From: Vamshi Krishna Gopal vamshi.krishna.gopal@intel.com

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 harshapriya.n@intel.com Signed-off-by: Vamshi Krishna Gopal vamshi.krishna.gopal@intel.com Tested-by: Lukasz Majczak lma@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 --- --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 26 ++++++++++++------- 1 file changed, 17 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..b261b1c466a8 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -379,22 +379,30 @@ 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; + + /* + * This macro will be called for playback stream + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) + rtd_dpcm = dpcm; + /* + * This macro will be called for capture stream + */ + for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) + rtd_dpcm = dpcm;

/* * 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 +413,7 @@ 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;