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, 183 insertions, 0 deletions

diff --git a/kernel/sound/soc/blackfin/bfin-eval-adau1373.c b/kernel/sound/soc/blackfin/bfin-eval-adau1373.c
new file mode 100644
index 000000000..523baf582
--- /dev/null
+++ b/kernel/sound/soc/blackfin/bfin-eval-adau1373.c
@@ -0,0 +1,183 @@
+/*
+ * Machine driver for EVAL-ADAU1373 on Analog Devices bfin
+ * evaluation boards.
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "../codecs/adau1373.h"
+
+static const struct snd_soc_dapm_widget bfin_eval_adau1373_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("Line In1", NULL),
+	SND_SOC_DAPM_LINE("Line In2", NULL),
+	SND_SOC_DAPM_LINE("Line In3", NULL),
+	SND_SOC_DAPM_LINE("Line In4", NULL),
+
+	SND_SOC_DAPM_LINE("Line Out1", NULL),
+	SND_SOC_DAPM_LINE("Line Out2", NULL),
+	SND_SOC_DAPM_LINE("Stereo Out", NULL),
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_HP("Earpiece", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_soc_dapm_route bfin_eval_adau1373_dapm_routes[] = {
+	{ "AIN1L", NULL, "Line In1" },
+	{ "AIN1R", NULL, "Line In1" },
+	{ "AIN2L", NULL, "Line In2" },
+	{ "AIN2R", NULL, "Line In2" },
+	{ "AIN3L", NULL, "Line In3" },
+	{ "AIN3R", NULL, "Line In3" },
+	{ "AIN4L", NULL, "Line In4" },
+	{ "AIN4R", NULL, "Line In4" },
+
+	/* MICBIAS can be connected via a jumper to the line-in jack, since w
+	   don't know which one is going to be used, just power both. */
+	{ "Line In1", NULL, "MICBIAS1" },
+	{ "Line In2", NULL, "MICBIAS1" },
+	{ "Line In3", NULL, "MICBIAS1" },
+	{ "Line In4", NULL, "MICBIAS1" },
+	{ "Line In1", NULL, "MICBIAS2" },
+	{ "Line In2", NULL, "MICBIAS2" },
+	{ "Line In3", NULL, "MICBIAS2" },
+	{ "Line In4", NULL, "MICBIAS2" },
+
+	{ "Line Out1", NULL, "LOUT1L" },
+	{ "Line Out1", NULL, "LOUT1R" },
+	{ "Line Out2", NULL, "LOUT2L" },
+	{ "Line Out2", NULL, "LOUT2R" },
+	{ "Headphone", NULL, "HPL" },
+	{ "Headphone", NULL, "HPR" },
+	{ "Earpiece", NULL, "EP" },
+	{ "Speaker", NULL, "SPKL" },
+	{ "Stereo Out", NULL, "SPKR" },
+};
+
+static int bfin_eval_adau1373_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;
+	int ret;
+	int pll_rate;
+
+	switch (params_rate(params)) {
+	case 48000:
+	case 8000:
+	case 12000:
+	case 16000:
+	case 24000:
+	case 32000:
+		pll_rate = 48000 * 1024;
+		break;
+	case 44100:
+	case 7350:
+	case 11025:
+	case 14700:
+	case 22050:
+	case 29400:
+		pll_rate = 44100 * 1024;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1,
+			ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate,
+			SND_SOC_CLOCK_IN);
+
+	return ret;
+}
+
+static int bfin_eval_adau1373_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	unsigned int pll_rate = 48000 * 1024;
+	int ret;
+
+	ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1,
+			ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate,
+			SND_SOC_CLOCK_IN);
+
+	return ret;
+}
+static struct snd_soc_ops bfin_eval_adau1373_ops = {
+	.hw_params = bfin_eval_adau1373_hw_params,
+};
+
+static struct snd_soc_dai_link bfin_eval_adau1373_dai = {
+	.name = "adau1373",
+	.stream_name = "adau1373",
+	.cpu_dai_name = "bfin-i2s.0",
+	.codec_dai_name = "adau1373-aif1",
+	.platform_name = "bfin-i2s-pcm-audio",
+	.codec_name = "adau1373.0-001a",
+	.ops = &bfin_eval_adau1373_ops,
+	.init = bfin_eval_adau1373_codec_init,
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBM_CFM,
+};
+
+static struct snd_soc_card bfin_eval_adau1373 = {
+	.name = "bfin-eval-adau1373",
+	.owner = THIS_MODULE,
+	.dai_link = &bfin_eval_adau1373_dai,
+	.num_links = 1,
+
+	.dapm_widgets		= bfin_eval_adau1373_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(bfin_eval_adau1373_dapm_widgets),
+	.dapm_routes		= bfin_eval_adau1373_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(bfin_eval_adau1373_dapm_routes),
+};
+
+static int bfin_eval_adau1373_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &bfin_eval_adau1373;
+
+	card->dev = &pdev->dev;
+
+	return snd_soc_register_card(&bfin_eval_adau1373);
+}
+
+static int bfin_eval_adau1373_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static struct platform_driver bfin_eval_adau1373_driver = {
+	.driver = {
+		.name = "bfin-eval-adau1373",
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = bfin_eval_adau1373_probe,
+	.remove = bfin_eval_adau1373_remove,
+};
+
+module_platform_driver(bfin_eval_adau1373_driver);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ALSA SoC bfin adau1373 driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:bfin-eval-adau1373");