Re: [PATCH v1] ASoC: Intel: kbl_da7219_max98927: Fix kabylake_ssp_fixup function

[Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>]

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;
>