annotate qcelpdec.c @ 11034:fd5921186064 libavcodec

Make the fast loop filter path work with unavailable left MBs. This prevents the issue with having to switch between slow and fast code paths in each row. 0.5% faster loopfilter for cathedral

/*
 * 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 libavcodec/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 "get_bits.h"

#include "qcelpdata.h"

#include "celp_math.h"
#include "celp_filters.h"
#include "acelp_vectors.h"
#include "lsp.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;

/**
 * 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.
        ff_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 != 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
        {
            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;
        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)
        ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i,
                                                ff_dot_productf(v_ref + i,
                                                                v_ref + i, 40),
                                                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.; 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
            {
                assert(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));
    }
}

/**
 * Reconstructs LPC coefficients from the line spectral pair frequencies
 * and performs 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;
    }
}

/**
 * Interpolates LSP frequencies and computes 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 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 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 is missing, guessing the bitrate from packet size.\n");
    }else
        return I_F_Q;

    if(bitrate == SILENCE)
    {
        //FIXME: Remove experimental warning when tested with samples.
        av_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");
    }
    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,
                              AVPacket *avpkt)
{
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->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"),
};