From f70575805708cabdedea7498aaa3f710fde4d920 Mon Sep 17 00:00:00 2001 From: hc <hc@nodka.com> Date: Wed, 31 Jan 2024 03:29:01 +0000 Subject: [PATCH] add lvds1024*800 --- kernel/sound/soc/fsl/p1022_rdk.c | 73 +++++++++++++++++++++++------------- 1 files changed, 46 insertions(+), 27 deletions(-) diff --git a/kernel/sound/soc/fsl/p1022_rdk.c b/kernel/sound/soc/fsl/p1022_rdk.c index 4afbdd6..714515b 100644 --- a/kernel/sound/soc/fsl/p1022_rdk.c +++ b/kernel/sound/soc/fsl/p1022_rdk.c @@ -1,21 +1,17 @@ -/** - * Freescale P1022RDK ALSA SoC Machine driver - * - * Author: Timur Tabi <timur@freescale.com> - * - * Copyright 2012 Freescale Semiconductor, Inc. - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - * - * Note: in order for audio to work correctly, the output controls need - * to be enabled, because they control the clock. So for playback, for - * example: - * - * amixer sset 'Left Output Mixer PCM' on - * amixer sset 'Right Output Mixer PCM' on - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale P1022RDK ALSA SoC Machine driver +// +// Author: Timur Tabi <timur@freescale.com> +// +// Copyright 2012 Freescale Semiconductor, Inc. +// +// Note: in order for audio to work correctly, the output controls need +// to be enabled, because they control the clock. So for playback, for +// example: +// +// amixer sset 'Left Output Mixer PCM' on +// amixer sset 'Right Output Mixer PCM' on #include <linux/module.h> #include <linux/fsl/guts.h> @@ -131,21 +127,21 @@ */ static int p1022_rdk_startup(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct machine_data *mdata = container_of(rtd->card, struct machine_data, card); struct device *dev = rtd->card->dev; int ret = 0; /* Tell the codec driver what the serial protocol is. */ - ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format); + ret = snd_soc_dai_set_fmt(asoc_rtd_to_codec(rtd, 0), mdata->dai_format); if (ret < 0) { dev_err(dev, "could not set codec driver audio format (ret=%i)\n", ret); return ret; } - ret = snd_soc_dai_set_pll(rtd->codec_dai, 0, 0, mdata->clk_frequency, + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), 0, 0, mdata->clk_frequency, mdata->clk_frequency); if (ret < 0) { dev_err(dev, "could not set codec PLL frequency (ret=%i)\n", @@ -207,6 +203,7 @@ struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; struct machine_data *mdata; + struct snd_soc_dai_link_component *comp; const u32 *iprop; int ret; @@ -223,11 +220,33 @@ goto error_put; } - mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); + comp = devm_kzalloc(&pdev->dev, 6 * sizeof(*comp), GFP_KERNEL); + if (!comp) { + ret = -ENOMEM; + goto error_put; + } + + mdata->dai[0].cpus = &comp[0]; + mdata->dai[0].codecs = &comp[1]; + mdata->dai[0].platforms = &comp[2]; + + mdata->dai[0].num_cpus = 1; + mdata->dai[0].num_codecs = 1; + mdata->dai[0].num_platforms = 1; + + mdata->dai[1].cpus = &comp[3]; + mdata->dai[1].codecs = &comp[4]; + mdata->dai[1].platforms = &comp[5]; + + mdata->dai[1].num_cpus = 1; + mdata->dai[1].num_codecs = 1; + mdata->dai[1].num_platforms = 1; + + mdata->dai[0].cpus->dai_name = dev_name(&ssi_pdev->dev); mdata->dai[0].ops = &p1022_rdk_ops; /* ASoC core can match codec with device node */ - mdata->dai[0].codec_of_node = codec_np; + mdata->dai[0].codecs->of_node = codec_np; /* * We register two DAIs per SSI, one for playback and the other for @@ -237,8 +256,8 @@ memcpy(&mdata->dai[1], &mdata->dai[0], sizeof(struct snd_soc_dai_link)); /* The DAI names from the codec (snd_soc_dai_driver.name) */ - mdata->dai[0].codec_dai_name = "wm8960-hifi"; - mdata->dai[1].codec_dai_name = mdata->dai[0].codec_dai_name; + mdata->dai[0].codecs->dai_name = "wm8960-hifi"; + mdata->dai[1].codecs->dai_name = mdata->dai[0].codecs->dai_name; /* * Configure the SSI for I2S slave mode. Older device trees have @@ -270,7 +289,7 @@ } /* Find the playback DMA channel to use. */ - mdata->dai[0].platform_name = mdata->platform_name[0]; + mdata->dai[0].platforms->name = mdata->platform_name[0]; ret = fsl_asoc_get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0], &mdata->dma_channel_id[0], &mdata->dma_id[0]); @@ -281,7 +300,7 @@ } /* Find the capture DMA channel to use. */ - mdata->dai[1].platform_name = mdata->platform_name[1]; + mdata->dai[1].platforms->name = mdata->platform_name[1]; ret = fsl_asoc_get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1], &mdata->dma_channel_id[1], &mdata->dma_id[1]); -- Gitblit v1.6.2