/* * G.723.1 compatible decoder * Copyright (c) 2006 Benjamin Larsson * Copyright (c) 2010 Mohamed Naufal Basheer * * This file is part of FFmpeg. * * FFmpeg 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. * * FFmpeg 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 FFmpeg; 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/audioconvert.h" #include "libavutil/lzo.h" #include "libavutil/opt.h" #include "avcodec.h" #include "internal.h" #include "get_bits.h" #include "acelp_vectors.h" #include "celp_filters.h" #include "celp_math.h" #include "g723_1_data.h" typedef struct g723_1_context { AVClass *class; AVFrame frame; 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 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 interp_index; int interp_gain; int sid_gain; int cur_gain; int reflection_coef; int pf_gain; ///< formant postfilter ///< gain scaling unit memory int postfilter; int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX]; int16_t prev_data[HALF_FRAME_LEN]; int16_t prev_weight_sig[PITCH_MAX]; int16_t hpf_fir_mem; ///< highpass filter fir int hpf_iir_mem; ///< and iir memories int16_t perf_fir_mem[LPC_ORDER]; ///< perceptual filter fir int16_t perf_iir_mem[LPC_ORDER]; ///< and iir memories int16_t harmonic_mem[PITCH_MAX]; } 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; p->pf_gain = 1 << 12; avcodec_get_frame_defaults(&p->frame); avctx->coded_frame = &p->frame; memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp)); 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_bits1(&gb); p->subframe[1].grid_index = get_bits1(&gb); p->subframe[2].grid_index = get_bits1(&gb); p->subframe[3].grid_index = get_bits1(&gb); if (p->cur_rate == RATE_6300) { skip_bits1(&gb); /* 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) { return (ff_sqrt(val << 1) >> 1) & (~1); } /** * Calculate the number of left-shifts required for normalizing the input. * * @param num input number * @param width width of the input, 15 or 31 bits */ static int normalize_bits(int num, int width) { int i = 0; if (num) { if (num == -1) return width; if (num < 0) num = ~num; i= width - av_log2(num) - 1; i= FFMAX(i, 0); } return i; } #define normalize_bits_int16(num) normalize_bits(num, 15) #define normalize_bits_int32(num) normalize_bits(num, 31) /** * 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]); bits= 14 - av_log2_16bit(max); bits= FFMAX(bits, 0); 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) \ MULL(a,b,15) /** * Convert LSP frequencies to LPC coefficients. * * @param lpc buffer for LPC coefficients */ static void lsp2lpc(int16_t *lpc) { int f1[LPC_ORDER / 2 + 1]; int f2[LPC_ORDER / 2 + 1]; int i, j; /* Calculate negative cosine */ for (j = 0; j < LPC_ORDER; j++) { int index = lpc[j] >> 7; int offset = lpc[j] & 0x7f; 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 sum = ff_dot_product(a,b,length); return av_sat_add32(sum, 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 = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER); vector[i] = av_sat_dadd32(1 << 15, av_sat_add32(sum, 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] = av_clipl_int32(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 * @param width width of the output, 16 bits(0) / 32 bits(1) */ #define iir_filter(fir_coef, iir_coef, src, dest, width)\ {\ int m, n;\ int res_shift = 16 & ~-(width);\ int in_shift = 16 - res_shift;\ \ 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] >> in_shift);\ }\ \ (dest)[m] = av_clipl_int32(((src)[m] << 16) + (filter << 3) +\ (1 << 15)) >> res_shift;\ }\ } /** * 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, 1); lpc += LPC_ORDER; } memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t)); memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int)); 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 g723_1_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt) { G723_1_Context *p = avctx->priv_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; } p->frame.nb_samples = FRAME_LEN; if ((ret = avctx->get_buffer(avctx, &p->frame)) < 0) { av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); return ret; } out = (int16_t *)p->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(p->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)); } } } else { memset(out, 0, FRAME_LEN * 2); av_log(avctx, AV_LOG_WARNING, "G.723.1: Comfort noise generation not supported yet\n"); *got_frame_ptr = 1; *(AVFrame *)data = p->frame; return frame_size[dec_mode]; } 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; *(AVFrame *)data = p->frame; 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, { 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", .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, .long_name = NULL_IF_CONFIG_SMALL("G.723.1"), .capabilities = CODEC_CAP_SUBFRAMES | 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]; 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 = dot_product(lsp + (offset), temp, size) << 1;\ error -= 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, 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, 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, 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, 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++) { 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++] = 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(int16_t *residual, 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; 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 = 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 = 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 = 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 = -1 << 30; 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) 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); } 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 = (const int16_t *)frame->data[0]; HFParam hf[4]; int i, j; 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); 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); inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0); 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); 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 */ 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; } if ((ret = ff_alloc_packet2(avctx, avpkt, 24))) 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", .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, .long_name = NULL_IF_CONFIG_SMALL("G.723.1"), .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE}, }; #endif