chasing indexing error

1 files changed, 0 insertions, 1279 deletions

diff --git a/ffmpeg/libavcodec/amrwbdec.c b/ffmpeg/libavcodec/amrwbdec.c
deleted file mode 100644
index bf668bb..0000000
--- a/ffmpeg/libavcodec/amrwbdec.c
+++ /dev/null
@@ -1,1279 +0,0 @@
-/*
- * AMR wideband decoder
- * Copyright (c) 2010 Marcelo Galvao Povoa
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A particular PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * AMR wideband decoder
- */
-
-#include "libavutil/channel_layout.h"
-#include "libavutil/common.h"
-#include "libavutil/float_dsp.h"
-#include "libavutil/lfg.h"
-
-#include "avcodec.h"
-#include "lsp.h"
-#include "celp_filters.h"
-#include "celp_math.h"
-#include "acelp_filters.h"
-#include "acelp_vectors.h"
-#include "acelp_pitch_delay.h"
-#include "internal.h"
-
-#define AMR_USE_16BIT_TABLES
-#include "amr.h"
-
-#include "amrwbdata.h"
-#include "mips/amrwbdec_mips.h"
-
-typedef struct {
-    AMRWBFrame                             frame; ///< AMRWB parameters decoded from bitstream
-    enum Mode                        fr_cur_mode; ///< mode index of current frame
-    uint8_t                           fr_quality; ///< frame quality index (FQI)
-    float                      isf_cur[LP_ORDER]; ///< working ISF vector from current frame
-    float                   isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame
-    float               isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame
-    double                      isp[4][LP_ORDER]; ///< ISP vectors from current frame
-    double               isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame
-
-    float                   lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector
-
-    uint8_t                       base_pitch_lag; ///< integer part of pitch lag for the next relative subframe
-    uint8_t                        pitch_lag_int; ///< integer part of pitch lag of the previous subframe
-
-    float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history
-    float                            *excitation; ///< points to current excitation in excitation_buf[]
-
-    float           pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe
-    float           fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe
-
-    float                    prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
-    float                          pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes
-    float                          fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes
-
-    float                              tilt_coef; ///< {beta_1} related to the voicing of the previous subframe
-
-    float                 prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness to determine "onset"
-    uint8_t                    prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
-    float                           prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold
-
-    float samples_az[LP_ORDER + AMRWB_SFR_SIZE];         ///< low-band samples and memory from synthesis at 12.8kHz
-    float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE];     ///< low-band samples and memory processed for upsampling
-    float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< high-band samples and memory from synthesis at 16kHz
-
-    float          hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters
-    float                           demph_mem[1]; ///< previous value in the de-emphasis filter
-    float               bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band band pass filter
-    float                 lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the high-band low pass filter
-
-    AVLFG                                   prng; ///< random number generator for white noise excitation
-    uint8_t                          first_frame; ///< flag active during decoding of the first frame
-    ACELPFContext                     acelpf_ctx; ///< context for filters for ACELP-based codecs
-    ACELPVContext                     acelpv_ctx; ///< context for vector operations for ACELP-based codecs
-    CELPFContext                       celpf_ctx; ///< context for filters for CELP-based codecs
-    CELPMContext                       celpm_ctx; ///< context for fixed point math operations
-
-} AMRWBContext;
-
-static av_cold int amrwb_decode_init(AVCodecContext *avctx)
-{
-    AMRWBContext *ctx = avctx->priv_data;
-    int i;
-
-    if (avctx->channels > 1) {
-        avpriv_report_missing_feature(avctx, "multi-channel AMR");
-        return AVERROR_PATCHWELCOME;
-    }
-
-    avctx->channels       = 1;
-    avctx->channel_layout = AV_CH_LAYOUT_MONO;
-    if (!avctx->sample_rate)
-        avctx->sample_rate = 16000;
-    avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;
-
-    av_lfg_init(&ctx->prng, 1);
-
-    ctx->excitation  = &ctx->excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1];
-    ctx->first_frame = 1;
-
-    for (i = 0; i < LP_ORDER; i++)
-        ctx->isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15));
-
-    for (i = 0; i < 4; i++)
-        ctx->prediction_error[i] = MIN_ENERGY;
-
-    ff_acelp_filter_init(&ctx->acelpf_ctx);
-    ff_acelp_vectors_init(&ctx->acelpv_ctx);
-    ff_celp_filter_init(&ctx->celpf_ctx);
-    ff_celp_math_init(&ctx->celpm_ctx);
-
-    return 0;
-}
-
-/**
- * Decode the frame header in the "MIME/storage" format. This format
- * is simpler and does not carry the auxiliary frame information.
- *
- * @param[in] ctx                  The Context
- * @param[in] buf                  Pointer to the input buffer
- *
- * @return The decoded header length in bytes
- */
-static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf)
-{
-    /* Decode frame header (1st octet) */
-    ctx->fr_cur_mode  = buf[0] >> 3 & 0x0F;
-    ctx->fr_quality   = (buf[0] & 0x4) == 0x4;
-
-    return 1;
-}
-
-/**
- * Decode quantized ISF vectors using 36-bit indexes (6K60 mode only).
- *
- * @param[in]  ind                 Array of 5 indexes
- * @param[out] isf_q               Buffer for isf_q[LP_ORDER]
- *
- */
-static void decode_isf_indices_36b(uint16_t *ind, float *isf_q)
-{
-    int i;
-
-    for (i = 0; i < 9; i++)
-        isf_q[i]      = dico1_isf[ind[0]][i]      * (1.0f / (1 << 15));
-
-    for (i = 0; i < 7; i++)
-        isf_q[i + 9]  = dico2_isf[ind[1]][i]      * (1.0f / (1 << 15));
-
-    for (i = 0; i < 5; i++)
-        isf_q[i]     += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15));
-
-    for (i = 0; i < 4; i++)
-        isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15));
-
-    for (i = 0; i < 7; i++)
-        isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15));
-}
-
-/**
- * Decode quantized ISF vectors using 46-bit indexes (except 6K60 mode).
- *
- * @param[in]  ind                 Array of 7 indexes
- * @param[out] isf_q               Buffer for isf_q[LP_ORDER]
- *
- */
-static void decode_isf_indices_46b(uint16_t *ind, float *isf_q)
-{
-    int i;
-
-    for (i = 0; i < 9; i++)
-        isf_q[i]       = dico1_isf[ind[0]][i]  * (1.0f / (1 << 15));
-
-    for (i = 0; i < 7; i++)
-        isf_q[i + 9]   = dico2_isf[ind[1]][i]  * (1.0f / (1 << 15));
-
-    for (i = 0; i < 3; i++)
-        isf_q[i]      += dico21_isf[ind[2]][i] * (1.0f / (1 << 15));
-
-    for (i = 0; i < 3; i++)
-        isf_q[i + 3]  += dico22_isf[ind[3]][i] * (1.0f / (1 << 15));
-
-    for (i = 0; i < 3; i++)
-        isf_q[i + 6]  += dico23_isf[ind[4]][i] * (1.0f / (1 << 15));
-
-    for (i = 0; i < 3; i++)
-        isf_q[i + 9]  += dico24_isf[ind[5]][i] * (1.0f / (1 << 15));
-
-    for (i = 0; i < 4; i++)
-        isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15));
-}
-
-/**
- * Apply mean and past ISF values using the prediction factor.
- * Updates past ISF vector.
- *
- * @param[in,out] isf_q            Current quantized ISF
- * @param[in,out] isf_past         Past quantized ISF
- *
- */
-static void isf_add_mean_and_past(float *isf_q, float *isf_past)
-{
-    int i;
-    float tmp;
-
-    for (i = 0; i < LP_ORDER; i++) {
-        tmp = isf_q[i];
-        isf_q[i] += isf_mean[i] * (1.0f / (1 << 15));
-        isf_q[i] += PRED_FACTOR * isf_past[i];
-        isf_past[i] = tmp;
-    }
-}
-
-/**
- * Interpolate the fourth ISP vector from current and past frames
- * to obtain an ISP vector for each subframe.
- *
- * @param[in,out] isp_q            ISPs for each subframe
- * @param[in]     isp4_past        Past ISP for subframe 4
- */
-static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past)
-{
-    int i, k;
-
-    for (k = 0; k < 3; k++) {
-        float c = isfp_inter[k];
-        for (i = 0; i < LP_ORDER; i++)
-            isp_q[k][i] = (1.0 - c) * isp4_past[i] + c * isp_q[3][i];
-    }
-}
-
-/**
- * Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes).
- * Calculate integer lag and fractional lag always using 1/4 resolution.
- * In 1st and 3rd subframes the index is relative to last subframe integer lag.
- *
- * @param[out]    lag_int          Decoded integer pitch lag
- * @param[out]    lag_frac         Decoded fractional pitch lag
- * @param[in]     pitch_index      Adaptive codebook pitch index
- * @param[in,out] base_lag_int     Base integer lag used in relative subframes
- * @param[in]     subframe         Current subframe index (0 to 3)
- */
-static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index,
-                                  uint8_t *base_lag_int, int subframe)
-{
-    if (subframe == 0 || subframe == 2) {
-        if (pitch_index < 376) {
-            *lag_int  = (pitch_index + 137) >> 2;
-            *lag_frac = pitch_index - (*lag_int << 2) + 136;
-        } else if (pitch_index < 440) {
-            *lag_int  = (pitch_index + 257 - 376) >> 1;
-            *lag_frac = (pitch_index - (*lag_int << 1) + 256 - 376) << 1;
-            /* the actual resolution is 1/2 but expressed as 1/4 */
-        } else {
-            *lag_int  = pitch_index - 280;
-            *lag_frac = 0;
-        }
-        /* minimum lag for next subframe */
-        *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
-                                AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
-        // XXX: the spec states clearly that *base_lag_int should be
-        // the nearest integer to *lag_int (minus 8), but the ref code
-        // actually always uses its floor, I'm following the latter
-    } else {
-        *lag_int  = (pitch_index + 1) >> 2;
-        *lag_frac = pitch_index - (*lag_int << 2);
-        *lag_int += *base_lag_int;
-    }
-}
-
-/**
- * Decode an adaptive codebook index into pitch lag for 8k85 and 6k60 modes.
- * The description is analogous to decode_pitch_lag_high, but in 6k60 the
- * relative index is used for all subframes except the first.
- */
-static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index,
-                                 uint8_t *base_lag_int, int subframe, enum Mode mode)
-{
-    if (subframe == 0 || (subframe == 2 && mode != MODE_6k60)) {
-        if (pitch_index < 116) {
-            *lag_int  = (pitch_index + 69) >> 1;
-            *lag_frac = (pitch_index - (*lag_int << 1) + 68) << 1;
-        } else {
-            *lag_int  = pitch_index - 24;
-            *lag_frac = 0;
-        }
-        // XXX: same problem as before
-        *base_lag_int = av_clip(*lag_int - 8 - (*lag_frac < 0),
-                                AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX - 15);
-    } else {
-        *lag_int  = (pitch_index + 1) >> 1;
-        *lag_frac = (pitch_index - (*lag_int << 1)) << 1;
-        *lag_int += *base_lag_int;
-    }
-}
-
-/**
- * Find the pitch vector by interpolating the past excitation at the
- * pitch delay, which is obtained in this function.
- *
- * @param[in,out] ctx              The context
- * @param[in]     amr_subframe     Current subframe data
- * @param[in]     subframe         Current subframe index (0 to 3)
- */
-static void decode_pitch_vector(AMRWBContext *ctx,
-                                const AMRWBSubFrame *amr_subframe,
-                                const int subframe)
-{
-    int pitch_lag_int, pitch_lag_frac;
-    int i;
-    float *exc     = ctx->excitation;
-    enum Mode mode = ctx->fr_cur_mode;
-
-    if (mode <= MODE_8k85) {
-        decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
-                              &ctx->base_pitch_lag, subframe, mode);
-    } else
-        decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe->adap,
-                              &ctx->base_pitch_lag, subframe);
-
-    ctx->pitch_lag_int = pitch_lag_int;
-    pitch_lag_int += pitch_lag_frac > 0;
-
-    /* Calculate the pitch vector by interpolating the past excitation at the
-       pitch lag using a hamming windowed sinc function */
-    ctx->acelpf_ctx.acelp_interpolatef(exc,
-                          exc + 1 - pitch_lag_int,
-                          ac_inter, 4,
-                          pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4),
-                          LP_ORDER, AMRWB_SFR_SIZE + 1);
-
-    /* Check which pitch signal path should be used
-     * 6k60 and 8k85 modes have the ltp flag set to 0 */
-    if (amr_subframe->ltp) {
-        memcpy(ctx->pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float));
-    } else {
-        for (i = 0; i < AMRWB_SFR_SIZE; i++)
-            ctx->pitch_vector[i] = 0.18 * exc[i - 1] + 0.64 * exc[i] +
-                                   0.18 * exc[i + 1];
-        memcpy(exc, ctx->pitch_vector, AMRWB_SFR_SIZE * sizeof(float));
-    }
-}
-
-/** Get x bits in the index interval [lsb,lsb+len-1] inclusive */
-#define BIT_STR(x,lsb,len) (((x) >> (lsb)) & ((1 << (len)) - 1))
-
-/** Get the bit at specified position */
-#define BIT_POS(x, p) (((x) >> (p)) & 1)
-
-/**
- * The next six functions decode_[i]p_track decode exactly i pulses
- * positions and amplitudes (-1 or 1) in a subframe track using
- * an encoded pulse indexing (TS 26.190 section 5.8.2).
- *
- * The results are given in out[], in which a negative number means
- * amplitude -1 and vice versa (i.e., ampl(x) = x / abs(x) ).
- *
- * @param[out] out                 Output buffer (writes i elements)
- * @param[in]  code                Pulse index (no. of bits varies, see below)
- * @param[in]  m                   (log2) Number of potential positions
- * @param[in]  off                 Offset for decoded positions
- */
-static inline void decode_1p_track(int *out, int code, int m, int off)
-{
-    int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits
-
-    out[0] = BIT_POS(code, m) ? -pos : pos;
-}
-
-static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits
-{
-    int pos0 = BIT_STR(code, m, m) + off;
-    int pos1 = BIT_STR(code, 0, m) + off;
-
-    out[0] = BIT_POS(code, 2*m) ? -pos0 : pos0;
-    out[1] = BIT_POS(code, 2*m) ? -pos1 : pos1;
-    out[1] = pos0 > pos1 ? -out[1] : out[1];
-}
-
-static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits
-{
-    int half_2p = BIT_POS(code, 2*m - 1) << (m - 1);
-
-    decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
-                    m - 1, off + half_2p);
-    decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off);
-}
-
-static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits
-{
-    int half_4p, subhalf_2p;
-    int b_offset = 1 << (m - 1);
-
-    switch (BIT_STR(code, 4*m - 2, 2)) { /* case ID (2 bits) */
-    case 0: /* 0 pulses in A, 4 pulses in B or vice versa */
-        half_4p = BIT_POS(code, 4*m - 3) << (m - 1); // which has 4 pulses
-        subhalf_2p = BIT_POS(code, 2*m - 3) << (m - 2);
-
-        decode_2p_track(out, BIT_STR(code, 0, 2*m - 3),
-                        m - 2, off + half_4p + subhalf_2p);
-        decode_2p_track(out + 2, BIT_STR(code, 2*m - 2, 2*m - 1),
-                        m - 1, off + half_4p);
-        break;
-    case 1: /* 1 pulse in A, 3 pulses in B */
-        decode_1p_track(out, BIT_STR(code,  3*m - 2, m),
-                        m - 1, off);
-        decode_3p_track(out + 1, BIT_STR(code, 0, 3*m - 2),
-                        m - 1, off + b_offset);
-        break;
-    case 2: /* 2 pulses in each half */
-        decode_2p_track(out, BIT_STR(code, 2*m - 1, 2*m - 1),
-                        m - 1, off);
-        decode_2p_track(out + 2, BIT_STR(code, 0, 2*m - 1),
-                        m - 1, off + b_offset);
-        break;
-    case 3: /* 3 pulses in A, 1 pulse in B */
-        decode_3p_track(out, BIT_STR(code, m, 3*m - 2),
-                        m - 1, off);
-        decode_1p_track(out + 3, BIT_STR(code, 0, m),
-                        m - 1, off + b_offset);
-        break;
-    }
-}
-
-static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits
-{
-    int half_3p = BIT_POS(code, 5*m - 1) << (m - 1);
-
-    decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m - 2),
-                    m - 1, off + half_3p);
-
-    decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off);
-}
-
-static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m-2 bits
-{
-    int b_offset = 1 << (m - 1);
-    /* which half has more pulses in cases 0 to 2 */
-    int half_more  = BIT_POS(code, 6*m - 5) << (m - 1);
-    int half_other = b_offset - half_more;
-
-    switch (BIT_STR(code, 6*m - 4, 2)) { /* case ID (2 bits) */
-    case 0: /* 0 pulses in A, 6 pulses in B or vice versa */
-        decode_1p_track(out, BIT_STR(code, 0, m),
-                        m - 1, off + half_more);
-        decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
-                        m - 1, off + half_more);
-        break;
-    case 1: /* 1 pulse in A, 5 pulses in B or vice versa */
-        decode_1p_track(out, BIT_STR(code, 0, m),
-                        m - 1, off + half_other);
-        decode_5p_track(out + 1, BIT_STR(code, m, 5*m - 5),
-                        m - 1, off + half_more);
-        break;
-    case 2: /* 2 pulses in A, 4 pulses in B or vice versa */
-        decode_2p_track(out, BIT_STR(code, 0, 2*m - 1),
-                        m - 1, off + half_other);
-        decode_4p_track(out + 2, BIT_STR(code, 2*m - 1, 4*m - 4),
-                        m - 1, off + half_more);
-        break;
-    case 3: /* 3 pulses in A, 3 pulses in B */
-        decode_3p_track(out, BIT_STR(code, 3*m - 2, 3*m - 2),
-                        m - 1, off);
-        decode_3p_track(out + 3, BIT_STR(code, 0, 3*m - 2),
-                        m - 1, off + b_offset);
-        break;
-    }
-}
-
-/**
- * Decode the algebraic codebook index to pulse positions and signs,
- * then construct the algebraic codebook vector.
- *
- * @param[out] fixed_vector        Buffer for the fixed codebook excitation
- * @param[in]  pulse_hi            MSBs part of the pulse index array (higher modes only)
- * @param[in]  pulse_lo            LSBs part of the pulse index array
- * @param[in]  mode                Mode of the current frame
- */
-static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi,
-                                const uint16_t *pulse_lo, const enum Mode mode)
-{
-    /* sig_pos stores for each track the decoded pulse position indexes
-     * (1-based) multiplied by its corresponding amplitude (+1 or -1) */
-    int sig_pos[4][6];
-    int spacing = (mode == MODE_6k60) ? 2 : 4;
-    int i, j;
-
-    switch (mode) {
-    case MODE_6k60:
-        for (i = 0; i < 2; i++)
-            decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1);
-        break;
-    case MODE_8k85:
-        for (i = 0; i < 4; i++)
-            decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1);
-        break;
-    case MODE_12k65:
-        for (i = 0; i < 4; i++)
-            decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
-        break;
-    case MODE_14k25:
-        for (i = 0; i < 2; i++)
-            decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
-        for (i = 2; i < 4; i++)
-            decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1);
-        break;
-    case MODE_15k85:
-        for (i = 0; i < 4; i++)
-            decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1);
-        break;
-    case MODE_18k25:
-        for (i = 0; i < 4; i++)
-            decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
-                           ((int) pulse_hi[i] << 14), 4, 1);
-        break;
-    case MODE_19k85:
-        for (i = 0; i < 2; i++)
-            decode_5p_track(sig_pos[i], (int) pulse_lo[i] +
-                           ((int) pulse_hi[i] << 10), 4, 1);
-        for (i = 2; i < 4; i++)
-            decode_4p_track(sig_pos[i], (int) pulse_lo[i] +
-                           ((int) pulse_hi[i] << 14), 4, 1);
-        break;
-    case MODE_23k05:
-    case MODE_23k85:
-        for (i = 0; i < 4; i++)
-            decode_6p_track(sig_pos[i], (int) pulse_lo[i] +
-                           ((int) pulse_hi[i] << 11), 4, 1);
-        break;
-    }
-
-    memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE);
-
-    for (i = 0; i < 4; i++)
-        for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) {
-            int pos = (FFABS(sig_pos[i][j]) - 1) * spacing + i;
-
-            fixed_vector[pos] += sig_pos[i][j] < 0 ? -1.0 : 1.0;
-        }
-}
-
-/**
- * Decode pitch gain and fixed gain correction factor.
- *
- * @param[in]  vq_gain             Vector-quantized index for gains
- * @param[in]  mode                Mode of the current frame
- * @param[out] fixed_gain_factor   Decoded fixed gain correction factor
- * @param[out] pitch_gain          Decoded pitch gain
- */
-static void decode_gains(const uint8_t vq_gain, const enum Mode mode,
-                         float *fixed_gain_factor, float *pitch_gain)
-{
-    const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :
-                                                qua_gain_7b[vq_gain]);
-
-    *pitch_gain        = gains[0] * (1.0f / (1 << 14));
-    *fixed_gain_factor = gains[1] * (1.0f / (1 << 11));
-}
-
-/**
- * Apply pitch sharpening filters to the fixed codebook vector.
- *
- * @param[in]     ctx              The context
- * @param[in,out] fixed_vector     Fixed codebook excitation
- */
-// XXX: Spec states this procedure should be applied when the pitch
-// lag is less than 64, but this checking seems absent in reference and AMR-NB
-static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
-{
-    int i;
-
-    /* Tilt part */
-    for (i = AMRWB_SFR_SIZE - 1; i != 0; i--)
-        fixed_vector[i] -= fixed_vector[i - 1] * ctx->tilt_coef;
-
-    /* Periodicity enhancement part */
-    for (i = ctx->pitch_lag_int; i < AMRWB_SFR_SIZE; i++)
-        fixed_vector[i] += fixed_vector[i - ctx->pitch_lag_int] * 0.85;
-}
-
-/**
- * Calculate the voicing factor (-1.0 = unvoiced to 1.0 = voiced).
- *
- * @param[in] p_vector, f_vector   Pitch and fixed excitation vectors
- * @param[in] p_gain, f_gain       Pitch and fixed gains
- * @param[in] ctx                  The context
- */
-// XXX: There is something wrong with the precision here! The magnitudes
-// of the energies are not correct. Please check the reference code carefully
-static float voice_factor(float *p_vector, float p_gain,
-                          float *f_vector, float f_gain,
-                          CELPMContext *ctx)
-{
-    double p_ener = (double) ctx->dot_productf(p_vector, p_vector,
-                                                          AMRWB_SFR_SIZE) *
-                    p_gain * p_gain;
-    double f_ener = (double) ctx->dot_productf(f_vector, f_vector,
-                                                          AMRWB_SFR_SIZE) *
-                    f_gain * f_gain;
-
-    return (p_ener - f_ener) / (p_ener + f_ener);
-}
-
-/**
- * Reduce fixed vector sparseness by smoothing with one of three IR filters,
- * also known as "adaptive phase dispersion".
- *
- * @param[in]     ctx              The context
- * @param[in,out] fixed_vector     Unfiltered fixed vector
- * @param[out]    buf              Space for modified vector if necessary
- *
- * @return The potentially overwritten filtered fixed vector address
- */
-static float *anti_sparseness(AMRWBContext *ctx,
-                              float *fixed_vector, float *buf)
-{
-    int ir_filter_nr;
-
-    if (ctx->fr_cur_mode > MODE_8k85) // no filtering in higher modes
-        return fixed_vector;
-
-    if (ctx->pitch_gain[0] < 0.6) {
-        ir_filter_nr = 0;      // strong filtering
-    } else if (ctx->pitch_gain[0] < 0.9) {
-        ir_filter_nr = 1;      // medium filtering
-    } else
-        ir_filter_nr = 2;      // no filtering
-
-    /* detect 'onset' */
-    if (ctx->fixed_gain[0] > 3.0 * ctx->fixed_gain[1]) {
-        if (ir_filter_nr < 2)
-            ir_filter_nr++;
-    } else {
-        int i, count = 0;
-
-        for (i = 0; i < 6; i++)
-            if (ctx->pitch_gain[i] < 0.6)
-                count++;
-
-        if (count > 2)
-            ir_filter_nr = 0;
-
-        if (ir_filter_nr > ctx->prev_ir_filter_nr + 1)
-            ir_filter_nr--;
-    }
-
-    /* update ir filter strength history */
-    ctx->prev_ir_filter_nr = ir_filter_nr;
-
-    ir_filter_nr += (ctx->fr_cur_mode == MODE_8k85);
-
-    if (ir_filter_nr < 2) {
-        int i;
-        const float *coef = ir_filters_lookup[ir_filter_nr];
-
-        /* Circular convolution code in the reference
-         * decoder was modified to avoid using one
-         * extra array. The filtered vector is given by:
-         *
-         * c2(n) = sum(i,0,len-1){ c(i) * coef( (n - i + len) % len ) }
-         */
-
-        memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE);
-        for (i = 0; i < AMRWB_SFR_SIZE; i++)
-            if (fixed_vector[i])
-                ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i],
-                                  AMRWB_SFR_SIZE);
-        fixed_vector = buf;
-    }
-
-    return fixed_vector;
-}
-
-/**
- * Calculate a stability factor {teta} based on distance between
- * current and past isf. A value of 1 shows maximum signal stability.
- */
-static float stability_factor(const float *isf, const float *isf_past)
-{
-    int i;
-    float acc = 0.0;
-
-    for (i = 0; i < LP_ORDER - 1; i++)
-        acc += (isf[i] - isf_past[i]) * (isf[i] - isf_past[i]);
-
-    // XXX: This part is not so clear from the reference code
-    // the result is more accurate changing the "/ 256" to "* 512"
-    return FFMAX(0.0, 1.25 - acc * 0.8 * 512);
-}
-
-/**
- * Apply a non-linear fixed gain smoothing in order to reduce
- * fluctuation in the energy of excitation.
- *
- * @param[in]     fixed_gain       Unsmoothed fixed gain
- * @param[in,out] prev_tr_gain     Previous threshold gain (updated)
- * @param[in]     voice_fac        Frame voicing factor
- * @param[in]     stab_fac         Frame stability factor
- *
- * @return The smoothed gain
- */
-static float noise_enhancer(float fixed_gain, float *prev_tr_gain,
-                            float voice_fac,  float stab_fac)
-{
-    float sm_fac = 0.5 * (1 - voice_fac) * stab_fac;
-    float g0;
-
-    // XXX: the following fixed-point constants used to in(de)crement
-    // gain by 1.5dB were taken from the reference code, maybe it could
-    // be simpler
-    if (fixed_gain < *prev_tr_gain) {
-        g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain *
-                     (6226 * (1.0f / (1 << 15)))); // +1.5 dB
-    } else
-        g0 = FFMAX(*prev_tr_gain, fixed_gain *
-                    (27536 * (1.0f / (1 << 15)))); // -1.5 dB
-
-    *prev_tr_gain = g0; // update next frame threshold
-
-    return sm_fac * g0 + (1 - sm_fac) * fixed_gain;
-}
-
-/**
- * Filter the fixed_vector to emphasize the higher frequencies.
- *
- * @param[in,out] fixed_vector     Fixed codebook vector
- * @param[in]     voice_fac        Frame voicing factor
- */
-static void pitch_enhancer(float *fixed_vector, float voice_fac)
-{
-    int i;
-    float cpe  = 0.125 * (1 + voice_fac);
-    float last = fixed_vector[0]; // holds c(i - 1)
-
-    fixed_vector[0] -= cpe * fixed_vector[1];
-
-    for (i = 1; i < AMRWB_SFR_SIZE - 1; i++) {
-        float cur = fixed_vector[i];
-
-        fixed_vector[i] -= cpe * (last + fixed_vector[i + 1]);
-        last = cur;
-    }
-
-    fixed_vector[AMRWB_SFR_SIZE - 1] -= cpe * last;
-}
-
-/**
- * Conduct 16th order linear predictive coding synthesis from excitation.
- *
- * @param[in]     ctx              Pointer to the AMRWBContext
- * @param[in]     lpc              Pointer to the LPC coefficients
- * @param[out]    excitation       Buffer for synthesis final excitation
- * @param[in]     fixed_gain       Fixed codebook gain for synthesis
- * @param[in]     fixed_vector     Algebraic codebook vector
- * @param[in,out] samples          Pointer to the output samples and memory
- */
-static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
-                      float fixed_gain, const float *fixed_vector,
-                      float *samples)
-{
-    ctx->acelpv_ctx.weighted_vector_sumf(excitation, ctx->pitch_vector, fixed_vector,
-                            ctx->pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);
-
-    /* emphasize pitch vector contribution in low bitrate modes */
-    if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
-        int i;
-        float energy = ctx->celpm_ctx.dot_productf(excitation, excitation,
-                                                    AMRWB_SFR_SIZE);
-
-        // XXX: Weird part in both ref code and spec. A unknown parameter
-        // {beta} seems to be identical to the current pitch gain
-        float pitch_factor = 0.25 * ctx->pitch_gain[0] * ctx->pitch_gain[0];
-
-        for (i = 0; i < AMRWB_SFR_SIZE; i++)
-            excitation[i] += pitch_factor * ctx->pitch_vector[i];
-
-        ff_scale_vector_to_given_sum_of_squares(excitation, excitation,
-                                                energy, AMRWB_SFR_SIZE);
-    }
-
-    ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
-                                 AMRWB_SFR_SIZE, LP_ORDER);
-}
-
-/**
- * Apply to synthesis a de-emphasis filter of the form:
- * H(z) = 1 / (1 - m * z^-1)
- *
- * @param[out]    out              Output buffer
- * @param[in]     in               Input samples array with in[-1]
- * @param[in]     m                Filter coefficient
- * @param[in,out] mem              State from last filtering
- */
-static void de_emphasis(float *out, float *in, float m, float mem[1])
-{
-    int i;
-
-    out[0] = in[0] + m * mem[0];
-
-    for (i = 1; i < AMRWB_SFR_SIZE; i++)
-         out[i] = in[i] + out[i - 1] * m;
-
-    mem[0] = out[AMRWB_SFR_SIZE - 1];
-}
-
-/**
- * Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using
- * a FIR interpolation filter. Uses past data from before *in address.
- *
- * @param[out] out                 Buffer for interpolated signal
- * @param[in]  in                  Current signal data (length 0.8*o_size)
- * @param[in]  o_size              Output signal length
- * @param[in] ctx                  The context
- */
-static void upsample_5_4(float *out, const float *in, int o_size, CELPMContext *ctx)
-{
-    const float *in0 = in - UPS_FIR_SIZE + 1;
-    int i, j, k;
-    int int_part = 0, frac_part;
-
-    i = 0;
-    for (j = 0; j < o_size / 5; j++) {
-        out[i] = in[int_part];
-        frac_part = 4;
-        i++;
-
-        for (k = 1; k < 5; k++) {
-            out[i] = ctx->dot_productf(in0 + int_part,
-                                                  upsample_fir[4 - frac_part],
-                                                  UPS_MEM_SIZE);
-            int_part++;
-            frac_part--;
-            i++;
-        }
-    }
-}
-
-/**
- * Calculate the high-band gain based on encoded index (23k85 mode) or
- * on the low-band speech signal and the Voice Activity Detection flag.
- *
- * @param[in] ctx                  The context
- * @param[in] synth                LB speech synthesis at 12.8k
- * @param[in] hb_idx               Gain index for mode 23k85 only
- * @param[in] vad                  VAD flag for the frame
- */
-static float find_hb_gain(AMRWBContext *ctx, const float *synth,
-                          uint16_t hb_idx, uint8_t vad)
-{
-    int wsp = (vad > 0);
-    float tilt;
-
-    if (ctx->fr_cur_mode == MODE_23k85)
-        return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
-
-    tilt = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
-           ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);
-
-    /* return gain bounded by [0.1, 1.0] */
-    return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0);
-}
-
-/**
- * Generate the high-band excitation with the same energy from the lower
- * one and scaled by the given gain.
- *
- * @param[in]  ctx                 The context
- * @param[out] hb_exc              Buffer for the excitation
- * @param[in]  synth_exc           Low-band excitation used for synthesis
- * @param[in]  hb_gain             Wanted excitation gain
- */
-static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
-                                 const float *synth_exc, float hb_gain)
-{
-    int i;
-    float energy = ctx->celpm_ctx.dot_productf(synth_exc, synth_exc,
-                                                AMRWB_SFR_SIZE);
-
-    /* Generate a white-noise excitation */
-    for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
-        hb_exc[i] = 32768.0 - (uint16_t) av_lfg_get(&ctx->prng);
-
-    ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc,
-                                            energy * hb_gain * hb_gain,
-                                            AMRWB_SFR_SIZE_16k);
-}
-
-/**
- * Calculate the auto-correlation for the ISF difference vector.
- */
-static float auto_correlation(float *diff_isf, float mean, int lag)
-{
-    int i;
-    float sum = 0.0;
-
-    for (i = 7; i < LP_ORDER - 2; i++) {
-        float prod = (diff_isf[i] - mean) * (diff_isf[i - lag] - mean);
-        sum += prod * prod;
-    }
-    return sum;
-}
-
-/**
- * Extrapolate a ISF vector to the 16kHz range (20th order LP)
- * used at mode 6k60 LP filter for the high frequency band.
- *
- * @param[out] isf Buffer for extrapolated isf; contains LP_ORDER
- *                 values on input
- */
-static void extrapolate_isf(float isf[LP_ORDER_16k])
-{
-    float diff_isf[LP_ORDER - 2], diff_mean;
-    float corr_lag[3];
-    float est, scale;
-    int i, j, i_max_corr;
-
-    isf[LP_ORDER_16k - 1] = isf[LP_ORDER - 1];
-
-    /* Calculate the difference vector */
-    for (i = 0; i < LP_ORDER - 2; i++)
-        diff_isf[i] = isf[i + 1] - isf[i];
-
-    diff_mean = 0.0;
-    for (i = 2; i < LP_ORDER - 2; i++)
-        diff_mean += diff_isf[i] * (1.0f / (LP_ORDER - 4));
-
-    /* Find which is the maximum autocorrelation */
-    i_max_corr = 0;
-    for (i = 0; i < 3; i++) {
-        corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);
-
-        if (corr_lag[i] > corr_lag[i_max_corr])
-            i_max_corr = i;
-    }
-    i_max_corr++;
-
-    for (i = LP_ORDER - 1; i < LP_ORDER_16k - 1; i++)
-        isf[i] = isf[i - 1] + isf[i - 1 - i_max_corr]
-                            - isf[i - 2 - i_max_corr];
-
-    /* Calculate an estimate for ISF(18) and scale ISF based on the error */
-    est   = 7965 + (isf[2] - isf[3] - isf[4]) / 6.0;
-    scale = 0.5 * (FFMIN(est, 7600) - isf[LP_ORDER - 2]) /
-            (isf[LP_ORDER_16k - 2] - isf[LP_ORDER - 2]);
-
-    for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
-        diff_isf[j] = scale * (isf[i] - isf[i - 1]);
-
-    /* Stability insurance */
-    for (i = 1; i < LP_ORDER_16k - LP_ORDER; i++)
-        if (diff_isf[i] + diff_isf[i - 1] < 5.0) {
-            if (diff_isf[i] > diff_isf[i - 1]) {
-                diff_isf[i - 1] = 5.0 - diff_isf[i];
-            } else
-                diff_isf[i] = 5.0 - diff_isf[i - 1];
-        }
-
-    for (i = LP_ORDER - 1, j = 0; i < LP_ORDER_16k - 1; i++, j++)
-        isf[i] = isf[i - 1] + diff_isf[j] * (1.0f / (1 << 15));
-
-    /* Scale the ISF vector for 16000 Hz */
-    for (i = 0; i < LP_ORDER_16k - 1; i++)
-        isf[i] *= 0.8;
-}
-
-/**
- * Spectral expand the LP coefficients using the equation:
- *   y[i] = x[i] * (gamma ** i)
- *
- * @param[out] out                 Output buffer (may use input array)
- * @param[in]  lpc                 LP coefficients array
- * @param[in]  gamma               Weighting factor
- * @param[in]  size                LP array size
- */
-static void lpc_weighting(float *out, const float *lpc, float gamma, int size)
-{
-    int i;
-    float fac = gamma;
-
-    for (i = 0; i < size; i++) {
-        out[i] = lpc[i] * fac;
-        fac   *= gamma;
-    }
-}
-
-/**
- * Conduct 20th order linear predictive coding synthesis for the high
- * frequency band excitation at 16kHz.
- *
- * @param[in]     ctx              The context
- * @param[in]     subframe         Current subframe index (0 to 3)
- * @param[in,out] samples          Pointer to the output speech samples
- * @param[in]     exc              Generated white-noise scaled excitation
- * @param[in]     isf              Current frame isf vector
- * @param[in]     isf_past         Past frame final isf vector
- */
-static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples,
-                         const float *exc, const float *isf, const float *isf_past)
-{
-    float hb_lpc[LP_ORDER_16k];
-    enum Mode mode = ctx->fr_cur_mode;
-
-    if (mode == MODE_6k60) {
-        float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation
-        double e_isp[LP_ORDER_16k];
-
-        ctx->acelpv_ctx.weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe],
-                                1.0 - isfp_inter[subframe], LP_ORDER);
-
-        extrapolate_isf(e_isf);
-
-        e_isf[LP_ORDER_16k - 1] *= 2.0;
-        ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k);
-        ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k);
-
-        lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k);
-    } else {
-        lpc_weighting(hb_lpc, ctx->lp_coef[subframe], 0.6, LP_ORDER);
-    }
-
-    ctx->celpf_ctx.celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k,
-                                 (mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER);
-}
-
-/**
- * Apply a 15th order filter to high-band samples.
- * The filter characteristic depends on the given coefficients.
- *
- * @param[out]    out              Buffer for filtered output
- * @param[in]     fir_coef         Filter coefficients
- * @param[in,out] mem              State from last filtering (updated)
- * @param[in]     in               Input speech data (high-band)
- *
- * @remark It is safe to pass the same array in in and out parameters
- */
-
-#ifndef hb_fir_filter
-static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1],
-                          float mem[HB_FIR_SIZE], const float *in)
-{
-    int i, j;
-    float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples
-
-    memcpy(data, mem, HB_FIR_SIZE * sizeof(float));
-    memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float));
-
-    for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) {
-        out[i] = 0.0;
-        for (j = 0; j <= HB_FIR_SIZE; j++)
-            out[i] += data[i + j] * fir_coef[j];
-    }
-
-    memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float));
-}
-#endif /* hb_fir_filter */
-
-/**
- * Update context state before the next subframe.
- */
-static void update_sub_state(AMRWBContext *ctx)
-{
-    memmove(&ctx->excitation_buf[0], &ctx->excitation_buf[AMRWB_SFR_SIZE],
-            (AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float));
-
-    memmove(&ctx->pitch_gain[1], &ctx->pitch_gain[0], 5 * sizeof(float));
-    memmove(&ctx->fixed_gain[1], &ctx->fixed_gain[0], 1 * sizeof(float));
-
-    memmove(&ctx->samples_az[0], &ctx->samples_az[AMRWB_SFR_SIZE],
-            LP_ORDER * sizeof(float));
-    memmove(&ctx->samples_up[0], &ctx->samples_up[AMRWB_SFR_SIZE],
-            UPS_MEM_SIZE * sizeof(float));
-    memmove(&ctx->samples_hb[0], &ctx->samples_hb[AMRWB_SFR_SIZE_16k],
-            LP_ORDER_16k * sizeof(float));
-}
-
-static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
-                              int *got_frame_ptr, AVPacket *avpkt)
-{
-    AMRWBContext *ctx  = avctx->priv_data;
-    AVFrame *frame     = data;
-    AMRWBFrame   *cf   = &ctx->frame;
-    const uint8_t *buf = avpkt->data;
-    int buf_size       = avpkt->size;
-    int expected_fr_size, header_size;
-    float *buf_out;
-    float spare_vector[AMRWB_SFR_SIZE];      // extra stack space to hold result from anti-sparseness processing
-    float fixed_gain_factor;                 // fixed gain correction factor (gamma)
-    float *synth_fixed_vector;               // pointer to the fixed vector that synthesis should use
-    float synth_fixed_gain;                  // the fixed gain that synthesis should use
-    float voice_fac, stab_fac;               // parameters used for gain smoothing
-    float synth_exc[AMRWB_SFR_SIZE];         // post-processed excitation for synthesis
-    float hb_exc[AMRWB_SFR_SIZE_16k];        // excitation for the high frequency band
-    float hb_samples[AMRWB_SFR_SIZE_16k];    // filtered high-band samples from synthesis
-    float hb_gain;
-    int sub, i, ret;
-
-    /* get output buffer */
-    frame->nb_samples = 4 * AMRWB_SFR_SIZE_16k;
-    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
-        return ret;
-    buf_out = (float *)frame->data[0];
-
-    header_size      = decode_mime_header(ctx, buf);
-    if (ctx->fr_cur_mode > MODE_SID) {
-        av_log(avctx, AV_LOG_ERROR,
-               "Invalid mode %d\n", ctx->fr_cur_mode);
-        return AVERROR_INVALIDDATA;
-    }
-    expected_fr_size = ((cf_sizes_wb[ctx->fr_cur_mode] + 7) >> 3) + 1;
-
-    if (buf_size < expected_fr_size) {
-        av_log(avctx, AV_LOG_ERROR,
-            "Frame too small (%d bytes). Truncated file?\n", buf_size);
-        *got_frame_ptr = 0;
-        return AVERROR_INVALIDDATA;
-    }
-
-    if (!ctx->fr_quality || ctx->fr_cur_mode > MODE_SID)
-        av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");
-
-    if (ctx->fr_cur_mode == MODE_SID) { /* Comfort noise frame */
-        avpriv_request_sample(avctx, "SID mode");
-        return AVERROR_PATCHWELCOME;
-    }
-
-    ff_amr_bit_reorder((uint16_t *) &ctx->frame, sizeof(AMRWBFrame),
-        buf + header_size, amr_bit_orderings_by_mode[ctx->fr_cur_mode]);
-
-    /* Decode the quantized ISF vector */
-    if (ctx->fr_cur_mode == MODE_6k60) {
-        decode_isf_indices_36b(cf->isp_id, ctx->isf_cur);
-    } else {
-        decode_isf_indices_46b(cf->isp_id, ctx->isf_cur);
-    }
-
-    isf_add_mean_and_past(ctx->isf_cur, ctx->isf_q_past);
-    ff_set_min_dist_lsf(ctx->isf_cur, MIN_ISF_SPACING, LP_ORDER - 1);
-
-    stab_fac = stability_factor(ctx->isf_cur, ctx->isf_past_final);
-
-    ctx->isf_cur[LP_ORDER - 1] *= 2.0;
-    ff_acelp_lsf2lspd(ctx->isp[3], ctx->isf_cur, LP_ORDER);
-
-    /* Generate a ISP vector for each subframe */
-    if (ctx->first_frame) {
-        ctx->first_frame = 0;
-        memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(double));
-    }
-    interpolate_isp(ctx->isp, ctx->isp_sub4_past);
-
-    for (sub = 0; sub < 4; sub++)
-        ff_amrwb_lsp2lpc(ctx->isp[sub], ctx->lp_coef[sub], LP_ORDER);
-
-    for (sub = 0; sub < 4; sub++) {
-        const AMRWBSubFrame *cur_subframe = &cf->subframe[sub];
-        float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;
-
-        /* Decode adaptive codebook (pitch vector) */
-        decode_pitch_vector(ctx, cur_subframe, sub);
-        /* Decode innovative codebook (fixed vector) */
-        decode_fixed_vector(ctx->fixed_vector, cur_subframe->pul_ih,
-                            cur_subframe->pul_il, ctx->fr_cur_mode);
-
-        pitch_sharpening(ctx, ctx->fixed_vector);
-
-        decode_gains(cur_subframe->vq_gain, ctx->fr_cur_mode,
-                     &fixed_gain_factor, &ctx->pitch_gain[0]);
-
-        ctx->fixed_gain[0] =
-            ff_amr_set_fixed_gain(fixed_gain_factor,
-                                  ctx->celpm_ctx.dot_productf(ctx->fixed_vector,
-                                                               ctx->fixed_vector,
-                                                               AMRWB_SFR_SIZE) /
-                                  AMRWB_SFR_SIZE,
-                       ctx->prediction_error,
-                       ENERGY_MEAN, energy_pred_fac);
-
-        /* Calculate voice factor and store tilt for next subframe */
-        voice_fac      = voice_factor(ctx->pitch_vector, ctx->pitch_gain[0],
-                                      ctx->fixed_vector, ctx->fixed_gain[0],
-                                      &ctx->celpm_ctx);
-        ctx->tilt_coef = voice_fac * 0.25 + 0.25;
-
-        /* Construct current excitation */
-        for (i = 0; i < AMRWB_SFR_SIZE; i++) {
-            ctx->excitation[i] *= ctx->pitch_gain[0];
-            ctx->excitation[i] += ctx->fixed_gain[0] * ctx->fixed_vector[i];
-            ctx->excitation[i] = truncf(ctx->excitation[i]);
-        }
-
-        /* Post-processing of excitation elements */
-        synth_fixed_gain = noise_enhancer(ctx->fixed_gain[0], &ctx->prev_tr_gain,
-                                          voice_fac, stab_fac);
-
-        synth_fixed_vector = anti_sparseness(ctx, ctx->fixed_vector,
-                                             spare_vector);
-
-        pitch_enhancer(synth_fixed_vector, voice_fac);
-
-        synthesis(ctx, ctx->lp_coef[sub], synth_exc, synth_fixed_gain,
-                  synth_fixed_vector, &ctx->samples_az[LP_ORDER]);
-
-        /* Synthesis speech post-processing */
-        de_emphasis(&ctx->samples_up[UPS_MEM_SIZE],
-                    &ctx->samples_az[LP_ORDER], PREEMPH_FAC, ctx->demph_mem);
-
-        ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(&ctx->samples_up[UPS_MEM_SIZE],
-            &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles,
-            hpf_31_gain, ctx->hpf_31_mem, AMRWB_SFR_SIZE);
-
-        upsample_5_4(sub_buf, &ctx->samples_up[UPS_FIR_SIZE],
-                     AMRWB_SFR_SIZE_16k, &ctx->celpm_ctx);
-
-        /* High frequency band (6.4 - 7.0 kHz) generation part */
-        ctx->acelpf_ctx.acelp_apply_order_2_transfer_function(hb_samples,
-            &ctx->samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles,
-            hpf_400_gain, ctx->hpf_400_mem, AMRWB_SFR_SIZE);
-
-        hb_gain = find_hb_gain(ctx, hb_samples,
-                               cur_subframe->hb_gain, cf->vad);
-
-        scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain);
-
-        hb_synthesis(ctx, sub, &ctx->samples_hb[LP_ORDER_16k],
-                     hb_exc, ctx->isf_cur, ctx->isf_past_final);
-
-        /* High-band post-processing filters */
-        hb_fir_filter(hb_samples, bpf_6_7_coef, ctx->bpf_6_7_mem,
-                      &ctx->samples_hb[LP_ORDER_16k]);
-
-        if (ctx->fr_cur_mode == MODE_23k85)
-            hb_fir_filter(hb_samples, lpf_7_coef, ctx->lpf_7_mem,
-                          hb_samples);
-
-        /* Add the low and high frequency bands */
-        for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
-            sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15));
-
-        /* Update buffers and history */
-        update_sub_state(ctx);
-    }
-
-    /* update state for next frame */
-    memcpy(ctx->isp_sub4_past, ctx->isp[3], LP_ORDER * sizeof(ctx->isp[3][0]));
-    memcpy(ctx->isf_past_final, ctx->isf_cur, LP_ORDER * sizeof(float));
-
-    *got_frame_ptr = 1;
-
-    return expected_fr_size;
-}
-
-AVCodec ff_amrwb_decoder = {
-    .name           = "amrwb",
-    .long_name      = NULL_IF_CONFIG_SMALL("AMR-WB (Adaptive Multi-Rate WideBand)"),
-    .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = AV_CODEC_ID_AMR_WB,
-    .priv_data_size = sizeof(AMRWBContext),
-    .init           = amrwb_decode_init,
-    .decode         = amrwb_decode_frame,
-    .capabilities   = CODEC_CAP_DR1,
-    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
-                                                     AV_SAMPLE_FMT_NONE },
-};