Merge series "ASoC: Intel: machine driver updates for 5.9" from Pierre-Louis Bossart <[e-mail]>:

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

include/sound/soc-dai.h

···
 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,

sound/soc/codecs/max98373.c

···
 	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,
···
 	.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,

sound/soc/codecs/rt286.c

···
 		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;
 }

sound/soc/codecs/rt5682.c

···
 		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_MB, 0);
+		if (!snd_soc_dapm_get_pin_status(dapm, "Vref2"))
 			snd_soc_component_update_bits(component,
 				RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0);
-		else
-			snd_soc_component_update_bits(component,
-				RT5682_PWR_ANLG_1,
-				RT5682_PWR_VREF2 | RT5682_PWR_MB, 0);
 		snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3,
 			RT5682_PWR_CBJ, 0);
 		snd_soc_component_update_bits(component, RT5682_MICBIAS_2,
···
 			/* 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);
···
 		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 */
···
 	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");
···
 	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");

sound/soc/codecs/wm8974.c

···
 
 /* 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 */
···
 		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:

sound/soc/generic/audio-graph-card.c

···
 	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;
 

sound/soc/generic/simple-card.c

···
 	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;
 

sound/soc/intel/boards/bdw-rt5677.c

···
 	{
 		.name = "Codec DSP",
 		.stream_name = "Wake on Voice",
+		.capture_only = 1,
 		.ops = &bdw_rt5677_dsp_ops,
 		SND_SOC_DAILINK_REG(dsp),
 	},

sound/soc/intel/boards/bytcht_es8316.c

···
 
 	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);

sound/soc/intel/common/soc-acpi-intel-ehl-match.c

···
 
 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",

sound/soc/meson/axg-card.c

···
 
 	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;
 

sound/soc/soc-dai.c

···
 	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)
 {

sound/soc/sof/core.c

···
 	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);

sound/soc/sof/imx/imx8.c

···
 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,
+	},
 },
 };
 

sound/soc/sof/imx/imx8m.c

···
 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,
+	},
 },
 };