kabylake_ssp_fixup function uses snd_soc_dpcm to identify the codecs DAIs.
The hw parameters are changed based on the codec DAI,
the stream is intended for. 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.
The snd_soc_pcm_runtime holds 2 dpcm
instances (one for playback and one for capture).
The 2 codecs on this SSP are dmic and speakers.
One is for capture and one is for playback respectively.
Based on the direction of the stream,
the snd_soc_dpcm is extracted from the snd_soc_pcm_runtime structure.
Tested for all use cases of the driver.

Signed-off-by: Harsha Priya <harshapriy...@intel.com>
Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.go...@intel.com>
Tested-by: Lukasz Majczak <l...@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

---
---
 .../intel/boards/kbl_rt5663_rt5514_max98927.c | 38 ++++++++++++++-----
 1 file changed, 29 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..9f4b949cc39c 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -379,22 +379,42 @@ 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;
+       }
+
+       /*
+        * The above 2 loops are mutually exclusive based on the strem 
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 +425,7 @@ 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

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