diff mbox series

[v3,05/23] ASoC: fsl: Replace dpcm_playback/capture to playback/capture_assertion

Message ID 87a5lr8g19.wl-kuninori.morimoto.gx@renesas.com (mailing list archive)
State New
Headers show
Series ASoC: Replace dpcm_playback/capture to playback/capture_assertion | expand

Commit Message

Kuninori Morimoto April 18, 2024, 4:13 a.m. UTC 
soc_get_playback_capture() is now handling DPCM and normal
comprehensively for playback/capture stream in same code.
This patch converts dpcm_xxx flag to xxx_assertion.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
---
 sound/soc/fsl/fsl-asoc-card.c | 20 ++++++++++----------
 sound/soc/fsl/imx-audmix.c    |  8 ++++----
 sound/soc/fsl/imx-card.c      |  4 ++--
 3 files changed, 16 insertions(+), 16 deletions(-)
diff mbox series

Patch

diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index f6d2564864c6..50ca3e72eb21 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -320,8 +320,8 @@  static const struct snd_soc_dai_link fsl_asoc_card_dai[] = {
 	{
 		.name = "HiFi-ASRC-FE",
 		.stream_name = "HiFi-ASRC-FE",
-		.dpcm_playback = 1,
-		.dpcm_capture = 1,
+		.playback_assertion = 1,
+		.capture_assertion = 1,
 		.dynamic = 1,
 		SND_SOC_DAILINK_REG(hifi_fe),
 	},
@@ -330,8 +330,8 @@  static const struct snd_soc_dai_link fsl_asoc_card_dai[] = {
 		.stream_name = "HiFi-ASRC-BE",
 		.be_hw_params_fixup = be_hw_params_fixup,
 		.ops = &fsl_asoc_card_ops,
-		.dpcm_playback = 1,
-		.dpcm_capture = 1,
+		.playback_assertion = 1,
+		.capture_assertion = 1,
 		.no_pcm = 1,
 		SND_SOC_DAILINK_REG(hifi_be),
 	},
@@ -649,8 +649,8 @@  static int fsl_asoc_card_probe(struct platform_device *pdev)
 	} else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic31xx")) {
 		codec_dai_name = "tlv320dac31xx-hifi";
 		priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
-		priv->dai_link[1].dpcm_capture = 0;
-		priv->dai_link[2].dpcm_capture = 0;
+		priv->dai_link[1].capture_assertion = 0;
+		priv->dai_link[2].capture_assertion = 0;
 		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
 		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
 		priv->card.dapm_routes = audio_map_tx;
@@ -676,15 +676,15 @@  static int fsl_asoc_card_probe(struct platform_device *pdev)
 		priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J |
 				SND_SOC_DAIFMT_CBC_CFC |
 				SND_SOC_DAIFMT_NB_NF;
-		priv->dai_link[1].dpcm_capture = 0;
-		priv->dai_link[2].dpcm_capture = 0;
+		priv->dai_link[1].capture_assertion = 0;
+		priv->dai_link[2].capture_assertion = 0;
 		priv->card.dapm_routes = audio_map_tx;
 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
 	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) {
 		codec_dai_name = "wm8524-hifi";
 		priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC;
-		priv->dai_link[1].dpcm_capture = 0;
-		priv->dai_link[2].dpcm_capture = 0;
+		priv->dai_link[1].capture_assertion = 0;
+		priv->dai_link[2].capture_assertion = 0;
 		priv->cpu_priv.slot_width = 32;
 		priv->card.dapm_routes = audio_map_tx;
 		priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx);
diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c
index 2aeb18397bcb..ca768375e7c7 100644
--- a/sound/soc/fsl/imx-audmix.c
+++ b/sound/soc/fsl/imx-audmix.c
@@ -258,8 +258,8 @@  static int imx_audmix_probe(struct platform_device *pdev)
 		priv->dai[i].cpus->of_node = args.np;
 		priv->dai[i].cpus->dai_name = dev_name(&cpu_pdev->dev);
 		priv->dai[i].dynamic = 1;
-		priv->dai[i].dpcm_playback = 1;
-		priv->dai[i].dpcm_capture = (i == 0 ? 1 : 0);
+		priv->dai[i].playback_assertion = 1;
+		priv->dai[i].capture_assertion = (i == 0 ? 1 : 0);
 		priv->dai[i].ignore_pmdown_time = 1;
 		priv->dai[i].ops = &imx_audmix_fe_ops;
 
@@ -283,8 +283,8 @@  static int imx_audmix_probe(struct platform_device *pdev)
 		priv->dai[num_dai + i].cpus->of_node = audmix_np;
 		priv->dai[num_dai + i].cpus->dai_name = be_name;
 		priv->dai[num_dai + i].no_pcm = 1;
-		priv->dai[num_dai + i].dpcm_playback = 1;
-		priv->dai[num_dai + i].dpcm_capture  = 1;
+		priv->dai[num_dai + i].playback_assertion = 1;
+		priv->dai[num_dai + i].capture_assertion  = 1;
 		priv->dai[num_dai + i].ignore_pmdown_time = 1;
 		priv->dai[num_dai + i].ops = &imx_audmix_be_ops;
 
diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c
index 9c7e24cebd7b..873f6f89dcf5 100644
--- a/sound/soc/fsl/imx-card.c
+++ b/sound/soc/fsl/imx-card.c
@@ -814,8 +814,8 @@  static int imx_card_probe(struct platform_device *pdev)
 		}
 		for_each_card_prelinks(&data->card, i, link) {
 			if (link->dynamic == 1 && link_be) {
-				link->dpcm_playback = link_be->dpcm_playback;
-				link->dpcm_capture = link_be->dpcm_capture;
+				link->playback_assertion = link_be->playback_assertion;
+				link->capture_assertion  = link_be->capture_assertion;
 			}
 		}
 	}