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