From: Vamshi Krishna Gopal <vamshi.krishna.gopal@xxxxxxxxx>
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@xxxxxxxxx>
Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal@xxxxxxxxx>
Tested-by: Lukasz Majczak <lma@xxxxxxxxxxxx>
---
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;
--
2.17.1
