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FFmpeg4/libavcodec/opus_silk.c

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/*
* Copyright (c) 2012 Andrew D'Addesio
* Copyright (c) 2013-2014 Mozilla Corporation
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Opus SILK decoder
*/
#include <stdint.h>
#include "opus.h"
#include "opustab.h"
typedef struct SilkFrame {
int coded;
int log_gain;
int16_t nlsf[16];
float lpc[16];
float output [2 * SILK_HISTORY];
float lpc_history[2 * SILK_HISTORY];
int primarylag;
int prev_voiced;
} SilkFrame;
struct SilkContext {
AVCodecContext *avctx;
int output_channels;
int midonly;
int subframes;
int sflength;
int flength;
int nlsf_interp_factor;
enum OpusBandwidth bandwidth;
int wb;
SilkFrame frame[2];
float prev_stereo_weights[2];
float stereo_weights[2];
int prev_coded_channels;
};
static inline void silk_stabilize_lsf(int16_t nlsf[16], int order, const uint16_t min_delta[17])
{
int pass, i;
for (pass = 0; pass < 20; pass++) {
int k, min_diff = 0;
for (i = 0; i < order+1; i++) {
int low = i != 0 ? nlsf[i-1] : 0;
int high = i != order ? nlsf[i] : 32768;
int diff = (high - low) - (min_delta[i]);
if (diff < min_diff) {
min_diff = diff;
k = i;
if (pass == 20)
break;
}
}
if (min_diff == 0) /* no issues; stabilized */
return;
/* wiggle one or two LSFs */
if (k == 0) {
/* repel away from lower bound */
nlsf[0] = min_delta[0];
} else if (k == order) {
/* repel away from higher bound */
nlsf[order-1] = 32768 - min_delta[order];
} else {
/* repel away from current position */
int min_center = 0, max_center = 32768, center_val;
/* lower extent */
for (i = 0; i < k; i++)
min_center += min_delta[i];
min_center += min_delta[k] >> 1;
/* upper extent */
for (i = order; i > k; i--)
max_center -= min_delta[i];
max_center -= min_delta[k] >> 1;
/* move apart */
center_val = nlsf[k - 1] + nlsf[k];
center_val = (center_val >> 1) + (center_val & 1); // rounded divide by 2
center_val = FFMIN(max_center, FFMAX(min_center, center_val));
nlsf[k - 1] = center_val - (min_delta[k] >> 1);
nlsf[k] = nlsf[k - 1] + min_delta[k];
}
}
/* resort to the fall-back method, the standard method for LSF stabilization */
/* sort; as the LSFs should be nearly sorted, use insertion sort */
for (i = 1; i < order; i++) {
int j, value = nlsf[i];
for (j = i - 1; j >= 0 && nlsf[j] > value; j--)
nlsf[j + 1] = nlsf[j];
nlsf[j + 1] = value;
}
/* push forwards to increase distance */
if (nlsf[0] < min_delta[0])
nlsf[0] = min_delta[0];
for (i = 1; i < order; i++)
nlsf[i] = FFMAX(nlsf[i], FFMIN(nlsf[i - 1] + min_delta[i], 32767));
/* push backwards to increase distance */
if (nlsf[order-1] > 32768 - min_delta[order])
nlsf[order-1] = 32768 - min_delta[order];
for (i = order-2; i >= 0; i--)
if (nlsf[i] > nlsf[i + 1] - min_delta[i+1])
nlsf[i] = nlsf[i + 1] - min_delta[i+1];
return;
}
static inline int silk_is_lpc_stable(const int16_t lpc[16], int order)
{
int k, j, DC_resp = 0;
int32_t lpc32[2][16]; // Q24
int totalinvgain = 1 << 30; // 1.0 in Q30
int32_t *row = lpc32[0], *prevrow;
/* initialize the first row for the Levinson recursion */
for (k = 0; k < order; k++) {
DC_resp += lpc[k];
row[k] = lpc[k] * 4096;
}
if (DC_resp >= 4096)
return 0;
/* check if prediction gain pushes any coefficients too far */
for (k = order - 1; 1; k--) {
int rc; // Q31; reflection coefficient
int gaindiv; // Q30; inverse of the gain (the divisor)
int gain; // gain for this reflection coefficient
int fbits; // fractional bits used for the gain
int error; // Q29; estimate of the error of our partial estimate of 1/gaindiv
if (FFABS(row[k]) > 16773022)
return 0;
rc = -(row[k] * 128);
gaindiv = (1 << 30) - MULH(rc, rc);
totalinvgain = MULH(totalinvgain, gaindiv) << 2;
if (k == 0)
return (totalinvgain >= 107374);
/* approximate 1.0/gaindiv */
fbits = opus_ilog(gaindiv);
gain = ((1 << 29) - 1) / (gaindiv >> (fbits + 1 - 16)); // Q<fbits-16>
error = (1 << 29) - MULL(gaindiv << (15 + 16 - fbits), gain, 16);
gain = ((gain << 16) + (error * gain >> 13));
/* switch to the next row of the LPC coefficients */
prevrow = row;
row = lpc32[k & 1];
for (j = 0; j < k; j++) {
int x = av_sat_sub32(prevrow[j], ROUND_MULL(prevrow[k - j - 1], rc, 31));
int64_t tmp = ROUND_MULL(x, gain, fbits);
/* per RFC 8251 section 6, if this calculation overflows, the filter
is considered unstable. */
if (tmp < INT32_MIN || tmp > INT32_MAX)
return 0;
row[j] = (int32_t)tmp;
}
}
}
static void silk_lsp2poly(const int32_t lsp[16], int32_t pol[16], int half_order)
{
int i, j;
pol[0] = 65536; // 1.0 in Q16
pol[1] = -lsp[0];
for (i = 1; i < half_order; i++) {
pol[i + 1] = pol[i - 1] * 2 - ROUND_MULL(lsp[2 * i], pol[i], 16);
for (j = i; j > 1; j--)
pol[j] += pol[j - 2] - ROUND_MULL(lsp[2 * i], pol[j - 1], 16);
pol[1] -= lsp[2 * i];
}
}
static void silk_lsf2lpc(const int16_t nlsf[16], float lpcf[16], int order)
{
int i, k;
int32_t lsp[16]; // Q17; 2*cos(LSF)
int32_t p[9], q[9]; // Q16
int32_t lpc32[16]; // Q17
int16_t lpc[16]; // Q12
/* convert the LSFs to LSPs, i.e. 2*cos(LSF) */
for (k = 0; k < order; k++) {
int index = nlsf[k] >> 8;
int offset = nlsf[k] & 255;
int k2 = (order == 10) ? ff_silk_lsf_ordering_nbmb[k] : ff_silk_lsf_ordering_wb[k];
/* interpolate and round */
lsp[k2] = ff_silk_cosine[index] * 256;
lsp[k2] += (ff_silk_cosine[index + 1] - ff_silk_cosine[index]) * offset;
lsp[k2] = (lsp[k2] + 4) >> 3;
}
silk_lsp2poly(lsp , p, order >> 1);
silk_lsp2poly(lsp + 1, q, order >> 1);
/* reconstruct A(z) */
for (k = 0; k < order>>1; k++) {
int32_t p_tmp = p[k + 1] + p[k];
int32_t q_tmp = q[k + 1] - q[k];
lpc32[k] = -q_tmp - p_tmp;
lpc32[order-k-1] = q_tmp - p_tmp;
}
/* limit the range of the LPC coefficients to each fit within an int16_t */
for (i = 0; i < 10; i++) {
int j;
unsigned int maxabs = 0;
for (j = 0, k = 0; j < order; j++) {
unsigned int x = FFABS(lpc32[k]);
if (x > maxabs) {
maxabs = x; // Q17
k = j;
}
}
maxabs = (maxabs + 16) >> 5; // convert to Q12
if (maxabs > 32767) {
/* perform bandwidth expansion */
unsigned int chirp, chirp_base; // Q16
maxabs = FFMIN(maxabs, 163838); // anything above this overflows chirp's numerator
chirp_base = chirp = 65470 - ((maxabs - 32767) << 14) / ((maxabs * (k+1)) >> 2);
for (k = 0; k < order; k++) {
lpc32[k] = ROUND_MULL(lpc32[k], chirp, 16);
chirp = (chirp_base * chirp + 32768) >> 16;
}
} else break;
}
if (i == 10) {
/* time's up: just clamp */
for (k = 0; k < order; k++) {
int x = (lpc32[k] + 16) >> 5;
lpc[k] = av_clip_int16(x);
lpc32[k] = lpc[k] << 5; // shortcut mandated by the spec; drops lower 5 bits
}
} else {
for (k = 0; k < order; k++)
lpc[k] = (lpc32[k] + 16) >> 5;
}
/* if the prediction gain causes the LPC filter to become unstable,
apply further bandwidth expansion on the Q17 coefficients */
for (i = 1; i <= 16 && !silk_is_lpc_stable(lpc, order); i++) {
unsigned int chirp, chirp_base;
chirp_base = chirp = 65536 - (1 << i);
for (k = 0; k < order; k++) {
lpc32[k] = ROUND_MULL(lpc32[k], chirp, 16);
lpc[k] = (lpc32[k] + 16) >> 5;
chirp = (chirp_base * chirp + 32768) >> 16;
}
}
for (i = 0; i < order; i++)
lpcf[i] = lpc[i] / 4096.0f;
}
static inline void silk_decode_lpc(SilkContext *s, SilkFrame *frame,
OpusRangeCoder *rc,
float lpc_leadin[16], float lpc[16],
int *lpc_order, int *has_lpc_leadin, int voiced)
{
int i;
int order; // order of the LP polynomial; 10 for NB/MB and 16 for WB
int8_t lsf_i1, lsf_i2[16]; // stage-1 and stage-2 codebook indices
int16_t lsf_res[16]; // residual as a Q10 value
int16_t nlsf[16]; // Q15
*lpc_order = order = s->wb ? 16 : 10;
/* obtain LSF stage-1 and stage-2 indices */
lsf_i1 = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s1[s->wb][voiced]);
for (i = 0; i < order; i++) {
int index = s->wb ? ff_silk_lsf_s2_model_sel_wb [lsf_i1][i] :
ff_silk_lsf_s2_model_sel_nbmb[lsf_i1][i];
lsf_i2[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2[index]) - 4;
if (lsf_i2[i] == -4)
lsf_i2[i] -= ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2_ext);
else if (lsf_i2[i] == 4)
lsf_i2[i] += ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2_ext);
}
/* reverse the backwards-prediction step */
for (i = order - 1; i >= 0; i--) {
int qstep = s->wb ? 9830 : 11796;
lsf_res[i] = lsf_i2[i] * 1024;
if (lsf_i2[i] < 0) lsf_res[i] += 102;
else if (lsf_i2[i] > 0) lsf_res[i] -= 102;
lsf_res[i] = (lsf_res[i] * qstep) >> 16;
if (i + 1 < order) {
int weight = s->wb ? ff_silk_lsf_pred_weights_wb [ff_silk_lsf_weight_sel_wb [lsf_i1][i]][i] :
ff_silk_lsf_pred_weights_nbmb[ff_silk_lsf_weight_sel_nbmb[lsf_i1][i]][i];
lsf_res[i] += (lsf_res[i+1] * weight) >> 8;
}
}
/* reconstruct the NLSF coefficients from the supplied indices */
for (i = 0; i < order; i++) {
const uint8_t * codebook = s->wb ? ff_silk_lsf_codebook_wb [lsf_i1] :
ff_silk_lsf_codebook_nbmb[lsf_i1];
int cur, prev, next, weight_sq, weight, ipart, fpart, y, value;
/* find the weight of the residual */
/* TODO: precompute */
cur = codebook[i];
prev = i ? codebook[i - 1] : 0;
next = i + 1 < order ? codebook[i + 1] : 256;
weight_sq = (1024 / (cur - prev) + 1024 / (next - cur)) << 16;
/* approximate square-root with mandated fixed-point arithmetic */
ipart = opus_ilog(weight_sq);
fpart = (weight_sq >> (ipart-8)) & 127;
y = ((ipart & 1) ? 32768 : 46214) >> ((32 - ipart)>>1);
weight = y + ((213 * fpart * y) >> 16);
value = cur * 128 + (lsf_res[i] * 16384) / weight;
nlsf[i] = av_clip_uintp2(value, 15);
}
/* stabilize the NLSF coefficients */
silk_stabilize_lsf(nlsf, order, s->wb ? ff_silk_lsf_min_spacing_wb :
ff_silk_lsf_min_spacing_nbmb);
/* produce an interpolation for the first 2 subframes, */
/* and then convert both sets of NLSFs to LPC coefficients */
*has_lpc_leadin = 0;
if (s->subframes == 4) {
int offset = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_interpolation_offset);
if (offset != 4 && frame->coded) {
*has_lpc_leadin = 1;
if (offset != 0) {
int16_t nlsf_leadin[16];
for (i = 0; i < order; i++)
nlsf_leadin[i] = frame->nlsf[i] +
((nlsf[i] - frame->nlsf[i]) * offset >> 2);
silk_lsf2lpc(nlsf_leadin, lpc_leadin, order);
} else /* avoid re-computation for a (roughly) 1-in-4 occurrence */
memcpy(lpc_leadin, frame->lpc, 16 * sizeof(float));
} else
offset = 4;
s->nlsf_interp_factor = offset;
silk_lsf2lpc(nlsf, lpc, order);
} else {
s->nlsf_interp_factor = 4;
silk_lsf2lpc(nlsf, lpc, order);
}
memcpy(frame->nlsf, nlsf, order * sizeof(nlsf[0]));
memcpy(frame->lpc, lpc, order * sizeof(lpc[0]));
}
static inline void silk_count_children(OpusRangeCoder *rc, int model, int32_t total,
int32_t child[2])
{
if (total != 0) {
child[0] = ff_opus_rc_dec_cdf(rc,
ff_silk_model_pulse_location[model] + (((total - 1 + 5) * (total - 1)) >> 1));
child[1] = total - child[0];
} else {
child[0] = 0;
child[1] = 0;
}
}
static inline void silk_decode_excitation(SilkContext *s, OpusRangeCoder *rc,
float* excitationf,
int qoffset_high, int active, int voiced)
{
int i;
uint32_t seed;
int shellblocks;
int ratelevel;
uint8_t pulsecount[20]; // total pulses in each shell block
uint8_t lsbcount[20] = {0}; // raw lsbits defined for each pulse in each shell block
int32_t excitation[320]; // Q23
/* excitation parameters */
seed = ff_opus_rc_dec_cdf(rc, ff_silk_model_lcg_seed);
shellblocks = ff_silk_shell_blocks[s->bandwidth][s->subframes >> 2];
ratelevel = ff_opus_rc_dec_cdf(rc, ff_silk_model_exc_rate[voiced]);
for (i = 0; i < shellblocks; i++) {
pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[ratelevel]);
if (pulsecount[i] == 17) {
while (pulsecount[i] == 17 && ++lsbcount[i] != 10)
pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[9]);
if (lsbcount[i] == 10)
pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[10]);
}
}
/* decode pulse locations using PVQ */
for (i = 0; i < shellblocks; i++) {
if (pulsecount[i] != 0) {
int a, b, c, d;
int32_t * location = excitation + 16*i;
int32_t branch[4][2];
branch[0][0] = pulsecount[i];
/* unrolled tail recursion */
for (a = 0; a < 1; a++) {
silk_count_children(rc, 0, branch[0][a], branch[1]);
for (b = 0; b < 2; b++) {
silk_count_children(rc, 1, branch[1][b], branch[2]);
for (c = 0; c < 2; c++) {
silk_count_children(rc, 2, branch[2][c], branch[3]);
for (d = 0; d < 2; d++) {
silk_count_children(rc, 3, branch[3][d], location);
location += 2;
}
}
}
}
} else
memset(excitation + 16*i, 0, 16*sizeof(int32_t));
}
/* decode least significant bits */
for (i = 0; i < shellblocks << 4; i++) {
int bit;
for (bit = 0; bit < lsbcount[i >> 4]; bit++)
excitation[i] = (excitation[i] << 1) |
ff_opus_rc_dec_cdf(rc, ff_silk_model_excitation_lsb);
}
/* decode signs */
for (i = 0; i < shellblocks << 4; i++) {
if (excitation[i] != 0) {
int sign = ff_opus_rc_dec_cdf(rc, ff_silk_model_excitation_sign[active +
voiced][qoffset_high][FFMIN(pulsecount[i >> 4], 6)]);
if (sign == 0)
excitation[i] *= -1;
}
}
/* assemble the excitation */
for (i = 0; i < shellblocks << 4; i++) {
int value = excitation[i];
excitation[i] = value * 256 | ff_silk_quant_offset[voiced][qoffset_high];
if (value < 0) excitation[i] += 20;
else if (value > 0) excitation[i] -= 20;
/* invert samples pseudorandomly */
seed = 196314165 * seed + 907633515;
if (seed & 0x80000000)
excitation[i] *= -1;
seed += value;
excitationf[i] = excitation[i] / 8388608.0f;
}
}
/** Maximum residual history according to 4.2.7.6.1 */
#define SILK_MAX_LAG (288 + LTP_ORDER / 2)
/** Order of the LTP filter */
#define LTP_ORDER 5
static void silk_decode_frame(SilkContext *s, OpusRangeCoder *rc,
int frame_num, int channel, int coded_channels,
int active, int active1, int redundant)
{
/* per frame */
int voiced; // combines with active to indicate inactive, active, or active+voiced
int qoffset_high;
int order; // order of the LPC coefficients
float lpc_leadin[16], lpc_body[16], residual[SILK_MAX_LAG + SILK_HISTORY];
int has_lpc_leadin;
float ltpscale;
/* per subframe */
struct {
float gain;
int pitchlag;
float ltptaps[5];
} sf[4];
SilkFrame * const frame = s->frame + channel;
int i;
/* obtain stereo weights */
if (coded_channels == 2 && channel == 0) {
int n, wi[2], ws[2], w[2];
n = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s1);
wi[0] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s2) + 3 * (n / 5);
ws[0] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s3);
wi[1] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s2) + 3 * (n % 5);
ws[1] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s3);
for (i = 0; i < 2; i++)
w[i] = ff_silk_stereo_weights[wi[i]] +
(((ff_silk_stereo_weights[wi[i] + 1] - ff_silk_stereo_weights[wi[i]]) * 6554) >> 16)
* (ws[i]*2 + 1);
s->stereo_weights[0] = (w[0] - w[1]) / 8192.0;
s->stereo_weights[1] = w[1] / 8192.0;
/* and read the mid-only flag */
s->midonly = active1 ? 0 : ff_opus_rc_dec_cdf(rc, ff_silk_model_mid_only);
}
/* obtain frame type */
if (!active) {
qoffset_high = ff_opus_rc_dec_cdf(rc, ff_silk_model_frame_type_inactive);
voiced = 0;
} else {
int type = ff_opus_rc_dec_cdf(rc, ff_silk_model_frame_type_active);
qoffset_high = type & 1;
voiced = type >> 1;
}
/* obtain subframe quantization gains */
for (i = 0; i < s->subframes; i++) {
int log_gain; //Q7
int ipart, fpart, lingain;
if (i == 0 && (frame_num == 0 || !frame->coded)) {
/* gain is coded absolute */
int x = ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_highbits[active + voiced]);
log_gain = (x<<3) | ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_lowbits);
if (frame->coded)
log_gain = FFMAX(log_gain, frame->log_gain - 16);
} else {
/* gain is coded relative */
int delta_gain = ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_delta);
log_gain = av_clip_uintp2(FFMAX((delta_gain<<1) - 16,
frame->log_gain + delta_gain - 4), 6);
}
frame->log_gain = log_gain;
/* approximate 2**(x/128) with a Q7 (i.e. non-integer) input */
log_gain = (log_gain * 0x1D1C71 >> 16) + 2090;
ipart = log_gain >> 7;
fpart = log_gain & 127;
lingain = (1 << ipart) + ((-174 * fpart * (128-fpart) >>16) + fpart) * ((1<<ipart) >> 7);
sf[i].gain = lingain / 65536.0f;
}
/* obtain LPC filter coefficients */
silk_decode_lpc(s, frame, rc, lpc_leadin, lpc_body, &order, &has_lpc_leadin, voiced);
/* obtain pitch lags, if this is a voiced frame */
if (voiced) {
int lag_absolute = (!frame_num || !frame->prev_voiced);
int primarylag; // primary pitch lag for the entire SILK frame
int ltpfilter;
const int8_t * offsets;
if (!lag_absolute) {
int delta = ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_delta);
if (delta)
primarylag = frame->primarylag + delta - 9;
else
lag_absolute = 1;
}
if (lag_absolute) {
/* primary lag is coded absolute */
int highbits, lowbits;
static const uint16_t * const model[] = {
ff_silk_model_pitch_lowbits_nb, ff_silk_model_pitch_lowbits_mb,
ff_silk_model_pitch_lowbits_wb
};
highbits = ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_highbits);
lowbits = ff_opus_rc_dec_cdf(rc, model[s->bandwidth]);
primarylag = ff_silk_pitch_min_lag[s->bandwidth] +
highbits*ff_silk_pitch_scale[s->bandwidth] + lowbits;
}
frame->primarylag = primarylag;
if (s->subframes == 2)
offsets = (s->bandwidth == OPUS_BANDWIDTH_NARROWBAND)
? ff_silk_pitch_offset_nb10ms[ff_opus_rc_dec_cdf(rc,
ff_silk_model_pitch_contour_nb10ms)]
: ff_silk_pitch_offset_mbwb10ms[ff_opus_rc_dec_cdf(rc,
ff_silk_model_pitch_contour_mbwb10ms)];
else
offsets = (s->bandwidth == OPUS_BANDWIDTH_NARROWBAND)
? ff_silk_pitch_offset_nb20ms[ff_opus_rc_dec_cdf(rc,
ff_silk_model_pitch_contour_nb20ms)]
: ff_silk_pitch_offset_mbwb20ms[ff_opus_rc_dec_cdf(rc,
ff_silk_model_pitch_contour_mbwb20ms)];
for (i = 0; i < s->subframes; i++)
sf[i].pitchlag = av_clip(primarylag + offsets[i],
ff_silk_pitch_min_lag[s->bandwidth],
ff_silk_pitch_max_lag[s->bandwidth]);
/* obtain LTP filter coefficients */
ltpfilter = ff_opus_rc_dec_cdf(rc, ff_silk_model_ltp_filter);
for (i = 0; i < s->subframes; i++) {
int index, j;
static const uint16_t * const filter_sel[] = {
ff_silk_model_ltp_filter0_sel, ff_silk_model_ltp_filter1_sel,
ff_silk_model_ltp_filter2_sel
};
static const int8_t (* const filter_taps[])[5] = {
ff_silk_ltp_filter0_taps, ff_silk_ltp_filter1_taps, ff_silk_ltp_filter2_taps
};
index = ff_opus_rc_dec_cdf(rc, filter_sel[ltpfilter]);
for (j = 0; j < 5; j++)
sf[i].ltptaps[j] = filter_taps[ltpfilter][index][j] / 128.0f;
}
}
/* obtain LTP scale factor */
if (voiced && frame_num == 0)
ltpscale = ff_silk_ltp_scale_factor[ff_opus_rc_dec_cdf(rc,
ff_silk_model_ltp_scale_index)] / 16384.0f;
else ltpscale = 15565.0f/16384.0f;
/* generate the excitation signal for the entire frame */
silk_decode_excitation(s, rc, residual + SILK_MAX_LAG, qoffset_high,
active, voiced);
/* skip synthesising the output if we do not need it */
// TODO: implement error recovery
if (s->output_channels == channel || redundant)
return;
/* generate the output signal */
for (i = 0; i < s->subframes; i++) {
const float * lpc_coeff = (i < 2 && has_lpc_leadin) ? lpc_leadin : lpc_body;
float *dst = frame->output + SILK_HISTORY + i * s->sflength;
float *resptr = residual + SILK_MAX_LAG + i * s->sflength;
float *lpc = frame->lpc_history + SILK_HISTORY + i * s->sflength;
float sum;
int j, k;
if (voiced) {
int out_end;
float scale;
if (i < 2 || s->nlsf_interp_factor == 4) {
out_end = -i * s->sflength;
scale = ltpscale;
} else {
out_end = -(i - 2) * s->sflength;
scale = 1.0f;
}
/* when the LPC coefficients change, a re-whitening filter is used */
/* to produce a residual that accounts for the change */
for (j = - sf[i].pitchlag - LTP_ORDER/2; j < out_end; j++) {
sum = dst[j];
for (k = 0; k < order; k++)
sum -= lpc_coeff[k] * dst[j - k - 1];
resptr[j] = av_clipf(sum, -1.0f, 1.0f) * scale / sf[i].gain;
}
if (out_end) {
float rescale = sf[i-1].gain / sf[i].gain;
for (j = out_end; j < 0; j++)
resptr[j] *= rescale;
}
/* LTP synthesis */
for (j = 0; j < s->sflength; j++) {
sum = resptr[j];
for (k = 0; k < LTP_ORDER; k++)
sum += sf[i].ltptaps[k] * resptr[j - sf[i].pitchlag + LTP_ORDER/2 - k];
resptr[j] = sum;
}
}
/* LPC synthesis */
for (j = 0; j < s->sflength; j++) {
sum = resptr[j] * sf[i].gain;
for (k = 1; k <= order; k++)
sum += lpc_coeff[k - 1] * lpc[j - k];
lpc[j] = sum;
dst[j] = av_clipf(sum, -1.0f, 1.0f);
}
}
frame->prev_voiced = voiced;
memmove(frame->lpc_history, frame->lpc_history + s->flength, SILK_HISTORY * sizeof(float));
memmove(frame->output, frame->output + s->flength, SILK_HISTORY * sizeof(float));
frame->coded = 1;
}
static void silk_unmix_ms(SilkContext *s, float *l, float *r)
{
float *mid = s->frame[0].output + SILK_HISTORY - s->flength;
float *side = s->frame[1].output + SILK_HISTORY - s->flength;
float w0_prev = s->prev_stereo_weights[0];
float w1_prev = s->prev_stereo_weights[1];
float w0 = s->stereo_weights[0];
float w1 = s->stereo_weights[1];
int n1 = ff_silk_stereo_interp_len[s->bandwidth];
int i;
for (i = 0; i < n1; i++) {
float interp0 = w0_prev + i * (w0 - w0_prev) / n1;
float interp1 = w1_prev + i * (w1 - w1_prev) / n1;
float p0 = 0.25 * (mid[i - 2] + 2 * mid[i - 1] + mid[i]);
l[i] = av_clipf((1 + interp1) * mid[i - 1] + side[i - 1] + interp0 * p0, -1.0, 1.0);
r[i] = av_clipf((1 - interp1) * mid[i - 1] - side[i - 1] - interp0 * p0, -1.0, 1.0);
}
for (; i < s->flength; i++) {
float p0 = 0.25 * (mid[i - 2] + 2 * mid[i - 1] + mid[i]);
l[i] = av_clipf((1 + w1) * mid[i - 1] + side[i - 1] + w0 * p0, -1.0, 1.0);
r[i] = av_clipf((1 - w1) * mid[i - 1] - side[i - 1] - w0 * p0, -1.0, 1.0);
}
memcpy(s->prev_stereo_weights, s->stereo_weights, sizeof(s->stereo_weights));
}
static void silk_flush_frame(SilkFrame *frame)
{
if (!frame->coded)
return;
memset(frame->output, 0, sizeof(frame->output));
memset(frame->lpc_history, 0, sizeof(frame->lpc_history));
memset(frame->lpc, 0, sizeof(frame->lpc));
memset(frame->nlsf, 0, sizeof(frame->nlsf));
frame->log_gain = 0;
frame->primarylag = 0;
frame->prev_voiced = 0;
frame->coded = 0;
}
int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc,
float *output[2],
enum OpusBandwidth bandwidth,
int coded_channels,
int duration_ms)
{
int active[2][6], redundancy[2];
int nb_frames, i, j;
if (bandwidth > OPUS_BANDWIDTH_WIDEBAND ||
coded_channels > 2 || duration_ms > 60) {
av_log(s->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);
}
/* read the per-frame LBRR flags */
for (i = 0; i < coded_channels; i++)
if (redundancy[i] && duration_ms > 20) {
redundancy[i] = ff_opus_rc_dec_cdf(rc, duration_ms == 40 ?
ff_silk_model_lbrr_flags_40 : ff_silk_model_lbrr_flags_60);
}
/* decode the LBRR frames */
for (i = 0; i < nb_frames; i++) {
for (j = 0; j < coded_channels; j++)
if (redundancy[j] & (1 << i)) {
int active1 = (j == 0 && !(redundancy[1] & (1 << i))) ? 0 : 1;
silk_decode_frame(s, rc, i, j, coded_channels, 1, active1, 1);
}
}
for (i = 0; i < nb_frames; i++) {
for (j = 0; j < coded_channels && !s->midonly; j++)
silk_decode_frame(s, rc, i, j, coded_channels, active[j][i], active[1][i], 0);
/* reset the side channel if it is not coded */
if (s->midonly && s->frame[1].coded)
silk_flush_frame(&s->frame[1]);
if (coded_channels == 1 || s->output_channels == 1) {
for (j = 0; j < s->output_channels; j++) {
memcpy(output[j] + i * s->flength,
s->frame[0].output + SILK_HISTORY - s->flength - 2,
s->flength * sizeof(float));
}
} else {
silk_unmix_ms(s, output[0] + i * s->flength, output[1] + i * s->flength);
}
s->midonly = 0;
}
return nb_frames * s->flength;
}
void ff_silk_free(SilkContext **ps)
{
av_freep(ps);
}
void ff_silk_flush(SilkContext *s)
{
silk_flush_frame(&s->frame[0]);
silk_flush_frame(&s->frame[1]);
memset(s->prev_stereo_weights, 0, sizeof(s->prev_stereo_weights));
}
int ff_silk_init(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;
}
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