PeerStreamer

/*

 * 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 qcelpdec.c

 * QCELP decoder

 * @author Reynaldo H. Verdejo Pinochet

 * @remark FFmpeg merging spearheaded by Kenan Gillet

 * @remark Development mentored by Benjamin Larson

 */

#include <stddef.h>

#include "avcodec.h"

#include "internal.h"

#include "bitstream.h"

#include "qcelpdata.h"

#include "celp_math.h"

#include "celp_filters.h"

#undef NDEBUG

#include <assert.h>

typedef enum

{

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

    SILENCE,

    RATE_OCTAVE,

    RATE_QUARTER,

    RATE_HALF,

    RATE_FULL

} qcelp_packet_rate;

typedef struct

{

    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;

} QCELPContext;

/**

 * Reconstructs LPC coefficients from the line spectral pair frequencies.

 *

 * TIA/EIA/IS-733 2.4.3.3.5

 */

void ff_qcelp_lspf2lpc(const float *lspf, float *lpc);

static void weighted_vector_sumf(float *out, const float *in_a,

                                 const float *in_b, float weight_coeff_a,

                                 float weight_coeff_b, int length)

{

    int i;

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

        out[i] = weight_coeff_a * in_a[i]

               + weight_coeff_b * in_b[i];

}

/**

 * 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;

    avctx->sample_fmt = SAMPLE_FMT_FLT;

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

        q->prev_lspf[i] = (i+1)/11.;

    return 0;

}

/**

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

 * transmission codes of any bitrate and checks 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;

            assert(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.

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

    }else

    {

        q->octave_count = 0;

        tmp_lspf = 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;

}

/**

 * Converts 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 == 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

        {

            assert(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;

}

/**

 * Computes 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;

    }

}

/**

 * Apply generic gain control.

 *

 * @param v_out output vector

 * @param v_in gain-controlled vector

 * @param v_ref vector to control gain of

 *

 * FIXME: If v_ref is a zero vector, it energy is zero

 *        and the behavior of the gain control is

 *        undefined in the specs.

 *

 * TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6

 */

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

                            const float *v_in)

{

    int   i, j, len;

    float scalefactor;

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

    {

        scalefactor = ff_dot_productf(v_in + j, v_in + j, 40);

        if(scalefactor)

            scalefactor = sqrt(ff_dot_productf(v_ref + j, v_ref + j, 40)

                        / scalefactor);

        else

            ff_log_missing_feature(NULL, "Zero energy for gain control", 1);

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

            v_out[j] = scalefactor * v_in[j];

    }

}

/**

 * 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.; 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

 *

 * @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 == 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->erasure_count < 3)

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

             else

                max_pitch_gain = 0.0;

            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));

    }

}

/**

 * Interpolates LSP frequencies and computes LPC coefficients

 * for a given bitrate & pitch subframe.

 *

 * TIA/EIA/IS-733 2.4.3.3.4

 *

 * @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

 */

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)

    {

        weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,

                             weight, 1.0 - weight, 10);

        ff_qcelp_lspf2lpc(interpolated_lspf, lpc);

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

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

        ff_qcelp_lspf2lpc(curr_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 bufffer

 *

 * @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 int 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 is missing, guessing the bitrate from packet size.\n");

    }else

        return I_F_Q;

    if(bitrate == SILENCE)

    {

        // FIXME: the decoder should not handle SILENCE frames as I_F_Q frames

        ff_log_missing_feature(avctx, "Blank frame", 1);

        bitrate = I_F_Q;

    }

    return bitrate;

}

static void warn_insufficient_frame_quality(AVCodecContext *avctx,

                                            const char *message)

{

    av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,

           message);

}

static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size,

                              const uint8_t *buf, int buf_size)

{

    QCELPContext *q = avctx->priv_data;

    float *outbuffer = data;

    int   i;

    float quantized_lspf[10], lpc[10];

    float gain[16];

    float *formant_mem;

    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;

        init_get_bits(&q->gb, buf, 8*buf_size);

        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;

    }

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

    // FIXME: postfilter and final gain control should be here.

    // TIA/EIA/IS-733 2.4.8.6

    formant_mem = q->formant_mem + 10;

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

        *outbuffer++ = av_clipf(*formant_mem++, QCELP_CLIP_LOWER_BOUND,

                                QCELP_CLIP_UPPER_BOUND);

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

    q->prev_bitrate = q->bitrate;

    *data_size = 160 * sizeof(*outbuffer);

    return *data_size;

}

AVCodec qcelp_decoder =

{

    .name   = "qcelp",

    .type   = CODEC_TYPE_AUDIO,

    .id     = CODEC_ID_QCELP,

    .init   = qcelp_decode_init,

    .decode = qcelp_decode_frame,

    .priv_data_size = sizeof(QCELPContext),

    .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),

};