Libav 0.7.1
libavcodec/amrwbdec.c
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00001 /*
00002  * AMR wideband decoder
00003  * Copyright (c) 2010 Marcelo Galvao Povoa
00004  *
00005  * This file is part of Libav.
00006  *
00007  * Libav is free software; you can redistribute it and/or
00008  * modify it under the terms of the GNU Lesser General Public
00009  * License as published by the Free Software Foundation; either
00010  * version 2.1 of the License, or (at your option) any later version.
00011  *
00012  * Libav is distributed in the hope that it will be useful,
00013  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00014  * MERCHANTABILITY or FITNESS FOR A particular PURPOSE.  See the GNU
00015  * Lesser General Public License for more details.
00016  *
00017  * You should have received a copy of the GNU Lesser General Public
00018  * License along with Libav; if not, write to the Free Software
00019  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00020  */
00021 
00027 #include "libavutil/lfg.h"
00028 
00029 #include "avcodec.h"
00030 #include "get_bits.h"
00031 #include "lsp.h"
00032 #include "celp_math.h"
00033 #include "celp_filters.h"
00034 #include "acelp_filters.h"
00035 #include "acelp_vectors.h"
00036 #include "acelp_pitch_delay.h"
00037 
00038 #define AMR_USE_16BIT_TABLES
00039 #include "amr.h"
00040 
00041 #include "amrwbdata.h"
00042 
00043 typedef struct {
00044     AMRWBFrame                             frame; 
00045     enum Mode                        fr_cur_mode; 
00046     uint8_t                           fr_quality; 
00047     float                      isf_cur[LP_ORDER]; 
00048     float                   isf_q_past[LP_ORDER]; 
00049     float               isf_past_final[LP_ORDER]; 
00050     double                      isp[4][LP_ORDER]; 
00051     double               isp_sub4_past[LP_ORDER]; 
00052 
00053     float                   lp_coef[4][LP_ORDER]; 
00054 
00055     uint8_t                       base_pitch_lag; 
00056     uint8_t                        pitch_lag_int; 
00057 
00058     float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; 
00059     float                            *excitation; 
00060 
00061     float           pitch_vector[AMRWB_SFR_SIZE]; 
00062     float           fixed_vector[AMRWB_SFR_SIZE]; 
00063 
00064     float                    prediction_error[4]; 
00065     float                          pitch_gain[6]; 
00066     float                          fixed_gain[2]; 
00067 
00068     float                              tilt_coef; 
00069 
00070     float                 prev_sparse_fixed_gain; 
00071     uint8_t                    prev_ir_filter_nr; 
00072     float                           prev_tr_gain; 
00073 
00074     float samples_az[LP_ORDER + AMRWB_SFR_SIZE];         
00075     float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE];     
00076     float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; 
00077 
00078     float          hpf_31_mem[2], hpf_400_mem[2]; 
00079     float                           demph_mem[1]; 
00080     float               bpf_6_7_mem[HB_FIR_SIZE]; 
00081     float                 lpf_7_mem[HB_FIR_SIZE]; 
00082 
00083     AVLFG                                   prng; 
00084     uint8_t                          first_frame; 
00085 } AMRWBContext;
00086 
00087 static av_cold int amrwb_decode_init(AVCodecContext *avctx)
00088 {
00089     AMRWBContext *ctx = avctx->priv_data;
00090     int i;
00091 
00092     avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
00093 
00094     av_lfg_init(&ctx->prng, 1);
00095 
00096     ctx->excitation  = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1];
00097     ctx->first_frame = 1;
00098 
00099     for (i = 0; i < LP_ORDER; i++)
00100         ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15));
00101 
00102     for (i = 0; i < 4; i++)
00103         ctx->prediction_error[i] = MIN_ENERGY;
00104 
00105     return 0;
00106 }
00107 
00117 static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
00118 {
00119     GetBitContext gb;
00120     init_get_bits(&gb, buf, 8);
00121 
00122     /* Decode frame header (1st octet) */
00123     skip_bits(&gb, 1);  // padding bit
00124     ctx->fr_cur_mode  = get_bits(&gb, 4);
00125     ctx->fr_quality   = get_bits1(&gb);
00126     skip_bits(&gb, 2);  // padding bits
00127 
00128     return 1;
00129 }
00130 
00138 static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
00139 {
00140     int i;
00141 
00142     for (i = 0; i < 9; i++)
00143         isf_q[i]      = dico1_isf[ind[0]][i]      * (1.0f / (1 << 15));
00144 
00145     for (i = 0; i < 7; i++)
00146         isf_q[i + 9]  = dico2_isf[ind[1]][i]      * (1.0f / (1 << 15));
00147 
00148     for (i = 0; i < 5; i++)
00149         isf_q[i]     += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15));
00150 
00151     for (i = 0; i < 4; i++)
00152         isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15));
00153 
00154     for (i = 0; i < 7; i++)
00155         isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15));
00156 }
00157 
00165 static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
00166 {
00167     int i;
00168 
00169     for (i = 0; i < 9; i++)
00170         isf_q[i]       = dico1_isf[ind[0]][i]  * (1.0f / (1 << 15));
00171 
00172     for (i = 0; i < 7; i++)
00173         isf_q[i + 9]   = dico2_isf[ind[1]][i]  * (1.0f / (1 << 15));
00174 
00175     for (i = 0; i < 3; i++)
00176         isf_q[i]      += dico21_isf[ind[2]][i] * (1.0f / (1 << 15));
00177 
00178     for (i = 0; i < 3; i++)
00179         isf_q[i + 3]  += dico22_isf[ind[3]][i] * (1.0f / (1 << 15));
00180 
00181     for (i = 0; i < 3; i++)
00182         isf_q[i + 6]  += dico23_isf[ind[4]][i] * (1.0f / (1 << 15));
00183 
00184     for (i = 0; i < 3; i++)
00185         isf_q[i + 9]  += dico24_isf[ind[5]][i] * (1.0f / (1 << 15));
00186 
00187     for (i = 0; i < 4; i++)
00188         isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15));
00189 }
00190 
00199 static void isf_add_mean_and_past(float *isf_q, float *isf_past)
00200 {
00201     int i;
00202     float tmp;
00203 
00204     for (i = 0; i < LP_ORDER; i++) {
00205         tmp = isf_q[i];
00206         isf_q[i] += isf_mean[i] * (1.0f / (1 << 15));
00207         isf_q[i] += PRED_FACTOR * isf_past[i];
00208         isf_past[i] = tmp;
00209     }
00210 }
00211 
00219 static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
00220 {
00221     int i, k;
00222 
00223     for (k = 0; k < 3; k++) {
00224         float c = isfp_inter[k];
00225         for (i = 0; i < LP_ORDER; i++)
00226             isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i];
00227     }
00228 }
00229 
00241 static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
00242                                   uint8_t *base_lag_int, int subframe)
00243 {
00244     if (subframe == 0 || subframe == 2) {
00245         if (pitch_index < 376) {
00246             *lag_int  = (pitch_index + 137) >> 2;
00247             *lag_frac = pitch_index - (*lag_int << 2) + 136;
00248         } else if (pitch_index < 440) {
00249             *lag_int  = (pitch_index + 257 - 376) >> 1;
00250             *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) << 1;
00251             /* the actual resolution is 1/2 but expressed as 1/4 */
00252         } else {
00253             *lag_int  = pitch_index - 280;
00254             *lag_frac = 0;
00255         }
00256         /* minimum lag for next subframe */
00257         *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
00258                                 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
00259         // XXX: the spec states clearly that *base_lag_int should be
00260         // the nearest integer to *lag_int (minus 8), but the ref code
00261         // actually always uses its floor, I'm following the latter
00262     } else {
00263         *lag_int  = (pitch_index + 1) >> 2;
00264         *lag_frac = pitch_index - (*lag_int << 2);
00265         *lag_int += *base_lag_int;
00266     }
00267 }
00268 
00274 static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
00275                                  uint8_t *base_lag_int, int subframe, enum Mode mode)
00276 {
00277     if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {
00278         if (pitch_index < 116) {
00279             *lag_int  = (pitch_index + 69) >> 1;
00280             *lag_frac = (pitch_index - (*lag_int << 1) + 68) << 1;
00281         } else {
00282             *lag_int  = pitch_index - 24;
00283             *lag_frac = 0;
00284         }
00285         // XXX: same problem as before
00286         *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
00287                                 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
00288     } else {
00289         *lag_int  = (pitch_index + 1) >> 1;
00290         *lag_frac = (pitch_index - (*lag_int << 1)) << 1;
00291         *lag_int += *base_lag_int;
00292     }
00293 }
00294 
00303 static void decode_pitch_vector(AMRWBContext *ctx,
00304                                 const AMRWBSubFrame *amr_subframe,
00305                                 const int subframe)
00306 {
00307     int pitch_lag_int, pitch_lag_frac;
00308     int i;
00309     float *exc     = ctx->excitation;
00310     enum Mode mode = ctx->fr_cur_mode;
00311 
00312     if (mode <= MODE_8k85) {
00313         decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
00314                               &ctx->base_pitch_lag, subframe, mode);
00315     } else
00316         decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
00317                               &ctx->base_pitch_lag, subframe);
00318 
00319     ctx->pitch_lag_int = pitch_lag_int;
00320     pitch_lag_int += pitch_lag_frac > 0;
00321 
00322     /* Calculate the pitch vector by interpolating the past excitation at the
00323        pitch lag using a hamming windowed sinc function */
00324     ff_acelp_interpolatef(exc, exc + 1 - pitch_lag_int,
00325                           ac_inter, 4,
00326                           pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
00327                           LP_ORDER, AMRWB_SFR_SIZE + 1);
00328 
00329     /* Check which pitch signal path should be used
00330      * 6k60 and 8k85 modes have the ltp flag set to 0 */
00331     if (amr_subframe->ltp) {
00332         memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float));
00333     } else {
00334         for (i = 0; i < AMRWB_SFR_SIZE; i++)
00335             ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] +
00336                                    0.18 * exc[i + 1];
00337         memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float));
00338     }
00339 }
00340 
00342 #define BIT_STR(x,lsb,len) (((x) >> (lsb)) & ((1 << (len)) - 1))
00343 
00345 #define BIT_POS(x, p) (((x) >> (p)) & 1)
00346 
00360 static inline void decode_1p_track(int *out, int code, int m, int off)
00361 {
00362     int pos = BIT_STR(code, 0, m) + off; 
00363 
00364     out[0] = BIT_POS(code, m) ? -pos : pos;
00365 }
00366 
00367 static inline void decode_2p_track(int *out, int code, int m, int off) 
00368 {
00369     int pos0 = BIT_STR(code, m, m) + off;
00370     int pos1 = BIT_STR(code, 0, m) + off;
00371 
00372     out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0;
00373     out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1;
00374     out[1] = pos0 > pos1 ? -out[1] : out[1];
00375 }
00376 
00377 static void decode_3p_track(int *out, int code, int m, int off) 
00378 {
00379     int half_2p = BIT_POS(code, 2*m - 1) << (m - 1);
00380 
00381     decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
00382                     m - 1, off + half_2p);
00383     decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off);
00384 }
00385 
00386 static void decode_4p_track(int *out, int code, int m, int off) 
00387 {
00388     int half_4p, subhalf_2p;
00389     int b_offset = 1 << (m - 1);
00390 
00391     switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */
00392     case 0: /* 0 pulses in A, 4 pulses in B or vice versa */
00393         half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses
00394         subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);
00395 
00396         decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),
00397                         m - 2, off + half_4p + subhalf_2p);
00398         decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),
00399                         m - 1, off + half_4p);
00400         break;
00401     case 1: /* 1 pulse in A, 3 pulses in B */
00402         decode_1p_track(out, BIT_STR(code,  3*m - 2, m),
00403                         m - 1, off);
00404         decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),
00405                         m - 1, off + b_offset);
00406         break;
00407     case 2: /* 2 pulses in each half */
00408         decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),
00409                         m - 1, off);
00410         decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),
00411                         m - 1, off + b_offset);
00412         break;
00413     case 3: /* 3 pulses in A, 1 pulse in B */
00414         decode_3p_track(out, BIT_STR(code, m, 3*m - 2),
00415                         m - 1, off);
00416         decode_1p_track(out + 3, BIT_STR(code, 0, m),
00417                         m - 1, off + b_offset);
00418         break;
00419     }
00420 }
00421 
00422 static void decode_5p_track(int *out, int code, int m, int off) 
00423 {
00424     int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);
00425 
00426     decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),
00427                     m - 1, off + half_3p);
00428 
00429     decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);
00430 }
00431 
00432 static void decode_6p_track(int *out, int code, int m, int off) 
00433 {
00434     int b_offset = 1 << (m - 1);
00435     /* which half has more pulses in cases 0 to 2 */
00436     int half_more  = BIT_POS(code, 6*m - 5) << (m - 1);
00437     int half_other = b_offset - half_more;
00438 
00439     switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */
00440     case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
00441         decode_1p_track(out, BIT_STR(code, 0, m),
00442                         m - 1, off + half_more);
00443         decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
00444                         m - 1, off + half_more);
00445         break;
00446     case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
00447         decode_1p_track(out, BIT_STR(code, 0, m),
00448                         m - 1, off + half_other);
00449         decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
00450                         m - 1, off + half_more);
00451         break;
00452     case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
00453         decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
00454                         m - 1, off + half_other);
00455         decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),
00456                         m - 1, off + half_more);
00457         break;
00458     case 3: /* 3 pulses in A, 3 pulses in B */
00459         decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),
00460                         m - 1, off);
00461         decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2),
00462                         m - 1, off + b_offset);
00463         break;
00464     }
00465 }
00466 
00476 static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
00477                                 const uint16_t *pulse_lo, const enum Mode mode)
00478 {
00479     /* sig_pos stores for each track the decoded pulse position indexes
00480      * (1-based) multiplied by its corresponding amplitude (+1 or -1) */
00481     int sig_pos[4][6];
00482     int spacing = (mode == MODE_6k60) ? 2 : 4;
00483     int i, j;
00484 
00485     switch (mode) {
00486     case MODE_6k60:
00487         for (i = 0; i < 2; i++)
00488             decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1);
00489         break;
00490     case MODE_8k85:
00491         for (i = 0; i < 4; i++)
00492             decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1);
00493         break;
00494     case MODE_12k65:
00495         for (i = 0; i < 4; i++)
00496             decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
00497         break;
00498     case MODE_14k25:
00499         for (i = 0; i < 2; i++)
00500             decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
00501         for (i = 2; i < 4; i++)
00502             decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
00503         break;
00504     case MODE_15k85:
00505         for (i = 0; i < 4; i++)
00506             decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
00507         break;
00508     case MODE_18k25:
00509         for (i = 0; i < 4; i++)
00510             decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
00511                            ((int) pulse_hi[i] << 14), 4, 1);
00512         break;
00513     case MODE_19k85:
00514         for (i = 0; i < 2; i++)
00515             decode_5p_track(sig_pos[i], (int) pulse_lo[i] +
00516                            ((int) pulse_hi[i] << 10), 4, 1);
00517         for (i = 2; i < 4; i++)
00518             decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
00519                            ((int) pulse_hi[i] << 14), 4, 1);
00520         break;
00521     case MODE_23k05:
00522     case MODE_23k85:
00523         for (i = 0; i < 4; i++)
00524             decode_6p_track(sig_pos[i], (int) pulse_lo[i] +
00525                            ((int) pulse_hi[i] << 11), 4, 1);
00526         break;
00527     }
00528 
00529     memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE);
00530 
00531     for (i = 0; i < 4; i++)
00532         for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) {
00533             int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i;
00534 
00535             fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0;
00536         }
00537 }
00538 
00547 static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
00548                          float *fixed_gain_factor, float *pitch_gain)
00549 {
00550     const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :
00551                                                 qua_gain_7b[vq_gain]);
00552 
00553     *pitch_gain        = gains[0] * (1.0f / (1 << 14));
00554     *fixed_gain_factor = gains[1] * (1.0f / (1 << 11));
00555 }
00556 
00563 // XXX: Spec states this procedure should be applied when the pitch
00564 // lag is less than 64, but this checking seems absent in reference and AMR-NB
00565 static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
00566 {
00567     int i;
00568 
00569     /* Tilt part */
00570     for (i = AMRWB_SFR_SIZE - 1; i != 0; i--)
00571         fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef;
00572 
00573     /* Periodicity enhancement part */
00574     for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++)
00575         fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85;
00576 }
00577 
00584 // XXX: There is something wrong with the precision here! The magnitudes
00585 // of the energies are not correct. Please check the reference code carefully
00586 static float voice_factor(float *p_vector, float p_gain,
00587                           float *f_vector, float f_gain)
00588 {
00589     double p_ener = (double) ff_dot_productf(p_vector, p_vector,
00590                                              AMRWB_SFR_SIZE) * p_gain * p_gain;
00591     double f_ener = (double) ff_dot_productf(f_vector, f_vector,
00592                                              AMRWB_SFR_SIZE) * f_gain * f_gain;
00593 
00594     return (p_ener - f_ener) / (p_ener + f_ener);
00595 }
00596 
00607 static float *anti_sparseness(AMRWBContext *ctx,
00608                               float *fixed_vector, float *buf)
00609 {
00610     int ir_filter_nr;
00611 
00612     if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes
00613         return fixed_vector;
00614 
00615     if (ctx->pitch_gain[0] < 0.6) {
00616         ir_filter_nr = 0;      // strong filtering
00617     } else if (ctx->pitch_gain[0] < 0.9) {
00618         ir_filter_nr = 1;      // medium filtering
00619     } else
00620         ir_filter_nr = 2;      // no filtering
00621 
00622     /* detect 'onset' */
00623     if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) {
00624         if (ir_filter_nr < 2)
00625             ir_filter_nr++;
00626     } else {
00627         int i, count = 0;
00628 
00629         for (i = 0; i < 6; i++)
00630             if (ctx->pitch_gain[i] < 0.6)
00631                 count++;
00632 
00633         if (count > 2)
00634             ir_filter_nr = 0;
00635 
00636         if (ir_filter_nr > ctx->prev_ir_filter_nr + 1)
00637             ir_filter_nr--;
00638     }
00639 
00640     /* update ir filter strength history */
00641     ctx->prev_ir_filter_nr = ir_filter_nr;
00642 
00643     ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85);
00644 
00645     if (ir_filter_nr < 2) {
00646         int i;
00647         const float *coef = ir_filters_lookup[ir_filter_nr];
00648 
00649         /* Circular convolution code in the reference
00650          * decoder was modified to avoid using one
00651          * extra array. The filtered vector is given by:
00652          *
00653          * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }
00654          */
00655 
00656         memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE);
00657         for (i = 0; i < AMRWB_SFR_SIZE; i++)
00658             if (fixed_vector[i])
00659                 ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],
00660                                   AMRWB_SFR_SIZE);
00661         fixed_vector = buf;
00662     }
00663 
00664     return fixed_vector;
00665 }
00666 
00671 static float stability_factor(const float *isf, const float *isf_past)
00672 {
00673     int i;
00674     float acc = 0.0;
00675 
00676     for (i = 0; i < LP_ORDER - 1; i++)
00677         acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);
00678 
00679     // XXX: This part is not so clear from the reference code
00680     // the result is more accurate changing the "/ 256" to "* 512"
00681     return FFMAX(0.0, 1.25 - acc * 0.8 * 512);
00682 }
00683 
00695 static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
00696                             float voice_fac,  float stab_fac)
00697 {
00698     float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;
00699     float g0;
00700 
00701     // XXX: the following fixed-point constants used to in(de)crement
00702     // gain by 1.5dB were taken from the reference code, maybe it could
00703     // be simpler
00704     if (fixed_gain < *prev_tr_gain) {
00705         g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *
00706                      (6226 * (1.0f / (1 << 15)))); // +1.5 dB
00707     } else
00708         g0 = FFMAX(*prev_tr_gain, fixed_gain *
00709                     (27536 * (1.0f / (1 << 15)))); // -1.5 dB
00710 
00711     *prev_tr_gain = g0; // update next frame threshold
00712 
00713     return sm_fac * g0 + (1 - sm_fac) * fixed_gain;
00714 }
00715 
00722 static void pitch_enhancer(float *fixed_vector, float voice_fac)
00723 {
00724     int i;
00725     float cpe  = 0.125 * (1 + voice_fac);
00726     float last = fixed_vector[0]; // holds c(i - 1)
00727 
00728     fixed_vector[0] -= cpe * fixed_vector[1];
00729 
00730     for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) {
00731         float cur = fixed_vector[i];
00732 
00733         fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]);
00734         last = cur;
00735     }
00736 
00737     fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last;
00738 }
00739 
00750 static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
00751                       float fixed_gain, const float *fixed_vector,
00752                       float *samples)
00753 {
00754     ff_weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,
00755                             ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);
00756 
00757     /* emphasize pitch vector contribution in low bitrate modes */
00758     if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
00759         int i;
00760         float energy = ff_dot_productf(excitation, excitation,
00761                                        AMRWB_SFR_SIZE);
00762 
00763         // XXX: Weird part in both ref code and spec. A unknown parameter
00764         // {beta} seems to be identical to the current pitch gain
00765         float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0];
00766 
00767         for (i = 0; i < AMRWB_SFR_SIZE; i++)
00768             excitation[i] += pitch_factor * ctx->pitch_vector[i];
00769 
00770         ff_scale_vector_to_given_sum_of_squares(excitation, excitation,
00771                                                 energy, AMRWB_SFR_SIZE);
00772     }
00773 
00774     ff_celp_lp_synthesis_filterf(samples, lpc, excitation,
00775                                  AMRWB_SFR_SIZE, LP_ORDER);
00776 }
00777 
00787 static void de_emphasis(float *out, float *in, float m, float mem[1])
00788 {
00789     int i;
00790 
00791     out[0] = in[0] + m * mem[0];
00792 
00793     for (i = 1; i < AMRWB_SFR_SIZE; i++)
00794          out[i] = in[i] + out[i - 1] * m;
00795 
00796     mem[0] = out[AMRWB_SFR_SIZE - 1];
00797 }
00798 
00807 static void upsample_5_4(float *out, const float *in, int o_size)
00808 {
00809     const float *in0 = in - UPS_FIR_SIZE + 1;
00810     int i, j, k;
00811     int int_part = 0, frac_part;
00812 
00813     i = 0;
00814     for (j = 0; j < o_size / 5; j++) {
00815         out[i] = in[int_part];
00816         frac_part = 4;
00817         i++;
00818 
00819         for (k = 1; k < 5; k++) {
00820             out[i] = ff_dot_productf(in0 + int_part, upsample_fir[4 - frac_part],
00821                                      UPS_MEM_SIZE);
00822             int_part++;
00823             frac_part--;
00824             i++;
00825         }
00826     }
00827 }
00828 
00838 static float find_hb_gain(AMRWBContext *ctx, const float *synth,
00839                           uint16_t hb_idx, uint8_t vad)
00840 {
00841     int wsp = (vad > 0);
00842     float tilt;
00843 
00844     if (ctx->fr_cur_mode == MODE_23k85)
00845         return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
00846 
00847     tilt = ff_dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
00848            ff_dot_productf(synth, synth, AMRWB_SFR_SIZE);
00849 
00850     /* return gain bounded by [0.1, 1.0] */
00851     return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0);
00852 }
00853 
00863 static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
00864                                  const float *synth_exc, float hb_gain)
00865 {
00866     int i;
00867     float energy = ff_dot_productf(synth_exc, synth_exc, AMRWB_SFR_SIZE);
00868 
00869     /* Generate a white-noise excitation */
00870     for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
00871         hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);
00872 
00873     ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc,
00874                                             energy * hb_gain * hb_gain,
00875                                             AMRWB_SFR_SIZE_16k);
00876 }
00877 
00881 static float auto_correlation(float *diff_isf, float mean, int lag)
00882 {
00883     int i;
00884     float sum = 0.0;
00885 
00886     for (i = 7; i < LP_ORDER - 2; i++) {
00887         float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean);
00888         sum += prod * prod;
00889     }
00890     return sum;
00891 }
00892 
00900 static void extrapolate_isf(float out[LP_ORDER_16k], float isf[LP_ORDER])
00901 {
00902     float diff_isf[LP_ORDER - 2], diff_mean;
00903     float *diff_hi = diff_isf - LP_ORDER + 1; // diff array for extrapolated indexes
00904     float corr_lag[3];
00905     float est, scale;
00906     int i, i_max_corr;
00907 
00908     memcpy(out, isf, (LP_ORDER - 1) * sizeof(float));
00909     out[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
00910 
00911     /* Calculate the difference vector */
00912     for (i = 0; i < LP_ORDER - 2; i++)
00913         diff_isf[i] = isf[i + 1] - isf[i];
00914 
00915     diff_mean = 0.0;
00916     for (i = 2; i < LP_ORDER - 2; i++)
00917         diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4));
00918 
00919     /* Find which is the maximum autocorrelation */
00920     i_max_corr = 0;
00921     for (i = 0; i < 3; i++) {
00922         corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);
00923 
00924         if (corr_lag[i] > corr_lag[i_max_corr])
00925             i_max_corr = i;
00926     }
00927     i_max_corr++;
00928 
00929     for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
00930         out[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
00931                             - isf[i - 2 - i_max_corr];
00932 
00933     /* Calculate an estimate for ISF(18) and scale ISF based on the error */
00934     est   = 7965 + (out[2] - out[3] - out[4]) / 6.0;
00935     scale = 0.5 * (FFMIN(est, 7600) - out[LP_ORDER - 2]) /
00936             (out[LP_ORDER_16k - 2] - out[LP_ORDER - 2]);
00937 
00938     for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
00939         diff_hi[i] = scale * (out[i] - out[i - 1]);
00940 
00941     /* Stability insurance */
00942     for (i = LP_ORDER; i < LP_ORDER_16k - 1; i++)
00943         if (diff_hi[i] + diff_hi[i - 1] < 5.0) {
00944             if (diff_hi[i] > diff_hi[i - 1]) {
00945                 diff_hi[i - 1] = 5.0 - diff_hi[i];
00946             } else
00947                 diff_hi[i] = 5.0 - diff_hi[i - 1];
00948         }
00949 
00950     for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
00951         out[i] = out[i - 1] + diff_hi[i] * (1.0f / (1 << 15));
00952 
00953     /* Scale the ISF vector for 16000 Hz */
00954     for (i = 0; i < LP_ORDER_16k - 1; i++)
00955         out[i] *= 0.8;
00956 }
00957 
00967 static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
00968 {
00969     int i;
00970     float fac = gamma;
00971 
00972     for (i = 0; i < size; i++) {
00973         out[i] = lpc[i] * fac;
00974         fac   *= gamma;
00975     }
00976 }
00977 
00989 static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
00990                          const float *exc, const float *isf, const float *isf_past)
00991 {
00992     float hb_lpc[LP_ORDER_16k];
00993     enum Mode mode = ctx->fr_cur_mode;
00994 
00995     if (mode == MODE_6k60) {
00996         float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation
00997         double e_isp[LP_ORDER_16k];
00998 
00999         ff_weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],
01000                                 1.0 - isfp_inter[subframe], LP_ORDER);
01001 
01002         extrapolate_isf(e_isf, e_isf);
01003 
01004         e_isf[LP_ORDER_16k - 1] *= 2.0;
01005         ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);
01006         ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k);
01007 
01008         lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k);
01009     } else {
01010         lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER);
01011     }
01012 
01013     ff_celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,
01014                                  (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);
01015 }
01016 
01028 static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
01029                           float mem[HB_FIR_SIZE], const float *in)
01030 {
01031     int i, j;
01032     float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples
01033 
01034     memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
01035     memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float));
01036 
01037     for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {
01038         out[i] = 0.0;
01039         for (j = 0; j <= HB_FIR_SIZE; j++)
01040             out[i] += data[i + j] * fir_coef[j];
01041     }
01042 
01043     memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));
01044 }
01045 
01049 static void update_sub_state(AMRWBContext *ctx)
01050 {
01051     memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE],
01052             (AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float));
01053 
01054     memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float));
01055     memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float));
01056 
01057     memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE],
01058             LP_ORDER * sizeof(float));
01059     memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE],
01060             UPS_MEM_SIZE * sizeof(float));
01061     memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k],
01062             LP_ORDER_16k * sizeof(float));
01063 }
01064 
01065 static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
01066                               AVPacket *avpkt)
01067 {
01068     AMRWBContext *ctx  = avctx->priv_data;
01069     AMRWBFrame   *cf   = &ctx->frame;
01070     const uint8_t *buf = avpkt->data;
01071     int buf_size       = avpkt->size;
01072     int expected_fr_size, header_size;
01073     float *buf_out = data;
01074     float spare_vector[AMRWB_SFR_SIZE];      // extra stack space to hold result from anti-sparseness processing
01075     float fixed_gain_factor;                 // fixed gain correction factor (gamma)
01076     float *synth_fixed_vector;               // pointer to the fixed vector that synthesis should use
01077     float synth_fixed_gain;                  // the fixed gain that synthesis should use
01078     float voice_fac, stab_fac;               // parameters used for gain smoothing
01079     float synth_exc[AMRWB_SFR_SIZE];         // post-processed excitation for synthesis
01080     float hb_exc[AMRWB_SFR_SIZE_16k];        // excitation for the high frequency band
01081     float hb_samples[AMRWB_SFR_SIZE_16k];    // filtered high-band samples from synthesis
01082     float hb_gain;
01083     int sub, i;
01084 
01085     header_size      = decode_mime_header(ctx, buf);
01086     expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
01087 
01088     if (buf_size < expected_fr_size) {
01089         av_log(avctx, AV_LOG_ERROR,
01090             "Frame too small (%d bytes). Truncated file?\n", buf_size);
01091         *data_size = 0;
01092         return buf_size;
01093     }
01094 
01095     if (!ctx->fr_quality || ctx->fr_cur_mode > MODE_SID)
01096         av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");
01097 
01098     if (ctx->fr_cur_mode == MODE_SID) /* Comfort noise frame */
01099         av_log_missing_feature(avctx, "SID mode", 1);
01100 
01101     if (ctx->fr_cur_mode >= MODE_SID)
01102         return -1;
01103 
01104     ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
01105         buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);
01106 
01107     /* Decode the quantized ISF vector */
01108     if (ctx->fr_cur_mode == MODE_6k60) {
01109         decode_isf_indices_36b(cf->isp_id, ctx->isf_cur);
01110     } else {
01111         decode_isf_indices_46b(cf->isp_id, ctx->isf_cur);
01112     }
01113 
01114     isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past);
01115     ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1);
01116 
01117     stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);
01118 
01119     ctx->isf_cur[LP_ORDER - 1] *= 2.0;
01120     ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER);
01121 
01122     /* Generate a ISP vector for each subframe */
01123     if (ctx->first_frame) {
01124         ctx->first_frame = 0;
01125         memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double));
01126     }
01127     interpolate_isp(ctx->isp, ctx->isp_sub4_past);
01128 
01129     for (sub = 0; sub < 4; sub++)
01130         ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER);
01131 
01132     for (sub = 0; sub < 4; sub++) {
01133         const AMRWBSubFrame *cur_subframe = &cf->subframe[sub];
01134         float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;
01135 
01136         /* Decode adaptive codebook (pitch vector) */
01137         decode_pitch_vector(ctx, cur_subframe, sub);
01138         /* Decode innovative codebook (fixed vector) */
01139         decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih,
01140                             cur_subframe->pul_il, ctx->fr_cur_mode);
01141 
01142         pitch_sharpening(ctx, ctx->fixed_vector);
01143 
01144         decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode,
01145                      &fixed_gain_factor, &ctx->pitch_gain[0]);
01146 
01147         ctx->fixed_gain[0] =
01148             ff_amr_set_fixed_gain(fixed_gain_factor,
01149                        ff_dot_productf(ctx->fixed_vector, ctx->fixed_vector,
01150                                        AMRWB_SFR_SIZE) / AMRWB_SFR_SIZE,
01151                        ctx->prediction_error,
01152                        ENERGY_MEAN, energy_pred_fac);
01153 
01154         /* Calculate voice factor and store tilt for next subframe */
01155         voice_fac      = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0],
01156                                       ctx->fixed_vector, ctx->fixed_gain[0]);
01157         ctx->tilt_coef = voice_fac * 0.25 + 0.25;
01158 
01159         /* Construct current excitation */
01160         for (i = 0; i < AMRWB_SFR_SIZE; i++) {
01161             ctx->excitation[i] *= ctx->pitch_gain[0];
01162             ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i];
01163             ctx->excitation[i] = truncf(ctx->excitation[i]);
01164         }
01165 
01166         /* Post-processing of excitation elements */
01167         synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain,
01168                                           voice_fac, stab_fac);
01169 
01170         synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector,
01171                                              spare_vector);
01172 
01173         pitch_enhancer(synth_fixed_vector, voice_fac);
01174 
01175         synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
01176                   synth_fixed_vector, &ctx->samples_az[LP_ORDER]);
01177 
01178         /* Synthesis speech post-processing */
01179         de_emphasis(&ctx->samples_up[UPS_MEM_SIZE],
01180                     &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);
01181 
01182         ff_acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE],
01183             &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles,
01184             hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE);
01185 
01186         upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],
01187                      AMRWB_SFR_SIZE_16k);
01188 
01189         /* High frequency band (6.4 - 7.0 kHz) generation part */
01190         ff_acelp_apply_order_2_transfer_function(hb_samples,
01191             &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles,
01192             hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE);
01193 
01194         hb_gain = find_hb_gain(ctx, hb_samples,
01195                                cur_subframe->hb_gain, cf->vad);
01196 
01197         scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);
01198 
01199         hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k],
01200                      hb_exc, ctx->isf_cur, ctx->isf_past_final);
01201 
01202         /* High-band post-processing filters */
01203         hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,
01204                       &ctx->samples_hb[LP_ORDER_16k]);
01205 
01206         if (ctx->fr_cur_mode == MODE_23k85)
01207             hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,
01208                           hb_samples);
01209 
01210         /* Add the low and high frequency bands */
01211         for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
01212             sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15));
01213 
01214         /* Update buffers and history */
01215         update_sub_state(ctx);
01216     }
01217 
01218     /* update state for next frame */
01219     memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
01220     memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
01221 
01222     /* report how many samples we got */
01223     *data_size = 4 * AMRWB_SFR_SIZE_16k * sizeof(float);
01224 
01225     return expected_fr_size;
01226 }
01227 
01228 AVCodec ff_amrwb_decoder = {
01229     .name           = "amrwb",
01230     .type           = AVMEDIA_TYPE_AUDIO,
01231     .id             = CODEC_ID_AMR_WB,
01232     .priv_data_size = sizeof(AMRWBContext),
01233     .init           = amrwb_decode_init,
01234     .decode         = amrwb_decode_frame,
01235     .long_name      = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate WideBand"),
01236     .sample_fmts    = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
01237 };