Merge series "ASoC: meson: tdm fixes" from Jerome Brunet <jbrunet@baylibre.com>:

This patcheset is collection of fixes for the TDM input and output the
axg audio architecture. Its fixes:
 - slave mode format setting
 - g12 and sm1 skew offset
 - tdm clock inversion
 - standard daifmt props names which don't require a specific prefix

Jerome Brunet (4):
  ASoC: meson: axg-tdm-interface: fix link fmt setup
  ASoC: meson: axg-tdmin: fix g12a skew
  ASoC: meson: axg-tdm-formatters: fix sclk inversion
  ASoC: meson: cards: remove DT_PREFIX for standard daifmt properties

 sound/soc/meson/axg-tdm-formatter.c | 11 ++++++-----
 sound/soc/meson/axg-tdm-formatter.h |  1 -
 sound/soc/meson/axg-tdm-interface.c | 26 +++++++++++++++++---------
 sound/soc/meson/axg-tdmin.c         | 16 +++++++++++++++-
 sound/soc/meson/axg-tdmout.c        |  3 ---
 sound/soc/meson/meson-card-utils.c  |  2 +-
 6 files changed, 39 insertions(+), 20 deletions(-)

--
2.25.4

drivers/spi/spi-lantiq-ssc.c

@@ -899,7 +899,7 @@ static int lantiq_ssc_probe(struct platform_device *pdev)
 	master->bits_per_word_mask = SPI_BPW_RANGE_MASK(2, 8) |
 				     SPI_BPW_MASK(16) | SPI_BPW_MASK(32);
 
-	spi->wq = alloc_ordered_workqueue(dev_name(dev), 0);
+	spi->wq = alloc_ordered_workqueue(dev_name(dev), WQ_MEM_RECLAIM);
 	if (!spi->wq) {
 		err = -ENOMEM;
 		goto err_clk_put;

sound/soc/codecs/max98357a.c

@@ -23,36 +23,61 @@
 struct max98357a_priv {
 	struct gpio_desc *sdmode;
 	unsigned int sdmode_delay;
+	int sdmode_switch;
 };
 
-static int max98357a_sdmode_event(struct snd_soc_dapm_widget *w,
-		struct snd_kcontrol *kcontrol, int event)
+static int max98357a_daiops_trigger(struct snd_pcm_substream *substream,
+		int cmd, struct snd_soc_dai *dai)
 {
-	struct snd_soc_component *component =
-		snd_soc_dapm_to_component(w->dapm);
+	struct snd_soc_component *component = dai->component;
 	struct max98357a_priv *max98357a =
 		snd_soc_component_get_drvdata(component);
 
 	if (!max98357a->sdmode)
 		return 0;
 
-	if (event & SND_SOC_DAPM_POST_PMU) {
-		msleep(max98357a->sdmode_delay);
-		gpiod_set_value(max98357a->sdmode, 1);
-		dev_dbg(component->dev, "set sdmode to 1");
-	} else if (event & SND_SOC_DAPM_PRE_PMD) {
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		mdelay(max98357a->sdmode_delay);
+		if (max98357a->sdmode_switch) {
+			gpiod_set_value(max98357a->sdmode, 1);
+			dev_dbg(component->dev, "set sdmode to 1");
+		}
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		gpiod_set_value(max98357a->sdmode, 0);
 		dev_dbg(component->dev, "set sdmode to 0");
+		break;
 	}
 
 	return 0;
 }
 
+static int max98357a_sdmode_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_component *component =
+		snd_soc_dapm_to_component(w->dapm);
+	struct max98357a_priv *max98357a =
+		snd_soc_component_get_drvdata(component);
+
+	if (event & SND_SOC_DAPM_POST_PMU)
+		max98357a->sdmode_switch = 1;
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		max98357a->sdmode_switch = 0;
+
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget max98357a_dapm_widgets[] = {
 	SND_SOC_DAPM_OUTPUT("Speaker"),
 	SND_SOC_DAPM_OUT_DRV_E("SD_MODE", SND_SOC_NOPM, 0, 0, NULL, 0,
 			max98357a_sdmode_event,
-			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+			SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 };
 
 static const struct snd_soc_dapm_route max98357a_dapm_routes[] = {
@@ -71,6 +96,10 @@ static const struct snd_soc_component_driver max98357a_component_driver = {
 	.non_legacy_dai_naming	= 1,
 };
 
+static const struct snd_soc_dai_ops max98357a_dai_ops = {
+	.trigger        = max98357a_daiops_trigger,
+};
+
 static struct snd_soc_dai_driver max98357a_dai_driver = {
 	.name = "HiFi",
 	.playback = {
@@ -90,6 +119,7 @@ static struct snd_soc_dai_driver max98357a_dai_driver = {
 		.channels_min	= 1,
 		.channels_max	= 2,
 	},
+	.ops    = &max98357a_dai_ops,
 };
 
 static int max98357a_platform_probe(struct platform_device *pdev)

sound/soc/codecs/max98390.c

@@ -678,7 +678,7 @@ static const struct snd_kcontrol_new max98390_dai_controls =
 
 static const struct snd_soc_dapm_widget max98390_dapm_widgets[] = {
 	SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback",
-		MAX98390_R203A_AMP_EN, 0, 0, max98390_dac_event,
+		SND_SOC_NOPM, 0, 0, max98390_dac_event,
 		SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0,
 		&max98390_dai_controls),

sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c

@@ -336,22 +336,45 @@ 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(
-			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;
+	struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL;
+
+	/*
+	 * 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") ||
-	    !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
-	    !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
+	if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") ||
+	    !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") ||
+	    !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);
 		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
-	} else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) {
+	} else if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio DMIC cap")) {
 		if (params_channels(params) == 2 ||
 				DMIC_CH(dmic_constraints) == 2)
 			chan->min = chan->max = 2;
@@ -362,7 +385,7 @@ 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/boards/skl_hda_dsp_common.h

@@ -33,6 +33,7 @@ struct skl_hda_private {
 	int dai_index;
 	const char *platform_name;
 	bool common_hdmi_codec_drv;
+	bool idisp_codec;
 };
 
 extern struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS];

sound/soc/intel/boards/skl_hda_dsp_generic.c

@@ -79,6 +79,9 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link)
 	link->platforms->name = ctx->platform_name;
 	link->nonatomic = 1;
 
+	if (!ctx->idisp_codec)
+		return 0;
+
 	if (strstr(link->name, "HDMI")) {
 		ret = skl_hda_hdmi_add_pcm(card, ctx->pcm_count);
 
@@ -118,19 +121,20 @@ static char hda_soc_components[30];
 static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params)
 {
 	struct snd_soc_card *card = &hda_soc_card;
+	struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card);
 	struct snd_soc_dai_link *dai_link;
-	u32 codec_count, codec_mask, idisp_mask;
+	u32 codec_count, codec_mask;
 	int i, num_links, num_route;
 
 	codec_mask = mach_params->codec_mask;
 	codec_count = hweight_long(codec_mask);
-	idisp_mask = codec_mask & IDISP_CODEC_MASK;
+	ctx->idisp_codec = !!(codec_mask & IDISP_CODEC_MASK);
 
 	if (!codec_count || codec_count > 2 ||
-	    (codec_count == 2 && !idisp_mask))
+	    (codec_count == 2 && !ctx->idisp_codec))
 		return -EINVAL;
 
-	if (codec_mask == idisp_mask) {
+	if (codec_mask == IDISP_CODEC_MASK) {
 		/* topology with iDisp as the only HDA codec */
 		num_links = IDISP_DAI_COUNT + DMIC_DAI_COUNT;
 		num_route = IDISP_ROUTE_COUNT;
@@ -152,7 +156,7 @@ static int skl_hda_fill_card_info(struct snd_soc_acpi_mach_params *mach_params)
 		num_route = ARRAY_SIZE(skl_hda_map);
 		card->dapm_widgets = skl_hda_widgets;
 		card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets);
-		if (!idisp_mask) {
+		if (!ctx->idisp_codec) {
 			for (i = 0; i < IDISP_DAI_COUNT; i++) {
 				skl_hda_be_dai_links[i].codecs = dummy_codec;
 				skl_hda_be_dai_links[i].num_codecs =
@@ -211,6 +215,8 @@ static int skl_hda_audio_probe(struct platform_device *pdev)
 	if (!mach)
 		return -EINVAL;
 
+	snd_soc_card_set_drvdata(&hda_soc_card, ctx);
+
 	ret = skl_hda_fill_card_info(&mach->mach_params);
 	if (ret < 0) {
 		dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n");
@@ -223,7 +229,6 @@ static int skl_hda_audio_probe(struct platform_device *pdev)
 	ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv;
 
 	hda_soc_card.dev = &pdev->dev;
-	snd_soc_card_set_drvdata(&hda_soc_card, ctx);
 
 	if (mach->mach_params.dmic_num > 0) {
 		snprintf(hda_soc_components, sizeof(hda_soc_components),

sound/soc/meson/axg-tdm-formatter.c

@@ -70,7 +70,7 @@ EXPORT_SYMBOL_GPL(axg_tdm_formatter_set_channel_masks);
 static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter)
 {
 	struct axg_tdm_stream *ts = formatter->stream;
-	bool invert = formatter->drv->quirks->invert_sclk;
+	bool invert;
 	int ret;
 
 	/* Do nothing if the formatter is already enabled */
@@ -96,11 +96,12 @@ static int axg_tdm_formatter_enable(struct axg_tdm_formatter *formatter)
 		return ret;
 
 	/*
-	 * If sclk is inverted, invert it back and provide the inversion
-	 * required by the formatter
+	 * If sclk is inverted, it means the bit should latched on the
+	 * rising edge which is what our HW expects. If not, we need to
+	 * invert it before the formatter.
 	 */
-	invert ^= axg_tdm_sclk_invert(ts->iface->fmt);
-	ret = clk_set_phase(formatter->sclk, invert ? 180 : 0);
+	invert = axg_tdm_sclk_invert(ts->iface->fmt);
+	ret = clk_set_phase(formatter->sclk, invert ? 0 : 180);
 	if (ret)
 		return ret;
 

sound/soc/meson/axg-tdm-formatter.h

@@ -16,7 +16,6 @@ struct snd_kcontrol;
 
 struct axg_tdm_formatter_hw {
 	unsigned int skew_offset;
-	bool invert_sclk;
 };
 
 struct axg_tdm_formatter_ops {

sound/soc/meson/axg-tdm-interface.c

@@ -119,18 +119,25 @@ static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
 	struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai);
 
-	/* These modes are not supported */
-	if (fmt & (SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_CBM_CFS)) {
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		if (!iface->mclk) {
+			dev_err(dai->dev, "cpu clock master: mclk missing\n");
+			return -ENODEV;
+		}
+		break;
+
+	case SND_SOC_DAIFMT_CBM_CFM:
+		break;
+
+	case SND_SOC_DAIFMT_CBS_CFM:
+	case SND_SOC_DAIFMT_CBM_CFS:
 		dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n");
+		/* Fall-through */
+	default:
 		return -EINVAL;
 	}
 
-	/* If the TDM interface is the clock master, it requires mclk */
-	if (!iface->mclk && (fmt & SND_SOC_DAIFMT_CBS_CFS)) {
-		dev_err(dai->dev, "cpu clock master: mclk missing\n");
-		return -ENODEV;
-	}
-
 	iface->fmt = fmt;
 	return 0;
 }
@@ -319,7 +326,8 @@ static int axg_tdm_iface_hw_params(struct snd_pcm_substream *substream,
 	if (ret)
 		return ret;
 
-	if (iface->fmt & SND_SOC_DAIFMT_CBS_CFS) {
+	if ((iface->fmt & SND_SOC_DAIFMT_MASTER_MASK) ==
+	    SND_SOC_DAIFMT_CBS_CFS) {
 		ret = axg_tdm_iface_set_sclk(dai, params);
 		if (ret)
 			return ret;

sound/soc/meson/axg-tdmin.c

@@ -228,15 +228,29 @@ static const struct axg_tdm_formatter_driver axg_tdmin_drv = {
 	.regmap_cfg	= &axg_tdmin_regmap_cfg,
 	.ops		= &axg_tdmin_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk	= false,
 		.skew_offset	= 2,
 	},
 };
 
+static const struct axg_tdm_formatter_driver g12a_tdmin_drv = {
+	.component_drv	= &axg_tdmin_component_drv,
+	.regmap_cfg	= &axg_tdmin_regmap_cfg,
+	.ops		= &axg_tdmin_ops,
+	.quirks		= &(const struct axg_tdm_formatter_hw) {
+		.skew_offset	= 3,
+	},
+};
+
 static const struct of_device_id axg_tdmin_of_match[] = {
 	{
 		.compatible = "amlogic,axg-tdmin",
 		.data = &axg_tdmin_drv,
+	}, {
+		.compatible = "amlogic,g12a-tdmin",
+		.data = &g12a_tdmin_drv,
+	}, {
+		.compatible = "amlogic,sm1-tdmin",
+		.data = &g12a_tdmin_drv,
 	}, {}
 };
 MODULE_DEVICE_TABLE(of, axg_tdmin_of_match);

sound/soc/meson/axg-tdmout.c

@@ -238,7 +238,6 @@ static const struct axg_tdm_formatter_driver axg_tdmout_drv = {
 	.regmap_cfg	= &axg_tdmout_regmap_cfg,
 	.ops		= &axg_tdmout_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk = true,
 		.skew_offset = 1,
 	},
 };
@@ -248,7 +247,6 @@ static const struct axg_tdm_formatter_driver g12a_tdmout_drv = {
 	.regmap_cfg	= &axg_tdmout_regmap_cfg,
 	.ops		= &axg_tdmout_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk = true,
 		.skew_offset = 2,
 	},
 };
@@ -309,7 +307,6 @@ static const struct axg_tdm_formatter_driver sm1_tdmout_drv = {
 	.regmap_cfg	= &axg_tdmout_regmap_cfg,
 	.ops		= &axg_tdmout_ops,
 	.quirks		= &(const struct axg_tdm_formatter_hw) {
-		.invert_sclk = true,
 		.skew_offset = 2,
 	},
 };

sound/soc/meson/meson-card-utils.c

@@ -119,7 +119,7 @@ unsigned int meson_card_parse_daifmt(struct device_node *node,
 	struct device_node *framemaster = NULL;
 	unsigned int daifmt;
 
-	daifmt = snd_soc_of_parse_daifmt(node, DT_PREFIX,
+	daifmt = snd_soc_of_parse_daifmt(node, "",
 					 &bitclkmaster, &framemaster);
 	daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK;