/* * SoC audio for EDB93xx * * Copyright (c) 2010 Alexander Sverdlin * * This program is free software; you can redistribute it and/or * modify it under the terms of the GNU General Public License * as published by the Free Software Foundation; either version 2 * of the License, or (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * This driver support CS4271 codec being master or slave, working * in control port mode, connected either via SPI or I2C. * The data format accepted is I2S or left-justified. * DAPM support not implemented. */ #include #include #include #include #include #include #include #include "ep93xx-pcm.h" #define edb93xx_has_audio() (machine_is_edb9301() || \ machine_is_edb9302() || \ machine_is_edb9302a() || \ machine_is_edb9307a() || \ machine_is_edb9315a()) static int edb93xx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int err; unsigned int mclk_rate; unsigned int rate = params_rate(params); /* * According to CS4271 datasheet we use MCLK/LRCK=256 for * rates below 50kHz and 128 for higher sample rates */ if (rate < 50000) mclk_rate = rate * 64 * 4; else mclk_rate = rate * 64 * 2; err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS); if (err) return err; err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS); if (err) return err; err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate, SND_SOC_CLOCK_IN); if (err) return err; return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate, SND_SOC_CLOCK_OUT); } static struct snd_soc_ops edb93xx_ops = { .hw_params = edb93xx_hw_params, }; static struct snd_soc_dai_link edb93xx_dai = { .name = "CS4271", .stream_name = "CS4271 HiFi", .platform_name = "ep93xx-pcm-audio", .cpu_dai_name = "ep93xx-i2s", .codec_name = "spi0.0", .codec_dai_name = "cs4271-hifi", .ops = &edb93xx_ops, }; static struct snd_soc_card snd_soc_edb93xx = { .name = "EDB93XX", .dai_link = &edb93xx_dai, .num_links = 1, }; static struct platform_device *edb93xx_snd_device; static int __init edb93xx_init(void) { int ret; if (!edb93xx_has_audio()) return -ENODEV; ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97, EP93XX_SYSCON_I2SCLKDIV_ORIDE | EP93XX_SYSCON_I2SCLKDIV_SPOL); if (ret) return ret; edb93xx_snd_device = platform_device_alloc("soc-audio", -1); if (!edb93xx_snd_device) { ret = -ENOMEM; goto free_i2s; } platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx); ret = platform_device_add(edb93xx_snd_device); if (ret) goto device_put; return 0; device_put: platform_device_put(edb93xx_snd_device); free_i2s: ep93xx_i2s_release(); return ret; } module_init(edb93xx_init); static void __exit edb93xx_exit(void) { platform_device_unregister(edb93xx_snd_device); ep93xx_i2s_release(); } module_exit(edb93xx_exit); MODULE_AUTHOR("Alexander Sverdlin "); MODULE_DESCRIPTION("ALSA SoC EDB93xx"); MODULE_LICENSE("GPL");