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 <harshapriy...@intel.com> and
Vamshi Krishna Gopal <vamshi.krishna.go...@intel.com>

Cc: <sta...@vger.kernel.org> # 5.4+
Signed-off-by: Lukasz Majczak <l...@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-harshapriy...@intel.com/
just applied to the kbl_da7219_max98927.
Best regards,
Lukasz

Acked-by: Pierre-Louis Bossart <pierre-louis.boss...@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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