/* * G.723.1 compatible encoder * Copyright (c) Mohamed Naufal * * 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 encoder */ #include #include #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 *s = avctx->priv_data; G723_1_ChannelContext *p = &s->ch[0]; 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, "Use bitrate 6300 instead of 5300.\n"); avpriv_report_missing_feature(avctx, "Bitrate 5300"); 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]; buf[i] = av_clipl_int32((int64_t)*iir + (1 << 15)) >> 16; } } /** * Estimate autocorrelation of the input vector. * * @param buf input buffer * @param autocorr autocorrelation coefficients vector */ static void comp_autocorr(int16_t *buf, int16_t *autocorr) { int i, scale, temp; int16_t vector[LPC_FRAME]; ff_g723_1_scale_vector(vector, buf, LPC_FRAME); /* Apply the Hamming window */ for (i = 0; i < LPC_FRAME; i++) vector[i] = (vector[i] * hamming_window[i] + (1 << 14)) >> 15; /* Compute the first autocorrelation coefficient */ temp = ff_dot_product(vector, vector, LPC_FRAME); /* Apply a white noise correlation factor of (1025/1024) */ temp += temp >> 10; /* Normalize */ scale = 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_ChannelContext *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_ChannelContext *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_ChannelContext *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_ChannelContext *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 *s = avctx->priv_data; G723_1_ChannelContext *p = &s->ch[0]; 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], p->cur_rate); sub_acb_contrib(residual, impulse_resp, in); fcb_search(p, impulse_resp, in, i); /* Reconstruct the excitation */ ff_g723_1_gen_acb_excitation(impulse_resp, p->prev_excitation, p->pitch_lag[i >> 1], &p->subframe[i], RATE_6300); memmove(p->prev_excitation, p->prev_excitation + SUBFRAME_LEN, sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN)); for (j = 0; j < SUBFRAME_LEN; j++) in[j] = av_clip_int16((in[j] << 1) + impulse_resp[j]); memcpy(p->prev_excitation + PITCH_MAX - SUBFRAME_LEN, in, sizeof(int16_t) * SUBFRAME_LEN); /* Update filter memories */ synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1), p->perf_fir_mem, p->perf_iir_mem, in, vector + PITCH_MAX, 0); memmove(p->harmonic_mem, p->harmonic_mem + SUBFRAME_LEN, sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN)); memcpy(p->harmonic_mem + PITCH_MAX - SUBFRAME_LEN, vector + PITCH_MAX, sizeof(int16_t) * SUBFRAME_LEN); in += SUBFRAME_LEN; offset += LPC_ORDER; } av_free(start); if ((ret = ff_alloc_packet2(avctx, avpkt, 24, 0)) < 0) return ret; *got_packet_ptr = 1; avpkt->size = pack_bitstream(p, avpkt); return 0; } static const AVCodecDefault defaults[] = { { "b", "6300" }, { NULL }, }; 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, .defaults = defaults, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, };