PeerStreamer

/*

 * 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/lfg.h"

#include "avcodec.h"

#include "get_bits.h"

#include "lsp.h"

#include "celp_math.h"

#include "celp_filters.h"

#include "acelp_filters.h"

#include "acelp_vectors.h"

#include "acelp_pitch_delay.h"

#define AMR_USE_16BIT_TABLES

#include "amr.h"

#include "amrwbdata.h"

typedef struct {

    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

} AMRWBContext;

static av_cold int amrwb_decode_init(AVCodecContext *avctx)

{

    AMRWBContext *ctx = avctx->priv_data;

    int i;

    avctx->sample_fmt = 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;

    return 0;

}

/**

 * Decode the frame header in the "MIME/storage" format. This format

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

 *

 * @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)

{

    GetBitContext gb;

    init_get_bits(&gb, buf, 8);

    /* Decode frame header (1st octet) */

    skip_bits(&gb, 1);  // padding bit

    ctx->fr_cur_mode  = get_bits(&gb, 4);

    ctx->fr_quality   = get_bits1(&gb);

    skip_bits(&gb, 2);  // padding bits

    return 1;

}

/**

 * Decodes 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));

}

/**

 * Decodes 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 a 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 a adaptive codebook index into pitch lag for 8k85 and 6k60 modes

 * Description is analogous to decode_pitch_lag_high, but in 6k60 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 */

    ff_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

 */

// 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)

{

    double p_ener = (double) ff_dot_productf(p_vector, p_vector,

                                             AMRWB_SFR_SIZE) * p_gain * p_gain;

    double f_ener = (double) ff_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)

{

    ff_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 = ff_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);

    }

    ff_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

 */

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

{

    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] = ff_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 = ff_dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /

           ff_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 = ff_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] out                 Buffer for extrapolated isf

 * @param[in]  isf                 Input isf vector

 */

static void extrapolate_isf(float out[LP_ORDER_16k], float isf[LP_ORDER])

{

    float diff_isf[LP_ORDER - 2], diff_mean;

    float *diff_hi = diff_isf - LP_ORDER + 1; // diff array for extrapolated indexes

    float corr_lag[3];

    float est, scale;

    int i, i_max_corr;

    memcpy(out, isf, (LP_ORDER - 1) * sizeof(float));

    out[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++)

        out[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 + (out[2] - out[3] - out[4]) / 6.0;

    scale = 0.5 * (FFMIN(est, 7600) - out[LP_ORDER - 2]) /

            (out[LP_ORDER_16k - 2] - out[LP_ORDER - 2]);

    for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)

        diff_hi[i] = scale * (out[i] - out[i - 1]);

    /* Stability insurance */

    for (i = LP_ORDER; i < LP_ORDER_16k - 1; i++)

        if (diff_hi[i] + diff_hi[i - 1] < 5.0) {

            if (diff_hi[i] > diff_hi[i - 1]) {

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

            } else

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

        }

    for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)

        out[i] = out[i - 1] + diff_hi[i] * (1.0f / (1 << 15));

    /* Scale the ISF vector for 16000 Hz */

    for (i = 0; i < LP_ORDER_16k - 1; i++)

        out[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];

        ff_weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],

                                1.0 - isfp_inter[subframe], LP_ORDER);

        extrapolate_isf(e_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);

    }

    ff_celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,

                                 (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);

}

/**

 * Apply to high-band samples a 15th order filter

 * 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

 */

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));

}

/**

 * 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 *data_size,

                              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 = data;

    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;

    header_size      = decode_mime_header(ctx, buf);

    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);

        *data_size = 0;

        return buf_size;

    }

    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);

    if (ctx->fr_cur_mode >= MODE_SID)

        return -1;

    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,

                       ff_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->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);

        ff_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);

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

        ff_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));

    /* report how many samples we got */

    *data_size = 4 * AMRWB_SFR_SIZE_16k * sizeof(float);

    return expected_fr_size;

}

AVCodec ff_amrwb_decoder = {

    .name           = "amrwb",

    .type           = AVMEDIA_TYPE_AUDIO,

    .id             = CODEC_ID_AMR_WB,

    .priv_data_size = sizeof(AMRWBContext),

    .init           = amrwb_decode_init,

    .decode         = amrwb_decode_frame,

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

    .sample_fmts    = (enum AVSampleFormat[]){SAMPLE_FMT_FLT,SAMPLE_FMT_NONE},

};