Add the rt linux 4.1.3-rt3 as base

Import the rt linux 4.1.3-rt3 as OPNFV kvm base.

It's from git://git.kernel.org/pub/scm/linux/kernel/git/rt/linux-rt-devel.git linux-4.1.y-rt and
the base is:

commit 0917f823c59692d751951bf5ea699a2d1e2f26a2
Author: Sebastian Andrzej Siewior <bigeasy@linutronix.de>
Date:   Sat Jul 25 12:13:34 2015 +0200

    Prepare v4.1.3-rt3

    Signed-off-by: Sebastian Andrzej Siewior <bigeasy@linutronix.de>

We lose all the git history this way and it's not good. We
should apply another opnfv project repo in future.

Change-Id: I87543d81c9df70d99c5001fbdf646b202c19f423
Signed-off-by: Yunhong Jiang <yunhong.jiang@intel.com>

1 files changed, 262 insertions, 0 deletions

diff --git a/kernel/sound/soc/samsung/h1940_uda1380.c b/kernel/sound/soc/samsung/h1940_uda1380.c
new file mode 100644
index 000000000..c72e9fb26
--- /dev/null
+++ b/kernel/sound/soc/samsung/h1940_uda1380.c
@@ -0,0 +1,262 @@
+/*
+ * 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/types.h>
+#include <linux/gpio.h>
+#include <linux/module.h>
+
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "regs-iis.h"
+#include <asm/mach-types.h>
+
+#include <mach/gpio-samsung.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;
+
+	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;
+	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;
+	}
+
+	/* 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(S3C_GPIO_END + 9, 1);
+	else
+		gpio_set_value(S3C_GPIO_END + 9, 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)
+{
+	snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE,
+		&hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins));
+
+	snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
+		hp_jack_gpios);
+
+	return 0;
+}
+
+static int h1940_uda1380_card_remove(struct snd_soc_card *card)
+{
+	snd_soc_jack_free_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	= "s3c24xx-iis",
+		.codec_name	= "uda1380-codec.0-001a",
+		.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBS_CFS,
+		.ops		= &h1940_ops,
+	},
+};
+
+static struct snd_soc_card h1940_asoc = {
+	.name = "h1940",
+	.owner = THIS_MODULE,
+	.remove = h1940_uda1380_card_remove,
+	.dai_link = h1940_uda1380_dai,
+	.num_links = ARRAY_SIZE(h1940_uda1380_dai),
+
+	.dapm_widgets = uda1380_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static int __init h1940_init(void)
+{
+	int ret;
+
+	if (!machine_is_h1940())
+		return -ENODEV;
+
+	/* configure some gpios */
+	ret = gpio_request(S3C_GPIO_END + 9, "speaker-power");
+	if (ret)
+		goto err_out;
+
+	ret = gpio_direction_output(S3C_GPIO_END + 9, 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(S3C_GPIO_END + 9);
+
+err_out:
+	return ret;
+}
+
+static void __exit h1940_exit(void)
+{
+	platform_device_unregister(s3c24xx_snd_device);
+	gpio_free(S3C_GPIO_END + 9);
+}
+
+module_init(h1940_init);
+module_exit(h1940_exit);
+
+/* Module information */
+MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
+MODULE_DESCRIPTION("ALSA SoC H1940");
+MODULE_LICENSE("GPL");