git.sesse.net Git - ffmpeg/blob - libavcodec/g729dec.c

   1 /*
   2  * G.729, G729 Annex D decoders
   3  * Copyright (c) 2008 Vladimir Voroshilov
   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 <inttypes.h>
  23 #include <string.h>
  24
  25 #include "avcodec.h"
  26 #include "libavutil/avutil.h"
  27 #include "get_bits.h"
  28 #include "audiodsp.h"
  29 #include "internal.h"
  30
  31
  32 #include "g729.h"
  33 #include "lsp.h"
  34 #include "celp_math.h"
  35 #include "celp_filters.h"
  36 #include "acelp_filters.h"
  37 #include "acelp_pitch_delay.h"
  38 #include "acelp_vectors.h"
  39 #include "g729data.h"
  40 #include "g729postfilter.h"
  41
  42 /**
  43  * minimum quantized LSF value (3.2.4)
  44  * 0.005 in Q13
  45  */
  46 #define LSFQ_MIN                   40
  47
  48 /**
  49  * maximum quantized LSF value (3.2.4)
  50  * 3.135 in Q13
  51  */
  52 #define LSFQ_MAX                   25681
  53
  54 /**
  55  * minimum LSF distance (3.2.4)
  56  * 0.0391 in Q13
  57  */
  58 #define LSFQ_DIFF_MIN              321
  59
  60 /// interpolation filter length
  61 #define INTERPOL_LEN              11
  62
  63 /**
  64  * minimum gain pitch value (3.8, Equation 47)
  65  * 0.2 in (1.14)
  66  */
  67 #define SHARP_MIN                  3277
  68
  69 /**
  70  * maximum gain pitch value (3.8, Equation 47)
  71  * (EE) This does not comply with the specification.
  72  * Specification says about 0.8, which should be
  73  * 13107 in (1.14), but reference C code uses
  74  * 13017 (equals to 0.7945) instead of it.
  75  */
  76 #define SHARP_MAX                  13017
  77
  78 /**
  79  * MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26  * subframe_size) in (7.13)
  80  */
  81 #define MR_ENERGY 1018156
  82
  83 #define DECISION_NOISE        0
  84 #define DECISION_INTERMEDIATE 1
  85 #define DECISION_VOICE        2
  86
  87 typedef enum {
  88     FORMAT_G729_8K = 0,
  89     FORMAT_G729D_6K4,
  90     FORMAT_COUNT,
  91 } G729Formats;
  92
  93 typedef struct {
  94     uint8_t ac_index_bits[2];   ///< adaptive codebook index for second subframe (size in bits)
  95     uint8_t parity_bit;         ///< parity bit for pitch delay
  96     uint8_t gc_1st_index_bits;  ///< gain codebook (first stage) index (size in bits)
  97     uint8_t gc_2nd_index_bits;  ///< gain codebook (second stage) index (size in bits)
  98     uint8_t fc_signs_bits;      ///< number of pulses in fixed-codebook vector
  99     uint8_t fc_indexes_bits;    ///< size (in bits) of fixed-codebook index entry
 100     uint8_t block_size;
 101 } G729FormatDescription;
 102
 103 typedef struct {
 104     /// past excitation signal buffer
 105     int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];
 106
 107     int16_t* exc;               ///< start of past excitation data in buffer
 108     int pitch_delay_int_prev;   ///< integer part of previous subframe's pitch delay (4.1.3)
 109
 110     /// (2.13) LSP quantizer outputs
 111     int16_t  past_quantizer_output_buf[MA_NP + 1][10];
 112     int16_t* past_quantizer_outputs[MA_NP + 1];
 113
 114     int16_t lsfq[10];           ///< (2.13) quantized LSF coefficients from previous frame
 115     int16_t lsp_buf[2][10];     ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
 116     int16_t *lsp[2];            ///< pointers to lsp_buf
 117
 118     int16_t quant_energy[4];    ///< (5.10) past quantized energy
 119
 120     /// previous speech data for LP synthesis filter
 121     int16_t syn_filter_data[10];
 122
 123
 124     /// residual signal buffer (used in long-term postfilter)
 125     int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
 126
 127     /// previous speech data for residual calculation filter
 128     int16_t res_filter_data[SUBFRAME_SIZE+10];
 129
 130     /// previous speech data for short-term postfilter
 131     int16_t pos_filter_data[SUBFRAME_SIZE+10];
 132
 133     /// (1.14) pitch gain of current and five previous subframes
 134     int16_t past_gain_pitch[6];
 135
 136     /// (14.1) gain code from current and previous subframe
 137     int16_t past_gain_code[2];
 138
 139     /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
 140     int16_t voice_decision;
 141
 142     int16_t onset;              ///< detected onset level (0-2)
 143     int16_t was_periodic;       ///< whether previous frame was declared as periodic or not (4.4)
 144     int16_t ht_prev_data;       ///< previous data for 4.2.3, equation 86
 145     int gain_coeff;             ///< (1.14) gain coefficient (4.2.4)
 146     uint16_t rand_value;        ///< random number generator value (4.4.4)
 147     int ma_predictor_prev;      ///< switched MA predictor of LSP quantizer from last good frame
 148
 149     /// (14.14) high-pass filter data (past input)
 150     int hpf_f[2];
 151
 152     /// high-pass filter data (past output)
 153     int16_t hpf_z[2];
 154 }  G729ChannelContext;
 155
 156 typedef struct {
 157     AudioDSPContext adsp;
 158
 159     G729ChannelContext *channel_context;
 160 } G729Context;
 161
 162 static const G729FormatDescription format_g729_8k = {
 163     .ac_index_bits     = {8,5},
 164     .parity_bit        = 1,
 165     .gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
 166     .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
 167     .fc_signs_bits     = 4,
 168     .fc_indexes_bits   = 13,
 169     .block_size        = G729_8K_BLOCK_SIZE,
 170 };
 171
 172 static const G729FormatDescription format_g729d_6k4 = {
 173     .ac_index_bits     = {8,4},
 174     .parity_bit        = 0,
 175     .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
 176     .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
 177     .fc_signs_bits     = 2,
 178     .fc_indexes_bits   = 9,
 179     .block_size        = G729D_6K4_BLOCK_SIZE,
 180 };
 181
 182 /**
 183  * @brief pseudo random number generator
 184  */
 185 static inline uint16_t g729_prng(uint16_t value)
 186 {
 187     return 31821 * value + 13849;
 188 }
 189
 190 /**
 191  * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
 192  * @param[out] lsfq (2.13) quantized LSF coefficients
 193  * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
 194  * @param ma_predictor switched MA predictor of LSP quantizer
 195  * @param vq_1st first stage vector of quantizer
 196  * @param vq_2nd_low second stage lower vector of LSP quantizer
 197  * @param vq_2nd_high second stage higher vector of LSP quantizer
 198  */
 199 static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
 200                        int16_t ma_predictor,
 201                        int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
 202 {
 203     int i,j;
 204     static const uint8_t min_distance[2]={10, 5}; //(2.13)
 205     int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
 206
 207     for (i = 0; i < 5; i++) {
 208         quantizer_output[i]     = cb_lsp_1st[vq_1st][i    ] + cb_lsp_2nd[vq_2nd_low ][i    ];
 209         quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
 210     }
 211
 212     for (j = 0; j < 2; j++) {
 213         for (i = 1; i < 10; i++) {
 214             int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
 215             if (diff > 0) {
 216                 quantizer_output[i - 1] -= diff;
 217                 quantizer_output[i    ] += diff;
 218             }
 219         }
 220     }
 221
 222     for (i = 0; i < 10; i++) {
 223         int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
 224         for (j = 0; j < MA_NP; j++)
 225             sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
 226
 227         lsfq[i] = sum >> 15;
 228     }
 229
 230     ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
 231 }
 232
 233 /**
 234  * Restores past LSP quantizer output using LSF from previous frame
 235  * @param[in,out] lsfq (2.13) quantized LSF coefficients
 236  * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
 237  * @param ma_predictor_prev MA predictor from previous frame
 238  * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
 239  */
 240 static void lsf_restore_from_previous(int16_t* lsfq,
 241                                       int16_t* past_quantizer_outputs[MA_NP + 1],
 242                                       int ma_predictor_prev)
 243 {
 244     int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
 245     int i,k;
 246
 247     for (i = 0; i < 10; i++) {
 248         int tmp = lsfq[i] << 15;
 249
 250         for (k = 0; k < MA_NP; k++)
 251             tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
 252
 253         quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
 254     }
 255 }
 256
 257 /**
 258  * Constructs new excitation signal and applies phase filter to it
 259  * @param[out] out constructed speech signal
 260  * @param in original excitation signal
 261  * @param fc_cur (2.13) original fixed-codebook vector
 262  * @param gain_code (14.1) gain code
 263  * @param subframe_size length of the subframe
 264  */
 265 static void g729d_get_new_exc(
 266         int16_t* out,
 267         const int16_t* in,
 268         const int16_t* fc_cur,
 269         int dstate,
 270         int gain_code,
 271         int subframe_size)
 272 {
 273     int i;
 274     int16_t fc_new[SUBFRAME_SIZE];
 275
 276     ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
 277
 278     for (i = 0; i < subframe_size; i++) {
 279         out[i]  = in[i];
 280         out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
 281         out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
 282     }
 283 }
 284
 285 /**
 286  * Makes decision about onset in current subframe
 287  * @param past_onset decision result of previous subframe
 288  * @param past_gain_code gain code of current and previous subframe
 289  *
 290  * @return onset decision result for current subframe
 291  */
 292 static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
 293 {
 294     if ((past_gain_code[0] >> 1) > past_gain_code[1])
 295         return 2;
 296
 297     return FFMAX(past_onset-1, 0);
 298 }
 299
 300 /**
 301  * Makes decision about voice presence in current subframe
 302  * @param onset onset level
 303  * @param prev_voice_decision voice decision result from previous subframe
 304  * @param past_gain_pitch pitch gain of current and previous subframes
 305  *
 306  * @return voice decision result for current subframe
 307  */
 308 static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
 309 {
 310     int i, low_gain_pitch_cnt, voice_decision;
 311
 312     if (past_gain_pitch[0] >= 14745) {       // 0.9
 313         voice_decision = DECISION_VOICE;
 314     } else if (past_gain_pitch[0] <= 9830) { // 0.6
 315         voice_decision = DECISION_NOISE;
 316     } else {
 317         voice_decision = DECISION_INTERMEDIATE;
 318     }
 319
 320     for (i = 0, low_gain_pitch_cnt = 0; i < 6; i++)
 321         if (past_gain_pitch[i] < 9830)
 322             low_gain_pitch_cnt++;
 323
 324     if (low_gain_pitch_cnt > 2 && !onset)
 325         voice_decision = DECISION_NOISE;
 326
 327     if (!onset && voice_decision > prev_voice_decision + 1)
 328         voice_decision--;
 329
 330     if (onset && voice_decision < DECISION_VOICE)
 331         voice_decision++;
 332
 333     return voice_decision;
 334 }
 335
 336 static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)
 337 {
 338     int64_t res = 0;
 339
 340     while (order--)
 341         res += *v1++ * *v2++;
 342
 343     if      (res > INT32_MAX) return INT32_MAX;
 344     else if (res < INT32_MIN) return INT32_MIN;
 345
 346     return res;
 347 }
 348
 349 static av_cold int decoder_init(AVCodecContext * avctx)
 350 {
 351     G729Context *s = avctx->priv_data;
 352     G729ChannelContext *ctx;
 353     int c,i,k;
 354
 355     if (avctx->channels < 1 || avctx->channels > 2) {
 356         av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n", avctx->channels);
 357         return AVERROR(EINVAL);
 358     }
 359     avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
 360
 361     /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
 362     avctx->frame_size = SUBFRAME_SIZE << 1;
 363
 364     ctx =
 365     s->channel_context = av_mallocz(sizeof(G729ChannelContext) * avctx->channels);
 366     if (!ctx)
 367         return AVERROR(ENOMEM);
 368
 369     for (c = 0; c < avctx->channels; c++) {
 370         ctx->gain_coeff = 16384; // 1.0 in (1.14)
 371
 372         for (k = 0; k < MA_NP + 1; k++) {
 373             ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
 374             for (i = 1; i < 11; i++)
 375                 ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
 376         }
 377
 378         ctx->lsp[0] = ctx->lsp_buf[0];
 379         ctx->lsp[1] = ctx->lsp_buf[1];
 380         memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
 381
 382         ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
 383
 384         ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
 385
 386         /* random seed initialization */
 387         ctx->rand_value = 21845;
 388
 389         /* quantized prediction error */
 390         for (i = 0; i < 4; i++)
 391             ctx->quant_energy[i] = -14336; // -14 in (5.10)
 392
 393         ctx++;
 394     }
 395
 396     ff_audiodsp_init(&s->adsp);
 397     s->adsp.scalarproduct_int16 = scalarproduct_int16_c;
 398
 399     return 0;
 400 }
 401
 402 static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
 403                         AVPacket *avpkt)
 404 {
 405     const uint8_t *buf = avpkt->data;
 406     int buf_size       = avpkt->size;
 407     int16_t *out_frame;
 408     GetBitContext gb;
 409     const G729FormatDescription *format;
 410     int c, i;
 411     int16_t *tmp;
 412     G729Formats packet_type;
 413     G729Context *s = avctx->priv_data;
 414     G729ChannelContext *ctx = s->channel_context;
 415     int16_t lp[2][11];           // (3.12)
 416     uint8_t ma_predictor;     ///< switched MA predictor of LSP quantizer
 417     uint8_t quantizer_1st;    ///< first stage vector of quantizer
 418     uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
 419     uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
 420
 421     int pitch_delay_int[2];      // pitch delay, integer part
 422     int pitch_delay_3x;          // pitch delay, multiplied by 3
 423     int16_t fc[SUBFRAME_SIZE];   // fixed-codebook vector
 424     int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
 425     int j, ret;
 426     int gain_before, gain_after;
 427     AVFrame *frame = data;
 428
 429     frame->nb_samples = SUBFRAME_SIZE<<1;
 430     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
 431         return ret;
 432
 433     if (buf_size && buf_size % ((G729_8K_BLOCK_SIZE + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * avctx->channels) == 0) {
 434         packet_type = FORMAT_G729_8K;
 435         format = &format_g729_8k;
 436         //Reset voice decision
 437         ctx->onset = 0;
 438         ctx->voice_decision = DECISION_VOICE;
 439         av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
 440     } else if (buf_size == G729D_6K4_BLOCK_SIZE * avctx->channels && avctx->codec_id != AV_CODEC_ID_ACELP_KELVIN) {
 441         packet_type = FORMAT_G729D_6K4;
 442         format = &format_g729d_6k4;
 443         av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
 444     } else {
 445         av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
 446         return AVERROR_INVALIDDATA;
 447     }
 448
 449     for (c = 0; c < avctx->channels; c++) {
 450         int frame_erasure = 0; ///< frame erasure detected during decoding
 451         int bad_pitch = 0;     ///< parity check failed
 452         int is_periodic = 0;   ///< whether one of the subframes is declared as periodic or not
 453         out_frame = (int16_t*)frame->data[c];
 454         if (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN) {
 455             if (*buf != ((avctx->channels - 1 - c) * 0x80 | 2))
 456                 avpriv_request_sample(avctx, "First byte value %x for channel %d", *buf, c);
 457             buf++;
 458         }
 459
 460         for (i = 0; i < format->block_size; i++)
 461             frame_erasure |= buf[i];
 462         frame_erasure = !frame_erasure;
 463
 464         init_get_bits8(&gb, buf, format->block_size);
 465
 466         ma_predictor     = get_bits(&gb, 1);
 467         quantizer_1st    = get_bits(&gb, VQ_1ST_BITS);
 468         quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
 469         quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
 470
 471         if (frame_erasure) {
 472             lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
 473                                       ctx->ma_predictor_prev);
 474         } else {
 475             lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
 476                        ma_predictor,
 477                        quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
 478             ctx->ma_predictor_prev = ma_predictor;
 479         }
 480
 481         tmp = ctx->past_quantizer_outputs[MA_NP];
 482         memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
 483                 MA_NP * sizeof(int16_t*));
 484         ctx->past_quantizer_outputs[0] = tmp;
 485
 486         ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
 487
 488         ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
 489
 490         FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
 491
 492         for (i = 0; i < 2; i++) {
 493             int gain_corr_factor;
 494
 495             uint8_t ac_index;      ///< adaptive codebook index
 496             uint8_t pulses_signs;  ///< fixed-codebook vector pulse signs
 497             int fc_indexes;        ///< fixed-codebook indexes
 498             uint8_t gc_1st_index;  ///< gain codebook (first stage) index
 499             uint8_t gc_2nd_index;  ///< gain codebook (second stage) index
 500
 501             ac_index      = get_bits(&gb, format->ac_index_bits[i]);
 502             if (!i && format->parity_bit)
 503                 bad_pitch = av_parity(ac_index >> 2) == get_bits1(&gb);
 504             fc_indexes    = get_bits(&gb, format->fc_indexes_bits);
 505             pulses_signs  = get_bits(&gb, format->fc_signs_bits);
 506             gc_1st_index  = get_bits(&gb, format->gc_1st_index_bits);
 507             gc_2nd_index  = get_bits(&gb, format->gc_2nd_index_bits);
 508
 509             if (frame_erasure) {
 510                 pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
 511             } else if (!i) {
 512                 if (bad_pitch) {
 513                     pitch_delay_3x = 3 * ctx->pitch_delay_int_prev;
 514                 } else {
 515                     pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
 516                 }
 517             } else {
 518                 int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
 519                                               PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
 520
 521                 if (packet_type == FORMAT_G729D_6K4) {
 522                     pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
 523                 } else {
 524                     pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
 525                 }
 526             }
 527
 528             /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
 529             pitch_delay_int[i]  = (pitch_delay_3x + 1) / 3;
 530             if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
 531                 av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
 532                 pitch_delay_int[i] = PITCH_DELAY_MAX;
 533             }
 534
 535             if (frame_erasure) {
 536                 ctx->rand_value = g729_prng(ctx->rand_value);
 537                 fc_indexes   = av_mod_uintp2(ctx->rand_value, format->fc_indexes_bits);
 538
 539                 ctx->rand_value = g729_prng(ctx->rand_value);
 540                 pulses_signs = ctx->rand_value;
 541             }
 542
 543
 544             memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
 545             switch (packet_type) {
 546                 case FORMAT_G729_8K:
 547                     ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
 548                                                 ff_fc_4pulses_8bits_track_4,
 549                                                 fc_indexes, pulses_signs, 3, 3);
 550                     break;
 551                 case FORMAT_G729D_6K4:
 552                     ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
 553                                                 ff_fc_2pulses_9bits_track2_gray,
 554                                                 fc_indexes, pulses_signs, 1, 4);
 555                     break;
 556             }
 557
 558             /*
 559               This filter enhances harmonic components of the fixed-codebook vector to
 560               improve the quality of the reconstructed speech.
 561
 562                          / fc_v[i],                                    i < pitch_delay
 563               fc_v[i] = <
 564                          \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
 565             */
 566             if (SUBFRAME_SIZE > pitch_delay_int[i])
 567                 ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
 568                                              fc + pitch_delay_int[i],
 569                                              fc, 1 << 14,
 570                                              av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
 571                                              0, 14,
 572                                              SUBFRAME_SIZE - pitch_delay_int[i]);
 573
 574             memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
 575             ctx->past_gain_code[1] = ctx->past_gain_code[0];
 576
 577             if (frame_erasure) {
 578                 ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
 579                 ctx->past_gain_code[0]  = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
 580
 581                 gain_corr_factor = 0;
 582             } else {
 583                 if (packet_type == FORMAT_G729D_6K4) {
 584                     ctx->past_gain_pitch[0]  = cb_gain_1st_6k4[gc_1st_index][0] +
 585                                                cb_gain_2nd_6k4[gc_2nd_index][0];
 586                     gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
 587                                        cb_gain_2nd_6k4[gc_2nd_index][1];
 588
 589                     /* Without check below overflow can occur in ff_acelp_update_past_gain.
 590                        It is not issue for G.729, because gain_corr_factor in it's case is always
 591                        greater than 1024, while in G.729D it can be even zero. */
 592                     gain_corr_factor = FFMAX(gain_corr_factor, 1024);
 593     #ifndef G729_BITEXACT
 594                     gain_corr_factor >>= 1;
 595     #endif
 596                 } else {
 597                     ctx->past_gain_pitch[0]  = cb_gain_1st_8k[gc_1st_index][0] +
 598                                                cb_gain_2nd_8k[gc_2nd_index][0];
 599                     gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
 600                                        cb_gain_2nd_8k[gc_2nd_index][1];
 601                 }
 602
 603                 /* Decode the fixed-codebook gain. */
 604                 ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&s->adsp, gain_corr_factor,
 605                                                                    fc, MR_ENERGY,
 606                                                                    ctx->quant_energy,
 607                                                                    ma_prediction_coeff,
 608                                                                    SUBFRAME_SIZE, 4);
 609     #ifdef G729_BITEXACT
 610                 /*
 611                   This correction required to get bit-exact result with
 612                   reference code, because gain_corr_factor in G.729D is
 613                   two times larger than in original G.729.
 614
 615                   If bit-exact result is not issue then gain_corr_factor
 616                   can be simpler divided by 2 before call to g729_get_gain_code
 617                   instead of using correction below.
 618                 */
 619                 if (packet_type == FORMAT_G729D_6K4) {
 620                     gain_corr_factor >>= 1;
 621                     ctx->past_gain_code[0] >>= 1;
 622                 }
 623     #endif
 624             }
 625             ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
 626
 627             /* Routine requires rounding to lowest. */
 628             ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
 629                                  ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
 630                                  ff_acelp_interp_filter, 6,
 631                                  (pitch_delay_3x % 3) << 1,
 632                                  10, SUBFRAME_SIZE);
 633
 634             ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
 635                                          ctx->exc + i * SUBFRAME_SIZE, fc,
 636                                          (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
 637                                          ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
 638                                          1 << 13, 14, SUBFRAME_SIZE);
 639
 640             memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
 641
 642             if (ff_celp_lp_synthesis_filter(
 643                 synth+10,
 644                 &lp[i][1],
 645                 ctx->exc  + i * SUBFRAME_SIZE,
 646                 SUBFRAME_SIZE,
 647                 10,
 648                 1,
 649                 0,
 650                 0x800))
 651                 /* Overflow occurred, downscale excitation signal... */
 652                 for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
 653                     ctx->exc_base[j] >>= 2;
 654
 655             /* ... and make synthesis again. */
 656             if (packet_type == FORMAT_G729D_6K4) {
 657                 int16_t exc_new[SUBFRAME_SIZE];
 658
 659                 ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
 660                 ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
 661
 662                 g729d_get_new_exc(exc_new, ctx->exc  + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
 663
 664                 ff_celp_lp_synthesis_filter(
 665                         synth+10,
 666                         &lp[i][1],
 667                         exc_new,
 668                         SUBFRAME_SIZE,
 669                         10,
 670                         0,
 671                         0,
 672                         0x800);
 673             } else {
 674                 ff_celp_lp_synthesis_filter(
 675                         synth+10,
 676                         &lp[i][1],
 677                         ctx->exc  + i * SUBFRAME_SIZE,
 678                         SUBFRAME_SIZE,
 679                         10,
 680                         0,
 681                         0,
 682                         0x800);
 683             }
 684             /* Save data (without postfilter) for use in next subframe. */
 685             memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
 686
 687             /* Calculate gain of unfiltered signal for use in AGC. */
 688             gain_before = 0;
 689             for (j = 0; j < SUBFRAME_SIZE; j++)
 690                 gain_before += FFABS(synth[j+10]);
 691
 692             /* Call postfilter and also update voicing decision for use in next frame. */
 693             ff_g729_postfilter(
 694                     &s->adsp,
 695                     &ctx->ht_prev_data,
 696                     &is_periodic,
 697                     &lp[i][0],
 698                     pitch_delay_int[0],
 699                     ctx->residual,
 700                     ctx->res_filter_data,
 701                     ctx->pos_filter_data,
 702                     synth+10,
 703                     SUBFRAME_SIZE);
 704
 705             /* Calculate gain of filtered signal for use in AGC. */
 706             gain_after = 0;
 707             for (j = 0; j < SUBFRAME_SIZE; j++)
 708                 gain_after += FFABS(synth[j+10]);
 709
 710             ctx->gain_coeff = ff_g729_adaptive_gain_control(
 711                     gain_before,
 712                     gain_after,
 713                     synth+10,
 714                     SUBFRAME_SIZE,
 715                     ctx->gain_coeff);
 716
 717             if (frame_erasure) {
 718                 ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
 719             } else {
 720                 ctx->pitch_delay_int_prev = pitch_delay_int[i];
 721             }
 722
 723             memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
 724             ff_acelp_high_pass_filter(
 725                     out_frame + i*SUBFRAME_SIZE,
 726                     ctx->hpf_f,
 727                     synth+10,
 728                     SUBFRAME_SIZE);
 729             memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
 730         }
 731
 732         ctx->was_periodic = is_periodic;
 733
 734         /* Save signal for use in next frame. */
 735         memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
 736
 737         buf += format->block_size;
 738         ctx++;
 739     }
 740
 741     *got_frame_ptr = 1;
 742     return (format->block_size + (avctx->codec_id == AV_CODEC_ID_ACELP_KELVIN)) * avctx->channels;
 743 }
 744
 745 static av_cold int decode_close(AVCodecContext *avctx)
 746 {
 747     G729Context *s = avctx->priv_data;
 748     av_freep(&s->channel_context);
 749
 750     return 0;
 751 }
 752
 753 AVCodec ff_g729_decoder = {
 754     .name           = "g729",
 755     .long_name      = NULL_IF_CONFIG_SMALL("G.729"),
 756     .type           = AVMEDIA_TYPE_AUDIO,
 757     .id             = AV_CODEC_ID_G729,
 758     .priv_data_size = sizeof(G729Context),
 759     .init           = decoder_init,
 760     .decode         = decode_frame,
 761     .close          = decode_close,
 762     .capabilities   = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
 763 };
 764
 765 AVCodec ff_acelp_kelvin_decoder = {
 766     .name           = "acelp.kelvin",
 767     .long_name      = NULL_IF_CONFIG_SMALL("Sipro ACELP.KELVIN"),
 768     .type           = AVMEDIA_TYPE_AUDIO,
 769     .id             = AV_CODEC_ID_ACELP_KELVIN,
 770     .priv_data_size = sizeof(G729Context),
 771     .init           = decoder_init,
 772     .decode         = decode_frame,
 773     .close          = decode_close,
 774     .capabilities   = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
 775 };