mirror of https://github.com/ossrs/srs.git
Squash: Fix build fail for arm/aarch64
winlin
4 years ago
parent
75530efbc4
commit
9f5d011687
@ -0,0 +1,882 @@
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/*
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* Copyright (c) 2012 Andrew D'Addesio
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* Copyright (c) 2013-2014 Mozilla Corporation
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Opus SILK decoder
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*/
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#include <stdint.h>
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#include "opus.h"
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#include "opustab.h"
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typedef struct SilkFrame {
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int coded;
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int log_gain;
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int16_t nlsf[16];
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float lpc[16];
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float output [2 * SILK_HISTORY];
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float lpc_history[2 * SILK_HISTORY];
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int primarylag;
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int prev_voiced;
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} SilkFrame;
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struct SilkContext {
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AVCodecContext *avctx;
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int output_channels;
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int midonly;
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int subframes;
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int sflength;
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int flength;
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int nlsf_interp_factor;
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enum OpusBandwidth bandwidth;
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int wb;
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SilkFrame frame[2];
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float prev_stereo_weights[2];
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float stereo_weights[2];
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int prev_coded_channels;
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};
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static inline void silk_stabilize_lsf(int16_t nlsf[16], int order, const uint16_t min_delta[17])
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{
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int pass, i;
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for (pass = 0; pass < 20; pass++) {
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int k, min_diff = 0;
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for (i = 0; i < order+1; i++) {
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int low = i != 0 ? nlsf[i-1] : 0;
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int high = i != order ? nlsf[i] : 32768;
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int diff = (high - low) - (min_delta[i]);
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if (diff < min_diff) {
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min_diff = diff;
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k = i;
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if (pass == 20)
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break;
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}
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}
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if (min_diff == 0) /* no issues; stabilized */
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return;
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/* wiggle one or two LSFs */
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if (k == 0) {
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/* repel away from lower bound */
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nlsf[0] = min_delta[0];
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} else if (k == order) {
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/* repel away from higher bound */
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nlsf[order-1] = 32768 - min_delta[order];
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} else {
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/* repel away from current position */
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int min_center = 0, max_center = 32768, center_val;
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/* lower extent */
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for (i = 0; i < k; i++)
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min_center += min_delta[i];
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min_center += min_delta[k] >> 1;
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/* upper extent */
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for (i = order; i > k; i--)
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max_center -= min_delta[i];
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max_center -= min_delta[k] >> 1;
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/* move apart */
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center_val = nlsf[k - 1] + nlsf[k];
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center_val = (center_val >> 1) + (center_val & 1); // rounded divide by 2
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center_val = FFMIN(max_center, FFMAX(min_center, center_val));
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nlsf[k - 1] = center_val - (min_delta[k] >> 1);
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nlsf[k] = nlsf[k - 1] + min_delta[k];
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}
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}
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/* resort to the fall-back method, the standard method for LSF stabilization */
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/* sort; as the LSFs should be nearly sorted, use insertion sort */
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for (i = 1; i < order; i++) {
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int j, value = nlsf[i];
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for (j = i - 1; j >= 0 && nlsf[j] > value; j--)
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nlsf[j + 1] = nlsf[j];
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nlsf[j + 1] = value;
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}
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/* push forwards to increase distance */
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if (nlsf[0] < min_delta[0])
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nlsf[0] = min_delta[0];
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for (i = 1; i < order; i++)
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nlsf[i] = FFMAX(nlsf[i], FFMIN(nlsf[i - 1] + min_delta[i], 32767));
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/* push backwards to increase distance */
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if (nlsf[order-1] > 32768 - min_delta[order])
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nlsf[order-1] = 32768 - min_delta[order];
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for (i = order-2; i >= 0; i--)
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if (nlsf[i] > nlsf[i + 1] - min_delta[i+1])
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nlsf[i] = nlsf[i + 1] - min_delta[i+1];
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return;
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}
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static inline int silk_is_lpc_stable(const int16_t lpc[16], int order)
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{
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int k, j, DC_resp = 0;
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int32_t lpc32[2][16]; // Q24
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int totalinvgain = 1 << 30; // 1.0 in Q30
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int32_t *row = lpc32[0], *prevrow;
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/* initialize the first row for the Levinson recursion */
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for (k = 0; k < order; k++) {
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DC_resp += lpc[k];
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row[k] = lpc[k] * 4096;
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}
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if (DC_resp >= 4096)
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return 0;
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/* check if prediction gain pushes any coefficients too far */
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for (k = order - 1; 1; k--) {
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int rc; // Q31; reflection coefficient
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int gaindiv; // Q30; inverse of the gain (the divisor)
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int gain; // gain for this reflection coefficient
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int fbits; // fractional bits used for the gain
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int error; // Q29; estimate of the error of our partial estimate of 1/gaindiv
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if (FFABS(row[k]) > 16773022)
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return 0;
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rc = -(row[k] * 128);
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gaindiv = (1 << 30) - MULH(rc, rc);
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totalinvgain = MULH(totalinvgain, gaindiv) << 2;
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if (k == 0)
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return (totalinvgain >= 107374);
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/* approximate 1.0/gaindiv */
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fbits = opus_ilog(gaindiv);
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gain = ((1 << 29) - 1) / (gaindiv >> (fbits + 1 - 16)); // Q<fbits-16>
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error = (1 << 29) - MULL(gaindiv << (15 + 16 - fbits), gain, 16);
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gain = ((gain << 16) + (error * gain >> 13));
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/* switch to the next row of the LPC coefficients */
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prevrow = row;
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row = lpc32[k & 1];
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for (j = 0; j < k; j++) {
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int x = av_sat_sub32(prevrow[j], ROUND_MULL(prevrow[k - j - 1], rc, 31));
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int64_t tmp = ROUND_MULL(x, gain, fbits);
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/* per RFC 8251 section 6, if this calculation overflows, the filter
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is considered unstable. */
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if (tmp < INT32_MIN || tmp > INT32_MAX)
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return 0;
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row[j] = (int32_t)tmp;
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}
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}
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}
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static void silk_lsp2poly(const int32_t lsp[16], int32_t pol[16], int half_order)
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{
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int i, j;
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pol[0] = 65536; // 1.0 in Q16
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pol[1] = -lsp[0];
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for (i = 1; i < half_order; i++) {
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pol[i + 1] = pol[i - 1] * 2 - ROUND_MULL(lsp[2 * i], pol[i], 16);
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for (j = i; j > 1; j--)
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pol[j] += pol[j - 2] - ROUND_MULL(lsp[2 * i], pol[j - 1], 16);
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pol[1] -= lsp[2 * i];
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}
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}
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static void silk_lsf2lpc(const int16_t nlsf[16], float lpcf[16], int order)
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{
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int i, k;
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int32_t lsp[16]; // Q17; 2*cos(LSF)
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int32_t p[9], q[9]; // Q16
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int32_t lpc32[16]; // Q17
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int16_t lpc[16]; // Q12
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/* convert the LSFs to LSPs, i.e. 2*cos(LSF) */
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for (k = 0; k < order; k++) {
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int index = nlsf[k] >> 8;
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int offset = nlsf[k] & 255;
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int k2 = (order == 10) ? ff_silk_lsf_ordering_nbmb[k] : ff_silk_lsf_ordering_wb[k];
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/* interpolate and round */
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lsp[k2] = ff_silk_cosine[index] * 256;
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lsp[k2] += (ff_silk_cosine[index + 1] - ff_silk_cosine[index]) * offset;
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lsp[k2] = (lsp[k2] + 4) >> 3;
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}
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silk_lsp2poly(lsp , p, order >> 1);
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silk_lsp2poly(lsp + 1, q, order >> 1);
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/* reconstruct A(z) */
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for (k = 0; k < order>>1; k++) {
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int32_t p_tmp = p[k + 1] + p[k];
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int32_t q_tmp = q[k + 1] - q[k];
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lpc32[k] = -q_tmp - p_tmp;
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lpc32[order-k-1] = q_tmp - p_tmp;
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}
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/* limit the range of the LPC coefficients to each fit within an int16_t */
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for (i = 0; i < 10; i++) {
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int j;
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unsigned int maxabs = 0;
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for (j = 0, k = 0; j < order; j++) {
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unsigned int x = FFABS(lpc32[k]);
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if (x > maxabs) {
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maxabs = x; // Q17
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k = j;
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}
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}
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maxabs = (maxabs + 16) >> 5; // convert to Q12
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if (maxabs > 32767) {
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/* perform bandwidth expansion */
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unsigned int chirp, chirp_base; // Q16
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maxabs = FFMIN(maxabs, 163838); // anything above this overflows chirp's numerator
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chirp_base = chirp = 65470 - ((maxabs - 32767) << 14) / ((maxabs * (k+1)) >> 2);
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for (k = 0; k < order; k++) {
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lpc32[k] = ROUND_MULL(lpc32[k], chirp, 16);
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chirp = (chirp_base * chirp + 32768) >> 16;
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}
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} else break;
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}
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if (i == 10) {
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/* time's up: just clamp */
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for (k = 0; k < order; k++) {
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int x = (lpc32[k] + 16) >> 5;
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lpc[k] = av_clip_int16(x);
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lpc32[k] = lpc[k] << 5; // shortcut mandated by the spec; drops lower 5 bits
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}
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} else {
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for (k = 0; k < order; k++)
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lpc[k] = (lpc32[k] + 16) >> 5;
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}
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/* if the prediction gain causes the LPC filter to become unstable,
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apply further bandwidth expansion on the Q17 coefficients */
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for (i = 1; i <= 16 && !silk_is_lpc_stable(lpc, order); i++) {
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unsigned int chirp, chirp_base;
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chirp_base = chirp = 65536 - (1 << i);
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for (k = 0; k < order; k++) {
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lpc32[k] = ROUND_MULL(lpc32[k], chirp, 16);
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lpc[k] = (lpc32[k] + 16) >> 5;
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chirp = (chirp_base * chirp + 32768) >> 16;
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}
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}
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for (i = 0; i < order; i++)
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lpcf[i] = lpc[i] / 4096.0f;
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}
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static inline void silk_decode_lpc(SilkContext *s, SilkFrame *frame,
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OpusRangeCoder *rc,
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float lpc_leadin[16], float lpc[16],
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int *lpc_order, int *has_lpc_leadin, int voiced)
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{
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int i;
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int order; // order of the LP polynomial; 10 for NB/MB and 16 for WB
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int8_t lsf_i1, lsf_i2[16]; // stage-1 and stage-2 codebook indices
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int16_t lsf_res[16]; // residual as a Q10 value
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int16_t nlsf[16]; // Q15
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*lpc_order = order = s->wb ? 16 : 10;
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/* obtain LSF stage-1 and stage-2 indices */
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lsf_i1 = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s1[s->wb][voiced]);
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for (i = 0; i < order; i++) {
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int index = s->wb ? ff_silk_lsf_s2_model_sel_wb [lsf_i1][i] :
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ff_silk_lsf_s2_model_sel_nbmb[lsf_i1][i];
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lsf_i2[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2[index]) - 4;
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if (lsf_i2[i] == -4)
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lsf_i2[i] -= ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2_ext);
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else if (lsf_i2[i] == 4)
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lsf_i2[i] += ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2_ext);
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}
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/* reverse the backwards-prediction step */
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for (i = order - 1; i >= 0; i--) {
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int qstep = s->wb ? 9830 : 11796;
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lsf_res[i] = lsf_i2[i] * 1024;
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if (lsf_i2[i] < 0) lsf_res[i] += 102;
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else if (lsf_i2[i] > 0) lsf_res[i] -= 102;
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lsf_res[i] = (lsf_res[i] * qstep) >> 16;
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if (i + 1 < order) {
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int weight = s->wb ? ff_silk_lsf_pred_weights_wb [ff_silk_lsf_weight_sel_wb [lsf_i1][i]][i] :
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ff_silk_lsf_pred_weights_nbmb[ff_silk_lsf_weight_sel_nbmb[lsf_i1][i]][i];
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lsf_res[i] += (lsf_res[i+1] * weight) >> 8;
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}
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}
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/* reconstruct the NLSF coefficients from the supplied indices */
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for (i = 0; i < order; i++) {
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const uint8_t * codebook = s->wb ? ff_silk_lsf_codebook_wb [lsf_i1] :
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ff_silk_lsf_codebook_nbmb[lsf_i1];
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int cur, prev, next, weight_sq, weight, ipart, fpart, y, value;
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/* find the weight of the residual */
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/* TODO: precompute */
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cur = codebook[i];
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prev = i ? codebook[i - 1] : 0;
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next = i + 1 < order ? codebook[i + 1] : 256;
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weight_sq = (1024 / (cur - prev) + 1024 / (next - cur)) << 16;
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/* approximate square-root with mandated fixed-point arithmetic */
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ipart = opus_ilog(weight_sq);
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fpart = (weight_sq >> (ipart-8)) & 127;
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y = ((ipart & 1) ? 32768 : 46214) >> ((32 - ipart)>>1);
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weight = y + ((213 * fpart * y) >> 16);
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value = cur * 128 + (lsf_res[i] * 16384) / weight;
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nlsf[i] = av_clip_uintp2(value, 15);
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}
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/* stabilize the NLSF coefficients */
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silk_stabilize_lsf(nlsf, order, s->wb ? ff_silk_lsf_min_spacing_wb :
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ff_silk_lsf_min_spacing_nbmb);
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/* produce an interpolation for the first 2 subframes, */
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/* and then convert both sets of NLSFs to LPC coefficients */
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*has_lpc_leadin = 0;
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if (s->subframes == 4) {
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int offset = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_interpolation_offset);
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if (offset != 4 && frame->coded) {
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*has_lpc_leadin = 1;
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if (offset != 0) {
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int16_t nlsf_leadin[16];
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for (i = 0; i < order; i++)
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nlsf_leadin[i] = frame->nlsf[i] +
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((nlsf[i] - frame->nlsf[i]) * offset >> 2);
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silk_lsf2lpc(nlsf_leadin, lpc_leadin, order);
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} else /* avoid re-computation for a (roughly) 1-in-4 occurrence */
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memcpy(lpc_leadin, frame->lpc, 16 * sizeof(float));
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} else
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offset = 4;
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s->nlsf_interp_factor = offset;
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silk_lsf2lpc(nlsf, lpc, order);
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} else {
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s->nlsf_interp_factor = 4;
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silk_lsf2lpc(nlsf, lpc, order);
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}
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memcpy(frame->nlsf, nlsf, order * sizeof(nlsf[0]));
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memcpy(frame->lpc, lpc, order * sizeof(lpc[0]));
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}
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static inline void silk_count_children(OpusRangeCoder *rc, int model, int32_t total,
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int32_t child[2])
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{
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if (total != 0) {
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child[0] = ff_opus_rc_dec_cdf(rc,
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ff_silk_model_pulse_location[model] + (((total - 1 + 5) * (total - 1)) >> 1));
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child[1] = total - child[0];
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} else {
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child[0] = 0;
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child[1] = 0;
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}
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}
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static inline void silk_decode_excitation(SilkContext *s, OpusRangeCoder *rc,
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float* excitationf,
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int qoffset_high, int active, int voiced)
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{
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int i;
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uint32_t seed;
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int shellblocks;
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int ratelevel;
|
||||
uint8_t pulsecount[20]; // total pulses in each shell block
|
||||
uint8_t lsbcount[20] = {0}; // raw lsbits defined for each pulse in each shell block
|
||||
int32_t excitation[320]; // Q23
|
||||
|
||||
/* excitation parameters */
|
||||
seed = ff_opus_rc_dec_cdf(rc, ff_silk_model_lcg_seed);
|
||||
shellblocks = ff_silk_shell_blocks[s->bandwidth][s->subframes >> 2];
|
||||
ratelevel = ff_opus_rc_dec_cdf(rc, ff_silk_model_exc_rate[voiced]);
|
||||
|
||||
for (i = 0; i < shellblocks; i++) {
|
||||
pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[ratelevel]);
|
||||
if (pulsecount[i] == 17) {
|
||||
while (pulsecount[i] == 17 && ++lsbcount[i] != 10)
|
||||
pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[9]);
|
||||
if (lsbcount[i] == 10)
|
||||
pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[10]);
|
||||
}
|
||||
}
|
||||
|
||||
/* decode pulse locations using PVQ */
|
||||
for (i = 0; i < shellblocks; i++) {
|
||||
if (pulsecount[i] != 0) {
|
||||
int a, b, c, d;
|
||||
int32_t * location = excitation + 16*i;
|
||||
int32_t branch[4][2];
|
||||
branch[0][0] = pulsecount[i];
|
||||
|
||||
/* unrolled tail recursion */
|
||||
for (a = 0; a < 1; a++) {
|
||||
silk_count_children(rc, 0, branch[0][a], branch[1]);
|
||||
for (b = 0; b < 2; b++) {
|
||||
silk_count_children(rc, 1, branch[1][b], branch[2]);
|
||||
for (c = 0; c < 2; c++) {
|
||||
silk_count_children(rc, 2, branch[2][c], branch[3]);
|
||||
for (d = 0; d < 2; d++) {
|
||||
silk_count_children(rc, 3, branch[3][d], location);
|
||||
location += 2;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
} else
|
||||
memset(excitation + 16*i, 0, 16*sizeof(int32_t));
|
||||
}
|
||||
|
||||
/* decode least significant bits */
|
||||
for (i = 0; i < shellblocks << 4; i++) {
|
||||
int bit;
|
||||
for (bit = 0; bit < lsbcount[i >> 4]; bit++)
|
||||
excitation[i] = (excitation[i] << 1) |
|
||||
ff_opus_rc_dec_cdf(rc, ff_silk_model_excitation_lsb);
|
||||
}
|
||||
|
||||
/* decode signs */
|
||||
for (i = 0; i < shellblocks << 4; i++) {
|
||||
if (excitation[i] != 0) {
|
||||
int sign = ff_opus_rc_dec_cdf(rc, ff_silk_model_excitation_sign[active +
|
||||
voiced][qoffset_high][FFMIN(pulsecount[i >> 4], 6)]);
|
||||
if (sign == 0)
|
||||
excitation[i] *= -1;
|
||||
}
|
||||
}
|
||||
|
||||
/* assemble the excitation */
|
||||
for (i = 0; i < shellblocks << 4; i++) {
|
||||
int value = excitation[i];
|
||||
excitation[i] = value * 256 | ff_silk_quant_offset[voiced][qoffset_high];
|
||||
if (value < 0) excitation[i] += 20;
|
||||
else if (value > 0) excitation[i] -= 20;
|
||||
|
||||
/* invert samples pseudorandomly */
|
||||
seed = 196314165 * seed + 907633515;
|
||||
if (seed & 0x80000000)
|
||||
excitation[i] *= -1;
|
||||
seed += value;
|
||||
|
||||
excitationf[i] = excitation[i] / 8388608.0f;
|
||||
}
|
||||
}
|
||||
|
||||
/** Maximum residual history according to 4.2.7.6.1 */
|
||||
#define SILK_MAX_LAG (288 + LTP_ORDER / 2)
|
||||
|
||||
/** Order of the LTP filter */
|
||||
#define LTP_ORDER 5
|
||||
|
||||
static void silk_decode_frame(SilkContext *s, OpusRangeCoder *rc,
|
||||
int frame_num, int channel, int coded_channels, int active, int active1)
|
||||
{
|
||||
/* per frame */
|
||||
int voiced; // combines with active to indicate inactive, active, or active+voiced
|
||||
int qoffset_high;
|
||||
int order; // order of the LPC coefficients
|
||||
float lpc_leadin[16], lpc_body[16], residual[SILK_MAX_LAG + SILK_HISTORY];
|
||||
int has_lpc_leadin;
|
||||
float ltpscale;
|
||||
|
||||
/* per subframe */
|
||||
struct {
|
||||
float gain;
|
||||
int pitchlag;
|
||||
float ltptaps[5];
|
||||
} sf[4];
|
||||
|
||||
SilkFrame * const frame = s->frame + channel;
|
||||
|
||||
int i;
|
||||
|
||||
/* obtain stereo weights */
|
||||
if (coded_channels == 2 && channel == 0) {
|
||||
int n, wi[2], ws[2], w[2];
|
||||
n = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s1);
|
||||
wi[0] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s2) + 3 * (n / 5);
|
||||
ws[0] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s3);
|
||||
wi[1] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s2) + 3 * (n % 5);
|
||||
ws[1] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s3);
|
||||
|
||||
for (i = 0; i < 2; i++)
|
||||
w[i] = ff_silk_stereo_weights[wi[i]] +
|
||||
(((ff_silk_stereo_weights[wi[i] + 1] - ff_silk_stereo_weights[wi[i]]) * 6554) >> 16)
|
||||
* (ws[i]*2 + 1);
|
||||
|
||||
s->stereo_weights[0] = (w[0] - w[1]) / 8192.0;
|
||||
s->stereo_weights[1] = w[1] / 8192.0;
|
||||
|
||||
/* and read the mid-only flag */
|
||||
s->midonly = active1 ? 0 : ff_opus_rc_dec_cdf(rc, ff_silk_model_mid_only);
|
||||
}
|
||||
|
||||
/* obtain frame type */
|
||||
if (!active) {
|
||||
qoffset_high = ff_opus_rc_dec_cdf(rc, ff_silk_model_frame_type_inactive);
|
||||
voiced = 0;
|
||||
} else {
|
||||
int type = ff_opus_rc_dec_cdf(rc, ff_silk_model_frame_type_active);
|
||||
qoffset_high = type & 1;
|
||||
voiced = type >> 1;
|
||||
}
|
||||
|
||||
/* obtain subframe quantization gains */
|
||||
for (i = 0; i < s->subframes; i++) {
|
||||
int log_gain; //Q7
|
||||
int ipart, fpart, lingain;
|
||||
|
||||
if (i == 0 && (frame_num == 0 || !frame->coded)) {
|
||||
/* gain is coded absolute */
|
||||
int x = ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_highbits[active + voiced]);
|
||||
log_gain = (x<<3) | ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_lowbits);
|
||||
|
||||
if (frame->coded)
|
||||
log_gain = FFMAX(log_gain, frame->log_gain - 16);
|
||||
} else {
|
||||
/* gain is coded relative */
|
||||
int delta_gain = ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_delta);
|
||||
log_gain = av_clip_uintp2(FFMAX((delta_gain<<1) - 16,
|
||||
frame->log_gain + delta_gain - 4), 6);
|
||||
}
|
||||
|
||||
frame->log_gain = log_gain;
|
||||
|
||||
/* approximate 2**(x/128) with a Q7 (i.e. non-integer) input */
|
||||
log_gain = (log_gain * 0x1D1C71 >> 16) + 2090;
|
||||
ipart = log_gain >> 7;
|
||||
fpart = log_gain & 127;
|
||||
lingain = (1 << ipart) + ((-174 * fpart * (128-fpart) >>16) + fpart) * ((1<<ipart) >> 7);
|
||||
sf[i].gain = lingain / 65536.0f;
|
||||
}
|
||||
|
||||
/* obtain LPC filter coefficients */
|
||||
silk_decode_lpc(s, frame, rc, lpc_leadin, lpc_body, &order, &has_lpc_leadin, voiced);
|
||||
|
||||
/* obtain pitch lags, if this is a voiced frame */
|
||||
if (voiced) {
|
||||
int lag_absolute = (!frame_num || !frame->prev_voiced);
|
||||
int primarylag; // primary pitch lag for the entire SILK frame
|
||||
int ltpfilter;
|
||||
const int8_t * offsets;
|
||||
|
||||
if (!lag_absolute) {
|
||||
int delta = ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_delta);
|
||||
if (delta)
|
||||
primarylag = frame->primarylag + delta - 9;
|
||||
else
|
||||
lag_absolute = 1;
|
||||
}
|
||||
|
||||
if (lag_absolute) {
|
||||
/* primary lag is coded absolute */
|
||||
int highbits, lowbits;
|
||||
static const uint16_t * const model[] = {
|
||||
ff_silk_model_pitch_lowbits_nb, ff_silk_model_pitch_lowbits_mb,
|
||||
ff_silk_model_pitch_lowbits_wb
|
||||
};
|
||||
highbits = ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_highbits);
|
||||
lowbits = ff_opus_rc_dec_cdf(rc, model[s->bandwidth]);
|
||||
|
||||
primarylag = ff_silk_pitch_min_lag[s->bandwidth] +
|
||||
highbits*ff_silk_pitch_scale[s->bandwidth] + lowbits;
|
||||
}
|
||||
frame->primarylag = primarylag;
|
||||
|
||||
if (s->subframes == 2)
|
||||
offsets = (s->bandwidth == OPUS_BANDWIDTH_NARROWBAND)
|
||||
? ff_silk_pitch_offset_nb10ms[ff_opus_rc_dec_cdf(rc,
|
||||
ff_silk_model_pitch_contour_nb10ms)]
|
||||
: ff_silk_pitch_offset_mbwb10ms[ff_opus_rc_dec_cdf(rc,
|
||||
ff_silk_model_pitch_contour_mbwb10ms)];
|
||||
else
|
||||
offsets = (s->bandwidth == OPUS_BANDWIDTH_NARROWBAND)
|
||||
? ff_silk_pitch_offset_nb20ms[ff_opus_rc_dec_cdf(rc,
|
||||
ff_silk_model_pitch_contour_nb20ms)]
|
||||
: ff_silk_pitch_offset_mbwb20ms[ff_opus_rc_dec_cdf(rc,
|
||||
ff_silk_model_pitch_contour_mbwb20ms)];
|
||||
|
||||
for (i = 0; i < s->subframes; i++)
|
||||
sf[i].pitchlag = av_clip(primarylag + offsets[i],
|
||||
ff_silk_pitch_min_lag[s->bandwidth],
|
||||
ff_silk_pitch_max_lag[s->bandwidth]);
|
||||
|
||||
/* obtain LTP filter coefficients */
|
||||
ltpfilter = ff_opus_rc_dec_cdf(rc, ff_silk_model_ltp_filter);
|
||||
for (i = 0; i < s->subframes; i++) {
|
||||
int index, j;
|
||||
static const uint16_t * const filter_sel[] = {
|
||||
ff_silk_model_ltp_filter0_sel, ff_silk_model_ltp_filter1_sel,
|
||||
ff_silk_model_ltp_filter2_sel
|
||||
};
|
||||
static const int8_t (* const filter_taps[])[5] = {
|
||||
ff_silk_ltp_filter0_taps, ff_silk_ltp_filter1_taps, ff_silk_ltp_filter2_taps
|
||||
};
|
||||
index = ff_opus_rc_dec_cdf(rc, filter_sel[ltpfilter]);
|
||||
for (j = 0; j < 5; j++)
|
||||
sf[i].ltptaps[j] = filter_taps[ltpfilter][index][j] / 128.0f;
|
||||
}
|
||||
}
|
||||
|
||||
/* obtain LTP scale factor */
|
||||
if (voiced && frame_num == 0)
|
||||
ltpscale = ff_silk_ltp_scale_factor[ff_opus_rc_dec_cdf(rc,
|
||||
ff_silk_model_ltp_scale_index)] / 16384.0f;
|
||||
else ltpscale = 15565.0f/16384.0f;
|
||||
|
||||
/* generate the excitation signal for the entire frame */
|
||||
silk_decode_excitation(s, rc, residual + SILK_MAX_LAG, qoffset_high,
|
||||
active, voiced);
|
||||
|
||||
/* skip synthesising the side channel if we want mono-only */
|
||||
if (s->output_channels == channel)
|
||||
return;
|
||||
|
||||
/* generate the output signal */
|
||||
for (i = 0; i < s->subframes; i++) {
|
||||
const float * lpc_coeff = (i < 2 && has_lpc_leadin) ? lpc_leadin : lpc_body;
|
||||
float *dst = frame->output + SILK_HISTORY + i * s->sflength;
|
||||
float *resptr = residual + SILK_MAX_LAG + i * s->sflength;
|
||||
float *lpc = frame->lpc_history + SILK_HISTORY + i * s->sflength;
|
||||
float sum;
|
||||
int j, k;
|
||||
|
||||
if (voiced) {
|
||||
int out_end;
|
||||
float scale;
|
||||
|
||||
if (i < 2 || s->nlsf_interp_factor == 4) {
|
||||
out_end = -i * s->sflength;
|
||||
scale = ltpscale;
|
||||
} else {
|
||||
out_end = -(i - 2) * s->sflength;
|
||||
scale = 1.0f;
|
||||
}
|
||||
|
||||
/* when the LPC coefficients change, a re-whitening filter is used */
|
||||
/* to produce a residual that accounts for the change */
|
||||
for (j = - sf[i].pitchlag - LTP_ORDER/2; j < out_end; j++) {
|
||||
sum = dst[j];
|
||||
for (k = 0; k < order; k++)
|
||||
sum -= lpc_coeff[k] * dst[j - k - 1];
|
||||
resptr[j] = av_clipf(sum, -1.0f, 1.0f) * scale / sf[i].gain;
|
||||
}
|
||||
|
||||
if (out_end) {
|
||||
float rescale = sf[i-1].gain / sf[i].gain;
|
||||
for (j = out_end; j < 0; j++)
|
||||
resptr[j] *= rescale;
|
||||
}
|
||||
|
||||
/* LTP synthesis */
|
||||
for (j = 0; j < s->sflength; j++) {
|
||||
sum = resptr[j];
|
||||
for (k = 0; k < LTP_ORDER; k++)
|
||||
sum += sf[i].ltptaps[k] * resptr[j - sf[i].pitchlag + LTP_ORDER/2 - k];
|
||||
resptr[j] = sum;
|
||||
}
|
||||
}
|
||||
|
||||
/* LPC synthesis */
|
||||
for (j = 0; j < s->sflength; j++) {
|
||||
sum = resptr[j] * sf[i].gain;
|
||||
for (k = 1; k <= order; k++)
|
||||
sum += lpc_coeff[k - 1] * lpc[j - k];
|
||||
|
||||
lpc[j] = sum;
|
||||
dst[j] = av_clipf(sum, -1.0f, 1.0f);
|
||||
}
|
||||
}
|
||||
|
||||
frame->prev_voiced = voiced;
|
||||
memmove(frame->lpc_history, frame->lpc_history + s->flength, SILK_HISTORY * sizeof(float));
|
||||
memmove(frame->output, frame->output + s->flength, SILK_HISTORY * sizeof(float));
|
||||
|
||||
frame->coded = 1;
|
||||
}
|
||||
|
||||
static void silk_unmix_ms(SilkContext *s, float *l, float *r)
|
||||
{
|
||||
float *mid = s->frame[0].output + SILK_HISTORY - s->flength;
|
||||
float *side = s->frame[1].output + SILK_HISTORY - s->flength;
|
||||
float w0_prev = s->prev_stereo_weights[0];
|
||||
float w1_prev = s->prev_stereo_weights[1];
|
||||
float w0 = s->stereo_weights[0];
|
||||
float w1 = s->stereo_weights[1];
|
||||
int n1 = ff_silk_stereo_interp_len[s->bandwidth];
|
||||
int i;
|
||||
|
||||
for (i = 0; i < n1; i++) {
|
||||
float interp0 = w0_prev + i * (w0 - w0_prev) / n1;
|
||||
float interp1 = w1_prev + i * (w1 - w1_prev) / n1;
|
||||
float p0 = 0.25 * (mid[i - 2] + 2 * mid[i - 1] + mid[i]);
|
||||
|
||||
l[i] = av_clipf((1 + interp1) * mid[i - 1] + side[i - 1] + interp0 * p0, -1.0, 1.0);
|
||||
r[i] = av_clipf((1 - interp1) * mid[i - 1] - side[i - 1] - interp0 * p0, -1.0, 1.0);
|
||||
}
|
||||
|
||||
for (; i < s->flength; i++) {
|
||||
float p0 = 0.25 * (mid[i - 2] + 2 * mid[i - 1] + mid[i]);
|
||||
|
||||
l[i] = av_clipf((1 + w1) * mid[i - 1] + side[i - 1] + w0 * p0, -1.0, 1.0);
|
||||
r[i] = av_clipf((1 - w1) * mid[i - 1] - side[i - 1] - w0 * p0, -1.0, 1.0);
|
||||
}
|
||||
|
||||
memcpy(s->prev_stereo_weights, s->stereo_weights, sizeof(s->stereo_weights));
|
||||
}
|
||||
|
||||
static void silk_flush_frame(SilkFrame *frame)
|
||||
{
|
||||
if (!frame->coded)
|
||||
return;
|
||||
|
||||
memset(frame->output, 0, sizeof(frame->output));
|
||||
memset(frame->lpc_history, 0, sizeof(frame->lpc_history));
|
||||
|
||||
memset(frame->lpc, 0, sizeof(frame->lpc));
|
||||
memset(frame->nlsf, 0, sizeof(frame->nlsf));
|
||||
|
||||
frame->log_gain = 0;
|
||||
|
||||
frame->primarylag = 0;
|
||||
frame->prev_voiced = 0;
|
||||
frame->coded = 0;
|
||||
}
|
||||
|
||||
int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc,
|
||||
float *output[2],
|
||||
enum OpusBandwidth bandwidth,
|
||||
int coded_channels,
|
||||
int duration_ms)
|
||||
{
|
||||
int active[2][6], redundancy[2];
|
||||
int nb_frames, i, j;
|
||||
|
||||
if (bandwidth > OPUS_BANDWIDTH_WIDEBAND ||
|
||||
coded_channels > 2 || duration_ms > 60) {
|
||||
av_log(s->avctx, AV_LOG_ERROR, "Invalid parameters passed "
|
||||
"to the SILK decoder.\n");
|
||||
return AVERROR(EINVAL);
|
||||
}
|
||||
|
||||
nb_frames = 1 + (duration_ms > 20) + (duration_ms > 40);
|
||||
s->subframes = duration_ms / nb_frames / 5; // 5ms subframes
|
||||
s->sflength = 20 * (bandwidth + 2);
|
||||
s->flength = s->sflength * s->subframes;
|
||||
s->bandwidth = bandwidth;
|
||||
s->wb = bandwidth == OPUS_BANDWIDTH_WIDEBAND;
|
||||
|
||||
/* make sure to flush the side channel when switching from mono to stereo */
|
||||
if (coded_channels > s->prev_coded_channels)
|
||||
silk_flush_frame(&s->frame[1]);
|
||||
s->prev_coded_channels = coded_channels;
|
||||
|
||||
/* read the LP-layer header bits */
|
||||
for (i = 0; i < coded_channels; i++) {
|
||||
for (j = 0; j < nb_frames; j++)
|
||||
active[i][j] = ff_opus_rc_dec_log(rc, 1);
|
||||
|
||||
redundancy[i] = ff_opus_rc_dec_log(rc, 1);
|
||||
if (redundancy[i]) {
|
||||
avpriv_report_missing_feature(s->avctx, "LBRR frames");
|
||||
return AVERROR_PATCHWELCOME;
|
||||
}
|
||||
}
|
||||
|
||||
for (i = 0; i < nb_frames; i++) {
|
||||
for (j = 0; j < coded_channels && !s->midonly; j++)
|
||||
silk_decode_frame(s, rc, i, j, coded_channels, active[j][i], active[1][i]);
|
||||
|
||||
/* reset the side channel if it is not coded */
|
||||
if (s->midonly && s->frame[1].coded)
|
||||
silk_flush_frame(&s->frame[1]);
|
||||
|
||||
if (coded_channels == 1 || s->output_channels == 1) {
|
||||
for (j = 0; j < s->output_channels; j++) {
|
||||
memcpy(output[j] + i * s->flength,
|
||||
s->frame[0].output + SILK_HISTORY - s->flength - 2,
|
||||
s->flength * sizeof(float));
|
||||
}
|
||||
} else {
|
||||
silk_unmix_ms(s, output[0] + i * s->flength, output[1] + i * s->flength);
|
||||
}
|
||||
|
||||
s->midonly = 0;
|
||||
}
|
||||
|
||||
return nb_frames * s->flength;
|
||||
}
|
||||
|
||||
void ff_silk_free(SilkContext **ps)
|
||||
{
|
||||
av_freep(ps);
|
||||
}
|
||||
|
||||
void ff_silk_flush(SilkContext *s)
|
||||
{
|
||||
silk_flush_frame(&s->frame[0]);
|
||||
silk_flush_frame(&s->frame[1]);
|
||||
|
||||
memset(s->prev_stereo_weights, 0, sizeof(s->prev_stereo_weights));
|
||||
}
|
||||
|
||||
int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels)
|
||||
{
|
||||
SilkContext *s;
|
||||
|
||||
if (output_channels != 1 && output_channels != 2) {
|
||||
av_log(avctx, AV_LOG_ERROR, "Invalid number of output channels: %d\n",
|
||||
output_channels);
|
||||
return AVERROR(EINVAL);
|
||||
}
|
||||
|
||||
s = av_mallocz(sizeof(*s));
|
||||
if (!s)
|
||||
return AVERROR(ENOMEM);
|
||||
|
||||
s->avctx = avctx;
|
||||
s->output_channels = output_channels;
|
||||
|
||||
ff_silk_flush(s);
|
||||
|
||||
*ps = s;
|
||||
|
||||
return 0;
|
||||
}
|
||||
@ -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 <stdint.h>
|
||||
|
||||
#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,
|
||||
};
|
||||
Loading…
Reference in New Issue