/* * Copyright (c) 2012 Andrew D'Addesio * Copyright (c) 2013-2014 Mozilla Corporation * * 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 * Opus SILK decoder */ #include <stdint.h> #include "opus.h" #include "opustab.h" typedef struct SilkFrame { int coded; int log_gain; int16_t nlsf[16]; float lpc[16]; float output [2 * SILK_HISTORY]; float lpc_history[2 * SILK_HISTORY]; int primarylag; int prev_voiced; } SilkFrame; struct SilkContext { AVCodecContext *avctx; int output_channels; int midonly; int subframes; int sflength; int flength; int nlsf_interp_factor; enum OpusBandwidth bandwidth; int wb; SilkFrame frame[2]; float prev_stereo_weights[2]; float stereo_weights[2]; int prev_coded_channels; }; static inline void silk_stabilize_lsf(int16_t nlsf[16], int order, const uint16_t min_delta[17]) { int pass, i; for (pass = 0; pass < 20; pass++) { int k, min_diff = 0; for (i = 0; i < order+1; i++) { int low = i != 0 ? nlsf[i-1] : 0; int high = i != order ? nlsf[i] : 32768; int diff = (high - low) - (min_delta[i]); if (diff < min_diff) { min_diff = diff; k = i; if (pass == 20) break; } } if (min_diff == 0) /* no issues; stabilized */ return; /* wiggle one or two LSFs */ if (k == 0) { /* repel away from lower bound */ nlsf[0] = min_delta[0]; } else if (k == order) { /* repel away from higher bound */ nlsf[order-1] = 32768 - min_delta[order]; } else { /* repel away from current position */ int min_center = 0, max_center = 32768, center_val; /* lower extent */ for (i = 0; i < k; i++) min_center += min_delta[i]; min_center += min_delta[k] >> 1; /* upper extent */ for (i = order; i > k; i--) max_center -= min_delta[i]; max_center -= min_delta[k] >> 1; /* move apart */ center_val = nlsf[k - 1] + nlsf[k]; center_val = (center_val >> 1) + (center_val & 1); // rounded divide by 2 center_val = FFMIN(max_center, FFMAX(min_center, center_val)); nlsf[k - 1] = center_val - (min_delta[k] >> 1); nlsf[k] = nlsf[k - 1] + min_delta[k]; } } /* resort to the fall-back method, the standard method for LSF stabilization */ /* sort; as the LSFs should be nearly sorted, use insertion sort */ for (i = 1; i < order; i++) { int j, value = nlsf[i]; for (j = i - 1; j >= 0 && nlsf[j] > value; j--) nlsf[j + 1] = nlsf[j]; nlsf[j + 1] = value; } /* push forwards to increase distance */ if (nlsf[0] < min_delta[0]) nlsf[0] = min_delta[0]; for (i = 1; i < order; i++) nlsf[i] = FFMAX(nlsf[i], FFMIN(nlsf[i - 1] + min_delta[i], 32767)); /* push backwards to increase distance */ if (nlsf[order-1] > 32768 - min_delta[order]) nlsf[order-1] = 32768 - min_delta[order]; for (i = order-2; i >= 0; i--) if (nlsf[i] > nlsf[i + 1] - min_delta[i+1]) nlsf[i] = nlsf[i + 1] - min_delta[i+1]; return; } static inline int silk_is_lpc_stable(const int16_t lpc[16], int order) { int k, j, DC_resp = 0; int32_t lpc32[2][16]; // Q24 int totalinvgain = 1 << 30; // 1.0 in Q30 int32_t *row = lpc32[0], *prevrow; /* initialize the first row for the Levinson recursion */ for (k = 0; k < order; k++) { DC_resp += lpc[k]; row[k] = lpc[k] * 4096; } if (DC_resp >= 4096) return 0; /* check if prediction gain pushes any coefficients too far */ for (k = order - 1; 1; k--) { int rc; // Q31; reflection coefficient int gaindiv; // Q30; inverse of the gain (the divisor) int gain; // gain for this reflection coefficient int fbits; // fractional bits used for the gain int error; // Q29; estimate of the error of our partial estimate of 1/gaindiv if (FFABS(row[k]) > 16773022) return 0; rc = -(row[k] * 128); gaindiv = (1 << 30) - MULH(rc, rc); totalinvgain = MULH(totalinvgain, gaindiv) << 2; if (k == 0) return (totalinvgain >= 107374); /* approximate 1.0/gaindiv */ fbits = opus_ilog(gaindiv); gain = ((1 << 29) - 1) / (gaindiv >> (fbits + 1 - 16)); // Q<fbits-16> error = (1 << 29) - MULL(gaindiv << (15 + 16 - fbits), gain, 16); gain = ((gain << 16) + (error * gain >> 13)); /* switch to the next row of the LPC coefficients */ prevrow = row; row = lpc32[k & 1]; for (j = 0; j < k; j++) { int x = av_sat_sub32(prevrow[j], ROUND_MULL(prevrow[k - j - 1], rc, 31)); int64_t tmp = ROUND_MULL(x, gain, fbits); /* per RFC 8251 section 6, if this calculation overflows, the filter is considered unstable. */ if (tmp < INT32_MIN || tmp > INT32_MAX) return 0; row[j] = (int32_t)tmp; } } } static void silk_lsp2poly(const int32_t lsp[/* 2 * half_order - 1 */], int32_t pol[/* half_order + 1 */], int half_order) { int i, j; pol[0] = 65536; // 1.0 in Q16 pol[1] = -lsp[0]; for (i = 1; i < half_order; i++) { pol[i + 1] = pol[i - 1] * 2 - ROUND_MULL(lsp[2 * i], pol[i], 16); for (j = i; j > 1; j--) pol[j] += pol[j - 2] - ROUND_MULL(lsp[2 * i], pol[j - 1], 16); pol[1] -= lsp[2 * i]; } } static void silk_lsf2lpc(const int16_t nlsf[16], float lpcf[16], int order) { int i, k; int32_t lsp[16]; // Q17; 2*cos(LSF) int32_t p[9], q[9]; // Q16 int32_t lpc32[16]; // Q17 int16_t lpc[16]; // Q12 /* convert the LSFs to LSPs, i.e. 2*cos(LSF) */ for (k = 0; k < order; k++) { int index = nlsf[k] >> 8; int offset = nlsf[k] & 255; int k2 = (order == 10) ? ff_silk_lsf_ordering_nbmb[k] : ff_silk_lsf_ordering_wb[k]; /* interpolate and round */ lsp[k2] = ff_silk_cosine[index] * 256; lsp[k2] += (ff_silk_cosine[index + 1] - ff_silk_cosine[index]) * offset; lsp[k2] = (lsp[k2] + 4) >> 3; } silk_lsp2poly(lsp , p, order >> 1); silk_lsp2poly(lsp + 1, q, order >> 1); /* reconstruct A(z) */ for (k = 0; k < order>>1; k++) { int32_t p_tmp = p[k + 1] + p[k]; int32_t q_tmp = q[k + 1] - q[k]; lpc32[k] = -q_tmp - p_tmp; lpc32[order-k-1] = q_tmp - p_tmp; } /* limit the range of the LPC coefficients to each fit within an int16_t */ for (i = 0; i < 10; i++) { int j; unsigned int maxabs = 0; for (j = 0, k = 0; j < order; j++) { unsigned int x = FFABS(lpc32[k]); if (x > maxabs) { maxabs = x; // Q17 k = j; } } maxabs = (maxabs + 16) >> 5; // convert to Q12 if (maxabs > 32767) { /* perform bandwidth expansion */ unsigned int chirp, chirp_base; // Q16 maxabs = FFMIN(maxabs, 163838); // anything above this overflows chirp's numerator chirp_base = chirp = 65470 - ((maxabs - 32767) << 14) / ((maxabs * (k+1)) >> 2); for (k = 0; k < order; k++) { lpc32[k] = ROUND_MULL(lpc32[k], chirp, 16); chirp = (chirp_base * chirp + 32768) >> 16; } } else break; } if (i == 10) { /* time's up: just clamp */ for (k = 0; k < order; k++) { int x = (lpc32[k] + 16) >> 5; lpc[k] = av_clip_int16(x); lpc32[k] = lpc[k] << 5; // shortcut mandated by the spec; drops lower 5 bits } } else { for (k = 0; k < order; k++) lpc[k] = (lpc32[k] + 16) >> 5; } /* if the prediction gain causes the LPC filter to become unstable, apply further bandwidth expansion on the Q17 coefficients */ for (i = 1; i <= 16 && !silk_is_lpc_stable(lpc, order); i++) { unsigned int chirp, chirp_base; chirp_base = chirp = 65536 - (1 << i); for (k = 0; k < order; k++) { lpc32[k] = ROUND_MULL(lpc32[k], chirp, 16); lpc[k] = (lpc32[k] + 16) >> 5; chirp = (chirp_base * chirp + 32768) >> 16; } } for (i = 0; i < order; i++) lpcf[i] = lpc[i] / 4096.0f; } static inline void silk_decode_lpc(SilkContext *s, SilkFrame *frame, OpusRangeCoder *rc, float lpc_leadin[16], float lpc[16], int *lpc_order, int *has_lpc_leadin, int voiced) { int i; int order; // order of the LP polynomial; 10 for NB/MB and 16 for WB int8_t lsf_i1, lsf_i2[16]; // stage-1 and stage-2 codebook indices int16_t lsf_res[16]; // residual as a Q10 value int16_t nlsf[16]; // Q15 *lpc_order = order = s->wb ? 16 : 10; /* obtain LSF stage-1 and stage-2 indices */ lsf_i1 = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s1[s->wb][voiced]); for (i = 0; i < order; i++) { int index = s->wb ? ff_silk_lsf_s2_model_sel_wb [lsf_i1][i] : ff_silk_lsf_s2_model_sel_nbmb[lsf_i1][i]; lsf_i2[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2[index]) - 4; if (lsf_i2[i] == -4) lsf_i2[i] -= ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2_ext); else if (lsf_i2[i] == 4) lsf_i2[i] += ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2_ext); } /* reverse the backwards-prediction step */ for (i = order - 1; i >= 0; i--) { int qstep = s->wb ? 9830 : 11796; lsf_res[i] = lsf_i2[i] * 1024; if (lsf_i2[i] < 0) lsf_res[i] += 102; else if (lsf_i2[i] > 0) lsf_res[i] -= 102; lsf_res[i] = (lsf_res[i] * qstep) >> 16; if (i + 1 < order) { int weight = s->wb ? ff_silk_lsf_pred_weights_wb [ff_silk_lsf_weight_sel_wb [lsf_i1][i]][i] : ff_silk_lsf_pred_weights_nbmb[ff_silk_lsf_weight_sel_nbmb[lsf_i1][i]][i]; lsf_res[i] += (lsf_res[i+1] * weight) >> 8; } } /* reconstruct the NLSF coefficients from the supplied indices */ for (i = 0; i < order; i++) { const uint8_t * codebook = s->wb ? ff_silk_lsf_codebook_wb [lsf_i1] : ff_silk_lsf_codebook_nbmb[lsf_i1]; int cur, prev, next, weight_sq, weight, ipart, fpart, y, value; /* find the weight of the residual */ /* TODO: precompute */ cur = codebook[i]; prev = i ? codebook[i - 1] : 0; next = i + 1 < order ? codebook[i + 1] : 256; weight_sq = (1024 / (cur - prev) + 1024 / (next - cur)) << 16; /* approximate square-root with mandated fixed-point arithmetic */ ipart = opus_ilog(weight_sq); fpart = (weight_sq >> (ipart-8)) & 127; y = ((ipart & 1) ? 32768 : 46214) >> ((32 - ipart)>>1); weight = y + ((213 * fpart * y) >> 16); value = cur * 128 + (lsf_res[i] * 16384) / weight; nlsf[i] = av_clip_uintp2(value, 15); } /* stabilize the NLSF coefficients */ silk_stabilize_lsf(nlsf, order, s->wb ? ff_silk_lsf_min_spacing_wb : ff_silk_lsf_min_spacing_nbmb); /* produce an interpolation for the first 2 subframes, */ /* and then convert both sets of NLSFs to LPC coefficients */ *has_lpc_leadin = 0; if (s->subframes == 4) { int offset = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_interpolation_offset); if (offset != 4 && frame->coded) { *has_lpc_leadin = 1; if (offset != 0) { int16_t nlsf_leadin[16]; for (i = 0; i < order; i++) nlsf_leadin[i] = frame->nlsf[i] + ((nlsf[i] - frame->nlsf[i]) * offset >> 2); silk_lsf2lpc(nlsf_leadin, lpc_leadin, order); } else /* avoid re-computation for a (roughly) 1-in-4 occurrence */ memcpy(lpc_leadin, frame->lpc, 16 * sizeof(float)); } else offset = 4; s->nlsf_interp_factor = offset; silk_lsf2lpc(nlsf, lpc, order); } else { s->nlsf_interp_factor = 4; silk_lsf2lpc(nlsf, lpc, order); } memcpy(frame->nlsf, nlsf, order * sizeof(nlsf[0])); memcpy(frame->lpc, lpc, order * sizeof(lpc[0])); } static inline void silk_count_children(OpusRangeCoder *rc, int model, int32_t total, int32_t child[2]) { if (total != 0) { child[0] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_location[model] + (((total - 1 + 5) * (total - 1)) >> 1)); child[1] = total - child[0]; } else { child[0] = 0; child[1] = 0; } } static inline void silk_decode_excitation(SilkContext *s, OpusRangeCoder *rc, float* excitationf, int qoffset_high, int active, int voiced) { int i; uint32_t seed; int shellblocks; int ratelevel; uint8_t pulsecount[20]; // total pulses in each shell block uint8_t lsbcount[20] = {0}; // raw lsbits defined for each pulse in each shell block int32_t excitation[320]; // Q23 /* excitation parameters */ seed = ff_opus_rc_dec_cdf(rc, ff_silk_model_lcg_seed); shellblocks = ff_silk_shell_blocks[s->bandwidth][s->subframes >> 2]; ratelevel = ff_opus_rc_dec_cdf(rc, ff_silk_model_exc_rate[voiced]); for (i = 0; i < shellblocks; i++) { pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[ratelevel]); if (pulsecount[i] == 17) { while (pulsecount[i] == 17 && ++lsbcount[i] != 10) pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[9]); if (lsbcount[i] == 10) pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[10]); } } /* decode pulse locations using PVQ */ for (i = 0; i < shellblocks; i++) { if (pulsecount[i] != 0) { int a, b, c, d; int32_t * location = excitation + 16*i; int32_t branch[4][2]; branch[0][0] = pulsecount[i]; /* unrolled tail recursion */ for (a = 0; a < 1; a++) { silk_count_children(rc, 0, branch[0][a], branch[1]); for (b = 0; b < 2; b++) { silk_count_children(rc, 1, branch[1][b], branch[2]); for (c = 0; c < 2; c++) { silk_count_children(rc, 2, branch[2][c], branch[3]); for (d = 0; d < 2; d++) { silk_count_children(rc, 3, branch[3][d], location); location += 2; } } } } } else memset(excitation + 16*i, 0, 16*sizeof(int32_t)); } /* decode least significant bits */ for (i = 0; i < shellblocks << 4; i++) { int bit; for (bit = 0; bit < lsbcount[i >> 4]; bit++) excitation[i] = (excitation[i] << 1) | ff_opus_rc_dec_cdf(rc, ff_silk_model_excitation_lsb); } /* decode signs */ for (i = 0; i < shellblocks << 4; i++) { if (excitation[i] != 0) { int sign = ff_opus_rc_dec_cdf(rc, ff_silk_model_excitation_sign[active + voiced][qoffset_high][FFMIN(pulsecount[i >> 4], 6)]); if (sign == 0) excitation[i] *= -1; } } /* assemble the excitation */ for (i = 0; i < shellblocks << 4; i++) { int value = excitation[i]; excitation[i] = value * 256 | ff_silk_quant_offset[voiced][qoffset_high]; if (value < 0) excitation[i] += 20; else if (value > 0) excitation[i] -= 20; /* invert samples pseudorandomly */ seed = 196314165 * seed + 907633515; if (seed & 0x80000000) excitation[i] *= -1; seed += value; excitationf[i] = excitation[i] / 8388608.0f; } } /** Maximum residual history according to 4.2.7.6.1 */ #define SILK_MAX_LAG (288 + LTP_ORDER / 2) /** Order of the LTP filter */ #define LTP_ORDER 5 static void silk_decode_frame(SilkContext *s, OpusRangeCoder *rc, int frame_num, int channel, int coded_channels, int active, int active1, int redundant) { /* per frame */ int voiced; // combines with active to indicate inactive, active, or active+voiced int qoffset_high; int order; // order of the LPC coefficients float lpc_leadin[16], lpc_body[16], residual[SILK_MAX_LAG + SILK_HISTORY]; int has_lpc_leadin; float ltpscale; /* per subframe */ struct { float gain; int pitchlag; float ltptaps[5]; } sf[4]; SilkFrame * const frame = s->frame + channel; int i; /* obtain stereo weights */ if (coded_channels == 2 && channel == 0) { int n, wi[2], ws[2], w[2]; n = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s1); wi[0] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s2) + 3 * (n / 5); ws[0] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s3); wi[1] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s2) + 3 * (n % 5); ws[1] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s3); for (i = 0; i < 2; i++) w[i] = ff_silk_stereo_weights[wi[i]] + (((ff_silk_stereo_weights[wi[i] + 1] - ff_silk_stereo_weights[wi[i]]) * 6554) >> 16) * (ws[i]*2 + 1); s->stereo_weights[0] = (w[0] - w[1]) / 8192.0; s->stereo_weights[1] = w[1] / 8192.0; /* and read the mid-only flag */ s->midonly = active1 ? 0 : ff_opus_rc_dec_cdf(rc, ff_silk_model_mid_only); } /* obtain frame type */ if (!active) { qoffset_high = ff_opus_rc_dec_cdf(rc, ff_silk_model_frame_type_inactive); voiced = 0; } else { int type = ff_opus_rc_dec_cdf(rc, ff_silk_model_frame_type_active); qoffset_high = type & 1; voiced = type >> 1; } /* obtain subframe quantization gains */ for (i = 0; i < s->subframes; i++) { int log_gain; //Q7 int ipart, fpart, lingain; if (i == 0 && (frame_num == 0 || !frame->coded)) { /* gain is coded absolute */ int x = ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_highbits[active + voiced]); log_gain = (x<<3) | ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_lowbits); if (frame->coded) log_gain = FFMAX(log_gain, frame->log_gain - 16); } else { /* gain is coded relative */ int delta_gain = ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_delta); log_gain = av_clip_uintp2(FFMAX((delta_gain<<1) - 16, frame->log_gain + delta_gain - 4), 6); } frame->log_gain = log_gain; /* approximate 2**(x/128) with a Q7 (i.e. non-integer) input */ log_gain = (log_gain * 0x1D1C71 >> 16) + 2090; ipart = log_gain >> 7; fpart = log_gain & 127; lingain = (1 << ipart) + ((-174 * fpart * (128-fpart) >>16) + fpart) * ((1<<ipart) >> 7); sf[i].gain = lingain / 65536.0f; } /* obtain LPC filter coefficients */ silk_decode_lpc(s, frame, rc, lpc_leadin, lpc_body, &order, &has_lpc_leadin, voiced); /* obtain pitch lags, if this is a voiced frame */ if (voiced) { int lag_absolute = (!frame_num || !frame->prev_voiced); int primarylag; // primary pitch lag for the entire SILK frame int ltpfilter; const int8_t * offsets; if (!lag_absolute) { int delta = ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_delta); if (delta) primarylag = frame->primarylag + delta - 9; else lag_absolute = 1; } if (lag_absolute) { /* primary lag is coded absolute */ int highbits, lowbits; static const uint16_t * const model[] = { ff_silk_model_pitch_lowbits_nb, ff_silk_model_pitch_lowbits_mb, ff_silk_model_pitch_lowbits_wb }; highbits = ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_highbits); lowbits = ff_opus_rc_dec_cdf(rc, model[s->bandwidth]); primarylag = ff_silk_pitch_min_lag[s->bandwidth] + highbits*ff_silk_pitch_scale[s->bandwidth] + lowbits; } frame->primarylag = primarylag; if (s->subframes == 2) offsets = (s->bandwidth == OPUS_BANDWIDTH_NARROWBAND) ? ff_silk_pitch_offset_nb10ms[ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_contour_nb10ms)] : ff_silk_pitch_offset_mbwb10ms[ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_contour_mbwb10ms)]; else offsets = (s->bandwidth == OPUS_BANDWIDTH_NARROWBAND) ? ff_silk_pitch_offset_nb20ms[ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_contour_nb20ms)] : ff_silk_pitch_offset_mbwb20ms[ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_contour_mbwb20ms)]; for (i = 0; i < s->subframes; i++) sf[i].pitchlag = av_clip(primarylag + offsets[i], ff_silk_pitch_min_lag[s->bandwidth], ff_silk_pitch_max_lag[s->bandwidth]); /* obtain LTP filter coefficients */ ltpfilter = ff_opus_rc_dec_cdf(rc, ff_silk_model_ltp_filter); for (i = 0; i < s->subframes; i++) { int index, j; static const uint16_t * const filter_sel[] = { ff_silk_model_ltp_filter0_sel, ff_silk_model_ltp_filter1_sel, ff_silk_model_ltp_filter2_sel }; static const int8_t (* const filter_taps[])[5] = { ff_silk_ltp_filter0_taps, ff_silk_ltp_filter1_taps, ff_silk_ltp_filter2_taps }; index = ff_opus_rc_dec_cdf(rc, filter_sel[ltpfilter]); for (j = 0; j < 5; j++) sf[i].ltptaps[j] = filter_taps[ltpfilter][index][j] / 128.0f; } } /* obtain LTP scale factor */ if (voiced && frame_num == 0) ltpscale = ff_silk_ltp_scale_factor[ff_opus_rc_dec_cdf(rc, ff_silk_model_ltp_scale_index)] / 16384.0f; else ltpscale = 15565.0f/16384.0f; /* generate the excitation signal for the entire frame */ silk_decode_excitation(s, rc, residual + SILK_MAX_LAG, qoffset_high, active, voiced); /* skip synthesising the output if we do not need it */ // TODO: implement error recovery if (s->output_channels == channel || redundant) return; /* generate the output signal */ for (i = 0; i < s->subframes; i++) { const float * lpc_coeff = (i < 2 && has_lpc_leadin) ? lpc_leadin : lpc_body; float *dst = frame->output + SILK_HISTORY + i * s->sflength; float *resptr = residual + SILK_MAX_LAG + i * s->sflength; float *lpc = frame->lpc_history + SILK_HISTORY + i * s->sflength; float sum; int j, k; if (voiced) { int out_end; float scale; if (i < 2 || s->nlsf_interp_factor == 4) { out_end = -i * s->sflength; scale = ltpscale; } else { out_end = -(i - 2) * s->sflength; scale = 1.0f; } /* when the LPC coefficients change, a re-whitening filter is used */ /* to produce a residual that accounts for the change */ for (j = - sf[i].pitchlag - LTP_ORDER/2; j < out_end; j++) { sum = dst[j]; for (k = 0; k < order; k++) sum -= lpc_coeff[k] * dst[j - k - 1]; resptr[j] = av_clipf(sum, -1.0f, 1.0f) * scale / sf[i].gain; } if (out_end) { float rescale = sf[i-1].gain / sf[i].gain; for (j = out_end; j < 0; j++) resptr[j] *= rescale; } /* LTP synthesis */ for (j = 0; j < s->sflength; j++) { sum = resptr[j]; for (k = 0; k < LTP_ORDER; k++) sum += sf[i].ltptaps[k] * resptr[j - sf[i].pitchlag + LTP_ORDER/2 - k]; resptr[j] = sum; } } /* LPC synthesis */ for (j = 0; j < s->sflength; j++) { sum = resptr[j] * sf[i].gain; for (k = 1; k <= order; k++) sum += lpc_coeff[k - 1] * lpc[j - k]; lpc[j] = sum; dst[j] = av_clipf(sum, -1.0f, 1.0f); } } frame->prev_voiced = voiced; memmove(frame->lpc_history, frame->lpc_history + s->flength, SILK_HISTORY * sizeof(float)); memmove(frame->output, frame->output + s->flength, SILK_HISTORY * sizeof(float)); frame->coded = 1; } static void silk_unmix_ms(SilkContext *s, float *l, float *r) { float *mid = s->frame[0].output + SILK_HISTORY - s->flength; float *side = s->frame[1].output + SILK_HISTORY - s->flength; float w0_prev = s->prev_stereo_weights[0]; float w1_prev = s->prev_stereo_weights[1]; float w0 = s->stereo_weights[0]; float w1 = s->stereo_weights[1]; int n1 = ff_silk_stereo_interp_len[s->bandwidth]; int i; for (i = 0; i < n1; i++) { float interp0 = w0_prev + i * (w0 - w0_prev) / n1; float interp1 = w1_prev + i * (w1 - w1_prev) / n1; float p0 = 0.25 * (mid[i - 2] + 2 * mid[i - 1] + mid[i]); l[i] = av_clipf((1 + interp1) * mid[i - 1] + side[i - 1] + interp0 * p0, -1.0, 1.0); r[i] = av_clipf((1 - interp1) * mid[i - 1] - side[i - 1] - interp0 * p0, -1.0, 1.0); } for (; i < s->flength; i++) { float p0 = 0.25 * (mid[i - 2] + 2 * mid[i - 1] + mid[i]); l[i] = av_clipf((1 + w1) * mid[i - 1] + side[i - 1] + w0 * p0, -1.0, 1.0); r[i] = av_clipf((1 - w1) * mid[i - 1] - side[i - 1] - w0 * p0, -1.0, 1.0); } memcpy(s->prev_stereo_weights, s->stereo_weights, sizeof(s->stereo_weights)); } static void silk_flush_frame(SilkFrame *frame) { if (!frame->coded) return; memset(frame->output, 0, sizeof(frame->output)); memset(frame->lpc_history, 0, sizeof(frame->lpc_history)); memset(frame->lpc, 0, sizeof(frame->lpc)); memset(frame->nlsf, 0, sizeof(frame->nlsf)); frame->log_gain = 0; frame->primarylag = 0; frame->prev_voiced = 0; frame->coded = 0; } int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc, float *output[2], enum OpusBandwidth bandwidth, int coded_channels, int duration_ms) { int active[2][6], redundancy[2]; int nb_frames, i, j; if (bandwidth > OPUS_BANDWIDTH_WIDEBAND || coded_channels > 2 || duration_ms > 60) { av_log(s->avctx, AV_LOG_ERROR, "Invalid parameters passed " "to the SILK decoder.\n"); return AVERROR(EINVAL); } nb_frames = 1 + (duration_ms > 20) + (duration_ms > 40); s->subframes = duration_ms / nb_frames / 5; // 5ms subframes s->sflength = 20 * (bandwidth + 2); s->flength = s->sflength * s->subframes; s->bandwidth = bandwidth; s->wb = bandwidth == OPUS_BANDWIDTH_WIDEBAND; /* make sure to flush the side channel when switching from mono to stereo */ if (coded_channels > s->prev_coded_channels) silk_flush_frame(&s->frame[1]); s->prev_coded_channels = coded_channels; /* read the LP-layer header bits */ for (i = 0; i < coded_channels; i++) { for (j = 0; j < nb_frames; j++) active[i][j] = ff_opus_rc_dec_log(rc, 1); redundancy[i] = ff_opus_rc_dec_log(rc, 1); } /* read the per-frame LBRR flags */ for (i = 0; i < coded_channels; i++) if (redundancy[i] && duration_ms > 20) { redundancy[i] = ff_opus_rc_dec_cdf(rc, duration_ms == 40 ? ff_silk_model_lbrr_flags_40 : ff_silk_model_lbrr_flags_60); } /* decode the LBRR frames */ for (i = 0; i < nb_frames; i++) { for (j = 0; j < coded_channels; j++) if (redundancy[j] & (1 << i)) { int active1 = (j == 0 && !(redundancy[1] & (1 << i))) ? 0 : 1; silk_decode_frame(s, rc, i, j, coded_channels, 1, active1, 1); } } for (i = 0; i < nb_frames; i++) { for (j = 0; j < coded_channels && !s->midonly; j++) silk_decode_frame(s, rc, i, j, coded_channels, active[j][i], active[1][i], 0); /* reset the side channel if it is not coded */ if (s->midonly && s->frame[1].coded) silk_flush_frame(&s->frame[1]); if (coded_channels == 1 || s->output_channels == 1) { for (j = 0; j < s->output_channels; j++) { memcpy(output[j] + i * s->flength, s->frame[0].output + SILK_HISTORY - s->flength - 2, s->flength * sizeof(float)); } } else { silk_unmix_ms(s, output[0] + i * s->flength, output[1] + i * s->flength); } s->midonly = 0; } return nb_frames * s->flength; } void ff_silk_free(SilkContext **ps) { av_freep(ps); } void ff_silk_flush(SilkContext *s) { silk_flush_frame(&s->frame[0]); silk_flush_frame(&s->frame[1]); memset(s->prev_stereo_weights, 0, sizeof(s->prev_stereo_weights)); } int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels) { SilkContext *s; if (output_channels != 1 && output_channels != 2) { av_log(avctx, AV_LOG_ERROR, "Invalid number of output channels: %d\n", output_channels); return AVERROR(EINVAL); } s = av_mallocz(sizeof(*s)); if (!s) return AVERROR(ENOMEM); s->avctx = avctx; s->output_channels = output_channels; ff_silk_flush(s); *ps = s; return 0; }