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Subject: Re: [PATCH v6] ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix
 kabylake_ssp_fixup function
To:     Harsha Priya <harshapriya.n@intel.com>,
        alsa-devel@alsa-project.org, broonie@kernel.org
Cc:     lma@semihalf.com, yang.jie@linux.intel.com,
        liam.r.girdwood@linux.intel.com, rad@semihalf.com,
        zwisler@google.com, linux-kernel@vger.kernel.org,
        sathya.prakash.m.r@intel.com, brndt@google.com, mw@semihalf.com,
        levinale@chromium.org,
        Vamshi Krishna Gopal <vamshi.krishna.gopal@intel.com>
References: <1594919637-31460-1-git-send-email-harshapriya.n@intel.com>
From:   Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Message-ID: <48a6db3a-2aef-cfca-19ec-7825573fc3da@linux.intel.com>
Date:   Thu, 16 Jul 2020 12:49:09 -0500
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> 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;
>