/* * G.723.1 compatible decoder * Copyright (c) 2006 Benjamin Larsson * Copyright (c) 2010 Mohamed Naufal Basheer * * This file is part of Libav. * * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * G.723.1 compatible decoder */ #define BITSTREAM_READER_LE #include "libavutil/audioconvert.h" #include "libavutil/lzo.h" #include "libavutil/opt.h" #include "avcodec.h" #include "get_bits.h" #include "acelp_vectors.h" #include "celp_filters.h" #include "g723_1_data.h" /** * G723.1 frame types */ enum FrameType { ACTIVE_FRAME, ///< Active speech SID_FRAME, ///< Silence Insertion Descriptor frame UNTRANSMITTED_FRAME }; enum Rate { RATE_6300, RATE_5300 }; /** * G723.1 unpacked data subframe */ typedef struct { int ad_cb_lag; ///< adaptive codebook lag int ad_cb_gain; int dirac_train; int pulse_sign; int grid_index; int amp_index; int pulse_pos; } G723_1_Subframe; /** * Pitch postfilter parameters */ typedef struct { int index; ///< postfilter backward/forward lag int16_t opt_gain; ///< optimal gain int16_t sc_gain; ///< scaling gain } PPFParam; typedef struct g723_1_context { AVClass *class; 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; int postfilter; int16_t audio[FRAME_LEN + LPC_ORDER + 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; avctx->sample_rate = 8000; 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_bits(&gb, 1); p->subframe[1].grid_index = get_bits(&gb, 1); p->subframe[2].grid_index = get_bits(&gb, 1); p->subframe[3].grid_index = get_bits(&gb, 1); if (p->cur_rate == RATE_6300) { skip_bits(&gb, 1); /* skip reserved bit */ /* Compute pulse_pos index using the 13-bit combined position index */ temp = get_bits(&gb, 13); p->subframe[0].pulse_pos = temp / 810; temp -= p->subframe[0].pulse_pos * 810; p->subframe[1].pulse_pos = FASTDIV(temp, 90); temp -= p->subframe[1].pulse_pos * 90; p->subframe[2].pulse_pos = FASTDIV(temp, 9); p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9; p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) + get_bits(&gb, 16); p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) + get_bits(&gb, 14); p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) + get_bits(&gb, 16); p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) + get_bits(&gb, 14); p->subframe[0].pulse_sign = get_bits(&gb, 6); p->subframe[1].pulse_sign = get_bits(&gb, 5); p->subframe[2].pulse_sign = get_bits(&gb, 6); p->subframe[3].pulse_sign = get_bits(&gb, 5); } else { /* 5300 bps */ p->subframe[0].pulse_pos = get_bits(&gb, 12); p->subframe[1].pulse_pos = get_bits(&gb, 12); p->subframe[2].pulse_pos = get_bits(&gb, 12); p->subframe[3].pulse_pos = get_bits(&gb, 12); p->subframe[0].pulse_sign = get_bits(&gb, 4); p->subframe[1].pulse_sign = get_bits(&gb, 4); p->subframe[2].pulse_sign = get_bits(&gb, 4); p->subframe[3].pulse_sign = get_bits(&gb, 4); } return 0; } /** * Bitexact implementation of sqrt(val/2). */ static int16_t square_root(int val) { int16_t res = 0; int16_t exp = 0x4000; int i; for (i = 0; i < 14; i ++) { int res_exp = res + exp; if (val >= res_exp * res_exp << 1) res += exp; exp >>= 1; } return res; } /** * Calculate the number of left-shifts required for normalizing the input. * * @param num input number * @param width width of the input, 16 bits(0) / 32 bits(1) */ static int normalize_bits(int num, int width) { if (!num) return 0; if (num == -1) return width; if (num < 0) num = ~num; return width - av_log2(num) - 1; } /** * Scale vector contents based on the largest of their absolutes. */ static int scale_vector(int16_t *dst, const int16_t *vector, int length) { int bits, max = 0; int i; for (i = 0; i < length; i++) max |= FFABS(vector[i]); max = FFMIN(max, 0x7FFF); bits = normalize_bits(max, 15); if (bits == 15) for (i = 0; i < length; i++) dst[i] = vector[i] * 0x7fff >> 3; else for (i = 0; i < length; i++) dst[i] = vector[i] << bits >> 3; return bits - 3; } /** * Perform inverse quantization of LSP frequencies. * * @param cur_lsp the current LSP vector * @param prev_lsp the previous LSP vector * @param lsp_index VQ indices * @param bad_frame bad frame flag */ static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp, uint8_t *lsp_index, int bad_frame) { int min_dist, pred; int i, j, temp, stable; /* Check for frame erasure */ if (!bad_frame) { min_dist = 0x100; pred = 12288; } else { min_dist = 0x200; pred = 23552; lsp_index[0] = lsp_index[1] = lsp_index[2] = 0; } /* Get the VQ table entry corresponding to the transmitted index */ cur_lsp[0] = lsp_band0[lsp_index[0]][0]; cur_lsp[1] = lsp_band0[lsp_index[0]][1]; cur_lsp[2] = lsp_band0[lsp_index[0]][2]; cur_lsp[3] = lsp_band1[lsp_index[1]][0]; cur_lsp[4] = lsp_band1[lsp_index[1]][1]; cur_lsp[5] = lsp_band1[lsp_index[1]][2]; cur_lsp[6] = lsp_band2[lsp_index[2]][0]; cur_lsp[7] = lsp_band2[lsp_index[2]][1]; cur_lsp[8] = lsp_band2[lsp_index[2]][2]; cur_lsp[9] = lsp_band2[lsp_index[2]][3]; /* Add predicted vector & DC component to the previously quantized vector */ for (i = 0; i < LPC_ORDER; i++) { temp = ((prev_lsp[i] - dc_lsp[i]) * pred + (1 << 14)) >> 15; cur_lsp[i] += dc_lsp[i] + temp; } for (i = 0; i < LPC_ORDER; i++) { cur_lsp[0] = FFMAX(cur_lsp[0], 0x180); cur_lsp[LPC_ORDER - 1] = FFMIN(cur_lsp[LPC_ORDER - 1], 0x7e00); /* Stability check */ for (j = 1; j < LPC_ORDER; j++) { temp = min_dist + cur_lsp[j - 1] - cur_lsp[j]; if (temp > 0) { temp >>= 1; cur_lsp[j - 1] -= temp; cur_lsp[j] += temp; } } stable = 1; for (j = 1; j < LPC_ORDER; j++) { temp = cur_lsp[j - 1] + min_dist - cur_lsp[j] - 4; if (temp > 0) { stable = 0; break; } } if (stable) break; } if (!stable) memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(*cur_lsp)); } /** * Bitexact implementation of 2ab scaled by 1/2^16. * * @param a 32 bit multiplicand * @param b 16 bit multiplier */ #define MULL2(a, b) \ ((((a) >> 16) * (b) << 1) + (((a) & 0xffff) * (b) >> 15)) /** * Convert LSP frequencies to LPC coefficients. * * @param lpc buffer for LPC coefficients */ static void lsp2lpc(int16_t *lpc) { int f1[LPC_ORDER / 2 + 1]; int f2[LPC_ORDER / 2 + 1]; int i, j; /* Calculate negative cosine */ for (j = 0; j < LPC_ORDER; j++) { int index = lpc[j] >> 7; int offset = lpc[j] & 0x7f; int temp1 = cos_tab[index] << 16; int temp2 = (cos_tab[index + 1] - cos_tab[index]) * ((offset << 8) + 0x80) << 1; lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 16); } /* * Compute sum and difference polynomial coefficients * (bitexact alternative to lsp2poly() in lsp.c) */ /* Initialize with values in Q28 */ f1[0] = 1 << 28; f1[1] = (lpc[0] << 14) + (lpc[2] << 14); f1[2] = lpc[0] * lpc[2] + (2 << 28); f2[0] = 1 << 28; f2[1] = (lpc[1] << 14) + (lpc[3] << 14); f2[2] = lpc[1] * lpc[3] + (2 << 28); /* * Calculate and scale the coefficients by 1/2 in * each iteration for a final scaling factor of Q25 */ for (i = 2; i < LPC_ORDER / 2; i++) { f1[i + 1] = f1[i - 1] + MULL2(f1[i], lpc[2 * i]); f2[i + 1] = f2[i - 1] + MULL2(f2[i], lpc[2 * i + 1]); for (j = i; j >= 2; j--) { f1[j] = MULL2(f1[j - 1], lpc[2 * i]) + (f1[j] >> 1) + (f1[j - 2] >> 1); f2[j] = MULL2(f2[j - 1], lpc[2 * i + 1]) + (f2[j] >> 1) + (f2[j - 2] >> 1); } f1[0] >>= 1; f2[0] >>= 1; f1[1] = ((lpc[2 * i] << 16 >> i) + f1[1]) >> 1; f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1; } /* Convert polynomial coefficients to LPC coefficients */ for (i = 0; i < LPC_ORDER / 2; i++) { int64_t ff1 = f1[i + 1] + f1[i]; int64_t ff2 = f2[i + 1] - f2[i]; lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) + (1 << 15)) >> 16; lpc[LPC_ORDER - i - 1] = av_clipl_int32(((ff1 - ff2) << 3) + (1 << 15)) >> 16; } } /** * Quantize LSP frequencies by interpolation and convert them to * the corresponding LPC coefficients. * * @param lpc buffer for LPC coefficients * @param cur_lsp the current LSP vector * @param prev_lsp the previous LSP vector */ static void lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp) { int i; int16_t *lpc_ptr = lpc; /* cur_lsp * 0.25 + prev_lsp * 0.75 */ ff_acelp_weighted_vector_sum(lpc, cur_lsp, prev_lsp, 4096, 12288, 1 << 13, 14, LPC_ORDER); ff_acelp_weighted_vector_sum(lpc + LPC_ORDER, cur_lsp, prev_lsp, 8192, 8192, 1 << 13, 14, LPC_ORDER); ff_acelp_weighted_vector_sum(lpc + 2 * LPC_ORDER, cur_lsp, prev_lsp, 12288, 4096, 1 << 13, 14, LPC_ORDER); memcpy(lpc + 3 * LPC_ORDER, cur_lsp, LPC_ORDER * sizeof(*lpc)); for (i = 0; i < SUBFRAMES; i++) { lsp2lpc(lpc_ptr); lpc_ptr += LPC_ORDER; } } /** * Generate a train of dirac functions with period as pitch lag. */ static void gen_dirac_train(int16_t *buf, int pitch_lag) { int16_t vector[SUBFRAME_LEN]; int i, j; memcpy(vector, buf, SUBFRAME_LEN * sizeof(*vector)); for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) { for (j = 0; j < SUBFRAME_LEN - i; j++) buf[i + j] += vector[j]; } } /** * Generate fixed codebook excitation vector. * * @param vector decoded excitation vector * @param subfrm current subframe * @param cur_rate current bitrate * @param pitch_lag closed loop pitch lag * @param index current subframe index */ static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe subfrm, enum Rate cur_rate, int pitch_lag, int index) { int temp, i, j; memset(vector, 0, SUBFRAME_LEN * sizeof(*vector)); if (cur_rate == RATE_6300) { if (subfrm.pulse_pos >= max_pos[index]) return; /* Decode amplitudes and positions */ j = PULSE_MAX - pulses[index]; temp = subfrm.pulse_pos; for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) { temp -= combinatorial_table[j][i]; if (temp >= 0) continue; temp += combinatorial_table[j++][i]; if (subfrm.pulse_sign & (1 << (PULSE_MAX - j))) { vector[subfrm.grid_index + GRID_SIZE * i] = -fixed_cb_gain[subfrm.amp_index]; } else { vector[subfrm.grid_index + GRID_SIZE * i] = fixed_cb_gain[subfrm.amp_index]; } if (j == PULSE_MAX) break; } if (subfrm.dirac_train == 1) gen_dirac_train(vector, pitch_lag); } else { /* 5300 bps */ int cb_gain = fixed_cb_gain[subfrm.amp_index]; int cb_shift = subfrm.grid_index; int cb_sign = subfrm.pulse_sign; int cb_pos = subfrm.pulse_pos; int offset, beta, lag; for (i = 0; i < 8; i += 2) { offset = ((cb_pos & 7) << 3) + cb_shift + i; vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain; cb_pos >>= 3; cb_sign >>= 1; } /* Enhance harmonic components */ lag = pitch_contrib[subfrm.ad_cb_gain << 1] + pitch_lag + subfrm.ad_cb_lag - 1; beta = pitch_contrib[(subfrm.ad_cb_gain << 1) + 1]; if (lag < SUBFRAME_LEN - 2) { for (i = lag; i < SUBFRAME_LEN; i++) vector[i] += beta * vector[i - lag] >> 15; } } } /** * Get delayed contribution from the previous excitation vector. */ static void get_residual(int16_t *residual, int16_t *prev_excitation, int lag) { int offset = PITCH_MAX - PITCH_ORDER / 2 - lag; int i; residual[0] = prev_excitation[offset]; residual[1] = prev_excitation[offset + 1]; offset += 2; for (i = 2; i < SUBFRAME_LEN + PITCH_ORDER - 1; i++) residual[i] = prev_excitation[offset + (i - 2) % lag]; } static int dot_product(const int16_t *a, const int16_t *b, int length) { int i, sum = 0; for (i = 0; i < length; i++) { int prod = a[i] * b[i]; sum = av_sat_dadd32(sum, prod); } return sum; } /** * Generate adaptive codebook excitation. */ static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation, int pitch_lag, G723_1_Subframe subfrm, enum Rate cur_rate) { int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1]; const int16_t *cb_ptr; int lag = pitch_lag + subfrm.ad_cb_lag - 1; int i; int sum; get_residual(residual, prev_excitation, lag); /* Select quantization table */ if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2) cb_ptr = adaptive_cb_gain85; else cb_ptr = adaptive_cb_gain170; /* Calculate adaptive vector */ cb_ptr += subfrm.ad_cb_gain * 20; for (i = 0; i < SUBFRAME_LEN; i++) { sum = dot_product(residual + i, cb_ptr, PITCH_ORDER); vector[i] = av_sat_dadd32(1 << 15, sum) >> 16; } } /** * Estimate maximum auto-correlation around pitch lag. * * @param buf buffer with offset applied * @param offset offset of the excitation vector * @param ccr_max pointer to the maximum auto-correlation * @param pitch_lag decoded pitch lag * @param length length of autocorrelation * @param dir forward lag(1) / backward lag(-1) */ static int autocorr_max(const int16_t *buf, int offset, int *ccr_max, int pitch_lag, int length, int dir) { int limit, ccr, lag = 0; int i; pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag); if (dir > 0) limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3); else limit = pitch_lag + 3; for (i = pitch_lag - 3; i <= limit; i++) { ccr = dot_product(buf, buf + dir * i, length); if (ccr > *ccr_max) { *ccr_max = ccr; lag = i; } } return lag; } /** * Calculate pitch postfilter optimal and scaling gains. * * @param lag pitch postfilter forward/backward lag * @param ppf pitch postfilter parameters * @param cur_rate current bitrate * @param tgt_eng target energy * @param ccr cross-correlation * @param res_eng residual energy */ static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate, int tgt_eng, int ccr, int res_eng) { int pf_residual; /* square of postfiltered residual */ int temp1, temp2; ppf->index = lag; temp1 = tgt_eng * res_eng >> 1; temp2 = ccr * ccr << 1; if (temp2 > temp1) { if (ccr >= res_eng) { ppf->opt_gain = ppf_gain_weight[cur_rate]; } else { ppf->opt_gain = (ccr << 15) / res_eng * ppf_gain_weight[cur_rate] >> 15; } /* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */ temp1 = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1); temp2 = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng; pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16; if (tgt_eng >= pf_residual << 1) { temp1 = 0x7fff; } else { temp1 = (tgt_eng << 14) / pf_residual; } /* scaling_gain = sqrt(tgt_eng/pf_res^2) */ ppf->sc_gain = square_root(temp1 << 16); } else { ppf->opt_gain = 0; ppf->sc_gain = 0x7fff; } ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15); } /** * Calculate pitch postfilter parameters. * * @param p the context * @param offset offset of the excitation vector * @param pitch_lag decoded pitch lag * @param ppf pitch postfilter parameters * @param cur_rate current bitrate */ static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag, PPFParam *ppf, enum Rate cur_rate) { int16_t scale; int i; int temp1, temp2; /* * 0 - target energy * 1 - forward cross-correlation * 2 - forward residual energy * 3 - backward cross-correlation * 4 - backward residual energy */ int energy[5] = {0, 0, 0, 0, 0}; int16_t *buf = p->audio + LPC_ORDER + offset; int fwd_lag = autocorr_max(buf, offset, &energy[1], pitch_lag, SUBFRAME_LEN, 1); int back_lag = autocorr_max(buf, offset, &energy[3], pitch_lag, SUBFRAME_LEN, -1); ppf->index = 0; ppf->opt_gain = 0; ppf->sc_gain = 0x7fff; /* Case 0, Section 3.6 */ if (!back_lag && !fwd_lag) return; /* Compute target energy */ energy[0] = dot_product(buf, buf, SUBFRAME_LEN); /* Compute forward residual energy */ if (fwd_lag) energy[2] = dot_product(buf + fwd_lag, buf + fwd_lag, SUBFRAME_LEN); /* Compute backward residual energy */ if (back_lag) energy[4] = dot_product(buf - back_lag, buf - back_lag, SUBFRAME_LEN); /* Normalize and shorten */ temp1 = 0; for (i = 0; i < 5; i++) temp1 = FFMAX(energy[i], temp1); scale = normalize_bits(temp1, 31); for (i = 0; i < 5; i++) energy[i] = (energy[i] << scale) >> 16; if (fwd_lag && !back_lag) { /* Case 1 */ comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1], energy[2]); } else if (!fwd_lag) { /* Case 2 */ comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3], energy[4]); } else { /* Case 3 */ /* * Select the largest of energy[1]^2/energy[2] * and energy[3]^2/energy[4] */ temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15); temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15); if (temp1 >= temp2) { comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1], energy[2]); } else { comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3], energy[4]); } } } /** * Classify frames as voiced/unvoiced. * * @param p the context * @param pitch_lag decoded pitch_lag * @param exc_eng excitation energy estimation * @param scale scaling factor of exc_eng * * @return residual interpolation index if voiced, 0 otherwise */ static int comp_interp_index(G723_1_Context *p, int pitch_lag, int *exc_eng, int *scale) { int offset = PITCH_MAX + 2 * SUBFRAME_LEN; int16_t *buf = p->audio + LPC_ORDER; int index, ccr, tgt_eng, best_eng, temp; *scale = scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX); buf += offset; /* Compute maximum backward cross-correlation */ ccr = 0; index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1); ccr = av_sat_add32(ccr, 1 << 15) >> 16; /* Compute target energy */ tgt_eng = dot_product(buf, buf, SUBFRAME_LEN * 2); *exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16; if (ccr <= 0) return 0; /* Compute best energy */ best_eng = dot_product(buf - index, buf - index, SUBFRAME_LEN * 2); best_eng = av_sat_add32(best_eng, 1 << 15) >> 16; temp = best_eng * *exc_eng >> 3; if (temp < ccr * ccr) return index; else return 0; } /** * Peform residual interpolation based on frame classification. * * @param buf decoded excitation vector * @param out output vector * @param lag decoded pitch lag * @param gain interpolated gain * @param rseed seed for random number generator */ static void residual_interp(int16_t *buf, int16_t *out, int lag, int gain, int *rseed) { int i; if (lag) { /* Voiced */ int16_t *vector_ptr = buf + PITCH_MAX; /* Attenuate */ for (i = 0; i < lag; i++) out[i] = vector_ptr[i - lag] * 3 >> 2; av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out), (FRAME_LEN - lag) * sizeof(*out)); } else { /* Unvoiced */ for (i = 0; i < FRAME_LEN; i++) { *rseed = *rseed * 521 + 259; out[i] = gain * *rseed >> 15; } memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(*buf)); } } /** * Perform IIR filtering. * * @param fir_coef FIR coefficients * @param iir_coef IIR coefficients * @param src source vector * @param dest destination vector */ static inline void iir_filter(int16_t *fir_coef, int16_t *iir_coef, int16_t *src, int *dest) { int m, n; for (m = 0; m < SUBFRAME_LEN; m++) { int64_t filter = 0; for (n = 1; n <= LPC_ORDER; n++) { filter -= fir_coef[n - 1] * src[m - n] - iir_coef[n - 1] * (dest[m - n] >> 16); } dest[m] = av_clipl_int32((src[m] << 16) + (filter << 3) + (1 << 15)); } } /** * Adjust gain of postfiltered signal. * * @param p the context * @param buf postfiltered output vector * @param energy input energy coefficient */ static void gain_scale(G723_1_Context *p, int16_t * buf, int energy) { int num, denom, gain, bits1, bits2; int i; num = energy; denom = 0; for (i = 0; i < SUBFRAME_LEN; i++) { int temp = buf[i] >> 2; temp *= temp; denom = av_sat_dadd32(denom, temp); } if (num && denom) { bits1 = normalize_bits(num, 31); bits2 = normalize_bits(denom, 31); num = num << bits1 >> 1; denom <<= bits2; bits2 = 5 + bits1 - bits2; bits2 = FFMAX(0, bits2); gain = (num >> 1) / (denom >> 16); gain = square_root(gain << 16 >> bits2); } else { gain = 1 << 12; } for (i = 0; i < SUBFRAME_LEN; i++) { p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4; buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) + (1 << 10)) >> 11); } } /** * Perform formant filtering. * * @param p the context * @param lpc quantized lpc coefficients * @param buf input buffer * @param dst output buffer */ static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int16_t *buf, int16_t *dst) { int16_t filter_coef[2][LPC_ORDER]; int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr; int i, j, k; memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf)); memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal)); for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) { for (k = 0; k < LPC_ORDER; k++) { filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] + (1 << 14)) >> 15; filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] + (1 << 14)) >> 15; } iir_filter(filter_coef[0], filter_coef[1], buf + i, filter_signal + i); lpc += LPC_ORDER; } memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(*p->fir_mem)); memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(*p->iir_mem)); buf += LPC_ORDER; signal_ptr = filter_signal + LPC_ORDER; for (i = 0; i < SUBFRAMES; i++) { int temp; int auto_corr[2]; int scale, energy; /* Normalize */ scale = scale_vector(dst, buf, SUBFRAME_LEN); /* Compute auto correlation coefficients */ auto_corr[0] = dot_product(dst, dst + 1, SUBFRAME_LEN - 1); auto_corr[1] = dot_product(dst, dst, SUBFRAME_LEN); /* Compute reflection coefficient */ temp = auto_corr[1] >> 16; if (temp) { temp = (auto_corr[0] >> 2) / temp; } p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2; temp = -p->reflection_coef >> 1 & ~3; /* Compensation filter */ for (j = 0; j < SUBFRAME_LEN; j++) { dst[j] = av_sat_dadd32(signal_ptr[j], (signal_ptr[j - 1] >> 16) * temp) >> 16; } /* Compute normalized signal energy */ temp = 2 * scale + 4; if (temp < 0) { energy = av_clipl_int32((int64_t)auto_corr[1] << -temp); } else energy = auto_corr[1] >> temp; gain_scale(p, dst, energy); buf += SUBFRAME_LEN; signal_ptr += SUBFRAME_LEN; dst += SUBFRAME_LEN; } } static int 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, .priv_class = &g723_1dec_class, };