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libavcodec/g729dec.c

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00001 /*
00002  * G.729, G729 Annex D decoders
00003  * Copyright (c) 2008 Vladimir Voroshilov
00004  *
00005  * This file is part of FFmpeg.
00006  *
00007  * FFmpeg is free software; you can redistribute it and/or
00008  * modify it under the terms of the GNU Lesser General Public
00009  * License as published by the Free Software Foundation; either
00010  * version 2.1 of the License, or (at your option) any later version.
00011  *
00012  * FFmpeg is distributed in the hope that it will be useful,
00013  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00014  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00015  * Lesser General Public License for more details.
00016  *
00017  * You should have received a copy of the GNU Lesser General Public
00018  * License along with FFmpeg; if not, write to the Free Software
00019  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00020  */
00021 
00022 #include <inttypes.h>
00023 #include <string.h>
00024 
00025 #include "avcodec.h"
00026 #include "libavutil/avutil.h"
00027 #include "get_bits.h"
00028 #include "dsputil.h"
00029 
00030 #include "g729.h"
00031 #include "lsp.h"
00032 #include "celp_math.h"
00033 #include "celp_filters.h"
00034 #include "acelp_filters.h"
00035 #include "acelp_pitch_delay.h"
00036 #include "acelp_vectors.h"
00037 #include "g729data.h"
00038 #include "g729postfilter.h"
00039 
00044 #define LSFQ_MIN                   40
00045 
00050 #define LSFQ_MAX                   25681
00051 
00056 #define LSFQ_DIFF_MIN              321
00057 
00059 #define INTERPOL_LEN              11
00060 
00065 #define SHARP_MIN                  3277
00066 
00074 #define SHARP_MAX                  13017
00075 
00079 #define MR_ENERGY 1018156
00080 
00081 #define DECISION_NOISE        0
00082 #define DECISION_INTERMEDIATE 1
00083 #define DECISION_VOICE        2
00084 
00085 typedef enum {
00086     FORMAT_G729_8K = 0,
00087     FORMAT_G729D_6K4,
00088     FORMAT_COUNT,
00089 } G729Formats;
00090 
00091 typedef struct {
00092     uint8_t ac_index_bits[2];   
00093     uint8_t parity_bit;         
00094     uint8_t gc_1st_index_bits;  
00095     uint8_t gc_2nd_index_bits;  
00096     uint8_t fc_signs_bits;      
00097     uint8_t fc_indexes_bits;    
00098 } G729FormatDescription;
00099 
00100 typedef struct {
00101     DSPContext dsp;
00102     AVFrame frame;
00103 
00105     int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];
00106 
00107     int16_t* exc;               
00108     int pitch_delay_int_prev;   
00109 
00111     int16_t  past_quantizer_output_buf[MA_NP + 1][10];
00112     int16_t* past_quantizer_outputs[MA_NP + 1];
00113 
00114     int16_t lsfq[10];           
00115     int16_t lsp_buf[2][10];     
00116     int16_t *lsp[2];            
00117 
00118     int16_t quant_energy[4];    
00119 
00121     int16_t syn_filter_data[10];
00122 
00123 
00125     int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
00126 
00128     int16_t res_filter_data[SUBFRAME_SIZE+10];
00129 
00131     int16_t pos_filter_data[SUBFRAME_SIZE+10];
00132 
00134     int16_t past_gain_pitch[6];
00135 
00137     int16_t past_gain_code[2];
00138 
00140     int16_t voice_decision;
00141 
00142     int16_t onset;              
00143     int16_t was_periodic;       
00144     int16_t ht_prev_data;       
00145     int gain_coeff;             
00146     uint16_t rand_value;        
00147     int ma_predictor_prev;      
00148 
00150     int hpf_f[2];
00151 
00153     int16_t hpf_z[2];
00154 }  G729Context;
00155 
00156 static const G729FormatDescription format_g729_8k = {
00157     .ac_index_bits     = {8,5},
00158     .parity_bit        = 1,
00159     .gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
00160     .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
00161     .fc_signs_bits     = 4,
00162     .fc_indexes_bits   = 13,
00163 };
00164 
00165 static const G729FormatDescription format_g729d_6k4 = {
00166     .ac_index_bits     = {8,4},
00167     .parity_bit        = 0,
00168     .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
00169     .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
00170     .fc_signs_bits     = 2,
00171     .fc_indexes_bits   = 9,
00172 };
00173 
00177 static inline uint16_t g729_prng(uint16_t value)
00178 {
00179     return 31821 * value + 13849;
00180 }
00181 
00185 static inline int get_parity(uint8_t value)
00186 {
00187    return (0x6996966996696996ULL >> (value >> 2)) & 1;
00188 }
00189 
00190 /*
00191  * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
00192  * @param lsfq [out] (2.13) quantized LSF coefficients
00193  * @param past_quantizer_outputs [in/out] (2.13) quantizer outputs from previous frames
00194  * @param ma_predictor switched MA predictor of LSP quantizer
00195  * @param vq_1st first stage vector of quantizer
00196  * @param vq_2nd_low second stage lower vector of LSP quantizer
00197  * @param vq_2nd_high second stage higher vector of LSP quantizer
00198  */
00199 static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
00200                        int16_t ma_predictor,
00201                        int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
00202 {
00203     int i,j;
00204     static const uint8_t min_distance[2]={10, 5}; //(2.13)
00205     int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
00206 
00207     for (i = 0; i < 5; i++) {
00208         quantizer_output[i]     = cb_lsp_1st[vq_1st][i    ] + cb_lsp_2nd[vq_2nd_low ][i    ];
00209         quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
00210     }
00211 
00212     for (j = 0; j < 2; j++) {
00213         for (i = 1; i < 10; i++) {
00214             int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
00215             if (diff > 0) {
00216                 quantizer_output[i - 1] -= diff;
00217                 quantizer_output[i    ] += diff;
00218             }
00219         }
00220     }
00221 
00222     for (i = 0; i < 10; i++) {
00223         int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
00224         for (j = 0; j < MA_NP; j++)
00225             sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
00226 
00227         lsfq[i] = sum >> 15;
00228     }
00229 
00230     ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
00231 }
00232 
00240 static void lsf_restore_from_previous(int16_t* lsfq,
00241                                       int16_t* past_quantizer_outputs[MA_NP + 1],
00242                                       int ma_predictor_prev)
00243 {
00244     int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
00245     int i,k;
00246 
00247     for (i = 0; i < 10; i++) {
00248         int tmp = lsfq[i] << 15;
00249 
00250         for (k = 0; k < MA_NP; k++)
00251             tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
00252 
00253         quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
00254     }
00255 }
00256 
00265 static void g729d_get_new_exc(
00266         int16_t* out,
00267         const int16_t* in,
00268         const int16_t* fc_cur,
00269         int dstate,
00270         int gain_code,
00271         int subframe_size)
00272 {
00273     int i;
00274     int16_t fc_new[SUBFRAME_SIZE];
00275 
00276     ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
00277 
00278     for(i=0; i<subframe_size; i++)
00279     {
00280         out[i]  = in[i];
00281         out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
00282         out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
00283     }
00284 }
00285 
00293 static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
00294 {
00295     if((past_gain_code[0] >> 1) > past_gain_code[1])
00296         return 2;
00297     else
00298         return FFMAX(past_onset-1, 0);
00299 }
00300 
00309 static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
00310 {
00311     int i, low_gain_pitch_cnt, voice_decision;
00312 
00313     if(past_gain_pitch[0] >= 14745)      // 0.9
00314         voice_decision = DECISION_VOICE;
00315     else if (past_gain_pitch[0] <= 9830) // 0.6
00316         voice_decision = DECISION_NOISE;
00317     else
00318         voice_decision = DECISION_INTERMEDIATE;
00319 
00320     for(i=0, low_gain_pitch_cnt=0; i<6; i++)
00321         if(past_gain_pitch[i] < 9830)
00322             low_gain_pitch_cnt++;
00323 
00324     if(low_gain_pitch_cnt > 2 && !onset)
00325         voice_decision = DECISION_NOISE;
00326 
00327     if(!onset && voice_decision > prev_voice_decision + 1)
00328         voice_decision--;
00329 
00330     if(onset && voice_decision < DECISION_VOICE)
00331         voice_decision++;
00332 
00333     return voice_decision;
00334 }
00335 
00336 static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order, int shift)
00337 {
00338     int res = 0;
00339 
00340     while (order--)
00341         res += (*v1++ * *v2++) >> shift;
00342 
00343     return res;
00344 }
00345 
00346 static av_cold int decoder_init(AVCodecContext * avctx)
00347 {
00348     G729Context* ctx = avctx->priv_data;
00349     int i,k;
00350 
00351     if (avctx->channels != 1) {
00352         av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels);
00353         return AVERROR(EINVAL);
00354     }
00355     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
00356 
00357     /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
00358     avctx->frame_size = SUBFRAME_SIZE << 1;
00359 
00360     ctx->gain_coeff = 16384; // 1.0 in (1.14)
00361 
00362     for (k = 0; k < MA_NP + 1; k++) {
00363         ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
00364         for (i = 1; i < 11; i++)
00365             ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
00366     }
00367 
00368     ctx->lsp[0] = ctx->lsp_buf[0];
00369     ctx->lsp[1] = ctx->lsp_buf[1];
00370     memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
00371 
00372     ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
00373 
00374     /* random seed initialization */
00375     ctx->rand_value = 21845;
00376 
00377     /* quantized prediction error */
00378     for(i=0; i<4; i++)
00379         ctx->quant_energy[i] = -14336; // -14 in (5.10)
00380 
00381     dsputil_init(&ctx->dsp, avctx);
00382     ctx->dsp.scalarproduct_int16 = scalarproduct_int16_c;
00383 
00384     avcodec_get_frame_defaults(&ctx->frame);
00385     avctx->coded_frame = &ctx->frame;
00386 
00387     return 0;
00388 }
00389 
00390 static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
00391                         AVPacket *avpkt)
00392 {
00393     const uint8_t *buf = avpkt->data;
00394     int buf_size       = avpkt->size;
00395     int16_t *out_frame;
00396     GetBitContext gb;
00397     const G729FormatDescription *format;
00398     int frame_erasure = 0;    
00399     int bad_pitch = 0;        
00400     int i;
00401     int16_t *tmp;
00402     G729Formats packet_type;
00403     G729Context *ctx = avctx->priv_data;
00404     int16_t lp[2][11];           // (3.12)
00405     uint8_t ma_predictor;     
00406     uint8_t quantizer_1st;    
00407     uint8_t quantizer_2nd_lo; 
00408     uint8_t quantizer_2nd_hi; 
00409 
00410     int pitch_delay_int[2];      // pitch delay, integer part
00411     int pitch_delay_3x;          // pitch delay, multiplied by 3
00412     int16_t fc[SUBFRAME_SIZE];   // fixed-codebook vector
00413     int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
00414     int j, ret;
00415     int gain_before, gain_after;
00416     int is_periodic = 0;         // whether one of the subframes is declared as periodic or not
00417 
00418     ctx->frame.nb_samples = SUBFRAME_SIZE<<1;
00419     if ((ret = avctx->get_buffer(avctx, &ctx->frame)) < 0) {
00420         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
00421         return ret;
00422     }
00423     out_frame= ctx->frame.data[0];
00424 
00425     if (buf_size == 10) {
00426         packet_type = FORMAT_G729_8K;
00427         format = &format_g729_8k;
00428         //Reset voice decision
00429         ctx->onset = 0;
00430         ctx->voice_decision = DECISION_VOICE;
00431         av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
00432     } else if (buf_size == 8) {
00433         packet_type = FORMAT_G729D_6K4;
00434         format = &format_g729d_6k4;
00435         av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
00436     } else {
00437         av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
00438         return AVERROR_INVALIDDATA;
00439     }
00440 
00441     for (i=0; i < buf_size; i++)
00442         frame_erasure |= buf[i];
00443     frame_erasure = !frame_erasure;
00444 
00445     init_get_bits(&gb, buf, 8*buf_size);
00446 
00447     ma_predictor     = get_bits(&gb, 1);
00448     quantizer_1st    = get_bits(&gb, VQ_1ST_BITS);
00449     quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
00450     quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
00451 
00452     if(frame_erasure)
00453         lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
00454                                   ctx->ma_predictor_prev);
00455     else {
00456         lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
00457                    ma_predictor,
00458                    quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
00459         ctx->ma_predictor_prev = ma_predictor;
00460     }
00461 
00462     tmp = ctx->past_quantizer_outputs[MA_NP];
00463     memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
00464             MA_NP * sizeof(int16_t*));
00465     ctx->past_quantizer_outputs[0] = tmp;
00466 
00467     ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
00468 
00469     ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
00470 
00471     FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
00472 
00473     for (i = 0; i < 2; i++) {
00474         int gain_corr_factor;
00475 
00476         uint8_t ac_index;      
00477         uint8_t pulses_signs;  
00478         int fc_indexes;        
00479         uint8_t gc_1st_index;  
00480         uint8_t gc_2nd_index;  
00481 
00482         ac_index      = get_bits(&gb, format->ac_index_bits[i]);
00483         if(!i && format->parity_bit)
00484             bad_pitch = get_parity(ac_index) == get_bits1(&gb);
00485         fc_indexes    = get_bits(&gb, format->fc_indexes_bits);
00486         pulses_signs  = get_bits(&gb, format->fc_signs_bits);
00487         gc_1st_index  = get_bits(&gb, format->gc_1st_index_bits);
00488         gc_2nd_index  = get_bits(&gb, format->gc_2nd_index_bits);
00489 
00490         if (frame_erasure)
00491             pitch_delay_3x   = 3 * ctx->pitch_delay_int_prev;
00492         else if(!i) {
00493             if (bad_pitch)
00494                 pitch_delay_3x   = 3 * ctx->pitch_delay_int_prev;
00495             else
00496                 pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
00497         } else {
00498             int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
00499                                           PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
00500 
00501             if(packet_type == FORMAT_G729D_6K4)
00502                 pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
00503             else
00504                 pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
00505         }
00506 
00507         /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
00508         pitch_delay_int[i]  = (pitch_delay_3x + 1) / 3;
00509 
00510         if (frame_erasure) {
00511             ctx->rand_value = g729_prng(ctx->rand_value);
00512             fc_indexes   = ctx->rand_value & ((1 << format->fc_indexes_bits) - 1);
00513 
00514             ctx->rand_value = g729_prng(ctx->rand_value);
00515             pulses_signs = ctx->rand_value;
00516         }
00517 
00518 
00519         memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
00520         switch (packet_type) {
00521             case FORMAT_G729_8K:
00522                 ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
00523                                             ff_fc_4pulses_8bits_track_4,
00524                                             fc_indexes, pulses_signs, 3, 3);
00525                 break;
00526             case FORMAT_G729D_6K4:
00527                 ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
00528                                             ff_fc_2pulses_9bits_track2_gray,
00529                                             fc_indexes, pulses_signs, 1, 4);
00530                 break;
00531         }
00532 
00533         /*
00534           This filter enhances harmonic components of the fixed-codebook vector to
00535           improve the quality of the reconstructed speech.
00536 
00537                      / fc_v[i],                                    i < pitch_delay
00538           fc_v[i] = <
00539                      \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
00540         */
00541         ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
00542                                      fc + pitch_delay_int[i],
00543                                      fc, 1 << 14,
00544                                      av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
00545                                      0, 14,
00546                                      SUBFRAME_SIZE - pitch_delay_int[i]);
00547 
00548         memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
00549         ctx->past_gain_code[1] = ctx->past_gain_code[0];
00550 
00551         if (frame_erasure) {
00552             ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
00553             ctx->past_gain_code[0]  = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
00554 
00555             gain_corr_factor = 0;
00556         } else {
00557             if (packet_type == FORMAT_G729D_6K4) {
00558                 ctx->past_gain_pitch[0]  = cb_gain_1st_6k4[gc_1st_index][0] +
00559                                            cb_gain_2nd_6k4[gc_2nd_index][0];
00560                 gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
00561                                    cb_gain_2nd_6k4[gc_2nd_index][1];
00562 
00563                 /* Without check below overflow can occure in ff_acelp_update_past_gain.
00564                    It is not issue for G.729, because gain_corr_factor in it's case is always
00565                    greater than 1024, while in G.729D it can be even zero. */
00566                 gain_corr_factor = FFMAX(gain_corr_factor, 1024);
00567 #ifndef G729_BITEXACT
00568                 gain_corr_factor >>= 1;
00569 #endif
00570             } else {
00571                 ctx->past_gain_pitch[0]  = cb_gain_1st_8k[gc_1st_index][0] +
00572                                            cb_gain_2nd_8k[gc_2nd_index][0];
00573                 gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
00574                                    cb_gain_2nd_8k[gc_2nd_index][1];
00575             }
00576 
00577             /* Decode the fixed-codebook gain. */
00578             ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&ctx->dsp, gain_corr_factor,
00579                                                                fc, MR_ENERGY,
00580                                                                ctx->quant_energy,
00581                                                                ma_prediction_coeff,
00582                                                                SUBFRAME_SIZE, 4);
00583 #ifdef G729_BITEXACT
00584             /*
00585               This correction required to get bit-exact result with
00586               reference code, because gain_corr_factor in G.729D is
00587               two times larger than in original G.729.
00588 
00589               If bit-exact result is not issue then gain_corr_factor
00590               can be simpler devided by 2 before call to g729_get_gain_code
00591               instead of using correction below.
00592             */
00593             if (packet_type == FORMAT_G729D_6K4) {
00594                 gain_corr_factor >>= 1;
00595                 ctx->past_gain_code[0] >>= 1;
00596             }
00597 #endif
00598         }
00599         ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
00600 
00601         /* Routine requires rounding to lowest. */
00602         ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
00603                              ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
00604                              ff_acelp_interp_filter, 6,
00605                              (pitch_delay_3x % 3) << 1,
00606                              10, SUBFRAME_SIZE);
00607 
00608         ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
00609                                      ctx->exc + i * SUBFRAME_SIZE, fc,
00610                                      (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
00611                                      ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
00612                                      1 << 13, 14, SUBFRAME_SIZE);
00613 
00614         memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
00615 
00616         if (ff_celp_lp_synthesis_filter(
00617             synth+10,
00618             &lp[i][1],
00619             ctx->exc  + i * SUBFRAME_SIZE,
00620             SUBFRAME_SIZE,
00621             10,
00622             1,
00623             0,
00624             0x800))
00625             /* Overflow occured, downscale excitation signal... */
00626             for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
00627                 ctx->exc_base[j] >>= 2;
00628 
00629         /* ... and make synthesis again. */
00630         if (packet_type == FORMAT_G729D_6K4) {
00631             int16_t exc_new[SUBFRAME_SIZE];
00632 
00633             ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
00634             ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
00635 
00636             g729d_get_new_exc(exc_new, ctx->exc  + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
00637 
00638             ff_celp_lp_synthesis_filter(
00639                     synth+10,
00640                     &lp[i][1],
00641                     exc_new,
00642                     SUBFRAME_SIZE,
00643                     10,
00644                     0,
00645                     0,
00646                     0x800);
00647         } else {
00648             ff_celp_lp_synthesis_filter(
00649                     synth+10,
00650                     &lp[i][1],
00651                     ctx->exc  + i * SUBFRAME_SIZE,
00652                     SUBFRAME_SIZE,
00653                     10,
00654                     0,
00655                     0,
00656                     0x800);
00657         }
00658         /* Save data (without postfilter) for use in next subframe. */
00659         memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
00660 
00661         /* Calculate gain of unfiltered signal for use in AGC. */
00662         gain_before = 0;
00663         for (j = 0; j < SUBFRAME_SIZE; j++)
00664             gain_before += FFABS(synth[j+10]);
00665 
00666         /* Call postfilter and also update voicing decision for use in next frame. */
00667         ff_g729_postfilter(
00668                 &ctx->dsp,
00669                 &ctx->ht_prev_data,
00670                 &is_periodic,
00671                 &lp[i][0],
00672                 pitch_delay_int[0],
00673                 ctx->residual,
00674                 ctx->res_filter_data,
00675                 ctx->pos_filter_data,
00676                 synth+10,
00677                 SUBFRAME_SIZE);
00678 
00679         /* Calculate gain of filtered signal for use in AGC. */
00680         gain_after = 0;
00681         for(j=0; j<SUBFRAME_SIZE; j++)
00682             gain_after += FFABS(synth[j+10]);
00683 
00684         ctx->gain_coeff = ff_g729_adaptive_gain_control(
00685                 gain_before,
00686                 gain_after,
00687                 synth+10,
00688                 SUBFRAME_SIZE,
00689                 ctx->gain_coeff);
00690 
00691         if (frame_erasure)
00692             ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
00693         else
00694             ctx->pitch_delay_int_prev = pitch_delay_int[i];
00695 
00696         memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
00697         ff_acelp_high_pass_filter(
00698                 out_frame + i*SUBFRAME_SIZE,
00699                 ctx->hpf_f,
00700                 synth+10,
00701                 SUBFRAME_SIZE);
00702         memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
00703     }
00704 
00705     ctx->was_periodic = is_periodic;
00706 
00707     /* Save signal for use in next frame. */
00708     memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
00709 
00710     *got_frame_ptr = 1;
00711     *(AVFrame*)data = ctx->frame;
00712     return buf_size;
00713 }
00714 
00715 AVCodec ff_g729_decoder =
00716 {
00717     .name           = "g729",
00718     .type           = AVMEDIA_TYPE_AUDIO,
00719     .id             = CODEC_ID_G729,
00720     .priv_data_size = sizeof(G729Context),
00721     .init           = decoder_init,
00722     .decode         = decode_frame,
00723     .capabilities = CODEC_CAP_DR1,
00724     .long_name = NULL_IF_CONFIG_SMALL("G.729"),
00725 };
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