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/*
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 * QCELP decoder
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 * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
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 *
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg 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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 * FFmpeg 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 FFmpeg; 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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/**
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 * @file qcelpdec.c
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 * QCELP decoder
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 * @author Reynaldo H. Verdejo Pinochet
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 */
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#include <stddef.h>
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#include "avcodec.h"
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#include "bitstream.h"
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#include "qcelp.h"
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#include "qcelpdata.h"
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#include "celp_math.h"
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#include "celp_filters.h"
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#undef NDEBUG
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#include <assert.h>
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static void weighted_vector_sumf(float *out,
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                                 const float *in_a,
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                                 const float *in_b,
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                                 float weight_coeff_a,
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                                 float weight_coeff_b,
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                                 int length) {
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    int   i;
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    for (i = 0; i < length; i++)
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        out[i] = weight_coeff_a * in_a[i]
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               + weight_coeff_b * in_b[i];
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}
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/**
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 * Initialize the speech codec according to the specification.
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 *
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 * TIA/EIA/IS-733 2.4.9
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 */
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static av_cold int qcelp_decode_init(AVCodecContext *avctx) {
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    QCELPContext *q = avctx->priv_data;
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    int i;
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    avctx->sample_fmt = SAMPLE_FMT_FLT;
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    for (i = 0; i < 10; i++)
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        q->prev_lspf[i] = (i + 1) / 11.;
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    return 0;
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}
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/**
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 * Computes the scaled codebook vector Cdn From INDEX and GAIN
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 * for all rates.
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 *
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 * The specification lacks some information here.
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 *
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 * TIA/EIA/IS-733 has an omission on the codebook index determination
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 * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
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 * you have to subtract the decoded index parameter from the given scaled
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 * codebook vector index 'n' to get the desired circular codebook index, but
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 * it does not mention that you have to clamp 'n' to [0-9] in order to get
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 * RI-compliant results.
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 *
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 * The reason for this mistake seems to be the fact they forgot to mention you
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 * have to do these calculations per codebook subframe and adjust given
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 * equation values accordingly.
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 *
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 * @param q the context
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 * @param gain array holding the 4 pitch subframe gain values
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 * @param cdn_vector array for the generated scaled codebook vector
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 */
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static void compute_svector(const QCELPContext *q,
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                            const float *gain,
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                            float *cdn_vector) {
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    int      i, j, k;
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    uint16_t cbseed, cindex;
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    float    *rnd, tmp_gain, fir_filter_value;
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    switch (q->framerate) {
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    case RATE_FULL:
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        for (i = 0; i < 16; i++) {
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            tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
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            cindex = -q->cindex[i];
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            for (j = 0; j < 10; j++)
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                *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
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        }
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        break;
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    case RATE_HALF:
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        for (i = 0; i < 4; i++) {
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            tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
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            cindex = -q->cindex[i];
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            for (j = 0; j < 40; j++)
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                *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
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        }
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        break;
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    case RATE_QUARTER:
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        cbseed = (0x0003 & q->lspv[4])<<14 |
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                 (0x003F & q->lspv[3])<< 8 |
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                 (0x0060 & q->lspv[2])<< 1 |
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                 (0x0007 & q->lspv[1])<< 3 |
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                 (0x0038 & q->lspv[0])>> 3 ;
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        rnd = q->rnd_fir_filter_mem + 20;
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        for (i = 0; i < 8; i++) {
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            tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
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            for (k = 0; k < 20; k++) {
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                cbseed = 521 * cbseed + 259;
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                *rnd = (int16_t)cbseed;
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                // FIR filter
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                fir_filter_value = 0.0;
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                for (j = 0; j < 10; j++)
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                    fir_filter_value += qcelp_rnd_fir_coefs[j ] * (rnd[-j ] + rnd[-20+j]);
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                fir_filter_value     += qcelp_rnd_fir_coefs[10] *  rnd[-10];
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                *cdn_vector++ = tmp_gain * fir_filter_value;
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                rnd++;
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            }
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        }
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        memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));
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        break;
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    case RATE_OCTAVE:
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        cbseed = q->first16bits;
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        for (i = 0; i < 8; i++) {
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            tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
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            for (j = 0; j < 20; j++) {
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                cbseed = 521 * cbseed + 259;
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                *cdn_vector++ = tmp_gain * (int16_t)cbseed;
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            }
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        }
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        break;
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    case I_F_Q:
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        cbseed = -44; // random codebook index
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        for (i = 0; i < 4; i++) {
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            tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
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            for (j = 0; j < 40; j++)
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                *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
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        }
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        break;
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    }
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}
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/**
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 * Apply generic gain control.
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 *
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 * @param v_out output vector
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 * @param v_in gain-controlled vector
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 * @param v_ref vector to control gain of
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 *
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 * FIXME: If v_ref is a zero vector, it energy is zero
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 *        and the behavior of the gain control is
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 *        undefined in the specs.
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 *
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 * TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6
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 */
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static void apply_gain_ctrl(float *v_out,
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                            const float *v_ref,
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                            const float *v_in) {
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    int   i, j, len;
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    float scalefactor;
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    for (i = 0, j = 0; i < 4; i++) {
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        scalefactor = ff_dot_productf(v_in + j, v_in + j, 40);
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        if (scalefactor)
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            scalefactor = sqrt(ff_dot_productf(v_ref + j, v_ref + j, 40) / scalefactor);
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        else
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            av_log_missing_feature(NULL, "Zero energy for gain control", 1);
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        for (len = j + 40; j < len; j++)
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            v_out[j] = scalefactor * v_in[j];
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    }
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}
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/**
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 * Apply filter in pitch-subframe steps.
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 *
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 * @param memory buffer for the previous state of the filter
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 *        - must be able to contain 303 elements
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 *        - the 143 first elements are from the previous state
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 *        - the next 160 are for output
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 * @param v_in input filter vector
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 * @param gain per-subframe gain array, each element is between 0.0 and 2.0
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 * @param lag per-subframe lag array, each element is
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 *        - between 16 and 143 if its corresponding pfrac is 0,
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 *        - between 16 and 139 otherwise
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 * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0 otherwise
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 *
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 * @return filter output vector
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 */
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static const float *do_pitchfilter(float memory[303],
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                                   const float v_in[160],
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                                   const float gain[4],
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                                   const uint8_t *lag,
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                                   const uint8_t pfrac[4]) {
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    int         i, j;
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    float       *v_lag, *v_out;
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    const float *v_len;
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    v_out = memory + 143; // Output vector starts at memory[143].
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    for (i = 0; i < 4; i++)
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        if (gain[i]) {
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            v_lag = memory + 143 + 40 * i - lag[i];
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            for (v_len = v_in + 40; v_in < v_len; v_in++) {
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                if (pfrac[i]) { // If it is a fractional lag...
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                    for (j = 0, *v_out = 0.; j < 4; j++)
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                        *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
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                } else
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                    *v_out = *v_lag;
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                *v_out = *v_in + gain[i] * *v_out;
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                v_lag++;
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                v_out++;
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            }
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        } else {
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            memcpy(v_out, v_in, 40 * sizeof(float));
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            v_in  += 40;
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            v_out += 40;
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        }
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    memmove(memory, memory + 160, 143 * sizeof(float));
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    return memory + 143;
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}
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/**
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 * Interpolates LSP frequencies and computes LPC coefficients
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 * for a given framerate & pitch subframe.
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 *
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 * TIA/EIA/IS-733 2.4.3.3.4
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 *
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 * @param q the context
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 * @param curr_lspf LSP frequencies vector of the current frame
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 * @param lpc float vector for the resulting LPC
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 * @param subframe_num frame number in decoded stream
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 */
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void interpolate_lpc(QCELPContext *q,
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                     const float *curr_lspf,
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                     float *lpc,
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                     const int subframe_num) {
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    float interpolated_lspf[10];
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    float weight;
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    if (q->framerate >= RATE_QUARTER) {
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        weight = 0.25 * (subframe_num + 1);
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    } else if (q->framerate == RATE_OCTAVE && !subframe_num) {
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        weight = 0.625;
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    } else {
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        weight = 1.0;
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    }
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    if (weight != 1.0) {
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        weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf, weight, 1.0 - weight, 10);
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        qcelp_lspf2lpc(interpolated_lspf, lpc);
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    } else if (q->framerate >= RATE_QUARTER || (q->framerate == I_F_Q && !subframe_num))
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        qcelp_lspf2lpc(curr_lspf, lpc);
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}
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static int buf_size2framerate(const int buf_size) {
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    switch (buf_size) {
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    case 35:
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        return RATE_FULL;
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    case 17:
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        return RATE_HALF;
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    case  8:
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        return RATE_QUARTER;
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    case  4:
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        return RATE_OCTAVE;
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    case  1:
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        return SILENCE;
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    }
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    return -1;
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}
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static void warn_insufficient_frame_quality(AVCodecContext *avctx,
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                                            const char *message) {
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    av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number, message);
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}
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AVCodec qcelp_decoder =
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{
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    .name   = "qcelp",
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    .type   = CODEC_TYPE_AUDIO,
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    .id     = CODEC_ID_QCELP,
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    .init   = qcelp_decode_init,
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    .decode = qcelp_decode_frame,
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    .priv_data_size = sizeof(QCELPContext),
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    .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
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};