drivers/isdn/mISDN/dsp_audio.c

   1 /*
   2  * Audio support data for mISDN_dsp.
   3  *
   4  * Copyright 2002/2003 by Andreas Eversberg (jolly@eversberg.eu)
   5  * Rewritten by Peter
   6  *
   7  * This software may be used and distributed according to the terms
   8  * of the GNU General Public License, incorporated herein by reference.
   9  *
  10  */
  11
  12 #include <linux/delay.h>
  13 #include <linux/mISDNif.h>
  14 #include <linux/mISDNdsp.h>
  15 #include <linux/export.h>
  16 #include "core.h"
  17 #include "dsp.h"
  18
  19 /* ulaw[unsigned char] -> signed 16-bit */
  20 s32 dsp_audio_ulaw_to_s32[256];
  21 /* alaw[unsigned char] -> signed 16-bit */
  22 s32 dsp_audio_alaw_to_s32[256];
  23
  24 s32 *dsp_audio_law_to_s32;
  25 EXPORT_SYMBOL(dsp_audio_law_to_s32);
  26
  27 /* signed 16-bit -> law */
  28 u8 dsp_audio_s16_to_law[65536];
  29 EXPORT_SYMBOL(dsp_audio_s16_to_law);
  30
  31 /* alaw -> ulaw */
  32 u8 dsp_audio_alaw_to_ulaw[256];
  33 /* ulaw -> alaw */
  34 static u8 dsp_audio_ulaw_to_alaw[256];
  35 u8 dsp_silence;
  36
  37
  38 /*****************************************************
  39  * generate table for conversion of s16 to alaw/ulaw *
  40  *****************************************************/
  41
  42 #define AMI_MASK 0x55
  43
  44 static inline unsigned char linear2alaw(short int linear)
  45 {
  46         int mask;
  47         int seg;
  48         int pcm_val;
  49         static int seg_end[8] = {
  50                 0xFF, 0x1FF, 0x3FF, 0x7FF, 0xFFF, 0x1FFF, 0x3FFF, 0x7FFF
  51         };
  52
  53         pcm_val = linear;
  54         if (pcm_val >= 0) {
  55                 /* Sign (7th) bit = 1 */
  56                 mask = AMI_MASK | 0x80;
  57         } else {
  58                 /* Sign bit = 0 */
  59                 mask = AMI_MASK;
  60                 pcm_val = -pcm_val;
  61         }
  62
  63         /* Convert the scaled magnitude to segment number. */
  64         for (seg = 0;  seg < 8;  seg++) {
  65                 if (pcm_val <= seg_end[seg])
  66                         break;
  67         }
  68         /* Combine the sign, segment, and quantization bits. */
  69         return  ((seg << 4) |
  70                  ((pcm_val >> ((seg)  ?  (seg + 3)  :  4)) & 0x0F)) ^ mask;
  71 }
  72
  73
  74 static inline short int alaw2linear(unsigned char alaw)
  75 {
  76         int i;
  77         int seg;
  78
  79         alaw ^= AMI_MASK;
  80         i = ((alaw & 0x0F) << 4) + 8 /* rounding error */;
  81         seg = (((int) alaw & 0x70) >> 4);
  82         if (seg)
  83                 i = (i + 0x100) << (seg - 1);
  84         return (short int) ((alaw & 0x80)  ?  i  :  -i);
  85 }
  86
  87 static inline short int ulaw2linear(unsigned char ulaw)
  88 {
  89         short mu, e, f, y;
  90         static short etab[] = {0, 132, 396, 924, 1980, 4092, 8316, 16764};
  91
  92         mu = 255 - ulaw;
  93         e = (mu & 0x70) / 16;
  94         f = mu & 0x0f;
  95         y = f * (1 << (e + 3));
  96         y += etab[e];
  97         if (mu & 0x80)
  98                 y = -y;
  99         return y;
 100 }
 101
 102 #define BIAS 0x84   /*!< define the add-in bias for 16 bit samples */
 103
 104 static unsigned char linear2ulaw(short sample)
 105 {
 106         static int exp_lut[256] = {
 107                 0, 0, 1, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3, 3, 3, 3,
 108                 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
 109                 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
 110                 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
 111                 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
 112                 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
 113                 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
 114                 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6,
 115                 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
 116                 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
 117                 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
 118                 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
 119                 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
 120                 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
 121                 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
 122                 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7};
 123         int sign, exponent, mantissa;
 124         unsigned char ulawbyte;
 125
 126         /* Get the sample into sign-magnitude. */
 127         sign = (sample >> 8) & 0x80;      /* set aside the sign */
 128         if (sign != 0)
 129                 sample = -sample;             /* get magnitude */
 130
 131         /* Convert from 16 bit linear to ulaw. */
 132         sample = sample + BIAS;
 133         exponent = exp_lut[(sample >> 7) & 0xFF];
 134         mantissa = (sample >> (exponent + 3)) & 0x0F;
 135         ulawbyte = ~(sign | (exponent << 4) | mantissa);
 136
 137         return ulawbyte;
 138 }
 139
 140 static int reverse_bits(int i)
 141 {
 142         int z, j;
 143         z = 0;
 144
 145         for (j = 0; j < 8; j++) {
 146                 if ((i & (1 << j)) != 0)
 147                         z |= 1 << (7 - j);
 148         }
 149         return z;
 150 }
 151
 152
 153 void dsp_audio_generate_law_tables(void)
 154 {
 155         int i;
 156         for (i = 0; i < 256; i++)
 157                 dsp_audio_alaw_to_s32[i] = alaw2linear(reverse_bits(i));
 158
 159         for (i = 0; i < 256; i++)
 160                 dsp_audio_ulaw_to_s32[i] = ulaw2linear(reverse_bits(i));
 161
 162         for (i = 0; i < 256; i++) {
 163                 dsp_audio_alaw_to_ulaw[i] =
 164                         linear2ulaw(dsp_audio_alaw_to_s32[i]);
 165                 dsp_audio_ulaw_to_alaw[i] =
 166                         linear2alaw(dsp_audio_ulaw_to_s32[i]);
 167         }
 168 }
 169
 170 void
 171 dsp_audio_generate_s2law_table(void)
 172 {
 173         int i;
 174
 175         if (dsp_options & DSP_OPT_ULAW) {
 176                 /* generating ulaw-table */
 177                 for (i = -32768; i < 32768; i++) {
 178                         dsp_audio_s16_to_law[i & 0xffff] =
 179                                 reverse_bits(linear2ulaw(i));
 180                 }
 181         } else {
 182                 /* generating alaw-table */
 183                 for (i = -32768; i < 32768; i++) {
 184                         dsp_audio_s16_to_law[i & 0xffff] =
 185                                 reverse_bits(linear2alaw(i));
 186                 }
 187         }
 188 }
 189
 190
 191 /*
 192  * the seven bit sample is the number of every second alaw-sample ordered by
 193  * aplitude. 0x00 is negative, 0x7f is positive amplitude.
 194  */
 195 u8 dsp_audio_seven2law[128];
 196 u8 dsp_audio_law2seven[256];
 197
 198 /********************************************************************
 199  * generate table for conversion law from/to 7-bit alaw-like sample *
 200  ********************************************************************/
 201
 202 void
 203 dsp_audio_generate_seven(void)
 204 {
 205         int i, j, k;
 206         u8 spl;
 207         u8 sorted_alaw[256];
 208
 209         /* generate alaw table, sorted by the linear value */
 210         for (i = 0; i < 256; i++) {
 211                 j = 0;
 212                 for (k = 0; k < 256; k++) {
 213                         if (dsp_audio_alaw_to_s32[k]
 214                             < dsp_audio_alaw_to_s32[i])
 215                                 j++;
 216                 }
 217                 sorted_alaw[j] = i;
 218         }
 219
 220         /* generate tabels */
 221         for (i = 0; i < 256; i++) {
 222                 /* spl is the source: the law-sample (converted to alaw) */
 223                 spl = i;
 224                 if (dsp_options & DSP_OPT_ULAW)
 225                         spl = dsp_audio_ulaw_to_alaw[i];
 226                 /* find the 7-bit-sample */
 227                 for (j = 0; j < 256; j++) {
 228                         if (sorted_alaw[j] == spl)
 229                                 break;
 230                 }
 231                 /* write 7-bit audio value */
 232                 dsp_audio_law2seven[i] = j >> 1;
 233         }
 234         for (i = 0; i < 128; i++) {
 235                 spl = sorted_alaw[i << 1];
 236                 if (dsp_options & DSP_OPT_ULAW)
 237                         spl = dsp_audio_alaw_to_ulaw[spl];
 238                 dsp_audio_seven2law[i] = spl;
 239         }
 240 }
 241
 242
 243 /* mix 2*law -> law */
 244 u8 dsp_audio_mix_law[65536];
 245
 246 /******************************************************
 247  * generate mix table to mix two law samples into one *
 248  ******************************************************/
 249
 250 void
 251 dsp_audio_generate_mix_table(void)
 252 {
 253         int i, j;
 254         s32 sample;
 255
 256         i = 0;
 257         while (i < 256) {
 258                 j = 0;
 259                 while (j < 256) {
 260                         sample = dsp_audio_law_to_s32[i];
 261                         sample += dsp_audio_law_to_s32[j];
 262                         if (sample > 32767)
 263                                 sample = 32767;
 264                         if (sample < -32768)
 265                                 sample = -32768;
 266                         dsp_audio_mix_law[(i<<8)|j] =
 267                                 dsp_audio_s16_to_law[sample & 0xffff];
 268                         j++;
 269                 }
 270                 i++;
 271         }
 272 }
 273
 274
 275 /*************************************
 276  * generate different volume changes *
 277  *************************************/
 278
 279 static u8 dsp_audio_reduce8[256];
 280 static u8 dsp_audio_reduce7[256];
 281 static u8 dsp_audio_reduce6[256];
 282 static u8 dsp_audio_reduce5[256];
 283 static u8 dsp_audio_reduce4[256];
 284 static u8 dsp_audio_reduce3[256];
 285 static u8 dsp_audio_reduce2[256];
 286 static u8 dsp_audio_reduce1[256];
 287 static u8 dsp_audio_increase1[256];
 288 static u8 dsp_audio_increase2[256];
 289 static u8 dsp_audio_increase3[256];
 290 static u8 dsp_audio_increase4[256];
 291 static u8 dsp_audio_increase5[256];
 292 static u8 dsp_audio_increase6[256];
 293 static u8 dsp_audio_increase7[256];
 294 static u8 dsp_audio_increase8[256];
 295
 296 static u8 *dsp_audio_volume_change[16] = {
 297         dsp_audio_reduce8,
 298         dsp_audio_reduce7,
 299         dsp_audio_reduce6,
 300         dsp_audio_reduce5,
 301         dsp_audio_reduce4,
 302         dsp_audio_reduce3,
 303         dsp_audio_reduce2,
 304         dsp_audio_reduce1,
 305         dsp_audio_increase1,
 306         dsp_audio_increase2,
 307         dsp_audio_increase3,
 308         dsp_audio_increase4,
 309         dsp_audio_increase5,
 310         dsp_audio_increase6,
 311         dsp_audio_increase7,
 312         dsp_audio_increase8,
 313 };
 314
 315 void
 316 dsp_audio_generate_volume_changes(void)
 317 {
 318         register s32 sample;
 319         int i;
 320         int num[]   = { 110, 125, 150, 175, 200, 300, 400, 500 };
 321         int denum[] = { 100, 100, 100, 100, 100, 100, 100, 100 };
 322
 323         i = 0;
 324         while (i < 256) {
 325                 dsp_audio_reduce8[i] = dsp_audio_s16_to_law[
 326                         (dsp_audio_law_to_s32[i] * denum[7] / num[7]) & 0xffff];
 327                 dsp_audio_reduce7[i] = dsp_audio_s16_to_law[
 328                         (dsp_audio_law_to_s32[i] * denum[6] / num[6]) & 0xffff];
 329                 dsp_audio_reduce6[i] = dsp_audio_s16_to_law[
 330                         (dsp_audio_law_to_s32[i] * denum[5] / num[5]) & 0xffff];
 331                 dsp_audio_reduce5[i] = dsp_audio_s16_to_law[
 332                         (dsp_audio_law_to_s32[i] * denum[4] / num[4]) & 0xffff];
 333                 dsp_audio_reduce4[i] = dsp_audio_s16_to_law[
 334                         (dsp_audio_law_to_s32[i] * denum[3] / num[3]) & 0xffff];
 335                 dsp_audio_reduce3[i] = dsp_audio_s16_to_law[
 336                         (dsp_audio_law_to_s32[i] * denum[2] / num[2]) & 0xffff];
 337                 dsp_audio_reduce2[i] = dsp_audio_s16_to_law[
 338                         (dsp_audio_law_to_s32[i] * denum[1] / num[1]) & 0xffff];
 339                 dsp_audio_reduce1[i] = dsp_audio_s16_to_law[
 340                         (dsp_audio_law_to_s32[i] * denum[0] / num[0]) & 0xffff];
 341                 sample = dsp_audio_law_to_s32[i] * num[0] / denum[0];
 342                 if (sample < -32768)
 343                         sample = -32768;
 344                 else if (sample > 32767)
 345                         sample = 32767;
 346                 dsp_audio_increase1[i] = dsp_audio_s16_to_law[sample & 0xffff];
 347                 sample = dsp_audio_law_to_s32[i] * num[1] / denum[1];
 348                 if (sample < -32768)
 349                         sample = -32768;
 350                 else if (sample > 32767)
 351                         sample = 32767;
 352                 dsp_audio_increase2[i] = dsp_audio_s16_to_law[sample & 0xffff];
 353                 sample = dsp_audio_law_to_s32[i] * num[2] / denum[2];
 354                 if (sample < -32768)
 355                         sample = -32768;
 356                 else if (sample > 32767)
 357                         sample = 32767;
 358                 dsp_audio_increase3[i] = dsp_audio_s16_to_law[sample & 0xffff];
 359                 sample = dsp_audio_law_to_s32[i] * num[3] / denum[3];
 360                 if (sample < -32768)
 361                         sample = -32768;
 362                 else if (sample > 32767)
 363                         sample = 32767;
 364                 dsp_audio_increase4[i] = dsp_audio_s16_to_law[sample & 0xffff];
 365                 sample = dsp_audio_law_to_s32[i] * num[4] / denum[4];
 366                 if (sample < -32768)
 367                         sample = -32768;
 368                 else if (sample > 32767)
 369                         sample = 32767;
 370                 dsp_audio_increase5[i] = dsp_audio_s16_to_law[sample & 0xffff];
 371                 sample = dsp_audio_law_to_s32[i] * num[5] / denum[5];
 372                 if (sample < -32768)
 373                         sample = -32768;
 374                 else if (sample > 32767)
 375                         sample = 32767;
 376                 dsp_audio_increase6[i] = dsp_audio_s16_to_law[sample & 0xffff];
 377                 sample = dsp_audio_law_to_s32[i] * num[6] / denum[6];
 378                 if (sample < -32768)
 379                         sample = -32768;
 380                 else if (sample > 32767)
 381                         sample = 32767;
 382                 dsp_audio_increase7[i] = dsp_audio_s16_to_law[sample & 0xffff];
 383                 sample = dsp_audio_law_to_s32[i] * num[7] / denum[7];
 384                 if (sample < -32768)
 385                         sample = -32768;
 386                 else if (sample > 32767)
 387                         sample = 32767;
 388                 dsp_audio_increase8[i] = dsp_audio_s16_to_law[sample & 0xffff];
 389
 390                 i++;
 391         }
 392 }
 393
 394
 395 /**************************************
 396  * change the volume of the given skb *
 397  **************************************/
 398
 399 /* this is a helper function for changing volume of skb. the range may be
 400  * -8 to 8, which is a shift to the power of 2. 0 == no volume, 3 == volume*8
 401  */
 402 void
 403 dsp_change_volume(struct sk_buff *skb, int volume)
 404 {
 405         u8 *volume_change;
 406         int i, ii;
 407         u8 *p;
 408         int shift;
 409
 410         if (volume == 0)
 411                 return;
 412
 413         /* get correct conversion table */
 414         if (volume < 0) {
 415                 shift = volume + 8;
 416                 if (shift < 0)
 417                         shift = 0;
 418         } else {
 419                 shift = volume + 7;
 420                 if (shift > 15)
 421                         shift = 15;
 422         }
 423         volume_change = dsp_audio_volume_change[shift];
 424         i = 0;
 425         ii = skb->len;
 426         p = skb->data;
 427         /* change volume */
 428         while (i < ii) {
 429                 *p = volume_change[*p];
 430                 p++;
 431                 i++;
 432         }
 433 }
 434