ASoC: Fixes for v5.10

A batch of driver specific fixes that have come up since the merge
 window, nothing particularly major here but all good to have.
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Merge tag 'asoc-fix-v5.10-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v5.10

A batch of driver specific fixes that have come up since the merge
window, nothing particularly major here but all good to have.

sound/soc/atmel/mchp-spdiftx.c

@@ -487,7 +487,6 @@ static int mchp_spdiftx_hw_params(struct snd_pcm_substream *substream,
 	}
 	mchp_spdiftx_channel_status_write(dev);
 	spin_unlock_irqrestore(&ctrl->lock, flags);
-	mr |= SPDIFTX_MR_VALID1 | SPDIFTX_MR_VALID2;
 
 	if (dev->gclk_enabled) {
 		clk_disable_unprepare(dev->gclk);

sound/soc/codecs/cs42l51.c

@@ -254,8 +254,28 @@ static const struct snd_soc_dapm_widget cs42l51_dapm_widgets[] = {
 		&cs42l51_adcr_mux_controls),
 };
 
+static int mclk_event(struct snd_soc_dapm_widget *w,
+		      struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm);
+	struct cs42l51_private *cs42l51 = snd_soc_component_get_drvdata(comp);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		return clk_prepare_enable(cs42l51->mclk_handle);
+	case SND_SOC_DAPM_POST_PMD:
+		/* Delay mclk shutdown to fulfill power-down sequence requirements */
+		msleep(20);
+		clk_disable_unprepare(cs42l51->mclk_handle);
+		break;
+	}
+
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget cs42l51_dapm_mclk_widgets[] = {
-	SND_SOC_DAPM_CLOCK_SUPPLY("MCLK")
+	SND_SOC_DAPM_SUPPLY("MCLK", SND_SOC_NOPM, 0, 0, mclk_event,
+			    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
 };
 
 static const struct snd_soc_dapm_route cs42l51_routes[] = {

sound/soc/codecs/wcd9335.c

@@ -618,7 +618,7 @@ static const char * const sb_tx8_mux_text[] = {
 	"ZERO", "RX_MIX_TX8", "DEC8", "DEC8_192"
 };
 
-static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0);
+static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400);
 static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1);
 static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1);
 static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0);

sound/soc/codecs/wcd934x.c

@@ -551,7 +551,7 @@ struct wcd_iir_filter_ctl {
 	struct soc_bytes_ext bytes_ext;
 };
 
-static const DECLARE_TLV_DB_SCALE(digital_gain, 0, 1, 0);
+static const DECLARE_TLV_DB_SCALE(digital_gain, -8400, 100, -8400);
 static const DECLARE_TLV_DB_SCALE(line_gain, 0, 7, 1);
 static const DECLARE_TLV_DB_SCALE(analog_gain, 0, 25, 1);
 static const DECLARE_TLV_DB_SCALE(ear_pa_gain, 0, 150, 0);

sound/soc/codecs/wsa881x.c

@@ -1026,6 +1026,8 @@ static struct snd_soc_dai_driver wsa881x_dais[] = {
 		.id = 0,
 		.playback = {
 			.stream_name = "SPKR Playback",
+			.rates = SNDRV_PCM_RATE_48000,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
 			.rate_max = 48000,
 			.rate_min = 48000,
 			.channels_min = 1,

sound/soc/intel/Kconfig

@@ -15,22 +15,6 @@ config SND_SOC_INTEL_SST_TOPLEVEL
 
 if SND_SOC_INTEL_SST_TOPLEVEL
 
-config SND_SST_IPC
-	tristate
-	# This option controls the IPC core for HiFi2 platforms
-
-config SND_SST_IPC_PCI
-	tristate
-	select SND_SST_IPC
-	# This option controls the PCI-based IPC for HiFi2 platforms
-	#  (Medfield, Merrifield).
-
-config SND_SST_IPC_ACPI
-	tristate
-	select SND_SST_IPC
-	# This option controls the ACPI-based IPC for HiFi2 platforms
-	# (Baytrail, Cherrytrail)
-
 config SND_SOC_INTEL_SST
 	tristate
 
@@ -57,7 +41,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM
 config SND_SST_ATOM_HIFI2_PLATFORM_PCI
 	tristate "PCI HiFi2 (Merrifield) Platforms"
 	depends on X86 && PCI
-	select SND_SST_IPC_PCI
 	select SND_SST_ATOM_HIFI2_PLATFORM
 	help
 	  If you have a Intel Merrifield/Edison platform, then
@@ -70,7 +53,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI
 	tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms"
 	default ACPI
 	depends on X86 && ACPI && PCI
-	select SND_SST_IPC_ACPI
 	select SND_SST_ATOM_HIFI2_PLATFORM
 	select SND_SOC_ACPI_INTEL_MATCH
 	select IOSF_MBI

sound/soc/intel/atom/Makefile

@@ -6,4 +6,4 @@ snd-soc-sst-atom-hifi2-platform-objs :=	sst-mfld-platform-pcm.o \
 obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-soc-sst-atom-hifi2-platform.o
 
 # DSP driver
-obj-$(CONFIG_SND_SST_IPC) += sst/
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += sst/

sound/soc/intel/atom/sst/Makefile

@@ -3,6 +3,6 @@ snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_
 snd-intel-sst-pci-objs += sst_pci.o
 snd-intel-sst-acpi-objs += sst_acpi.o
 
-obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst-core.o
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-intel-sst-core.o
-obj-$(CONFIG_SND_SST_IPC_PCI) += snd-intel-sst-pci.o
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_PCI) += snd-intel-sst-pci.o
-obj-$(CONFIG_SND_SST_IPC_ACPI) += snd-intel-sst-acpi.o
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) += snd-intel-sst-acpi.o

sound/soc/intel/boards/kbl_rt5663_max98927.c

@@ -401,17 +401,40 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	struct snd_interval *chan = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
-	struct snd_soc_dpcm *dpcm = container_of(
+	struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL;
-			params, struct snd_soc_dpcm, hw_params);
+
-	struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link;
+	/*
-	struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link;
+	 * The following loop will be called only for playback stream
+	 * In this platform, there is only one playback device on every SSP
+	 */
+	for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
+		rtd_dpcm = dpcm;
+		break;
+	}
+
+	/*
+	 * This following loop will be called only for capture stream
+	 * In this platform, there is only one capture device on every SSP
+	 */
+	for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) {
+		rtd_dpcm = dpcm;
+		break;
+	}
+
+	if (!rtd_dpcm)
+		return -EINVAL;
+
+	/*
+	 * The above 2 loops are mutually exclusive based on the stream direction,
+	 * thus rtd_dpcm variable will never be overwritten
+	 */
 
 	/*
 	 * The ADSP will convert the FE rate to 48k, stereo, 24 bit
 	 */
-	if (!strcmp(fe_dai_link->name, "Kbl Audio Port") ||
+	if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") ||
-	    !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
+	    !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") ||
-	    !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
+	    !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) {
 		rate->min = rate->max = 48000;
 		chan->min = chan->max = 2;
 		snd_mask_none(fmt);
@@ -421,7 +444,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	 * The speaker on the SSP0 supports S16_LE and not S24_LE.
 	 * thus changing the mask here
 	 */
-	if (!strcmp(be_dai_link->name, "SSP0-Codec"))
+	if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec"))
 		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
 
 	return 0;

sound/soc/intel/catpt/dsp.c

@@ -267,9 +267,12 @@ static int catpt_dsp_select_lpclock(struct catpt_dev *cdev, bool lp, bool waiti)
 					    reg, (reg & CATPT_ISD_DCPWM),
 					    500, 10000);
 		if (ret) {
-			dev_err(cdev->dev, "await WAITI timeout\n");
+			dev_warn(cdev->dev, "await WAITI timeout\n");
-			mutex_unlock(&cdev->clk_mutex);
+			/* no signal - only high clock selection allowed */
-			return ret;
+			if (lp) {
+				mutex_unlock(&cdev->clk_mutex);
+				return 0;
+			}
 		}
 	}
 

sound/soc/intel/catpt/pcm.c

@@ -667,7 +667,17 @@ static int catpt_dai_pcm_new(struct snd_soc_pcm_runtime *rtm,
 		break;
 	}
 
+	/* see if this is a new configuration */
+	if (!memcmp(&cdev->devfmt[devfmt.iface], &devfmt, sizeof(devfmt)))
+		return 0;
+
+	pm_runtime_get_sync(cdev->dev);
+
 	ret = catpt_ipc_set_device_format(cdev, &devfmt);
+
+	pm_runtime_mark_last_busy(cdev->dev);
+	pm_runtime_put_autosuspend(cdev->dev);
+
 	if (ret)
 		return CATPT_IPC_ERROR(ret);
 

sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c

@@ -630,15 +630,34 @@ static struct snd_soc_codec_conf mt8183_da7219_rt1015_codec_conf[] = {
 	},
 };
 
+static const struct snd_kcontrol_new mt8183_da7219_rt1015_snd_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Left Spk"),
+	SOC_DAPM_PIN_SWITCH("Right Spk"),
+};
+
+static const
+struct snd_soc_dapm_widget mt8183_da7219_rt1015_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Left Spk", NULL),
+	SND_SOC_DAPM_SPK("Right Spk", NULL),
+	SND_SOC_DAPM_PINCTRL("TDM_OUT_PINCTRL",
+			     "aud_tdm_out_on", "aud_tdm_out_off"),
+};
+
+static const struct snd_soc_dapm_route mt8183_da7219_rt1015_dapm_routes[] = {
+	{"Left Spk", NULL, "Left SPO"},
+	{"Right Spk", NULL, "Right SPO"},
+	{"I2S Playback", NULL, "TDM_OUT_PINCTRL"},
+};
+
 static struct snd_soc_card mt8183_da7219_rt1015_card = {
 	.name = "mt8183_da7219_rt1015",
 	.owner = THIS_MODULE,
-	.controls = mt8183_da7219_max98357_snd_controls,
+	.controls = mt8183_da7219_rt1015_snd_controls,
-	.num_controls = ARRAY_SIZE(mt8183_da7219_max98357_snd_controls),
+	.num_controls = ARRAY_SIZE(mt8183_da7219_rt1015_snd_controls),
-	.dapm_widgets = mt8183_da7219_max98357_dapm_widgets,
+	.dapm_widgets = mt8183_da7219_rt1015_dapm_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_max98357_dapm_widgets),
+	.num_dapm_widgets = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_widgets),
-	.dapm_routes = mt8183_da7219_max98357_dapm_routes,
+	.dapm_routes = mt8183_da7219_rt1015_dapm_routes,
-	.num_dapm_routes = ARRAY_SIZE(mt8183_da7219_max98357_dapm_routes),
+	.num_dapm_routes = ARRAY_SIZE(mt8183_da7219_rt1015_dapm_routes),
 	.dai_link = mt8183_da7219_dai_links,
 	.num_links = ARRAY_SIZE(mt8183_da7219_dai_links),
 	.aux_dev = &mt8183_da7219_max98357_headset_dev,

sound/soc/qcom/lpass-cpu.c

@@ -80,6 +80,12 @@ static int lpass_cpu_daiops_startup(struct snd_pcm_substream *substream,
 		dev_err(dai->dev, "error in enabling mi2s osr clk: %d\n", ret);
 		return ret;
 	}
+	ret = clk_prepare(drvdata->mi2s_bit_clk[dai->driver->id]);
+	if (ret) {
+		dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret);
+		clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]);
+		return ret;
+	}
 	return 0;
 }
 
@@ -88,9 +94,8 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream,
 {
 	struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai);
 
-	clk_disable_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]);
-
 	clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]);
+	clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]);
 }
 
 static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream,
@@ -303,10 +308,10 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
 			dev_err(dai->dev, "error writing to i2sctl reg: %d\n",
 				ret);
 
-		ret = clk_prepare_enable(drvdata->mi2s_bit_clk[id]);
+		ret = clk_enable(drvdata->mi2s_bit_clk[id]);
 		if (ret) {
 			dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret);
-			clk_disable_unprepare(drvdata->mi2s_osr_clk[id]);
+			clk_disable(drvdata->mi2s_osr_clk[id]);
 			return ret;
 		}
 
@@ -324,6 +329,7 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream,
 		if (ret)
 			dev_err(dai->dev, "error writing to i2sctl reg: %d\n",
 				ret);
+		clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]);
 		break;
 	}
 

sound/soc/qcom/lpass-sc7180.c

@@ -188,7 +188,7 @@ static struct lpass_variant sc7180_data = {
 	.micmode		= REG_FIELD_ID(0x1000, 4, 8, 3, 0x1000),
 	.micmono		= REG_FIELD_ID(0x1000, 3, 3, 3, 0x1000),
 	.wssrc			= REG_FIELD_ID(0x1000, 2, 2, 3, 0x1000),
-	.bitwidth		= REG_FIELD_ID(0x1000, 0, 0, 3, 0x1000),
+	.bitwidth		= REG_FIELD_ID(0x1000, 0, 1, 3, 0x1000),
 
 	.rdma_dyncclk		= REG_FIELD_ID(0xC000, 21, 21, 5, 0x1000),
 	.rdma_bursten		= REG_FIELD_ID(0xC000, 20, 20, 5, 0x1000),

sound/soc/qcom/sdm845.c

@@ -17,6 +17,7 @@
 #include "qdsp6/q6afe.h"
 #include "../codecs/rt5663.h"
 
+#define DRIVER_NAME	"sdm845"
 #define DEFAULT_SAMPLE_RATE_48K		48000
 #define DEFAULT_MCLK_RATE		24576000
 #define TDM_BCLK_RATE		6144000
@@ -552,6 +553,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev)
 	if (!data)
 		return -ENOMEM;
 
+	card->driver_name = DRIVER_NAME;
 	card->dapm_widgets = sdm845_snd_widgets;
 	card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets);
 	card->dev = dev;

sound/soc/sof/loader.c

@@ -118,6 +118,11 @@ int snd_sof_fw_parse_ext_data(struct snd_sof_dev *sdev, u32 bar, u32 offset)
 		case SOF_IPC_EXT_CC_INFO:
 			ret = get_cc_info(sdev, ext_hdr);
 			break;
+		case SOF_IPC_EXT_UNUSED:
+		case SOF_IPC_EXT_PROBE_INFO:
+		case SOF_IPC_EXT_USER_ABI_INFO:
+			/* They are supported but we don't do anything here */
+			break;
 		default:
 			dev_warn(sdev->dev, "warning: unknown ext header type %d size 0x%x\n",
 				 ext_hdr->type, ext_hdr->hdr.size);