1 files changed, 287 insertions, 0 deletions

diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c
new file mode 100644
index 000000000000..241f55d00660
--- /dev/null
+++ b/sound/soc/samsung/h1940_uda1380.c
@@ -0,0 +1,287 @@
+/*
+ * h1940-uda1380.c  --  ALSA Soc Audio Layer
+ *
+ * Copyright (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
+ * Copyright (c) 2010 Vasily Khoruzhick <anarsoul@gmail.com>
+ *
+ * Based on version from Arnaud Patard <arnaud.patard@rtp-net.org>
+ *
+ * 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.
+ *
+ */
+
+#include <linux/gpio.h>
+
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include <plat/regs-iis.h>
+#include <mach/h1940-latch.h>
+#include <asm/mach-types.h>
+
+#include "s3c24xx-i2s.h"
+
+static unsigned int rates[] = {
+        11025,
+        22050,
+        44100,
+};
+
+static struct snd_pcm_hw_constraint_list hw_rates = {
+        .count = ARRAY_SIZE(rates),
+        .list = rates,
+        .mask = 0,
+};
+
+static struct snd_soc_jack hp_jack;
+
+static struct snd_soc_jack_pin hp_jack_pins[] = {
+        {
+                .pin    = "Headphone Jack",
+                .mask   = SND_JACK_HEADPHONE,
+        },
+        {
+                .pin    = "Speaker",
+                .mask   = SND_JACK_HEADPHONE,
+                .invert = 1,
+        },
+};
+
+static struct snd_soc_jack_gpio hp_jack_gpios[] = {
+        {
+                .gpio                   = S3C2410_GPG(4),
+                .name                   = "hp-gpio",
+                .report                 = SND_JACK_HEADPHONE,
+                .invert                 = 1,
+                .debounce_time          = 200,
+        },
+};
+
+static int h1940_startup(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+
+        runtime->hw.rate_min = hw_rates.list[0];
+        runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
+        runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
+
+        return snd_pcm_hw_constraint_list(runtime, 0,
+                                        SNDRV_PCM_HW_PARAM_RATE,
+                                        &hw_rates);
+}
+
+static int h1940_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 *cpu_dai = rtd->cpu_dai;
+        struct snd_soc_dai *codec_dai = rtd->codec_dai;
+        int div;
+        int ret;
+        unsigned int rate = params_rate(params);
+
+        switch (rate) {
+        case 11025:
+        case 22050:
+        case 44100:
+                div = s3c24xx_i2s_get_clockrate() / (384 * rate);
+                if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate))
+                        div++;
+                break;
+        default:
+                dev_err(&rtd->dev, "%s: rate %d is not supported\n",
+                        __func__, rate);
+                return -EINVAL;
+        }
+
+        /* set codec DAI configuration */
+        ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+                SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+        if (ret < 0)
+                return ret;
+
+        /* set cpu DAI configuration */
+        ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+                SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+        if (ret < 0)
+                return ret;
+
+        /* select clock source */
+        ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate,
+                        SND_SOC_CLOCK_OUT);
+        if (ret < 0)
+                return ret;
+
+        /* set MCLK division for sample rate */
+        ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
+                S3C2410_IISMOD_384FS);
+        if (ret < 0)
+                return ret;
+
+        /* set BCLK division for sample rate */
+        ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
+                S3C2410_IISMOD_32FS);
+        if (ret < 0)
+                return ret;
+
+        /* set prescaler division for sample rate */
+        ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
+                S3C24XX_PRESCALE(div, div));
+        if (ret < 0)
+                return ret;
+
+        return 0;
+}
+
+static struct snd_soc_ops h1940_ops = {
+        .startup        = h1940_startup,
+        .hw_params      = h1940_hw_params,
+};
+
+static int h1940_spk_power(struct snd_soc_dapm_widget *w,
+                                struct snd_kcontrol *kcontrol, int event)
+{
+        if (SND_SOC_DAPM_EVENT_ON(event))
+                gpio_set_value(H1940_LATCH_AUDIO_POWER, 1);
+        else
+                gpio_set_value(H1940_LATCH_AUDIO_POWER, 0);
+
+        return 0;
+}
+
+/* h1940 machine dapm widgets */
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+        SND_SOC_DAPM_HP("Headphone Jack", NULL),
+        SND_SOC_DAPM_MIC("Mic Jack", NULL),
+        SND_SOC_DAPM_SPK("Speaker", h1940_spk_power),
+};
+
+/* h1940 machine audio_map */
+static const struct snd_soc_dapm_route audio_map[] = {
+        /* headphone connected to VOUTLHP, VOUTRHP */
+        {"Headphone Jack", NULL, "VOUTLHP"},
+        {"Headphone Jack", NULL, "VOUTRHP"},
+
+        /* ext speaker connected to VOUTL, VOUTR  */
+        {"Speaker", NULL, "VOUTL"},
+        {"Speaker", NULL, "VOUTR"},
+
+        /* mic is connected to VINM */
+        {"VINM", NULL, "Mic Jack"},
+};
+
+static struct platform_device *s3c24xx_snd_device;
+
+static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
+{
+        struct snd_soc_codec *codec = rtd->codec;
+        struct snd_soc_dapm_context *dapm = &codec->dapm;
+        int err;
+
+        /* Add h1940 specific widgets */
+        err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
+                                  ARRAY_SIZE(uda1380_dapm_widgets));
+        if (err)
+                return err;
+
+        /* Set up h1940 specific audio path audio_mapnects */
+        err = snd_soc_dapm_add_routes(dapm, audio_map,
+                                      ARRAY_SIZE(audio_map));
+        if (err)
+                return err;
+
+        snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+        snd_soc_dapm_enable_pin(dapm, "Speaker");
+        snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+
+        snd_soc_dapm_sync(dapm);
+
+        snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
+                &hp_jack);
+
+        snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
+                hp_jack_pins);
+
+        snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
+                hp_jack_gpios);
+
+        return 0;
+}
+
+/* s3c24xx digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link h1940_uda1380_dai[] = {
+        {
+                .name           = "uda1380",
+                .stream_name    = "UDA1380 Duplex",
+                .cpu_dai_name   = "s3c24xx-iis",
+                .codec_dai_name = "uda1380-hifi",
+                .init           = h1940_uda1380_init,
+                .platform_name  = "samsung-audio",
+                .codec_name     = "uda1380-codec.0-001a",
+                .ops            = &h1940_ops,
+        },
+};
+
+static struct snd_soc_card h1940_asoc = {
+        .name = "h1940",
+        .dai_link = h1940_uda1380_dai,
+        .num_links = ARRAY_SIZE(h1940_uda1380_dai),
+};
+
+static int __init h1940_init(void)
+{
+        int ret;
+
+        if (!machine_is_h1940())
+                return -ENODEV;
+
+        /* configure some gpios */
+        ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power");
+        if (ret)
+                goto err_out;
+
+        ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0);
+        if (ret)
+                goto err_gpio;
+
+        s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
+        if (!s3c24xx_snd_device) {
+                ret = -ENOMEM;
+                goto err_gpio;
+        }
+
+        platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc);
+        ret = platform_device_add(s3c24xx_snd_device);
+
+        if (ret)
+                goto err_plat;
+
+        return 0;
+
+err_plat:
+        platform_device_put(s3c24xx_snd_device);
+err_gpio:
+        gpio_free(H1940_LATCH_AUDIO_POWER);
+
+err_out:
+        return ret;
+}
+
+static void __exit h1940_exit(void)
+{
+        platform_device_unregister(s3c24xx_snd_device);
+        snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
+                hp_jack_gpios);
+        gpio_free(H1940_LATCH_AUDIO_POWER);
+}
+
+module_init(h1940_init);
+module_exit(h1940_exit);
+
+/* Module information */
+MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
+MODULE_DESCRIPTION("ALSA SoC H1940");
+MODULE_LICENSE("GPL");