g723_1: Rename files to better reflect their purpose

Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>

1 files changed, 0 insertions, 1378 deletions

diff --git a/libavcodec/g723_1.c b/libavcodec/g723_1.c
deleted file mode 100644
index 80c7d1f491..0000000000
--- a/libavcodec/g723_1.c
+++ /dev/null
@@ -1,1378 +0,0 @@
-/*
- * 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
-#include "libavutil/channel_layout.h"
-#include "libavutil/mem.h"
-#include "libavutil/opt.h"
-#include "avcodec.h"
-#include "get_bits.h"
-#include "acelp_vectors.h"
-#include "celp_filters.h"
-#include "g723_1_data.h"
-#include "internal.h"
-
-#define CNG_RANDOM_SEED 12345
-
-/**
- * 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
- */
-typedef struct G723_1_Subframe {
-    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
- */
-typedef struct PPFParam {
-    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];
-    int16_t sid_lsp[LPC_ORDER];
-    int16_t prev_excitation[PITCH_MAX];
-    int16_t excitation[PITCH_MAX + FRAME_LEN + 4];
-    int16_t synth_mem[LPC_ORDER];
-    int16_t fir_mem[LPC_ORDER];
-    int     iir_mem[LPC_ORDER];
-
-    int random_seed;
-    int cng_random_seed;
-    int interp_index;
-    int interp_gain;
-    int sid_gain;
-    int cur_gain;
-    int reflection_coef;
-    int pf_gain;
-    int postfilter;
-
-    int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4];
-} 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));
-    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;
-
-    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)
-{
-    return width - av_log2(num) - 1;
-}
-
-/**
- * Scale vector contents based on the largest of their absolutes.
- */
-static int scale_vector(int16_t *dst, const int16_t *vector, int length)
-{
-    int bits, max = 0;
-    int i;
-
-
-    for (i = 0; i < length; i++)
-        max |= FFABS(vector[i]);
-
-    max   = FFMIN(max, 0x7FFF);
-    bits  = normalize_bits(max, 15);
-
-    for (i = 0; i < length; i++)
-        dst[i] = vector[i] << bits >> 3;
-
-    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) & 0x1FF;
-        int offset    = lpc[j] & 0x7f;
-        int temp1     = cos_tab[index] << 16;
-        int temp2     = (cos_tab[index + 1] - cos_tab[index]) *
-                          ((offset << 8) + 0x80) << 1;
-
-        lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 16);
-    }
-
-    /*
-     * 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
- */
-static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
-                               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) {
-        if (subfrm->pulse_pos >= max_pos[index])
-            return;
-
-        /* Decode amplitudes and positions */
-        j = PULSE_MAX - pulses[index];
-        temp = subfrm->pulse_pos;
-        for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {
-            temp -= combinatorial_table[j][i];
-            if (temp >= 0)
-                continue;
-            temp += combinatorial_table[j++][i];
-            if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) {
-                vector[subfrm->grid_index + GRID_SIZE * i] =
-                                        -fixed_cb_gain[subfrm->amp_index];
-            } else {
-                vector[subfrm->grid_index + GRID_SIZE * i] =
-                                         fixed_cb_gain[subfrm->amp_index];
-            }
-            if (j == PULSE_MAX)
-                break;
-        }
-        if (subfrm->dirac_train == 1)
-            gen_dirac_train(vector, pitch_lag);
-    } else { /* 5300 bps */
-        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;
-        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 */
-        lag  = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag +
-               subfrm->ad_cb_lag - 1;
-        beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1];
-
-        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];
-}
-
-static int dot_product(const int16_t *a, const int16_t *b, int length)
-{
-    int i, sum = 0;
-
-    for (i = 0; i < length; i++) {
-        int prod = a[i] * b[i];
-        sum = av_sat_dadd32(sum, prod);
-    }
-    return sum;
-}
-
-/**
- * Generate adaptive codebook excitation.
- */
-static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
-                               int pitch_lag, G723_1_Subframe *subfrm,
-                               enum Rate cur_rate)
-{
-    int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
-    const int16_t *cb_ptr;
-    int lag = pitch_lag + subfrm->ad_cb_lag - 1;
-
-    int i;
-    int sum;
-
-    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 */
-    cb_ptr += subfrm->ad_cb_gain * 20;
-    for (i = 0; i < SUBFRAME_LEN; i++) {
-        sum = dot_product(residual + i, cb_ptr, PITCH_ORDER);
-        vector[i] = av_sat_dadd32(1 << 15, sum) >> 16;
-    }
-}
-
-/**
- * Estimate maximum auto-correlation around pitch lag.
- *
- * @param buf       buffer with offset applied
- * @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)
- */
-static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,
-                        int pitch_lag, int length, int dir)
-{
-    int limit, ccr, lag = 0;
-    int i;
-
-    pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);
-    if (dir > 0)
-        limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);
-    else
-        limit = pitch_lag + 3;
-
-    for (i = pitch_lag - 3; i <= limit; i++) {
-        ccr = dot_product(buf, buf + dir * i, length);
-
-        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 */
-    int temp1, temp2;
-
-    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;
-        pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16;
-
-        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;
-    int temp1, temp2;
-
-    /*
-     * 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};
-    int16_t *buf  = p->audio + LPC_ORDER + offset;
-    int fwd_lag   = autocorr_max(buf, offset, &energy[1], pitch_lag,
-                                 SUBFRAME_LEN, 1);
-    int back_lag  = autocorr_max(buf, offset, &energy[3], pitch_lag,
-                                 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 */
-    energy[0] = dot_product(buf, buf, SUBFRAME_LEN);
-
-    /* Compute forward residual energy */
-    if (fwd_lag)
-        energy[2] = dot_product(buf + fwd_lag, buf + fwd_lag, SUBFRAME_LEN);
-
-    /* Compute backward residual energy */
-    if (back_lag)
-        energy[4] = dot_product(buf - back_lag, buf - back_lag, SUBFRAME_LEN);
-
-    /* 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;
-    int16_t *buf = p->audio + LPC_ORDER;
-
-    int index, ccr, tgt_eng, best_eng, temp;
-
-    *scale = scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX);
-    buf   += offset;
-
-    /* Compute maximum backward cross-correlation */
-    ccr   = 0;
-    index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
-    ccr   = av_sat_add32(ccr, 1 << 15) >> 16;
-
-    /* Compute target energy */
-    tgt_eng  = dot_product(buf, buf, SUBFRAME_LEN * 2);
-    *exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16;
-
-    if (ccr <= 0)
-        return 0;
-
-    /* Compute best energy */
-    best_eng = dot_product(buf - index, buf - index, SUBFRAME_LEN * 2);
-    best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;
-
-    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++)
-            out[i] = vector_ptr[i - lag] * 3 >> 2;
-        av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out),
-                          (FRAME_LEN - lag) * sizeof(*out));
-    } 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++) {
-        int temp = buf[i] >> 2;
-        temp *= temp;
-        denom = av_sat_dadd32(denom, temp);
-    }
-
-    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++) {
-        p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4;
-        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
- * @param buf input buffer
- * @param dst output buffer
- */
-static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
-                               int16_t *buf, int16_t *dst)
-{
-    int16_t filter_coef[2][LPC_ORDER];
-    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);
-        lpc += LPC_ORDER;
-    }
-
-    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));
-
-    buf += LPC_ORDER;
-    signal_ptr = filter_signal + LPC_ORDER;
-    for (i = 0; i < SUBFRAMES; i++) {
-        int temp;
-        int auto_corr[2];
-        int scale, energy;
-
-        /* Normalize */
-        scale = scale_vector(dst, buf, SUBFRAME_LEN);
-
-        /* Compute auto correlation coefficients */
-        auto_corr[0] = dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
-        auto_corr[1] = dot_product(dst, dst,     SUBFRAME_LEN);
-
-        /* Compute reflection coefficient */
-        temp = auto_corr[1] >> 16;
-        if (temp) {
-            temp = (auto_corr[0] >> 2) / temp;
-        }
-        p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2;
-        temp = -p->reflection_coef >> 1 & ~3;
-
-        /* Compensation filter */
-        for (j = 0; j < SUBFRAME_LEN; j++) {
-            dst[j] = av_sat_dadd32(signal_ptr[j],
-                                   (signal_ptr[j - 1] >> 16) * temp) >> 16;
-        }
-
-        /* 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;
-
-        gain_scale(p, dst, energy);
-
-        buf        += SUBFRAME_LEN;
-        signal_ptr += SUBFRAME_LEN;
-        dst        += SUBFRAME_LEN;
-    }
-}
-
-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));
-}
-
-static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
-                               int *got_frame_ptr, AVPacket *avpkt)
-{
-    G723_1_Context *p  = avctx->priv_data;
-    AVFrame *frame     = data;
-    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];
-    int16_t *out;
-    int bad_frame = 0, i, j, ret;
-    int16_t *audio = p->audio;
-
-    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;
-    }
-
-    frame->nb_samples = FRAME_LEN;
-    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
-         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
-         return ret;
-    }
-
-    out = (int16_t *)frame->data[0];
-
-    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) {
-            int16_t *vector_ptr = p->excitation + PITCH_MAX;
-
-            /* 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++) {
-                gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
-                                   p->pitch_lag[i >> 1], i);
-                gen_acb_excitation(acb_vector, &p->excitation[SUBFRAME_LEN * i],
-                                   p->pitch_lag[i >> 1], &p->subframe[i],
-                                   p->cur_rate);
-                /* Get the total excitation */
-                for (j = 0; j < SUBFRAME_LEN; j++) {
-                    int v = av_clip_int16(vector_ptr[j] << 1);
-                    vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
-                }
-                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);
-
-            /* Peform pitch postfiltering */
-            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);
-            } else {
-                audio = vector_ptr - LPC_ORDER;
-            }
-
-            /* Save the excitation for the next frame */
-            memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
-                   PITCH_MAX * sizeof(*p->excitation));
-        } 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));
-                memset(p->prev_excitation, 0,
-                       PITCH_MAX * sizeof(*p->excitation));
-                memset(frame->data[0], 0,
-                       (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
-            } else {
-                int16_t *buf = p->audio + LPC_ORDER;
-
-                /* Regenerate frame */
-                residual_interp(p->excitation, buf, p->interp_index,
-                                p->interp_gain, &p->random_seed);
-
-                /* Save the excitation for the next frame */
-                memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
-                       PITCH_MAX * sizeof(*p->excitation));
-            }
-        }
-        p->cng_random_seed = CNG_RANDOM_SEED;
-    } else {
-        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);
-        }
-
-        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));
-    }
-
-    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],
-                                    audio + i, SUBFRAME_LEN, LPC_ORDER,
-                                    0, 1, 1 << 12);
-    memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
-
-    if (p->postfilter) {
-        formant_postfilter(p, lpc, p->audio, out);
-    } 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);
-    }
-
-    *got_frame_ptr = 1;
-
-    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,
-      { .i64 = 1 }, 0, 1, AD },
-    { 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",
-    .long_name      = NULL_IF_CONFIG_SMALL("G.723.1"),
-    .type           = AVMEDIA_TYPE_AUDIO,
-    .id             = AV_CODEC_ID_G723_1,
-    .priv_data_size = sizeof(G723_1_Context),
-    .init           = g723_1_decode_init,
-    .decode         = g723_1_decode_frame,
-    .capabilities   = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
-    .priv_class     = &g723_1dec_class,
-};