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/*

 * G.729, G729 Annex D decoders

 * 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 

#include 

#include "avcodec.h"

#include "libavutil/avutil.h"

#include "get_bits.h"

#include "audiodsp.h"

#include "internal.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"

#include "g729postfilter.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 {

    AudioDSPContext adsp;

    /// 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];

    /// residual signal buffer (used in long-term postfilter)

    int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];

    /// previous speech data for residual calculation filter

    int16_t res_filter_data[SUBFRAME_SIZE+10];

    /// previous speech data for short-term postfilter

    int16_t pos_filter_data[SUBFRAME_SIZE+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)

    int16_t ht_prev_data;       ///< previous data for 4.2.3, equation 86

    int gain_coeff;             ///< (1.14) gain coefficient (4.2.4)

    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

    /// (14.14) high-pass filter data (past input)

    int hpf_f[2];

    /// high-pass filter data (past output)

    int16_t hpf_z[2];

}  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[out] lsfq (2.13) quantized LSF coefficients

 * @param[in,out] past_quantizer_outputs (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[in,out] lsfq (2.13) quantized LSF coefficients

 * @param[in,out] past_quantizer_outputs (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

 */

static 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

    {

        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

 */

static 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 int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)

{

    int res = 0;

    while (order--)

        res += *v1++ * *v2++;

    return res;

}

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);

    }

    avctx->sample_fmt = AV_SAMPLE_FMT_S16;

    /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */

    avctx->frame_size = SUBFRAME_SIZE << 1;

    ctx->gain_coeff = 16384; // 1.0 in (1.14)

    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];

    ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;

    /* 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)

    ff_audiodsp_init(&ctx->adsp);

    ctx->adsp.scalarproduct_int16 = scalarproduct_int16_c;

    return 0;

}

static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,

                        AVPacket *avpkt)

{

    const uint8_t *buf = avpkt->data;

    int buf_size       = avpkt->size;

    int16_t *out_frame;

    GetBitContext gb;

    const 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[2];      // 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, ret;

    int gain_before, gain_after;

    int is_periodic = 0;         // whether one of the subframes is declared as periodic or not

    AVFrame *frame = data;

    frame->nb_samples = SUBFRAME_SIZE<<1;

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

        return ret;

    out_frame = (int16_t*) frame->data[0];

    if (buf_size % 10 == 0) {

        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, 8*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[i]  = (pitch_delay_3x + 1) / 3;

        if (pitch_delay_int[i] > PITCH_DELAY_MAX) {

            av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);

            pitch_delay_int[i] = PITCH_DELAY_MAX;

        }

        if (frame_erasure) {

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

            fc_indexes   = av_mod_uintp2(ctx->rand_value, format->fc_indexes_bits);

            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[i],

                                     fc + pitch_delay_int[i],

                                     fc, 1 << 14,

                                     av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),

                                     0, 14,

                                     SUBFRAME_SIZE - pitch_delay_int[i]);

        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 occur 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->adsp, 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 divided 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));

        if (ff_celp_lp_synthesis_filter(

            synth+10,

            &lp[i][1],

            ctx->exc  + i * SUBFRAME_SIZE,

            SUBFRAME_SIZE,

            10,

            1,

            0,

            0x800))

            /* Overflow occurred, 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,

                    0,

                    0x800);

        } else {

            ff_celp_lp_synthesis_filter(

                    synth+10,

                    &lp[i][1],

                    ctx->exc  + i * SUBFRAME_SIZE,

                    SUBFRAME_SIZE,

                    10,

                    0,

                    0,

                    0x800);

        }

        /* Save data (without postfilter) for use in next subframe. */

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

        /* Calculate gain of unfiltered signal for use in AGC. */

        gain_before = 0;

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

            gain_before += FFABS(synth[j+10]);

        /* Call postfilter and also update voicing decision for use in next frame. */

        ff_g729_postfilter(

                &ctx->adsp,

                &ctx->ht_prev_data,

                &is_periodic,

                &lp[i][0],

                pitch_delay_int[0],

                ctx->residual,

                ctx->res_filter_data,

                ctx->pos_filter_data,

                synth+10,

                SUBFRAME_SIZE);

        /* Calculate gain of filtered signal for use in AGC. */

        gain_after = 0;

        for(j=0; j

            gain_after += FFABS(synth[j+10]);

        ctx->gain_coeff = ff_g729_adaptive_gain_control(

                gain_before,

                gain_after,

                synth+10,

                SUBFRAME_SIZE,

                ctx->gain_coeff);

        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[i];

        memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));

        ff_acelp_high_pass_filter(

                out_frame + i*SUBFRAME_SIZE,

                ctx->hpf_f,

                synth+10,

                SUBFRAME_SIZE);

        memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*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));

    *got_frame_ptr = 1;

    return packet_type == FORMAT_G729_8K ? 10 : 8;

}

AVCodec ff_g729_decoder = {

    .name           = "g729",

    .long_name      = NULL_IF_CONFIG_SMALL("G.729"),

    .type           = AVMEDIA_TYPE_AUDIO,

    .id             = AV_CODEC_ID_G729,

    .priv_data_size = sizeof(G729Context),

    .init           = decoder_init,

    .decode         = decode_frame,

    .capabilities   = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,

};