path: root/drivers/isdn/mISDN/dsp_audio.c

/*
 * 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++;
	}
}