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SubjectRe: [PATCH v1] ASoC: Intel: kbl_da7219_max98927: Fix kabylake_ssp_fixup function
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On 4/15/21 7:43 AM, Lukasz Majczak 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.
> Based on similar fix in kbl_rt5663_rt5514_max98927.c
> from Harsha Priya <harshapriya.n@intel.com> and
> Vamshi Krishna Gopal <vamshi.krishna.gopal@intel.com>
>
> Cc: <stable@vger.kernel.org> # 5.4+
> Signed-off-by: Lukasz Majczak <lma@semihalf.com>
> ---
> Hi,
> This is basically a cherry-pick of this change:
> https://patchwork.kernel.org/project/alsa-devel/patch/1595432147-11166-1-git-send-email-harshapriya.n@intel.com/
> just applied to the kbl_da7219_max98927.
> Best regards,
> Lukasz

Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>

>
> sound/soc/intel/boards/kbl_da7219_max98927.c | 38 +++++++++++++++-----
> 1 file changed, 30 insertions(+), 8 deletions(-)
>
> diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
> index 9dfe5bd9180d..4b7b4a044f81 100644
> --- a/sound/soc/intel/boards/kbl_da7219_max98927.c
> +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
> @@ -284,11 +284,33 @@ 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
> + */
> /*
> * Topology for kblda7219m98373 & kblmax98373 supports only S24_LE,
> * where as kblda7219m98927 & kblmax98927 supports S16_LE by default.
> @@ -311,9 +333,9 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
> /*
> * 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);
> @@ -324,7 +346,7 @@ 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;
>

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