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, 433 insertions, 0 deletions

diff --git a/kernel/drivers/isdn/mISDN/dsp_audio.c b/kernel/drivers/isdn/mISDN/dsp_audio.c
new file mode 100644
index 000000000..06022952a
--- /dev/null
+++ b/kernel/drivers/isdn/mISDN/dsp_audio.c
@@ -0,0 +1,433 @@
+/*
+ * Audio support data for mISDN_dsp.
+ *
+ * Copyright 2002/2003 by Andreas Eversberg (jolly@eversberg.eu)
+ * Rewritten by Peter
+ *
+ * This software may be used and distributed according to the terms
+ * of the GNU General Public License, incorporated herein by reference.
+ *
+ */
+
+#include <linux/delay.h>
+#include <linux/mISDNif.h>
+#include <linux/mISDNdsp.h>
+#include <linux/export.h>
+#include "core.h"
+#include "dsp.h"
+
+/* ulaw[unsigned char] -> signed 16-bit */
+s32 dsp_audio_ulaw_to_s32[256];
+/* alaw[unsigned char] -> signed 16-bit */
+s32 dsp_audio_alaw_to_s32[256];
+
+s32 *dsp_audio_law_to_s32;
+EXPORT_SYMBOL(dsp_audio_law_to_s32);
+
+/* signed 16-bit -> law */
+u8 dsp_audio_s16_to_law[65536];
+EXPORT_SYMBOL(dsp_audio_s16_to_law);
+
+/* alaw -> ulaw */
+u8 dsp_audio_alaw_to_ulaw[256];
+/* ulaw -> alaw */
+static u8 dsp_audio_ulaw_to_alaw[256];
+u8 dsp_silence;
+
+
+/*****************************************************
+ * generate table for conversion of s16 to alaw/ulaw *
+ *****************************************************/
+
+#define AMI_MASK 0x55
+
+static inline unsigned char linear2alaw(short int linear)
+{
+	int mask;
+	int seg;
+	int pcm_val;
+	static int seg_end[8] = {
+		0xFF, 0x1FF, 0x3FF, 0x7FF, 0xFFF, 0x1FFF, 0x3FFF, 0x7FFF
+	};
+
+	pcm_val = linear;
+	if (pcm_val >= 0) {
+		/* Sign (7th) bit = 1 */
+		mask = AMI_MASK | 0x80;
+	} else {
+		/* Sign bit = 0 */
+		mask = AMI_MASK;
+		pcm_val = -pcm_val;
+	}
+
+	/* Convert the scaled magnitude to segment number. */
+	for (seg = 0; seg < 8; seg++) {
+		if (pcm_val <= seg_end[seg])
+			break;
+	}
+	/* Combine the sign, segment, and quantization bits. */
+	return  ((seg << 4) |
+		 ((pcm_val >> ((seg)  ?  (seg + 3)  :  4)) & 0x0F)) ^ mask;
+}
+
+
+static inline short int alaw2linear(unsigned char alaw)
+{
+	int i;
+	int seg;
+
+	alaw ^= AMI_MASK;
+	i = ((alaw & 0x0F) << 4) + 8 /* rounding error */;
+	seg = (((int) alaw & 0x70) >> 4);
+	if (seg)
+		i = (i + 0x100) << (seg - 1);
+	return (short int) ((alaw & 0x80)  ?  i  :  -i);
+}
+
+static inline short int ulaw2linear(unsigned char ulaw)
+{
+	short mu, e, f, y;
+	static short etab[] = {0, 132, 396, 924, 1980, 4092, 8316, 16764};
+
+	mu = 255 - ulaw;
+	e = (mu & 0x70) / 16;
+	f = mu & 0x0f;
+	y = f * (1 << (e + 3));
+	y += etab[e];
+	if (mu & 0x80)
+		y = -y;
+	return y;
+}
+
+#define BIAS 0x84   /*!< define the add-in bias for 16 bit samples */
+
+static unsigned char linear2ulaw(short sample)
+{
+	static int exp_lut[256] = {
+		0, 0, 1, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3,
+		4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
+		5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+		5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
+		6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
+		6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
+		6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
+		6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
+		7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
+		7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
+		7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
+		7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
+		7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
+		7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
+		7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
+		7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7};
+	int sign, exponent, mantissa;
+	unsigned char ulawbyte;
+
+	/* Get the sample into sign-magnitude. */
+	sign = (sample >> 8) & 0x80;	  /* set aside the sign */
+	if (sign != 0)
+		sample = -sample;	      /* get magnitude */
+
+	/* Convert from 16 bit linear to ulaw. */
+	sample = sample + BIAS;
+	exponent = exp_lut[(sample >> 7) & 0xFF];
+	mantissa = (sample >> (exponent + 3)) & 0x0F;
+	ulawbyte = ~(sign | (exponent << 4) | mantissa);
+
+	return ulawbyte;
+}
+
+static int reverse_bits(int i)
+{
+	int z, j;
+	z = 0;
+
+	for (j = 0; j < 8; j++) {
+		if ((i & (1 << j)) != 0)
+			z |= 1 << (7 - j);
+	}
+	return z;
+}
+
+
+void dsp_audio_generate_law_tables(void)
+{
+	int i;
+	for (i = 0; i < 256; i++)
+		dsp_audio_alaw_to_s32[i] = alaw2linear(reverse_bits(i));
+
+	for (i = 0; i < 256; i++)
+		dsp_audio_ulaw_to_s32[i] = ulaw2linear(reverse_bits(i));
+
+	for (i = 0; i < 256; i++) {
+		dsp_audio_alaw_to_ulaw[i] =
+			linear2ulaw(dsp_audio_alaw_to_s32[i]);
+		dsp_audio_ulaw_to_alaw[i] =
+			linear2alaw(dsp_audio_ulaw_to_s32[i]);
+	}
+}
+
+void
+dsp_audio_generate_s2law_table(void)
+{
+	int i;
+
+	if (dsp_options & DSP_OPT_ULAW) {
+		/* generating ulaw-table */
+		for (i = -32768; i < 32768; i++) {
+			dsp_audio_s16_to_law[i & 0xffff] =
+				reverse_bits(linear2ulaw(i));
+		}
+	} else {
+		/* generating alaw-table */
+		for (i = -32768; i < 32768; i++) {
+			dsp_audio_s16_to_law[i & 0xffff] =
+				reverse_bits(linear2alaw(i));
+		}
+	}
+}
+
+
+/*
+ * the seven bit sample is the number of every second alaw-sample ordered by
+ * aplitude. 0x00 is negative, 0x7f is positive amplitude.
+ */
+u8 dsp_audio_seven2law[128];
+u8 dsp_audio_law2seven[256];
+
+/********************************************************************
+ * generate table for conversion law from/to 7-bit alaw-like sample *
+ ********************************************************************/
+
+void
+dsp_audio_generate_seven(void)
+{
+	int i, j, k;
+	u8 spl;
+	u8 sorted_alaw[256];
+
+	/* generate alaw table, sorted by the linear value */
+	for (i = 0; i < 256; i++) {
+		j = 0;
+		for (k = 0; k < 256; k++) {
+			if (dsp_audio_alaw_to_s32[k]
+			    < dsp_audio_alaw_to_s32[i])
+				j++;
+		}
+		sorted_alaw[j] = i;
+	}
+
+	/* generate tabels */
+	for (i = 0; i < 256; i++) {
+		/* spl is the source: the law-sample (converted to alaw) */
+		spl = i;
+		if (dsp_options & DSP_OPT_ULAW)
+			spl = dsp_audio_ulaw_to_alaw[i];
+		/* find the 7-bit-sample */
+		for (j = 0; j < 256; j++) {
+			if (sorted_alaw[j] == spl)
+				break;
+		}
+		/* write 7-bit audio value */
+		dsp_audio_law2seven[i] = j >> 1;
+	}
+	for (i = 0; i < 128; i++) {
+		spl = sorted_alaw[i << 1];
+		if (dsp_options & DSP_OPT_ULAW)
+			spl = dsp_audio_alaw_to_ulaw[spl];
+		dsp_audio_seven2law[i] = spl;
+	}
+}
+
+
+/* mix 2*law -> law */
+u8 dsp_audio_mix_law[65536];
+
+/******************************************************
+ * generate mix table to mix two law samples into one *
+ ******************************************************/
+
+void
+dsp_audio_generate_mix_table(void)
+{
+	int i, j;
+	s32 sample;
+
+	i = 0;
+	while (i < 256) {
+		j = 0;
+		while (j < 256) {
+			sample = dsp_audio_law_to_s32[i];
+			sample += dsp_audio_law_to_s32[j];
+			if (sample > 32767)
+				sample = 32767;
+			if (sample < -32768)
+				sample = -32768;
+			dsp_audio_mix_law[(i << 8) | j] =
+				dsp_audio_s16_to_law[sample & 0xffff];
+			j++;
+		}
+		i++;
+	}
+}
+
+
+/*************************************
+ * generate different volume changes *
+ *************************************/
+
+static u8 dsp_audio_reduce8[256];
+static u8 dsp_audio_reduce7[256];
+static u8 dsp_audio_reduce6[256];
+static u8 dsp_audio_reduce5[256];
+static u8 dsp_audio_reduce4[256];
+static u8 dsp_audio_reduce3[256];
+static u8 dsp_audio_reduce2[256];
+static u8 dsp_audio_reduce1[256];
+static u8 dsp_audio_increase1[256];
+static u8 dsp_audio_increase2[256];
+static u8 dsp_audio_increase3[256];
+static u8 dsp_audio_increase4[256];
+static u8 dsp_audio_increase5[256];
+static u8 dsp_audio_increase6[256];
+static u8 dsp_audio_increase7[256];
+static u8 dsp_audio_increase8[256];
+
+static u8 *dsp_audio_volume_change[16] = {
+	dsp_audio_reduce8,
+	dsp_audio_reduce7,
+	dsp_audio_reduce6,
+	dsp_audio_reduce5,
+	dsp_audio_reduce4,
+	dsp_audio_reduce3,
+	dsp_audio_reduce2,
+	dsp_audio_reduce1,
+	dsp_audio_increase1,
+	dsp_audio_increase2,
+	dsp_audio_increase3,
+	dsp_audio_increase4,
+	dsp_audio_increase5,
+	dsp_audio_increase6,
+	dsp_audio_increase7,
+	dsp_audio_increase8,
+};
+
+void
+dsp_audio_generate_volume_changes(void)
+{
+	register s32 sample;
+	int i;
+	int num[]   = { 110, 125, 150, 175, 200, 300, 400, 500 };
+	int denum[] = { 100, 100, 100, 100, 100, 100, 100, 100 };
+
+	i = 0;
+	while (i < 256) {
+		dsp_audio_reduce8[i] = dsp_audio_s16_to_law[
+			(dsp_audio_law_to_s32[i] * denum[7] / num[7]) & 0xffff];
+		dsp_audio_reduce7[i] = dsp_audio_s16_to_law[
+			(dsp_audio_law_to_s32[i] * denum[6] / num[6]) & 0xffff];
+		dsp_audio_reduce6[i] = dsp_audio_s16_to_law[
+			(dsp_audio_law_to_s32[i] * denum[5] / num[5]) & 0xffff];
+		dsp_audio_reduce5[i] = dsp_audio_s16_to_law[
+			(dsp_audio_law_to_s32[i] * denum[4] / num[4]) & 0xffff];
+		dsp_audio_reduce4[i] = dsp_audio_s16_to_law[
+			(dsp_audio_law_to_s32[i] * denum[3] / num[3]) & 0xffff];
+		dsp_audio_reduce3[i] = dsp_audio_s16_to_law[
+			(dsp_audio_law_to_s32[i] * denum[2] / num[2]) & 0xffff];
+		dsp_audio_reduce2[i] = dsp_audio_s16_to_law[
+			(dsp_audio_law_to_s32[i] * denum[1] / num[1]) & 0xffff];
+		dsp_audio_reduce1[i] = dsp_audio_s16_to_law[
+			(dsp_audio_law_to_s32[i] * denum[0] / num[0]) & 0xffff];
+		sample = dsp_audio_law_to_s32[i] * num[0] / denum[0];
+		if (sample < -32768)
+			sample = -32768;
+		else if (sample > 32767)
+			sample = 32767;
+		dsp_audio_increase1[i] = dsp_audio_s16_to_law[sample & 0xffff];
+		sample = dsp_audio_law_to_s32[i] * num[1] / denum[1];
+		if (sample < -32768)
+			sample = -32768;
+		else if (sample > 32767)
+			sample = 32767;
+		dsp_audio_increase2[i] = dsp_audio_s16_to_law[sample & 0xffff];
+		sample = dsp_audio_law_to_s32[i] * num[2] / denum[2];
+		if (sample < -32768)
+			sample = -32768;
+		else if (sample > 32767)
+			sample = 32767;
+		dsp_audio_increase3[i] = dsp_audio_s16_to_law[sample & 0xffff];
+		sample = dsp_audio_law_to_s32[i] * num[3] / denum[3];
+		if (sample < -32768)
+			sample = -32768;
+		else if (sample > 32767)
+			sample = 32767;
+		dsp_audio_increase4[i] = dsp_audio_s16_to_law[sample & 0xffff];
+		sample = dsp_audio_law_to_s32[i] * num[4] / denum[4];
+		if (sample < -32768)
+			sample = -32768;
+		else if (sample > 32767)
+			sample = 32767;
+		dsp_audio_increase5[i] = dsp_audio_s16_to_law[sample & 0xffff];
+		sample = dsp_audio_law_to_s32[i] * num[5] / denum[5];
+		if (sample < -32768)
+			sample = -32768;
+		else if (sample > 32767)
+			sample = 32767;
+		dsp_audio_increase6[i] = dsp_audio_s16_to_law[sample & 0xffff];
+		sample = dsp_audio_law_to_s32[i] * num[6] / denum[6];
+		if (sample < -32768)
+			sample = -32768;
+		else if (sample > 32767)
+			sample = 32767;
+		dsp_audio_increase7[i] = dsp_audio_s16_to_law[sample & 0xffff];
+		sample = dsp_audio_law_to_s32[i] * num[7] / denum[7];
+		if (sample < -32768)
+			sample = -32768;
+		else if (sample > 32767)
+			sample = 32767;
+		dsp_audio_increase8[i] = dsp_audio_s16_to_law[sample & 0xffff];
+
+		i++;
+	}
+}
+
+
+/**************************************
+ * change the volume of the given skb *
+ **************************************/
+
+/* this is a helper function for changing volume of skb. the range may be
+ * -8 to 8, which is a shift to the power of 2. 0 == no volume, 3 == volume*8
+ */
+void
+dsp_change_volume(struct sk_buff *skb, int volume)
+{
+	u8 *volume_change;
+	int i, ii;
+	u8 *p;
+	int shift;
+
+	if (volume == 0)
+		return;
+
+	/* get correct conversion table */
+	if (volume < 0) {
+		shift = volume + 8;
+		if (shift < 0)
+			shift = 0;
+	} else {
+		shift = volume + 7;
+		if (shift > 15)
+			shift = 15;
+	}
+	volume_change = dsp_audio_volume_change[shift];
+	i = 0;
+	ii = skb->len;
+	p = skb->data;
+	/* change volume */
+	while (i < ii) {
+		*p = volume_change[*p];
+		p++;
+		i++;
+	}
+}