From mboxrd@z Thu Jan 1 00:00:00 1970 From: Mark Brown Subject: Applied "ASoC: fsl: imx-audmix: don't select unnecessary Platform" to the asoc tree Date: Wed, 19 Jun 2019 13:12:02 +0100 (BST) Message-ID: <20190619121202.969B0440049@finisterre.sirena.org.uk> References: <87imt2v0iy.wl-kuninori.morimoto.gx@renesas.com> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Return-path: Received: from heliosphere.sirena.org.uk (heliosphere.sirena.org.uk [IPv6:2a01:7e01::f03c:91ff:fed4:a3b6]) (using TLSv1.2 with cipher ECDHE-RSA-AES256-GCM-SHA384 (256/256 bits)) (No client certificate requested) by alsa1.perex.cz (Postfix) with ESMTPS id 3074DF8973F for ; Wed, 19 Jun 2019 14:12:04 +0200 (CEST) In-Reply-To: <87imt2v0iy.wl-kuninori.morimoto.gx@renesas.com> List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: alsa-devel-bounces@alsa-project.org Sender: "Alsa-devel" To: Kuninori Morimoto Cc: Linux-ALSA , Mark Brown List-Id: alsa-devel@alsa-project.org The patch ASoC: fsl: imx-audmix: don't select unnecessary Platform has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.3 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >>From d8893261a7d327302520eb9bebb72c5040c2219f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 19 Jun 2019 10:17:10 +0900 Subject: [PATCH] ASoC: fsl: imx-audmix: don't select unnecessary Platform ALSA SoC is now supporting "no Platform". Sound card doesn't need to select "CPU component" as "Platform" anymore if it doesn't need special Platform. This patch removes such settings. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmix.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index 9e1cb18859ce..9d41266a5264 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -207,8 +207,8 @@ static int imx_audmix_probe(struct platform_device *pdev) for (i = 0; i < num_dai; i++) { struct snd_soc_dai_link_component *dlc; - /* for CPU/Codec/Platform x 2 */ - dlc = devm_kzalloc(&pdev->dev, 6 * sizeof(*dlc), GFP_KERNEL); + /* for CPU/Codec x 2 */ + dlc = devm_kzalloc(&pdev->dev, 4 * sizeof(*dlc), GFP_KERNEL); if (!dlc) { dev_err(&pdev->dev, "failed to allocate dai_link\n"); return -ENOMEM; @@ -242,11 +242,9 @@ static int imx_audmix_probe(struct platform_device *pdev) priv->dai[i].cpus = &dlc[0]; priv->dai[i].codecs = &dlc[1]; - priv->dai[i].platforms = &dlc[2]; priv->dai[i].num_cpus = 1; priv->dai[i].num_codecs = 1; - priv->dai[i].num_platforms = 1; priv->dai[i].name = dai_name; priv->dai[i].stream_name = "HiFi-AUDMIX-FE"; @@ -254,7 +252,6 @@ static int imx_audmix_probe(struct platform_device *pdev) priv->dai[i].codecs->name = "snd-soc-dummy"; priv->dai[i].cpus->of_node = args.np; priv->dai[i].cpus->dai_name = dev_name(&cpu_pdev->dev); - priv->dai[i].platforms->of_node = args.np; priv->dai[i].dynamic = 1; priv->dai[i].dpcm_playback = 1; priv->dai[i].dpcm_capture = (i == 0 ? 1 : 0); @@ -269,20 +266,17 @@ static int imx_audmix_probe(struct platform_device *pdev) be_cp = devm_kasprintf(&pdev->dev, GFP_KERNEL, "AUDMIX-Capture-%d", i); - priv->dai[num_dai + i].cpus = &dlc[3]; - priv->dai[num_dai + i].codecs = &dlc[4]; - priv->dai[num_dai + i].platforms = &dlc[5]; + priv->dai[num_dai + i].cpus = &dlc[2]; + priv->dai[num_dai + i].codecs = &dlc[3]; priv->dai[num_dai + i].num_cpus = 1; priv->dai[num_dai + i].num_codecs = 1; - priv->dai[num_dai + i].num_platforms = 1; priv->dai[num_dai + i].name = be_name; priv->dai[num_dai + i].codecs->dai_name = "snd-soc-dummy-dai"; priv->dai[num_dai + i].codecs->name = "snd-soc-dummy"; 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].platforms->name = "snd-soc-dummy"; priv->dai[num_dai + i].no_pcm = 1; priv->dai[num_dai + i].dpcm_playback = 1; priv->dai[num_dai + i].dpcm_capture = 1; -- 2.20.1