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, 179 insertions, 0 deletions

diff --git a/kernel/Documentation/sound/alsa/soc/codec.txt b/kernel/Documentation/sound/alsa/soc/codec.txt
new file mode 100644
index 000000000..db5f9c9ae
--- /dev/null
+++ b/kernel/Documentation/sound/alsa/soc/codec.txt
@@ -0,0 +1,179 @@
+ASoC Codec Class Driver
+=======================
+
+The codec class driver is generic and hardware independent code that configures
+the codec, FM, MODEM, BT or external DSP to provide audio capture and playback.
+It should contain no code that is specific to the target platform or machine.
+All platform and machine specific code should be added to the platform and
+machine drivers respectively.
+
+Each codec class driver *must* provide the following features:-
+
+ 1) Codec DAI and PCM configuration
+ 2) Codec control IO - using RegMap API
+ 3) Mixers and audio controls
+ 4) Codec audio operations
+ 5) DAPM description.
+ 6) DAPM event handler.
+
+Optionally, codec drivers can also provide:-
+
+ 7) DAC Digital mute control.
+
+Its probably best to use this guide in conjunction with the existing codec
+driver code in sound/soc/codecs/
+
+ASoC Codec driver breakdown
+===========================
+
+1 - Codec DAI and PCM configuration
+-----------------------------------
+Each codec driver must have a struct snd_soc_dai_driver to define its DAI and
+PCM capabilities and operations. This struct is exported so that it can be
+registered with the core by your machine driver.
+
+e.g.
+
+static struct snd_soc_dai_ops wm8731_dai_ops = {
+	.prepare	= wm8731_pcm_prepare,
+	.hw_params	= wm8731_hw_params,
+	.shutdown	= wm8731_shutdown,
+	.digital_mute	= wm8731_mute,
+	.set_sysclk	= wm8731_set_dai_sysclk,
+	.set_fmt	= wm8731_set_dai_fmt,
+};
+
+struct snd_soc_dai_driver wm8731_dai = {
+	.name = "wm8731-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8731_RATES,
+		.formats = WM8731_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8731_RATES,
+		.formats = WM8731_FORMATS,},
+	.ops = &wm8731_dai_ops,
+	.symmetric_rates = 1,
+};
+
+
+2 - Codec control IO
+--------------------
+The codec can usually be controlled via an I2C or SPI style interface
+(AC97 combines control with data in the DAI). The codec driver should use the
+Regmap API for all codec IO. Please see include/linux/regmap.h and existing
+codec drivers for example regmap usage.
+
+
+3 - Mixers and audio controls
+-----------------------------
+All the codec mixers and audio controls can be defined using the convenience
+macros defined in soc.h.
+
+    #define SOC_SINGLE(xname, reg, shift, mask, invert)
+
+Defines a single control as follows:-
+
+  xname = Control name e.g. "Playback Volume"
+  reg = codec register
+  shift = control bit(s) offset in register
+  mask = control bit size(s) e.g. mask of 7 = 3 bits
+  invert = the control is inverted
+
+Other macros include:-
+
+    #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert)
+
+A stereo control
+
+    #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert)
+
+A stereo control spanning 2 registers
+
+    #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts)
+
+Defines an single enumerated control as follows:-
+
+   xreg = register
+   xshift = control bit(s) offset in register
+   xmask = control bit(s) size
+   xtexts = pointer to array of strings that describe each setting
+
+   #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts)
+
+Defines a stereo enumerated control
+
+
+4 - Codec Audio Operations
+--------------------------
+The codec driver also supports the following ALSA PCM operations:-
+
+/* SoC audio ops */
+struct snd_soc_ops {
+	int (*startup)(struct snd_pcm_substream *);
+	void (*shutdown)(struct snd_pcm_substream *);
+	int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *);
+	int (*hw_free)(struct snd_pcm_substream *);
+	int (*prepare)(struct snd_pcm_substream *);
+};
+
+Please refer to the ALSA driver PCM documentation for details.
+http://www.alsa-project.org/~iwai/writing-an-alsa-driver/
+
+
+5 - DAPM description.
+---------------------
+The Dynamic Audio Power Management description describes the codec power
+components and their relationships and registers to the ASoC core.
+Please read dapm.txt for details of building the description.
+
+Please also see the examples in other codec drivers.
+
+
+6 - DAPM event handler
+----------------------
+This function is a callback that handles codec domain PM calls and system
+domain PM calls (e.g. suspend and resume). It is used to put the codec
+to sleep when not in use.
+
+Power states:-
+
+	SNDRV_CTL_POWER_D0: /* full On */
+	/* vref/mid, clk and osc on, active */
+
+	SNDRV_CTL_POWER_D1: /* partial On */
+	SNDRV_CTL_POWER_D2: /* partial On */
+
+	SNDRV_CTL_POWER_D3hot: /* Off, with power */
+	/* everything off except vref/vmid, inactive */
+
+	SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */
+
+
+7 - Codec DAC digital mute control
+----------------------------------
+Most codecs have a digital mute before the DACs that can be used to
+minimise any system noise.  The mute stops any digital data from
+entering the DAC.
+
+A callback can be created that is called by the core for each codec DAI
+when the mute is applied or freed.
+
+i.e.
+
+static int wm8974_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf;
+
+	if (mute)
+		snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40);
+	else
+		snd_soc_write(codec, WM8974_DAC, mute_reg);
+	return 0;
+}