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