/*

  * 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 <inttypes.h>
 #include <string.h>
 
 #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<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
  */

 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) {

         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   = 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[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<SUBFRAME_SIZE; 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 buf_size;
 }
 

 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   = CODEC_CAP_DR1,

 };