CHROMIUM: ASoC: Import entire upstream ASoC tree

sound/soc/ep93xx/edb93xx.c

+/*
+ * SoC audio for EDB93xx
+ *
+ * Copyright (c) 2010 Alexander Sverdlin <subaparts@yandex.ru>
+ *
+ * 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 <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#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 rate = params_rate(params);
+        /*
+         * We set LRCLK equal to `rate' and SCLK = LRCLK * 64,
+         * because our sample size is 32 bit * 2 channels.
+         * I2S standard permits us to transmit more bits than
+         * the codec uses.
+         * MCLK = SCLK * 4 is the best recommended value,
+         * but we have to fall back to ratio 2 for higher
+         * sample rates.
+         */
+        unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 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 <subaparts@yandex.ru>");
+MODULE_DESCRIPTION("ALSA SoC EDB93xx");
+MODULE_LICENSE("GPL");