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@renesas.com --- 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; } } }