
+		snd_soc_dapm_sync(dapm);
+	}
 
 	return 0;
 }
diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c
index fab066a75ce0..a4713bd6508d 100644
--- a/sound/soc/codecs/rt5682.c
+++ b/sound/soc/codecs/rt5682.c
@@ -970,13 +970,12 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert)
 		rt5682_enable_push_button_irq(component, false);
 		snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1,
 			RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW);
-		if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
+		if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS"))
 			snd_soc_component_update_bits(component,
-				RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
-		else
+				RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0);
+		if (!snd_soc_dapm_get_pin_status(dapm, "Vref2"))
 			snd_soc_component_update_bits(component,
-				RT5682_PWR_ANLG_1,
-				RT5682_PWR_VREF2 | RT5682_PWR_MB, 0);
+				RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
 		snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
 			RT5682_PWR_CBJ, 0);
 		snd_soc_component_update_bits(component, RT5682_MICBIAS_2,
@@ -1088,7 +1087,8 @@ void rt5682_jack_detect_handler(struct work_struct *work)
 			/* jack was out, report jack type */
 			rt5682->jack_type =
 				rt5682_headset_detect(rt5682->component, 1);
-		} else {
+		} else if ((rt5682->jack_type & SND_JACK_HEADSET) ==
+			SND_JACK_HEADSET) {
 			/* jack is already in, report button event */
 			rt5682->jack_type = SND_JACK_HEADSET;
 			btn_type = rt5682_button_detect(rt5682->component);
@@ -1614,8 +1614,7 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = {
 		0, set_filter_clk, SND_SOC_DAPM_PRE_PMU),
 	SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0,
 		rt5682_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
-	SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0,
-		NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Vref2", SND_SOC_NOPM, 0, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0),
 
 	/* ASRC */
@@ -2505,6 +2504,15 @@ static int rt5682_wclk_prepare(struct clk_hw *hw)
 	snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS");
 	snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
 				RT5682_PWR_MB, RT5682_PWR_MB);
+
+	snd_soc_dapm_force_enable_pin_unlocked(dapm, "Vref2");
+	snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+			RT5682_PWR_VREF2 | RT5682_PWR_FV2,
+			RT5682_PWR_VREF2);
+	usleep_range(55000, 60000);
+	snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+			RT5682_PWR_FV2, RT5682_PWR_FV2);
+
 	snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1");
 	snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F");
 	snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B");
@@ -2530,9 +2538,12 @@ static void rt5682_wclk_unprepare(struct clk_hw *hw)
 	snd_soc_dapm_mutex_lock(dapm);
 
 	snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS");
+	snd_soc_dapm_disable_pin_unlocked(dapm, "Vref2");
 	if (!rt5682->jack_type)
 		snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1,
+				RT5682_PWR_VREF2 | RT5682_PWR_FV2 |
 				RT5682_PWR_MB, 0);
+
 	snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1");
 	snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F");
 	snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B");
diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c
index 89c6507d5566..c86231dfcf4f 100644
--- a/sound/soc/codecs/wm8974.c
+++ b/sound/soc/codecs/wm8974.c
@@ -186,7 +186,7 @@ SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 0),
 
 /* Boost mixer */
 static const struct snd_kcontrol_new wm8974_boost_mixer[] = {
-SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 0),
+SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 1),
 };
 
 /* Input PGA */
@@ -474,6 +474,10 @@ static int wm8974_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		iface |= 0x0008;
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
+		if ((fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_IB_IF ||
+		    (fmt & SND_SOC_DAIFMT_INV_MASK) == SND_SOC_DAIFMT_NB_IF) {
+			return -EINVAL;
+		}
 		iface |= 0x00018;
 		break;
 	default:
diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c
index 9ad35d9940fe..97b4f5480a31 100644
--- a/sound/soc/generic/audio-graph-card.c
+++ b/sound/soc/generic/audio-graph-card.c
@@ -317,8 +317,8 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 	if (ret < 0)
 		goto out_put_node;
 
-	dai_link->dpcm_playback		= 1;
-	dai_link->dpcm_capture		= 1;
+	snd_soc_dai_link_set_capabilities(dai_link);
+
 	dai_link->ops			= &graph_ops;
 	dai_link->init			= asoc_simple_dai_init;
 
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 55e9f8800b3e..04d4d28ed511 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -231,8 +231,8 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv,
 	if (ret < 0)
 		goto out_put_node;
 
-	dai_link->dpcm_playback		= 1;
-	dai_link->dpcm_capture		= 1;
+	snd_soc_dai_link_set_capabilities(dai_link);
+
 	dai_link->ops			= &simple_ops;
 	dai_link->init			= asoc_simple_dai_init;
 
diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c
index c9da91147770..725304779426 100644
--- a/sound/soc/intel/boards/bdw-rt5677.c
+++ b/sound/soc/intel/boards/bdw-rt5677.c
@@ -367,6 +367,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = {
 	{
 		.name = "Codec DSP",
 		.stream_name = "Wake on Voice",
+		.capture_only = 1,
 		.ops = &bdw_rt5677_dsp_ops,
 		SND_SOC_DAILINK_REG(dsp),
 	},
diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c
index 71b39e579af9..414ae4bb5224 100644
--- a/sound/soc/intel/boards/bytcht_es8316.c
+++ b/sound/soc/intel/boards/bytcht_es8316.c
@@ -552,8 +552,10 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev)
 
 	if (cnt) {
 		ret = device_add_properties(codec_dev, props);
-		if (ret)
+		if (ret) {
+			put_device(codec_dev);
 			return ret;
+		}
 	}
 
 	devm_acpi_dev_add_driver_gpios(codec_dev, byt_cht_es8316_gpios);
diff --git a/sound/soc/intel/common/soc-acpi-intel-ehl-match.c b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c
index 45e07d886013..badafc1d54d2 100644
--- a/sound/soc/intel/common/soc-acpi-intel-ehl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c
@@ -12,7 +12,7 @@
 
 struct snd_soc_acpi_mach snd_soc_acpi_intel_ehl_machines[] = {
 	{
-		.id = "INTC1027",
+		.id = "10EC5660",
 		.drv_name = "ehl_rt5660",
 		.sof_fw_filename = "sof-ehl.ri",
 		.sof_tplg_filename = "sof-ehl-rt5660.tplg",
diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c
index 89f7f64747cd..47f2d93224fe 100644
--- a/sound/soc/meson/axg-card.c
+++ b/sound/soc/meson/axg-card.c
@@ -116,7 +116,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card,
 
 	lb = &card->dai_link[*index + 1];
 
-	lb->name = kasprintf(GFP_KERNEL, "%s-lb", pad->name);
+	lb->name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-lb", pad->name);
 	if (!lb->name)
 		return -ENOMEM;
 
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index c89a1929d141..98f0c98b06bb 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -393,6 +393,44 @@ bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int dir)
 	return stream->channels_min;
 }
 
+/*
+ * snd_soc_dai_link_set_capabilities() - set dai_link properties based on its DAIs
+ */
+void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link)
+{
+	struct snd_soc_dai_link_component *cpu;
+	struct snd_soc_dai_link_component *codec;
+	struct snd_soc_dai *dai;
+	bool supported[SNDRV_PCM_STREAM_LAST + 1];
+	int direction;
+	int i;
+
+	for_each_pcm_streams(direction) {
+		supported[direction] = true;
+
+		for_each_link_cpus(dai_link, i, cpu) {
+			dai = snd_soc_find_dai(cpu);
+			if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
+				supported[direction] = false;
+				break;
+			}
+		}
+		if (!supported[direction])
+			continue;
+		for_each_link_codecs(dai_link, i, codec) {
+			dai = snd_soc_find_dai(codec);
+			if (!dai || !snd_soc_dai_stream_valid(dai, direction)) {
+				supported[direction] = false;
+				break;
+			}
+		}
+	}
+
+	dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK];
+	dai_link->dpcm_capture  = supported[SNDRV_PCM_STREAM_CAPTURE];
+}
+EXPORT_SYMBOL_GPL(snd_soc_dai_link_set_capabilities);
+
 void snd_soc_dai_action(struct snd_soc_dai *dai,
 			int stream, int action)
 {
diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 339c4930b0c0..adc7c37145d6 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -345,15 +345,15 @@ int snd_sof_device_remove(struct device *dev)
 	struct snd_sof_pdata *pdata = sdev->pdata;
 	int ret;
 
-	ret = snd_sof_dsp_power_down_notify(sdev);
-	if (ret < 0)
-		dev_warn(dev, "error: %d failed to prepare DSP for device removal",
-			 ret);
-
 	if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE))
 		cancel_work_sync(&sdev->probe_work);
 
 	if (sdev->fw_state > SOF_FW_BOOT_NOT_STARTED) {
+		ret = snd_sof_dsp_power_down_notify(sdev);
+		if (ret < 0)
+			dev_warn(dev, "error: %d failed to prepare DSP for device removal",
+				 ret);
+
 		snd_sof_fw_unload(sdev);
 		snd_sof_ipc_free(sdev);
 		snd_sof_free_debug(sdev);
diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index 63f9c20a1bac..a4fa8451d8cb 100644
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -375,6 +375,14 @@ static int imx8_ipc_pcm_params(struct snd_sof_dev *sdev,
 static struct snd_soc_dai_driver imx8_dai[] = {
 {
 	.name = "esai-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 8,
+	},
 },
 };
 
diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c
index fa86a9e2990f..287114a37688 100644
--- a/sound/soc/sof/imx/imx8m.c
+++ b/sound/soc/sof/imx/imx8m.c
@@ -240,6 +240,14 @@ static int imx8m_ipc_pcm_params(struct snd_sof_dev *sdev,
 static struct snd_soc_dai_driver imx8m_dai[] = {
 {
 	.name = "sai-port",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 32,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 32,
+	},
 },
 };