Merge remote-tracking branch 'asoc/for-5.12' into asoc-linus

2 files changed, 33 insertions, 10 deletions

diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
index cc9a2509ace2..e0149cf6127d 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98927.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -282,12 +282,34 @@ 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
+	 */
+	/*
 	 * Topology for kblda7219m98373 & kblmax98373 supports only S24_LE,
 	 * where as kblda7219m98927 & kblmax98927 supports S16_LE by default.
 	 * Skipping the port wise FE and BE configuration for kblda7219m98373 &
@@ -309,9 +331,9 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	/*
 	 * 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);
@@ -322,7 +344,7 @@ 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;
diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c
index 0fd1e8f62c89..96741c7c0fba 100644
--- a/sound/soc/intel/keembay/kmb_platform.c
+++ b/sound/soc/intel/keembay/kmb_platform.c
@@ -105,14 +105,15 @@ static unsigned int kmb_pcm_tx_fn(struct kmb_i2s_info *kmb_i2s,
 	void *buf = runtime->dma_area;
 	int i;
 
+	if (kmb_i2s->iec958_fmt)
+		hdmi_reformat_iec958(runtime, kmb_i2s, tx_ptr);
+
 	/* KMB i2s uses two separate L/R FIFO */
 	for (i = 0; i < kmb_i2s->fifo_th; i++) {
 		if (kmb_i2s->config.data_width == 16) {
 			writel(((u16(*)[2])buf)[tx_ptr][0], i2s_base + LRBR_LTHR(0));
 			writel(((u16(*)[2])buf)[tx_ptr][1], i2s_base + RRBR_RTHR(0));
 		} else {
-			if (kmb_i2s->iec958_fmt)
-				hdmi_reformat_iec958(runtime, kmb_i2s, tx_ptr);
 			writel(((u32(*)[2])buf)[tx_ptr][0], i2s_base + LRBR_LTHR(0));
 			writel(((u32(*)[2])buf)[tx_ptr][1], i2s_base + RRBR_RTHR(0));
 		}