soc_get_playback_capture() is now handling DPCM and normal comprehensively
for playback/capture stream. We can use playback/capture_only flag
instead of using dpcm_playback/capture. This patch replace these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
---
sound/soc/fsl/fsl-asoc-card.c | 16 ++++++----------
sound/soc/fsl/imx-audmix.c | 6 ++----
sound/soc/fsl/imx-card.c | 4 ++--
3 files changed, 10 insertions(+), 16 deletions(-)
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 40870668ee24..917d9da5c57f 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -311,8 +311,6 @@ 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,
.dynamic = 1,
SND_SOC_DAILINK_REG(hifi_fe),
},
@@ -321,8 +319,6 @@ 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,
.no_pcm = 1,
SND_SOC_DAILINK_REG(hifi_be),
},
@@ -633,8 +629,8 @@ 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].playback_only = 1;
+ priv->dai_link[2].playback_only = 1;
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;
@@ -660,15 +656,15 @@ 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].playback_only = 1;
+ priv->dai_link[2].playback_only = 1;
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].playback_only = 1;
+ priv->dai_link[2].playback_only = 1;
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 efbcd4a65ca8..5cf7bb861698 100644
--- a/sound/soc/fsl/imx-audmix.c
+++ b/sound/soc/fsl/imx-audmix.c
@@ -255,10 +255,10 @@ 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].ignore_pmdown_time = 1;
priv->dai[i].ops = &imx_audmix_fe_ops;
+ if (i)
+ priv->dai[i].playback_only = 1;
/* Add AUDMIX Backend */
be_name = devm_kasprintf(&pdev->dev, GFP_KERNEL,
@@ -278,8 +278,6 @@ 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].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 78e2e3932ba5..6e3ce0817478 100644
--- a/sound/soc/fsl/imx-card.c
+++ b/sound/soc/fsl/imx-card.c
@@ -826,8 +826,8 @@ 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_only = link_be->playback_only;
+ link->capture_only = link_be->capture_only;
}
}
}
--
2.25.1
