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  1. /*
  2.  * G.729, G729 Annex D decoders
  3.  * Copyright (c) 2008 Vladimir Voroshilov
  4.  *
  5.  * This file is part of FFmpeg.
  6.  *
  7.  * FFmpeg is free software; you can redistribute it and/or
  8.  * modify it under the terms of the GNU Lesser General Public
  9.  * License as published by the Free Software Foundation; either
  10.  * version 2.1 of the License, or (at your option) any later version.
  11.  *
  12.  * FFmpeg is distributed in the hope that it will be useful,
  13.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15.  * Lesser General Public License for more details.
  16.  *
  17.  * You should have received a copy of the GNU Lesser General Public
  18.  * License along with FFmpeg; if not, write to the Free Software
  19.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20.  */
  21.  
  22. #include <inttypes.h>
  23. #include <string.h>
  24.  
  25. #include "avcodec.h"
  26. #include "libavutil/avutil.h"
  27. #include "get_bits.h"
  28. #include "dsputil.h"
  29. #include "internal.h"
  30.  
  31.  
  32. #include "g729.h"
  33. #include "lsp.h"
  34. #include "celp_math.h"
  35. #include "celp_filters.h"
  36. #include "acelp_filters.h"
  37. #include "acelp_pitch_delay.h"
  38. #include "acelp_vectors.h"
  39. #include "g729data.h"
  40. #include "g729postfilter.h"
  41.  
  42. /**
  43.  * minimum quantized LSF value (3.2.4)
  44.  * 0.005 in Q13
  45.  */
  46. #define LSFQ_MIN                   40
  47.  
  48. /**
  49.  * maximum quantized LSF value (3.2.4)
  50.  * 3.135 in Q13
  51.  */
  52. #define LSFQ_MAX                   25681
  53.  
  54. /**
  55.  * minimum LSF distance (3.2.4)
  56.  * 0.0391 in Q13
  57.  */
  58. #define LSFQ_DIFF_MIN              321
  59.  
  60. /// interpolation filter length
  61. #define INTERPOL_LEN              11
  62.  
  63. /**
  64.  * minimum gain pitch value (3.8, Equation 47)
  65.  * 0.2 in (1.14)
  66.  */
  67. #define SHARP_MIN                  3277
  68.  
  69. /**
  70.  * maximum gain pitch value (3.8, Equation 47)
  71.  * (EE) This does not comply with the specification.
  72.  * Specification says about 0.8, which should be
  73.  * 13107 in (1.14), but reference C code uses
  74.  * 13017 (equals to 0.7945) instead of it.
  75.  */
  76. #define SHARP_MAX                  13017
  77.  
  78. /**
  79.  * MR_ENERGY (mean removed energy) = mean_energy + 10 * log10(2^26  * subframe_size) in (7.13)
  80.  */
  81. #define MR_ENERGY 1018156
  82.  
  83. #define DECISION_NOISE        0
  84. #define DECISION_INTERMEDIATE 1
  85. #define DECISION_VOICE        2
  86.  
  87. typedef enum {
  88.     FORMAT_G729_8K = 0,
  89.     FORMAT_G729D_6K4,
  90.     FORMAT_COUNT,
  91. } G729Formats;
  92.  
  93. typedef struct {
  94.     uint8_t ac_index_bits[2];   ///< adaptive codebook index for second subframe (size in bits)
  95.     uint8_t parity_bit;         ///< parity bit for pitch delay
  96.     uint8_t gc_1st_index_bits;  ///< gain codebook (first stage) index (size in bits)
  97.     uint8_t gc_2nd_index_bits;  ///< gain codebook (second stage) index (size in bits)
  98.     uint8_t fc_signs_bits;      ///< number of pulses in fixed-codebook vector
  99.     uint8_t fc_indexes_bits;    ///< size (in bits) of fixed-codebook index entry
  100. } G729FormatDescription;
  101.  
  102. typedef struct {
  103.     DSPContext dsp;
  104.  
  105.     /// past excitation signal buffer
  106.     int16_t exc_base[2*SUBFRAME_SIZE+PITCH_DELAY_MAX+INTERPOL_LEN];
  107.  
  108.     int16_t* exc;               ///< start of past excitation data in buffer
  109.     int pitch_delay_int_prev;   ///< integer part of previous subframe's pitch delay (4.1.3)
  110.  
  111.     /// (2.13) LSP quantizer outputs
  112.     int16_t  past_quantizer_output_buf[MA_NP + 1][10];
  113.     int16_t* past_quantizer_outputs[MA_NP + 1];
  114.  
  115.     int16_t lsfq[10];           ///< (2.13) quantized LSF coefficients from previous frame
  116.     int16_t lsp_buf[2][10];     ///< (0.15) LSP coefficients (previous and current frames) (3.2.5)
  117.     int16_t *lsp[2];            ///< pointers to lsp_buf
  118.  
  119.     int16_t quant_energy[4];    ///< (5.10) past quantized energy
  120.  
  121.     /// previous speech data for LP synthesis filter
  122.     int16_t syn_filter_data[10];
  123.  
  124.  
  125.     /// residual signal buffer (used in long-term postfilter)
  126.     int16_t residual[SUBFRAME_SIZE + RES_PREV_DATA_SIZE];
  127.  
  128.     /// previous speech data for residual calculation filter
  129.     int16_t res_filter_data[SUBFRAME_SIZE+10];
  130.  
  131.     /// previous speech data for short-term postfilter
  132.     int16_t pos_filter_data[SUBFRAME_SIZE+10];
  133.  
  134.     /// (1.14) pitch gain of current and five previous subframes
  135.     int16_t past_gain_pitch[6];
  136.  
  137.     /// (14.1) gain code from current and previous subframe
  138.     int16_t past_gain_code[2];
  139.  
  140.     /// voice decision on previous subframe (0-noise, 1-intermediate, 2-voice), G.729D
  141.     int16_t voice_decision;
  142.  
  143.     int16_t onset;              ///< detected onset level (0-2)
  144.     int16_t was_periodic;       ///< whether previous frame was declared as periodic or not (4.4)
  145.     int16_t ht_prev_data;       ///< previous data for 4.2.3, equation 86
  146.     int gain_coeff;             ///< (1.14) gain coefficient (4.2.4)
  147.     uint16_t rand_value;        ///< random number generator value (4.4.4)
  148.     int ma_predictor_prev;      ///< switched MA predictor of LSP quantizer from last good frame
  149.  
  150.     /// (14.14) high-pass filter data (past input)
  151.     int hpf_f[2];
  152.  
  153.     /// high-pass filter data (past output)
  154.     int16_t hpf_z[2];
  155. }  G729Context;
  156.  
  157. static const G729FormatDescription format_g729_8k = {
  158.     .ac_index_bits     = {8,5},
  159.     .parity_bit        = 1,
  160.     .gc_1st_index_bits = GC_1ST_IDX_BITS_8K,
  161.     .gc_2nd_index_bits = GC_2ND_IDX_BITS_8K,
  162.     .fc_signs_bits     = 4,
  163.     .fc_indexes_bits   = 13,
  164. };
  165.  
  166. static const G729FormatDescription format_g729d_6k4 = {
  167.     .ac_index_bits     = {8,4},
  168.     .parity_bit        = 0,
  169.     .gc_1st_index_bits = GC_1ST_IDX_BITS_6K4,
  170.     .gc_2nd_index_bits = GC_2ND_IDX_BITS_6K4,
  171.     .fc_signs_bits     = 2,
  172.     .fc_indexes_bits   = 9,
  173. };
  174.  
  175. /**
  176.  * @brief pseudo random number generator
  177.  */
  178. static inline uint16_t g729_prng(uint16_t value)
  179. {
  180.     return 31821 * value + 13849;
  181. }
  182.  
  183. /**
  184.  * Get parity bit of bit 2..7
  185.  */
  186. static inline int get_parity(uint8_t value)
  187. {
  188.    return (0x6996966996696996ULL >> (value >> 2)) & 1;
  189. }
  190.  
  191. /**
  192.  * Decodes LSF (Line Spectral Frequencies) from L0-L3 (3.2.4).
  193.  * @param[out] lsfq (2.13) quantized LSF coefficients
  194.  * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
  195.  * @param ma_predictor switched MA predictor of LSP quantizer
  196.  * @param vq_1st first stage vector of quantizer
  197.  * @param vq_2nd_low second stage lower vector of LSP quantizer
  198.  * @param vq_2nd_high second stage higher vector of LSP quantizer
  199.  */
  200. static void lsf_decode(int16_t* lsfq, int16_t* past_quantizer_outputs[MA_NP + 1],
  201.                        int16_t ma_predictor,
  202.                        int16_t vq_1st, int16_t vq_2nd_low, int16_t vq_2nd_high)
  203. {
  204.     int i,j;
  205.     static const uint8_t min_distance[2]={10, 5}; //(2.13)
  206.     int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
  207.  
  208.     for (i = 0; i < 5; i++) {
  209.         quantizer_output[i]     = cb_lsp_1st[vq_1st][i    ] + cb_lsp_2nd[vq_2nd_low ][i    ];
  210.         quantizer_output[i + 5] = cb_lsp_1st[vq_1st][i + 5] + cb_lsp_2nd[vq_2nd_high][i + 5];
  211.     }
  212.  
  213.     for (j = 0; j < 2; j++) {
  214.         for (i = 1; i < 10; i++) {
  215.             int diff = (quantizer_output[i - 1] - quantizer_output[i] + min_distance[j]) >> 1;
  216.             if (diff > 0) {
  217.                 quantizer_output[i - 1] -= diff;
  218.                 quantizer_output[i    ] += diff;
  219.             }
  220.         }
  221.     }
  222.  
  223.     for (i = 0; i < 10; i++) {
  224.         int sum = quantizer_output[i] * cb_ma_predictor_sum[ma_predictor][i];
  225.         for (j = 0; j < MA_NP; j++)
  226.             sum += past_quantizer_outputs[j][i] * cb_ma_predictor[ma_predictor][j][i];
  227.  
  228.         lsfq[i] = sum >> 15;
  229.     }
  230.  
  231.     ff_acelp_reorder_lsf(lsfq, LSFQ_DIFF_MIN, LSFQ_MIN, LSFQ_MAX, 10);
  232. }
  233.  
  234. /**
  235.  * Restores past LSP quantizer output using LSF from previous frame
  236.  * @param[in,out] lsfq (2.13) quantized LSF coefficients
  237.  * @param[in,out] past_quantizer_outputs (2.13) quantizer outputs from previous frames
  238.  * @param ma_predictor_prev MA predictor from previous frame
  239.  * @param lsfq_prev (2.13) quantized LSF coefficients from previous frame
  240.  */
  241. static void lsf_restore_from_previous(int16_t* lsfq,
  242.                                       int16_t* past_quantizer_outputs[MA_NP + 1],
  243.                                       int ma_predictor_prev)
  244. {
  245.     int16_t* quantizer_output = past_quantizer_outputs[MA_NP];
  246.     int i,k;
  247.  
  248.     for (i = 0; i < 10; i++) {
  249.         int tmp = lsfq[i] << 15;
  250.  
  251.         for (k = 0; k < MA_NP; k++)
  252.             tmp -= past_quantizer_outputs[k][i] * cb_ma_predictor[ma_predictor_prev][k][i];
  253.  
  254.         quantizer_output[i] = ((tmp >> 15) * cb_ma_predictor_sum_inv[ma_predictor_prev][i]) >> 12;
  255.     }
  256. }
  257.  
  258. /**
  259.  * Constructs new excitation signal and applies phase filter to it
  260.  * @param[out] out constructed speech signal
  261.  * @param in original excitation signal
  262.  * @param fc_cur (2.13) original fixed-codebook vector
  263.  * @param gain_code (14.1) gain code
  264.  * @param subframe_size length of the subframe
  265.  */
  266. static void g729d_get_new_exc(
  267.         int16_t* out,
  268.         const int16_t* in,
  269.         const int16_t* fc_cur,
  270.         int dstate,
  271.         int gain_code,
  272.         int subframe_size)
  273. {
  274.     int i;
  275.     int16_t fc_new[SUBFRAME_SIZE];
  276.  
  277.     ff_celp_convolve_circ(fc_new, fc_cur, phase_filter[dstate], subframe_size);
  278.  
  279.     for(i=0; i<subframe_size; i++)
  280.     {
  281.         out[i]  = in[i];
  282.         out[i] -= (gain_code * fc_cur[i] + 0x2000) >> 14;
  283.         out[i] += (gain_code * fc_new[i] + 0x2000) >> 14;
  284.     }
  285. }
  286.  
  287. /**
  288.  * Makes decision about onset in current subframe
  289.  * @param past_onset decision result of previous subframe
  290.  * @param past_gain_code gain code of current and previous subframe
  291.  *
  292.  * @return onset decision result for current subframe
  293.  */
  294. static int g729d_onset_decision(int past_onset, const int16_t* past_gain_code)
  295. {
  296.     if((past_gain_code[0] >> 1) > past_gain_code[1])
  297.         return 2;
  298.     else
  299.         return FFMAX(past_onset-1, 0);
  300. }
  301.  
  302. /**
  303.  * Makes decision about voice presence in current subframe
  304.  * @param onset onset level
  305.  * @param prev_voice_decision voice decision result from previous subframe
  306.  * @param past_gain_pitch pitch gain of current and previous subframes
  307.  *
  308.  * @return voice decision result for current subframe
  309.  */
  310. static int16_t g729d_voice_decision(int onset, int prev_voice_decision, const int16_t* past_gain_pitch)
  311. {
  312.     int i, low_gain_pitch_cnt, voice_decision;
  313.  
  314.     if(past_gain_pitch[0] >= 14745)      // 0.9
  315.         voice_decision = DECISION_VOICE;
  316.     else if (past_gain_pitch[0] <= 9830) // 0.6
  317.         voice_decision = DECISION_NOISE;
  318.     else
  319.         voice_decision = DECISION_INTERMEDIATE;
  320.  
  321.     for(i=0, low_gain_pitch_cnt=0; i<6; i++)
  322.         if(past_gain_pitch[i] < 9830)
  323.             low_gain_pitch_cnt++;
  324.  
  325.     if(low_gain_pitch_cnt > 2 && !onset)
  326.         voice_decision = DECISION_NOISE;
  327.  
  328.     if(!onset && voice_decision > prev_voice_decision + 1)
  329.         voice_decision--;
  330.  
  331.     if(onset && voice_decision < DECISION_VOICE)
  332.         voice_decision++;
  333.  
  334.     return voice_decision;
  335. }
  336.  
  337. static int32_t scalarproduct_int16_c(const int16_t * v1, const int16_t * v2, int order)
  338. {
  339.     int res = 0;
  340.  
  341.     while (order--)
  342.         res += *v1++ * *v2++;
  343.  
  344.     return res;
  345. }
  346.  
  347. static av_cold int decoder_init(AVCodecContext * avctx)
  348. {
  349.     G729Context* ctx = avctx->priv_data;
  350.     int i,k;
  351.  
  352.     if (avctx->channels != 1) {
  353.         av_log(avctx, AV_LOG_ERROR, "Only mono sound is supported (requested channels: %d).\n", avctx->channels);
  354.         return AVERROR(EINVAL);
  355.     }
  356.     avctx->sample_fmt = AV_SAMPLE_FMT_S16;
  357.  
  358.     /* Both 8kbit/s and 6.4kbit/s modes uses two subframes per frame. */
  359.     avctx->frame_size = SUBFRAME_SIZE << 1;
  360.  
  361.     ctx->gain_coeff = 16384; // 1.0 in (1.14)
  362.  
  363.     for (k = 0; k < MA_NP + 1; k++) {
  364.         ctx->past_quantizer_outputs[k] = ctx->past_quantizer_output_buf[k];
  365.         for (i = 1; i < 11; i++)
  366.             ctx->past_quantizer_outputs[k][i - 1] = (18717 * i) >> 3;
  367.     }
  368.  
  369.     ctx->lsp[0] = ctx->lsp_buf[0];
  370.     ctx->lsp[1] = ctx->lsp_buf[1];
  371.     memcpy(ctx->lsp[0], lsp_init, 10 * sizeof(int16_t));
  372.  
  373.     ctx->exc = &ctx->exc_base[PITCH_DELAY_MAX+INTERPOL_LEN];
  374.  
  375.     ctx->pitch_delay_int_prev = PITCH_DELAY_MIN;
  376.  
  377.     /* random seed initialization */
  378.     ctx->rand_value = 21845;
  379.  
  380.     /* quantized prediction error */
  381.     for(i=0; i<4; i++)
  382.         ctx->quant_energy[i] = -14336; // -14 in (5.10)
  383.  
  384.     ff_dsputil_init(&ctx->dsp, avctx);
  385.     ctx->dsp.scalarproduct_int16 = scalarproduct_int16_c;
  386.  
  387.     return 0;
  388. }
  389.  
  390. static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
  391.                         AVPacket *avpkt)
  392. {
  393.     const uint8_t *buf = avpkt->data;
  394.     int buf_size       = avpkt->size;
  395.     int16_t *out_frame;
  396.     GetBitContext gb;
  397.     const G729FormatDescription *format;
  398.     int frame_erasure = 0;    ///< frame erasure detected during decoding
  399.     int bad_pitch = 0;        ///< parity check failed
  400.     int i;
  401.     int16_t *tmp;
  402.     G729Formats packet_type;
  403.     G729Context *ctx = avctx->priv_data;
  404.     int16_t lp[2][11];           // (3.12)
  405.     uint8_t ma_predictor;     ///< switched MA predictor of LSP quantizer
  406.     uint8_t quantizer_1st;    ///< first stage vector of quantizer
  407.     uint8_t quantizer_2nd_lo; ///< second stage lower vector of quantizer (size in bits)
  408.     uint8_t quantizer_2nd_hi; ///< second stage higher vector of quantizer (size in bits)
  409.  
  410.     int pitch_delay_int[2];      // pitch delay, integer part
  411.     int pitch_delay_3x;          // pitch delay, multiplied by 3
  412.     int16_t fc[SUBFRAME_SIZE];   // fixed-codebook vector
  413.     int16_t synth[SUBFRAME_SIZE+10]; // fixed-codebook vector
  414.     int j, ret;
  415.     int gain_before, gain_after;
  416.     int is_periodic = 0;         // whether one of the subframes is declared as periodic or not
  417.     AVFrame *frame = data;
  418.  
  419.     frame->nb_samples = SUBFRAME_SIZE<<1;
  420.     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
  421.         return ret;
  422.     out_frame = (int16_t*) frame->data[0];
  423.  
  424.     if (buf_size == 10) {
  425.         packet_type = FORMAT_G729_8K;
  426.         format = &format_g729_8k;
  427.         //Reset voice decision
  428.         ctx->onset = 0;
  429.         ctx->voice_decision = DECISION_VOICE;
  430.         av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729 @ 8kbit/s");
  431.     } else if (buf_size == 8) {
  432.         packet_type = FORMAT_G729D_6K4;
  433.         format = &format_g729d_6k4;
  434.         av_log(avctx, AV_LOG_DEBUG, "Packet type: %s\n", "G.729D @ 6.4kbit/s");
  435.     } else {
  436.         av_log(avctx, AV_LOG_ERROR, "Packet size %d is unknown.\n", buf_size);
  437.         return AVERROR_INVALIDDATA;
  438.     }
  439.  
  440.     for (i=0; i < buf_size; i++)
  441.         frame_erasure |= buf[i];
  442.     frame_erasure = !frame_erasure;
  443.  
  444.     init_get_bits(&gb, buf, 8*buf_size);
  445.  
  446.     ma_predictor     = get_bits(&gb, 1);
  447.     quantizer_1st    = get_bits(&gb, VQ_1ST_BITS);
  448.     quantizer_2nd_lo = get_bits(&gb, VQ_2ND_BITS);
  449.     quantizer_2nd_hi = get_bits(&gb, VQ_2ND_BITS);
  450.  
  451.     if(frame_erasure)
  452.         lsf_restore_from_previous(ctx->lsfq, ctx->past_quantizer_outputs,
  453.                                   ctx->ma_predictor_prev);
  454.     else {
  455.         lsf_decode(ctx->lsfq, ctx->past_quantizer_outputs,
  456.                    ma_predictor,
  457.                    quantizer_1st, quantizer_2nd_lo, quantizer_2nd_hi);
  458.         ctx->ma_predictor_prev = ma_predictor;
  459.     }
  460.  
  461.     tmp = ctx->past_quantizer_outputs[MA_NP];
  462.     memmove(ctx->past_quantizer_outputs + 1, ctx->past_quantizer_outputs,
  463.             MA_NP * sizeof(int16_t*));
  464.     ctx->past_quantizer_outputs[0] = tmp;
  465.  
  466.     ff_acelp_lsf2lsp(ctx->lsp[1], ctx->lsfq, 10);
  467.  
  468.     ff_acelp_lp_decode(&lp[0][0], &lp[1][0], ctx->lsp[1], ctx->lsp[0], 10);
  469.  
  470.     FFSWAP(int16_t*, ctx->lsp[1], ctx->lsp[0]);
  471.  
  472.     for (i = 0; i < 2; i++) {
  473.         int gain_corr_factor;
  474.  
  475.         uint8_t ac_index;      ///< adaptive codebook index
  476.         uint8_t pulses_signs;  ///< fixed-codebook vector pulse signs
  477.         int fc_indexes;        ///< fixed-codebook indexes
  478.         uint8_t gc_1st_index;  ///< gain codebook (first stage) index
  479.         uint8_t gc_2nd_index;  ///< gain codebook (second stage) index
  480.  
  481.         ac_index      = get_bits(&gb, format->ac_index_bits[i]);
  482.         if(!i && format->parity_bit)
  483.             bad_pitch = get_parity(ac_index) == get_bits1(&gb);
  484.         fc_indexes    = get_bits(&gb, format->fc_indexes_bits);
  485.         pulses_signs  = get_bits(&gb, format->fc_signs_bits);
  486.         gc_1st_index  = get_bits(&gb, format->gc_1st_index_bits);
  487.         gc_2nd_index  = get_bits(&gb, format->gc_2nd_index_bits);
  488.  
  489.         if (frame_erasure)
  490.             pitch_delay_3x   = 3 * ctx->pitch_delay_int_prev;
  491.         else if(!i) {
  492.             if (bad_pitch)
  493.                 pitch_delay_3x   = 3 * ctx->pitch_delay_int_prev;
  494.             else
  495.                 pitch_delay_3x = ff_acelp_decode_8bit_to_1st_delay3(ac_index);
  496.         } else {
  497.             int pitch_delay_min = av_clip(ctx->pitch_delay_int_prev - 5,
  498.                                           PITCH_DELAY_MIN, PITCH_DELAY_MAX - 9);
  499.  
  500.             if(packet_type == FORMAT_G729D_6K4)
  501.                 pitch_delay_3x = ff_acelp_decode_4bit_to_2nd_delay3(ac_index, pitch_delay_min);
  502.             else
  503.                 pitch_delay_3x = ff_acelp_decode_5_6_bit_to_2nd_delay3(ac_index, pitch_delay_min);
  504.         }
  505.  
  506.         /* Round pitch delay to nearest (used everywhere except ff_acelp_interpolate). */
  507.         pitch_delay_int[i]  = (pitch_delay_3x + 1) / 3;
  508.         if (pitch_delay_int[i] > PITCH_DELAY_MAX) {
  509.             av_log(avctx, AV_LOG_WARNING, "pitch_delay_int %d is too large\n", pitch_delay_int[i]);
  510.             pitch_delay_int[i] = PITCH_DELAY_MAX;
  511.         }
  512.  
  513.         if (frame_erasure) {
  514.             ctx->rand_value = g729_prng(ctx->rand_value);
  515.             fc_indexes   = ctx->rand_value & ((1 << format->fc_indexes_bits) - 1);
  516.  
  517.             ctx->rand_value = g729_prng(ctx->rand_value);
  518.             pulses_signs = ctx->rand_value;
  519.         }
  520.  
  521.  
  522.         memset(fc, 0, sizeof(int16_t) * SUBFRAME_SIZE);
  523.         switch (packet_type) {
  524.             case FORMAT_G729_8K:
  525.                 ff_acelp_fc_pulse_per_track(fc, ff_fc_4pulses_8bits_tracks_13,
  526.                                             ff_fc_4pulses_8bits_track_4,
  527.                                             fc_indexes, pulses_signs, 3, 3);
  528.                 break;
  529.             case FORMAT_G729D_6K4:
  530.                 ff_acelp_fc_pulse_per_track(fc, ff_fc_2pulses_9bits_track1_gray,
  531.                                             ff_fc_2pulses_9bits_track2_gray,
  532.                                             fc_indexes, pulses_signs, 1, 4);
  533.                 break;
  534.         }
  535.  
  536.         /*
  537.           This filter enhances harmonic components of the fixed-codebook vector to
  538.           improve the quality of the reconstructed speech.
  539.  
  540.                      / fc_v[i],                                    i < pitch_delay
  541.           fc_v[i] = <
  542.                      \ fc_v[i] + gain_pitch * fc_v[i-pitch_delay], i >= pitch_delay
  543.         */
  544.         ff_acelp_weighted_vector_sum(fc + pitch_delay_int[i],
  545.                                      fc + pitch_delay_int[i],
  546.                                      fc, 1 << 14,
  547.                                      av_clip(ctx->past_gain_pitch[0], SHARP_MIN, SHARP_MAX),
  548.                                      0, 14,
  549.                                      SUBFRAME_SIZE - pitch_delay_int[i]);
  550.  
  551.         memmove(ctx->past_gain_pitch+1, ctx->past_gain_pitch, 5 * sizeof(int16_t));
  552.         ctx->past_gain_code[1] = ctx->past_gain_code[0];
  553.  
  554.         if (frame_erasure) {
  555.             ctx->past_gain_pitch[0] = (29491 * ctx->past_gain_pitch[0]) >> 15; // 0.90 (0.15)
  556.             ctx->past_gain_code[0]  = ( 2007 * ctx->past_gain_code[0] ) >> 11; // 0.98 (0.11)
  557.  
  558.             gain_corr_factor = 0;
  559.         } else {
  560.             if (packet_type == FORMAT_G729D_6K4) {
  561.                 ctx->past_gain_pitch[0]  = cb_gain_1st_6k4[gc_1st_index][0] +
  562.                                            cb_gain_2nd_6k4[gc_2nd_index][0];
  563.                 gain_corr_factor = cb_gain_1st_6k4[gc_1st_index][1] +
  564.                                    cb_gain_2nd_6k4[gc_2nd_index][1];
  565.  
  566.                 /* Without check below overflow can occur in ff_acelp_update_past_gain.
  567.                    It is not issue for G.729, because gain_corr_factor in it's case is always
  568.                    greater than 1024, while in G.729D it can be even zero. */
  569.                 gain_corr_factor = FFMAX(gain_corr_factor, 1024);
  570. #ifndef G729_BITEXACT
  571.                 gain_corr_factor >>= 1;
  572. #endif
  573.             } else {
  574.                 ctx->past_gain_pitch[0]  = cb_gain_1st_8k[gc_1st_index][0] +
  575.                                            cb_gain_2nd_8k[gc_2nd_index][0];
  576.                 gain_corr_factor = cb_gain_1st_8k[gc_1st_index][1] +
  577.                                    cb_gain_2nd_8k[gc_2nd_index][1];
  578.             }
  579.  
  580.             /* Decode the fixed-codebook gain. */
  581.             ctx->past_gain_code[0] = ff_acelp_decode_gain_code(&ctx->dsp, gain_corr_factor,
  582.                                                                fc, MR_ENERGY,
  583.                                                                ctx->quant_energy,
  584.                                                                ma_prediction_coeff,
  585.                                                                SUBFRAME_SIZE, 4);
  586. #ifdef G729_BITEXACT
  587.             /*
  588.               This correction required to get bit-exact result with
  589.               reference code, because gain_corr_factor in G.729D is
  590.               two times larger than in original G.729.
  591.  
  592.               If bit-exact result is not issue then gain_corr_factor
  593.               can be simpler divided by 2 before call to g729_get_gain_code
  594.               instead of using correction below.
  595.             */
  596.             if (packet_type == FORMAT_G729D_6K4) {
  597.                 gain_corr_factor >>= 1;
  598.                 ctx->past_gain_code[0] >>= 1;
  599.             }
  600. #endif
  601.         }
  602.         ff_acelp_update_past_gain(ctx->quant_energy, gain_corr_factor, 2, frame_erasure);
  603.  
  604.         /* Routine requires rounding to lowest. */
  605.         ff_acelp_interpolate(ctx->exc + i * SUBFRAME_SIZE,
  606.                              ctx->exc + i * SUBFRAME_SIZE - pitch_delay_3x / 3,
  607.                              ff_acelp_interp_filter, 6,
  608.                              (pitch_delay_3x % 3) << 1,
  609.                              10, SUBFRAME_SIZE);
  610.  
  611.         ff_acelp_weighted_vector_sum(ctx->exc + i * SUBFRAME_SIZE,
  612.                                      ctx->exc + i * SUBFRAME_SIZE, fc,
  613.                                      (!ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_pitch[0],
  614.                                      ( ctx->was_periodic && frame_erasure) ? 0 : ctx->past_gain_code[0],
  615.                                      1 << 13, 14, SUBFRAME_SIZE);
  616.  
  617.         memcpy(synth, ctx->syn_filter_data, 10 * sizeof(int16_t));
  618.  
  619.         if (ff_celp_lp_synthesis_filter(
  620.             synth+10,
  621.             &lp[i][1],
  622.             ctx->exc  + i * SUBFRAME_SIZE,
  623.             SUBFRAME_SIZE,
  624.             10,
  625.             1,
  626.             0,
  627.             0x800))
  628.             /* Overflow occurred, downscale excitation signal... */
  629.             for (j = 0; j < 2 * SUBFRAME_SIZE + PITCH_DELAY_MAX + INTERPOL_LEN; j++)
  630.                 ctx->exc_base[j] >>= 2;
  631.  
  632.         /* ... and make synthesis again. */
  633.         if (packet_type == FORMAT_G729D_6K4) {
  634.             int16_t exc_new[SUBFRAME_SIZE];
  635.  
  636.             ctx->onset = g729d_onset_decision(ctx->onset, ctx->past_gain_code);
  637.             ctx->voice_decision = g729d_voice_decision(ctx->onset, ctx->voice_decision, ctx->past_gain_pitch);
  638.  
  639.             g729d_get_new_exc(exc_new, ctx->exc  + i * SUBFRAME_SIZE, fc, ctx->voice_decision, ctx->past_gain_code[0], SUBFRAME_SIZE);
  640.  
  641.             ff_celp_lp_synthesis_filter(
  642.                     synth+10,
  643.                     &lp[i][1],
  644.                     exc_new,
  645.                     SUBFRAME_SIZE,
  646.                     10,
  647.                     0,
  648.                     0,
  649.                     0x800);
  650.         } else {
  651.             ff_celp_lp_synthesis_filter(
  652.                     synth+10,
  653.                     &lp[i][1],
  654.                     ctx->exc  + i * SUBFRAME_SIZE,
  655.                     SUBFRAME_SIZE,
  656.                     10,
  657.                     0,
  658.                     0,
  659.                     0x800);
  660.         }
  661.         /* Save data (without postfilter) for use in next subframe. */
  662.         memcpy(ctx->syn_filter_data, synth+SUBFRAME_SIZE, 10 * sizeof(int16_t));
  663.  
  664.         /* Calculate gain of unfiltered signal for use in AGC. */
  665.         gain_before = 0;
  666.         for (j = 0; j < SUBFRAME_SIZE; j++)
  667.             gain_before += FFABS(synth[j+10]);
  668.  
  669.         /* Call postfilter and also update voicing decision for use in next frame. */
  670.         ff_g729_postfilter(
  671.                 &ctx->dsp,
  672.                 &ctx->ht_prev_data,
  673.                 &is_periodic,
  674.                 &lp[i][0],
  675.                 pitch_delay_int[0],
  676.                 ctx->residual,
  677.                 ctx->res_filter_data,
  678.                 ctx->pos_filter_data,
  679.                 synth+10,
  680.                 SUBFRAME_SIZE);
  681.  
  682.         /* Calculate gain of filtered signal for use in AGC. */
  683.         gain_after = 0;
  684.         for(j=0; j<SUBFRAME_SIZE; j++)
  685.             gain_after += FFABS(synth[j+10]);
  686.  
  687.         ctx->gain_coeff = ff_g729_adaptive_gain_control(
  688.                 gain_before,
  689.                 gain_after,
  690.                 synth+10,
  691.                 SUBFRAME_SIZE,
  692.                 ctx->gain_coeff);
  693.  
  694.         if (frame_erasure)
  695.             ctx->pitch_delay_int_prev = FFMIN(ctx->pitch_delay_int_prev + 1, PITCH_DELAY_MAX);
  696.         else
  697.             ctx->pitch_delay_int_prev = pitch_delay_int[i];
  698.  
  699.         memcpy(synth+8, ctx->hpf_z, 2*sizeof(int16_t));
  700.         ff_acelp_high_pass_filter(
  701.                 out_frame + i*SUBFRAME_SIZE,
  702.                 ctx->hpf_f,
  703.                 synth+10,
  704.                 SUBFRAME_SIZE);
  705.         memcpy(ctx->hpf_z, synth+8+SUBFRAME_SIZE, 2*sizeof(int16_t));
  706.     }
  707.  
  708.     ctx->was_periodic = is_periodic;
  709.  
  710.     /* Save signal for use in next frame. */
  711.     memmove(ctx->exc_base, ctx->exc_base + 2 * SUBFRAME_SIZE, (PITCH_DELAY_MAX+INTERPOL_LEN)*sizeof(int16_t));
  712.  
  713.     *got_frame_ptr = 1;
  714.     return buf_size;
  715. }
  716.  
  717. AVCodec ff_g729_decoder = {
  718.     .name           = "g729",
  719.     .long_name      = NULL_IF_CONFIG_SMALL("G.729"),
  720.     .type           = AVMEDIA_TYPE_AUDIO,
  721.     .id             = AV_CODEC_ID_G729,
  722.     .priv_data_size = sizeof(G729Context),
  723.     .init           = decoder_init,
  724.     .decode         = decode_frame,
  725.     .capabilities   = CODEC_CAP_DR1,
  726. };
  727.