[PATCH v8] ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function

[Harsha Priya]

kabylake_ssp_fixup function uses snd_soc_dpcm to identify the
codecs DAIs. The HW parameters are changed based on the codec DAI of the
stream. The earlier approach to get snd_soc_dpcm was using container_of()
macro on snd_pcm_hw_params.

The structures have been modified over time and snd_soc_dpcm does not have
snd_pcm_hw_params as a reference but as a copy. This causes the current
driver to crash when used.

This patch changes the way snd_soc_dpcm is extracted. snd_soc_pcm_runtime
holds 2 dpcm instances (one for playback and one for capture). 2 codecs
on the SSP are dmic (capture) and speakers (playback). Based on the
stream direction, snd_soc_dpcm is extracted from snd_soc_pcm_runtime.

Tested for all use cases of the driver.

Signed-off-by: Harsha Priya <harshapriya.n at intel.com>
Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal at intel.com>
Tested-by: Lukasz Majczak <lma at semihalf.com>
---
v1 -> v2:
- Extract dmic from SSP0 as every BE should have own fixup function.
v2 -> v3:
- Restore naming in the dapm route table to not confuse with other
drivers
- Fixed indentations
v3 -> v4:
- Updated code and commit description according to
solution proposed by Harsha
v4 -> v5:
- Cosmetic Changes
v5 -> v6:
- Dmic regression seen with v4 fixed 
- Using available routines for obtaining dpcm information
v6 -> v7:
- Updated comments
- initilize rtd_dpcm variable
- added break statement in the loop
v7 -> v8:
- Updated commit message
- return on rtd_dpcm being NULL
- typo fixed
---
---
 .../intel/boards/kbl_rt5663_rt5514_max98927.c | 41 +++++++++++++++----
 1 file changed, 32 insertions(+), 9 deletions(-)

diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 584e4f9cedc2..21808fe13481 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -379,22 +379,45 @@ 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;
@@ -405,7 +428,7 @@ 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;
-- 
2.17.1