Unverified Commit 5c5f1bae authored by Harsha Priya's avatar Harsha Priya Committed by Mark Brown
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ASoC: Intel: kbl_rt5663_rt5514_max98927: Fix kabylake_ssp_fixup function



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: default avatarHarsha Priya <harshapriya.n@intel.com>
Signed-off-by: default avatarVamshi Krishna Gopal <vamshi.krishna.gopal@intel.com>
Tested-by: default avatarLukasz Majczak <lma@semihalf.com>
Acked-by: default avatarPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/1595432147-11166-1-git-send-email-harshapriya.n@intel.com


Signed-off-by: default avatarMark Brown <broonie@kernel.org>
parent ffc6d45d
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+32 −9
Original line number Diff line number Diff line
@@ -336,22 +336,45 @@ 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;
@@ -362,7 +385,7 @@ 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;