/*
 * 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 <linux/bitrev.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;
}

void dsp_audio_generate_law_tables(void)
{
        int i;
        for (i = 0; i < 256; i++)
                dsp_audio_alaw_to_s32[i] = alaw2linear(bitrev8((u8)i));

        for (i = 0; i < 256; i++)
                dsp_audio_ulaw_to_s32[i] = ulaw2linear(bitrev8((u8)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] =
                                bitrev8(linear2ulaw(i));
                }
        } else {
                /* generating alaw-table */
                for (i = -32768; i < 32768; i++) {
                        dsp_audio_s16_to_law[i & 0xffff] =
                                bitrev8(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++;
        }
}