release/src/router/ffmpeg/libavcodec/alacenc.c

   1 /**
   2  * ALAC audio encoder
   3  * Copyright (c) 2008  Jaikrishnan Menon <realityman@gmx.net>
   4  *
   5  * This file is part of FFmpeg.
   6  *
   7  * FFmpeg is free software; you can redistribute it and/or
   8  * modify it under the terms of the GNU Lesser General Public
   9  * License as published by the Free Software Foundation; either
  10  * version 2.1 of the License, or (at your option) any later version.
  11  *
  12  * FFmpeg is distributed in the hope that it will be useful,
  13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15  * Lesser General Public License for more details.
  16  *
  17  * You should have received a copy of the GNU Lesser General Public
  18  * License along with FFmpeg; if not, write to the Free Software
  19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20  */
  21
  22 #include "avcodec.h"
  23 #include "put_bits.h"
  24 #include "dsputil.h"
  25 #include "lpc.h"
  26 #include "mathops.h"
  27
  28 #define DEFAULT_FRAME_SIZE        4096
  29 #define DEFAULT_SAMPLE_SIZE       16
  30 #define MAX_CHANNELS              8
  31 #define ALAC_EXTRADATA_SIZE       36
  32 #define ALAC_FRAME_HEADER_SIZE    55
  33 #define ALAC_FRAME_FOOTER_SIZE    3
  34
  35 #define ALAC_ESCAPE_CODE          0x1FF
  36 #define ALAC_MAX_LPC_ORDER        30
  37 #define DEFAULT_MAX_PRED_ORDER    6
  38 #define DEFAULT_MIN_PRED_ORDER    4
  39 #define ALAC_MAX_LPC_PRECISION    9
  40 #define ALAC_MAX_LPC_SHIFT        9
  41
  42 #define ALAC_CHMODE_LEFT_RIGHT    0
  43 #define ALAC_CHMODE_LEFT_SIDE     1
  44 #define ALAC_CHMODE_RIGHT_SIDE    2
  45 #define ALAC_CHMODE_MID_SIDE      3
  46
  47 typedef struct RiceContext {
  48     int history_mult;
  49     int initial_history;
  50     int k_modifier;
  51     int rice_modifier;
  52 } RiceContext;
  53
  54 typedef struct LPCContext {
  55     int lpc_order;
  56     int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
  57     int lpc_quant;
  58 } LPCContext;
  59
  60 typedef struct AlacEncodeContext {
  61     int compression_level;
  62     int min_prediction_order;
  63     int max_prediction_order;
  64     int max_coded_frame_size;
  65     int write_sample_size;
  66     int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE];
  67     int32_t predictor_buf[DEFAULT_FRAME_SIZE];
  68     int interlacing_shift;
  69     int interlacing_leftweight;
  70     PutBitContext pbctx;
  71     RiceContext rc;
  72     LPCContext lpc[MAX_CHANNELS];
  73     DSPContext dspctx;
  74     AVCodecContext *avctx;
  75 } AlacEncodeContext;
  76
  77
  78 static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples)
  79 {
  80     int ch, i;
  81
  82     for(ch=0;ch<s->avctx->channels;ch++) {
  83         int16_t *sptr = input_samples + ch;
  84         for(i=0;i<s->avctx->frame_size;i++) {
  85             s->sample_buf[ch][i] = *sptr;
  86             sptr += s->avctx->channels;
  87         }
  88     }
  89 }
  90
  91 static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
  92 {
  93     int divisor, q, r;
  94
  95     k = FFMIN(k, s->rc.k_modifier);
  96     divisor = (1<<k) - 1;
  97     q = x / divisor;
  98     r = x % divisor;
  99
 100     if(q > 8) {
 101         // write escape code and sample value directly
 102         put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);
 103         put_bits(&s->pbctx, write_sample_size, x);
 104     } else {
 105         if(q)
 106             put_bits(&s->pbctx, q, (1<<q) - 1);
 107         put_bits(&s->pbctx, 1, 0);
 108
 109         if(k != 1) {
 110             if(r > 0)
 111                 put_bits(&s->pbctx, k, r+1);
 112             else
 113                 put_bits(&s->pbctx, k-1, 0);
 114         }
 115     }
 116 }
 117
 118 static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
 119 {
 120     put_bits(&s->pbctx, 3,  s->avctx->channels-1);          // No. of channels -1
 121     put_bits(&s->pbctx, 16, 0);                             // Seems to be zero
 122     put_bits(&s->pbctx, 1,  1);                             // Sample count is in the header
 123     put_bits(&s->pbctx, 2,  0);                             // FIXME: Wasted bytes field
 124     put_bits(&s->pbctx, 1,  is_verbatim);                   // Audio block is verbatim
 125     put_bits32(&s->pbctx, s->avctx->frame_size);            // No. of samples in the frame
 126 }
 127
 128 static void calc_predictor_params(AlacEncodeContext *s, int ch)
 129 {
 130     int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];
 131     int shift[MAX_LPC_ORDER];
 132     int opt_order;
 133
 134     if (s->compression_level == 1) {
 135         s->lpc[ch].lpc_order = 6;
 136         s->lpc[ch].lpc_quant = 6;
 137         s->lpc[ch].lpc_coeff[0] =  160;
 138         s->lpc[ch].lpc_coeff[1] = -190;
 139         s->lpc[ch].lpc_coeff[2] =  170;
 140         s->lpc[ch].lpc_coeff[3] = -130;
 141         s->lpc[ch].lpc_coeff[4] =   80;
 142         s->lpc[ch].lpc_coeff[5] =  -25;
 143     } else {
 144         opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch],
 145                                       s->avctx->frame_size,
 146                                       s->min_prediction_order,
 147                                       s->max_prediction_order,
 148                                       ALAC_MAX_LPC_PRECISION, coefs, shift, 1,
 149                                       ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
 150
 151         s->lpc[ch].lpc_order = opt_order;
 152         s->lpc[ch].lpc_quant = shift[opt_order-1];
 153         memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));
 154     }
 155 }
 156
 157 static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
 158 {
 159     int i, best;
 160     int32_t lt, rt;
 161     uint64_t sum[4];
 162     uint64_t score[4];
 163
 164     /* calculate sum of 2nd order residual for each channel */
 165     sum[0] = sum[1] = sum[2] = sum[3] = 0;
 166     for(i=2; i<n; i++) {
 167         lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2];
 168         rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2];
 169         sum[2] += FFABS((lt + rt) >> 1);
 170         sum[3] += FFABS(lt - rt);
 171         sum[0] += FFABS(lt);
 172         sum[1] += FFABS(rt);
 173     }
 174
 175     /* calculate score for each mode */
 176     score[0] = sum[0] + sum[1];
 177     score[1] = sum[0] + sum[3];
 178     score[2] = sum[1] + sum[3];
 179     score[3] = sum[2] + sum[3];
 180
 181     /* return mode with lowest score */
 182     best = 0;
 183     for(i=1; i<4; i++) {
 184         if(score[i] < score[best]) {
 185             best = i;
 186         }
 187     }
 188     return best;
 189 }
 190
 191 static void alac_stereo_decorrelation(AlacEncodeContext *s)
 192 {
 193     int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
 194     int i, mode, n = s->avctx->frame_size;
 195     int32_t tmp;
 196
 197     mode = estimate_stereo_mode(left, right, n);
 198
 199     switch(mode)
 200     {
 201         case ALAC_CHMODE_LEFT_RIGHT:
 202             s->interlacing_leftweight = 0;
 203             s->interlacing_shift = 0;
 204             break;
 205
 206         case ALAC_CHMODE_LEFT_SIDE:
 207             for(i=0; i<n; i++) {
 208                 right[i] = left[i] - right[i];
 209             }
 210             s->interlacing_leftweight = 1;
 211             s->interlacing_shift = 0;
 212             break;
 213
 214         case ALAC_CHMODE_RIGHT_SIDE:
 215             for(i=0; i<n; i++) {
 216                 tmp = right[i];
 217                 right[i] = left[i] - right[i];
 218                 left[i] = tmp + (right[i] >> 31);
 219             }
 220             s->interlacing_leftweight = 1;
 221             s->interlacing_shift = 31;
 222             break;
 223
 224         default:
 225             for(i=0; i<n; i++) {
 226                 tmp = left[i];
 227                 left[i] = (tmp + right[i]) >> 1;
 228                 right[i] = tmp - right[i];
 229             }
 230             s->interlacing_leftweight = 1;
 231             s->interlacing_shift = 1;
 232             break;
 233     }
 234 }
 235
 236 static void alac_linear_predictor(AlacEncodeContext *s, int ch)
 237 {
 238     int i;
 239     LPCContext lpc = s->lpc[ch];
 240
 241     if(lpc.lpc_order == 31) {
 242         s->predictor_buf[0] = s->sample_buf[ch][0];
 243
 244         for(i=1; i<s->avctx->frame_size; i++)
 245             s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1];
 246
 247         return;
 248     }
 249
 250     // generalised linear predictor
 251
 252     if(lpc.lpc_order > 0) {
 253         int32_t *samples  = s->sample_buf[ch];
 254         int32_t *residual = s->predictor_buf;
 255
 256         // generate warm-up samples
 257         residual[0] = samples[0];
 258         for(i=1;i<=lpc.lpc_order;i++)
 259             residual[i] = samples[i] - samples[i-1];
 260
 261         // perform lpc on remaining samples
 262         for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {
 263             int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
 264
 265             for (j = 0; j < lpc.lpc_order; j++) {
 266                 sum += (samples[lpc.lpc_order-j] - samples[0]) *
 267                         lpc.lpc_coeff[j];
 268             }
 269
 270             sum >>= lpc.lpc_quant;
 271             sum += samples[0];
 272             residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,
 273                                       s->write_sample_size);
 274             res_val = residual[i];
 275
 276             if(res_val) {
 277                 int index = lpc.lpc_order - 1;
 278                 int neg = (res_val < 0);
 279
 280                 while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) {
 281                     int val = samples[0] - samples[lpc.lpc_order - index];
 282                     int sign = (val ? FFSIGN(val) : 0);
 283
 284                     if(neg)
 285                         sign*=-1;
 286
 287                     lpc.lpc_coeff[index] -= sign;
 288                     val *= sign;
 289                     res_val -= ((val >> lpc.lpc_quant) *
 290                             (lpc.lpc_order - index));
 291                     index--;
 292                 }
 293             }
 294             samples++;
 295         }
 296     }
 297 }
 298
 299 static void alac_entropy_coder(AlacEncodeContext *s)
 300 {
 301     unsigned int history = s->rc.initial_history;
 302     int sign_modifier = 0, i, k;
 303     int32_t *samples = s->predictor_buf;
 304
 305     for(i=0;i < s->avctx->frame_size;) {
 306         int x;
 307
 308         k = av_log2((history >> 9) + 3);
 309
 310         x = -2*(*samples)-1;
 311         x ^= (x>>31);
 312
 313         samples++;
 314         i++;
 315
 316         encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
 317
 318         history += x * s->rc.history_mult
 319                    - ((history * s->rc.history_mult) >> 9);
 320
 321         sign_modifier = 0;
 322         if(x > 0xFFFF)
 323             history = 0xFFFF;
 324
 325         if((history < 128) && (i < s->avctx->frame_size)) {
 326             unsigned int block_size = 0;
 327
 328             k = 7 - av_log2(history) + ((history + 16) >> 6);
 329
 330             while((*samples == 0) && (i < s->avctx->frame_size)) {
 331                 samples++;
 332                 i++;
 333                 block_size++;
 334             }
 335             encode_scalar(s, block_size, k, 16);
 336
 337             sign_modifier = (block_size <= 0xFFFF);
 338
 339             history = 0;
 340         }
 341
 342     }
 343 }
 344
 345 static void write_compressed_frame(AlacEncodeContext *s)
 346 {
 347     int i, j;
 348
 349     if(s->avctx->channels == 2)
 350         alac_stereo_decorrelation(s);
 351     put_bits(&s->pbctx, 8, s->interlacing_shift);
 352     put_bits(&s->pbctx, 8, s->interlacing_leftweight);
 353
 354     for(i=0;i<s->avctx->channels;i++) {
 355
 356         calc_predictor_params(s, i);
 357
 358         put_bits(&s->pbctx, 4, 0);  // prediction type : currently only type 0 has been RE'd
 359         put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);
 360
 361         put_bits(&s->pbctx, 3, s->rc.rice_modifier);
 362         put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
 363         // predictor coeff. table
 364         for(j=0;j<s->lpc[i].lpc_order;j++) {
 365             put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
 366         }
 367     }
 368
 369     // apply lpc and entropy coding to audio samples
 370
 371     for(i=0;i<s->avctx->channels;i++) {
 372         alac_linear_predictor(s, i);
 373         alac_entropy_coder(s);
 374     }
 375 }
 376
 377 static av_cold int alac_encode_init(AVCodecContext *avctx)
 378 {
 379     AlacEncodeContext *s    = avctx->priv_data;
 380     uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);
 381
 382     avctx->frame_size      = DEFAULT_FRAME_SIZE;
 383     avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE;
 384
 385     if(avctx->sample_fmt != SAMPLE_FMT_S16) {
 386         av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
 387         return -1;
 388     }
 389
 390     // Set default compression level
 391     if(avctx->compression_level == FF_COMPRESSION_DEFAULT)
 392         s->compression_level = 2;
 393     else
 394         s->compression_level = av_clip(avctx->compression_level, 0, 2);
 395
 396     // Initialize default Rice parameters
 397     s->rc.history_mult    = 40;
 398     s->rc.initial_history = 10;
 399     s->rc.k_modifier      = 14;
 400     s->rc.rice_modifier   = 4;
 401
 402     s->max_coded_frame_size = 8 + (avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample>>3);
 403
 404     s->write_sample_size  = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes
 405
 406     AV_WB32(alac_extradata,    ALAC_EXTRADATA_SIZE);
 407     AV_WB32(alac_extradata+4,  MKBETAG('a','l','a','c'));
 408     AV_WB32(alac_extradata+12, avctx->frame_size);
 409     AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample);
 410     AV_WB8 (alac_extradata+21, avctx->channels);
 411     AV_WB32(alac_extradata+24, s->max_coded_frame_size);
 412     AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate
 413     AV_WB32(alac_extradata+32, avctx->sample_rate);
 414
 415     // Set relevant extradata fields
 416     if(s->compression_level > 0) {
 417         AV_WB8(alac_extradata+18, s->rc.history_mult);
 418         AV_WB8(alac_extradata+19, s->rc.initial_history);
 419         AV_WB8(alac_extradata+20, s->rc.k_modifier);
 420     }
 421
 422     s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;
 423     if(avctx->min_prediction_order >= 0) {
 424         if(avctx->min_prediction_order < MIN_LPC_ORDER ||
 425            avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {
 426             av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order);
 427                 return -1;
 428         }
 429
 430         s->min_prediction_order = avctx->min_prediction_order;
 431     }
 432
 433     s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;
 434     if(avctx->max_prediction_order >= 0) {
 435         if(avctx->max_prediction_order < MIN_LPC_ORDER ||
 436            avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {
 437             av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order);
 438                 return -1;
 439         }
 440
 441         s->max_prediction_order = avctx->max_prediction_order;
 442     }
 443
 444     if(s->max_prediction_order < s->min_prediction_order) {
 445         av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n",
 446                s->min_prediction_order, s->max_prediction_order);
 447         return -1;
 448     }
 449
 450     avctx->extradata = alac_extradata;
 451     avctx->extradata_size = ALAC_EXTRADATA_SIZE;
 452
 453     avctx->coded_frame = avcodec_alloc_frame();
 454     avctx->coded_frame->key_frame = 1;
 455
 456     s->avctx = avctx;
 457     dsputil_init(&s->dspctx, avctx);
 458
 459     return 0;
 460 }
 461
 462 static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
 463                              int buf_size, void *data)
 464 {
 465     AlacEncodeContext *s = avctx->priv_data;
 466     PutBitContext *pb = &s->pbctx;
 467     int i, out_bytes, verbatim_flag = 0;
 468
 469     if(avctx->frame_size > DEFAULT_FRAME_SIZE) {
 470         av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n");
 471         return -1;
 472     }
 473
 474     if(buf_size < 2*s->max_coded_frame_size) {
 475         av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n");
 476         return -1;
 477     }
 478
 479 verbatim:
 480     init_put_bits(pb, frame, buf_size);
 481
 482     if((s->compression_level == 0) || verbatim_flag) {
 483         // Verbatim mode
 484         int16_t *samples = data;
 485         write_frame_header(s, 1);
 486         for(i=0; i<avctx->frame_size*avctx->channels; i++) {
 487             put_sbits(pb, 16, *samples++);
 488         }
 489     } else {
 490         init_sample_buffers(s, data);
 491         write_frame_header(s, 0);
 492         write_compressed_frame(s);
 493     }
 494
 495     put_bits(pb, 3, 7);
 496     flush_put_bits(pb);
 497     out_bytes = put_bits_count(pb) >> 3;
 498
 499     if(out_bytes > s->max_coded_frame_size) {
 500         /* frame too large. use verbatim mode */
 501         if(verbatim_flag || (s->compression_level == 0)) {
 502             /* still too large. must be an error. */
 503             av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
 504             return -1;
 505         }
 506         verbatim_flag = 1;
 507         goto verbatim;
 508     }
 509
 510     return out_bytes;
 511 }
 512
 513 static av_cold int alac_encode_close(AVCodecContext *avctx)
 514 {
 515     av_freep(&avctx->extradata);
 516     avctx->extradata_size = 0;
 517     av_freep(&avctx->coded_frame);
 518     return 0;
 519 }
 520
 521 AVCodec alac_encoder = {
 522     "alac",
 523     AVMEDIA_TYPE_AUDIO,
 524     CODEC_ID_ALAC,
 525     sizeof(AlacEncodeContext),
 526     alac_encode_init,
 527     alac_encode_frame,
 528     alac_encode_close,
 529     .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
 530     .sample_fmts = (const enum SampleFormat[]){ SAMPLE_FMT_S16, SAMPLE_FMT_NONE},
 531     .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
 532 };