g723_1.c 41.7 KB
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/*
 * G.723.1 compatible decoder
 * Copyright (c) 2006 Benjamin Larsson
 * Copyright (c) 2010 Mohamed Naufal Basheer
 *
 * This file is part of Libav.
 *
 * Libav is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * Libav is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with Libav; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

/**
 * @file
 * G.723.1 compatible decoder
 */

#define BITSTREAM_READER_LE
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#include "libavutil/channel_layout.h"
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#include "libavutil/mem.h"
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#include "libavutil/opt.h"
#include "avcodec.h"
#include "get_bits.h"
#include "acelp_vectors.h"
#include "celp_filters.h"
#include "g723_1_data.h"
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#include "internal.h"
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#define CNG_RANDOM_SEED 12345

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/**
 * G723.1 frame types
 */
enum FrameType {
    ACTIVE_FRAME,        ///< Active speech
    SID_FRAME,           ///< Silence Insertion Descriptor frame
    UNTRANSMITTED_FRAME
};

enum Rate {
    RATE_6300,
    RATE_5300
};

/**
 * G723.1 unpacked data subframe
 */
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typedef struct G723_1_Subframe {
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    int ad_cb_lag;     ///< adaptive codebook lag
    int ad_cb_gain;
    int dirac_train;
    int pulse_sign;
    int grid_index;
    int amp_index;
    int pulse_pos;
} G723_1_Subframe;

/**
 * Pitch postfilter parameters
 */
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typedef struct PPFParam {
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    int     index;    ///< postfilter backward/forward lag
    int16_t opt_gain; ///< optimal gain
    int16_t sc_gain;  ///< scaling gain
} PPFParam;

typedef struct g723_1_context {
    AVClass *class;

    G723_1_Subframe subframe[4];
    enum FrameType cur_frame_type;
    enum FrameType past_frame_type;
    enum Rate cur_rate;
    uint8_t lsp_index[LSP_BANDS];
    int pitch_lag[2];
    int erased_frames;

    int16_t prev_lsp[LPC_ORDER];
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    int16_t sid_lsp[LPC_ORDER];
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    int16_t prev_excitation[PITCH_MAX];
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    int16_t excitation[PITCH_MAX + FRAME_LEN + 4];
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    int16_t synth_mem[LPC_ORDER];
    int16_t fir_mem[LPC_ORDER];
    int     iir_mem[LPC_ORDER];

    int random_seed;
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    int cng_random_seed;
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    int interp_index;
    int interp_gain;
    int sid_gain;
    int cur_gain;
    int reflection_coef;
    int pf_gain;
    int postfilter;

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    int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4];
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} G723_1_Context;

static av_cold int g723_1_decode_init(AVCodecContext *avctx)
{
    G723_1_Context *p = avctx->priv_data;

    avctx->channel_layout = AV_CH_LAYOUT_MONO;
    avctx->sample_fmt     = AV_SAMPLE_FMT_S16;
    avctx->channels       = 1;
    avctx->sample_rate    = 8000;
    p->pf_gain            = 1 << 12;

    memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
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    memcpy(p->sid_lsp,  dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp));

    p->cng_random_seed = CNG_RANDOM_SEED;
    p->past_frame_type = SID_FRAME;
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    return 0;
}

/**
 * Unpack the frame into parameters.
 *
 * @param p           the context
 * @param buf         pointer to the input buffer
 * @param buf_size    size of the input buffer
 */
static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf,
                            int buf_size)
{
    GetBitContext gb;
    int ad_cb_len;
    int temp, info_bits, i;

    init_get_bits(&gb, buf, buf_size * 8);

    /* Extract frame type and rate info */
    info_bits = get_bits(&gb, 2);

    if (info_bits == 3) {
        p->cur_frame_type = UNTRANSMITTED_FRAME;
        return 0;
    }

    /* Extract 24 bit lsp indices, 8 bit for each band */
    p->lsp_index[2] = get_bits(&gb, 8);
    p->lsp_index[1] = get_bits(&gb, 8);
    p->lsp_index[0] = get_bits(&gb, 8);

    if (info_bits == 2) {
        p->cur_frame_type = SID_FRAME;
        p->subframe[0].amp_index = get_bits(&gb, 6);
        return 0;
    }

    /* Extract the info common to both rates */
    p->cur_rate       = info_bits ? RATE_5300 : RATE_6300;
    p->cur_frame_type = ACTIVE_FRAME;

    p->pitch_lag[0] = get_bits(&gb, 7);
    if (p->pitch_lag[0] > 123)       /* test if forbidden code */
        return -1;
    p->pitch_lag[0] += PITCH_MIN;
    p->subframe[1].ad_cb_lag = get_bits(&gb, 2);

    p->pitch_lag[1] = get_bits(&gb, 7);
    if (p->pitch_lag[1] > 123)
        return -1;
    p->pitch_lag[1] += PITCH_MIN;
    p->subframe[3].ad_cb_lag = get_bits(&gb, 2);
    p->subframe[0].ad_cb_lag = 1;
    p->subframe[2].ad_cb_lag = 1;

    for (i = 0; i < SUBFRAMES; i++) {
        /* Extract combined gain */
        temp = get_bits(&gb, 12);
        ad_cb_len = 170;
        p->subframe[i].dirac_train = 0;
        if (p->cur_rate == RATE_6300 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) {
            p->subframe[i].dirac_train = temp >> 11;
            temp &= 0x7FF;
            ad_cb_len = 85;
        }
        p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS);
        if (p->subframe[i].ad_cb_gain < ad_cb_len) {
            p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain *
                                       GAIN_LEVELS;
        } else {
            return -1;
        }
    }

    p->subframe[0].grid_index = get_bits(&gb, 1);
    p->subframe[1].grid_index = get_bits(&gb, 1);
    p->subframe[2].grid_index = get_bits(&gb, 1);
    p->subframe[3].grid_index = get_bits(&gb, 1);

    if (p->cur_rate == RATE_6300) {
        skip_bits(&gb, 1);  /* skip reserved bit */

        /* Compute pulse_pos index using the 13-bit combined position index */
        temp = get_bits(&gb, 13);
        p->subframe[0].pulse_pos = temp / 810;

        temp -= p->subframe[0].pulse_pos * 810;
        p->subframe[1].pulse_pos = FASTDIV(temp, 90);

        temp -= p->subframe[1].pulse_pos * 90;
        p->subframe[2].pulse_pos = FASTDIV(temp, 9);
        p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9;

        p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) +
                                   get_bits(&gb, 16);
        p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) +
                                   get_bits(&gb, 14);
        p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) +
                                   get_bits(&gb, 16);
        p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) +
                                   get_bits(&gb, 14);

        p->subframe[0].pulse_sign = get_bits(&gb, 6);
        p->subframe[1].pulse_sign = get_bits(&gb, 5);
        p->subframe[2].pulse_sign = get_bits(&gb, 6);
        p->subframe[3].pulse_sign = get_bits(&gb, 5);
    } else { /* 5300 bps */
        p->subframe[0].pulse_pos  = get_bits(&gb, 12);
        p->subframe[1].pulse_pos  = get_bits(&gb, 12);
        p->subframe[2].pulse_pos  = get_bits(&gb, 12);
        p->subframe[3].pulse_pos  = get_bits(&gb, 12);

        p->subframe[0].pulse_sign = get_bits(&gb, 4);
        p->subframe[1].pulse_sign = get_bits(&gb, 4);
        p->subframe[2].pulse_sign = get_bits(&gb, 4);
        p->subframe[3].pulse_sign = get_bits(&gb, 4);
    }

    return 0;
}

/**
 * Bitexact implementation of sqrt(val/2).
 */
static int16_t square_root(int val)
{
    int16_t res = 0;
    int16_t exp = 0x4000;
    int i;

    for (i = 0; i < 14; i ++) {
        int res_exp = res + exp;
        if (val >= res_exp * res_exp << 1)
            res += exp;
        exp >>= 1;
    }
    return res;
}

/**
 * Calculate the number of left-shifts required for normalizing the input.
 *
 * @param num   input number
 * @param width width of the input, 16 bits(0) / 32 bits(1)
 */
static int normalize_bits(int num, int width)
{
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    return width - av_log2(num) - 1;
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}

/**
 * Scale vector contents based on the largest of their absolutes.
 */
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static int scale_vector(int16_t *dst, const int16_t *vector, int length)
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{
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    int bits, max = 0;
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    int i;


    for (i = 0; i < length; i++)
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        max |= FFABS(vector[i]);
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    max   = FFMIN(max, 0x7FFF);
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    bits  = normalize_bits(max, 15);

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    for (i = 0; i < length; i++)
        dst[i] = vector[i] << bits >> 3;
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    return bits - 3;
}

/**
 * Perform inverse quantization of LSP frequencies.
 *
 * @param cur_lsp    the current LSP vector
 * @param prev_lsp   the previous LSP vector
 * @param lsp_index  VQ indices
 * @param bad_frame  bad frame flag
 */
static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp,
                          uint8_t *lsp_index, int bad_frame)
{
    int min_dist, pred;
    int i, j, temp, stable;

    /* Check for frame erasure */
    if (!bad_frame) {
        min_dist     = 0x100;
        pred         = 12288;
    } else {
        min_dist     = 0x200;
        pred         = 23552;
        lsp_index[0] = lsp_index[1] = lsp_index[2] = 0;
    }

    /* Get the VQ table entry corresponding to the transmitted index */
    cur_lsp[0] = lsp_band0[lsp_index[0]][0];
    cur_lsp[1] = lsp_band0[lsp_index[0]][1];
    cur_lsp[2] = lsp_band0[lsp_index[0]][2];
    cur_lsp[3] = lsp_band1[lsp_index[1]][0];
    cur_lsp[4] = lsp_band1[lsp_index[1]][1];
    cur_lsp[5] = lsp_band1[lsp_index[1]][2];
    cur_lsp[6] = lsp_band2[lsp_index[2]][0];
    cur_lsp[7] = lsp_band2[lsp_index[2]][1];
    cur_lsp[8] = lsp_band2[lsp_index[2]][2];
    cur_lsp[9] = lsp_band2[lsp_index[2]][3];

    /* Add predicted vector & DC component to the previously quantized vector */
    for (i = 0; i < LPC_ORDER; i++) {
        temp        = ((prev_lsp[i] - dc_lsp[i]) * pred + (1 << 14)) >> 15;
        cur_lsp[i] += dc_lsp[i] + temp;
    }

    for (i = 0; i < LPC_ORDER; i++) {
        cur_lsp[0]             = FFMAX(cur_lsp[0],  0x180);
        cur_lsp[LPC_ORDER - 1] = FFMIN(cur_lsp[LPC_ORDER - 1], 0x7e00);

        /* Stability check */
        for (j = 1; j < LPC_ORDER; j++) {
            temp = min_dist + cur_lsp[j - 1] - cur_lsp[j];
            if (temp > 0) {
                temp >>= 1;
                cur_lsp[j - 1] -= temp;
                cur_lsp[j]     += temp;
            }
        }
        stable = 1;
        for (j = 1; j < LPC_ORDER; j++) {
            temp = cur_lsp[j - 1] + min_dist - cur_lsp[j] - 4;
            if (temp > 0) {
                stable = 0;
                break;
            }
        }
        if (stable)
            break;
    }
    if (!stable)
        memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(*cur_lsp));
}

/**
 * Bitexact implementation of 2ab scaled by 1/2^16.
 *
 * @param a 32 bit multiplicand
 * @param b 16 bit multiplier
 */
#define MULL2(a, b) \
        ((((a) >> 16) * (b) << 1) + (((a) & 0xffff) * (b) >> 15))

/**
 * Convert LSP frequencies to LPC coefficients.
 *
 * @param lpc buffer for LPC coefficients
 */
static void lsp2lpc(int16_t *lpc)
{
    int f1[LPC_ORDER / 2 + 1];
    int f2[LPC_ORDER / 2 + 1];
    int i, j;

    /* Calculate negative cosine */
    for (j = 0; j < LPC_ORDER; j++) {
        int index     = lpc[j] >> 7;
        int offset    = lpc[j] & 0x7f;
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        int temp1     = cos_tab[index] << 16;
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        int temp2     = (cos_tab[index + 1] - cos_tab[index]) *
                          ((offset << 8) + 0x80) << 1;

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        lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 16);
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    }

    /*
     * Compute sum and difference polynomial coefficients
     * (bitexact alternative to lsp2poly() in lsp.c)
     */
    /* Initialize with values in Q28 */
    f1[0] = 1 << 28;
    f1[1] = (lpc[0] << 14) + (lpc[2] << 14);
    f1[2] = lpc[0] * lpc[2] + (2 << 28);

    f2[0] = 1 << 28;
    f2[1] = (lpc[1] << 14) + (lpc[3] << 14);
    f2[2] = lpc[1] * lpc[3] + (2 << 28);

    /*
     * Calculate and scale the coefficients by 1/2 in
     * each iteration for a final scaling factor of Q25
     */
    for (i = 2; i < LPC_ORDER / 2; i++) {
        f1[i + 1] = f1[i - 1] + MULL2(f1[i], lpc[2 * i]);
        f2[i + 1] = f2[i - 1] + MULL2(f2[i], lpc[2 * i + 1]);

        for (j = i; j >= 2; j--) {
            f1[j] = MULL2(f1[j - 1], lpc[2 * i]) +
                    (f1[j] >> 1) + (f1[j - 2] >> 1);
            f2[j] = MULL2(f2[j - 1], lpc[2 * i + 1]) +
                    (f2[j] >> 1) + (f2[j - 2] >> 1);
        }

        f1[0] >>= 1;
        f2[0] >>= 1;
        f1[1] = ((lpc[2 * i]     << 16 >> i) + f1[1]) >> 1;
        f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1;
    }

    /* Convert polynomial coefficients to LPC coefficients */
    for (i = 0; i < LPC_ORDER / 2; i++) {
        int64_t ff1 = f1[i + 1] + f1[i];
        int64_t ff2 = f2[i + 1] - f2[i];

        lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) + (1 << 15)) >> 16;
        lpc[LPC_ORDER - i - 1] = av_clipl_int32(((ff1 - ff2) << 3) +
                                                (1 << 15)) >> 16;
    }
}

/**
 * Quantize LSP frequencies by interpolation and convert them to
 * the corresponding LPC coefficients.
 *
 * @param lpc      buffer for LPC coefficients
 * @param cur_lsp  the current LSP vector
 * @param prev_lsp the previous LSP vector
 */
static void lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp)
{
    int i;
    int16_t *lpc_ptr = lpc;

    /* cur_lsp * 0.25 + prev_lsp * 0.75 */
    ff_acelp_weighted_vector_sum(lpc, cur_lsp, prev_lsp,
                                 4096, 12288, 1 << 13, 14, LPC_ORDER);
    ff_acelp_weighted_vector_sum(lpc + LPC_ORDER, cur_lsp, prev_lsp,
                                 8192, 8192, 1 << 13, 14, LPC_ORDER);
    ff_acelp_weighted_vector_sum(lpc + 2 * LPC_ORDER, cur_lsp, prev_lsp,
                                 12288, 4096, 1 << 13, 14, LPC_ORDER);
    memcpy(lpc + 3 * LPC_ORDER, cur_lsp, LPC_ORDER * sizeof(*lpc));

    for (i = 0; i < SUBFRAMES; i++) {
        lsp2lpc(lpc_ptr);
        lpc_ptr += LPC_ORDER;
    }
}

/**
 * Generate a train of dirac functions with period as pitch lag.
 */
static void gen_dirac_train(int16_t *buf, int pitch_lag)
{
    int16_t vector[SUBFRAME_LEN];
    int i, j;

    memcpy(vector, buf, SUBFRAME_LEN * sizeof(*vector));
    for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) {
        for (j = 0; j < SUBFRAME_LEN - i; j++)
            buf[i + j] += vector[j];
    }
}

/**
 * Generate fixed codebook excitation vector.
 *
 * @param vector    decoded excitation vector
 * @param subfrm    current subframe
 * @param cur_rate  current bitrate
 * @param pitch_lag closed loop pitch lag
 * @param index     current subframe index
 */
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static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
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                               enum Rate cur_rate, int pitch_lag, int index)
{
    int temp, i, j;

    memset(vector, 0, SUBFRAME_LEN * sizeof(*vector));

    if (cur_rate == RATE_6300) {
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        if (subfrm->pulse_pos >= max_pos[index])
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            return;

        /* Decode amplitudes and positions */
        j = PULSE_MAX - pulses[index];
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        temp = subfrm->pulse_pos;
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        for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {
            temp -= combinatorial_table[j][i];
            if (temp >= 0)
                continue;
            temp += combinatorial_table[j++][i];
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            if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) {
                vector[subfrm->grid_index + GRID_SIZE * i] =
                                        -fixed_cb_gain[subfrm->amp_index];
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            } else {
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                vector[subfrm->grid_index + GRID_SIZE * i] =
                                         fixed_cb_gain[subfrm->amp_index];
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            }
            if (j == PULSE_MAX)
                break;
        }
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        if (subfrm->dirac_train == 1)
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            gen_dirac_train(vector, pitch_lag);
    } else { /* 5300 bps */
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        int cb_gain  = fixed_cb_gain[subfrm->amp_index];
        int cb_shift = subfrm->grid_index;
        int cb_sign  = subfrm->pulse_sign;
        int cb_pos   = subfrm->pulse_pos;
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        int offset, beta, lag;

        for (i = 0; i < 8; i += 2) {
            offset         = ((cb_pos & 7) << 3) + cb_shift + i;
            vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain;
            cb_pos  >>= 3;
            cb_sign >>= 1;
        }

        /* Enhance harmonic components */
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        lag  = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag +
               subfrm->ad_cb_lag - 1;
        beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1];
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        if (lag < SUBFRAME_LEN - 2) {
            for (i = lag; i < SUBFRAME_LEN; i++)
                vector[i] += beta * vector[i - lag] >> 15;
        }
    }
}

/**
 * Get delayed contribution from the previous excitation vector.
 */
static void get_residual(int16_t *residual, int16_t *prev_excitation, int lag)
{
    int offset = PITCH_MAX - PITCH_ORDER / 2 - lag;
    int i;

    residual[0] = prev_excitation[offset];
    residual[1] = prev_excitation[offset + 1];

    offset += 2;
    for (i = 2; i < SUBFRAME_LEN + PITCH_ORDER - 1; i++)
        residual[i] = prev_excitation[offset + (i - 2) % lag];
}

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static int dot_product(const int16_t *a, const int16_t *b, int length)
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{
    int i, sum = 0;

    for (i = 0; i < length; i++) {
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        int prod = a[i] * b[i];
        sum = av_sat_dadd32(sum, prod);
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    }
    return sum;
}

/**
 * Generate adaptive codebook excitation.
 */
static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
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                               int pitch_lag, G723_1_Subframe *subfrm,
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                               enum Rate cur_rate)
{
    int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
    const int16_t *cb_ptr;
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    int lag = pitch_lag + subfrm->ad_cb_lag - 1;
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    int i;
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    int sum;
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    get_residual(residual, prev_excitation, lag);

    /* Select quantization table */
    if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2)
        cb_ptr = adaptive_cb_gain85;
    else
        cb_ptr = adaptive_cb_gain170;

    /* Calculate adaptive vector */
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    cb_ptr += subfrm->ad_cb_gain * 20;
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    for (i = 0; i < SUBFRAME_LEN; i++) {
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        sum = dot_product(residual + i, cb_ptr, PITCH_ORDER);
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        vector[i] = av_sat_dadd32(1 << 15, sum) >> 16;
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    }
}

/**
 * Estimate maximum auto-correlation around pitch lag.
 *
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 * @param buf       buffer with offset applied
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 * @param offset    offset of the excitation vector
 * @param ccr_max   pointer to the maximum auto-correlation
 * @param pitch_lag decoded pitch lag
 * @param length    length of autocorrelation
 * @param dir       forward lag(1) / backward lag(-1)
 */
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static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,
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                        int pitch_lag, int length, int dir)
{
    int limit, ccr, lag = 0;
    int i;

    pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);
626 627 628 629
    if (dir > 0)
        limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);
    else
        limit = pitch_lag + 3;
630 631

    for (i = pitch_lag - 3; i <= limit; i++) {
632
        ccr = dot_product(buf, buf + dir * i, length);
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        if (ccr > *ccr_max) {
            *ccr_max = ccr;
            lag = i;
        }
    }
    return lag;
}

/**
 * Calculate pitch postfilter optimal and scaling gains.
 *
 * @param lag      pitch postfilter forward/backward lag
 * @param ppf      pitch postfilter parameters
 * @param cur_rate current bitrate
 * @param tgt_eng  target energy
 * @param ccr      cross-correlation
 * @param res_eng  residual energy
 */
static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate,
                           int tgt_eng, int ccr, int res_eng)
{
    int pf_residual;     /* square of postfiltered residual */
656
    int temp1, temp2;
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    ppf->index = lag;

    temp1 = tgt_eng * res_eng >> 1;
    temp2 = ccr * ccr << 1;

    if (temp2 > temp1) {
        if (ccr >= res_eng) {
            ppf->opt_gain = ppf_gain_weight[cur_rate];
        } else {
            ppf->opt_gain = (ccr << 15) / res_eng *
                            ppf_gain_weight[cur_rate] >> 15;
        }
        /* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */
        temp1       = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1);
        temp2       = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng;
673
        pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16;
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        if (tgt_eng >= pf_residual << 1) {
            temp1 = 0x7fff;
        } else {
            temp1 = (tgt_eng << 14) / pf_residual;
        }

        /* scaling_gain = sqrt(tgt_eng/pf_res^2) */
        ppf->sc_gain = square_root(temp1 << 16);
    } else {
        ppf->opt_gain = 0;
        ppf->sc_gain  = 0x7fff;
    }

    ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15);
}

/**
 * Calculate pitch postfilter parameters.
 *
 * @param p         the context
 * @param offset    offset of the excitation vector
 * @param pitch_lag decoded pitch lag
 * @param ppf       pitch postfilter parameters
 * @param cur_rate  current bitrate
 */
static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,
                           PPFParam *ppf, enum Rate cur_rate)
{

    int16_t scale;
    int i;
706
    int temp1, temp2;
707 708 709 710 711 712 713 714 715

    /*
     * 0 - target energy
     * 1 - forward cross-correlation
     * 2 - forward residual energy
     * 3 - backward cross-correlation
     * 4 - backward residual energy
     */
    int energy[5] = {0, 0, 0, 0, 0};
716
    int16_t *buf  = p->audio + LPC_ORDER + offset;
717
    int fwd_lag   = autocorr_max(buf, offset, &energy[1], pitch_lag,
718
                                 SUBFRAME_LEN, 1);
719
    int back_lag  = autocorr_max(buf, offset, &energy[3], pitch_lag,
720 721 722 723 724 725 726 727 728 729 730
                                 SUBFRAME_LEN, -1);

    ppf->index    = 0;
    ppf->opt_gain = 0;
    ppf->sc_gain  = 0x7fff;

    /* Case 0, Section 3.6 */
    if (!back_lag && !fwd_lag)
        return;

    /* Compute target energy */
731
    energy[0] = dot_product(buf, buf, SUBFRAME_LEN);
732 733 734

    /* Compute forward residual energy */
    if (fwd_lag)
735
        energy[2] = dot_product(buf + fwd_lag, buf + fwd_lag, SUBFRAME_LEN);
736 737 738

    /* Compute backward residual energy */
    if (back_lag)
739
        energy[4] = dot_product(buf - back_lag, buf - back_lag, SUBFRAME_LEN);
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    /* Normalize and shorten */
    temp1 = 0;
    for (i = 0; i < 5; i++)
        temp1 = FFMAX(energy[i], temp1);

    scale = normalize_bits(temp1, 31);
    for (i = 0; i < 5; i++)
        energy[i] = (energy[i] << scale) >> 16;

    if (fwd_lag && !back_lag) {  /* Case 1 */
        comp_ppf_gains(fwd_lag,  ppf, cur_rate, energy[0], energy[1],
                       energy[2]);
    } else if (!fwd_lag) {       /* Case 2 */
        comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
                       energy[4]);
    } else {                     /* Case 3 */

        /*
         * Select the largest of energy[1]^2/energy[2]
         * and energy[3]^2/energy[4]
         */
        temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15);
        temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15);
        if (temp1 >= temp2) {
            comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
                           energy[2]);
        } else {
            comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
                           energy[4]);
        }
    }
}

/**
 * Classify frames as voiced/unvoiced.
 *
 * @param p         the context
 * @param pitch_lag decoded pitch_lag
 * @param exc_eng   excitation energy estimation
 * @param scale     scaling factor of exc_eng
 *
 * @return residual interpolation index if voiced, 0 otherwise
 */
static int comp_interp_index(G723_1_Context *p, int pitch_lag,
                             int *exc_eng, int *scale)
{
    int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
788
    int16_t *buf = p->audio + LPC_ORDER;
789 790 791

    int index, ccr, tgt_eng, best_eng, temp;

792 793
    *scale = scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX);
    buf   += offset;
794 795 796

    /* Compute maximum backward cross-correlation */
    ccr   = 0;
797
    index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
798
    ccr   = av_sat_add32(ccr, 1 << 15) >> 16;
799 800

    /* Compute target energy */
801
    tgt_eng  = dot_product(buf, buf, SUBFRAME_LEN * 2);
802
    *exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16;
803 804 805 806 807

    if (ccr <= 0)
        return 0;

    /* Compute best energy */
808
    best_eng = dot_product(buf - index, buf - index, SUBFRAME_LEN * 2);
809
    best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;
810 811 812 813 814 815 816 817 818 819 820 821 822 823 824 825 826 827 828 829 830 831 832 833 834 835

    temp = best_eng * *exc_eng >> 3;

    if (temp < ccr * ccr)
        return index;
    else
        return 0;
}

/**
 * Peform residual interpolation based on frame classification.
 *
 * @param buf   decoded excitation vector
 * @param out   output vector
 * @param lag   decoded pitch lag
 * @param gain  interpolated gain
 * @param rseed seed for random number generator
 */
static void residual_interp(int16_t *buf, int16_t *out, int lag,
                            int gain, int *rseed)
{
    int i;
    if (lag) { /* Voiced */
        int16_t *vector_ptr = buf + PITCH_MAX;
        /* Attenuate */
        for (i = 0; i < lag; i++)
836 837 838
            out[i] = vector_ptr[i - lag] * 3 >> 2;
        av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out),
                          (FRAME_LEN - lag) * sizeof(*out));
839 840 841 842 843 844 845 846 847 848 849 850 851 852 853 854 855 856 857 858 859 860 861 862 863 864 865 866 867 868 869 870 871 872 873 874 875 876 877 878 879 880 881 882 883 884 885 886
    } else {  /* Unvoiced */
        for (i = 0; i < FRAME_LEN; i++) {
            *rseed = *rseed * 521 + 259;
            out[i] = gain * *rseed >> 15;
        }
        memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(*buf));
    }
}

/**
 * Perform IIR filtering.
 *
 * @param fir_coef FIR coefficients
 * @param iir_coef IIR coefficients
 * @param src      source vector
 * @param dest     destination vector
 */
static inline void iir_filter(int16_t *fir_coef, int16_t *iir_coef,
                              int16_t *src, int *dest)
{
    int m, n;

    for (m = 0; m < SUBFRAME_LEN; m++) {
        int64_t filter = 0;
        for (n = 1; n <= LPC_ORDER; n++) {
            filter -= fir_coef[n - 1] * src[m - n] -
                      iir_coef[n - 1] * (dest[m - n] >> 16);
        }

        dest[m] = av_clipl_int32((src[m] << 16) + (filter << 3) + (1 << 15));
    }
}

/**
 * Adjust gain of postfiltered signal.
 *
 * @param p      the context
 * @param buf    postfiltered output vector
 * @param energy input energy coefficient
 */
static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
{
    int num, denom, gain, bits1, bits2;
    int i;

    num   = energy;
    denom = 0;
    for (i = 0; i < SUBFRAME_LEN; i++) {
887 888
        int temp = buf[i] >> 2;
        temp *= temp;
889
        denom = av_sat_dadd32(denom, temp);
890 891 892 893 894 895 896 897 898 899 900 901 902 903 904 905 906 907
    }

    if (num && denom) {
        bits1   = normalize_bits(num,   31);
        bits2   = normalize_bits(denom, 31);
        num     = num << bits1 >> 1;
        denom <<= bits2;

        bits2 = 5 + bits1 - bits2;
        bits2 = FFMAX(0, bits2);

        gain = (num >> 1) / (denom >> 16);
        gain = square_root(gain << 16 >> bits2);
    } else {
        gain = 1 << 12;
    }

    for (i = 0; i < SUBFRAME_LEN; i++) {
908
        p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4;
909 910 911 912 913 914 915 916 917 918
        buf[i]     = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) +
                                   (1 << 10)) >> 11);
    }
}

/**
 * Perform formant filtering.
 *
 * @param p   the context
 * @param lpc quantized lpc coefficients
919 920
 * @param buf input buffer
 * @param dst output buffer
921
 */
922 923
static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
                               int16_t *buf, int16_t *dst)
924
{
925
    int16_t filter_coef[2][LPC_ORDER];
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    int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;
    int i, j, k;

    memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf));
    memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal));

    for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
        for (k = 0; k < LPC_ORDER; k++) {
            filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] +
                                 (1 << 14)) >> 15;
            filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] +
                                 (1 << 14)) >> 15;
        }
        iir_filter(filter_coef[0], filter_coef[1], buf + i,
                   filter_signal + i);
941
        lpc += LPC_ORDER;
942 943 944 945 946 947
    }

    memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(*p->fir_mem));
    memcpy(p->iir_mem, filter_signal + FRAME_LEN,
           LPC_ORDER * sizeof(*p->iir_mem));

948
    buf += LPC_ORDER;
949 950
    signal_ptr = filter_signal + LPC_ORDER;
    for (i = 0; i < SUBFRAMES; i++) {
951
        int temp;
952 953 954 955
        int auto_corr[2];
        int scale, energy;

        /* Normalize */
956
        scale = scale_vector(dst, buf, SUBFRAME_LEN);
957 958

        /* Compute auto correlation coefficients */
959 960
        auto_corr[0] = dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
        auto_corr[1] = dot_product(dst, dst,     SUBFRAME_LEN);
961 962 963 964 965 966

        /* Compute reflection coefficient */
        temp = auto_corr[1] >> 16;
        if (temp) {
            temp = (auto_corr[0] >> 2) / temp;
        }
967
        p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2;
968
        temp = -p->reflection_coef >> 1 & ~3;
969 970 971

        /* Compensation filter */
        for (j = 0; j < SUBFRAME_LEN; j++) {
972
            dst[j] = av_sat_dadd32(signal_ptr[j],
973
                                   (signal_ptr[j - 1] >> 16) * temp) >> 16;
974 975 976 977 978 979 980 981 982
        }

        /* Compute normalized signal energy */
        temp = 2 * scale + 4;
        if (temp < 0) {
            energy = av_clipl_int32((int64_t)auto_corr[1] << -temp);
        } else
            energy = auto_corr[1] >> temp;

983
        gain_scale(p, dst, energy);
984

985
        buf        += SUBFRAME_LEN;
986
        signal_ptr += SUBFRAME_LEN;
987
        dst        += SUBFRAME_LEN;
988 989 990
    }
}

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static int sid_gain_to_lsp_index(int gain)
{
    if (gain < 0x10)
        return gain << 6;
    else if (gain < 0x20)
        return gain - 8 << 7;
    else
        return gain - 20 << 8;
}

static inline int cng_rand(int *state, int base)
{
    *state = (*state * 521 + 259) & 0xFFFF;
    return (*state & 0x7FFF) * base >> 15;
}

static int estimate_sid_gain(G723_1_Context *p)
{
    int i, shift, seg, seg2, t, val, val_add, x, y;

    shift = 16 - p->cur_gain * 2;
    if (shift > 0)
        t = p->sid_gain << shift;
    else
        t = p->sid_gain >> -shift;
    x = t * cng_filt[0] >> 16;

    if (x >= cng_bseg[2])
        return 0x3F;

    if (x >= cng_bseg[1]) {
        shift = 4;
        seg   = 3;
    } else {
        shift = 3;
        seg   = (x >= cng_bseg[0]);
    }
    seg2 = FFMIN(seg, 3);

    val     = 1 << shift;
    val_add = val >> 1;
    for (i = 0; i < shift; i++) {
        t = seg * 32 + (val << seg2);
        t *= t;
        if (x >= t)
            val += val_add;
        else
            val -= val_add;
        val_add >>= 1;
    }

    t = seg * 32 + (val << seg2);
    y = t * t - x;
    if (y <= 0) {
        t = seg * 32 + (val + 1 << seg2);
        t = t * t - x;
        val = (seg2 - 1 << 4) + val;
        if (t >= y)
            val++;
    } else {
        t = seg * 32 + (val - 1 << seg2);
        t = t * t - x;
        val = (seg2 - 1 << 4) + val;
        if (t >= y)
            val--;
    }

    return val;
}

static void generate_noise(G723_1_Context *p)
{
    int i, j, idx, t;
    int off[SUBFRAMES];
    int signs[SUBFRAMES / 2 * 11], pos[SUBFRAMES / 2 * 11];
    int tmp[SUBFRAME_LEN * 2];
    int16_t *vector_ptr;
    int64_t sum;
    int b0, c, delta, x, shift;

    p->pitch_lag[0] = cng_rand(&p->cng_random_seed, 21) + 123;
    p->pitch_lag[1] = cng_rand(&p->cng_random_seed, 19) + 123;

    for (i = 0; i < SUBFRAMES; i++) {
        p->subframe[i].ad_cb_gain = cng_rand(&p->cng_random_seed, 50) + 1;
        p->subframe[i].ad_cb_lag  = cng_adaptive_cb_lag[i];
    }

    for (i = 0; i < SUBFRAMES / 2; i++) {
        t = cng_rand(&p->cng_random_seed, 1 << 13);
        off[i * 2]     =   t       & 1;
        off[i * 2 + 1] = ((t >> 1) & 1) + SUBFRAME_LEN;
        t >>= 2;
        for (j = 0; j < 11; j++) {
            signs[i * 11 + j] = (t & 1) * 2 - 1 << 14;
            t >>= 1;
        }
    }

    idx = 0;
    for (i = 0; i < SUBFRAMES; i++) {
        for (j = 0; j < SUBFRAME_LEN / 2; j++)
            tmp[j] = j;
        t = SUBFRAME_LEN / 2;
        for (j = 0; j < pulses[i]; j++, idx++) {
            int idx2 = cng_rand(&p->cng_random_seed, t);

            pos[idx]  = tmp[idx2] * 2 + off[i];
            tmp[idx2] = tmp[--t];
        }
    }

    vector_ptr = p->audio + LPC_ORDER;
    memcpy(vector_ptr, p->prev_excitation,
           PITCH_MAX * sizeof(*p->excitation));
    for (i = 0; i < SUBFRAMES; i += 2) {
        gen_acb_excitation(vector_ptr, vector_ptr,
                           p->pitch_lag[i >> 1], &p->subframe[i],
                           p->cur_rate);
        gen_acb_excitation(vector_ptr + SUBFRAME_LEN,
                           vector_ptr + SUBFRAME_LEN,
                           p->pitch_lag[i >> 1], &p->subframe[i + 1],
                           p->cur_rate);

        t = 0;
        for (j = 0; j < SUBFRAME_LEN * 2; j++)
            t |= FFABS(vector_ptr[j]);
        t = FFMIN(t, 0x7FFF);
        if (!t) {
            shift = 0;
        } else {
            shift = -10 + av_log2(t);
            if (shift < -2)
                shift = -2;
        }
        sum = 0;
        if (shift < 0) {
           for (j = 0; j < SUBFRAME_LEN * 2; j++) {
               t      = vector_ptr[j] << -shift;
               sum   += t * t;
               tmp[j] = t;
           }
        } else {
           for (j = 0; j < SUBFRAME_LEN * 2; j++) {
               t      = vector_ptr[j] >> shift;
               sum   += t * t;
               tmp[j] = t;
           }
        }

        b0 = 0;
        for (j = 0; j < 11; j++)
            b0 += tmp[pos[(i / 2) * 11 + j]] * signs[(i / 2) * 11 + j];
        b0 = b0 * 2 * 2979LL + (1 << 29) >> 30; // approximated division by 11

        c = p->cur_gain * (p->cur_gain * SUBFRAME_LEN >> 5);
        if (shift * 2 + 3 >= 0)
            c >>= shift * 2 + 3;
        else
            c <<= -(shift * 2 + 3);
        c = (av_clipl_int32(sum << 1) - c) * 2979LL >> 15;

        delta = b0 * b0 * 2 - c;
        if (delta <= 0) {
            x = -b0;
        } else {
            delta = square_root(delta);
            x     = delta - b0;
            t     = delta + b0;
            if (FFABS(t) < FFABS(x))
                x = -t;
        }
        shift++;
        if (shift < 0)
           x >>= -shift;
        else
           x <<= shift;
        x = av_clip(x, -10000, 10000);

        for (j = 0; j < 11; j++) {
            idx = (i / 2) * 11 + j;
            vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] +
                                                 (x * signs[idx] >> 15));
        }

        /* copy decoded data to serve as a history for the next decoded subframes */
        memcpy(vector_ptr + PITCH_MAX, vector_ptr,
               sizeof(*vector_ptr) * SUBFRAME_LEN * 2);
        vector_ptr += SUBFRAME_LEN * 2;
    }
    /* Save the excitation for the next frame */
    memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN,
           PITCH_MAX * sizeof(*p->excitation));
}

1186 1187 1188 1189
static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
                               int *got_frame_ptr, AVPacket *avpkt)
{
    G723_1_Context *p  = avctx->priv_data;
1190
    AVFrame *frame     = data;
1191 1192 1193 1194 1195 1196 1197 1198
    const uint8_t *buf = avpkt->data;
    int buf_size       = avpkt->size;
    int dec_mode       = buf[0] & 3;

    PPFParam ppf[SUBFRAMES];
    int16_t cur_lsp[LPC_ORDER];
    int16_t lpc[SUBFRAMES * LPC_ORDER];
    int16_t acb_vector[SUBFRAME_LEN];
1199
    int16_t *out;
1200
    int bad_frame = 0, i, j, ret;
1201
    int16_t *audio = p->audio;
1202 1203 1204 1205 1206 1207 1208 1209 1210 1211 1212 1213 1214 1215 1216 1217 1218 1219

    if (buf_size < frame_size[dec_mode]) {
        if (buf_size)
            av_log(avctx, AV_LOG_WARNING,
                   "Expected %d bytes, got %d - skipping packet\n",
                   frame_size[dec_mode], buf_size);
        *got_frame_ptr = 0;
        return buf_size;
    }

    if (unpack_bitstream(p, buf, buf_size) < 0) {
        bad_frame = 1;
        if (p->past_frame_type == ACTIVE_FRAME)
            p->cur_frame_type = ACTIVE_FRAME;
        else
            p->cur_frame_type = UNTRANSMITTED_FRAME;
    }

1220
    frame->nb_samples = FRAME_LEN;
1221
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
1222 1223 1224 1225
         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;
    }

1226
    out = (int16_t *)frame->data[0];
1227

1228 1229 1230 1231 1232 1233 1234 1235 1236 1237 1238 1239 1240 1241 1242 1243
    if (p->cur_frame_type == ACTIVE_FRAME) {
        if (!bad_frame)
            p->erased_frames = 0;
        else if (p->erased_frames != 3)
            p->erased_frames++;

        inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
        lsp_interpolate(lpc, cur_lsp, p->prev_lsp);

        /* Save the lsp_vector for the next frame */
        memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));

        /* Generate the excitation for the frame */
        memcpy(p->excitation, p->prev_excitation,
               PITCH_MAX * sizeof(*p->excitation));
        if (!p->erased_frames) {
1244 1245
            int16_t *vector_ptr = p->excitation + PITCH_MAX;

1246 1247 1248 1249
            /* Update interpolation gain memory */
            p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index +
                                            p->subframe[3].amp_index) >> 1];
            for (i = 0; i < SUBFRAMES; i++) {
1250
                gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
1251 1252
                                   p->pitch_lag[i >> 1], i);
                gen_acb_excitation(acb_vector, &p->excitation[SUBFRAME_LEN * i],
1253
                                   p->pitch_lag[i >> 1], &p->subframe[i],
1254 1255 1256
                                   p->cur_rate);
                /* Get the total excitation */
                for (j = 0; j < SUBFRAME_LEN; j++) {
1257 1258
                    int v = av_clip_int16(vector_ptr[j] << 1);
                    vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
1259 1260 1261 1262 1263 1264 1265 1266 1267
                }
                vector_ptr += SUBFRAME_LEN;
            }

            vector_ptr = p->excitation + PITCH_MAX;

            p->interp_index = comp_interp_index(p, p->pitch_lag[1],
                                                &p->sid_gain, &p->cur_gain);

1268
            /* Peform pitch postfiltering */
1269 1270 1271 1272 1273 1274 1275 1276 1277 1278 1279 1280 1281
            if (p->postfilter) {
                i = PITCH_MAX;
                for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
                    comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
                                   ppf + j, p->cur_rate);

                for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
                    ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
                                                 vector_ptr + i,
                                                 vector_ptr + i + ppf[j].index,
                                                 ppf[j].sc_gain,
                                                 ppf[j].opt_gain,
                                                 1 << 14, 15, SUBFRAME_LEN);
1282 1283 1284
            } else {
                audio = vector_ptr - LPC_ORDER;
            }
1285

1286 1287 1288
            /* Save the excitation for the next frame */
            memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
                   PITCH_MAX * sizeof(*p->excitation));
1289 1290 1291 1292 1293 1294
        } else {
            p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
            if (p->erased_frames == 3) {
                /* Mute output */
                memset(p->excitation, 0,
                       (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
1295 1296
                memset(p->prev_excitation, 0,
                       PITCH_MAX * sizeof(*p->excitation));
1297
                memset(frame->data[0], 0,
1298 1299
                       (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
            } else {
1300 1301
                int16_t *buf = p->audio + LPC_ORDER;

1302
                /* Regenerate frame */
1303
                residual_interp(p->excitation, buf, p->interp_index,
1304
                                p->interp_gain, &p->random_seed);
1305 1306 1307 1308

                /* Save the excitation for the next frame */
                memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
                       PITCH_MAX * sizeof(*p->excitation));
1309 1310
            }
        }
1311
        p->cng_random_seed = CNG_RANDOM_SEED;
1312
    } else {
1313 1314 1315 1316 1317 1318
        if (p->cur_frame_type == SID_FRAME) {
            p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index);
            inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);
        } else if (p->past_frame_type == ACTIVE_FRAME) {
            p->sid_gain = estimate_sid_gain(p);
        }
1319

1320 1321 1322 1323 1324 1325 1326 1327
        if (p->past_frame_type == ACTIVE_FRAME)
            p->cur_gain = p->sid_gain;
        else
            p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
        generate_noise(p);
        lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);
        /* Save the lsp_vector for the next frame */
        memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
1328 1329 1330 1331 1332 1333 1334
    }

    p->past_frame_type = p->cur_frame_type;

    memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
    for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
        ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
1335
                                    audio + i, SUBFRAME_LEN, LPC_ORDER,
1336 1337 1338
                                    0, 1, 1 << 12);
    memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));

1339
    if (p->postfilter) {
1340
        formant_postfilter(p, lpc, p->audio, out);
1341 1342 1343 1344
    } else { // if output is not postfiltered it should be scaled by 2
        for (i = 0; i < FRAME_LEN; i++)
            out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);
    }
1345

1346
    *got_frame_ptr = 1;
1347 1348 1349 1350 1351 1352 1353 1354 1355

    return frame_size[dec_mode];
}

#define OFFSET(x) offsetof(G723_1_Context, x)
#define AD     AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM

static const AVOption options[] = {
    { "postfilter", "postfilter on/off", OFFSET(postfilter), AV_OPT_TYPE_INT,
1356
      { .i64 = 1 }, 0, 1, AD },
1357 1358 1359 1360 1361 1362 1363 1364 1365 1366 1367 1368 1369
    { NULL }
};


static const AVClass g723_1dec_class = {
    .class_name = "G.723.1 decoder",
    .item_name  = av_default_item_name,
    .option     = options,
    .version    = LIBAVUTIL_VERSION_INT,
};

AVCodec ff_g723_1_decoder = {
    .name           = "g723_1",
1370
    .long_name      = NULL_IF_CONFIG_SMALL("G.723.1"),
1371
    .type           = AVMEDIA_TYPE_AUDIO,
1372
    .id             = AV_CODEC_ID_G723_1,
1373 1374 1375
    .priv_data_size = sizeof(G723_1_Context),
    .init           = g723_1_decode_init,
    .decode         = g723_1_decode_frame,
1376
    .capabilities   = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
1377 1378
    .priv_class     = &g723_1dec_class,
};