| Message ID | 1595365649-8019-1-git-send-email-harshapriya.n@intel.com (mailing list archive) |
|---|---|
| State | New, archived |
| Headers | show |
| Series | [v7] ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function | expand |
Hi Harsha, this looks mostly good to me, only have a couple of nit-picks below. Thanks! > 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. Maybe reformat a bit and add newlines, this is difficult to read. > 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> > 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 idea of initializing was also to test before dereferencing this pointer. Without the test, this is somewhat useless, tools may still complain about dereferencing a NULL pointer? > + > + /* > + * 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; > + } add if (!rtd_dpcm) return -EINVAL here? > + /* > + * The above 2 loops are mutually exclusive based on the strem direction, typo: stream > + * thus rtd_dpcm variable will never be overwritten > + */
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;