From 81a5c1b8da31b29e73665972d6ad8fe0cb35ffba Mon Sep 17 00:00:00 2001 From: winlin Date: Wed, 30 Jun 2021 08:20:53 +0800 Subject: [PATCH 1/2] Move AUTHORS.txt to trunk for docker to access it --- CHANGELOG.md | 2 +- PERFORMANCE.md | 2 +- README.md | 2 +- AUTHORS.txt => trunk/AUTHORS.txt | 4 +++- trunk/auto/auto_headers.sh | 4 ++-- trunk/scripts/new_authors.sh | 2 +- trunk/src/core/srs_core.hpp | 2 +- 7 files changed, 10 insertions(+), 8 deletions(-) rename AUTHORS.txt => trunk/AUTHORS.txt (96%) diff --git a/CHANGELOG.md b/CHANGELOG.md index 451e04c15..4113af929 100644 --- a/CHANGELOG.md +++ b/CHANGELOG.md @@ -794,7 +794,7 @@ 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 7dad6ea17..800821b01 100755 --- a/README.md +++ b/README.md @@ -378,7 +378,7 @@ 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/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 112b76017..4d0df0543 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" From b8dcf202376837a40f971bc8f1c6bcee7c371b06 Mon Sep 17 00:00:00 2001 From: winlin Date: Wed, 30 Jun 2021 10:02:05 +0800 Subject: [PATCH 2/2] Fix build fail for arm/aarch64 --- .../ffmpeg-4-fit/libavcodec/opus_silk.c | 882 ++++++++++++++++++ .../ffmpeg-4-fit/libavcodec/opusdec.c | 741 +++++++++++++++ 2 files changed, 1623 insertions(+) create mode 100644 trunk/3rdparty/ffmpeg-4-fit/libavcodec/opus_silk.c create mode 100644 trunk/3rdparty/ffmpeg-4-fit/libavcodec/opusdec.c 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, +};