text: /*
   * 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++;
  	}
  }
  