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

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),