/*

 * QCELP decoder

 * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet

 *

 * 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

 * QCELP decoder

 * @author Reynaldo H. Verdejo Pinochet

 * @remark FFmpeg merging spearheaded by Kenan Gillet

 * @remark Development mentored by Benjamin Larson

 */

#include "libavutil/avassert.h"

#include "libavutil/channel_layout.h"

#include "libavutil/float_dsp.h"

#include "avcodec.h"

#include "codec_internal.h"

#include "decode.h"

#include "get_bits.h"

#include "qcelpdata.h"

#include "celp_filters.h"

#include "acelp_filters.h"

#include "acelp_vectors.h"

#include "lsp.h"

typedef enum {

    I_F_Q = -1,    /**< insufficient frame quality */

    SILENCE,

    RATE_OCTAVE,

    RATE_QUARTER,

    RATE_HALF,

    RATE_FULL

} qcelp_packet_rate;

typedef struct QCELPContext {

    GetBitContext     gb;

    qcelp_packet_rate bitrate;

    QCELPFrame        frame;    /**< unpacked data frame */

    uint8_t  erasure_count;

    uint8_t  octave_count;      /**< count the consecutive RATE_OCTAVE frames */

    float    prev_lspf[10];

    float    predictor_lspf[10];/**< LSP predictor for RATE_OCTAVE and I_F_Q */

    float    pitch_synthesis_filter_mem[303];

    float    pitch_pre_filter_mem[303];

    float    rnd_fir_filter_mem[180];

    float    formant_mem[170];

    float    last_codebook_gain;

    int      prev_g1[2];

    int      prev_bitrate;

    float    pitch_gain[4];

    uint8_t  pitch_lag[4];

    uint16_t first16bits;

    uint8_t  warned_buf_mismatch_bitrate;

    /* postfilter */

    float    postfilter_synth_mem[10];

    float    postfilter_agc_mem;

    float    postfilter_tilt_mem;

} QCELPContext;

/**

 * Initialize the speech codec according to the specification.

 *

 * TIA/EIA/IS-733 2.4.9

 */

static av_cold int qcelp_decode_init(AVCodecContext *avctx)

{

    QCELPContext *q = avctx->priv_data;

    int i;

    av_channel_layout_uninit(&avctx->ch_layout);

    avctx->ch_layout      = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;

    avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;

    if (!avctx->sample_rate)

        avctx->sample_rate = 8000;

    for (i = 0; i < 10; i++)

        q->prev_lspf[i] = (i + 1) / 11.0;

    return 0;

}

/**

 * Decode the 10 quantized LSP frequencies from the LSPV/LSP

 * transmission codes of any bitrate and check for badly received packets.

 *

 * @param q the context

 * @param lspf line spectral pair frequencies

 *

 * @return 0 on success, -1 if the packet is badly received

 *

 * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3

 */

static int decode_lspf(QCELPContext *q, float *lspf)

{

    int i;

    float tmp_lspf, smooth, erasure_coeff;

    const float *predictors;

    if (q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q) {

        predictors = q->prev_bitrate != RATE_OCTAVE &&

                     q->prev_bitrate != I_F_Q ? q->prev_lspf

                                              : q->predictor_lspf;

        if (q->bitrate == RATE_OCTAVE) {

            q->octave_count++;

            for (i = 0; i < 10; i++) {

                q->predictor_lspf[i] =

                             lspf[i] = (q->frame.lspv[i] ?  QCELP_LSP_SPREAD_FACTOR

                                                         : -QCELP_LSP_SPREAD_FACTOR) +

                                        predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR   +

                                        (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR) / 11);

            }

            smooth = q->octave_count < 10 ? .875 : 0.1;

        } else {

            erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;

            av_assert2(q->bitrate == I_F_Q);

            if (q->erasure_count > 1)

                erasure_coeff *= q->erasure_count < 4 ? 0.9 : 0.7;

            for (i = 0; i < 10; i++) {

                q->predictor_lspf[i] =

                             lspf[i] = (i + 1) * (1 - erasure_coeff) / 11 +

                                       erasure_coeff * predictors[i];

            }

            smooth = 0.125;

        }

        // Check the stability of the LSP frequencies.

        lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);

        for (i = 1; i < 10; i++)

            lspf[i] = FFMAX(lspf[i], lspf[i - 1] + QCELP_LSP_SPREAD_FACTOR);

        lspf[9] = FFMIN(lspf[9], 1.0 - QCELP_LSP_SPREAD_FACTOR);

        for (i = 9; i > 0; i--)

            lspf[i - 1] = FFMIN(lspf[i - 1], lspf[i] - QCELP_LSP_SPREAD_FACTOR);

        // Low-pass filter the LSP frequencies.

        ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0 - smooth, 10);

    } else {

        q->octave_count = 0;

        tmp_lspf = 0.0;

        for (i = 0; i < 5; i++) {

            lspf[2 * i + 0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;

            lspf[2 * i + 1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;

        }

        // Check for badly received packets.

        if (q->bitrate == RATE_QUARTER) {

            if (lspf[9] <= .70 || lspf[9] >= .97)

                return -1;

            for (i = 3; i < 10; i++)

                if (fabs(lspf[i] - lspf[i - 2]) < .08)

                    return -1;

        } else {

            if (lspf[9] <= .66 || lspf[9] >= .985)

                return -1;

            for (i = 4; i < 10; i++)

                if (fabs(lspf[i] - lspf[i - 4]) < .0931)

                    return -1;

        }

    }

    return 0;

}

/**

 * Convert codebook transmission codes to GAIN and INDEX.

 *

 * @param q the context

 * @param gain array holding the decoded gain

 *

 * TIA/EIA/IS-733 2.4.6.2

 */

static void decode_gain_and_index(QCELPContext *q, float *gain)

{

    int i, subframes_count, g1[16];

    float slope;

    if (q->bitrate >= RATE_QUARTER) {

        switch (q->bitrate) {

        case RATE_FULL: subframes_count = 16; break;

        case RATE_HALF: subframes_count =  4; break;

        default:        subframes_count =  5;

        }

        for (i = 0; i < subframes_count; i++) {

            g1[i] = 4 * q->frame.cbgain[i];

            if (q->bitrate == RATE_FULL && !((i + 1) & 3)) {

                g1[i] += av_clip((g1[i - 1] + g1[i - 2] + g1[i - 3]) / 3 - 6, 0, 32);

            }

            gain[i] = qcelp_g12ga[g1[i]];

            if (q->frame.cbsign[i]) {

                gain[i] = -gain[i];

                q->frame.cindex[i] = (q->frame.cindex[i] - 89) & 127;

            }

        }

        q->prev_g1[0]         = g1[i - 2];

        q->prev_g1[1]         = g1[i - 1];

        q->last_codebook_gain = qcelp_g12ga[g1[i - 1]];

        if (q->bitrate == RATE_QUARTER) {

            // Provide smoothing of the unvoiced excitation energy.

            gain[7] =       gain[4];

            gain[6] = 0.4 * gain[3] + 0.6 * gain[4];

            gain[5] =       gain[3];

            gain[4] = 0.8 * gain[2] + 0.2 * gain[3];

            gain[3] = 0.2 * gain[1] + 0.8 * gain[2];

            gain[2] =       gain[1];

            gain[1] = 0.6 * gain[0] + 0.4 * gain[1];

        }

    } else if (q->bitrate != SILENCE) {

        if (q->bitrate == RATE_OCTAVE) {

            g1[0] = 2 * q->frame.cbgain[0] +

                    av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);

            subframes_count = 8;

        } else {

            av_assert2(q->bitrate == I_F_Q);

            g1[0] = q->prev_g1[1];

            switch (q->erasure_count) {

            case 1 : break;

            case 2 : g1[0] -= 1; break;

            case 3 : g1[0] -= 2; break;

            default: g1[0] -= 6;

            }

            if (g1[0] < 0)

                g1[0] = 0;

            subframes_count = 4;

        }

        // This interpolation is done to produce smoother background noise.

        slope = 0.5 * (qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;

        for (i = 1; i <= subframes_count; i++)

                gain[i - 1] = q->last_codebook_gain + slope * i;

        q->last_codebook_gain = gain[i - 2];

        q->prev_g1[0]         = q->prev_g1[1];

        q->prev_g1[1]         = g1[0];

    }

}

/**

 * If the received packet is Rate 1/4 a further sanity check is made of the

 * codebook gain.

 *

 * @param cbgain the unpacked cbgain array

 * @return -1 if the sanity check fails, 0 otherwise

 *

 * TIA/EIA/IS-733 2.4.8.7.3

 */

static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)

{

    int i, diff, prev_diff = 0;

    for (i = 1; i < 5; i++) {

        diff = cbgain[i] - cbgain[i-1];

        if (FFABS(diff) > 10)

            return -1;

        else if (FFABS(diff - prev_diff) > 12)

            return -1;

        prev_diff = diff;

    }

    return 0;

}

/**

 * Compute the scaled codebook vector Cdn From INDEX and GAIN

 * for all rates.

 *

 * The specification lacks some information here.

 *

 * TIA/EIA/IS-733 has an omission on the codebook index determination

 * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says

 * you have to subtract the decoded index parameter from the given scaled

 * codebook vector index 'n' to get the desired circular codebook index, but

 * it does not mention that you have to clamp 'n' to [0-9] in order to get

 * RI-compliant results.

 *

 * The reason for this mistake seems to be the fact they forgot to mention you

 * have to do these calculations per codebook subframe and adjust given

 * equation values accordingly.

 *

 * @param q the context

 * @param gain array holding the 4 pitch subframe gain values

 * @param cdn_vector array for the generated scaled codebook vector

 */

static void compute_svector(QCELPContext *q, const float *gain,

                            float *cdn_vector)

{

    int i, j, k;

    uint16_t cbseed, cindex;

    float *rnd, tmp_gain, fir_filter_value;

    switch (q->bitrate) {

    case RATE_FULL:

        for (i = 0; i < 16; i++) {

            tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;

            cindex   = -q->frame.cindex[i];

            for (j = 0; j < 10; j++)

                *cdn_vector++ = tmp_gain *

                                qcelp_rate_full_codebook[cindex++ & 127];

        }

        break;

    case RATE_HALF:

        for (i = 0; i < 4; i++) {

            tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;

            cindex   = -q->frame.cindex[i];

            for (j = 0; j < 40; j++)

                *cdn_vector++ = tmp_gain *

                                qcelp_rate_half_codebook[cindex++ & 127];

        }

        break;

    case RATE_QUARTER:

        cbseed = (0x0003 & q->frame.lspv[4]) << 14 |

                 (0x003F & q->frame.lspv[3]) <<  8 |

                 (0x0060 & q->frame.lspv[2]) <<  1 |

                 (0x0007 & q->frame.lspv[1]) <<  3 |

                 (0x0038 & q->frame.lspv[0]) >>  3;

        rnd    = q->rnd_fir_filter_mem + 20;

        for (i = 0; i < 8; i++) {

            tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);

            for (k = 0; k < 20; k++) {

                cbseed = 521 * cbseed + 259;

                *rnd   = (int16_t) cbseed;

                    // FIR filter

                fir_filter_value = 0.0;

                for (j = 0; j < 10; j++)

                    fir_filter_value += qcelp_rnd_fir_coefs[j] *

                                        (rnd[-j] + rnd[-20+j]);

                fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];

                *cdn_vector++     = tmp_gain * fir_filter_value;

                rnd++;

            }

        }

        memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160,

               20 * sizeof(float));

        break;

    case RATE_OCTAVE:

        cbseed = q->first16bits;

        for (i = 0; i < 8; i++) {

            tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);

            for (j = 0; j < 20; j++) {

                cbseed        = 521 * cbseed + 259;

                *cdn_vector++ = tmp_gain * (int16_t) cbseed;

            }

        }

        break;

    case I_F_Q:

        cbseed = -44; // random codebook index

        for (i = 0; i < 4; i++) {

            tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;

            for (j = 0; j < 40; j++)

                *cdn_vector++ = tmp_gain *

                                qcelp_rate_full_codebook[cbseed++ & 127];

        }

        break;

    case SILENCE:

        memset(cdn_vector, 0, 160 * sizeof(float));

        break;

    }

}

/**

 * Apply generic gain control.

 *

 * @param v_out output vector

 * @param v_in gain-controlled vector

 * @param v_ref vector to control gain of

 *

 * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6

 */

static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in)

{

    int i;

    for (i = 0; i < 160; i += 40) {

        float res = ff_scalarproduct_float_c(v_ref + i, v_ref + i, 40);

        ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i, res, 40);

    }

}

/**

 * Apply filter in pitch-subframe steps.

 *

 * @param memory buffer for the previous state of the filter

 *        - must be able to contain 303 elements

 *        - the 143 first elements are from the previous state

 *        - the next 160 are for output

 * @param v_in input filter vector

 * @param gain per-subframe gain array, each element is between 0.0 and 2.0

 * @param lag per-subframe lag array, each element is

 *        - between 16 and 143 if its corresponding pfrac is 0,

 *        - between 16 and 139 otherwise

 * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0

 *        otherwise

 *

 * @return filter output vector

 */

static const float *do_pitchfilter(float memory[303], const float v_in[160],

                                   const float gain[4], const uint8_t *lag,

                                   const uint8_t pfrac[4])

{

    int i, j;

    float *v_lag, *v_out;

    const float *v_len;

    v_out = memory + 143; // Output vector starts at memory[143].

    for (i = 0; i < 4; i++) {

        if (gain[i]) {

            v_lag = memory + 143 + 40 * i - lag[i];

            for (v_len = v_in + 40; v_in < v_len; v_in++) {

                if (pfrac[i]) { // If it is a fractional lag...

                    for (j = 0, *v_out = 0.0; j < 4; j++)

                        *v_out += qcelp_hammsinc_table[j] *

                                  (v_lag[j - 4] + v_lag[3 - j]);

                } else

                    *v_out = *v_lag;

                *v_out = *v_in + gain[i] * *v_out;

                v_lag++;

                v_out++;

            }

        } else {

            memcpy(v_out, v_in, 40 * sizeof(float));

            v_in  += 40;

            v_out += 40;

        }

    }

    memmove(memory, memory + 160, 143 * sizeof(float));

    return memory + 143;

}

/**

 * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.

 * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2

 *

 * @param q the context

 * @param cdn_vector the scaled codebook vector

 */

static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)

{

    int i;

    const float *v_synthesis_filtered, *v_pre_filtered;

    if (q->bitrate >= RATE_HALF || q->bitrate == SILENCE ||

        (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF))) {

        if (q->bitrate >= RATE_HALF) {

            // Compute gain & lag for the whole frame.

            for (i = 0; i < 4; i++) {

                q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;

                q->pitch_lag[i] = q->frame.plag[i] + 16;

            }

        } else {

            float max_pitch_gain;

            if (q->bitrate == I_F_Q) {

                  if (q->erasure_count < 3)

                      max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);

                  else

                      max_pitch_gain = 0.0;

            } else {

                av_assert2(q->bitrate == SILENCE);

                max_pitch_gain = 1.0;

            }

            for (i = 0; i < 4; i++)

                q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);

            memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));

        }

        // pitch synthesis filter

        v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,

                                              cdn_vector, q->pitch_gain,

                                              q->pitch_lag, q->frame.pfrac);

        // pitch prefilter update

        for (i = 0; i < 4; i++)

            q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);

        v_pre_filtered       = do_pitchfilter(q->pitch_pre_filter_mem,

                                              v_synthesis_filtered,

                                              q->pitch_gain, q->pitch_lag,

                                              q->frame.pfrac);

        apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);

    } else {

        memcpy(q->pitch_synthesis_filter_mem,

               cdn_vector + 17, 143 * sizeof(float));

        memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));

        memset(q->pitch_gain, 0, sizeof(q->pitch_gain));

        memset(q->pitch_lag,  0, sizeof(q->pitch_lag));

    }

}

/**

 * Reconstruct LPC coefficients from the line spectral pair frequencies

 * and perform bandwidth expansion.

 *

 * @param lspf line spectral pair frequencies

 * @param lpc linear predictive coding coefficients

 *

 * @note: bandwidth_expansion_coeff could be precalculated into a table

 *        but it seems to be slower on x86

 *

 * TIA/EIA/IS-733 2.4.3.3.5

 */

static void lspf2lpc(const float *lspf, float *lpc)

{

    double lsp[10];

    double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;

    int i;

    for (i = 0; i < 10; i++)

        lsp[i] = cos(M_PI * lspf[i]);

    ff_acelp_lspd2lpc(lsp, lpc, 5);

    for (i = 0; i < 10; i++) {

        lpc[i]                    *= bandwidth_expansion_coeff;

        bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;

    }

}

/**

 * Interpolate LSP frequencies and compute LPC coefficients

 * for a given bitrate & pitch subframe.

 *

 * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2

 *

 * @param q the context

 * @param curr_lspf LSP frequencies vector of the current frame

 * @param lpc float vector for the resulting LPC

 * @param subframe_num frame number in decoded stream

 */

static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,

                            float *lpc, const int subframe_num)

{

    float interpolated_lspf[10];

    float weight;

    if (q->bitrate >= RATE_QUARTER)

        weight = 0.25 * (subframe_num + 1);

    else if (q->bitrate == RATE_OCTAVE && !subframe_num)

        weight = 0.625;

    else

        weight = 1.0;

    if (weight != 1.0) {

        ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,

                                weight, 1.0 - weight, 10);

        lspf2lpc(interpolated_lspf, lpc);

    } else if (q->bitrate >= RATE_QUARTER ||

               (q->bitrate == I_F_Q && !subframe_num))

        lspf2lpc(curr_lspf, lpc);

    else if (q->bitrate == SILENCE && !subframe_num)

        lspf2lpc(q->prev_lspf, lpc);

}

static qcelp_packet_rate buf_size2bitrate(const int buf_size)

{

    switch (buf_size) {

    case 35: return RATE_FULL;

    case 17: return RATE_HALF;

    case  8: return RATE_QUARTER;

    case  4: return RATE_OCTAVE;

    case  1: return SILENCE;

    }

    return I_F_Q;

}

/**

 * Determine the bitrate from the frame size and/or the first byte of the frame.

 *

 * @param avctx the AV codec context

 * @param buf_size length of the buffer

 * @param buf the buffer

 *

 * @return the bitrate on success,

 *         I_F_Q  if the bitrate cannot be satisfactorily determined

 *

 * TIA/EIA/IS-733 2.4.8.7.1

 */

static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx,

                                           const int buf_size,

                                           const uint8_t **buf)

{

    qcelp_packet_rate bitrate;

    if ((bitrate = buf_size2bitrate(buf_size)) >= 0) {

        if (bitrate > **buf) {

            QCELPContext *q = avctx->priv_data;

            if (!q->warned_buf_mismatch_bitrate) {

            av_log(avctx, AV_LOG_WARNING,

                   "Claimed bitrate and buffer size mismatch.\n");

                q->warned_buf_mismatch_bitrate = 1;

            }

            bitrate = **buf;

        } else if (bitrate < **buf) {

            av_log(avctx, AV_LOG_ERROR,

                   "Buffer is too small for the claimed bitrate.\n");

            return I_F_Q;

        }

        (*buf)++;

    } else if ((bitrate = buf_size2bitrate(buf_size + 1)) >= 0) {

        av_log(avctx, AV_LOG_WARNING,

               "Bitrate byte missing, guessing bitrate from packet size.\n");

    } else

        return I_F_Q;

    if (bitrate == SILENCE) {

        // FIXME: Remove this warning when tested with samples.

        avpriv_request_sample(avctx, "Blank frame handling");

    }

    return bitrate;

}

static void warn_insufficient_frame_quality(AVCodecContext *avctx,

                                            const char *message)

{

    av_log(avctx, AV_LOG_WARNING, "Frame #%"PRId64", IFQ: %s\n",

           avctx->frame_num, message);

}

static void postfilter(QCELPContext *q, float *samples, float *lpc)

{

    static const float pow_0_775[10] = {

        0.775000, 0.600625, 0.465484, 0.360750, 0.279582,

        0.216676, 0.167924, 0.130141, 0.100859, 0.078166

    }, pow_0_625[10] = {

        0.625000, 0.390625, 0.244141, 0.152588, 0.095367,

        0.059605, 0.037253, 0.023283, 0.014552, 0.009095

    };

    float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];

    int n;

    for (n = 0; n < 10; n++) {

        lpc_s[n] = lpc[n] * pow_0_625[n];

        lpc_p[n] = lpc[n] * pow_0_775[n];

    }

    ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,

                                      q->formant_mem + 10, 160, 10);

    memcpy(pole_out, q->postfilter_synth_mem, sizeof(float) * 10);

    ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);

    memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);

    ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);

    ff_adaptive_gain_control(samples, pole_out + 10,

                             ff_scalarproduct_float_c(q->formant_mem + 10,

                                                      q->formant_mem + 10,

                                                      160),

                             160, 0.9375, &q->postfilter_agc_mem);

}

static int qcelp_decode_frame(AVCodecContext *avctx, AVFrame *frame,

                              int *got_frame_ptr, AVPacket *avpkt)

{

    const uint8_t *buf = avpkt->data;

    int buf_size       = avpkt->size;

    QCELPContext *q    = avctx->priv_data;

    float *outbuffer;

    int   i, ret;

    float quantized_lspf[10], lpc[10];

    float gain[16];

    float *formant_mem;

    /* get output buffer */

    frame->nb_samples = 160;

    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)

        return ret;

    outbuffer = (float *)frame->data[0];

    if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) {

        warn_insufficient_frame_quality(avctx, "Bitrate cannot be determined.");

        goto erasure;

    }

    if (q->bitrate == RATE_OCTAVE &&

        (q->first16bits = AV_RB16(buf)) == 0xFFFF) {

        warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");

        goto erasure;

    }

    if (q->bitrate > SILENCE) {

        const QCELPBitmap *bitmaps     = qcelp_unpacking_bitmaps_per_rate[q->bitrate];

        const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate] +

                                         qcelp_unpacking_bitmaps_lengths[q->bitrate];

        uint8_t *unpacked_data         = (uint8_t *)&q->frame;

        if ((ret = init_get_bits8(&q->gb, buf, buf_size)) < 0)

            return ret;

        memset(&q->frame, 0, sizeof(QCELPFrame));

        for (; bitmaps < bitmaps_end; bitmaps++)

            unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;

        // Check for erasures/blanks on rates 1, 1/4 and 1/8.

        if (q->frame.reserved) {

            warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");

            goto erasure;

        }

        if (q->bitrate == RATE_QUARTER &&

            codebook_sanity_check_for_rate_quarter(q->frame.cbgain)) {

            warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");

            goto erasure;

        }

        if (q->bitrate >= RATE_HALF) {

            for (i = 0; i < 4; i++) {

                if (q->frame.pfrac[i] && q->frame.plag[i] >= 124) {

                    warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");

                    goto erasure;

                }

            }

        }

    }

    decode_gain_and_index(q, gain);

    compute_svector(q, gain, outbuffer);

    if (decode_lspf(q, quantized_lspf) < 0) {

        warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");

        goto erasure;

    }

    apply_pitch_filters(q, outbuffer);

    if (q->bitrate == I_F_Q) {

erasure:

        q->bitrate = I_F_Q;

        q->erasure_count++;

        decode_gain_and_index(q, gain);

        compute_svector(q, gain, outbuffer);

        decode_lspf(q, quantized_lspf);

        apply_pitch_filters(q, outbuffer);

    } else

        q->erasure_count = 0;

    formant_mem = q->formant_mem + 10;

    for (i = 0; i < 4; i++) {

        interpolate_lpc(q, quantized_lspf, lpc, i);

        ff_celp_lp_synthesis_filterf(formant_mem, lpc,

                                     outbuffer + i * 40, 40, 10);

        formant_mem += 40;

    }

    // postfilter, as per TIA/EIA/IS-733 2.4.8.6

    postfilter(q, outbuffer, lpc);

    memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));

    memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));

    q->prev_bitrate  = q->bitrate;

    *got_frame_ptr = 1;

    return buf_size;

}

const FFCodec ff_qcelp_decoder = {

    .p.name         = "qcelp",

    CODEC_LONG_NAME("QCELP / PureVoice"),

    .p.type         = AVMEDIA_TYPE_AUDIO,

    .p.id           = AV_CODEC_ID_QCELP,

    .init           = qcelp_decode_init,

    FF_CODEC_DECODE_CB(qcelp_decode_frame),

    .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,

    .priv_data_size = sizeof(QCELPContext),

};