Merge commit 'f023d57d355ff3b917f1aad9b03db5c293ec4244'

* commit 'f023d57d355ff3b917f1aad9b03db5c293ec4244':
  lavc: G.723.1 encoder

Split existing FFmpeg G.723.1 encoder into a new file.

Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>

1 files changed, 0 insertions, 1145 deletions

diff --git a/libavcodec/g723_1dec.c b/libavcodec/g723_1dec.c
index 47d22b54bc..3e8c4897d2 100644
--- a/libavcodec/g723_1dec.c
+++ b/libavcodec/g723_1dec.c
@@ -186,9 +186,6 @@ static int16_t square_root(unsigned val)
     return (ff_sqrt(val << 1) >> 1) & (~1);
 }
 
-#define normalize_bits_int16(num) ff_g723_1_normalize_bits(num, 15)
-#define normalize_bits_int32(num) ff_g723_1_normalize_bits(num, 31)
-
 /**
  * Generate fixed codebook excitation vector.
  *
@@ -1028,1145 +1025,3 @@ AVCodec ff_g723_1_decoder = {
     .capabilities   = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
     .priv_class     = &g723_1dec_class,
 };
-
-#if CONFIG_G723_1_ENCODER
-#define BITSTREAM_WRITER_LE
-#include "put_bits.h"
-
-static av_cold int g723_1_encode_init(AVCodecContext *avctx)
-{
-    G723_1_Context *p = avctx->priv_data;
-
-    if (avctx->sample_rate != 8000) {
-        av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");
-        return -1;
-    }
-
-    if (avctx->channels != 1) {
-        av_log(avctx, AV_LOG_ERROR, "Only mono supported\n");
-        return AVERROR(EINVAL);
-    }
-
-    if (avctx->bit_rate == 6300) {
-        p->cur_rate = RATE_6300;
-    } else if (avctx->bit_rate == 5300) {
-        av_log(avctx, AV_LOG_ERROR, "Bitrate not supported yet, use 6.3k\n");
-        return AVERROR_PATCHWELCOME;
-    } else {
-        av_log(avctx, AV_LOG_ERROR,
-               "Bitrate not supported, use 6.3k\n");
-        return AVERROR(EINVAL);
-    }
-    avctx->frame_size = 240;
-    memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t));
-
-    return 0;
-}
-
-/**
- * Remove DC component from the input signal.
- *
- * @param buf input signal
- * @param fir zero memory
- * @param iir pole memory
- */
-static void highpass_filter(int16_t *buf, int16_t *fir, int *iir)
-{
-    int i;
-    for (i = 0; i < FRAME_LEN; i++) {
-        *iir   = (buf[i] << 15) + ((-*fir) << 15) + MULL2(*iir, 0x7f00);
-        *fir   = buf[i];
-        buf[i] = av_clipl_int32((int64_t)*iir + (1 << 15)) >> 16;
-    }
-}
-
-/**
- * Estimate autocorrelation of the input vector.
- *
- * @param buf      input buffer
- * @param autocorr autocorrelation coefficients vector
- */
-static void comp_autocorr(int16_t *buf, int16_t *autocorr)
-{
-    int i, scale, temp;
-    int16_t vector[LPC_FRAME];
-
-    ff_g723_1_scale_vector(vector, buf, LPC_FRAME);
-
-    /* Apply the Hamming window */
-    for (i = 0; i < LPC_FRAME; i++)
-        vector[i] = (vector[i] * hamming_window[i] + (1 << 14)) >> 15;
-
-    /* Compute the first autocorrelation coefficient */
-    temp = ff_dot_product(vector, vector, LPC_FRAME);
-
-    /* Apply a white noise correlation factor of (1025/1024) */
-    temp += temp >> 10;
-
-    /* Normalize */
-    scale = normalize_bits_int32(temp);
-    autocorr[0] = av_clipl_int32((int64_t)(temp << scale) +
-                                 (1 << 15)) >> 16;
-
-    /* Compute the remaining coefficients */
-    if (!autocorr[0]) {
-        memset(autocorr + 1, 0, LPC_ORDER * sizeof(int16_t));
-    } else {
-        for (i = 1; i <= LPC_ORDER; i++) {
-           temp = ff_dot_product(vector, vector + i, LPC_FRAME - i);
-           temp = MULL2((temp << scale), binomial_window[i - 1]);
-           autocorr[i] = av_clipl_int32((int64_t)temp + (1 << 15)) >> 16;
-        }
-    }
-}
-
-/**
- * Use Levinson-Durbin recursion to compute LPC coefficients from
- * autocorrelation values.
- *
- * @param lpc      LPC coefficients vector
- * @param autocorr autocorrelation coefficients vector
- * @param error    prediction error
- */
-static void levinson_durbin(int16_t *lpc, int16_t *autocorr, int16_t error)
-{
-    int16_t vector[LPC_ORDER];
-    int16_t partial_corr;
-    int i, j, temp;
-
-    memset(lpc, 0, LPC_ORDER * sizeof(int16_t));
-
-    for (i = 0; i < LPC_ORDER; i++) {
-        /* Compute the partial correlation coefficient */
-        temp = 0;
-        for (j = 0; j < i; j++)
-            temp -= lpc[j] * autocorr[i - j - 1];
-        temp = ((autocorr[i] << 13) + temp) << 3;
-
-        if (FFABS(temp) >= (error << 16))
-            break;
-
-        partial_corr = temp / (error << 1);
-
-        lpc[i] = av_clipl_int32((int64_t)(partial_corr << 14) +
-                                (1 << 15)) >> 16;
-
-        /* Update the prediction error */
-        temp  = MULL2(temp, partial_corr);
-        error = av_clipl_int32((int64_t)(error << 16) - temp +
-                                (1 << 15)) >> 16;
-
-        memcpy(vector, lpc, i * sizeof(int16_t));
-        for (j = 0; j < i; j++) {
-            temp = partial_corr * vector[i - j - 1] << 1;
-            lpc[j] = av_clipl_int32((int64_t)(lpc[j] << 16) - temp +
-                                    (1 << 15)) >> 16;
-        }
-    }
-}
-
-/**
- * Calculate LPC coefficients for the current frame.
- *
- * @param buf       current frame
- * @param prev_data 2 trailing subframes of the previous frame
- * @param lpc       LPC coefficients vector
- */
-static void comp_lpc_coeff(int16_t *buf, int16_t *lpc)
-{
-    int16_t autocorr[(LPC_ORDER + 1) * SUBFRAMES];
-    int16_t *autocorr_ptr = autocorr;
-    int16_t *lpc_ptr      = lpc;
-    int i, j;
-
-    for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
-        comp_autocorr(buf + i, autocorr_ptr);
-        levinson_durbin(lpc_ptr, autocorr_ptr + 1, autocorr_ptr[0]);
-
-        lpc_ptr += LPC_ORDER;
-        autocorr_ptr += LPC_ORDER + 1;
-    }
-}
-
-static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp)
-{
-    int f[LPC_ORDER + 2]; ///< coefficients of the sum and difference
-                          ///< polynomials (F1, F2) ordered as
-                          ///< f1[0], f2[0], ...., f1[5], f2[5]
-
-    int max, shift, cur_val, prev_val, count, p;
-    int i, j;
-    int64_t temp;
-
-    /* Initialize f1[0] and f2[0] to 1 in Q25 */
-    for (i = 0; i < LPC_ORDER; i++)
-        lsp[i] = (lpc[i] * bandwidth_expand[i] + (1 << 14)) >> 15;
-
-    /* Apply bandwidth expansion on the LPC coefficients */
-    f[0] = f[1] = 1 << 25;
-
-    /* Compute the remaining coefficients */
-    for (i = 0; i < LPC_ORDER / 2; i++) {
-        /* f1 */
-        f[2 * i + 2] = -f[2 * i] - ((lsp[i] + lsp[LPC_ORDER - 1 - i]) << 12);
-        /* f2 */
-        f[2 * i + 3] = f[2 * i + 1] - ((lsp[i] - lsp[LPC_ORDER - 1 - i]) << 12);
-    }
-
-    /* Divide f1[5] and f2[5] by 2 for use in polynomial evaluation */
-    f[LPC_ORDER] >>= 1;
-    f[LPC_ORDER + 1] >>= 1;
-
-    /* Normalize and shorten */
-    max = FFABS(f[0]);
-    for (i = 1; i < LPC_ORDER + 2; i++)
-        max = FFMAX(max, FFABS(f[i]));
-
-    shift = normalize_bits_int32(max);
-
-    for (i = 0; i < LPC_ORDER + 2; i++)
-        f[i] = av_clipl_int32((int64_t)(f[i] << shift) + (1 << 15)) >> 16;
-
-    /**
-     * Evaluate F1 and F2 at uniform intervals of pi/256 along the
-     * unit circle and check for zero crossings.
-     */
-    p    = 0;
-    temp = 0;
-    for (i = 0; i <= LPC_ORDER / 2; i++)
-        temp += f[2 * i] * cos_tab[0];
-    prev_val = av_clipl_int32(temp << 1);
-    count    = 0;
-    for ( i = 1; i < COS_TBL_SIZE / 2; i++) {
-        /* Evaluate */
-        temp = 0;
-        for (j = 0; j <= LPC_ORDER / 2; j++)
-            temp += f[LPC_ORDER - 2 * j + p] * cos_tab[i * j % COS_TBL_SIZE];
-        cur_val = av_clipl_int32(temp << 1);
-
-        /* Check for sign change, indicating a zero crossing */
-        if ((cur_val ^ prev_val) < 0) {
-            int abs_cur  = FFABS(cur_val);
-            int abs_prev = FFABS(prev_val);
-            int sum      = abs_cur + abs_prev;
-
-            shift        = normalize_bits_int32(sum);
-            sum          <<= shift;
-            abs_prev     = abs_prev << shift >> 8;
-            lsp[count++] = ((i - 1) << 7) + (abs_prev >> 1) / (sum >> 16);
-
-            if (count == LPC_ORDER)
-                break;
-
-            /* Switch between sum and difference polynomials */
-            p ^= 1;
-
-            /* Evaluate */
-            temp = 0;
-            for (j = 0; j <= LPC_ORDER / 2; j++){
-                temp += f[LPC_ORDER - 2 * j + p] *
-                        cos_tab[i * j % COS_TBL_SIZE];
-            }
-            cur_val = av_clipl_int32(temp<<1);
-        }
-        prev_val = cur_val;
-    }
-
-    if (count != LPC_ORDER)
-        memcpy(lsp, prev_lsp, LPC_ORDER * sizeof(int16_t));
-}
-
-/**
- * Quantize the current LSP subvector.
- *
- * @param num    band number
- * @param offset offset of the current subvector in an LPC_ORDER vector
- * @param size   size of the current subvector
- */
-#define get_index(num, offset, size) \
-{\
-    int error, max = -1;\
-    int16_t temp[4];\
-    int i, j;\
-    for (i = 0; i < LSP_CB_SIZE; i++) {\
-        for (j = 0; j < size; j++){\
-            temp[j] = (weight[j + (offset)] * lsp_band##num[i][j] +\
-                      (1 << 14)) >> 15;\
-        }\
-        error =  ff_g723_1_dot_product(lsp + (offset), temp, size) << 1;\
-        error -= ff_g723_1_dot_product(lsp_band##num[i], temp, size);\
-        if (error > max) {\
-            max = error;\
-            lsp_index[num] = i;\
-        }\
-    }\
-}
-
-/**
- * Vector quantize the LSP frequencies.
- *
- * @param lsp      the current lsp vector
- * @param prev_lsp the previous lsp vector
- */
-static void lsp_quantize(uint8_t *lsp_index, int16_t *lsp, int16_t *prev_lsp)
-{
-    int16_t weight[LPC_ORDER];
-    int16_t min, max;
-    int shift, i;
-
-    /* Calculate the VQ weighting vector */
-    weight[0] = (1 << 20) / (lsp[1] - lsp[0]);
-    weight[LPC_ORDER - 1] = (1 << 20) /
-                            (lsp[LPC_ORDER - 1] - lsp[LPC_ORDER - 2]);
-
-    for (i = 1; i < LPC_ORDER - 1; i++) {
-        min  = FFMIN(lsp[i] - lsp[i - 1], lsp[i + 1] - lsp[i]);
-        if (min > 0x20)
-            weight[i] = (1 << 20) / min;
-        else
-            weight[i] = INT16_MAX;
-    }
-
-    /* Normalize */
-    max = 0;
-    for (i = 0; i < LPC_ORDER; i++)
-        max = FFMAX(weight[i], max);
-
-    shift = normalize_bits_int16(max);
-    for (i = 0; i < LPC_ORDER; i++) {
-        weight[i] <<= shift;
-    }
-
-    /* Compute the VQ target vector */
-    for (i = 0; i < LPC_ORDER; i++) {
-        lsp[i] -= dc_lsp[i] +
-                  (((prev_lsp[i] - dc_lsp[i]) * 12288 + (1 << 14)) >> 15);
-    }
-
-    get_index(0, 0, 3);
-    get_index(1, 3, 3);
-    get_index(2, 6, 4);
-}
-
-/**
- * Apply the formant perceptual weighting filter.
- *
- * @param flt_coef filter coefficients
- * @param unq_lpc  unquantized lpc vector
- */
-static void perceptual_filter(G723_1_Context *p, int16_t *flt_coef,
-                              int16_t *unq_lpc, int16_t *buf)
-{
-    int16_t vector[FRAME_LEN + LPC_ORDER];
-    int i, j, k, l = 0;
-
-    memcpy(buf, p->iir_mem, sizeof(int16_t) * LPC_ORDER);
-    memcpy(vector, p->fir_mem, sizeof(int16_t) * LPC_ORDER);
-    memcpy(vector + LPC_ORDER, buf + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
-
-    for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
-        for (k = 0; k < LPC_ORDER; k++) {
-            flt_coef[k + 2 * l] = (unq_lpc[k + l] * percept_flt_tbl[0][k] +
-                                  (1 << 14)) >> 15;
-            flt_coef[k + 2 * l + LPC_ORDER] = (unq_lpc[k + l] *
-                                             percept_flt_tbl[1][k] +
-                                             (1 << 14)) >> 15;
-        }
-        iir_filter(flt_coef + 2 * l, flt_coef + 2 * l + LPC_ORDER, vector + i,
-                   buf + i, 0);
-        l += LPC_ORDER;
-    }
-    memcpy(p->iir_mem, buf + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
-    memcpy(p->fir_mem, vector + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
-}
-
-/**
- * Estimate the open loop pitch period.
- *
- * @param buf   perceptually weighted speech
- * @param start estimation is carried out from this position
- */
-static int estimate_pitch(int16_t *buf, int start)
-{
-    int max_exp = 32;
-    int max_ccr = 0x4000;
-    int max_eng = 0x7fff;
-    int index   = PITCH_MIN;
-    int offset  = start - PITCH_MIN + 1;
-
-    int ccr, eng, orig_eng, ccr_eng, exp;
-    int diff, temp;
-
-    int i;
-
-    orig_eng = ff_dot_product(buf + offset, buf + offset, HALF_FRAME_LEN);
-
-    for (i = PITCH_MIN; i <= PITCH_MAX - 3; i++) {
-        offset--;
-
-        /* Update energy and compute correlation */
-        orig_eng += buf[offset] * buf[offset] -
-                    buf[offset + HALF_FRAME_LEN] * buf[offset + HALF_FRAME_LEN];
-        ccr      =  ff_dot_product(buf + start, buf + offset, HALF_FRAME_LEN);
-        if (ccr <= 0)
-            continue;
-
-        /* Split into mantissa and exponent to maintain precision */
-        exp  =   normalize_bits_int32(ccr);
-        ccr  =   av_clipl_int32((int64_t)(ccr << exp) + (1 << 15)) >> 16;
-        exp  <<= 1;
-        ccr  *=  ccr;
-        temp =   normalize_bits_int32(ccr);
-        ccr  =   ccr << temp >> 16;
-        exp  +=  temp;
-
-        temp =   normalize_bits_int32(orig_eng);
-        eng  =   av_clipl_int32((int64_t)(orig_eng << temp) + (1 << 15)) >> 16;
-        exp  -=  temp;
-
-        if (ccr >= eng) {
-            exp--;
-            ccr >>= 1;
-        }
-        if (exp > max_exp)
-            continue;
-
-        if (exp + 1 < max_exp)
-            goto update;
-
-        /* Equalize exponents before comparison */
-        if (exp + 1 == max_exp)
-            temp = max_ccr >> 1;
-        else
-            temp = max_ccr;
-        ccr_eng = ccr * max_eng;
-        diff    = ccr_eng - eng * temp;
-        if (diff > 0 && (i - index < PITCH_MIN || diff > ccr_eng >> 2)) {
-update:
-            index   = i;
-            max_exp = exp;
-            max_ccr = ccr;
-            max_eng = eng;
-        }
-    }
-    return index;
-}
-
-/**
- * Compute harmonic noise filter parameters.
- *
- * @param buf       perceptually weighted speech
- * @param pitch_lag open loop pitch period
- * @param hf        harmonic filter parameters
- */
-static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf)
-{
-    int ccr, eng, max_ccr, max_eng;
-    int exp, max, diff;
-    int energy[15];
-    int i, j;
-
-    for (i = 0, j = pitch_lag - 3; j <= pitch_lag + 3; i++, j++) {
-        /* Compute residual energy */
-        energy[i << 1] = ff_dot_product(buf - j, buf - j, SUBFRAME_LEN);
-        /* Compute correlation */
-        energy[(i << 1) + 1] = ff_dot_product(buf, buf - j, SUBFRAME_LEN);
-    }
-
-    /* Compute target energy */
-    energy[14] = ff_dot_product(buf, buf, SUBFRAME_LEN);
-
-    /* Normalize */
-    max = 0;
-    for (i = 0; i < 15; i++)
-        max = FFMAX(max, FFABS(energy[i]));
-
-    exp = normalize_bits_int32(max);
-    for (i = 0; i < 15; i++) {
-        energy[i] = av_clipl_int32((int64_t)(energy[i] << exp) +
-                                   (1 << 15)) >> 16;
-    }
-
-    hf->index = -1;
-    hf->gain  =  0;
-    max_ccr   =  1;
-    max_eng   =  0x7fff;
-
-    for (i = 0; i <= 6; i++) {
-        eng = energy[i << 1];
-        ccr = energy[(i << 1) + 1];
-
-        if (ccr <= 0)
-            continue;
-
-        ccr  = (ccr * ccr + (1 << 14)) >> 15;
-        diff = ccr * max_eng - eng * max_ccr;
-        if (diff > 0) {
-            max_ccr   = ccr;
-            max_eng   = eng;
-            hf->index = i;
-        }
-    }
-
-    if (hf->index == -1) {
-        hf->index = pitch_lag;
-        return;
-    }
-
-    eng = energy[14] * max_eng;
-    eng = (eng >> 2) + (eng >> 3);
-    ccr = energy[(hf->index << 1) + 1] * energy[(hf->index << 1) + 1];
-    if (eng < ccr) {
-        eng = energy[(hf->index << 1) + 1];
-
-        if (eng >= max_eng)
-            hf->gain = 0x2800;
-        else
-            hf->gain = ((eng << 15) / max_eng * 0x2800 + (1 << 14)) >> 15;
-    }
-    hf->index += pitch_lag - 3;
-}
-
-/**
- * Apply the harmonic noise shaping filter.
- *
- * @param hf filter parameters
- */
-static void harmonic_filter(HFParam *hf, const int16_t *src, int16_t *dest)
-{
-    int i;
-
-    for (i = 0; i < SUBFRAME_LEN; i++) {
-        int64_t temp = hf->gain * src[i - hf->index] << 1;
-        dest[i] = av_clipl_int32((src[i] << 16) - temp + (1 << 15)) >> 16;
-    }
-}
-
-static void harmonic_noise_sub(HFParam *hf, const int16_t *src, int16_t *dest)
-{
-    int i;
-    for (i = 0; i < SUBFRAME_LEN; i++) {
-        int64_t temp = hf->gain * src[i - hf->index] << 1;
-        dest[i] = av_clipl_int32(((dest[i] - src[i]) << 16) + temp +
-                                 (1 << 15)) >> 16;
-
-    }
-}
-
-/**
- * Combined synthesis and formant perceptual weighting filer.
- *
- * @param qnt_lpc  quantized lpc coefficients
- * @param perf_lpc perceptual filter coefficients
- * @param perf_fir perceptual filter fir memory
- * @param perf_iir perceptual filter iir memory
- * @param scale    the filter output will be scaled by 2^scale
- */
-static void synth_percept_filter(int16_t *qnt_lpc, int16_t *perf_lpc,
-                                 int16_t *perf_fir, int16_t *perf_iir,
-                                 const int16_t *src, int16_t *dest, int scale)
-{
-    int i, j;
-    int16_t buf_16[SUBFRAME_LEN + LPC_ORDER];
-    int64_t buf[SUBFRAME_LEN];
-
-    int16_t *bptr_16 = buf_16 + LPC_ORDER;
-
-    memcpy(buf_16, perf_fir, sizeof(int16_t) * LPC_ORDER);
-    memcpy(dest - LPC_ORDER, perf_iir, sizeof(int16_t) * LPC_ORDER);
-
-    for (i = 0; i < SUBFRAME_LEN; i++) {
-        int64_t temp = 0;
-        for (j = 1; j <= LPC_ORDER; j++)
-            temp -= qnt_lpc[j - 1] * bptr_16[i - j];
-
-        buf[i]     = (src[i] << 15) + (temp << 3);
-        bptr_16[i] = av_clipl_int32(buf[i] + (1 << 15)) >> 16;
-    }
-
-    for (i = 0; i < SUBFRAME_LEN; i++) {
-        int64_t fir = 0, iir = 0;
-        for (j = 1; j <= LPC_ORDER; j++) {
-            fir -= perf_lpc[j - 1] * bptr_16[i - j];
-            iir += perf_lpc[j + LPC_ORDER - 1] * dest[i - j];
-        }
-        dest[i] = av_clipl_int32(((buf[i] + (fir << 3)) << scale) + (iir << 3) +
-                                 (1 << 15)) >> 16;
-    }
-    memcpy(perf_fir, buf_16 + SUBFRAME_LEN, sizeof(int16_t) * LPC_ORDER);
-    memcpy(perf_iir, dest + SUBFRAME_LEN - LPC_ORDER,
-           sizeof(int16_t) * LPC_ORDER);
-}
-
-/**
- * Compute the adaptive codebook contribution.
- *
- * @param buf   input signal
- * @param index the current subframe index
- */
-static void acb_search(G723_1_Context *p, int16_t *residual,
-                       int16_t *impulse_resp, const int16_t *buf,
-                       int index)
-{
-
-    int16_t flt_buf[PITCH_ORDER][SUBFRAME_LEN];
-
-    const int16_t *cb_tbl = adaptive_cb_gain85;
-
-    int ccr_buf[PITCH_ORDER * SUBFRAMES << 2];
-
-    int pitch_lag = p->pitch_lag[index >> 1];
-    int acb_lag   = 1;
-    int acb_gain  = 0;
-    int odd_frame = index & 1;
-    int iter      = 3 + odd_frame;
-    int count     = 0;
-    int tbl_size  = 85;
-
-    int i, j, k, l, max;
-    int64_t temp;
-
-    if (!odd_frame) {
-        if (pitch_lag == PITCH_MIN)
-            pitch_lag++;
-        else
-            pitch_lag = FFMIN(pitch_lag, PITCH_MAX - 5);
-    }
-
-    for (i = 0; i < iter; i++) {
-        ff_g723_1_get_residual(residual, p->prev_excitation, pitch_lag + i - 1);
-
-        for (j = 0; j < SUBFRAME_LEN; j++) {
-            temp = 0;
-            for (k = 0; k <= j; k++)
-                temp += residual[PITCH_ORDER - 1 + k] * impulse_resp[j - k];
-            flt_buf[PITCH_ORDER - 1][j] = av_clipl_int32((temp << 1) +
-                                                         (1 << 15)) >> 16;
-        }
-
-        for (j = PITCH_ORDER - 2; j >= 0; j--) {
-            flt_buf[j][0] = ((residual[j] << 13) + (1 << 14)) >> 15;
-            for (k = 1; k < SUBFRAME_LEN; k++) {
-                temp = (flt_buf[j + 1][k - 1] << 15) +
-                       residual[j] * impulse_resp[k];
-                flt_buf[j][k] = av_clipl_int32((temp << 1) + (1 << 15)) >> 16;
-            }
-        }
-
-        /* Compute crosscorrelation with the signal */
-        for (j = 0; j < PITCH_ORDER; j++) {
-            temp = ff_dot_product(buf, flt_buf[j], SUBFRAME_LEN);
-            ccr_buf[count++] = av_clipl_int32(temp << 1);
-        }
-
-        /* Compute energies */
-        for (j = 0; j < PITCH_ORDER; j++) {
-            ccr_buf[count++] = ff_g723_1_dot_product(flt_buf[j], flt_buf[j],
-                                           SUBFRAME_LEN);
-        }
-
-        for (j = 1; j < PITCH_ORDER; j++) {
-            for (k = 0; k < j; k++) {
-                temp = ff_dot_product(flt_buf[j], flt_buf[k], SUBFRAME_LEN);
-                ccr_buf[count++] = av_clipl_int32(temp<<2);
-            }
-        }
-    }
-
-    /* Normalize and shorten */
-    max = 0;
-    for (i = 0; i < 20 * iter; i++)
-        max = FFMAX(max, FFABS(ccr_buf[i]));
-
-    temp = normalize_bits_int32(max);
-
-    for (i = 0; i < 20 * iter; i++){
-        ccr_buf[i] = av_clipl_int32((int64_t)(ccr_buf[i] << temp) +
-                                    (1 << 15)) >> 16;
-    }
-
-    max = 0;
-    for (i = 0; i < iter; i++) {
-        /* Select quantization table */
-        if (!odd_frame && pitch_lag + i - 1 >= SUBFRAME_LEN - 2 ||
-            odd_frame && pitch_lag >= SUBFRAME_LEN - 2) {
-            cb_tbl = adaptive_cb_gain170;
-            tbl_size = 170;
-        }
-
-        for (j = 0, k = 0; j < tbl_size; j++, k += 20) {
-            temp = 0;
-            for (l = 0; l < 20; l++)
-                temp += ccr_buf[20 * i + l] * cb_tbl[k + l];
-            temp =  av_clipl_int32(temp);
-
-            if (temp > max) {
-                max      = temp;
-                acb_gain = j;
-                acb_lag  = i;
-            }
-        }
-    }
-
-    if (!odd_frame) {
-        pitch_lag += acb_lag - 1;
-        acb_lag   =  1;
-    }
-
-    p->pitch_lag[index >> 1]      = pitch_lag;
-    p->subframe[index].ad_cb_lag  = acb_lag;
-    p->subframe[index].ad_cb_gain = acb_gain;
-}
-
-/**
- * Subtract the adaptive codebook contribution from the input
- * to obtain the residual.
- *
- * @param buf target vector
- */
-static void sub_acb_contrib(const int16_t *residual, const int16_t *impulse_resp,
-                            int16_t *buf)
-{
-    int i, j;
-    /* Subtract adaptive CB contribution to obtain the residual */
-    for (i = 0; i < SUBFRAME_LEN; i++) {
-        int64_t temp = buf[i] << 14;
-        for (j = 0; j <= i; j++)
-            temp -= residual[j] * impulse_resp[i - j];
-
-        buf[i] = av_clipl_int32((temp << 2) + (1 << 15)) >> 16;
-    }
-}
-
-/**
- * Quantize the residual signal using the fixed codebook (MP-MLQ).
- *
- * @param optim optimized fixed codebook parameters
- * @param buf   excitation vector
- */
-static void get_fcb_param(FCBParam *optim, int16_t *impulse_resp,
-                          int16_t *buf, int pulse_cnt, int pitch_lag)
-{
-    FCBParam param;
-    int16_t impulse_r[SUBFRAME_LEN];
-    int16_t temp_corr[SUBFRAME_LEN];
-    int16_t impulse_corr[SUBFRAME_LEN];
-
-    int ccr1[SUBFRAME_LEN];
-    int ccr2[SUBFRAME_LEN];
-    int amp, err, max, max_amp_index, min, scale, i, j, k, l;
-
-    int64_t temp;
-
-    /* Update impulse response */
-    memcpy(impulse_r, impulse_resp, sizeof(int16_t) * SUBFRAME_LEN);
-    param.dirac_train = 0;
-    if (pitch_lag < SUBFRAME_LEN - 2) {
-        param.dirac_train = 1;
-        ff_g723_1_gen_dirac_train(impulse_r, pitch_lag);
-    }
-
-    for (i = 0; i < SUBFRAME_LEN; i++)
-        temp_corr[i] = impulse_r[i] >> 1;
-
-    /* Compute impulse response autocorrelation */
-    temp = ff_g723_1_dot_product(temp_corr, temp_corr, SUBFRAME_LEN);
-
-    scale = normalize_bits_int32(temp);
-    impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
-
-    for (i = 1; i < SUBFRAME_LEN; i++) {
-        temp = ff_g723_1_dot_product(temp_corr + i, temp_corr, SUBFRAME_LEN - i);
-        impulse_corr[i] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
-    }
-
-    /* Compute crosscorrelation of impulse response with residual signal */
-    scale -= 4;
-    for (i = 0; i < SUBFRAME_LEN; i++){
-        temp = ff_g723_1_dot_product(buf + i, impulse_r, SUBFRAME_LEN - i);
-        if (scale < 0)
-            ccr1[i] = temp >> -scale;
-        else
-            ccr1[i] = av_clipl_int32(temp << scale);
-    }
-
-    /* Search loop */
-    for (i = 0; i < GRID_SIZE; i++) {
-        /* Maximize the crosscorrelation */
-        max = 0;
-        for (j = i; j < SUBFRAME_LEN; j += GRID_SIZE) {
-            temp = FFABS(ccr1[j]);
-            if (temp >= max) {
-                max = temp;
-                param.pulse_pos[0] = j;
-            }
-        }
-
-        /* Quantize the gain (max crosscorrelation/impulse_corr[0]) */
-        amp = max;
-        min = 1 << 30;
-        max_amp_index = GAIN_LEVELS - 2;
-        for (j = max_amp_index; j >= 2; j--) {
-            temp = av_clipl_int32((int64_t)fixed_cb_gain[j] *
-                                  impulse_corr[0] << 1);
-            temp = FFABS(temp - amp);
-            if (temp < min) {
-                min = temp;
-                max_amp_index = j;
-            }
-        }
-
-        max_amp_index--;
-        /* Select additional gain values */
-        for (j = 1; j < 5; j++) {
-            for (k = i; k < SUBFRAME_LEN; k += GRID_SIZE) {
-                temp_corr[k] = 0;
-                ccr2[k]      = ccr1[k];
-            }
-            param.amp_index = max_amp_index + j - 2;
-            amp = fixed_cb_gain[param.amp_index];
-
-            param.pulse_sign[0] = (ccr2[param.pulse_pos[0]] < 0) ? -amp : amp;
-            temp_corr[param.pulse_pos[0]] = 1;
-
-            for (k = 1; k < pulse_cnt; k++) {
-                max = INT_MIN;
-                for (l = i; l < SUBFRAME_LEN; l += GRID_SIZE) {
-                    if (temp_corr[l])
-                        continue;
-                    temp = impulse_corr[FFABS(l - param.pulse_pos[k - 1])];
-                    temp = av_clipl_int32((int64_t)temp *
-                                          param.pulse_sign[k - 1] << 1);
-                    ccr2[l] -= temp;
-                    temp = FFABS(ccr2[l]);
-                    if (temp > max) {
-                        max = temp;
-                        param.pulse_pos[k] = l;
-                    }
-                }
-
-                param.pulse_sign[k] = (ccr2[param.pulse_pos[k]] < 0) ?
-                                      -amp : amp;
-                temp_corr[param.pulse_pos[k]] = 1;
-            }
-
-            /* Create the error vector */
-            memset(temp_corr, 0, sizeof(int16_t) * SUBFRAME_LEN);
-
-            for (k = 0; k < pulse_cnt; k++)
-                temp_corr[param.pulse_pos[k]] = param.pulse_sign[k];
-
-            for (k = SUBFRAME_LEN - 1; k >= 0; k--) {
-                temp = 0;
-                for (l = 0; l <= k; l++) {
-                    int prod = av_clipl_int32((int64_t)temp_corr[l] *
-                                              impulse_r[k - l] << 1);
-                    temp     = av_clipl_int32(temp + prod);
-                }
-                temp_corr[k] = temp << 2 >> 16;
-            }
-
-            /* Compute square of error */
-            err = 0;
-            for (k = 0; k < SUBFRAME_LEN; k++) {
-                int64_t prod;
-                prod = av_clipl_int32((int64_t)buf[k] * temp_corr[k] << 1);
-                err  = av_clipl_int32(err - prod);
-                prod = av_clipl_int32((int64_t)temp_corr[k] * temp_corr[k]);
-                err  = av_clipl_int32(err + prod);
-            }
-
-            /* Minimize */
-            if (err < optim->min_err) {
-                optim->min_err     = err;
-                optim->grid_index  = i;
-                optim->amp_index   = param.amp_index;
-                optim->dirac_train = param.dirac_train;
-
-                for (k = 0; k < pulse_cnt; k++) {
-                    optim->pulse_sign[k] = param.pulse_sign[k];
-                    optim->pulse_pos[k]  = param.pulse_pos[k];
-                }
-            }
-        }
-    }
-}
-
-/**
- * Encode the pulse position and gain of the current subframe.
- *
- * @param optim optimized fixed CB parameters
- * @param buf   excitation vector
- */
-static void pack_fcb_param(G723_1_Subframe *subfrm, FCBParam *optim,
-                           int16_t *buf, int pulse_cnt)
-{
-    int i, j;
-
-    j = PULSE_MAX - pulse_cnt;
-
-    subfrm->pulse_sign = 0;
-    subfrm->pulse_pos  = 0;
-
-    for (i = 0; i < SUBFRAME_LEN >> 1; i++) {
-        int val = buf[optim->grid_index + (i << 1)];
-        if (!val) {
-            subfrm->pulse_pos += combinatorial_table[j][i];
-        } else {
-            subfrm->pulse_sign <<= 1;
-            if (val < 0) subfrm->pulse_sign++;
-            j++;
-
-            if (j == PULSE_MAX) break;
-        }
-    }
-    subfrm->amp_index   = optim->amp_index;
-    subfrm->grid_index  = optim->grid_index;
-    subfrm->dirac_train = optim->dirac_train;
-}
-
-/**
- * Compute the fixed codebook excitation.
- *
- * @param buf          target vector
- * @param impulse_resp impulse response of the combined filter
- */
-static void fcb_search(G723_1_Context *p, int16_t *impulse_resp,
-                       int16_t *buf, int index)
-{
-    FCBParam optim;
-    int pulse_cnt = pulses[index];
-    int i;
-
-    optim.min_err = 1 << 30;
-    get_fcb_param(&optim, impulse_resp, buf, pulse_cnt, SUBFRAME_LEN);
-
-    if (p->pitch_lag[index >> 1] < SUBFRAME_LEN - 2) {
-        get_fcb_param(&optim, impulse_resp, buf, pulse_cnt,
-                      p->pitch_lag[index >> 1]);
-    }
-
-    /* Reconstruct the excitation */
-    memset(buf, 0, sizeof(int16_t) * SUBFRAME_LEN);
-    for (i = 0; i < pulse_cnt; i++)
-        buf[optim.pulse_pos[i]] = optim.pulse_sign[i];
-
-    pack_fcb_param(&p->subframe[index], &optim, buf, pulse_cnt);
-
-    if (optim.dirac_train)
-        ff_g723_1_gen_dirac_train(buf, p->pitch_lag[index >> 1]);
-}
-
-/**
- * Pack the frame parameters into output bitstream.
- *
- * @param frame output buffer
- * @param size  size of the buffer
- */
-static int pack_bitstream(G723_1_Context *p, unsigned char *frame, int size)
-{
-    PutBitContext pb;
-    int info_bits, i, temp;
-
-    init_put_bits(&pb, frame, size);
-
-    if (p->cur_rate == RATE_6300) {
-        info_bits = 0;
-        put_bits(&pb, 2, info_bits);
-    }else
-        av_assert0(0);
-
-    put_bits(&pb, 8, p->lsp_index[2]);
-    put_bits(&pb, 8, p->lsp_index[1]);
-    put_bits(&pb, 8, p->lsp_index[0]);
-
-    put_bits(&pb, 7, p->pitch_lag[0] - PITCH_MIN);
-    put_bits(&pb, 2, p->subframe[1].ad_cb_lag);
-    put_bits(&pb, 7, p->pitch_lag[1] - PITCH_MIN);
-    put_bits(&pb, 2, p->subframe[3].ad_cb_lag);
-
-    /* Write 12 bit combined gain */
-    for (i = 0; i < SUBFRAMES; i++) {
-        temp = p->subframe[i].ad_cb_gain * GAIN_LEVELS +
-               p->subframe[i].amp_index;
-        if (p->cur_rate ==  RATE_6300)
-            temp += p->subframe[i].dirac_train << 11;
-        put_bits(&pb, 12, temp);
-    }
-
-    put_bits(&pb, 1, p->subframe[0].grid_index);
-    put_bits(&pb, 1, p->subframe[1].grid_index);
-    put_bits(&pb, 1, p->subframe[2].grid_index);
-    put_bits(&pb, 1, p->subframe[3].grid_index);
-
-    if (p->cur_rate == RATE_6300) {
-        skip_put_bits(&pb, 1); /* reserved bit */
-
-        /* Write 13 bit combined position index */
-        temp = (p->subframe[0].pulse_pos >> 16) * 810 +
-               (p->subframe[1].pulse_pos >> 14) *  90 +
-               (p->subframe[2].pulse_pos >> 16) *   9 +
-               (p->subframe[3].pulse_pos >> 14);
-        put_bits(&pb, 13, temp);
-
-        put_bits(&pb, 16, p->subframe[0].pulse_pos & 0xffff);
-        put_bits(&pb, 14, p->subframe[1].pulse_pos & 0x3fff);
-        put_bits(&pb, 16, p->subframe[2].pulse_pos & 0xffff);
-        put_bits(&pb, 14, p->subframe[3].pulse_pos & 0x3fff);
-
-        put_bits(&pb, 6, p->subframe[0].pulse_sign);
-        put_bits(&pb, 5, p->subframe[1].pulse_sign);
-        put_bits(&pb, 6, p->subframe[2].pulse_sign);
-        put_bits(&pb, 5, p->subframe[3].pulse_sign);
-    }
-
-    flush_put_bits(&pb);
-    return frame_size[info_bits];
-}
-
-static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
-                            const AVFrame *frame, int *got_packet_ptr)
-{
-    G723_1_Context *p = avctx->priv_data;
-    int16_t unq_lpc[LPC_ORDER * SUBFRAMES];
-    int16_t qnt_lpc[LPC_ORDER * SUBFRAMES];
-    int16_t cur_lsp[LPC_ORDER];
-    int16_t weighted_lpc[LPC_ORDER * SUBFRAMES << 1];
-    int16_t vector[FRAME_LEN + PITCH_MAX];
-    int offset, ret;
-    int16_t *in_orig = av_memdup(frame->data[0], frame->nb_samples * sizeof(int16_t));
-    int16_t *in = in_orig;
-
-    HFParam hf[4];
-    int i, j;
-
-    if (!in)
-        return AVERROR(ENOMEM);
-
-    highpass_filter(in, &p->hpf_fir_mem, &p->hpf_iir_mem);
-
-    memcpy(vector, p->prev_data, HALF_FRAME_LEN * sizeof(int16_t));
-    memcpy(vector + HALF_FRAME_LEN, in, FRAME_LEN * sizeof(int16_t));
-
-    comp_lpc_coeff(vector, unq_lpc);
-    lpc2lsp(&unq_lpc[LPC_ORDER * 3], p->prev_lsp, cur_lsp);
-    lsp_quantize(p->lsp_index, cur_lsp, p->prev_lsp);
-
-    /* Update memory */
-    memcpy(vector + LPC_ORDER, p->prev_data + SUBFRAME_LEN,
-           sizeof(int16_t) * SUBFRAME_LEN);
-    memcpy(vector + LPC_ORDER + SUBFRAME_LEN, in,
-           sizeof(int16_t) * (HALF_FRAME_LEN + SUBFRAME_LEN));
-    memcpy(p->prev_data, in + HALF_FRAME_LEN,
-           sizeof(int16_t) * HALF_FRAME_LEN);
-    memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
-
-    perceptual_filter(p, weighted_lpc, unq_lpc, vector);
-
-    memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
-    memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
-    memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
-
-    ff_g723_1_scale_vector(vector, vector, FRAME_LEN + PITCH_MAX);
-
-    p->pitch_lag[0] = estimate_pitch(vector, PITCH_MAX);
-    p->pitch_lag[1] = estimate_pitch(vector, PITCH_MAX + HALF_FRAME_LEN);
-
-    for (i = PITCH_MAX, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
-        comp_harmonic_coeff(vector + i, p->pitch_lag[j >> 1], hf + j);
-
-    memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
-    memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
-    memcpy(p->prev_weight_sig, vector + FRAME_LEN, sizeof(int16_t) * PITCH_MAX);
-
-    for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
-        harmonic_filter(hf + j, vector + PITCH_MAX + i, in + i);
-
-    ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);
-    ff_g723_1_lsp_interpolate(qnt_lpc, cur_lsp, p->prev_lsp);
-
-    memcpy(p->prev_lsp, cur_lsp, sizeof(int16_t) * LPC_ORDER);
-
-    offset = 0;
-    for (i = 0; i < SUBFRAMES; i++) {
-        int16_t impulse_resp[SUBFRAME_LEN];
-        int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
-        int16_t flt_in[SUBFRAME_LEN];
-        int16_t zero[LPC_ORDER], fir[LPC_ORDER], iir[LPC_ORDER];
-
-        /**
-         * Compute the combined impulse response of the synthesis filter,
-         * formant perceptual weighting filter and harmonic noise shaping filter
-         */
-        memset(zero, 0, sizeof(int16_t) * LPC_ORDER);
-        memset(vector, 0, sizeof(int16_t) * PITCH_MAX);
-        memset(flt_in, 0, sizeof(int16_t) * SUBFRAME_LEN);
-
-        flt_in[0] = 1 << 13; /* Unit impulse */
-        synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
-                             zero, zero, flt_in, vector + PITCH_MAX, 1);
-        harmonic_filter(hf + i, vector + PITCH_MAX, impulse_resp);
-
-         /* Compute the combined zero input response */
-        flt_in[0] = 0;
-        memcpy(fir, p->perf_fir_mem, sizeof(int16_t) * LPC_ORDER);
-        memcpy(iir, p->perf_iir_mem, sizeof(int16_t) * LPC_ORDER);
-
-        synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
-                             fir, iir, flt_in, vector + PITCH_MAX, 0);
-        memcpy(vector, p->harmonic_mem, sizeof(int16_t) * PITCH_MAX);
-        harmonic_noise_sub(hf + i, vector + PITCH_MAX, in);
-
-        acb_search(p, residual, impulse_resp, in, i);
-        ff_g723_1_gen_acb_excitation(residual, p->prev_excitation,p->pitch_lag[i >> 1],
-                           &p->subframe[i], p->cur_rate);
-        sub_acb_contrib(residual, impulse_resp, in);
-
-        fcb_search(p, impulse_resp, in, i);
-
-        /* Reconstruct the excitation */
-        ff_g723_1_gen_acb_excitation(impulse_resp, p->prev_excitation, p->pitch_lag[i >> 1],
-                           &p->subframe[i], RATE_6300);
-
-        memmove(p->prev_excitation, p->prev_excitation + SUBFRAME_LEN,
-               sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
-        for (j = 0; j < SUBFRAME_LEN; j++)
-            in[j] = av_clip_int16((in[j] << 1) + impulse_resp[j]);
-        memcpy(p->prev_excitation + PITCH_MAX - SUBFRAME_LEN, in,
-               sizeof(int16_t) * SUBFRAME_LEN);
-
-        /* Update filter memories */
-        synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
-                             p->perf_fir_mem, p->perf_iir_mem,
-                             in, vector + PITCH_MAX, 0);
-        memmove(p->harmonic_mem, p->harmonic_mem + SUBFRAME_LEN,
-                sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
-        memcpy(p->harmonic_mem + PITCH_MAX - SUBFRAME_LEN, vector + PITCH_MAX,
-               sizeof(int16_t) * SUBFRAME_LEN);
-
-        in += SUBFRAME_LEN;
-        offset += LPC_ORDER;
-    }
-
-    av_freep(&in_orig); in = NULL;
-
-    if ((ret = ff_alloc_packet2(avctx, avpkt, 24, 0)) < 0)
-        return ret;
-
-    *got_packet_ptr = 1;
-    avpkt->size = pack_bitstream(p, avpkt->data, avpkt->size);
-    return 0;
-}
-
-AVCodec ff_g723_1_encoder = {
-    .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_encode_init,
-    .encode2        = g723_1_encode_frame,
-    .sample_fmts    = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,
-                                                    AV_SAMPLE_FMT_NONE},
-};
-#endif