libavcodec/g723_1enc.c
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 /*
  * G.723.1 compatible encoder
  * Copyright (c) Mohamed Naufal <naufal22@gmail.com>
  *
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  * This file is part of FFmpeg.
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  *
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  * FFmpeg is free software; you can redistribute it and/or
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  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
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  * FFmpeg is distributed in the hope that it will be useful,
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  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * 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
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  * License along with FFmpeg; if not, write to the Free Software
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  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 /**
  * @file
  * G.723.1 compatible encoder
  */
 
 #include <stdint.h>
 #include <string.h>
 
 #include "libavutil/channel_layout.h"
 #include "libavutil/common.h"
 #include "libavutil/mem.h"
 #include "libavutil/opt.h"
 
 #include "avcodec.h"
 #include "celp_math.h"
 #include "g723_1.h"
 #include "internal.h"
 
 #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 AVERROR(EINVAL);
     }
 
     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 6300\n");
         return AVERROR_PATCHWELCOME;
     } else {
         av_log(avctx, AV_LOG_ERROR, "Bitrate not supported, use 6300\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];
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         buf[i] = av_clipl_int32((int64_t)*iir + (1 << 15)) >> 16;
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     }
 }
 
 /**
  * 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       = ff_g723_1_normalize_bits(temp, 31);
     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 = ff_g723_1_normalize_bits(max, 31);
 
     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        = ff_g723_1_normalize_bits(sum, 31);
             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 = ff_g723_1_normalize_bits(max, 15);
     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);
 }
 
 /**
  * Perform IIR filtering.
  *
  * @param fir_coef FIR coefficients
  * @param iir_coef IIR coefficients
  * @param src      source vector
  * @param dest     destination vector
  */
 static void iir_filter(int16_t *fir_coef, int16_t *iir_coef,
                        int16_t *src, int16_t *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];
         }
 
         dest[m] = av_clipl_int32((src[m] << 16) + (filter << 3) +
                                  (1 << 15)) >> 16;
     }
 }
 
 /**
  * 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);
         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   = ff_g723_1_normalize_bits(ccr, 31);
         ccr   = av_clipl_int32((int64_t) (ccr << exp) + (1 << 15)) >> 16;
         exp <<= 1;
         ccr  *= ccr;
         temp  = ff_g723_1_normalize_bits(ccr, 31);
         ccr   = ccr << temp >> 16;
         exp  += temp;
 
         temp = ff_g723_1_normalize_bits(orig_eng, 31);
         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 = ff_g723_1_normalize_bits(max, 31);
     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 = ff_g723_1_normalize_bits(max, 31);
 
     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           = ff_g723_1_normalize_bits(temp, 31);
     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, AVPacket *avpkt)
 {
     PutBitContext pb;
     int info_bits = 0;
     int i, temp;
 
     init_put_bits(&pb, avpkt->data, avpkt->size);
 
     put_bits(&pb, 2, info_bits);
 
     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, i, j;
     int16_t *in, *start;
     HFParam hf[4];
 
     /* duplicate input */
     start = in = av_malloc(frame->nb_samples * sizeof(int16_t));
     if (!in)
         return AVERROR(ENOMEM);
     memcpy(in, frame->data[0], frame->nb_samples * sizeof(int16_t));
 
     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],
90c93fb1
                                      p->cur_rate);
f023d57d
         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_free(start);
 
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     if ((ret = ff_alloc_packet2(avctx, avpkt, 24, 0)) < 0)
f023d57d
         return ret;
 
     *got_packet_ptr = 1;
90c93fb1
     avpkt->size = pack_bitstream(p, avpkt);
     return 0;
f023d57d
 }
 
 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
     },
 };