/*
  * AMR wideband decoder
  * Copyright (c) 2010 Marcelo Galvao Povoa
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
  * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A particular PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 /**
  * @file
  * AMR wideband decoder
  */
 

 #include "libavutil/channel_layout.h"

 #include "libavutil/common.h"

 #include "libavutil/lfg.h"
 
 #include "avcodec.h"

 #include "dsputil.h"

 #include "lsp.h"
 #include "celp_filters.h"

 #include "celp_math.h"

 #include "acelp_filters.h"
 #include "acelp_vectors.h"
 #include "acelp_pitch_delay.h"

 #include "internal.h"

 
 #define AMR_USE_16BIT_TABLES
 #include "amr.h"
 
 #include "amrwbdata.h"

 #include "mips/amrwbdec_mips.h"

 
 typedef struct {

     AVFrame                              avframe; ///< AVFrame for decoded samples

     AMRWBFrame                             frame; ///< AMRWB parameters decoded from bitstream
     enum Mode                        fr_cur_mode; ///< mode index of current frame
     uint8_t                           fr_quality; ///< frame quality index (FQI)
     float                      isf_cur[LP_ORDER]; ///< working ISF vector from current frame
     float                   isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame
     float               isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame
     double                      isp[4][LP_ORDER]; ///< ISP vectors from current frame
     double               isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame
 
     float                   lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector
 
     uint8_t                       base_pitch_lag; ///< integer part of pitch lag for the next relative subframe
     uint8_t                        pitch_lag_int; ///< integer part of pitch lag of the previous subframe
 
     float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history
     float                            *excitation; ///< points to current excitation in excitation_buf[]
 
     float           pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe
     float           fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe
 
     float                    prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
     float                          pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes
     float                          fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes
 
     float                              tilt_coef; ///< {beta_1} related to the voicing of the previous subframe
 
     float                 prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness to determine "onset"
     uint8_t                    prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
     float                           prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold
 
     float samples_az[LP_ORDER + AMRWB_SFR_SIZE];         ///< low-band samples and memory from synthesis at 12.8kHz
     float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE];     ///< low-band samples and memory processed for upsampling
     float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band samples and memory from synthesis at 16kHz
 
     float          hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters
     float                           demph_mem[1]; ///< previous value in the de-emphasis filter
     float               bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter
     float                 lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter
 
     AVLFG                                   prng; ///< random number generator for white noise excitation
     uint8_t                          first_frame; ///< flag active during decoding of the first frame

     ACELPFContext                     acelpf_ctx; ///< context for filters for ACELP-based codecs
     ACELPVContext                     acelpv_ctx; ///< context for vector operations for ACELP-based codecs
     CELPFContext                       celpf_ctx; ///< context for filters for CELP-based codecs
     CELPMContext                       celpm_ctx; ///< context for fixed point math operations
 

 } AMRWBContext;
 
 static av_cold int amrwb_decode_init(AVCodecContext *avctx)
 {
     AMRWBContext *ctx = avctx->priv_data;
     int i;
 

     if (avctx->channels > 1) {
         av_log_missing_feature(avctx, "multi-channel AMR", 0);
         return AVERROR_PATCHWELCOME;
     }
 
     avctx->channels       = 1;
     avctx->channel_layout = AV_CH_LAYOUT_MONO;

     if (!avctx->sample_rate)
         avctx->sample_rate = 16000;

     avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;

 
     av_lfg_init(&ctx->prng, 1);
 
     ctx->excitation  = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1];
     ctx->first_frame = 1;
 
     for (i = 0; i < LP_ORDER; i++)
         ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15));
 
     for (i = 0; i < 4; i++)
         ctx->prediction_error[i] = MIN_ENERGY;
 

     avcodec_get_frame_defaults(&ctx->avframe);
     avctx->coded_frame = &ctx->avframe;
 

     ff_acelp_filter_init(&ctx->acelpf_ctx);
     ff_acelp_vectors_init(&ctx->acelpv_ctx);
     ff_celp_filter_init(&ctx->celpf_ctx);
     ff_celp_math_init(&ctx->celpm_ctx);
 

     return 0;
 }
 
 /**
  * Decode the frame header in the "MIME/storage" format. This format

  * is simpler and does not carry the auxiliary frame information.

  *
  * @param[in] ctx                  The Context
  * @param[in] buf                  Pointer to the input buffer
  *
  * @return The decoded header length in bytes
  */
 static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
 {
     /* Decode frame header (1st octet) */

     ctx->fr_cur_mode  = buf[0] >> 3 & 0x0F;

     ctx->fr_quality   = (buf[0] & 0x4) == 0x4;

 
     return 1;
 }
 
 /**

  * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).

  *
  * @param[in]  ind                 Array of 5 indexes
  * @param[out] isf_q               Buffer for isf_q[LP_ORDER]
  *
  */
 static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
 {
     int i;
 
     for (i = 0; i < 9; i++)
         isf_q[i]      = dico1_isf[ind[0]][i]      * (1.0f / (1 << 15));
 
     for (i = 0; i < 7; i++)
         isf_q[i + 9]  = dico2_isf[ind[1]][i]      * (1.0f / (1 << 15));
 
     for (i = 0; i < 5; i++)
         isf_q[i]     += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15));
 
     for (i = 0; i < 4; i++)
         isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15));
 
     for (i = 0; i < 7; i++)
         isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15));
 }
 
 /**

  * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).

  *
  * @param[in]  ind                 Array of 7 indexes
  * @param[out] isf_q               Buffer for isf_q[LP_ORDER]
  *
  */
 static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
 {
     int i;
 
     for (i = 0; i < 9; i++)
         isf_q[i]       = dico1_isf[ind[0]][i]  * (1.0f / (1 << 15));
 
     for (i = 0; i < 7; i++)
         isf_q[i + 9]   = dico2_isf[ind[1]][i]  * (1.0f / (1 << 15));
 
     for (i = 0; i < 3; i++)
         isf_q[i]      += dico21_isf[ind[2]][i] * (1.0f / (1 << 15));
 
     for (i = 0; i < 3; i++)
         isf_q[i + 3]  += dico22_isf[ind[3]][i] * (1.0f / (1 << 15));
 
     for (i = 0; i < 3; i++)
         isf_q[i + 6]  += dico23_isf[ind[4]][i] * (1.0f / (1 << 15));
 
     for (i = 0; i < 3; i++)
         isf_q[i + 9]  += dico24_isf[ind[5]][i] * (1.0f / (1 << 15));
 
     for (i = 0; i < 4; i++)
         isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15));
 }
 
 /**

  * Apply mean and past ISF values using the prediction factor.
  * Updates past ISF vector.

  *
  * @param[in,out] isf_q            Current quantized ISF
  * @param[in,out] isf_past         Past quantized ISF
  *
  */
 static void isf_add_mean_and_past(float *isf_q, float *isf_past)
 {
     int i;
     float tmp;
 
     for (i = 0; i < LP_ORDER; i++) {
         tmp = isf_q[i];
         isf_q[i] += isf_mean[i] * (1.0f / (1 << 15));
         isf_q[i] += PRED_FACTOR * isf_past[i];
         isf_past[i] = tmp;
     }
 }
 
 /**
  * Interpolate the fourth ISP vector from current and past frames

  * to obtain an ISP vector for each subframe.

  *
  * @param[in,out] isp_q            ISPs for each subframe
  * @param[in]     isp4_past        Past ISP for subframe 4
  */
 static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
 {
     int i, k;
 
     for (k = 0; k < 3; k++) {
         float c = isfp_inter[k];
         for (i = 0; i < LP_ORDER; i++)
             isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i];
     }
 }
 
 /**

  * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
  * Calculate integer lag and fractional lag always using 1/4 resolution.
  * In 1st and 3rd subframes the index is relative to last subframe integer lag.

  *
  * @param[out]    lag_int          Decoded integer pitch lag
  * @param[out]    lag_frac         Decoded fractional pitch lag
  * @param[in]     pitch_index      Adaptive codebook pitch index
  * @param[in,out] base_lag_int     Base integer lag used in relative subframes
  * @param[in]     subframe         Current subframe index (0 to 3)
  */
 static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
                                   uint8_t *base_lag_int, int subframe)
 {
     if (subframe == 0 || subframe == 2) {
         if (pitch_index < 376) {
             *lag_int  = (pitch_index + 137) >> 2;
             *lag_frac = pitch_index - (*lag_int << 2) + 136;
         } else if (pitch_index < 440) {
             *lag_int  = (pitch_index + 257 - 376) >> 1;
             *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) << 1;
             /* the actual resolution is 1/2 but expressed as 1/4 */
         } else {
             *lag_int  = pitch_index - 280;
             *lag_frac = 0;
         }
         /* minimum lag for next subframe */
         *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
                                 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
         // XXX: the spec states clearly that *base_lag_int should be
         // the nearest integer to *lag_int (minus 8), but the ref code
         // actually always uses its floor, I'm following the latter
     } else {
         *lag_int  = (pitch_index + 1) >> 2;
         *lag_frac = pitch_index - (*lag_int << 2);
         *lag_int += *base_lag_int;
     }
 }
 
 /**

  * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
  * The description is analogous to decode_pitch_lag_high, but in 6k60 the
  * relative index is used for all subframes except the first.

  */
 static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
                                  uint8_t *base_lag_int, int subframe, enum Mode mode)
 {
     if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {
         if (pitch_index < 116) {
             *lag_int  = (pitch_index + 69) >> 1;
             *lag_frac = (pitch_index - (*lag_int << 1) + 68) << 1;
         } else {
             *lag_int  = pitch_index - 24;
             *lag_frac = 0;
         }
         // XXX: same problem as before
         *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
                                 AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
     } else {
         *lag_int  = (pitch_index + 1) >> 1;
         *lag_frac = (pitch_index - (*lag_int << 1)) << 1;
         *lag_int += *base_lag_int;
     }
 }
 
 /**
  * Find the pitch vector by interpolating the past excitation at the

  * pitch delay, which is obtained in this function.

  *
  * @param[in,out] ctx              The context
  * @param[in]     amr_subframe     Current subframe data
  * @param[in]     subframe         Current subframe index (0 to 3)
  */
 static void decode_pitch_vector(AMRWBContext *ctx,
                                 const AMRWBSubFrame *amr_subframe,
                                 const int subframe)
 {
     int pitch_lag_int, pitch_lag_frac;
     int i;
     float *exc     = ctx->excitation;
     enum Mode mode = ctx->fr_cur_mode;
 
     if (mode <= MODE_8k85) {
         decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
                               &ctx->base_pitch_lag, subframe, mode);
     } else
         decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
                               &ctx->base_pitch_lag, subframe);
 
     ctx->pitch_lag_int = pitch_lag_int;
     pitch_lag_int += pitch_lag_frac > 0;
 
     /* Calculate the pitch vector by interpolating the past excitation at the
        pitch lag using a hamming windowed sinc function */

     ctx->acelpf_ctx.acelp_interpolatef(exc,
                           exc + 1 - pitch_lag_int,

                           ac_inter, 4,
                           pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
                           LP_ORDER, AMRWB_SFR_SIZE + 1);
 
     /* Check which pitch signal path should be used
      * 6k60 and 8k85 modes have the ltp flag set to 0 */
     if (amr_subframe->ltp) {
         memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float));
     } else {
         for (i = 0; i < AMRWB_SFR_SIZE; i++)
             ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] +
                                    0.18 * exc[i + 1];
         memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float));
     }
 }
 
 /** Get x bits in the index interval [lsb,lsb+len-1] inclusive */
 #define BIT_STR(x,lsb,len) (((x) >> (lsb)) & ((1 << (len)) - 1))
 
 /** Get the bit at specified position */
 #define BIT_POS(x, p) (((x) >> (p)) & 1)
 
 /**
  * The next six functions decode_[i]p_track decode exactly i pulses
  * positions and amplitudes (-1 or 1) in a subframe track using

  * an encoded pulse indexing (TS 26.190 section 5.8.2).

  *
  * The results are given in out[], in which a negative number means

  * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).

  *
  * @param[out] out                 Output buffer (writes i elements)
  * @param[in]  code                Pulse index (no. of bits varies, see below)
  * @param[in]  m                   (log2) Number of potential positions
  * @param[in]  off                 Offset for decoded positions
  */
 static inline void decode_1p_track(int *out, int code, int m, int off)
 {
     int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits
 
     out[0] = BIT_POS(code, m) ? -pos : pos;
 }
 
 static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits
 {
     int pos0 = BIT_STR(code, m, m) + off;
     int pos1 = BIT_STR(code, 0, m) + off;
 
     out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0;
     out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1;
     out[1] = pos0 > pos1 ? -out[1] : out[1];
 }
 
 static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits
 {
     int half_2p = BIT_POS(code, 2*m - 1) << (m - 1);
 
     decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
                     m - 1, off + half_2p);
     decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off);
 }
 
 static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits
 {
     int half_4p, subhalf_2p;
     int b_offset = 1 << (m - 1);
 
     switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */
     case 0: /* 0 pulses in A, 4 pulses in B or vice versa */
         half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses
         subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);
 
         decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),
                         m - 2, off + half_4p + subhalf_2p);
         decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),
                         m - 1, off + half_4p);
         break;
     case 1: /* 1 pulse in A, 3 pulses in B */
         decode_1p_track(out, BIT_STR(code,  3*m - 2, m),
                         m - 1, off);
         decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),
                         m - 1, off + b_offset);
         break;
     case 2: /* 2 pulses in each half */
         decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),
                         m - 1, off);
         decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),
                         m - 1, off + b_offset);
         break;
     case 3: /* 3 pulses in A, 1 pulse in B */
         decode_3p_track(out, BIT_STR(code, m, 3*m - 2),
                         m - 1, off);
         decode_1p_track(out + 3, BIT_STR(code, 0, m),
                         m - 1, off + b_offset);
         break;
     }
 }
 
 static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits
 {
     int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);
 
     decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),
                     m - 1, off + half_3p);
 
     decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);
 }
 
 static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits
 {
     int b_offset = 1 << (m - 1);
     /* which half has more pulses in cases 0 to 2 */
     int half_more  = BIT_POS(code, 6*m - 5) << (m - 1);
     int half_other = b_offset - half_more;
 
     switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */
     case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
         decode_1p_track(out, BIT_STR(code, 0, m),
                         m - 1, off + half_more);
         decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
                         m - 1, off + half_more);
         break;
     case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
         decode_1p_track(out, BIT_STR(code, 0, m),
                         m - 1, off + half_other);
         decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
                         m - 1, off + half_more);
         break;
     case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
         decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
                         m - 1, off + half_other);
         decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),
                         m - 1, off + half_more);
         break;
     case 3: /* 3 pulses in A, 3 pulses in B */
         decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),
                         m - 1, off);
         decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2),
                         m - 1, off + b_offset);
         break;
     }
 }
 
 /**
  * Decode the algebraic codebook index to pulse positions and signs,

  * then construct the algebraic codebook vector.

  *
  * @param[out] fixed_vector        Buffer for the fixed codebook excitation
  * @param[in]  pulse_hi            MSBs part of the pulse index array (higher modes only)
  * @param[in]  pulse_lo            LSBs part of the pulse index array
  * @param[in]  mode                Mode of the current frame
  */
 static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
                                 const uint16_t *pulse_lo, const enum Mode mode)
 {
     /* sig_pos stores for each track the decoded pulse position indexes
      * (1-based) multiplied by its corresponding amplitude (+1 or -1) */
     int sig_pos[4][6];
     int spacing = (mode == MODE_6k60) ? 2 : 4;
     int i, j;
 
     switch (mode) {
     case MODE_6k60:
         for (i = 0; i < 2; i++)
             decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1);
         break;
     case MODE_8k85:
         for (i = 0; i < 4; i++)
             decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1);
         break;
     case MODE_12k65:
         for (i = 0; i < 4; i++)
             decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
         break;
     case MODE_14k25:
         for (i = 0; i < 2; i++)
             decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
         for (i = 2; i < 4; i++)
             decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
         break;
     case MODE_15k85:
         for (i = 0; i < 4; i++)
             decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
         break;
     case MODE_18k25:
         for (i = 0; i < 4; i++)
             decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
                            ((int) pulse_hi[i] << 14), 4, 1);
         break;
     case MODE_19k85:
         for (i = 0; i < 2; i++)
             decode_5p_track(sig_pos[i], (int) pulse_lo[i] +
                            ((int) pulse_hi[i] << 10), 4, 1);
         for (i = 2; i < 4; i++)
             decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
                            ((int) pulse_hi[i] << 14), 4, 1);
         break;
     case MODE_23k05:
     case MODE_23k85:
         for (i = 0; i < 4; i++)
             decode_6p_track(sig_pos[i], (int) pulse_lo[i] +
                            ((int) pulse_hi[i] << 11), 4, 1);
         break;
     }
 
     memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE);
 
     for (i = 0; i < 4; i++)
         for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) {
             int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i;
 
             fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0;
         }
 }
 
 /**

  * Decode pitch gain and fixed gain correction factor.

  *
  * @param[in]  vq_gain             Vector-quantized index for gains
  * @param[in]  mode                Mode of the current frame
  * @param[out] fixed_gain_factor   Decoded fixed gain correction factor
  * @param[out] pitch_gain          Decoded pitch gain
  */
 static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
                          float *fixed_gain_factor, float *pitch_gain)
 {
     const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :
                                                 qua_gain_7b[vq_gain]);
 
     *pitch_gain        = gains[0] * (1.0f / (1 << 14));
     *fixed_gain_factor = gains[1] * (1.0f / (1 << 11));
 }
 
 /**

  * Apply pitch sharpening filters to the fixed codebook vector.

  *
  * @param[in]     ctx              The context
  * @param[in,out] fixed_vector     Fixed codebook excitation
  */
 // XXX: Spec states this procedure should be applied when the pitch
 // lag is less than 64, but this checking seems absent in reference and AMR-NB
 static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
 {
     int i;
 
     /* Tilt part */
     for (i = AMRWB_SFR_SIZE - 1; i != 0; i--)
         fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef;
 
     /* Periodicity enhancement part */
     for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++)
         fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85;
 }
 
 /**

  * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).

  *
  * @param[in] p_vector, f_vector   Pitch and fixed excitation vectors
  * @param[in] p_gain, f_gain       Pitch and fixed gains

  * @param[in] ctx                  The context

  */
 // XXX: There is something wrong with the precision here! The magnitudes
 // of the energies are not correct. Please check the reference code carefully
 static float voice_factor(float *p_vector, float p_gain,

                           float *f_vector, float f_gain,
                           CELPMContext *ctx)

 {

     double p_ener = (double) ctx->dot_productf(p_vector, p_vector,

                                              AMRWB_SFR_SIZE) *
                                              p_gain * p_gain;

     double f_ener = (double) ctx->dot_productf(f_vector, f_vector,

                                              AMRWB_SFR_SIZE) *
                                              f_gain * f_gain;

 
     return (p_ener - f_ener) / (p_ener + f_ener);
 }
 
 /**

  * Reduce fixed vector sparseness by smoothing with one of three IR filters,
  * also known as "adaptive phase dispersion".

  *
  * @param[in]     ctx              The context
  * @param[in,out] fixed_vector     Unfiltered fixed vector
  * @param[out]    buf              Space for modified vector if necessary
  *
  * @return The potentially overwritten filtered fixed vector address
  */
 static float *anti_sparseness(AMRWBContext *ctx,
                               float *fixed_vector, float *buf)
 {
     int ir_filter_nr;
 
     if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes
         return fixed_vector;
 
     if (ctx->pitch_gain[0] < 0.6) {
         ir_filter_nr = 0;      // strong filtering
     } else if (ctx->pitch_gain[0] < 0.9) {
         ir_filter_nr = 1;      // medium filtering
     } else
         ir_filter_nr = 2;      // no filtering
 
     /* detect 'onset' */
     if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) {
         if (ir_filter_nr < 2)
             ir_filter_nr++;
     } else {
         int i, count = 0;
 
         for (i = 0; i < 6; i++)
             if (ctx->pitch_gain[i] < 0.6)
                 count++;
 
         if (count > 2)
             ir_filter_nr = 0;
 
         if (ir_filter_nr > ctx->prev_ir_filter_nr + 1)
             ir_filter_nr--;
     }
 
     /* update ir filter strength history */
     ctx->prev_ir_filter_nr = ir_filter_nr;
 
     ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85);
 
     if (ir_filter_nr < 2) {
         int i;
         const float *coef = ir_filters_lookup[ir_filter_nr];
 
         /* Circular convolution code in the reference
          * decoder was modified to avoid using one
          * extra array. The filtered vector is given by:
          *
          * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }
          */
 
         memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE);
         for (i = 0; i < AMRWB_SFR_SIZE; i++)
             if (fixed_vector[i])
                 ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],
                                   AMRWB_SFR_SIZE);
         fixed_vector = buf;
     }
 
     return fixed_vector;
 }
 
 /**
  * Calculate a stability factor {teta} based on distance between

  * current and past isf. A value of 1 shows maximum signal stability.

  */
 static float stability_factor(const float *isf, const float *isf_past)
 {
     int i;
     float acc = 0.0;
 
     for (i = 0; i < LP_ORDER - 1; i++)
         acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);
 
     // XXX: This part is not so clear from the reference code
     // the result is more accurate changing the "/ 256" to "* 512"
     return FFMAX(0.0, 1.25 - acc * 0.8 * 512);
 }
 
 /**
  * Apply a non-linear fixed gain smoothing in order to reduce

  * fluctuation in the energy of excitation.

  *
  * @param[in]     fixed_gain       Unsmoothed fixed gain
  * @param[in,out] prev_tr_gain     Previous threshold gain (updated)
  * @param[in]     voice_fac        Frame voicing factor
  * @param[in]     stab_fac         Frame stability factor
  *
  * @return The smoothed gain
  */
 static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
                             float voice_fac,  float stab_fac)
 {
     float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;
     float g0;
 
     // XXX: the following fixed-point constants used to in(de)crement
     // gain by 1.5dB were taken from the reference code, maybe it could
     // be simpler
     if (fixed_gain < *prev_tr_gain) {
         g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *
                      (6226 * (1.0f / (1 << 15)))); // +1.5 dB
     } else
         g0 = FFMAX(*prev_tr_gain, fixed_gain *
                     (27536 * (1.0f / (1 << 15)))); // -1.5 dB
 
     *prev_tr_gain = g0; // update next frame threshold
 
     return sm_fac * g0 + (1 - sm_fac) * fixed_gain;
 }
 
 /**

  * Filter the fixed_vector to emphasize the higher frequencies.

  *
  * @param[in,out] fixed_vector     Fixed codebook vector
  * @param[in]     voice_fac        Frame voicing factor
  */
 static void pitch_enhancer(float *fixed_vector, float voice_fac)
 {
     int i;
     float cpe  = 0.125 * (1 + voice_fac);
     float last = fixed_vector[0]; // holds c(i - 1)
 
     fixed_vector[0] -= cpe * fixed_vector[1];
 
     for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) {
         float cur = fixed_vector[i];
 
         fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]);
         last = cur;
     }
 
     fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last;
 }
 
 /**

  * Conduct 16th order linear predictive coding synthesis from excitation.

  *
  * @param[in]     ctx              Pointer to the AMRWBContext
  * @param[in]     lpc              Pointer to the LPC coefficients
  * @param[out]    excitation       Buffer for synthesis final excitation
  * @param[in]     fixed_gain       Fixed codebook gain for synthesis
  * @param[in]     fixed_vector     Algebraic codebook vector
  * @param[in,out] samples          Pointer to the output samples and memory
  */
 static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
                       float fixed_gain, const float *fixed_vector,
                       float *samples)
 {

     ctx->acelpv_ctx.weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,

                             ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);
 
     /* emphasize pitch vector contribution in low bitrate modes */
     if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
         int i;

         float energy = ctx->celpm_ctx.dot_productf(excitation, excitation,

                                                 AMRWB_SFR_SIZE);

 
         // XXX: Weird part in both ref code and spec. A unknown parameter
         // {beta} seems to be identical to the current pitch gain
         float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0];
 
         for (i = 0; i < AMRWB_SFR_SIZE; i++)
             excitation[i] += pitch_factor * ctx->pitch_vector[i];
 
         ff_scale_vector_to_given_sum_of_squares(excitation, excitation,
                                                 energy, AMRWB_SFR_SIZE);
     }
 

     ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,

                                  AMRWB_SFR_SIZE, LP_ORDER);
 }
 
 /**
  * Apply to synthesis a de-emphasis filter of the form:
  * H(z) = 1 / (1 - m * z^-1)
  *
  * @param[out]    out              Output buffer
  * @param[in]     in               Input samples array with in[-1]
  * @param[in]     m                Filter coefficient
  * @param[in,out] mem              State from last filtering
  */
 static void de_emphasis(float *out, float *in, float m, float mem[1])
 {
     int i;
 
     out[0] = in[0] + m * mem[0];
 
     for (i = 1; i < AMRWB_SFR_SIZE; i++)
          out[i] = in[i] + out[i - 1] * m;
 
     mem[0] = out[AMRWB_SFR_SIZE - 1];
 }
 
 /**
  * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using

  * a FIR interpolation filter. Uses past data from before *in address.

  *
  * @param[out] out                 Buffer for interpolated signal
  * @param[in]  in                  Current signal data (length 0.8*o_size)
  * @param[in]  o_size              Output signal length

  * @param[in] ctx                  The context

  */

 static void upsample_5_4(float *out, const float *in, int o_size, CELPMContext *ctx)

 {
     const float *in0 = in - UPS_FIR_SIZE + 1;
     int i, j, k;
     int int_part = 0, frac_part;
 
     i = 0;
     for (j = 0; j < o_size / 5; j++) {
         out[i] = in[int_part];
         frac_part = 4;
         i++;
 
         for (k = 1; k < 5; k++) {

             out[i] = ctx->dot_productf(in0 + int_part,

                                               upsample_fir[4 - frac_part],
                                               UPS_MEM_SIZE);

             int_part++;
             frac_part--;
             i++;
         }
     }
 }
 
 /**
  * Calculate the high-band gain based on encoded index (23k85 mode) or

  * on the low-band speech signal and the Voice Activity Detection flag.

  *
  * @param[in] ctx                  The context
  * @param[in] synth                LB speech synthesis at 12.8k
  * @param[in] hb_idx               Gain index for mode 23k85 only
  * @param[in] vad                  VAD flag for the frame
  */
 static float find_hb_gain(AMRWBContext *ctx, const float *synth,
                           uint16_t hb_idx, uint8_t vad)
 {
     int wsp = (vad > 0);
     float tilt;
 
     if (ctx->fr_cur_mode == MODE_23k85)
         return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
 

     tilt = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
            ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);

 
     /* return gain bounded by [0.1, 1.0] */
     return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0);
 }
 
 /**
  * Generate the high-band excitation with the same energy from the lower

  * one and scaled by the given gain.

  *
  * @param[in]  ctx                 The context
  * @param[out] hb_exc              Buffer for the excitation
  * @param[in]  synth_exc           Low-band excitation used for synthesis
  * @param[in]  hb_gain             Wanted excitation gain
  */
 static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
                                  const float *synth_exc, float hb_gain)
 {
     int i;

     float energy = ctx->celpm_ctx.dot_productf(synth_exc, synth_exc, AMRWB_SFR_SIZE);

 
     /* Generate a white-noise excitation */
     for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
         hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);
 
     ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc,
                                             energy * hb_gain * hb_gain,
                                             AMRWB_SFR_SIZE_16k);
 }
 
 /**

  * Calculate the auto-correlation for the ISF difference vector.

  */
 static float auto_correlation(float *diff_isf, float mean, int lag)
 {
     int i;
     float sum = 0.0;
 
     for (i = 7; i < LP_ORDER - 2; i++) {
         float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean);
         sum += prod * prod;
     }
     return sum;
 }
 
 /**
  * Extrapolate a ISF vector to the 16kHz range (20th order LP)

  * used at mode 6k60 LP filter for the high frequency band.

  *

  * @param[out] isf Buffer for extrapolated isf; contains LP_ORDER
  *                 values on input

  */

 static void extrapolate_isf(float isf[LP_ORDER_16k])

 {
     float diff_isf[LP_ORDER - 2], diff_mean;
     float corr_lag[3];
     float est, scale;

     int i, j, i_max_corr;

     isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];

 
     /* Calculate the difference vector */
     for (i = 0; i < LP_ORDER - 2; i++)
         diff_isf[i] = isf[i + 1] - isf[i];
 
     diff_mean = 0.0;
     for (i = 2; i < LP_ORDER - 2; i++)
         diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4));
 
     /* Find which is the maximum autocorrelation */
     i_max_corr = 0;
     for (i = 0; i < 3; i++) {
         corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);
 
         if (corr_lag[i] > corr_lag[i_max_corr])
             i_max_corr = i;
     }
     i_max_corr++;
 
     for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)

         isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr]

                             - isf[i - 2 - i_max_corr];
 
     /* Calculate an estimate for ISF(18) and scale ISF based on the error */

     est   = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0;
     scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) /
             (isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]);

     for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
         diff_isf[j] = scale * (isf[i] - isf[i - 1]);

 
     /* Stability insurance */

     for (i = 1; i < LP_ORDER_16k - LP_ORDER; i++)
         if (diff_isf[i] + diff_isf[i - 1] < 5.0) {
             if (diff_isf[i] > diff_isf[i - 1]) {
                 diff_isf[i - 1] = 5.0 - diff_isf[i];

             } else

                 diff_isf[i] = 5.0 - diff_isf[i - 1];

         }
 

     for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
         isf[i] = isf[i - 1] + diff_isf[j] * (1.0f / (1 << 15));

 
     /* Scale the ISF vector for 16000 Hz */
     for (i = 0; i < LP_ORDER_16k - 1; i++)

         isf[i] *= 0.8;

 }
 
 /**
  * Spectral expand the LP coefficients using the equation:
  *   y[i] = x[i] * (gamma ** i)
  *
  * @param[out] out                 Output buffer (may use input array)
  * @param[in]  lpc                 LP coefficients array
  * @param[in]  gamma               Weighting factor
  * @param[in]  size                LP array size
  */
 static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
 {
     int i;
     float fac = gamma;
 
     for (i = 0; i < size; i++) {
         out[i] = lpc[i] * fac;
         fac   *= gamma;
     }
 }
 
 /**
  * Conduct 20th order linear predictive coding synthesis for the high

  * frequency band excitation at 16kHz.

  *
  * @param[in]     ctx              The context
  * @param[in]     subframe         Current subframe index (0 to 3)
  * @param[in,out] samples          Pointer to the output speech samples
  * @param[in]     exc              Generated white-noise scaled excitation
  * @param[in]     isf              Current frame isf vector
  * @param[in]     isf_past         Past frame final isf vector
  */
 static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
                          const float *exc, const float *isf, const float *isf_past)
 {
     float hb_lpc[LP_ORDER_16k];
     enum Mode mode = ctx->fr_cur_mode;
 
     if (mode == MODE_6k60) {
         float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation
         double e_isp[LP_ORDER_16k];
 

         ctx->acelpv_ctx.weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],

                                 1.0 - isfp_inter[subframe], LP_ORDER);
 

         extrapolate_isf(e_isf);

 
         e_isf[LP_ORDER_16k - 1] *= 2.0;
         ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);
         ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k);
 
         lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k);
     } else {
         lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER);
     }
 

     ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,

                                  (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);
 }
 
 /**

  * Apply a 15th order filter to high-band samples.
  * The filter characteristic depends on the given coefficients.

  *
  * @param[out]    out              Buffer for filtered output
  * @param[in]     fir_coef         Filter coefficients
  * @param[in,out] mem              State from last filtering (updated)
  * @param[in]     in               Input speech data (high-band)
  *
  * @remark It is safe to pass the same array in in and out parameters
  */

 
 #ifndef hb_fir_filter

 static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
                           float mem[HB_FIR_SIZE], const float *in)
 {
     int i, j;
     float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples
 
     memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
     memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float));
 
     for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {
         out[i] = 0.0;
         for (j = 0; j <= HB_FIR_SIZE; j++)
             out[i] += data[i + j] * fir_coef[j];
     }
 
     memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));
 }

 #endif /* hb_fir_filter */

 
 /**

  * Update context state before the next subframe.

  */
 static void update_sub_state(AMRWBContext *ctx)
 {
     memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE],
             (AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float));
 
     memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float));
     memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float));
 
     memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE],
             LP_ORDER * sizeof(float));
     memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE],
             UPS_MEM_SIZE * sizeof(float));
     memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k],
             LP_ORDER_16k * sizeof(float));
 }
 

 static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
                               int *got_frame_ptr, AVPacket *avpkt)

 {
     AMRWBContext *ctx  = avctx->priv_data;
     AMRWBFrame   *cf   = &ctx->frame;
     const uint8_t *buf = avpkt->data;
     int buf_size       = avpkt->size;
     int expected_fr_size, header_size;

     float *buf_out;

     float spare_vector[AMRWB_SFR_SIZE];      // extra stack space to hold result from anti-sparseness processing
     float fixed_gain_factor;                 // fixed gain correction factor (gamma)
     float *synth_fixed_vector;               // pointer to the fixed vector that synthesis should use
     float synth_fixed_gain;                  // the fixed gain that synthesis should use
     float voice_fac, stab_fac;               // parameters used for gain smoothing
     float synth_exc[AMRWB_SFR_SIZE];         // post-processed excitation for synthesis
     float hb_exc[AMRWB_SFR_SIZE_16k];        // excitation for the high frequency band
     float hb_samples[AMRWB_SFR_SIZE_16k];    // filtered high-band samples from synthesis
     float hb_gain;

     int sub, i, ret;
 
     /* get output buffer */
     ctx->avframe.nb_samples = 4 * AMRWB_SFR_SIZE_16k;

     if ((ret = ff_get_buffer(avctx, &ctx->avframe)) < 0) {

         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
     }
     buf_out = (float *)ctx->avframe.data[0];

 
     header_size      = decode_mime_header(ctx, buf);

     if (ctx->fr_cur_mode > MODE_SID) {
         av_log(avctx, AV_LOG_ERROR,
                "Invalid mode %d\n", ctx->fr_cur_mode);
         return AVERROR_INVALIDDATA;
     }

     expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
 
     if (buf_size < expected_fr_size) {
         av_log(avctx, AV_LOG_ERROR,
             "Frame too small (%d bytes). Truncated file?\n", buf_size);

         *got_frame_ptr = 0;

         return AVERROR_INVALIDDATA;

     }
 
     if (!ctx->fr_quality || ctx->fr_cur_mode > MODE_SID)
         av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");
 

     if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */

         av_log_missing_feature(avctx, "SID mode", 1);

         return AVERROR_PATCHWELCOME;

     }

 
     ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
         buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);
 
     /* Decode the quantized ISF vector */
     if (ctx->fr_cur_mode == MODE_6k60) {
         decode_isf_indices_36b(cf->isp_id, ctx->isf_cur);
     } else {
         decode_isf_indices_46b(cf->isp_id, ctx->isf_cur);
     }
 
     isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past);
     ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1);
 
     stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);
 
     ctx->isf_cur[LP_ORDER - 1] *= 2.0;
     ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER);
 
     /* Generate a ISP vector for each subframe */
     if (ctx->first_frame) {
         ctx->first_frame = 0;
         memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double));
     }
     interpolate_isp(ctx->isp, ctx->isp_sub4_past);
 
     for (sub = 0; sub < 4; sub++)
         ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER);
 
     for (sub = 0; sub < 4; sub++) {
         const AMRWBSubFrame *cur_subframe = &cf->subframe[sub];
         float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;
 
         /* Decode adaptive codebook (pitch vector) */
         decode_pitch_vector(ctx, cur_subframe, sub);
         /* Decode innovative codebook (fixed vector) */
         decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih,
                             cur_subframe->pul_il, ctx->fr_cur_mode);
 
         pitch_sharpening(ctx, ctx->fixed_vector);
 
         decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode,
                      &fixed_gain_factor, &ctx->pitch_gain[0]);
 
         ctx->fixed_gain[0] =
             ff_amr_set_fixed_gain(fixed_gain_factor,

                                   ctx->celpm_ctx.dot_productf(ctx->fixed_vector,

                                                            ctx->fixed_vector,
                                                            AMRWB_SFR_SIZE) /
                                   AMRWB_SFR_SIZE,

                        ctx->prediction_error,
                        ENERGY_MEAN, energy_pred_fac);
 
         /* Calculate voice factor and store tilt for next subframe */
         voice_fac      = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0],

                                       ctx->fixed_vector, ctx->fixed_gain[0],
                                       &ctx->celpm_ctx);

         ctx->tilt_coef = voice_fac * 0.25 + 0.25;
 
         /* Construct current excitation */
         for (i = 0; i < AMRWB_SFR_SIZE; i++) {
             ctx->excitation[i] *= ctx->pitch_gain[0];
             ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i];
             ctx->excitation[i] = truncf(ctx->excitation[i]);
         }
 
         /* Post-processing of excitation elements */
         synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain,
                                           voice_fac, stab_fac);
 
         synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector,
                                              spare_vector);
 
         pitch_enhancer(synth_fixed_vector, voice_fac);
 
         synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
                   synth_fixed_vector, &ctx->samples_az[LP_ORDER]);
 
         /* Synthesis speech post-processing */
         de_emphasis(&ctx->samples_up[UPS_MEM_SIZE],
                     &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);
 

         ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE],

             &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles,
             hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE);
 
         upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],

                      AMRWB_SFR_SIZE_16k, &ctx->celpm_ctx);

 
         /* High frequency band (6.4 - 7.0 kHz) generation part */

         ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(hb_samples,

             &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles,
             hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE);
 
         hb_gain = find_hb_gain(ctx, hb_samples,
                                cur_subframe->hb_gain, cf->vad);
 
         scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);
 
         hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k],
                      hb_exc, ctx->isf_cur, ctx->isf_past_final);
 
         /* High-band post-processing filters */
         hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,
                       &ctx->samples_hb[LP_ORDER_16k]);
 
         if (ctx->fr_cur_mode == MODE_23k85)
             hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,
                           hb_samples);
 
         /* Add the low and high frequency bands */
         for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
             sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15));
 
         /* Update buffers and history */
         update_sub_state(ctx);
     }
 
     /* update state for next frame */
     memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
     memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
 

     *got_frame_ptr   = 1;
     *(AVFrame *)data = ctx->avframe;

 
     return expected_fr_size;
 }
 

 AVCodec ff_amrwb_decoder = {

     .name           = "amrwb",

     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_AMR_WB,

     .priv_data_size = sizeof(AMRWBContext),
     .init           = amrwb_decode_init,
     .decode         = amrwb_decode_frame,

     .capabilities   = CODEC_CAP_DR1,

     .long_name      = NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"),

     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
                                                      AV_SAMPLE_FMT_NONE },

 };