| Message ID | 1595432147-11166-1-git-send-email-harshapriya.n@intel.com (mailing list archive) |
|---|---|
| State | Accepted |
| Commit | 5c5f1baee85ae48b1ff50da4cc5e89f496be702c |
| Headers | show |
| Series | [v8] ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function | expand |
On 7/22/20 10:35 AM, Harsha Priya wrote: > 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> Thanks Harsha and Lukasz!
On Wed, 22 Jul 2020 08:35:47 -0700, Harsha Priya wrote: > 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. > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function commit: 5c5f1baee85ae48b1ff50da4cc5e89f496be702c All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 584e4f9cedc2..21808fe13481 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -379,22 +379,45 @@ 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; @@ -405,7 +428,7 @@ 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;