| /* | |
| * 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 "mathops.h" | |
| #include "opus.h" | |
| #include "opus_rc.h" | |
| #include "opus_silk.h" | |
| #include "opustab.h" | |
| #define ROUND_MULL(a,b,s) (((MUL64(a, b) >> ((s) - 1)) + 1) >> 1) | |
| 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 { | |
| void *logctx; | |
| 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->logctx, 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); | |
| } | |
| s->midonly = 0; | |
| } | |
| 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(void *logctx, SilkContext **ps, int output_channels) | |
| { | |
| SilkContext *s; | |
| if (output_channels != 1 && output_channels != 2) { | |
| av_log(logctx, 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->logctx = logctx; | |
| s->output_channels = output_channels; | |
| ff_silk_flush(s); | |
| *ps = s; | |
| return 0; | |
| } |