ffmpeg/libavcodec/g729dec.c

/*
 * G.729 decoder
 * Copyright (c) 2008 Vladimir Voroshilov
 *
 * 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
 */
#include <stdlib.h>
#include <inttypes.h>
#include <limits.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include <assert.h>

#include "avcodec.h"
#include "libavutil/avutil.h"
#include "get_bits.h"
#include "dsputil.h"

#include "g729.h"
#include "lsp.h"
#include "celp_math.h"
#include "celp_filters.h"
#include "acelp_filters.h"
#include "acelp_pitch_delay.h"
#include "acelp_vectors.h"
#include "g729data.h"

/**
 * minimum quantized LSF value (3.2.4)
 * 0.005 in Q13
 */
#define LSFQ_MIN                   40

/**
 * maximum quantized LSF value (3.2.4)
 * 3.135 in Q13
 */
#define LSFQ_MAX                   25681

/**
 * minimum LSF distance (3.2.4)
 * 0.0391 in Q13
 */
#define LSFQ_DIFF_MIN              321

/// interpolation filter length
#define INTERPOL_LEN              11

/**
 * minimum gain pitch value (3.8, Equation 47)
 * 0.2 in (1.14)
 */
#define SHARP_MIN                  3277

/**
 * maximum gain pitch value (3.8, Equation 47)
 * (EE) This does not comply with the specification.
 * Specification says about 0.8, which should be
 * 13107 in (1.14), but reference C code uses
 * 13017 (equals to 0.7945) instead of it.
 */
#define SHARP_MAX                  13017

/**
 * MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26  * subframe_size) in (7.13)
 */
#define MR_ENERGY 1018156

#define DECISION_NOISE        0
#define DECISION_INTERMEDIATE 1
#define DECISION_VOICE        2

typedef enum {
    FORMAT_G729_8K = 0,
    FORMAT_G729D_6K4,
    FORMAT_COUNT,
} G729Formats;

typedef struct {
    uint8_t ac_index_bits[2];   ///< adaptive codebook index for second subframe (size in bits)
    uint8_t parity_bit;         ///< parity bit for pitch delay
    uint8_t gc_1st_index_bits;  ///< gain codebook (first stage) index (size in bits)
    uint8_t gc_2nd_index_bits;  ///< gain codebook (second stage) index (size in bits)
    uint8_t fc_signs_bits;      ///< number of pulses in fixed-codebook vector
    uint8_t fc_indexes_bits;    ///< size (in bits) of fixed-codebook index entry
} G729FormatDescription;

typedef struct {
    DSPContext dsp;

    /// past excitation signal buffer
    int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];

    int16_t* exc;               ///< start of past excitation data in buffer
    int pitch_delay_int_prev;   ///< integer part of previous subframe's pitch delay (4.1.3)

    /// (2.13) LSP quantizer outputs
    int16_t  past_quantizer_output_buf[MA_NP + 1][10];
    int16_t* past_quantizer_outputs[MA_NP + 1];

    int16_t lsfq[10];           ///< (2.13) quantized LSF coefficients from previous frame
    int16_t lsp_buf[2][10];     ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
    int16_t *lsp[2];            ///< pointers to lsp_buf

    int16_t quant_energy[4];    ///< (5.10) past quantized energy

    /// previous speech data for LP synthesis filter
    int16_t syn_filter_data[10];

    /// (1.14) pitch gain of current and five previous subframes
    int16_t past_gain_pitch[6];

    /// (14.1) gain code from current and previous subframe
    int16_t past_gain_code[2];

    /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
    int16_t voice_decision;

    int16_t onset;              ///< detected onset level (0-2)
    int16_t was_periodic;       ///< whether previous frame was declared as periodic or not (4.4)
    uint16_t rand_value;        ///< random number generator value (4.4.4)
    int ma_predictor_prev;      ///< switched MA predictor of LSP quantizer from last good frame
}  G729Context;

static const G729FormatDescription format_g729_8k = {
    .ac_index_bits     = {8,5},
    .parity_bit        = 1,
    .gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
    .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
    .fc_signs_bits     = 4,
    .fc_indexes_bits   = 13,
};

static const G729FormatDescription format_g729d_6k4 = {
    .ac_index_bits     = {8,4},
    .parity_bit        = 0,
    .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
    .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
    .fc_signs_bits     = 2,
    .fc_indexes_bits   = 9,
};

/**
 * @brief pseudo random number generator
 */
static inline uint16_t g729_prng(uint16_t value)
{
    return 31821 * value + 13849;
}

/**
 * Get parity bit of bit 2..7
 */
static inline int get_parity(uint8_t value)
{
   return (0x6996966996696996ULL >> (value >> 2)) & 1;
}

/*
 * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
 * @param lsfq [out] (2.13) quantized LSF coefficients
 * @param past_quantizer_outputs [in/out] (2.13) quantizer outputs from previous frames
 * @param ma_predictor switched MA predictor of LSP quantizer
 * @param vq_1st first stage vector of quantizer
 * @param vq_2nd_low second stage lower vector of LSP quantizer
 * @param vq_2nd_high second stage higher vector of LSP quantizer
 */
static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
                       int16_t ma_predictor,
                       int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
{
    int i,j;
    static const uint8_t min_distance[2]={10, 5}; //(2.13)
    int16_t* quantizer_output = past_quantizer_outputs[MA_NP];

    for (i = 0; i < 5; i++) {
        quantizer_output[i]     = cb_lsp_1st[vq_1st][i    ] + cb_lsp_2nd[vq_2nd_low ][i    ];
        quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
    }

    for (j = 0; j < 2; j++) {
        for (i = 1; i < 10; i++) {
            int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
            if (diff > 0) {
                quantizer_output[i - 1] -= diff;
                quantizer_output[i    ] += diff;
            }
        }
    }

    for (i = 0; i < 10; i++) {
        int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
        for (j = 0; j < MA_NP; j++)
            sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];

        lsfq[i] = sum >> 15;
    }

    ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
}

/**
 * Restores past LSP quantizer output using LSF from previous frame
 * @param lsfq [in/out] (2.13) quantized LSF coefficients
 * @param past_quantizer_outputs [in/out] (2.13) quantizer outputs from previous frames
 * @param ma_predictor_prev MA predictor from previous frame
 * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
 */
static void lsf_restore_from_previous(int16_t* lsfq,
                                      int16_t* past_quantizer_outputs[MA_NP + 1],
                                      int ma_predictor_prev)
{
    int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
    int i,k;

    for (i = 0; i < 10; i++) {
        int tmp = lsfq[i] << 15;

        for (k = 0; k < MA_NP; k++)
            tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];

        quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
    }
}

/**
 * Constructs new excitation signal and applies phase filter to it
 * @param out[out] constructed speech signal
 * @param in original excitation signal
 * @param fc_cur (2.13) original fixed-codebook vector
 * @param gain_code (14.1) gain code
 * @param subframe_size length of the subframe
 */
void g729d_get_new_exc(
        int16_t* out,
        const int16_t* in,
        const int16_t* fc_cur,
        int dstate,
        int gain_code,
        int subframe_size)
{
    int i;
    int16_t fc_new[SUBFRAME_SIZE];

    ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);

    for(i=0; i<subframe_size; i++)
    {
        out[i]  = in[i];
        out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
        out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
    }
}

/**
 * Makes decision about onset in current subframe
 * @param past_onset decision result of previous subframe
 * @param past_gain_code gain code of current and previous subframe
 *
 * @return onset decision result for current subframe
 */
int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
{
    if((past_gain_code[0] >> 1) > past_gain_code[1])
        return 2;
    else
        return FFMAX(past_onset-1, 0);
}

/**
 * Makes decision about voice presence in current subframe
 * @param onset onset level
 * @param prev_voice_decision voice decision result from previous subframe
 * @param past_gain_pitch pitch gain of current and previous subframes
 *
 * @return voice decision result for current subframe
 */
static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
{
    int i, low_gain_pitch_cnt, voice_decision;

    if(past_gain_pitch[0] >= 14745)      // 0.9
        voice_decision = DECISION_VOICE;
    else if (past_gain_pitch[0] <= 9830) // 0.6
        voice_decision = DECISION_NOISE;
    else
        voice_decision = DECISION_INTERMEDIATE;

    for(i=0, low_gain_pitch_cnt=0; i<6; i++)
        if(past_gain_pitch[i] < 9830)
            low_gain_pitch_cnt++;

    if(low_gain_pitch_cnt > 2 && !onset)
        voice_decision = DECISION_NOISE;

    if(!onset && voice_decision > prev_voice_decision + 1)
        voice_decision--;

    if(onset && voice_decision < DECISION_VOICE)
        voice_decision++;

    return voice_decision;
}

static av_cold int decoder_init(AVCodecContext * avctx)
{
    G729Context* ctx = avctx->priv_data;
    int i,k;

    if (avctx->channels != 1) {
        av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels);
        return AVERROR(EINVAL);
    }

    /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
    avctx->frame_size = SUBFRAME_SIZE << 1;

    for (k = 0; k < MA_NP + 1; k++) {
        ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
        for (i = 1; i < 11; i++)
            ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
    }

    ctx->lsp[0] = ctx->lsp_buf[0];
    ctx->lsp[1] = ctx->lsp_buf[1];
    memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));

    ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];

    /* random seed initialization */
    ctx->rand_value = 21845;

    /* quantized prediction error */
    for(i=0; i<4; i++)
        ctx->quant_energy[i] = -14336; // -14 in (5.10)

    dsputil_init(&ctx->dsp, avctx);

    return 0;
}

static int decode_frame(AVCodecContext *avctx, void *data, int *data_size,
                        AVPacket *avpkt)
{
    const uint8_t *buf = avpkt->data;
    int buf_size       = avpkt->size;
    int16_t *out_frame = data;
    GetBitContext gb;
    G729FormatDescription format;
    int frame_erasure = 0;    ///< frame erasure detected during decoding
    int bad_pitch = 0;        ///< parity check failed
    int i;
    int16_t *tmp;
    G729Formats packet_type;
    G729Context *ctx = avctx->priv_data;
    int16_t lp[2][11];           // (3.12)
    uint8_t ma_predictor;     ///< switched MA predictor of LSP quantizer
    uint8_t quantizer_1st;    ///< first stage vector of quantizer
    uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
    uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)

    int pitch_delay_int;         // pitch delay, integer part
    int pitch_delay_3x;          // pitch delay, multiplied by 3
    int16_t fc[SUBFRAME_SIZE];   // fixed-codebook vector
    int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
    int j;
    int is_periodic = 0;         // whether one of the subframes is declared as periodic or not

    if (*data_size < SUBFRAME_SIZE << 2) {
        av_log(avctx, AV_LOG_ERROR, "Error processing packet: output buffer too small\n");
        return AVERROR(EIO);
    }

    if (buf_size == 10) {
        packet_type = FORMAT_G729_8K;
        format = format_g729_8k;
        //Reset voice decision
        ctx->onset = 0;
        ctx->voice_decision = DECISION_VOICE;
        av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
    } else if (buf_size == 8) {
        packet_type = FORMAT_G729D_6K4;
        format = format_g729d_6k4;
        av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
    } else {
        av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
        return AVERROR_INVALIDDATA;
    }

    for (i=0; i < buf_size; i++)
        frame_erasure |= buf[i];
    frame_erasure = !frame_erasure;

    init_get_bits(&gb, buf, buf_size);

    ma_predictor     = get_bits(&gb, 1);
    quantizer_1st    = get_bits(&gb, VQ_1ST_BITS);
    quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
    quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);

    if(frame_erasure)
        lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
                                  ctx->ma_predictor_prev);
    else {
        lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
                   ma_predictor,
                   quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
        ctx->ma_predictor_prev = ma_predictor;
    }

    tmp = ctx->past_quantizer_outputs[MA_NP];
    memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
            MA_NP * sizeof(int16_t*));
    ctx->past_quantizer_outputs[0] = tmp;

    ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);

    ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);

    FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);

    for (i = 0; i < 2; i++) {
        int gain_corr_factor;

        uint8_t ac_index;      ///< adaptive codebook index
        uint8_t pulses_signs;  ///< fixed-codebook vector pulse signs
        int fc_indexes;        ///< fixed-codebook indexes
        uint8_t gc_1st_index;  ///< gain codebook (first stage) index
        uint8_t gc_2nd_index;  ///< gain codebook (second stage) index

        ac_index      = get_bits(&gb, format.ac_index_bits[i]);
        if(!i && format.parity_bit)
            bad_pitch = get_parity(ac_index) == get_bits1(&gb);
        fc_indexes    = get_bits(&gb, format.fc_indexes_bits);
        pulses_signs  = get_bits(&gb, format.fc_signs_bits);
        gc_1st_index  = get_bits(&gb, format.gc_1st_index_bits);
        gc_2nd_index  = get_bits(&gb, format.gc_2nd_index_bits);

        if (frame_erasure)
            pitch_delay_3x   = 3 * ctx->pitch_delay_int_prev;
        else if(!i) {
            if (bad_pitch)
                pitch_delay_3x   = 3 * ctx->pitch_delay_int_prev;
            else
                pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
        } else {
            int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
                                          PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);

            if(packet_type == FORMAT_G729D_6K4)
                pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
            else
                pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
        }

        /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
        pitch_delay_int  = (pitch_delay_3x + 1) / 3;

        if (frame_erasure) {
            ctx->rand_value = g729_prng(ctx->rand_value);
            fc_indexes   = ctx->rand_value & ((1 << format.fc_indexes_bits) - 1);

            ctx->rand_value = g729_prng(ctx->rand_value);
            pulses_signs = ctx->rand_value;
        }


        memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
        switch (packet_type) {
            case FORMAT_G729_8K:
                ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
                                            ff_fc_4pulses_8bits_track_4,
                                            fc_indexes, pulses_signs, 3, 3);
                break;
            case FORMAT_G729D_6K4:
                ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
                                            ff_fc_2pulses_9bits_track2_gray,
                                            fc_indexes, pulses_signs, 1, 4);
                break;
        }

        /*
          This filter enhances harmonic components of the fixed-codebook vector to
          improve the quality of the reconstructed speech.

                     / fc_v[i],                                    i < pitch_delay
          fc_v[i] = <
                     \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
        */
        ff_acelp_weighted_vector_sum(fc + pitch_delay_int,
                                     fc + pitch_delay_int,
                                     fc, 1 << 14,
                                     av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
                                     0, 14,
                                     SUBFRAME_SIZE - pitch_delay_int);

        memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
        ctx->past_gain_code[1] = ctx->past_gain_code[0];

        if (frame_erasure) {
            ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
            ctx->past_gain_code[0]  = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)

            gain_corr_factor = 0;
        } else {
            if (packet_type == FORMAT_G729D_6K4) {
                ctx->past_gain_pitch[0]  = cb_gain_1st_6k4[gc_1st_index][0] +
                                           cb_gain_2nd_6k4[gc_2nd_index][0];
                gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
                                   cb_gain_2nd_6k4[gc_2nd_index][1];

                /* Without check below overflow can occure in ff_acelp_update_past_gain.
                   It is not issue for G.729, because gain_corr_factor in it's case is always
                   greater than 1024, while in G.729D it can be even zero. */
                gain_corr_factor = FFMAX(gain_corr_factor, 1024);
#ifndef G729_BITEXACT
                gain_corr_factor >>= 1;
#endif
            } else {
                ctx->past_gain_pitch[0]  = cb_gain_1st_8k[gc_1st_index][0] +
                                           cb_gain_2nd_8k[gc_2nd_index][0];
                gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
                                   cb_gain_2nd_8k[gc_2nd_index][1];
            }

            /* Decode the fixed-codebook gain. */
            ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&ctx->dsp, gain_corr_factor,
                                                               fc, MR_ENERGY,
                                                               ctx->quant_energy,
                                                               ma_prediction_coeff,
                                                               SUBFRAME_SIZE, 4);
#ifdef G729_BITEXACT
            /*
              This correction required to get bit-exact result with
              reference code, because gain_corr_factor in G.729D is
              two times larger than in original G.729.

              If bit-exact result is not issue then gain_corr_factor
              can be simpler devided by 2 before call to g729_get_gain_code
              instead of using correction below.
            */
            if (packet_type == FORMAT_G729D_6K4) {
                gain_corr_factor >>= 1;
                ctx->past_gain_code[0] >>= 1;
            }
#endif
        }
        ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);

        /* Routine requires rounding to lowest. */
        ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
                             ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
                             ff_acelp_interp_filter, 6,
                             (pitch_delay_3x % 3) << 1,
                             10, SUBFRAME_SIZE);

        ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
                                     ctx->exc + i * SUBFRAME_SIZE, fc,
                                     (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
                                     ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
                                     1 << 13, 14, SUBFRAME_SIZE);

        memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));

        /* Temporary synth buffer is required since filter needs additional space at top of buffer and, thus,
           synthesis can not be done directly to output buffer. This buffer will be reused by future
           postprocessing filters. */
        if (ff_celp_lp_synthesis_filter(
            synth+10,
            &lp[i][1],
            ctx->exc  + i * SUBFRAME_SIZE,
            SUBFRAME_SIZE,
            10,
            1,
            0x800))
            /* Overflow occured, downscale excitation signal... */
            for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
                ctx->exc_base[j] >>= 2;

        /* ... and make synthesis again. */
        if (packet_type == FORMAT_G729D_6K4) {
            int16_t exc_new[SUBFRAME_SIZE];

            ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
            ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);

            g729d_get_new_exc(exc_new, ctx->exc  + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);

            ff_celp_lp_synthesis_filter(
                    synth+10,
                    &lp[i][1],
                    exc_new,
                    SUBFRAME_SIZE,
                    10,
                    0,
                    0x800);
        } else {
            ff_celp_lp_synthesis_filter(
                    synth+10,
                    &lp[i][1],
                    ctx->exc  + i * SUBFRAME_SIZE,
                    SUBFRAME_SIZE,
                    10,
                    0,
                    0x800);
        }
        /* Save data (without postfilter) for use in next subframe. */
        memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));

        if (frame_erasure)
            ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
        else
            ctx->pitch_delay_int_prev = pitch_delay_int;

        /* Dumb. Will be replaced by high-pass filter */
        memcpy(out_frame + i * SUBFRAME_SIZE, synth + 10, SUBFRAME_SIZE * sizeof(int16_t));
    }

    ctx->was_periodic = is_periodic;

    /* Save signal for use in next frame. */
    memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));

    *data_size = SUBFRAME_SIZE << 2;
    return buf_size;
}

AVCodec ff_g729_decoder =
{
    "g729",
    AVMEDIA_TYPE_AUDIO,
    CODEC_ID_G729,
    sizeof(G729Context),
    decoder_init,
    NULL,
    NULL,
    decode_frame,
    .long_name = NULL_IF_CONFIG_SMALL("G.729"),
};