AMR-WB decoder, written as part of Google Summer of Code 2010 by Marcelo

Galvão Póvoa <marspeoplester gmail com>, mentored by Robert Swain <robert
dot swain gmail com>.

Originally committed as revision 26051 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 1237 insertions, 0 deletions

diff --git a/libavcodec/amrwbdec.c b/libavcodec/amrwbdec.c
new file mode 100644
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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
+ */
+
+#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 amrwb_decoder = {
+    .name           = "amrwb",
+    .type           = CODEC_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 SampleFormat[]){SAMPLE_FMT_FLT,SAMPLE_FMT_NONE},
+};