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ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function 

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@intel.com>
Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal@intel.com>
Tested-by: Lukasz Majczak <lma@semihalf.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/1595432147-11166-1-git-send-email-harshapriya.n@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
zero-sugar-mainline-defconfig
Harsha Priya 2020-07-22 08:35:47 -07:00 committed by Mark Brown
parent ffc6d45d96
commit 5c5f1baee8
No known key found for this signature in database
GPG Key ID: 24D68B725D5487D0
1 changed files with 32 additions and 9 deletions
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41
sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
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@ -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;