| /* |
| * 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 |
| */ |
| |
| #include "libavutil/channel_layout.h" |
| #include "libavutil/mem.h" |
| #include "libavutil/opt.h" |
| |
| #define BITSTREAM_READER_LE |
| #include "acelp_vectors.h" |
| #include "avcodec.h" |
| #include "celp_filters.h" |
| #include "celp_math.h" |
| #include "get_bits.h" |
| #include "internal.h" |
| #include "g723_1.h" |
| |
| #define CNG_RANDOM_SEED 12345 |
| |
| static av_cold int g723_1_decode_init(AVCodecContext *avctx) |
| { |
| G723_1_Context *s = avctx->priv_data; |
| |
| avctx->sample_fmt = AV_SAMPLE_FMT_S16P; |
| if (avctx->channels < 1 || avctx->channels > 2) { |
| av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n", avctx->channels); |
| return AVERROR(EINVAL); |
| } |
| avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO; |
| for (int ch = 0; ch < avctx->channels; ch++) { |
| G723_1_ChannelContext *p = &s->ch[ch]; |
| |
| p->pf_gain = 1 << 12; |
| |
| memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp)); |
| memcpy(p->sid_lsp, dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp)); |
| |
| p->cng_random_seed = CNG_RANDOM_SEED; |
| p->past_frame_type = SID_FRAME; |
| } |
| |
| 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_ChannelContext *p, const uint8_t *buf, |
| int buf_size) |
| { |
| GetBitContext gb; |
| int ad_cb_len; |
| int temp, info_bits, i; |
| int ret; |
| |
| ret = init_get_bits8(&gb, buf, buf_size); |
| if (ret < 0) |
| return ret; |
| |
| /* 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(unsigned val) |
| { |
| av_assert2(!(val & 0x80000000)); |
| |
| return (ff_sqrt(val << 1) >> 1) & (~1); |
| } |
| |
| /** |
| * 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) |
| ff_g723_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; |
| } |
| } |
| } |
| |
| /** |
| * 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 = ff_g723_1_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_ChannelContext *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] = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN); |
| |
| /* Compute forward residual energy */ |
| if (fwd_lag) |
| energy[2] = ff_g723_1_dot_product(buf + fwd_lag, buf + fwd_lag, |
| SUBFRAME_LEN); |
| |
| /* Compute backward residual energy */ |
| if (back_lag) |
| energy[4] = ff_g723_1_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 = ff_g723_1_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_ChannelContext *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 = ff_g723_1_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 = ff_g723_1_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 = ff_g723_1_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; |
| } |
| |
| /** |
| * Perform 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 = (int16_t)(*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] * 65536) + (filter * 8) +\ |
| (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_ChannelContext *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 = ff_g723_1_normalize_bits(num, 31); |
| bits2 = ff_g723_1_normalize_bits(denom, 31); |
| num = num << bits1 >> 1; |
| denom <<= bits2; |
| |
| bits2 = 5 + bits1 - bits2; |
| bits2 = av_clip_uintp2(bits2, 5); |
| |
| 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_ChannelContext *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 = ff_g723_1_scale_vector(dst, buf, SUBFRAME_LEN); |
| |
| /* Compute auto correlation coefficients */ |
| auto_corr[0] = ff_g723_1_dot_product(dst, dst + 1, SUBFRAME_LEN - 1); |
| auto_corr[1] = ff_g723_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 sid_gain_to_lsp_index(int gain) |
| { |
| if (gain < 0x10) |
| return gain << 6; |
| else if (gain < 0x20) |
| return gain - 8 << 7; |
| else |
| return gain - 20 << 8; |
| } |
| |
| static inline int cng_rand(int *state, int base) |
| { |
| *state = (*state * 521 + 259) & 0xFFFF; |
| return (*state & 0x7FFF) * base >> 15; |
| } |
| |
| static int estimate_sid_gain(G723_1_ChannelContext *p) |
| { |
| int i, shift, seg, seg2, t, val, val_add, x, y; |
| |
| shift = 16 - p->cur_gain * 2; |
| if (shift > 0) { |
| if (p->sid_gain == 0) { |
| t = 0; |
| } else if (shift >= 31 || (int32_t)((uint32_t)p->sid_gain << shift) >> shift != p->sid_gain) { |
| if (p->sid_gain < 0) t = INT32_MIN; |
| else t = INT32_MAX; |
| } else |
| t = p->sid_gain << shift; |
| }else |
| t = p->sid_gain >> -shift; |
| x = av_clipl_int32(t * (int64_t)cng_filt[0] >> 16); |
| |
| if (x >= cng_bseg[2]) |
| return 0x3F; |
| |
| if (x >= cng_bseg[1]) { |
| shift = 4; |
| seg = 3; |
| } else { |
| shift = 3; |
| seg = (x >= cng_bseg[0]); |
| } |
| seg2 = FFMIN(seg, 3); |
| |
| val = 1 << shift; |
| val_add = val >> 1; |
| for (i = 0; i < shift; i++) { |
| t = seg * 32 + (val << seg2); |
| t *= t; |
| if (x >= t) |
| val += val_add; |
| else |
| val -= val_add; |
| val_add >>= 1; |
| } |
| |
| t = seg * 32 + (val << seg2); |
| y = t * t - x; |
| if (y <= 0) { |
| t = seg * 32 + (val + 1 << seg2); |
| t = t * t - x; |
| val = (seg2 - 1) * 16 + val; |
| if (t >= y) |
| val++; |
| } else { |
| t = seg * 32 + (val - 1 << seg2); |
| t = t * t - x; |
| val = (seg2 - 1) * 16 + val; |
| if (t >= y) |
| val--; |
| } |
| |
| return val; |
| } |
| |
| static void generate_noise(G723_1_ChannelContext *p) |
| { |
| int i, j, idx, t; |
| int off[SUBFRAMES]; |
| int signs[SUBFRAMES / 2 * 11], pos[SUBFRAMES / 2 * 11]; |
| int tmp[SUBFRAME_LEN * 2]; |
| int16_t *vector_ptr; |
| int64_t sum; |
| int b0, c, delta, x, shift; |
| |
| p->pitch_lag[0] = cng_rand(&p->cng_random_seed, 21) + 123; |
| p->pitch_lag[1] = cng_rand(&p->cng_random_seed, 19) + 123; |
| |
| for (i = 0; i < SUBFRAMES; i++) { |
| p->subframe[i].ad_cb_gain = cng_rand(&p->cng_random_seed, 50) + 1; |
| p->subframe[i].ad_cb_lag = cng_adaptive_cb_lag[i]; |
| } |
| |
| for (i = 0; i < SUBFRAMES / 2; i++) { |
| t = cng_rand(&p->cng_random_seed, 1 << 13); |
| off[i * 2] = t & 1; |
| off[i * 2 + 1] = ((t >> 1) & 1) + SUBFRAME_LEN; |
| t >>= 2; |
| for (j = 0; j < 11; j++) { |
| signs[i * 11 + j] = ((t & 1) * 2 - 1) * (1 << 14); |
| t >>= 1; |
| } |
| } |
| |
| idx = 0; |
| for (i = 0; i < SUBFRAMES; i++) { |
| for (j = 0; j < SUBFRAME_LEN / 2; j++) |
| tmp[j] = j; |
| t = SUBFRAME_LEN / 2; |
| for (j = 0; j < pulses[i]; j++, idx++) { |
| int idx2 = cng_rand(&p->cng_random_seed, t); |
| |
| pos[idx] = tmp[idx2] * 2 + off[i]; |
| tmp[idx2] = tmp[--t]; |
| } |
| } |
| |
| vector_ptr = p->audio + LPC_ORDER; |
| memcpy(vector_ptr, p->prev_excitation, |
| PITCH_MAX * sizeof(*p->excitation)); |
| for (i = 0; i < SUBFRAMES; i += 2) { |
| ff_g723_1_gen_acb_excitation(vector_ptr, vector_ptr, |
| p->pitch_lag[i >> 1], &p->subframe[i], |
| p->cur_rate); |
| ff_g723_1_gen_acb_excitation(vector_ptr + SUBFRAME_LEN, |
| vector_ptr + SUBFRAME_LEN, |
| p->pitch_lag[i >> 1], &p->subframe[i + 1], |
| p->cur_rate); |
| |
| t = 0; |
| for (j = 0; j < SUBFRAME_LEN * 2; j++) |
| t |= FFABS(vector_ptr[j]); |
| t = FFMIN(t, 0x7FFF); |
| if (!t) { |
| shift = 0; |
| } else { |
| shift = -10 + av_log2(t); |
| if (shift < -2) |
| shift = -2; |
| } |
| sum = 0; |
| if (shift < 0) { |
| for (j = 0; j < SUBFRAME_LEN * 2; j++) { |
| t = vector_ptr[j] * (1 << -shift); |
| sum += t * t; |
| tmp[j] = t; |
| } |
| } else { |
| for (j = 0; j < SUBFRAME_LEN * 2; j++) { |
| t = vector_ptr[j] >> shift; |
| sum += t * t; |
| tmp[j] = t; |
| } |
| } |
| |
| b0 = 0; |
| for (j = 0; j < 11; j++) |
| b0 += tmp[pos[(i / 2) * 11 + j]] * signs[(i / 2) * 11 + j]; |
| b0 = b0 * 2 * 2979LL + (1 << 29) >> 30; // approximated division by 11 |
| |
| c = p->cur_gain * (p->cur_gain * SUBFRAME_LEN >> 5); |
| if (shift * 2 + 3 >= 0) |
| c >>= shift * 2 + 3; |
| else |
| c <<= -(shift * 2 + 3); |
| c = (av_clipl_int32(sum << 1) - c) * 2979LL >> 15; |
| |
| delta = b0 * b0 * 2 - c; |
| if (delta <= 0) { |
| x = -b0; |
| } else { |
| delta = square_root(delta); |
| x = delta - b0; |
| t = delta + b0; |
| if (FFABS(t) < FFABS(x)) |
| x = -t; |
| } |
| shift++; |
| if (shift < 0) |
| x >>= -shift; |
| else |
| x *= 1 << shift; |
| x = av_clip(x, -10000, 10000); |
| |
| for (j = 0; j < 11; j++) { |
| idx = (i / 2) * 11 + j; |
| vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] + |
| (x * signs[idx] >> 15)); |
| } |
| |
| /* copy decoded data to serve as a history for the next decoded subframes */ |
| memcpy(vector_ptr + PITCH_MAX, vector_ptr, |
| sizeof(*vector_ptr) * SUBFRAME_LEN * 2); |
| vector_ptr += SUBFRAME_LEN * 2; |
| } |
| /* Save the excitation for the next frame */ |
| memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN, |
| PITCH_MAX * sizeof(*p->excitation)); |
| } |
| |
| static int g723_1_decode_frame(AVCodecContext *avctx, void *data, |
| int *got_frame_ptr, AVPacket *avpkt) |
| { |
| G723_1_Context *s = avctx->priv_data; |
| AVFrame *frame = 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; |
| |
| if (buf_size < frame_size[dec_mode] * avctx->channels) { |
| 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; |
| } |
| |
| frame->nb_samples = FRAME_LEN; |
| if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
| return ret; |
| |
| for (int ch = 0; ch < avctx->channels; ch++) { |
| G723_1_ChannelContext *p = &s->ch[ch]; |
| int16_t *audio = p->audio; |
| |
| if (unpack_bitstream(p, buf + ch * (buf_size / avctx->channels), |
| buf_size / avctx->channels) < 0) { |
| bad_frame = 1; |
| if (p->past_frame_type == ACTIVE_FRAME) |
| p->cur_frame_type = ACTIVE_FRAME; |
| else |
| p->cur_frame_type = UNTRANSMITTED_FRAME; |
| } |
| |
| out = (int16_t *)frame->extended_data[ch]; |
| |
| if (p->cur_frame_type == ACTIVE_FRAME) { |
| if (!bad_frame) |
| p->erased_frames = 0; |
| else if (p->erased_frames != 3) |
| p->erased_frames++; |
| |
| ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame); |
| ff_g723_1_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); |
| ff_g723_1_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] * 2); |
| 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); |
| |
| /* Perform pitch postfiltering */ |
| if (s->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(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)); |
| } |
| } |
| p->cng_random_seed = CNG_RANDOM_SEED; |
| } else { |
| if (p->cur_frame_type == SID_FRAME) { |
| p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index); |
| ff_g723_1_inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0); |
| } else if (p->past_frame_type == ACTIVE_FRAME) { |
| p->sid_gain = estimate_sid_gain(p); |
| } |
| |
| if (p->past_frame_type == ACTIVE_FRAME) |
| p->cur_gain = p->sid_gain; |
| else |
| p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3; |
| generate_noise(p); |
| ff_g723_1_lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp); |
| /* Save the lsp_vector for the next frame */ |
| memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp)); |
| } |
| |
| 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 (s->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; |
| |
| return frame_size[dec_mode] * avctx->channels; |
| } |
| |
| #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", "enable postfilter", OFFSET(postfilter), AV_OPT_TYPE_BOOL, |
| { .i64 = 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", |
| .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_decode_init, |
| .decode = g723_1_decode_frame, |
| .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1, |
| .priv_class = &g723_1dec_class, |
| }; |