LCOV - code coverage report
Current view: top level - src/libavcodec - amrwbdec.c (source / functions) Hit Total Coverage
Test: coverage.info Lines: 474 490 96.7 %
Date: 2017-03-25 17:02:41 Functions: 35 35 100.0 %

          Line data    Source code
       1             : /*
       2             :  * AMR wideband decoder
       3             :  * Copyright (c) 2010 Marcelo Galvao Povoa
       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             : /**
      23             :  * @file
      24             :  * AMR wideband decoder
      25             :  */
      26             : 
      27             : #include "libavutil/channel_layout.h"
      28             : #include "libavutil/common.h"
      29             : #include "libavutil/float_dsp.h"
      30             : #include "libavutil/lfg.h"
      31             : 
      32             : #include "avcodec.h"
      33             : #include "lsp.h"
      34             : #include "celp_filters.h"
      35             : #include "celp_math.h"
      36             : #include "acelp_filters.h"
      37             : #include "acelp_vectors.h"
      38             : #include "acelp_pitch_delay.h"
      39             : #include "internal.h"
      40             : 
      41             : #define AMR_USE_16BIT_TABLES
      42             : #include "amr.h"
      43             : 
      44             : #include "amrwbdata.h"
      45             : #include "mips/amrwbdec_mips.h"
      46             : 
      47             : typedef struct AMRWBContext {
      48             :     AMRWBFrame                             frame; ///< AMRWB parameters decoded from bitstream
      49             :     enum Mode                        fr_cur_mode; ///< mode index of current frame
      50             :     uint8_t                           fr_quality; ///< frame quality index (FQI)
      51             :     float                      isf_cur[LP_ORDER]; ///< working ISF vector from current frame
      52             :     float                   isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame
      53             :     float               isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame
      54             :     double                      isp[4][LP_ORDER]; ///< ISP vectors from current frame
      55             :     double               isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame
      56             : 
      57             :     float                   lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector
      58             : 
      59             :     uint8_t                       base_pitch_lag; ///< integer part of pitch lag for the next relative subframe
      60             :     uint8_t                        pitch_lag_int; ///< integer part of pitch lag of the previous subframe
      61             : 
      62             :     float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history
      63             :     float                            *excitation; ///< points to current excitation in excitation_buf[]
      64             : 
      65             :     float           pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe
      66             :     float           fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe
      67             : 
      68             :     float                    prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
      69             :     float                          pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes
      70             :     float                          fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes
      71             : 
      72             :     float                              tilt_coef; ///< {beta_1} related to the voicing of the previous subframe
      73             : 
      74             :     float                 prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness to determine "onset"
      75             :     uint8_t                    prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
      76             :     float                           prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold
      77             : 
      78             :     float samples_az[LP_ORDER + AMRWB_SFR_SIZE];         ///< low-band samples and memory from synthesis at 12.8kHz
      79             :     float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE];     ///< low-band samples and memory processed for upsampling
      80             :     float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band samples and memory from synthesis at 16kHz
      81             : 
      82             :     float          hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters
      83             :     float                           demph_mem[1]; ///< previous value in the de-emphasis filter
      84             :     float               bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter
      85             :     float                 lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter
      86             : 
      87             :     AVLFG                                   prng; ///< random number generator for white noise excitation
      88             :     uint8_t                          first_frame; ///< flag active during decoding of the first frame
      89             :     ACELPFContext                     acelpf_ctx; ///< context for filters for ACELP-based codecs
      90             :     ACELPVContext                     acelpv_ctx; ///< context for vector operations for ACELP-based codecs
      91             :     CELPFContext                       celpf_ctx; ///< context for filters for CELP-based codecs
      92             :     CELPMContext                       celpm_ctx; ///< context for fixed point math operations
      93             : 
      94             : } AMRWBContext;
      95             : 
      96          22 : static av_cold int amrwb_decode_init(AVCodecContext *avctx)
      97             : {
      98          22 :     AMRWBContext *ctx = avctx->priv_data;
      99             :     int i;
     100             : 
     101          22 :     if (avctx->channels > 1) {
     102           0 :         avpriv_report_missing_feature(avctx, "multi-channel AMR");
     103           0 :         return AVERROR_PATCHWELCOME;
     104             :     }
     105             : 
     106          22 :     avctx->channels       = 1;
     107          22 :     avctx->channel_layout = AV_CH_LAYOUT_MONO;
     108          22 :     if (!avctx->sample_rate)
     109           0 :         avctx->sample_rate = 16000;
     110          22 :     avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;
     111             : 
     112          22 :     av_lfg_init(&ctx->prng, 1);
     113             : 
     114          22 :     ctx->excitation  = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1];
     115          22 :     ctx->first_frame = 1;
     116             : 
     117         374 :     for (i = 0; i < LP_ORDER; i++)
     118         352 :         ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15));
     119             : 
     120         110 :     for (i = 0; i < 4; i++)
     121          88 :         ctx->prediction_error[i] = MIN_ENERGY;
     122             : 
     123          22 :     ff_acelp_filter_init(&ctx->acelpf_ctx);
     124          22 :     ff_acelp_vectors_init(&ctx->acelpv_ctx);
     125          22 :     ff_celp_filter_init(&ctx->celpf_ctx);
     126          22 :     ff_celp_math_init(&ctx->celpm_ctx);
     127             : 
     128          22 :     return 0;
     129             : }
     130             : 
     131             : /**
     132             :  * Decode the frame header in the "MIME/storage" format. This format
     133             :  * is simpler and does not carry the auxiliary frame information.
     134             :  *
     135             :  * @param[in] ctx                  The Context
     136             :  * @param[in] buf                  Pointer to the input buffer
     137             :  *
     138             :  * @return The decoded header length in bytes
     139             :  */
     140        6768 : static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
     141             : {
     142             :     /* Decode frame header (1st octet) */
     143        6768 :     ctx->fr_cur_mode  = buf[0] >> 3 & 0x0F;
     144        6768 :     ctx->fr_quality   = (buf[0] & 0x4) == 0x4;
     145             : 
     146        6768 :     return 1;
     147             : }
     148             : 
     149             : /**
     150             :  * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
     151             :  *
     152             :  * @param[in]  ind                 Array of 5 indexes
     153             :  * @param[out] isf_q               Buffer for isf_q[LP_ORDER]
     154             :  */
     155         512 : static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
     156             : {
     157             :     int i;
     158             : 
     159        5120 :     for (i = 0; i < 9; i++)
     160        4608 :         isf_q[i]      = dico1_isf[ind[0]][i]      * (1.0f / (1 << 15));
     161             : 
     162        4096 :     for (i = 0; i < 7; i++)
     163        3584 :         isf_q[i + 9]  = dico2_isf[ind[1]][i]      * (1.0f / (1 << 15));
     164             : 
     165        3072 :     for (i = 0; i < 5; i++)
     166        2560 :         isf_q[i]     += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15));
     167             : 
     168        2560 :     for (i = 0; i < 4; i++)
     169        2048 :         isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15));
     170             : 
     171        4096 :     for (i = 0; i < 7; i++)
     172        3584 :         isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15));
     173         512 : }
     174             : 
     175             : /**
     176             :  * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
     177             :  *
     178             :  * @param[in]  ind                 Array of 7 indexes
     179             :  * @param[out] isf_q               Buffer for isf_q[LP_ORDER]
     180             :  */
     181        6256 : static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
     182             : {
     183             :     int i;
     184             : 
     185       62560 :     for (i = 0; i < 9; i++)
     186       56304 :         isf_q[i]       = dico1_isf[ind[0]][i]  * (1.0f / (1 << 15));
     187             : 
     188       50048 :     for (i = 0; i < 7; i++)
     189       43792 :         isf_q[i + 9]   = dico2_isf[ind[1]][i]  * (1.0f / (1 << 15));
     190             : 
     191       25024 :     for (i = 0; i < 3; i++)
     192       18768 :         isf_q[i]      += dico21_isf[ind[2]][i] * (1.0f / (1 << 15));
     193             : 
     194       25024 :     for (i = 0; i < 3; i++)
     195       18768 :         isf_q[i + 3]  += dico22_isf[ind[3]][i] * (1.0f / (1 << 15));
     196             : 
     197       25024 :     for (i = 0; i < 3; i++)
     198       18768 :         isf_q[i + 6]  += dico23_isf[ind[4]][i] * (1.0f / (1 << 15));
     199             : 
     200       25024 :     for (i = 0; i < 3; i++)
     201       18768 :         isf_q[i + 9]  += dico24_isf[ind[5]][i] * (1.0f / (1 << 15));
     202             : 
     203       31280 :     for (i = 0; i < 4; i++)
     204       25024 :         isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15));
     205        6256 : }
     206             : 
     207             : /**
     208             :  * Apply mean and past ISF values using the prediction factor.
     209             :  * Updates past ISF vector.
     210             :  *
     211             :  * @param[in,out] isf_q            Current quantized ISF
     212             :  * @param[in,out] isf_past         Past quantized ISF
     213             :  */
     214        6768 : static void isf_add_mean_and_past(float *isf_q, float *isf_past)
     215             : {
     216             :     int i;
     217             :     float tmp;
     218             : 
     219      115056 :     for (i = 0; i < LP_ORDER; i++) {
     220      108288 :         tmp = isf_q[i];
     221      108288 :         isf_q[i] += isf_mean[i] * (1.0f / (1 << 15));
     222      108288 :         isf_q[i] += PRED_FACTOR * isf_past[i];
     223      108288 :         isf_past[i] = tmp;
     224             :     }
     225        6768 : }
     226             : 
     227             : /**
     228             :  * Interpolate the fourth ISP vector from current and past frames
     229             :  * to obtain an ISP vector for each subframe.
     230             :  *
     231             :  * @param[in,out] isp_q            ISPs for each subframe
     232             :  * @param[in]     isp4_past        Past ISP for subframe 4
     233             :  */
     234        6768 : static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
     235             : {
     236             :     int i, k;
     237             : 
     238       27072 :     for (k = 0; k < 3; k++) {
     239       20304 :         float c = isfp_inter[k];
     240      345168 :         for (i = 0; i < LP_ORDER; i++)
     241      324864 :             isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i];
     242             :     }
     243        6768 : }
     244             : 
     245             : /**
     246             :  * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
     247             :  * Calculate integer lag and fractional lag always using 1/4 resolution.
     248             :  * In 1st and 3rd subframes the index is relative to last subframe integer lag.
     249             :  *
     250             :  * @param[out]    lag_int          Decoded integer pitch lag
     251             :  * @param[out]    lag_frac         Decoded fractional pitch lag
     252             :  * @param[in]     pitch_index      Adaptive codebook pitch index
     253             :  * @param[in,out] base_lag_int     Base integer lag used in relative subframes
     254             :  * @param[in]     subframe         Current subframe index (0 to 3)
     255             :  */
     256       22976 : static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
     257             :                                   uint8_t *base_lag_int, int subframe)
     258             : {
     259       22976 :     if (subframe == 0 || subframe == 2) {
     260       11488 :         if (pitch_index < 376) {
     261        8295 :             *lag_int  = (pitch_index + 137) >> 2;
     262        8295 :             *lag_frac = pitch_index - (*lag_int << 2) + 136;
     263        3193 :         } else if (pitch_index < 440) {
     264        1402 :             *lag_int  = (pitch_index + 257 - 376) >> 1;
     265        1402 :             *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) * 2;
     266             :             /* the actual resolution is 1/2 but expressed as 1/4 */
     267             :         } else {
     268        1791 :             *lag_int  = pitch_index - 280;
     269        1791 :             *lag_frac = 0;
     270             :         }
     271             :         /* minimum lag for next subframe */
     272       11488 :         *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
     273             :                                 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
     274             :         // XXX: the spec states clearly that *base_lag_int should be
     275             :         // the nearest integer to *lag_int (minus 8), but the ref code
     276             :         // actually always uses its floor, I'm following the latter
     277             :     } else {
     278       11488 :         *lag_int  = (pitch_index + 1) >> 2;
     279       11488 :         *lag_frac = pitch_index - (*lag_int << 2);
     280       11488 :         *lag_int += *base_lag_int;
     281             :     }
     282       22976 : }
     283             : 
     284             : /**
     285             :  * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
     286             :  * The description is analogous to decode_pitch_lag_high, but in 6k60 the
     287             :  * relative index is used for all subframes except the first.
     288             :  */
     289        4096 : static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
     290             :                                  uint8_t *base_lag_int, int subframe, enum Mode mode)
     291             : {
     292        4096 :     if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {
     293        1536 :         if (pitch_index < 116) {
     294         847 :             *lag_int  = (pitch_index + 69) >> 1;
     295         847 :             *lag_frac = (pitch_index - (*lag_int << 1) + 68) * 2;
     296             :         } else {
     297         689 :             *lag_int  = pitch_index - 24;
     298         689 :             *lag_frac = 0;
     299             :         }
     300             :         // XXX: same problem as before
     301        1536 :         *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
     302             :                                 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
     303             :     } else {
     304        2560 :         *lag_int  = (pitch_index + 1) >> 1;
     305        2560 :         *lag_frac = (pitch_index - (*lag_int << 1)) * 2;
     306        2560 :         *lag_int += *base_lag_int;
     307             :     }
     308        4096 : }
     309             : 
     310             : /**
     311             :  * Find the pitch vector by interpolating the past excitation at the
     312             :  * pitch delay, which is obtained in this function.
     313             :  *
     314             :  * @param[in,out] ctx              The context
     315             :  * @param[in]     amr_subframe     Current subframe data
     316             :  * @param[in]     subframe         Current subframe index (0 to 3)
     317             :  */
     318       27072 : static void decode_pitch_vector(AMRWBContext *ctx,
     319             :                                 const AMRWBSubFrame *amr_subframe,
     320             :                                 const int subframe)
     321             : {
     322             :     int pitch_lag_int, pitch_lag_frac;
     323             :     int i;
     324       27072 :     float *exc     = ctx->excitation;
     325       27072 :     enum Mode mode = ctx->fr_cur_mode;
     326             : 
     327       27072 :     if (mode <= MODE_8k85) {
     328        4096 :         decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
     329             :                               &ctx->base_pitch_lag, subframe, mode);
     330             :     } else
     331       22976 :         decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
     332             :                               &ctx->base_pitch_lag, subframe);
     333             : 
     334       27072 :     ctx->pitch_lag_int = pitch_lag_int;
     335       27072 :     pitch_lag_int += pitch_lag_frac > 0;
     336             : 
     337             :     /* Calculate the pitch vector by interpolating the past excitation at the
     338             :        pitch lag using a hamming windowed sinc function */
     339      108288 :     ctx->acelpf_ctx.acelp_interpolatef(exc,
     340       54144 :                           exc + 1 - pitch_lag_int,
     341             :                           ac_inter, 4,
     342       27072 :                           pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
     343             :                           LP_ORDER, AMRWB_SFR_SIZE + 1);
     344             : 
     345             :     /* Check which pitch signal path should be used
     346             :      * 6k60 and 8k85 modes have the ltp flag set to 0 */
     347       27072 :     if (amr_subframe->ltp) {
     348        7913 :         memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float));
     349             :     } else {
     350     1245335 :         for (i = 0; i < AMRWB_SFR_SIZE; i++)
     351     2452352 :             ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] +
     352     1226176 :                                    0.18 * exc[i + 1];
     353       19159 :         memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float));
     354             :     }
     355       27072 : }
     356             : 
     357             : /** Get x bits in the index interval [lsb,lsb+len-1] inclusive */
     358             : #define BIT_STR(x,lsb,len) av_mod_uintp2((x) >> (lsb), (len))
     359             : 
     360             : /** Get the bit at specified position */
     361             : #define BIT_POS(x, p) (((x) >> (p)) & 1)
     362             : 
     363             : /**
     364             :  * The next six functions decode_[i]p_track decode exactly i pulses
     365             :  * positions and amplitudes (-1 or 1) in a subframe track using
     366             :  * an encoded pulse indexing (TS 26.190 section 5.8.2).
     367             :  *
     368             :  * The results are given in out[], in which a negative number means
     369             :  * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).
     370             :  *
     371             :  * @param[out] out                 Output buffer (writes i elements)
     372             :  * @param[in]  code                Pulse index (no. of bits varies, see below)
     373             :  * @param[in]  m                   (log2) Number of potential positions
     374             :  * @param[in]  off                 Offset for decoded positions
     375             :  */
     376      105998 : static inline void decode_1p_track(int *out, int code, int m, int off)
     377             : {
     378      105998 :     int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits
     379             : 
     380      105998 :     out[0] = BIT_POS(code, m) ? -pos : pos;
     381      105998 : }
     382             : 
     383      155129 : static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits
     384             : {
     385      155129 :     int pos0 = BIT_STR(code, m, m) + off;
     386      155129 :     int pos1 = BIT_STR(code, 0, m) + off;
     387             : 
     388      155129 :     out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0;
     389      155129 :     out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1;
     390      155129 :     out[1] = pos0 > pos1 ? -out[1] : out[1];
     391      155129 : }
     392             : 
     393       68636 : static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits
     394             : {
     395       68636 :     int half_2p = BIT_POS(code, 2*m - 1) << (m - 1);
     396             : 
     397       68636 :     decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
     398             :                     m - 1, off + half_2p);
     399       68636 :     decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off);
     400       68636 : }
     401             : 
     402       32497 : static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits
     403             : {
     404             :     int half_4p, subhalf_2p;
     405       32497 :     int b_offset = 1 << (m - 1);
     406             : 
     407       32497 :     switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */
     408             :     case 0: /* 0 pulses in A, 4 pulses in B or vice versa */
     409        3597 :         half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses
     410        3597 :         subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);
     411             : 
     412        3597 :         decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),
     413        3597 :                         m - 2, off + half_4p + subhalf_2p);
     414        3597 :         decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),
     415             :                         m - 1, off + half_4p);
     416        3597 :         break;
     417             :     case 1: /* 1 pulse in A, 3 pulses in B */
     418        7976 :         decode_1p_track(out, BIT_STR(code,  3*m - 2, m),
     419             :                         m - 1, off);
     420        7976 :         decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),
     421             :                         m - 1, off + b_offset);
     422        7976 :         break;
     423             :     case 2: /* 2 pulses in each half */
     424       12780 :         decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),
     425             :                         m - 1, off);
     426       12780 :         decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),
     427             :                         m - 1, off + b_offset);
     428       12780 :         break;
     429             :     case 3: /* 3 pulses in A, 1 pulse in B */
     430        8144 :         decode_3p_track(out, BIT_STR(code, m, 3*m - 2),
     431             :                         m - 1, off);
     432        8144 :         decode_1p_track(out + 3, BIT_STR(code, 0, m),
     433             :                         m - 1, off + b_offset);
     434        8144 :         break;
     435             :     }
     436       32497 : }
     437             : 
     438       13050 : static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits
     439             : {
     440       13050 :     int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);
     441             : 
     442       13050 :     decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),
     443             :                     m - 1, off + half_3p);
     444             : 
     445       13050 :     decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);
     446       13050 : }
     447             : 
     448       42752 : static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits
     449             : {
     450       42752 :     int b_offset = 1 << (m - 1);
     451             :     /* which half has more pulses in cases 0 to 2 */
     452       42752 :     int half_more  = BIT_POS(code, 6*m - 5) << (m - 1);
     453       42752 :     int half_other = b_offset - half_more;
     454             : 
     455       42752 :     switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */
     456             :     case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
     457        1288 :         decode_1p_track(out, BIT_STR(code, 0, m),
     458             :                         m - 1, off + half_more);
     459        1288 :         decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
     460             :                         m - 1, off + half_more);
     461        1288 :         break;
     462             :     case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
     463        7666 :         decode_1p_track(out, BIT_STR(code, 0, m),
     464             :                         m - 1, off + half_other);
     465        7666 :         decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
     466             :                         m - 1, off + half_more);
     467        7666 :         break;
     468             :     case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
     469       20209 :         decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
     470             :                         m - 1, off + half_other);
     471       20209 :         decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),
     472             :                         m - 1, off + half_more);
     473       20209 :         break;
     474             :     case 3: /* 3 pulses in A, 3 pulses in B */
     475       13589 :         decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),
     476             :                         m - 1, off);
     477       13589 :         decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2),
     478             :                         m - 1, off + b_offset);
     479       13589 :         break;
     480             :     }
     481       42752 : }
     482             : 
     483             : /**
     484             :  * Decode the algebraic codebook index to pulse positions and signs,
     485             :  * then construct the algebraic codebook vector.
     486             :  *
     487             :  * @param[out] fixed_vector        Buffer for the fixed codebook excitation
     488             :  * @param[in]  pulse_hi            MSBs part of the pulse index array (higher modes only)
     489             :  * @param[in]  pulse_lo            LSBs part of the pulse index array
     490             :  * @param[in]  mode                Mode of the current frame
     491             :  */
     492       27072 : static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
     493             :                                 const uint16_t *pulse_lo, const enum Mode mode)
     494             : {
     495             :     /* sig_pos stores for each track the decoded pulse position indexes
     496             :      * (1-based) multiplied by its corresponding amplitude (+1 or -1) */
     497             :     int sig_pos[4][6];
     498       27072 :     int spacing = (mode == MODE_6k60) ? 2 : 4;
     499             :     int i, j;
     500             : 
     501       27072 :     switch (mode) {
     502             :     case MODE_6k60:
     503        6144 :         for (i = 0; i < 2; i++)
     504        4096 :             decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1);
     505        2048 :         break;
     506             :     case MODE_8k85:
     507       10240 :         for (i = 0; i < 4; i++)
     508        8192 :             decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1);
     509        2048 :         break;
     510             :     case MODE_12k65:
     511       20480 :         for (i = 0; i < 4; i++)
     512       16384 :             decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
     513        4096 :         break;
     514             :     case MODE_14k25:
     515        6144 :         for (i = 0; i < 2; i++)
     516        4096 :             decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
     517        6144 :         for (i = 2; i < 4; i++)
     518        4096 :             decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
     519        2048 :         break;
     520             :     case MODE_15k85:
     521       10240 :         for (i = 0; i < 4; i++)
     522        8192 :             decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
     523        2048 :         break;
     524             :     case MODE_18k25:
     525       10240 :         for (i = 0; i < 4; i++)
     526       16384 :             decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
     527        8192 :                            ((int) pulse_hi[i] << 14), 4, 1);
     528        2048 :         break;
     529             :     case MODE_19k85:
     530        6144 :         for (i = 0; i < 2; i++)
     531        8192 :             decode_5p_track(sig_pos[i], (int) pulse_lo[i] +
     532        4096 :                            ((int) pulse_hi[i] << 10), 4, 1);
     533        6144 :         for (i = 2; i < 4; i++)
     534        8192 :             decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
     535        4096 :                            ((int) pulse_hi[i] << 14), 4, 1);
     536        2048 :         break;
     537             :     case MODE_23k05:
     538             :     case MODE_23k85:
     539       53440 :         for (i = 0; i < 4; i++)
     540       85504 :             decode_6p_track(sig_pos[i], (int) pulse_lo[i] +
     541       42752 :                            ((int) pulse_hi[i] << 11), 4, 1);
     542       10688 :         break;
     543             :     }
     544             : 
     545       27072 :     memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE);
     546             : 
     547      135360 :     for (i = 0; i < 4; i++)
     548      524544 :         for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) {
     549      416256 :             int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i;
     550             : 
     551      416256 :             fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0;
     552             :         }
     553       27072 : }
     554             : 
     555             : /**
     556             :  * Decode pitch gain and fixed gain correction factor.
     557             :  *
     558             :  * @param[in]  vq_gain             Vector-quantized index for gains
     559             :  * @param[in]  mode                Mode of the current frame
     560             :  * @param[out] fixed_gain_factor   Decoded fixed gain correction factor
     561             :  * @param[out] pitch_gain          Decoded pitch gain
     562             :  */
     563       27072 : static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
     564             :                          float *fixed_gain_factor, float *pitch_gain)
     565             : {
     566       50048 :     const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :
     567       22976 :                                                 qua_gain_7b[vq_gain]);
     568             : 
     569       27072 :     *pitch_gain        = gains[0] * (1.0f / (1 << 14));
     570       27072 :     *fixed_gain_factor = gains[1] * (1.0f / (1 << 11));
     571       27072 : }
     572             : 
     573             : /**
     574             :  * Apply pitch sharpening filters to the fixed codebook vector.
     575             :  *
     576             :  * @param[in]     ctx              The context
     577             :  * @param[in,out] fixed_vector     Fixed codebook excitation
     578             :  */
     579             : // XXX: Spec states this procedure should be applied when the pitch
     580             : // lag is less than 64, but this checking seems absent in reference and AMR-NB
     581       27072 : static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
     582             : {
     583             :     int i;
     584             : 
     585             :     /* Tilt part */
     586     1732608 :     for (i = AMRWB_SFR_SIZE - 1; i != 0; i--)
     587     1705536 :         fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef;
     588             : 
     589             :     /* Periodicity enhancement part */
     590      204025 :     for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++)
     591      176953 :         fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85;
     592       27072 : }
     593             : 
     594             : /**
     595             :  * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
     596             :  *
     597             :  * @param[in] p_vector, f_vector   Pitch and fixed excitation vectors
     598             :  * @param[in] p_gain, f_gain       Pitch and fixed gains
     599             :  * @param[in] ctx                  The context
     600             :  */
     601             : // XXX: There is something wrong with the precision here! The magnitudes
     602             : // of the energies are not correct. Please check the reference code carefully
     603       27072 : static float voice_factor(float *p_vector, float p_gain,
     604             :                           float *f_vector, float f_gain,
     605             :                           CELPMContext *ctx)
     606             : {
     607       54144 :     double p_ener = (double) ctx->dot_productf(p_vector, p_vector,
     608       27072 :                                                           AMRWB_SFR_SIZE) *
     609       27072 :                     p_gain * p_gain;
     610       54144 :     double f_ener = (double) ctx->dot_productf(f_vector, f_vector,
     611       27072 :                                                           AMRWB_SFR_SIZE) *
     612       27072 :                     f_gain * f_gain;
     613             : 
     614       27072 :     return (p_ener - f_ener) / (p_ener + f_ener);
     615             : }
     616             : 
     617             : /**
     618             :  * Reduce fixed vector sparseness by smoothing with one of three IR filters,
     619             :  * also known as "adaptive phase dispersion".
     620             :  *
     621             :  * @param[in]     ctx              The context
     622             :  * @param[in,out] fixed_vector     Unfiltered fixed vector
     623             :  * @param[out]    buf              Space for modified vector if necessary
     624             :  *
     625             :  * @return The potentially overwritten filtered fixed vector address
     626             :  */
     627       27072 : static float *anti_sparseness(AMRWBContext *ctx,
     628             :                               float *fixed_vector, float *buf)
     629             : {
     630             :     int ir_filter_nr;
     631             : 
     632       27072 :     if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes
     633       22976 :         return fixed_vector;
     634             : 
     635        4096 :     if (ctx->pitch_gain[0] < 0.6) {
     636        2254 :         ir_filter_nr = 0;      // strong filtering
     637        1842 :     } else if (ctx->pitch_gain[0] < 0.9) {
     638         712 :         ir_filter_nr = 1;      // medium filtering
     639             :     } else
     640        1130 :         ir_filter_nr = 2;      // no filtering
     641             : 
     642             :     /* detect 'onset' */
     643        4096 :     if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) {
     644          57 :         if (ir_filter_nr < 2)
     645          35 :             ir_filter_nr++;
     646             :     } else {
     647        4039 :         int i, count = 0;
     648             : 
     649       28273 :         for (i = 0; i < 6; i++)
     650       24234 :             if (ctx->pitch_gain[i] < 0.6)
     651       13350 :                 count++;
     652             : 
     653        4039 :         if (count > 2)
     654        2720 :             ir_filter_nr = 0;
     655             : 
     656        4039 :         if (ir_filter_nr > ctx->prev_ir_filter_nr + 1)
     657          83 :             ir_filter_nr--;
     658             :     }
     659             : 
     660             :     /* update ir filter strength history */
     661        4096 :     ctx->prev_ir_filter_nr = ir_filter_nr;
     662             : 
     663        4096 :     ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85);
     664             : 
     665        4096 :     if (ir_filter_nr < 2) {
     666             :         int i;
     667        3191 :         const float *coef = ir_filters_lookup[ir_filter_nr];
     668             : 
     669             :         /* Circular convolution code in the reference
     670             :          * decoder was modified to avoid using one
     671             :          * extra array. The filtered vector is given by:
     672             :          *
     673             :          * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }
     674             :          */
     675             : 
     676        3191 :         memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE);
     677      207415 :         for (i = 0; i < AMRWB_SFR_SIZE; i++)
     678      204224 :             if (fixed_vector[i])
     679       21425 :                 ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],
     680             :                                   AMRWB_SFR_SIZE);
     681        3191 :         fixed_vector = buf;
     682             :     }
     683             : 
     684        4096 :     return fixed_vector;
     685             : }
     686             : 
     687             : /**
     688             :  * Calculate a stability factor {teta} based on distance between
     689             :  * current and past isf. A value of 1 shows maximum signal stability.
     690             :  */
     691        6768 : static float stability_factor(const float *isf, const float *isf_past)
     692             : {
     693             :     int i;
     694        6768 :     float acc = 0.0;
     695             : 
     696      108288 :     for (i = 0; i < LP_ORDER - 1; i++)
     697      101520 :         acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);
     698             : 
     699             :     // XXX: This part is not so clear from the reference code
     700             :     // the result is more accurate changing the "/ 256" to "* 512"
     701        6768 :     return FFMAX(0.0, 1.25 - acc * 0.8 * 512);
     702             : }
     703             : 
     704             : /**
     705             :  * Apply a non-linear fixed gain smoothing in order to reduce
     706             :  * fluctuation in the energy of excitation.
     707             :  *
     708             :  * @param[in]     fixed_gain       Unsmoothed fixed gain
     709             :  * @param[in,out] prev_tr_gain     Previous threshold gain (updated)
     710             :  * @param[in]     voice_fac        Frame voicing factor
     711             :  * @param[in]     stab_fac         Frame stability factor
     712             :  *
     713             :  * @return The smoothed gain
     714             :  */
     715       27072 : static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
     716             :                             float voice_fac,  float stab_fac)
     717             : {
     718       27072 :     float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;
     719             :     float g0;
     720             : 
     721             :     // XXX: the following fixed-point constants used to in(de)crement
     722             :     // gain by 1.5dB were taken from the reference code, maybe it could
     723             :     // be simpler
     724       27072 :     if (fixed_gain < *prev_tr_gain) {
     725       14148 :         g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *
     726             :                      (6226 * (1.0f / (1 << 15)))); // +1.5 dB
     727             :     } else
     728       12924 :         g0 = FFMAX(*prev_tr_gain, fixed_gain *
     729             :                     (27536 * (1.0f / (1 << 15)))); // -1.5 dB
     730             : 
     731       27072 :     *prev_tr_gain = g0; // update next frame threshold
     732             : 
     733       27072 :     return sm_fac * g0 + (1 - sm_fac) * fixed_gain;
     734             : }
     735             : 
     736             : /**
     737             :  * Filter the fixed_vector to emphasize the higher frequencies.
     738             :  *
     739             :  * @param[in,out] fixed_vector     Fixed codebook vector
     740             :  * @param[in]     voice_fac        Frame voicing factor
     741             :  */
     742       27072 : static void pitch_enhancer(float *fixed_vector, float voice_fac)
     743             : {
     744             :     int i;
     745       27072 :     float cpe  = 0.125 * (1 + voice_fac);
     746       27072 :     float last = fixed_vector[0]; // holds c(i - 1)
     747             : 
     748       27072 :     fixed_vector[0] -= cpe * fixed_vector[1];
     749             : 
     750     1705536 :     for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) {
     751     1678464 :         float cur = fixed_vector[i];
     752             : 
     753     1678464 :         fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]);
     754     1678464 :         last = cur;
     755             :     }
     756             : 
     757       27072 :     fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last;
     758       27072 : }
     759             : 
     760             : /**
     761             :  * Conduct 16th order linear predictive coding synthesis from excitation.
     762             :  *
     763             :  * @param[in]     ctx              Pointer to the AMRWBContext
     764             :  * @param[in]     lpc              Pointer to the LPC coefficients
     765             :  * @param[out]    excitation       Buffer for synthesis final excitation
     766             :  * @param[in]     fixed_gain       Fixed codebook gain for synthesis
     767             :  * @param[in]     fixed_vector     Algebraic codebook vector
     768             :  * @param[in,out] samples          Pointer to the output samples and memory
     769             :  */
     770       27072 : static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
     771             :                       float fixed_gain, const float *fixed_vector,
     772             :                       float *samples)
     773             : {
     774       27072 :     ctx->acelpv_ctx.weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,
     775             :                             ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);
     776             : 
     777             :     /* emphasize pitch vector contribution in low bitrate modes */
     778       27072 :     if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
     779             :         int i;
     780        1895 :         float energy = ctx->celpm_ctx.dot_productf(excitation, excitation,
     781             :                                                     AMRWB_SFR_SIZE);
     782             : 
     783             :         // XXX: Weird part in both ref code and spec. A unknown parameter
     784             :         // {beta} seems to be identical to the current pitch gain
     785        1895 :         float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0];
     786             : 
     787      123175 :         for (i = 0; i < AMRWB_SFR_SIZE; i++)
     788      121280 :             excitation[i] += pitch_factor * ctx->pitch_vector[i];
     789             : 
     790        1895 :         ff_scale_vector_to_given_sum_of_squares(excitation, excitation,
     791             :                                                 energy, AMRWB_SFR_SIZE);
     792             :     }
     793             : 
     794       27072 :     ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
     795             :                                  AMRWB_SFR_SIZE, LP_ORDER);
     796       27072 : }
     797             : 
     798             : /**
     799             :  * Apply to synthesis a de-emphasis filter of the form:
     800             :  * H(z) = 1 / (1 - m * z^-1)
     801             :  *
     802             :  * @param[out]    out              Output buffer
     803             :  * @param[in]     in               Input samples array with in[-1]
     804             :  * @param[in]     m                Filter coefficient
     805             :  * @param[in,out] mem              State from last filtering
     806             :  */
     807       27072 : static void de_emphasis(float *out, float *in, float m, float mem[1])
     808             : {
     809             :     int i;
     810             : 
     811       27072 :     out[0] = in[0] + m * mem[0];
     812             : 
     813     1732608 :     for (i = 1; i < AMRWB_SFR_SIZE; i++)
     814     1705536 :          out[i] = in[i] + out[i - 1] * m;
     815             : 
     816       27072 :     mem[0] = out[AMRWB_SFR_SIZE - 1];
     817       27072 : }
     818             : 
     819             : /**
     820             :  * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
     821             :  * a FIR interpolation filter. Uses past data from before *in address.
     822             :  *
     823             :  * @param[out] out                 Buffer for interpolated signal
     824             :  * @param[in]  in                  Current signal data (length 0.8*o_size)
     825             :  * @param[in]  o_size              Output signal length
     826             :  * @param[in] ctx                  The context
     827             :  */
     828       27072 : static void upsample_5_4(float *out, const float *in, int o_size, CELPMContext *ctx)
     829             : {
     830       27072 :     const float *in0 = in - UPS_FIR_SIZE + 1;
     831             :     int i, j, k;
     832       27072 :     int int_part = 0, frac_part;
     833             : 
     834       27072 :     i = 0;
     835      460224 :     for (j = 0; j < o_size / 5; j++) {
     836      433152 :         out[i] = in[int_part];
     837      433152 :         frac_part = 4;
     838      433152 :         i++;
     839             : 
     840     2165760 :         for (k = 1; k < 5; k++) {
     841     3465216 :             out[i] = ctx->dot_productf(in0 + int_part,
     842     1732608 :                                                   upsample_fir[4 - frac_part],
     843             :                                                   UPS_MEM_SIZE);
     844     1732608 :             int_part++;
     845     1732608 :             frac_part--;
     846     1732608 :             i++;
     847             :         }
     848             :     }
     849       27072 : }
     850             : 
     851             : /**
     852             :  * Calculate the high-band gain based on encoded index (23k85 mode) or
     853             :  * on the low-band speech signal and the Voice Activity Detection flag.
     854             :  *
     855             :  * @param[in] ctx                  The context
     856             :  * @param[in] synth                LB speech synthesis at 12.8k
     857             :  * @param[in] hb_idx               Gain index for mode 23k85 only
     858             :  * @param[in] vad                  VAD flag for the frame
     859             :  */
     860       27072 : static float find_hb_gain(AMRWBContext *ctx, const float *synth,
     861             :                           uint16_t hb_idx, uint8_t vad)
     862             : {
     863       27072 :     int wsp = (vad > 0);
     864             :     float tilt;
     865             : 
     866       27072 :     if (ctx->fr_cur_mode == MODE_23k85)
     867        8640 :         return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
     868             : 
     869       36864 :     tilt = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
     870       18432 :            ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);
     871             : 
     872             :     /* return gain bounded by [0.1, 1.0] */
     873       18432 :     return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0);
     874             : }
     875             : 
     876             : /**
     877             :  * Generate the high-band excitation with the same energy from the lower
     878             :  * one and scaled by the given gain.
     879             :  *
     880             :  * @param[in]  ctx                 The context
     881             :  * @param[out] hb_exc              Buffer for the excitation
     882             :  * @param[in]  synth_exc           Low-band excitation used for synthesis
     883             :  * @param[in]  hb_gain             Wanted excitation gain
     884             :  */
     885       27072 : static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
     886             :                                  const float *synth_exc, float hb_gain)
     887             : {
     888             :     int i;
     889       27072 :     float energy = ctx->celpm_ctx.dot_productf(synth_exc, synth_exc,
     890             :                                                 AMRWB_SFR_SIZE);
     891             : 
     892             :     /* Generate a white-noise excitation */
     893     2192832 :     for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
     894     2165760 :         hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);
     895             : 
     896       27072 :     ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc,
     897       27072 :                                             energy * hb_gain * hb_gain,
     898             :                                             AMRWB_SFR_SIZE_16k);
     899       27072 : }
     900             : 
     901             : /**
     902             :  * Calculate the auto-correlation for the ISF difference vector.
     903             :  */
     904        6144 : static float auto_correlation(float *diff_isf, float mean, int lag)
     905             : {
     906             :     int i;
     907        6144 :     float sum = 0.0;
     908             : 
     909       49152 :     for (i = 7; i < LP_ORDER - 2; i++) {
     910       43008 :         float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean);
     911       43008 :         sum += prod * prod;
     912             :     }
     913        6144 :     return sum;
     914             : }
     915             : 
     916             : /**
     917             :  * Extrapolate a ISF vector to the 16kHz range (20th order LP)
     918             :  * used at mode 6k60 LP filter for the high frequency band.
     919             :  *
     920             :  * @param[out] isf Buffer for extrapolated isf; contains LP_ORDER
     921             :  *                 values on input
     922             :  */
     923        2048 : static void extrapolate_isf(float isf[LP_ORDER_16k])
     924             : {
     925             :     float diff_isf[LP_ORDER - 2], diff_mean;
     926             :     float corr_lag[3];
     927             :     float est, scale;
     928             :     int i, j, i_max_corr;
     929             : 
     930        2048 :     isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
     931             : 
     932             :     /* Calculate the difference vector */
     933       30720 :     for (i = 0; i < LP_ORDER - 2; i++)
     934       28672 :         diff_isf[i] = isf[i + 1] - isf[i];
     935             : 
     936        2048 :     diff_mean = 0.0;
     937       26624 :     for (i = 2; i < LP_ORDER - 2; i++)
     938       24576 :         diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4));
     939             : 
     940             :     /* Find which is the maximum autocorrelation */
     941        2048 :     i_max_corr = 0;
     942        8192 :     for (i = 0; i < 3; i++) {
     943        6144 :         corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);
     944             : 
     945        6144 :         if (corr_lag[i] > corr_lag[i_max_corr])
     946        2154 :             i_max_corr = i;
     947             :     }
     948        2048 :     i_max_corr++;
     949             : 
     950       10240 :     for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
     951       16384 :         isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
     952        8192 :                             - isf[i - 2 - i_max_corr];
     953             : 
     954             :     /* Calculate an estimate for ISF(18) and scale ISF based on the error */
     955        2048 :     est   = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0;
     956        4096 :     scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) /
     957        2048 :             (isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]);
     958             : 
     959       10240 :     for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
     960        8192 :         diff_isf[j] = scale * (isf[i] - isf[i - 1]);
     961             : 
     962             :     /* Stability insurance */
     963        8192 :     for (i = 1; i < LP_ORDER_16k - LP_ORDER; i++)
     964        6144 :         if (diff_isf[i] + diff_isf[i - 1] < 5.0) {
     965           0 :             if (diff_isf[i] > diff_isf[i - 1]) {
     966           0 :                 diff_isf[i - 1] = 5.0 - diff_isf[i];
     967             :             } else
     968           0 :                 diff_isf[i] = 5.0 - diff_isf[i - 1];
     969             :         }
     970             : 
     971       10240 :     for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
     972        8192 :         isf[i] = isf[i - 1] + diff_isf[j] * (1.0f / (1 << 15));
     973             : 
     974             :     /* Scale the ISF vector for 16000 Hz */
     975       40960 :     for (i = 0; i < LP_ORDER_16k - 1; i++)
     976       38912 :         isf[i] *= 0.8;
     977        2048 : }
     978             : 
     979             : /**
     980             :  * Spectral expand the LP coefficients using the equation:
     981             :  *   y[i] = x[i] * (gamma ** i)
     982             :  *
     983             :  * @param[out] out                 Output buffer (may use input array)
     984             :  * @param[in]  lpc                 LP coefficients array
     985             :  * @param[in]  gamma               Weighting factor
     986             :  * @param[in]  size                LP array size
     987             :  */
     988       27072 : static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
     989             : {
     990             :     int i;
     991       27072 :     float fac = gamma;
     992             : 
     993      468416 :     for (i = 0; i < size; i++) {
     994      441344 :         out[i] = lpc[i] * fac;
     995      441344 :         fac   *= gamma;
     996             :     }
     997       27072 : }
     998             : 
     999             : /**
    1000             :  * Conduct 20th order linear predictive coding synthesis for the high
    1001             :  * frequency band excitation at 16kHz.
    1002             :  *
    1003             :  * @param[in]     ctx              The context
    1004             :  * @param[in]     subframe         Current subframe index (0 to 3)
    1005             :  * @param[in,out] samples          Pointer to the output speech samples
    1006             :  * @param[in]     exc              Generated white-noise scaled excitation
    1007             :  * @param[in]     isf              Current frame isf vector
    1008             :  * @param[in]     isf_past         Past frame final isf vector
    1009             :  */
    1010       27072 : static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
    1011             :                          const float *exc, const float *isf, const float *isf_past)
    1012             : {
    1013             :     float hb_lpc[LP_ORDER_16k];
    1014       27072 :     enum Mode mode = ctx->fr_cur_mode;
    1015             : 
    1016       27072 :     if (mode == MODE_6k60) {
    1017             :         float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation
    1018             :         double e_isp[LP_ORDER_16k];
    1019             : 
    1020        4096 :         ctx->acelpv_ctx.weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],
    1021        2048 :                                 1.0 - isfp_inter[subframe], LP_ORDER);
    1022             : 
    1023        2048 :         extrapolate_isf(e_isf);
    1024             : 
    1025        2048 :         e_isf[LP_ORDER_16k - 1] *= 2.0;
    1026        2048 :         ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);
    1027        2048 :         ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k);
    1028             : 
    1029        2048 :         lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k);
    1030             :     } else {
    1031       25024 :         lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER);
    1032             :     }
    1033             : 
    1034       27072 :     ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,
    1035             :                                  (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);
    1036       27072 : }
    1037             : 
    1038             : /**
    1039             :  * Apply a 15th order filter to high-band samples.
    1040             :  * The filter characteristic depends on the given coefficients.
    1041             :  *
    1042             :  * @param[out]    out              Buffer for filtered output
    1043             :  * @param[in]     fir_coef         Filter coefficients
    1044             :  * @param[in,out] mem              State from last filtering (updated)
    1045             :  * @param[in]     in               Input speech data (high-band)
    1046             :  *
    1047             :  * @remark It is safe to pass the same array in in and out parameters
    1048             :  */
    1049             : 
    1050             : #ifndef hb_fir_filter
    1051       35712 : static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
    1052             :                           float mem[HB_FIR_SIZE], const float *in)
    1053             : {
    1054             :     int i, j;
    1055             :     float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples
    1056             : 
    1057       35712 :     memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
    1058       35712 :     memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float));
    1059             : 
    1060     2892672 :     for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {
    1061     2856960 :         out[i] = 0.0;
    1062    91422720 :         for (j = 0; j <= HB_FIR_SIZE; j++)
    1063    88565760 :             out[i] += data[i + j] * fir_coef[j];
    1064             :     }
    1065             : 
    1066       35712 :     memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));
    1067       35712 : }
    1068             : #endif /* hb_fir_filter */
    1069             : 
    1070             : /**
    1071             :  * Update context state before the next subframe.
    1072             :  */
    1073       27072 : static void update_sub_state(AMRWBContext *ctx)
    1074             : {
    1075       27072 :     memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE],
    1076             :             (AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float));
    1077             : 
    1078       27072 :     memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float));
    1079       27072 :     memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float));
    1080             : 
    1081       27072 :     memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE],
    1082             :             LP_ORDER * sizeof(float));
    1083       27072 :     memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE],
    1084             :             UPS_MEM_SIZE * sizeof(float));
    1085       27072 :     memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k],
    1086             :             LP_ORDER_16k * sizeof(float));
    1087       27072 : }
    1088             : 
    1089        6768 : static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
    1090             :                               int *got_frame_ptr, AVPacket *avpkt)
    1091             : {
    1092        6768 :     AMRWBContext *ctx  = avctx->priv_data;
    1093        6768 :     AVFrame *frame     = data;
    1094        6768 :     AMRWBFrame   *cf   = &ctx->frame;
    1095        6768 :     const uint8_t *buf = avpkt->data;
    1096        6768 :     int buf_size       = avpkt->size;
    1097             :     int expected_fr_size, header_size;
    1098             :     float *buf_out;
    1099             :     float spare_vector[AMRWB_SFR_SIZE];      // extra stack space to hold result from anti-sparseness processing
    1100             :     float fixed_gain_factor;                 // fixed gain correction factor (gamma)
    1101             :     float *synth_fixed_vector;               // pointer to the fixed vector that synthesis should use
    1102             :     float synth_fixed_gain;                  // the fixed gain that synthesis should use
    1103             :     float voice_fac, stab_fac;               // parameters used for gain smoothing
    1104             :     float synth_exc[AMRWB_SFR_SIZE];         // post-processed excitation for synthesis
    1105             :     float hb_exc[AMRWB_SFR_SIZE_16k];        // excitation for the high frequency band
    1106             :     float hb_samples[AMRWB_SFR_SIZE_16k];    // filtered high-band samples from synthesis
    1107             :     float hb_gain;
    1108             :     int sub, i, ret;
    1109             : 
    1110             :     /* get output buffer */
    1111        6768 :     frame->nb_samples = 4 * AMRWB_SFR_SIZE_16k;
    1112        6768 :     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
    1113           0 :         return ret;
    1114        6768 :     buf_out = (float *)frame->data[0];
    1115             : 
    1116        6768 :     header_size      = decode_mime_header(ctx, buf);
    1117        6768 :     if (ctx->fr_cur_mode > MODE_SID) {
    1118           0 :         av_log(avctx, AV_LOG_ERROR,
    1119           0 :                "Invalid mode %d\n", ctx->fr_cur_mode);
    1120           0 :         return AVERROR_INVALIDDATA;
    1121             :     }
    1122        6768 :     expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
    1123             : 
    1124        6768 :     if (buf_size < expected_fr_size) {
    1125           0 :         av_log(avctx, AV_LOG_ERROR,
    1126             :             "Frame too small (%d bytes). Truncated file?\n", buf_size);
    1127           0 :         *got_frame_ptr = 0;
    1128           0 :         return AVERROR_INVALIDDATA;
    1129             :     }
    1130             : 
    1131        6768 :     if (!ctx->fr_quality || ctx->fr_cur_mode > MODE_SID)
    1132           0 :         av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");
    1133             : 
    1134        6768 :     if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */
    1135           0 :         avpriv_request_sample(avctx, "SID mode");
    1136           0 :         return AVERROR_PATCHWELCOME;
    1137             :     }
    1138             : 
    1139        6768 :     ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
    1140        6768 :         buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);
    1141             : 
    1142             :     /* Decode the quantized ISF vector */
    1143        6768 :     if (ctx->fr_cur_mode == MODE_6k60) {
    1144         512 :         decode_isf_indices_36b(cf->isp_id, ctx->isf_cur);
    1145             :     } else {
    1146        6256 :         decode_isf_indices_46b(cf->isp_id, ctx->isf_cur);
    1147             :     }
    1148             : 
    1149        6768 :     isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past);
    1150        6768 :     ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1);
    1151             : 
    1152        6768 :     stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);
    1153             : 
    1154        6768 :     ctx->isf_cur[LP_ORDER - 1] *= 2.0;
    1155        6768 :     ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER);
    1156             : 
    1157             :     /* Generate a ISP vector for each subframe */
    1158        6768 :     if (ctx->first_frame) {
    1159          11 :         ctx->first_frame = 0;
    1160          11 :         memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double));
    1161             :     }
    1162        6768 :     interpolate_isp(ctx->isp, ctx->isp_sub4_past);
    1163             : 
    1164       33840 :     for (sub = 0; sub < 4; sub++)
    1165       27072 :         ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER);
    1166             : 
    1167       33840 :     for (sub = 0; sub < 4; sub++) {
    1168       27072 :         const AMRWBSubFrame *cur_subframe = &cf->subframe[sub];
    1169       27072 :         float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;
    1170             : 
    1171             :         /* Decode adaptive codebook (pitch vector) */
    1172       27072 :         decode_pitch_vector(ctx, cur_subframe, sub);
    1173             :         /* Decode innovative codebook (fixed vector) */
    1174       54144 :         decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih,
    1175       27072 :                             cur_subframe->pul_il, ctx->fr_cur_mode);
    1176             : 
    1177       27072 :         pitch_sharpening(ctx, ctx->fixed_vector);
    1178             : 
    1179       27072 :         decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode,
    1180             :                      &fixed_gain_factor, &ctx->pitch_gain[0]);
    1181             : 
    1182       27072 :         ctx->fixed_gain[0] =
    1183       27072 :             ff_amr_set_fixed_gain(fixed_gain_factor,
    1184       54144 :                                   ctx->celpm_ctx.dot_productf(ctx->fixed_vector,
    1185       27072 :                                                                ctx->fixed_vector,
    1186             :                                                                AMRWB_SFR_SIZE) /
    1187             :                                   AMRWB_SFR_SIZE,
    1188       27072 :                        ctx->prediction_error,
    1189             :                        ENERGY_MEAN, energy_pred_fac);
    1190             : 
    1191             :         /* Calculate voice factor and store tilt for next subframe */
    1192       54144 :         voice_fac      = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0],
    1193       27072 :                                       ctx->fixed_vector, ctx->fixed_gain[0],
    1194             :                                       &ctx->celpm_ctx);
    1195       27072 :         ctx->tilt_coef = voice_fac * 0.25 + 0.25;
    1196             : 
    1197             :         /* Construct current excitation */
    1198     1759680 :         for (i = 0; i < AMRWB_SFR_SIZE; i++) {
    1199     1732608 :             ctx->excitation[i] *= ctx->pitch_gain[0];
    1200     1732608 :             ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i];
    1201     1732608 :             ctx->excitation[i] = truncf(ctx->excitation[i]);
    1202             :         }
    1203             : 
    1204             :         /* Post-processing of excitation elements */
    1205       27072 :         synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain,
    1206             :                                           voice_fac, stab_fac);
    1207             : 
    1208       27072 :         synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector,
    1209             :                                              spare_vector);
    1210             : 
    1211       27072 :         pitch_enhancer(synth_fixed_vector, voice_fac);
    1212             : 
    1213       27072 :         synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
    1214             :                   synth_fixed_vector, &ctx->samples_az[LP_ORDER]);
    1215             : 
    1216             :         /* Synthesis speech post-processing */
    1217       27072 :         de_emphasis(&ctx->samples_up[UPS_MEM_SIZE],
    1218       27072 :                     &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);
    1219             : 
    1220       81216 :         ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE],
    1221       27072 :             &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles,
    1222       27072 :             hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE);
    1223             : 
    1224       27072 :         upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],
    1225             :                      AMRWB_SFR_SIZE_16k, &ctx->celpm_ctx);
    1226             : 
    1227             :         /* High frequency band (6.4 - 7.0 kHz) generation part */
    1228       81216 :         ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(hb_samples,
    1229       27072 :             &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles,
    1230       27072 :             hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE);
    1231             : 
    1232       54144 :         hb_gain = find_hb_gain(ctx, hb_samples,
    1233       54144 :                                cur_subframe->hb_gain, cf->vad);
    1234             : 
    1235       27072 :         scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);
    1236             : 
    1237       27072 :         hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k],
    1238       27072 :                      hb_exc, ctx->isf_cur, ctx->isf_past_final);
    1239             : 
    1240             :         /* High-band post-processing filters */
    1241       27072 :         hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,
    1242       27072 :                       &ctx->samples_hb[LP_ORDER_16k]);
    1243             : 
    1244       27072 :         if (ctx->fr_cur_mode == MODE_23k85)
    1245        8640 :             hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,
    1246             :                           hb_samples);
    1247             : 
    1248             :         /* Add the low and high frequency bands */
    1249     2192832 :         for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
    1250     2165760 :             sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15));
    1251             : 
    1252             :         /* Update buffers and history */
    1253       27072 :         update_sub_state(ctx);
    1254             :     }
    1255             : 
    1256             :     /* update state for next frame */
    1257        6768 :     memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
    1258        6768 :     memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
    1259             : 
    1260        6768 :     *got_frame_ptr = 1;
    1261             : 
    1262        6768 :     return expected_fr_size;
    1263             : }
    1264             : 
    1265             : AVCodec ff_amrwb_decoder = {
    1266             :     .name           = "amrwb",
    1267             :     .long_name      = NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"),
    1268             :     .type           = AVMEDIA_TYPE_AUDIO,
    1269             :     .id             = AV_CODEC_ID_AMR_WB,
    1270             :     .priv_data_size = sizeof(AMRWBContext),
    1271             :     .init           = amrwb_decode_init,
    1272             :     .decode         = amrwb_decode_frame,
    1273             :     .capabilities   = AV_CODEC_CAP_DR1,
    1274             :     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
    1275             :                                                      AV_SAMPLE_FMT_NONE },
    1276             : };

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