diff --git a/CHANGELOG.md b/CHANGELOG.md
index c7db753a0..fe7bf4889 100644
--- a/CHANGELOG.md
+++ b/CHANGELOG.md
@@ -806,7 +806,7 @@ Winlin 2021
[pr #558]: https://github.com/ossrs/srs/pull/558
[pr #559]: https://github.com/ossrs/srs/pull/559
-[authors]: https://github.com/ossrs/srs/blob/develop/AUTHORS.txt
+[authors]: https://github.com/ossrs/srs/blob/4.0release/trunk/AUTHORS.txt
[bigthanks]: https://github.com/ossrs/srs/wiki/v4_CN_Product#bigthanks
[st]: https://github.com/winlinvip/state-threads
[st2]: https://github.com/ossrs/state-threads/tree/srs
diff --git a/PERFORMANCE.md b/PERFORMANCE.md
index 303f33f16..ad4749ab7 100644
--- a/PERFORMANCE.md
+++ b/PERFORMANCE.md
@@ -143,7 +143,7 @@ Winlin 2021
[p22]: https://github.com/ossrs/srs/commit/5a4373d4835758188b9a1f03005cea0b6ddc62aa
[p23]: https://github.com/ossrs/srs/pull/239
-[authors]: https://github.com/ossrs/srs/blob/4.0release/AUTHORS.txt
+[authors]: https://github.com/ossrs/srs/blob/4.0release/trunk/AUTHORS.txt
[bigthanks]: https://github.com/ossrs/srs/wiki/Product#release40
[st]: https://github.com/ossrs/state-threads
[st2]: https://github.com/ossrs/state-threads/tree/srs
diff --git a/README.md b/README.md
index 71c18e9e9..8eab592eb 100755
--- a/README.md
+++ b/README.md
@@ -379,7 +379,7 @@ Supported operating systems and hardware:
Beijing, 2013.10
Winlin
-[authors]: https://github.com/ossrs/srs/blob/4.0release/AUTHORS.txt
+[authors]: https://github.com/ossrs/srs/blob/4.0release/trunk/AUTHORS.txt
[bigthanks]: https://github.com/ossrs/srs/wiki/Product#release40
[st]: https://github.com/ossrs/state-threads
[st2]: https://github.com/ossrs/state-threads/tree/srs
diff --git a/trunk/3rdparty/ffmpeg-4-fit/libavcodec/opus_silk.c b/trunk/3rdparty/ffmpeg-4-fit/libavcodec/opus_silk.c
new file mode 100644
index 000000000..2fcbf3b9d
--- /dev/null
+++ b/trunk/3rdparty/ffmpeg-4-fit/libavcodec/opus_silk.c
@@ -0,0 +1,882 @@
+/*
+ * 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
+
+#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
+ 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<> 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;
+}
diff --git a/trunk/3rdparty/ffmpeg-4-fit/libavcodec/opusdec.c b/trunk/3rdparty/ffmpeg-4-fit/libavcodec/opusdec.c
new file mode 100644
index 000000000..03086dea9
--- /dev/null
+++ b/trunk/3rdparty/ffmpeg-4-fit/libavcodec/opusdec.c
@@ -0,0 +1,741 @@
+/*
+ * Opus decoder
+ * 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 decoder
+ * @author Andrew D'Addesio, Anton Khirnov
+ *
+ * Codec homepage: http://opus-codec.org/
+ * Specification: http://tools.ietf.org/html/rfc6716
+ * Ogg Opus specification: https://tools.ietf.org/html/draft-ietf-codec-oggopus-03
+ *
+ * Ogg-contained .opus files can be produced with opus-tools:
+ * http://git.xiph.org/?p=opus-tools.git
+ */
+
+#include
+
+#include "libavutil/attributes.h"
+#include "libavutil/audio_fifo.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+
+#include "libswresample/swresample.h"
+
+#include "avcodec.h"
+#include "get_bits.h"
+#include "internal.h"
+#include "mathops.h"
+#include "opus.h"
+#include "opustab.h"
+#include "opus_celt.h"
+
+static const uint16_t silk_frame_duration_ms[16] = {
+ 10, 20, 40, 60,
+ 10, 20, 40, 60,
+ 10, 20, 40, 60,
+ 10, 20,
+ 10, 20,
+};
+
+/* number of samples of silence to feed to the resampler
+ * at the beginning */
+static const int silk_resample_delay[] = {
+ 4, 8, 11, 11, 11
+};
+
+static int get_silk_samplerate(int config)
+{
+ if (config < 4)
+ return 8000;
+ else if (config < 8)
+ return 12000;
+ return 16000;
+}
+
+static void opus_fade(float *out,
+ const float *in1, const float *in2,
+ const float *window, int len)
+{
+ int i;
+ for (i = 0; i < len; i++)
+ out[i] = in2[i] * window[i] + in1[i] * (1.0 - window[i]);
+}
+
+static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
+{
+ int celt_size = av_audio_fifo_size(s->celt_delay);
+ int ret, i;
+ ret = swr_convert(s->swr,
+ (uint8_t**)s->out, nb_samples,
+ NULL, 0);
+ if (ret < 0)
+ return ret;
+ else if (ret != nb_samples) {
+ av_log(s->avctx, AV_LOG_ERROR, "Wrong number of flushed samples: %d\n",
+ ret);
+ return AVERROR_BUG;
+ }
+
+ if (celt_size) {
+ if (celt_size != nb_samples) {
+ av_log(s->avctx, AV_LOG_ERROR, "Wrong number of CELT delay samples.\n");
+ return AVERROR_BUG;
+ }
+ av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, nb_samples);
+ for (i = 0; i < s->output_channels; i++) {
+ s->fdsp->vector_fmac_scalar(s->out[i],
+ s->celt_output[i], 1.0,
+ nb_samples);
+ }
+ }
+
+ if (s->redundancy_idx) {
+ for (i = 0; i < s->output_channels; i++)
+ opus_fade(s->out[i], s->out[i],
+ s->redundancy_output[i] + 120 + s->redundancy_idx,
+ ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
+ s->redundancy_idx = 0;
+ }
+
+ s->out[0] += nb_samples;
+ s->out[1] += nb_samples;
+ s->out_size -= nb_samples * sizeof(float);
+
+ return 0;
+}
+
+static int opus_init_resample(OpusStreamContext *s)
+{
+ static const float delay[16] = { 0.0 };
+ const uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };
+ int ret;
+
+ av_opt_set_int(s->swr, "in_sample_rate", s->silk_samplerate, 0);
+ ret = swr_init(s->swr);
+ if (ret < 0) {
+ av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n");
+ return ret;
+ }
+
+ ret = swr_convert(s->swr,
+ NULL, 0,
+ delayptr, silk_resample_delay[s->packet.bandwidth]);
+ if (ret < 0) {
+ av_log(s->avctx, AV_LOG_ERROR,
+ "Error feeding initial silence to the resampler.\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static int opus_decode_redundancy(OpusStreamContext *s, const uint8_t *data, int size)
+{
+ int ret = ff_opus_rc_dec_init(&s->redundancy_rc, data, size);
+ if (ret < 0)
+ goto fail;
+ ff_opus_rc_dec_raw_init(&s->redundancy_rc, data + size, size);
+
+ ret = ff_celt_decode_frame(s->celt, &s->redundancy_rc,
+ s->redundancy_output,
+ s->packet.stereo + 1, 240,
+ 0, ff_celt_band_end[s->packet.bandwidth]);
+ if (ret < 0)
+ goto fail;
+
+ return 0;
+fail:
+ av_log(s->avctx, AV_LOG_ERROR, "Error decoding the redundancy frame.\n");
+ return ret;
+}
+
+static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)
+{
+ int samples = s->packet.frame_duration;
+ int redundancy = 0;
+ int redundancy_size, redundancy_pos;
+ int ret, i, consumed;
+ int delayed_samples = s->delayed_samples;
+
+ ret = ff_opus_rc_dec_init(&s->rc, data, size);
+ if (ret < 0)
+ return ret;
+
+ /* decode the silk frame */
+ if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
+ if (!swr_is_initialized(s->swr)) {
+ ret = opus_init_resample(s);
+ if (ret < 0)
+ return ret;
+ }
+
+ samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output,
+ FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND),
+ s->packet.stereo + 1,
+ silk_frame_duration_ms[s->packet.config]);
+ if (samples < 0) {
+ av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
+ return samples;
+ }
+ samples = swr_convert(s->swr,
+ (uint8_t**)s->out, s->packet.frame_duration,
+ (const uint8_t**)s->silk_output, samples);
+ if (samples < 0) {
+ av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
+ return samples;
+ }
+ av_assert2((samples & 7) == 0);
+ s->delayed_samples += s->packet.frame_duration - samples;
+ } else
+ ff_silk_flush(s->silk);
+
+ // decode redundancy information
+ consumed = opus_rc_tell(&s->rc);
+ if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8)
+ redundancy = ff_opus_rc_dec_log(&s->rc, 12);
+ else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)
+ redundancy = 1;
+
+ if (redundancy) {
+ redundancy_pos = ff_opus_rc_dec_log(&s->rc, 1);
+
+ if (s->packet.mode == OPUS_MODE_HYBRID)
+ redundancy_size = ff_opus_rc_dec_uint(&s->rc, 256) + 2;
+ else
+ redundancy_size = size - (consumed + 7) / 8;
+ size -= redundancy_size;
+ if (size < 0) {
+ av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ if (redundancy_pos) {
+ ret = opus_decode_redundancy(s, data + size, redundancy_size);
+ if (ret < 0)
+ return ret;
+ ff_celt_flush(s->celt);
+ }
+ }
+
+ /* decode the CELT frame */
+ if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {
+ float *out_tmp[2] = { s->out[0], s->out[1] };
+ float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
+ out_tmp : s->celt_output;
+ int celt_output_samples = samples;
+ int delay_samples = av_audio_fifo_size(s->celt_delay);
+
+ if (delay_samples) {
+ if (s->packet.mode == OPUS_MODE_HYBRID) {
+ av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples);
+
+ for (i = 0; i < s->output_channels; i++) {
+ s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0,
+ delay_samples);
+ out_tmp[i] += delay_samples;
+ }
+ celt_output_samples -= delay_samples;
+ } else {
+ av_log(s->avctx, AV_LOG_WARNING,
+ "Spurious CELT delay samples present.\n");
+ av_audio_fifo_drain(s->celt_delay, delay_samples);
+ if (s->avctx->err_recognition & AV_EF_EXPLODE)
+ return AVERROR_BUG;
+ }
+ }
+
+ ff_opus_rc_dec_raw_init(&s->rc, data + size, size);
+
+ ret = ff_celt_decode_frame(s->celt, &s->rc, dst,
+ s->packet.stereo + 1,
+ s->packet.frame_duration,
+ (s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,
+ ff_celt_band_end[s->packet.bandwidth]);
+ if (ret < 0)
+ return ret;
+
+ if (s->packet.mode == OPUS_MODE_HYBRID) {
+ int celt_delay = s->packet.frame_duration - celt_output_samples;
+ void *delaybuf[2] = { s->celt_output[0] + celt_output_samples,
+ s->celt_output[1] + celt_output_samples };
+
+ for (i = 0; i < s->output_channels; i++) {
+ s->fdsp->vector_fmac_scalar(out_tmp[i],
+ s->celt_output[i], 1.0,
+ celt_output_samples);
+ }
+
+ ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay);
+ if (ret < 0)
+ return ret;
+ }
+ } else
+ ff_celt_flush(s->celt);
+
+ if (s->redundancy_idx) {
+ for (i = 0; i < s->output_channels; i++)
+ opus_fade(s->out[i], s->out[i],
+ s->redundancy_output[i] + 120 + s->redundancy_idx,
+ ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
+ s->redundancy_idx = 0;
+ }
+ if (redundancy) {
+ if (!redundancy_pos) {
+ ff_celt_flush(s->celt);
+ ret = opus_decode_redundancy(s, data + size, redundancy_size);
+ if (ret < 0)
+ return ret;
+
+ for (i = 0; i < s->output_channels; i++) {
+ opus_fade(s->out[i] + samples - 120 + delayed_samples,
+ s->out[i] + samples - 120 + delayed_samples,
+ s->redundancy_output[i] + 120,
+ ff_celt_window2, 120 - delayed_samples);
+ if (delayed_samples)
+ s->redundancy_idx = 120 - delayed_samples;
+ }
+ } else {
+ for (i = 0; i < s->output_channels; i++) {
+ memcpy(s->out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
+ opus_fade(s->out[i] + 120 + delayed_samples,
+ s->redundancy_output[i] + 120,
+ s->out[i] + 120 + delayed_samples,
+ ff_celt_window2, 120);
+ }
+ }
+ }
+
+ return samples;
+}
+
+static int opus_decode_subpacket(OpusStreamContext *s,
+ const uint8_t *buf, int buf_size,
+ float **out, int out_size,
+ int nb_samples)
+{
+ int output_samples = 0;
+ int flush_needed = 0;
+ int i, j, ret;
+
+ s->out[0] = out[0];
+ s->out[1] = out[1];
+ s->out_size = out_size;
+
+ /* check if we need to flush the resampler */
+ if (swr_is_initialized(s->swr)) {
+ if (buf) {
+ int64_t cur_samplerate;
+ av_opt_get_int(s->swr, "in_sample_rate", 0, &cur_samplerate);
+ flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
+ } else {
+ flush_needed = !!s->delayed_samples;
+ }
+ }
+
+ if (!buf && !flush_needed)
+ return 0;
+
+ /* use dummy output buffers if the channel is not mapped to anything */
+ if (!s->out[0] ||
+ (s->output_channels == 2 && !s->out[1])) {
+ av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size, s->out_size);
+ if (!s->out_dummy)
+ return AVERROR(ENOMEM);
+ if (!s->out[0])
+ s->out[0] = s->out_dummy;
+ if (!s->out[1])
+ s->out[1] = s->out_dummy;
+ }
+
+ /* flush the resampler if necessary */
+ if (flush_needed) {
+ ret = opus_flush_resample(s, s->delayed_samples);
+ if (ret < 0) {
+ av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");
+ return ret;
+ }
+ swr_close(s->swr);
+ output_samples += s->delayed_samples;
+ s->delayed_samples = 0;
+
+ if (!buf)
+ goto finish;
+ }
+
+ /* decode all the frames in the packet */
+ for (i = 0; i < s->packet.frame_count; i++) {
+ int size = s->packet.frame_size[i];
+ int samples = opus_decode_frame(s, buf + s->packet.frame_offset[i], size);
+
+ if (samples < 0) {
+ av_log(s->avctx, AV_LOG_ERROR, "Error decoding an Opus frame.\n");
+ if (s->avctx->err_recognition & AV_EF_EXPLODE)
+ return samples;
+
+ for (j = 0; j < s->output_channels; j++)
+ memset(s->out[j], 0, s->packet.frame_duration * sizeof(float));
+ samples = s->packet.frame_duration;
+ }
+ output_samples += samples;
+
+ for (j = 0; j < s->output_channels; j++)
+ s->out[j] += samples;
+ s->out_size -= samples * sizeof(float);
+ }
+
+finish:
+ s->out[0] = s->out[1] = NULL;
+ s->out_size = 0;
+
+ return output_samples;
+}
+
+static int opus_decode_packet(AVCodecContext *avctx, void *data,
+ int *got_frame_ptr, AVPacket *avpkt)
+{
+ OpusContext *c = avctx->priv_data;
+ AVFrame *frame = data;
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
+ int coded_samples = 0;
+ int decoded_samples = INT_MAX;
+ int delayed_samples = 0;
+ int i, ret;
+
+ /* calculate the number of delayed samples */
+ for (i = 0; i < c->nb_streams; i++) {
+ OpusStreamContext *s = &c->streams[i];
+ s->out[0] =
+ s->out[1] = NULL;
+ delayed_samples = FFMAX(delayed_samples,
+ s->delayed_samples + av_audio_fifo_size(c->sync_buffers[i]));
+ }
+
+ /* decode the header of the first sub-packet to find out the sample count */
+ if (buf) {
+ OpusPacket *pkt = &c->streams[0].packet;
+ ret = ff_opus_parse_packet(pkt, buf, buf_size, c->nb_streams > 1);
+ if (ret < 0) {
+ av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
+ return ret;
+ }
+ coded_samples += pkt->frame_count * pkt->frame_duration;
+ c->streams[0].silk_samplerate = get_silk_samplerate(pkt->config);
+ }
+
+ frame->nb_samples = coded_samples + delayed_samples;
+
+ /* no input or buffered data => nothing to do */
+ if (!frame->nb_samples) {
+ *got_frame_ptr = 0;
+ return 0;
+ }
+
+ /* setup the data buffers */
+ ret = ff_get_buffer(avctx, frame, 0);
+ if (ret < 0)
+ return ret;
+ frame->nb_samples = 0;
+
+ memset(c->out, 0, c->nb_streams * 2 * sizeof(*c->out));
+ for (i = 0; i < avctx->channels; i++) {
+ ChannelMap *map = &c->channel_maps[i];
+ if (!map->copy)
+ c->out[2 * map->stream_idx + map->channel_idx] = (float*)frame->extended_data[i];
+ }
+
+ /* read the data from the sync buffers */
+ for (i = 0; i < c->nb_streams; i++) {
+ float **out = c->out + 2 * i;
+ int sync_size = av_audio_fifo_size(c->sync_buffers[i]);
+
+ float sync_dummy[32];
+ int out_dummy = (!out[0]) | ((!out[1]) << 1);
+
+ if (!out[0])
+ out[0] = sync_dummy;
+ if (!out[1])
+ out[1] = sync_dummy;
+ if (out_dummy && sync_size > FF_ARRAY_ELEMS(sync_dummy))
+ return AVERROR_BUG;
+
+ ret = av_audio_fifo_read(c->sync_buffers[i], (void**)out, sync_size);
+ if (ret < 0)
+ return ret;
+
+ if (out_dummy & 1)
+ out[0] = NULL;
+ else
+ out[0] += ret;
+ if (out_dummy & 2)
+ out[1] = NULL;
+ else
+ out[1] += ret;
+
+ c->out_size[i] = frame->linesize[0] - ret * sizeof(float);
+ }
+
+ /* decode each sub-packet */
+ for (i = 0; i < c->nb_streams; i++) {
+ OpusStreamContext *s = &c->streams[i];
+
+ if (i && buf) {
+ ret = ff_opus_parse_packet(&s->packet, buf, buf_size, i != c->nb_streams - 1);
+ if (ret < 0) {
+ av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
+ return ret;
+ }
+ if (coded_samples != s->packet.frame_count * s->packet.frame_duration) {
+ av_log(avctx, AV_LOG_ERROR,
+ "Mismatching coded sample count in substream %d.\n", i);
+ return AVERROR_INVALIDDATA;
+ }
+
+ s->silk_samplerate = get_silk_samplerate(s->packet.config);
+ }
+
+ ret = opus_decode_subpacket(&c->streams[i], buf, s->packet.data_size,
+ c->out + 2 * i, c->out_size[i], coded_samples);
+ if (ret < 0)
+ return ret;
+ c->decoded_samples[i] = ret;
+ decoded_samples = FFMIN(decoded_samples, ret);
+
+ buf += s->packet.packet_size;
+ buf_size -= s->packet.packet_size;
+ }
+
+ /* buffer the extra samples */
+ for (i = 0; i < c->nb_streams; i++) {
+ int buffer_samples = c->decoded_samples[i] - decoded_samples;
+ if (buffer_samples) {
+ float *buf[2] = { c->out[2 * i + 0] ? c->out[2 * i + 0] : (float*)frame->extended_data[0],
+ c->out[2 * i + 1] ? c->out[2 * i + 1] : (float*)frame->extended_data[0] };
+ buf[0] += decoded_samples;
+ buf[1] += decoded_samples;
+ ret = av_audio_fifo_write(c->sync_buffers[i], (void**)buf, buffer_samples);
+ if (ret < 0)
+ return ret;
+ }
+ }
+
+ for (i = 0; i < avctx->channels; i++) {
+ ChannelMap *map = &c->channel_maps[i];
+
+ /* handle copied channels */
+ if (map->copy) {
+ memcpy(frame->extended_data[i],
+ frame->extended_data[map->copy_idx],
+ frame->linesize[0]);
+ } else if (map->silence) {
+ memset(frame->extended_data[i], 0, frame->linesize[0]);
+ }
+
+ if (c->gain_i && decoded_samples > 0) {
+ c->fdsp->vector_fmul_scalar((float*)frame->extended_data[i],
+ (float*)frame->extended_data[i],
+ c->gain, FFALIGN(decoded_samples, 8));
+ }
+ }
+
+ frame->nb_samples = decoded_samples;
+ *got_frame_ptr = !!decoded_samples;
+
+ return avpkt->size;
+}
+
+static av_cold void opus_decode_flush(AVCodecContext *ctx)
+{
+ OpusContext *c = ctx->priv_data;
+ int i;
+
+ for (i = 0; i < c->nb_streams; i++) {
+ OpusStreamContext *s = &c->streams[i];
+
+ memset(&s->packet, 0, sizeof(s->packet));
+ s->delayed_samples = 0;
+
+ if (s->celt_delay)
+ av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
+ swr_close(s->swr);
+
+ av_audio_fifo_drain(c->sync_buffers[i], av_audio_fifo_size(c->sync_buffers[i]));
+
+ ff_silk_flush(s->silk);
+ ff_celt_flush(s->celt);
+ }
+}
+
+static av_cold int opus_decode_close(AVCodecContext *avctx)
+{
+ OpusContext *c = avctx->priv_data;
+ int i;
+
+ for (i = 0; i < c->nb_streams; i++) {
+ OpusStreamContext *s = &c->streams[i];
+
+ ff_silk_free(&s->silk);
+ ff_celt_free(&s->celt);
+
+ av_freep(&s->out_dummy);
+ s->out_dummy_allocated_size = 0;
+
+ av_audio_fifo_free(s->celt_delay);
+ swr_free(&s->swr);
+ }
+
+ av_freep(&c->streams);
+
+ if (c->sync_buffers) {
+ for (i = 0; i < c->nb_streams; i++)
+ av_audio_fifo_free(c->sync_buffers[i]);
+ }
+ av_freep(&c->sync_buffers);
+ av_freep(&c->decoded_samples);
+ av_freep(&c->out);
+ av_freep(&c->out_size);
+
+ c->nb_streams = 0;
+
+ av_freep(&c->channel_maps);
+ av_freep(&c->fdsp);
+
+ return 0;
+}
+
+static av_cold int opus_decode_init(AVCodecContext *avctx)
+{
+ OpusContext *c = avctx->priv_data;
+ int ret, i, j;
+
+ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+ avctx->sample_rate = 48000;
+
+ c->fdsp = avpriv_float_dsp_alloc(0);
+ if (!c->fdsp)
+ return AVERROR(ENOMEM);
+
+ /* find out the channel configuration */
+ ret = ff_opus_parse_extradata(avctx, c);
+ if (ret < 0) {
+ av_freep(&c->fdsp);
+ return ret;
+ }
+
+ /* allocate and init each independent decoder */
+ c->streams = av_mallocz_array(c->nb_streams, sizeof(*c->streams));
+ c->out = av_mallocz_array(c->nb_streams, 2 * sizeof(*c->out));
+ c->out_size = av_mallocz_array(c->nb_streams, sizeof(*c->out_size));
+ c->sync_buffers = av_mallocz_array(c->nb_streams, sizeof(*c->sync_buffers));
+ c->decoded_samples = av_mallocz_array(c->nb_streams, sizeof(*c->decoded_samples));
+ if (!c->streams || !c->sync_buffers || !c->decoded_samples || !c->out || !c->out_size) {
+ c->nb_streams = 0;
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+
+ for (i = 0; i < c->nb_streams; i++) {
+ OpusStreamContext *s = &c->streams[i];
+ uint64_t layout;
+
+ s->output_channels = (i < c->nb_stereo_streams) ? 2 : 1;
+
+ s->avctx = avctx;
+
+ for (j = 0; j < s->output_channels; j++) {
+ s->silk_output[j] = s->silk_buf[j];
+ s->celt_output[j] = s->celt_buf[j];
+ s->redundancy_output[j] = s->redundancy_buf[j];
+ }
+
+ s->fdsp = c->fdsp;
+
+ s->swr =swr_alloc();
+ if (!s->swr)
+ goto fail;
+
+ layout = (s->output_channels == 1) ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
+ av_opt_set_int(s->swr, "in_sample_fmt", avctx->sample_fmt, 0);
+ av_opt_set_int(s->swr, "out_sample_fmt", avctx->sample_fmt, 0);
+ av_opt_set_int(s->swr, "in_channel_layout", layout, 0);
+ av_opt_set_int(s->swr, "out_channel_layout", layout, 0);
+ av_opt_set_int(s->swr, "out_sample_rate", avctx->sample_rate, 0);
+ av_opt_set_int(s->swr, "filter_size", 16, 0);
+
+ ret = ff_silk_init(avctx, &s->silk, s->output_channels);
+ if (ret < 0)
+ goto fail;
+
+ ret = ff_celt_init(avctx, &s->celt, s->output_channels, c->apply_phase_inv);
+ if (ret < 0)
+ goto fail;
+
+ s->celt_delay = av_audio_fifo_alloc(avctx->sample_fmt,
+ s->output_channels, 1024);
+ if (!s->celt_delay) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+
+ c->sync_buffers[i] = av_audio_fifo_alloc(avctx->sample_fmt,
+ s->output_channels, 32);
+ if (!c->sync_buffers[i]) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+ }
+
+ return 0;
+fail:
+ opus_decode_close(avctx);
+ return ret;
+}
+
+#define OFFSET(x) offsetof(OpusContext, x)
+#define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
+static const AVOption opus_options[] = {
+ { "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AD },
+ { NULL },
+};
+
+static const AVClass opus_class = {
+ .class_name = "Opus Decoder",
+ .item_name = av_default_item_name,
+ .option = opus_options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+AVCodec ff_opus_decoder = {
+ .name = "opus",
+ .long_name = NULL_IF_CONFIG_SMALL("Opus"),
+ .priv_class = &opus_class,
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_OPUS,
+ .priv_data_size = sizeof(OpusContext),
+ .init = opus_decode_init,
+ .close = opus_decode_close,
+ .decode = opus_decode_packet,
+ .flush = opus_decode_flush,
+ .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY,
+};
diff --git a/AUTHORS.txt b/trunk/AUTHORS.txt
similarity index 96%
rename from AUTHORS.txt
rename to trunk/AUTHORS.txt
index 37ac62d27..a4b17dbd0 100644
--- a/AUTHORS.txt
+++ b/trunk/AUTHORS.txt
@@ -89,4 +89,6 @@ CONTRIBUTORS ordered by first contribution.
* xbpeng121<53243357+xbpeng121@users.noreply.github.com>
* johzzy
* stone
-* cfw11<34058899+cfw11@users.noreply.github.com>
\ No newline at end of file
+* cfw11<34058899+cfw11@users.noreply.github.com>
+* Hung-YiChen
+* long
\ No newline at end of file
diff --git a/trunk/auto/auto_headers.sh b/trunk/auto/auto_headers.sh
index 589861869..824daab86 100755
--- a/trunk/auto/auto_headers.sh
+++ b/trunk/auto/auto_headers.sh
@@ -182,8 +182,8 @@ echo "" >> $SRS_AUTO_HEADERS_H
#####################################################################################
# generated the contributors from AUTHORS.txt
#####################################################################################
-if [[ -f ../AUTHORS.txt ]]; then
- SRS_CONSTRIBUTORS=`cat ../AUTHORS.txt|grep "*"|awk '{print $2}'`
+if [[ -f AUTHORS.txt ]]; then
+ SRS_CONSTRIBUTORS=`cat AUTHORS.txt|grep "*"|awk '{print $2}'`
echo "#define SRS_CONSTRIBUTORS \"\\" >> $SRS_AUTO_HEADERS_H
for CONTRIBUTOR in $SRS_CONSTRIBUTORS; do
CONTRIBUTOR=`echo $CONTRIBUTOR|sed 's/@users.noreply.github.com>/@github>/g'`
diff --git a/trunk/scripts/new_authors.sh b/trunk/scripts/new_authors.sh
index 21bba453b..3a4bc0b83 100755
--- a/trunk/scripts/new_authors.sh
+++ b/trunk/scripts/new_authors.sh
@@ -1,6 +1,6 @@
#!/bin/bash
-AFILE=`dirname $0`/../../AUTHORS.txt
+AFILE=`dirname $0`/../AUTHORS.txt
if [[ ! -f $AFILE ]]; then echo "No file at $AFILE"; exit -1; fi
authors=`git log --format='%ae'|grep -v localhost|grep -v demo|grep -v none|sort|uniq`
diff --git a/trunk/src/core/srs_core.hpp b/trunk/src/core/srs_core.hpp
index 61cbfc477..f8c60d9bd 100644
--- a/trunk/src/core/srs_core.hpp
+++ b/trunk/src/core/srs_core.hpp
@@ -23,7 +23,7 @@
#define RTMP_SIG_SRS_CODE "Leo"
#define RTMP_SIG_SRS_URL "https://github.com/ossrs/srs"
#define RTMP_SIG_SRS_LICENSE "MIT"
-#define RTMP_SIG_SRS_AUTHORS "https://github.com/ossrs/srs/blob/4.0release/AUTHORS.txt"
+#define RTMP_SIG_SRS_AUTHORS "https://github.com/ossrs/srs/blob/4.0release/trunk/AUTHORS.txt"
#define RTMP_SIG_SRS_VERSION SRS_XSTR(VERSION_MAJOR) "." SRS_XSTR(VERSION_MINOR) "." SRS_XSTR(VERSION_REVISION)
#define RTMP_SIG_SRS_SERVER RTMP_SIG_SRS_KEY "/" RTMP_SIG_SRS_VERSION "(" RTMP_SIG_SRS_CODE ")"
#define RTMP_SIG_SRS_DOMAIN "ossrs.net"