amrwbdec.c 46.7 KB
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/*
 * 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
 */

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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/lfg.h"

#include "avcodec.h"
#include "lsp.h"
#include "celp_filters.h"
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#include "celp_math.h"
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#include "acelp_filters.h"
#include "acelp_vectors.h"
#include "acelp_pitch_delay.h"
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#include "internal.h"
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#define AMR_USE_16BIT_TABLES
#include "amr.h"

#include "amrwbdata.h"
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#include "mips/amrwbdec_mips.h"
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typedef struct AMRWBContext {
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    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
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    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

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} AMRWBContext;

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

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    if (avctx->channels > 1) {
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        avpriv_report_missing_feature(avctx, "multi-channel AMR");
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        return AVERROR_PATCHWELCOME;
    }

    avctx->channels       = 1;
    avctx->channel_layout = AV_CH_LAYOUT_MONO;
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    if (!avctx->sample_rate)
        avctx->sample_rate = 16000;
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    avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;
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    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;

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    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);

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    return 0;
}

/**
 * Decode the frame header in the "MIME/storage" format. This format
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 * is simpler and does not carry the auxiliary frame information.
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 *
 * @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) */
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    ctx->fr_cur_mode  = buf[0] >> 3 & 0x0F;
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    ctx->fr_quality   = (buf[0] & 0x4) == 0x4;
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    return 1;
}

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

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

/**
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 * Apply mean and past ISF values using the prediction factor.
 * Updates past ISF vector.
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 *
 * @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
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 * to obtain an ISP vector for each subframe.
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 *
 * @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];
    }
}

/**
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 * 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.
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 *
 * @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;
    }
}

/**
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 * 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.
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 */
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
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 * pitch delay, which is obtained in this function.
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 *
 * @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 */
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    ctx->acelpf_ctx.acelp_interpolatef(exc,
                          exc + 1 - pitch_lag_int,
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                          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 */
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#define BIT_STR(x,lsb,len) av_mod_uintp2((x) >> (lsb), (len))
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/** 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
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 * an encoded pulse indexing (TS 26.190 section 5.8.2).
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 *
 * The results are given in out[], in which a negative number means
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 * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).
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 *
 * @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,
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 * then construct the algebraic codebook vector.
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 *
 * @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;
        }
}

/**
556
 * Decode pitch gain and fixed gain correction factor.
557 558 559 560 561 562 563 564 565 566 567 568 569 570 571 572 573
 *
 * @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));
}

/**
574
 * Apply pitch sharpening filters to the fixed codebook vector.
575 576 577 578 579 580 581 582 583 584 585 586 587 588 589 590 591 592 593 594
 *
 * @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;
}

/**
595
 * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
596 597 598
 *
 * @param[in] p_vector, f_vector   Pitch and fixed excitation vectors
 * @param[in] p_gain, f_gain       Pitch and fixed gains
599
 * @param[in] ctx                  The context
600 601 602 603
 */
// 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,
604 605
                          float *f_vector, float f_gain,
                          CELPMContext *ctx)
606
{
607
    double p_ener = (double) ctx->dot_productf(p_vector, p_vector,
608
                                                          AMRWB_SFR_SIZE) *
609
                    p_gain * p_gain;
610
    double f_ener = (double) ctx->dot_productf(f_vector, f_vector,
611
                                                          AMRWB_SFR_SIZE) *
612
                    f_gain * f_gain;
613 614 615 616 617

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

/**
618 619
 * Reduce fixed vector sparseness by smoothing with one of three IR filters,
 * also known as "adaptive phase dispersion".
620 621 622 623 624 625 626 627 628 629 630 631 632 633 634 635 636 637 638 639 640 641 642 643 644 645 646 647 648 649 650 651 652 653 654 655 656 657 658 659 660 661 662 663 664 665 666 667 668 669 670 671 672 673 674 675 676 677 678 679 680 681 682 683 684 685 686 687 688
 *
 * @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
689
 * current and past isf. A value of 1 shows maximum signal stability.
690 691 692 693 694 695 696 697 698 699 700 701 702 703 704 705
 */
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
706
 * fluctuation in the energy of excitation.
707 708 709 710 711 712 713 714 715 716 717 718 719 720 721 722 723 724 725 726 727 728 729 730 731 732 733 734 735 736
 *
 * @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;
}

/**
737
 * Filter the fixed_vector to emphasize the higher frequencies.
738 739 740 741 742 743 744 745 746 747 748 749 750 751 752 753 754 755 756 757 758 759 760
 *
 * @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;
}

/**
761
 * Conduct 16th order linear predictive coding synthesis from excitation.
762 763 764 765 766 767 768 769 770 771 772 773
 *
 * @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)
{
774
    ctx->acelpv_ctx.weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,
775 776 777 778 779
                            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;
780
        float energy = ctx->celpm_ctx.dot_productf(excitation, excitation,
781
                                                    AMRWB_SFR_SIZE);
782 783 784 785 786 787 788 789 790 791 792 793

        // 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);
    }

794
    ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
795 796 797 798 799 800 801 802 803 804 805 806 807 808 809 810 811 812 813 814 815 816 817 818 819 820
                                 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
821
 * a FIR interpolation filter. Uses past data from before *in address.
822 823 824 825
 *
 * @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
826
 * @param[in] ctx                  The context
827
 */
828
static void upsample_5_4(float *out, const float *in, int o_size, CELPMContext *ctx)
829 830 831 832 833 834 835 836 837 838 839 840
{
    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++) {
841
            out[i] = ctx->dot_productf(in0 + int_part,
842 843
                                                  upsample_fir[4 - frac_part],
                                                  UPS_MEM_SIZE);
844 845 846 847 848 849 850 851 852
            int_part++;
            frac_part--;
            i++;
        }
    }
}

/**
 * Calculate the high-band gain based on encoded index (23k85 mode) or
853
 * on the low-band speech signal and the Voice Activity Detection flag.
854 855 856 857 858 859 860 861 862 863 864 865 866 867 868
 *
 * @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));

869 870
    tilt = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
           ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);
871 872 873 874 875 876 877

    /* 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
878
 * one and scaled by the given gain.
879 880 881 882 883 884 885 886 887 888
 *
 * @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;
889
    float energy = ctx->celpm_ctx.dot_productf(synth_exc, synth_exc,
890
                                                AMRWB_SFR_SIZE);
891 892 893 894 895 896 897 898 899 900 901

    /* 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);
}

/**
902
 * Calculate the auto-correlation for the ISF difference vector.
903 904 905 906 907 908 909 910 911 912 913 914 915 916 917
 */
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)
918
 * used at mode 6k60 LP filter for the high frequency band.
919
 *
920 921
 * @param[out] isf Buffer for extrapolated isf; contains LP_ORDER
 *                 values on input
922
 */
923
static void extrapolate_isf(float isf[LP_ORDER_16k])
924 925 926 927
{
    float diff_isf[LP_ORDER - 2], diff_mean;
    float corr_lag[3];
    float est, scale;
928
    int i, j, i_max_corr;
929

930
    isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
931 932 933 934 935 936 937 938 939 940 941 942 943 944 945 946 947 948 949 950

    /* 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++)
951
        isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
952 953 954
                            - isf[i - 2 - i_max_corr];

    /* Calculate an estimate for ISF(18) and scale ISF based on the error */
955 956 957
    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]);
958

959 960
    for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
        diff_isf[j] = scale * (isf[i] - isf[i - 1]);
961 962

    /* Stability insurance */
963 964 965 966
    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];
967
            } else
968
                diff_isf[i] = 5.0 - diff_isf[i - 1];
969 970
        }

971 972
    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));
973 974 975

    /* Scale the ISF vector for 16000 Hz */
    for (i = 0; i < LP_ORDER_16k - 1; i++)
976
        isf[i] *= 0.8;
977 978 979 980 981 982 983 984 985 986 987 988 989 990 991 992 993 994 995 996 997 998 999 1000
}

/**
 * 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
1001
 * frequency band excitation at 16kHz.
1002 1003 1004 1005 1006 1007 1008 1009 1010 1011 1012 1013 1014 1015 1016 1017 1018 1019
 *
 * @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];

1020
        ctx->acelpv_ctx.weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],
1021 1022
                                1.0 - isfp_inter[subframe], LP_ORDER);

1023
        extrapolate_isf(e_isf);
1024 1025 1026 1027 1028 1029 1030 1031 1032 1033

        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);
    }

1034
    ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,
1035 1036 1037 1038
                                 (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);
}

/**
1039 1040
 * Apply a 15th order filter to high-band samples.
 * The filter characteristic depends on the given coefficients.
1041 1042 1043 1044 1045 1046 1047 1048
 *
 * @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
 */
1049 1050

#ifndef hb_fir_filter
1051 1052 1053 1054 1055 1056 1057 1058 1059 1060 1061 1062 1063 1064 1065 1066 1067
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));
}
1068
#endif /* hb_fir_filter */
1069 1070

/**
1071
 * Update context state before the next subframe.
1072 1073 1074 1075 1076 1077 1078 1079 1080 1081 1082 1083 1084 1085 1086 1087 1088
 */
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));
}

1089 1090
static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
                              int *got_frame_ptr, AVPacket *avpkt)
1091 1092
{
    AMRWBContext *ctx  = avctx->priv_data;
1093
    AVFrame *frame     = data;
1094 1095 1096 1097
    AMRWBFrame   *cf   = &ctx->frame;
    const uint8_t *buf = avpkt->data;
    int buf_size       = avpkt->size;
    int expected_fr_size, header_size;
1098
    float *buf_out;
1099 1100 1101 1102 1103 1104 1105 1106 1107
    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;
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    int sub, i, ret;

    /* get output buffer */
1111
    frame->nb_samples = 4 * AMRWB_SFR_SIZE_16k;
1112
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1113
        return ret;
1114
    buf_out = (float *)frame->data[0];
1115 1116

    header_size      = decode_mime_header(ctx, buf);
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    if (ctx->fr_cur_mode > MODE_SID) {
        av_log(avctx, AV_LOG_ERROR,
               "Invalid mode %d\n", ctx->fr_cur_mode);
        return AVERROR_INVALIDDATA;
    }
1122 1123 1124 1125 1126
    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);
1127
        *got_frame_ptr = 0;
1128
        return AVERROR_INVALIDDATA;
1129 1130 1131 1132 1133
    }

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

1134
    if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */
1135
        avpriv_request_sample(avctx, "SID mode");
1136
        return AVERROR_PATCHWELCOME;
1137
    }
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    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,
1184
                                  ctx->celpm_ctx.dot_productf(ctx->fixed_vector,
1185 1186
                                                               ctx->fixed_vector,
                                                               AMRWB_SFR_SIZE) /
1187
                                  AMRWB_SFR_SIZE,
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                       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],
1193 1194
                                      ctx->fixed_vector, ctx->fixed_gain[0],
                                      &ctx->celpm_ctx);
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        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);

1220
        ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE],
1221 1222 1223 1224
            &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],
1225
                     AMRWB_SFR_SIZE_16k, &ctx->celpm_ctx);
1226 1227

        /* High frequency band (6.4 - 7.0 kHz) generation part */
1228
        ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(hb_samples,
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            &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));

1260
    *got_frame_ptr = 1;
1261 1262 1263 1264

    return expected_fr_size;
}

1265
AVCodec ff_amrwb_decoder = {
1266
    .name           = "amrwb",
1267
    .long_name      = NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"),
1268
    .type           = AVMEDIA_TYPE_AUDIO,
1269
    .id             = AV_CODEC_ID_AMR_WB,
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    .priv_data_size = sizeof(AMRWBContext),
    .init           = amrwb_decode_init,
    .decode         = amrwb_decode_frame,
1273
    .capabilities   = AV_CODEC_CAP_DR1,
1274 1275
    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
                                                     AV_SAMPLE_FMT_NONE },
1276
};