libmpcodecs/ad_hwac3.c

   1
   2 // Reference: DOCS/tech/hwac3.txt !!!!!
   3
   4 /* DTS code based on "ac3/decode_dts.c" and "ac3/conversion.c" from "ogle 0.9"
   5    (see http://www.dtek.chalmers.se/~dvd/)
   6 */
   7
   8 #define _XOPEN_SOURCE 600
   9 #include <stdio.h>
  10 #include <stdlib.h>
  11 #include <string.h>
  12 #include <unistd.h>
  13
  14 #include "config.h"
  15 #include "mp_msg.h"
  16 #include "help_mp.h"
  17 #include "mpbswap.h"
  18
  19 #include "ad_internal.h"
  20
  21 #ifdef CONFIG_LIBA52_INTERNAL
  22 #include "liba52/a52.h"
  23 #else
  24 #include <a52dec/a52.h>
  25 #endif
  26
  27
  28 static int isdts = -1;
  29
  30 static ad_info_t info =
  31 {
  32   "AC3/DTS pass-through S/PDIF",
  33   "hwac3",
  34   "Nick Kurshev/Peter Schüller",
  35   "???",
  36   ""
  37 };
  38
  39 LIBAD_EXTERN(hwac3)
  40
  41
  42 static int dts_syncinfo(uint8_t *indata_ptr, int *flags, int *sample_rate, int *bit_rate);
  43 static int decode_audio_dts(unsigned char *indata_ptr, int len, unsigned char *buf);
  44
  45
  46 static int ac3dts_fillbuff(sh_audio_t *sh_audio)
  47 {
  48   int length = 0;
  49   int flags = 0;
  50   int sample_rate = 0;
  51   int bit_rate = 0;
  52
  53   sh_audio->a_in_buffer_len = 0;
  54   /* sync frame:*/
  55   while(1)
  56   {
  57     // Original code DTS has a 10 bytes header.
  58     // Now max 12 bytes for 14 bits DTS header.
  59     while(sh_audio->a_in_buffer_len < 12)
  60     {
  61       int c = demux_getc(sh_audio->ds);
  62       if(c<0)
  63         return -1; /* EOF*/
  64       sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++] = c;
  65     }
  66
  67     if (sh_audio->format == 0x2001)
  68     {
  69       length = dts_syncinfo(sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
  70       if(length >= 12)
  71       {
  72         if(isdts != 1)
  73         {
  74           mp_msg(MSGT_DECAUDIO, MSGL_STATUS, "hwac3: switched to DTS, %d bps, %d Hz\n", bit_rate, sample_rate);
  75           isdts = 1;
  76         }
  77         break;
  78       }
  79     }
  80     else
  81     {
  82       length = a52_syncinfo(sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
  83       if(length >= 7 && length <= 3840)
  84       {
  85         if(isdts != 0)
  86         {
  87           mp_msg(MSGT_DECAUDIO, MSGL_STATUS, "hwac3: switched to AC3, %d bps, %d Hz\n", bit_rate, sample_rate);
  88           isdts = 0;
  89         }
  90         break; /* we're done.*/
  91       }
  92     }
  93     /* bad file => resync*/
  94     memcpy(sh_audio->a_in_buffer, sh_audio->a_in_buffer + 1, 11);
  95     --sh_audio->a_in_buffer_len;
  96   }
  97   mp_msg(MSGT_DECAUDIO, MSGL_DBG2, "ac3dts: %s len=%d  flags=0x%X  %d Hz %d bit/s\n", isdts == 1 ? "DTS" : isdts == 0 ? "AC3" : "unknown", length, flags, sample_rate, bit_rate);
  98
  99   sh_audio->samplerate = sample_rate;
 100   sh_audio->i_bps = bit_rate / 8;
 101   demux_read_data(sh_audio->ds, sh_audio->a_in_buffer + 12, length - 12);
 102   sh_audio->a_in_buffer_len = length;
 103
 104   // TODO: is DTS also checksummed?
 105 #ifdef CONFIG_LIBA52_INTERNAL
 106   if(isdts == 0 && crc16_block(sh_audio->a_in_buffer + 2, length - 2) != 0)
 107     mp_msg(MSGT_DECAUDIO, MSGL_STATUS, "a52: CRC check failed!  \n");
 108 #endif
 109
 110   return length;
 111 }
 112
 113
 114 static int preinit(sh_audio_t *sh)
 115 {
 116   /* Dolby AC3 audio: */
 117   sh->audio_out_minsize = 128 * 32 * 2 * 2; // DTS seems to need more than AC3
 118   sh->audio_in_minsize = 8192;
 119   sh->channels = 2;
 120   sh->samplesize = 2;
 121   sh->sample_format = AF_FORMAT_AC3;
 122   return 1;
 123 }
 124
 125 static int init(sh_audio_t *sh_audio)
 126 {
 127   /* Dolby AC3 passthrough:*/
 128   a52_state_t *a52_state = a52_init(0);
 129   if(a52_state == NULL)
 130   {
 131     mp_msg(MSGT_DECAUDIO, MSGL_ERR, "A52 init failed\n");
 132     return 0;
 133   }
 134   if(ac3dts_fillbuff(sh_audio) < 0)
 135   {
 136     a52_free(a52_state);
 137     mp_msg(MSGT_DECAUDIO, MSGL_ERR, "AC3/DTS sync failed\n");
 138     return 0;
 139   }
 140   sh_audio->context = a52_state;
 141   return 1;
 142 }
 143
 144 static void uninit(sh_audio_t *sh)
 145 {
 146   a52_free(sh->context);
 147 }
 148
 149 static int control(sh_audio_t *sh,int cmd,void* arg, ...)
 150 {
 151   switch(cmd)
 152   {
 153   case ADCTRL_RESYNC_STREAM:
 154   case ADCTRL_SKIP_FRAME:
 155       ac3dts_fillbuff(sh);
 156       return CONTROL_TRUE;
 157   }
 158   return CONTROL_UNKNOWN;
 159 }
 160
 161
 162 static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
 163 {
 164   int len = sh_audio->a_in_buffer_len;
 165
 166   if(len <= 0)
 167     if((len = ac3dts_fillbuff(sh_audio)) <= 0)
 168       return len; /*EOF*/
 169   sh_audio->a_in_buffer_len = 0;
 170
 171   if(isdts == 1)
 172   {
 173     return decode_audio_dts(sh_audio->a_in_buffer, len, buf);
 174   }
 175   else if(isdts == 0)
 176   {
 177     uint16_t *buf16 = (uint16_t *)buf;
 178     buf16[0] = 0xF872;   // iec 61937 syncword 1
 179     buf16[1] = 0x4E1F;   // iec 61937 syncword 2
 180     buf16[2] = 0x0001;   // data-type ac3
 181     buf16[2] |= (sh_audio->a_in_buffer[5] & 0x7) << 8; // bsmod
 182     buf16[3] = len << 3; // number of bits in payload
 183 #ifdef WORDS_BIGENDIAN
 184     memcpy(buf + 8, sh_audio->a_in_buffer, len);
 185 #else
 186     swab(sh_audio->a_in_buffer, buf + 8, len);
 187     if (len & 1) {
 188       buf[8+len-1] = 0;
 189       buf[8+len] = sh_audio->a_in_buffer[len-1];
 190       len++;
 191     }
 192 #endif
 193     memset(buf + 8 + len, 0, 6144 - 8 - len);
 194
 195     return 6144;
 196   }
 197   else
 198     return -1;
 199 }
 200
 201
 202 static const int DTS_SAMPLEFREQS[16] =
 203 {
 204   0,
 205   8000,
 206   16000,
 207   32000,
 208   64000,
 209   128000,
 210   11025,
 211   22050,
 212   44100,
 213   88200,
 214   176400,
 215   12000,
 216   24000,
 217   48000,
 218   96000,
 219   192000
 220 };
 221
 222 static const int DTS_BITRATES[30] =
 223 {
 224   32000,
 225   56000,
 226   64000,
 227   96000,
 228   112000,
 229   128000,
 230   192000,
 231   224000,
 232   256000,
 233   320000,
 234   384000,
 235   448000,
 236   512000,
 237   576000,
 238   640000,
 239   768000,
 240   896000,
 241   1024000,
 242   1152000,
 243   1280000,
 244   1344000,
 245   1408000,
 246   1411200,
 247   1472000,
 248   1536000,
 249   1920000,
 250   2048000,
 251   3072000,
 252   3840000,
 253   4096000
 254 };
 255
 256 static int dts_decode_header(uint8_t *indata_ptr, int *rate, int *nblks, int *sfreq)
 257 {
 258   int ftype;
 259   int surp;
 260   int unknown_bit;
 261   int fsize;
 262   int amode;
 263
 264   int word_mode;
 265   int le_mode;
 266
 267   unsigned int first4bytes = indata_ptr[0] << 24 | indata_ptr[1] << 16
 268                              | indata_ptr[2] << 8 | indata_ptr[3];
 269
 270   switch(first4bytes)
 271   {
 272     /* 14 bits LE */
 273     case 0xff1f00e8:
 274       /* Also make sure frame type is 1. */
 275       if ((indata_ptr[4]&0xf0) != 0xf0 || indata_ptr[5] != 0x07)
 276         return -1;
 277       word_mode = 0;
 278       le_mode = 1;
 279       break;
 280     /* 14 bits BE */
 281     case 0x1fffe800:
 282       /* Also make sure frame type is 1. */
 283       if (indata_ptr[4] != 0x07 || (indata_ptr[5]&0xf0) != 0xf0)
 284         return -1;
 285       word_mode = 0;
 286       le_mode = 0;
 287       break;
 288     /* 16 bits LE */
 289     case 0xfe7f0180:
 290       word_mode = 1;
 291       le_mode = 1;
 292       break;
 293     /* 16 bits BE */
 294     case 0x7ffe8001:
 295       word_mode = 1;
 296       le_mode = 0;
 297       break;
 298     default:
 299       return -1;
 300   }
 301
 302   if(word_mode)
 303   {
 304     /* First bit after first 32 bits:
 305        Frame type ( 1: Normal frame; 0: Termination frame ) */
 306     ftype = indata_ptr[4+le_mode] >> 7;
 307
 308   if(ftype != 1)
 309   {
 310     mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: Termination frames not handled, REPORT BUG\n");
 311     return -1;
 312   }
 313     /* Next 5 bits: Surplus Sample Count V SURP 5 bits */
 314     surp = indata_ptr[4+le_mode] >> 2 & 0x1f;
 315     /* Number of surplus samples */
 316     surp = (surp + 1) % 32;
 317
 318     /* One unknown bit, crc? */
 319     unknown_bit = indata_ptr[4+le_mode] >> 1 & 0x01;
 320
 321     /* NBLKS 7 bits: Valid Range=5-127, Invalid Range=0-4 */
 322     *nblks = (indata_ptr[4+le_mode] & 0x01) << 6 | indata_ptr[5-le_mode] >> 2;
 323     /* NBLKS+1 indicates the number of 32 sample PCM audio blocks per channel
 324        encoded in the current frame per channel. */
 325     ++(*nblks);
 326
 327     /* Frame Byte Size V FSIZE 14 bits: 0-94=Invalid, 95-8191=Valid range-1
 328        (ie. 96 bytes to 8192 bytes), 8192-16383=Invalid
 329        FSIZE defines the byte size of the current audio frame. */
 330     fsize = (indata_ptr[5-le_mode] & 0x03) << 12 | indata_ptr[6+le_mode] << 4
 331             | indata_ptr[7-le_mode] >> 4;
 332     ++fsize;
 333
 334     /* Audio Channel Arrangement ACC AMODE 6 bits */
 335     amode = (indata_ptr[7-le_mode] & 0x0f) << 2 | indata_ptr[8+le_mode] >> 6;
 336
 337     /* Source Sampling rate ACC SFREQ 4 bits */
 338     *sfreq = indata_ptr[8+le_mode] >> 2 & 0x0f;
 339     /* Transmission Bit Rate ACC RATE 5 bits */
 340     *rate = (indata_ptr[8+le_mode] & 0x03) << 3
 341             | (indata_ptr[9-le_mode] >> 5 & 0x07);
 342   }
 343   else
 344   {
 345     /* in the case judgement, we assure this */
 346     ftype = 1;
 347     surp = 0;
 348     /* 14 bits support, every 2 bytes, & 0x3fff, got used 14 bits */
 349     /* Bits usage:
 350        32 bits: Sync code (28 + 4)      1th and 2th word, 4 bits in 3th word
 351        1  bits: Frame type              1 bits in 3th word
 352        5  bits: SURP                    5 bits in 3th word
 353        1  bits: crc?                    1 bits in 3th word
 354        7  bits: NBLKS                   3 bits in 3th word, 4 bits in 4th word
 355        14 bits: FSIZE                   10 bits in 4th word, 4 bits in 5th word
 356                                         in 14 bits mode, FSIZE = FSIZE*8/14*2
 357        6  bits: AMODE                   6 bits in 5th word
 358        4  bits: SFREQ                   4 bits in 5th word
 359        5  bits: RATE                    5 bits in 6th word
 360        total bits: 75 bits    */
 361
 362     /* NBLKS 7 bits: Valid Range=5-127, Invalid Range=0-4 */
 363     *nblks = (indata_ptr[5-le_mode] & 0x07) << 4
 364              | (indata_ptr[6+le_mode] & 0x3f) >> 2;
 365     /* NBLKS+1 indicates the number of 32 sample PCM audio blocks per channel
 366        encoded in the current frame per channel. */
 367     ++(*nblks);
 368
 369     /* Frame Byte Size V FSIZE 14 bits: 0-94=Invalid, 95-8191=Valid range-1
 370        (ie. 96 bytes to 8192 bytes), 8192-16383=Invalid
 371        FSIZE defines the byte size of the current audio frame. */
 372     fsize = (indata_ptr[6+le_mode] & 0x03) << 12 | indata_ptr[7-le_mode] << 4
 373             | (indata_ptr[8+le_mode] & 0x3f) >> 2;
 374     ++fsize;
 375     fsize = fsize * 8 / 14 * 2;
 376
 377     /* Audio Channel Arrangement ACC AMODE 6 bits */
 378     amode = (indata_ptr[8+le_mode] & 0x03) << 4
 379             | (indata_ptr[9-le_mode] & 0xf0) >> 4;
 380
 381     /* Source Sampling rate ACC SFREQ 4 bits */
 382     *sfreq = indata_ptr[9-le_mode] & 0x0f;
 383     /* Transmission Bit Rate ACC RATE 5 bits */
 384     *rate = (indata_ptr[10+le_mode] & 0x3f) >> 1;
 385   }
 386 #if 0
 387   if(*sfreq != 13)
 388   {
 389     mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: Only 48kHz supported, REPORT BUG\n");
 390     return -1;
 391   }
 392 #endif
 393   if((fsize > 8192) || (fsize < 96))
 394   {
 395     mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: fsize: %d invalid, REPORT BUG\n", fsize);
 396     return -1;
 397   }
 398
 399   if(*nblks != 8 &&
 400     *nblks != 16 &&
 401     *nblks != 32 &&
 402     *nblks != 64 &&
 403     *nblks != 128 &&
 404     ftype == 1)
 405   {
 406     mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: nblks %d not valid for normal frame, REPORT BUG\n", *nblks);
 407     return -1;
 408   }
 409
 410   return fsize;
 411 }
 412
 413 static int dts_syncinfo(uint8_t *indata_ptr, int *flags, int *sample_rate, int *bit_rate)
 414 {
 415   int nblks;
 416   int fsize;
 417   int rate;
 418   int sfreq;
 419
 420   fsize = dts_decode_header(indata_ptr, &rate, &nblks, &sfreq);
 421   if(fsize >= 0)
 422   {
 423     if(rate >= 0 && rate <= 29)
 424       *bit_rate = DTS_BITRATES[rate];
 425     else
 426       *bit_rate = 0;
 427     if(sfreq >= 1 && sfreq <= 15)
 428       *sample_rate = DTS_SAMPLEFREQS[sfreq];
 429     else
 430       *sample_rate = 0;
 431   }
 432   return fsize;
 433 }
 434
 435 static int convert_14bits_to_16bits(const unsigned char *src,
 436                                     unsigned char *dest,
 437                                     int len,
 438                                     int is_le)
 439 {
 440   uint16_t *p = (uint16_t *)dest;
 441   uint16_t buf = 0;
 442   int spacebits = 16;
 443   if (len <= 0) return 0;
 444   while (len > 0) {
 445     uint16_t v;
 446     if (len == 1)
 447       v = is_le ? src[0] : src[0] << 8;
 448     else
 449       v = is_le ? src[1] << 8 | src[0] : src[0] << 8 | src[1];
 450     v <<= 2;
 451     src += 2;
 452     len -= 2;
 453     buf |= v >> (16 - spacebits);
 454     spacebits -= 14;
 455     if (spacebits < 0) {
 456       *p++ = buf;
 457       spacebits += 16;
 458       buf = v << (spacebits - 2);
 459     }
 460   }
 461   *p++ = buf;
 462   return (unsigned char *)p - dest;
 463 }
 464
 465 static int decode_audio_dts(unsigned char *indata_ptr, int len, unsigned char *buf)
 466 {
 467   int nblks;
 468   int fsize;
 469   int rate;
 470   int sfreq;
 471   int nr_samples;
 472   int convert_16bits = 0;
 473   uint16_t *buf16 = (uint16_t *)buf;
 474
 475   fsize = dts_decode_header(indata_ptr, &rate, &nblks, &sfreq);
 476   if(fsize < 0)
 477     return -1;
 478   nr_samples = nblks * 32;
 479
 480   buf16[0] = 0xf872; /* iec 61937     */
 481   buf16[1] = 0x4e1f; /*  syncword     */
 482   switch(nr_samples)
 483   {
 484   case 512:
 485     buf16[2] = 0x000b;      /* DTS-1 (512-sample bursts) */
 486     break;
 487   case 1024:
 488     buf16[2] = 0x000c;      /* DTS-2 (1024-sample bursts) */
 489     break;
 490   case 2048:
 491     buf16[2] = 0x000d;      /* DTS-3 (2048-sample bursts) */
 492     break;
 493   default:
 494     mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: %d-sample bursts not supported\n", nr_samples);
 495     buf16[2] = 0x0000;
 496     break;
 497   }
 498
 499   if(fsize + 8 > nr_samples * 2 * 2)
 500   {
 501     // dts wav (14bits LE) match this condition, one way to passthrough
 502     // is not add iec 61937 header, decoders will notice the dts header
 503     // and identify the dts stream. Another way here is convert
 504     // the stream from 14 bits to 16 bits.
 505     if ((indata_ptr[0] == 0xff || indata_ptr[0] == 0x1f)
 506         && fsize * 14 / 16 + 8 <= nr_samples * 2 * 2) {
 507       // The input stream is 14 bits, we can shrink it to 16 bits
 508       // to save space for add the 61937 header
 509       fsize = convert_14bits_to_16bits(indata_ptr,
 510                                        &buf[8],
 511                                        fsize,
 512                                        indata_ptr[0] == 0xff /* is LE */
 513                                        );
 514       mp_msg(MSGT_DECAUDIO, MSGL_DBG3, "DTS: shrink 14 bits stream to "
 515              "16 bits %02x%02x%02x%02x => %02x%02x%02x%02x, new size %d.\n",
 516              indata_ptr[0], indata_ptr[1], indata_ptr[2], indata_ptr[3],
 517              buf[8], buf[9], buf[10], buf[11], fsize);
 518       convert_16bits = 1;
 519     }
 520     else
 521     mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: more data than fits\n");
 522   }
 523
 524   buf16[3] = fsize << 3;
 525
 526   if (!convert_16bits) {
 527 #ifdef WORDS_BIGENDIAN
 528   /* BE stream */
 529   if (indata_ptr[0] == 0x1f || indata_ptr[0] == 0x7f)
 530 #else
 531   /* LE stream */
 532   if (indata_ptr[0] == 0xff || indata_ptr[0] == 0xfe)
 533 #endif
 534   memcpy(&buf[8], indata_ptr, fsize);
 535   else
 536   {
 537   swab(indata_ptr, &buf[8], fsize);
 538   if (fsize & 1) {
 539     buf[8+fsize-1] = 0;
 540     buf[8+fsize] = indata_ptr[fsize-1];
 541     fsize++;
 542   }
 543   }
 544   }
 545   memset(&buf[fsize + 8], 0, nr_samples * 2 * 2 - (fsize + 8));
 546
 547   return nr_samples * 2 * 2;
 548 }