From: Vamshi Krishna Gopal <[email protected]>
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 <[email protected]>
Signed-off-by: Vamshi Krishna Gopal <[email protected]>
Tested-by: Lukasz Majczak <[email protected]>
---
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
> 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;
is the assumption that both of those loops return the same pointer?
If yes, why not stop for the first non-NULL dpcm value?
Also wondering if you are using a loop because there's no other helper
available?
>
> /*
> * 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;
>
>
>
>
> > 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;
>
> is the assumption that both of those loops return the same pointer?
no only one loop will enter based on the direction of the stream
If it’s a playback stream, the dpcm[1] will be empty
If it’s a capture stream, the dpcm[0] will be empty
> If yes, why not stop for the first non-NULL dpcm value?
> Also wondering if you are using a loop because there's no other helper
> available?
yes. I did not find any other helper function so I took the loop.