| /* |
| * 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[16], int32_t pol[16], 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) |
| { |
| /* 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 side channel if we want mono-only */ |
| if (s->output_channels == channel) |
| 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); |
| if (redundancy[i]) { |
| avpriv_report_missing_feature(s->avctx, "LBRR frames"); |
| return AVERROR_PATCHWELCOME; |
| } |
| } |
| |
| 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]); |
| |
| /* 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; |
| } |
