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ffmpeg / libavcodec / truespeech.c @ 2912e87a

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/*
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 * DSP Group TrueSpeech compatible decoder
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 * Copyright (c) 2005 Konstantin Shishkov
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 *
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 * This file is part of Libav.
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 *
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 * Libav is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * Libav is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with Libav; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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#include "libavutil/intreadwrite.h"
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#include "avcodec.h"
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#include "truespeech_data.h"
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/**
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 * @file
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 * TrueSpeech decoder.
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 */
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/**
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 * TrueSpeech decoder context
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 */
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typedef struct {
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    /* input data */
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    int16_t vector[8];  //< input vector: 5/5/4/4/4/3/3/3
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    int offset1[2];     //< 8-bit value, used in one copying offset
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    int offset2[4];     //< 7-bit value, encodes offsets for copying and for two-point filter
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    int pulseoff[4];    //< 4-bit offset of pulse values block
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    int pulsepos[4];    //< 27-bit variable, encodes 7 pulse positions
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    int pulseval[4];    //< 7x2-bit pulse values
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    int flag;           //< 1-bit flag, shows how to choose filters
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    /* temporary data */
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    int filtbuf[146];   // some big vector used for storing filters
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    int prevfilt[8];    // filter from previous frame
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    int16_t tmp1[8];    // coefficients for adding to out
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    int16_t tmp2[8];    // coefficients for adding to out
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    int16_t tmp3[8];    // coefficients for adding to out
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    int16_t cvector[8]; // correlated input vector
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    int filtval;        // gain value for one function
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    int16_t newvec[60]; // tmp vector
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    int16_t filters[32]; // filters for every subframe
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} TSContext;
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static av_cold int truespeech_decode_init(AVCodecContext * avctx)
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{
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//    TSContext *c = avctx->priv_data;
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    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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    return 0;
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}
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static void truespeech_read_frame(TSContext *dec, const uint8_t *input)
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{
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    uint32_t t;
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    /* first dword */
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    t = AV_RL32(input);
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    input += 4;
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    dec->flag = t & 1;
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    dec->vector[0] = ts_codebook[0][(t >>  1) & 0x1F];
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    dec->vector[1] = ts_codebook[1][(t >>  6) & 0x1F];
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    dec->vector[2] = ts_codebook[2][(t >> 11) &  0xF];
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    dec->vector[3] = ts_codebook[3][(t >> 15) &  0xF];
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    dec->vector[4] = ts_codebook[4][(t >> 19) &  0xF];
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    dec->vector[5] = ts_codebook[5][(t >> 23) &  0x7];
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    dec->vector[6] = ts_codebook[6][(t >> 26) &  0x7];
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    dec->vector[7] = ts_codebook[7][(t >> 29) &  0x7];
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    /* second dword */
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    t = AV_RL32(input);
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    input += 4;
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    dec->offset2[0] = (t >>  0) & 0x7F;
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    dec->offset2[1] = (t >>  7) & 0x7F;
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    dec->offset2[2] = (t >> 14) & 0x7F;
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    dec->offset2[3] = (t >> 21) & 0x7F;
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    dec->offset1[0] = ((t >> 28) & 0xF) << 4;
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    /* third dword */
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    t = AV_RL32(input);
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    input += 4;
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    dec->pulseval[0] = (t >>  0) & 0x3FFF;
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    dec->pulseval[1] = (t >> 14) & 0x3FFF;
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    dec->offset1[1] = (t >> 28) & 0x0F;
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    /* fourth dword */
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    t = AV_RL32(input);
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    input += 4;
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    dec->pulseval[2] = (t >>  0) & 0x3FFF;
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    dec->pulseval[3] = (t >> 14) & 0x3FFF;
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    dec->offset1[1] |= ((t >> 28) & 0x0F) << 4;
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    /* fifth dword */
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    t = AV_RL32(input);
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    input += 4;
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    dec->pulsepos[0] = (t >> 4) & 0x7FFFFFF;
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    dec->pulseoff[0] = (t >> 0) & 0xF;
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    dec->offset1[0] |= (t >> 31) & 1;
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    /* sixth dword */
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    t = AV_RL32(input);
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    input += 4;
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    dec->pulsepos[1] = (t >> 4) & 0x7FFFFFF;
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    dec->pulseoff[1] = (t >> 0) & 0xF;
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    dec->offset1[0] |= ((t >> 31) & 1) << 1;
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    /* seventh dword */
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    t = AV_RL32(input);
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    input += 4;
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    dec->pulsepos[2] = (t >> 4) & 0x7FFFFFF;
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    dec->pulseoff[2] = (t >> 0) & 0xF;
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    dec->offset1[0] |= ((t >> 31) & 1) << 2;
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    /* eighth dword */
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    t = AV_RL32(input);
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    input += 4;
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    dec->pulsepos[3] = (t >> 4) & 0x7FFFFFF;
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    dec->pulseoff[3] = (t >> 0) & 0xF;
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    dec->offset1[0] |= ((t >> 31) & 1) << 3;
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}
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static void truespeech_correlate_filter(TSContext *dec)
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{
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    int16_t tmp[8];
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    int i, j;
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    for(i = 0; i < 8; i++){
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        if(i > 0){
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            memcpy(tmp, dec->cvector, i * 2);
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            for(j = 0; j < i; j++)
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                dec->cvector[j] = ((tmp[i - j - 1] * dec->vector[i]) +
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                                   (dec->cvector[j] << 15) + 0x4000) >> 15;
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        }
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        dec->cvector[i] = (8 - dec->vector[i]) >> 3;
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    }
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    for(i = 0; i < 8; i++)
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        dec->cvector[i] = (dec->cvector[i] * ts_decay_994_1000[i]) >> 15;
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    dec->filtval = dec->vector[0];
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}
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static void truespeech_filters_merge(TSContext *dec)
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{
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    int i;
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    if(!dec->flag){
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        for(i = 0; i < 8; i++){
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            dec->filters[i + 0] = dec->prevfilt[i];
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            dec->filters[i + 8] = dec->prevfilt[i];
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        }
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    }else{
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        for(i = 0; i < 8; i++){
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            dec->filters[i + 0]=(dec->cvector[i] * 21846 + dec->prevfilt[i] * 10923 + 16384) >> 15;
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            dec->filters[i + 8]=(dec->cvector[i] * 10923 + dec->prevfilt[i] * 21846 + 16384) >> 15;
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        }
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    }
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    for(i = 0; i < 8; i++){
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        dec->filters[i + 16] = dec->cvector[i];
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        dec->filters[i + 24] = dec->cvector[i];
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    }
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}
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static void truespeech_apply_twopoint_filter(TSContext *dec, int quart)
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{
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    int16_t tmp[146 + 60], *ptr0, *ptr1;
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    const int16_t *filter;
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    int i, t, off;
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    t = dec->offset2[quart];
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    if(t == 127){
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        memset(dec->newvec, 0, 60 * 2);
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        return;
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    }
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    for(i = 0; i < 146; i++)
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        tmp[i] = dec->filtbuf[i];
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    off = (t / 25) + dec->offset1[quart >> 1] + 18;
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    ptr0 = tmp + 145 - off;
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    ptr1 = tmp + 146;
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    filter = (const int16_t*)ts_order2_coeffs + (t % 25) * 2;
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    for(i = 0; i < 60; i++){
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        t = (ptr0[0] * filter[0] + ptr0[1] * filter[1] + 0x2000) >> 14;
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        ptr0++;
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        dec->newvec[i] = t;
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        ptr1[i] = t;
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    }
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}
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static void truespeech_place_pulses(TSContext *dec, int16_t *out, int quart)
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{
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    int16_t tmp[7];
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    int i, j, t;
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    const int16_t *ptr1;
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    int16_t *ptr2;
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    int coef;
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    memset(out, 0, 60 * 2);
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    for(i = 0; i < 7; i++) {
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        t = dec->pulseval[quart] & 3;
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        dec->pulseval[quart] >>= 2;
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        tmp[6 - i] = ts_pulse_scales[dec->pulseoff[quart] * 4 + t];
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    }
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    coef = dec->pulsepos[quart] >> 15;
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    ptr1 = (const int16_t*)ts_pulse_values + 30;
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    ptr2 = tmp;
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    for(i = 0, j = 3; (i < 30) && (j > 0); i++){
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        t = *ptr1++;
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        if(coef >= t)
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            coef -= t;
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        else{
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            out[i] = *ptr2++;
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            ptr1 += 30;
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            j--;
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        }
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    }
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    coef = dec->pulsepos[quart] & 0x7FFF;
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    ptr1 = (const int16_t*)ts_pulse_values;
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    for(i = 30, j = 4; (i < 60) && (j > 0); i++){
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        t = *ptr1++;
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        if(coef >= t)
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            coef -= t;
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        else{
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            out[i] = *ptr2++;
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            ptr1 += 30;
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            j--;
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        }
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    }
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}
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static void truespeech_update_filters(TSContext *dec, int16_t *out, int quart)
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{
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    int i;
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    for(i = 0; i < 86; i++)
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        dec->filtbuf[i] = dec->filtbuf[i + 60];
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    for(i = 0; i < 60; i++){
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        dec->filtbuf[i + 86] = out[i] + dec->newvec[i] - (dec->newvec[i] >> 3);
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        out[i] += dec->newvec[i];
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    }
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}
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static void truespeech_synth(TSContext *dec, int16_t *out, int quart)
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{
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    int i,k;
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    int t[8];
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    int16_t *ptr0, *ptr1;
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    ptr0 = dec->tmp1;
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    ptr1 = dec->filters + quart * 8;
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    for(i = 0; i < 60; i++){
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        int sum = 0;
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        for(k = 0; k < 8; k++)
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            sum += ptr0[k] * ptr1[k];
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        sum = (sum + (out[i] << 12) + 0x800) >> 12;
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        out[i] = av_clip(sum, -0x7FFE, 0x7FFE);
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        for(k = 7; k > 0; k--)
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            ptr0[k] = ptr0[k - 1];
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        ptr0[0] = out[i];
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    }
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    for(i = 0; i < 8; i++)
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        t[i] = (ts_decay_35_64[i] * ptr1[i]) >> 15;
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    ptr0 = dec->tmp2;
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    for(i = 0; i < 60; i++){
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        int sum = 0;
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        for(k = 0; k < 8; k++)
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            sum += ptr0[k] * t[k];
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        for(k = 7; k > 0; k--)
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            ptr0[k] = ptr0[k - 1];
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        ptr0[0] = out[i];
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        out[i] = ((out[i] << 12) - sum) >> 12;
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    }
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    for(i = 0; i < 8; i++)
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        t[i] = (ts_decay_3_4[i] * ptr1[i]) >> 15;
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    ptr0 = dec->tmp3;
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    for(i = 0; i < 60; i++){
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        int sum = out[i] << 12;
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        for(k = 0; k < 8; k++)
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            sum += ptr0[k] * t[k];
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        for(k = 7; k > 0; k--)
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            ptr0[k] = ptr0[k - 1];
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        ptr0[0] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
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        sum = ((ptr0[1] * (dec->filtval - (dec->filtval >> 2))) >> 4) + sum;
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        sum = sum - (sum >> 3);
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        out[i] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
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    }
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}
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static void truespeech_save_prevvec(TSContext *c)
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{
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    int i;
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    for(i = 0; i < 8; i++)
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        c->prevfilt[i] = c->cvector[i];
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}
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static int truespeech_decode_frame(AVCodecContext *avctx,
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                void *data, int *data_size,
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                AVPacket *avpkt)
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{
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    const uint8_t *buf = avpkt->data;
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    int buf_size = avpkt->size;
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    TSContext *c = avctx->priv_data;
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    int i, j;
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    short *samples = data;
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    int consumed = 0;
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    int16_t out_buf[240];
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    int iterations;
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    if (!buf_size)
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        return 0;
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    if (buf_size < 32) {
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        av_log(avctx, AV_LOG_ERROR,
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               "Too small input buffer (%d bytes), need at least 32 bytes\n", buf_size);
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        return -1;
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    }
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    iterations = FFMIN(buf_size / 32, *data_size / 480);
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    for(j = 0; j < iterations; j++) {
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        truespeech_read_frame(c, buf + consumed);
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        consumed += 32;
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        truespeech_correlate_filter(c);
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        truespeech_filters_merge(c);
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        memset(out_buf, 0, 240 * 2);
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        for(i = 0; i < 4; i++) {
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            truespeech_apply_twopoint_filter(c, i);
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            truespeech_place_pulses(c, out_buf + i * 60, i);
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            truespeech_update_filters(c, out_buf + i * 60, i);
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            truespeech_synth(c, out_buf + i * 60, i);
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        }
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        truespeech_save_prevvec(c);
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        /* finally output decoded frame */
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        for(i = 0; i < 240; i++)
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            *samples++ = out_buf[i];
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    }
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    *data_size = consumed * 15;
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    return consumed;
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}
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AVCodec ff_truespeech_decoder = {
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    "truespeech",
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    AVMEDIA_TYPE_AUDIO,
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    CODEC_ID_TRUESPEECH,
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    sizeof(TSContext),
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    truespeech_decode_init,
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    NULL,
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    NULL,
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    truespeech_decode_frame,
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    .long_name = NULL_IF_CONFIG_SMALL("DSP Group TrueSpeech"),
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};