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ffmpeg/libavcodec/g723_1dec.c
Andreas Rheinhardt 4243da4ff4 avcodec/codec_internal: Use union for FFCodec decode/encode callbacks 
This is possible, because every given FFCodec has to implement
exactly one of these. Doing so decreases sizeof(FFCodec) and
therefore decreases the size of the binary.
Notice that in case of position-independent code the decrease
is in .data.rel.ro, so that this translates to decreased
memory consumption.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-04-05 20:02:37 +02:00

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/*
* G.723.1 compatible decoder
* Copyright (c) 2006 Benjamin Larsson
* Copyright (c) 2010 Mohamed Naufal Basheer
*
* 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
* G.723.1 compatible decoder
*/
#include "libavutil/channel_layout.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#define BITSTREAM_READER_LE
#include "acelp_vectors.h"
#include "avcodec.h"
#include "celp_filters.h"
#include "celp_math.h"
#include "codec_internal.h"
#include "get_bits.h"
#include "internal.h"
#include "g723_1.h"
#define CNG_RANDOM_SEED 12345
/**
* Postfilter gain weighting factors scaled by 2^15
*/
static const int16_t ppf_gain_weight[2] = {0x1800, 0x2000};
static const int16_t pitch_contrib[340] = {
60, 0, 0, 2489, 60, 0, 0, 5217,
1, 6171, 0, 3953, 0, 10364, 1, 9357,
-1, 8843, 1, 9396, 0, 5794, -1, 10816,
2, 11606, -2, 12072, 0, 8616, 1, 12170,
0, 14440, 0, 7787, -1, 13721, 0, 18205,
0, 14471, 0, 15807, 1, 15275, 0, 13480,
-1, 18375, -1, 0, 1, 11194, -1, 13010,
1, 18836, -2, 20354, 1, 16233, -1, 0,
60, 0, 0, 12130, 0, 13385, 1, 17834,
1, 20875, 0, 21996, 1, 0, 1, 18277,
-1, 21321, 1, 13738, -1, 19094, -1, 20387,
-1, 0, 0, 21008, 60, 0, -2, 22807,
0, 15900, 1, 0, 0, 17989, -1, 22259,
1, 24395, 1, 23138, 0, 23948, 1, 22997,
2, 22604, -1, 25942, 0, 26246, 1, 25321,
0, 26423, 0, 24061, 0, 27247, 60, 0,
-1, 25572, 1, 23918, 1, 25930, 2, 26408,
-1, 19049, 1, 27357, -1, 24538, 60, 0,
-1, 25093, 0, 28549, 1, 0, 0, 22793,
-1, 25659, 0, 29377, 0, 30276, 0, 26198,
1, 22521, -1, 28919, 0, 27384, 1, 30162,
-1, 0, 0, 24237, -1, 30062, 0, 21763,
1, 30917, 60, 0, 0, 31284, 0, 29433,
1, 26821, 1, 28655, 0, 31327, 2, 30799,
1, 31389, 0, 32322, 1, 31760, -2, 31830,
0, 26936, -1, 31180, 1, 30875, 0, 27873,
-1, 30429, 1, 31050, 0, 0, 0, 31912,
1, 31611, 0, 31565, 0, 25557, 0, 31357,
60, 0, 1, 29536, 1, 28985, -1, 26984,
-1, 31587, 2, 30836, -2, 31133, 0, 30243,
-1, 30742, -1, 32090, 60, 0, 2, 30902,
60, 0, 0, 30027, 0, 29042, 60, 0,
0, 31756, 0, 24553, 0, 25636, -2, 30501,
60, 0, -1, 29617, 0, 30649, 60, 0,
0, 29274, 2, 30415, 0, 27480, 0, 31213,
-1, 28147, 0, 30600, 1, 31652, 2, 29068,
60, 0, 1, 28571, 1, 28730, 1, 31422,
0, 28257, 0, 24797, 60, 0, 0, 0,
60, 0, 0, 22105, 0, 27852, 60, 0,
60, 0, -1, 24214, 0, 24642, 0, 23305,
60, 0, 60, 0, 1, 22883, 0, 21601,
60, 0, 2, 25650, 60, 0, -2, 31253,
-2, 25144, 0, 17998
};
/**
* Size of the MP-MLQ fixed excitation codebooks
*/
static const int32_t max_pos[4] = {593775, 142506, 593775, 142506};
/**
* 0.65^i (Zero part) and 0.75^i (Pole part) scaled by 2^15
*/
static const int16_t postfilter_tbl[2][LPC_ORDER] = {
/* Zero */
{21299, 13844, 8999, 5849, 3802, 2471, 1606, 1044, 679, 441},
/* Pole */
{24576, 18432, 13824, 10368, 7776, 5832, 4374, 3281, 2460, 1845}
};
static const int cng_adaptive_cb_lag[4] = { 1, 0, 1, 3 };
static const int cng_filt[4] = { 273, 998, 499, 333 };
static const int cng_bseg[3] = { 2048, 18432, 231233 };
static av_cold int g723_1_decode_init(AVCodecContext *avctx)
{
G723_1_Context *s = avctx->priv_data;
avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
if (avctx->ch_layout.nb_channels < 1 || avctx->ch_layout.nb_channels > 2) {
av_log(avctx, AV_LOG_ERROR, "Only mono and stereo are supported (requested channels: %d).\n",
avctx->ch_layout.nb_channels);
return AVERROR(EINVAL);
}
for (int ch = 0; ch < avctx->ch_layout.nb_channels; ch++) {
G723_1_ChannelContext *p = &s->ch[ch];
p->pf_gain = 1 << 12;
memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
memcpy(p->sid_lsp, dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp));
p->cng_random_seed = CNG_RANDOM_SEED;
p->past_frame_type = SID_FRAME;
}
return 0;
}
/**
* Unpack the frame into parameters.
*
* @param p the context
* @param buf pointer to the input buffer
* @param buf_size size of the input buffer
*/
static int unpack_bitstream(G723_1_ChannelContext *p, const uint8_t *buf,
int buf_size)
{
GetBitContext gb;
int ad_cb_len;
int temp, info_bits, i;
int ret;
ret = init_get_bits8(&gb, buf, buf_size);
if (ret < 0)
return ret;
/* Extract frame type and rate info */
info_bits = get_bits(&gb, 2);
if (info_bits == 3) {
p->cur_frame_type = UNTRANSMITTED_FRAME;
return 0;
}
/* Extract 24 bit lsp indices, 8 bit for each band */
p->lsp_index[2] = get_bits(&gb, 8);
p->lsp_index[1] = get_bits(&gb, 8);
p->lsp_index[0] = get_bits(&gb, 8);
if (info_bits == 2) {
p->cur_frame_type = SID_FRAME;
p->subframe[0].amp_index = get_bits(&gb, 6);
return 0;
}
/* Extract the info common to both rates */
p->cur_rate = info_bits ? RATE_5300 : RATE_6300;
p->cur_frame_type = ACTIVE_FRAME;
p->pitch_lag[0] = get_bits(&gb, 7);
if (p->pitch_lag[0] > 123) /* test if forbidden code */
return -1;
p->pitch_lag[0] += PITCH_MIN;
p->subframe[1].ad_cb_lag = get_bits(&gb, 2);
p->pitch_lag[1] = get_bits(&gb, 7);
if (p->pitch_lag[1] > 123)
return -1;
p->pitch_lag[1] += PITCH_MIN;
p->subframe[3].ad_cb_lag = get_bits(&gb, 2);
p->subframe[0].ad_cb_lag = 1;
p->subframe[2].ad_cb_lag = 1;
for (i = 0; i < SUBFRAMES; i++) {
/* Extract combined gain */
temp = get_bits(&gb, 12);
ad_cb_len = 170;
p->subframe[i].dirac_train = 0;
if (p->cur_rate == RATE_6300 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) {
p->subframe[i].dirac_train = temp >> 11;
temp &= 0x7FF;
ad_cb_len = 85;
}
p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS);
if (p->subframe[i].ad_cb_gain < ad_cb_len) {
p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain *
GAIN_LEVELS;
} else {
return -1;
}
}
p->subframe[0].grid_index = get_bits1(&gb);
p->subframe[1].grid_index = get_bits1(&gb);
p->subframe[2].grid_index = get_bits1(&gb);
p->subframe[3].grid_index = get_bits1(&gb);
if (p->cur_rate == RATE_6300) {
skip_bits1(&gb); /* skip reserved bit */
/* Compute pulse_pos index using the 13-bit combined position index */
temp = get_bits(&gb, 13);
p->subframe[0].pulse_pos = temp / 810;
temp -= p->subframe[0].pulse_pos * 810;
p->subframe[1].pulse_pos = FASTDIV(temp, 90);
temp -= p->subframe[1].pulse_pos * 90;
p->subframe[2].pulse_pos = FASTDIV(temp, 9);
p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9;
p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) +
get_bits(&gb, 16);
p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) +
get_bits(&gb, 14);
p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) +
get_bits(&gb, 16);
p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) +
get_bits(&gb, 14);
p->subframe[0].pulse_sign = get_bits(&gb, 6);
p->subframe[1].pulse_sign = get_bits(&gb, 5);
p->subframe[2].pulse_sign = get_bits(&gb, 6);
p->subframe[3].pulse_sign = get_bits(&gb, 5);
} else { /* 5300 bps */
p->subframe[0].pulse_pos = get_bits(&gb, 12);
p->subframe[1].pulse_pos = get_bits(&gb, 12);
p->subframe[2].pulse_pos = get_bits(&gb, 12);
p->subframe[3].pulse_pos = get_bits(&gb, 12);
p->subframe[0].pulse_sign = get_bits(&gb, 4);
p->subframe[1].pulse_sign = get_bits(&gb, 4);
p->subframe[2].pulse_sign = get_bits(&gb, 4);
p->subframe[3].pulse_sign = get_bits(&gb, 4);
}
return 0;
}
/**
* Bitexact implementation of sqrt(val/2).
*/
static int16_t square_root(unsigned val)
{
av_assert2(!(val & 0x80000000));
return (ff_sqrt(val << 1) >> 1) & (~1);
}
/**
* Generate fixed codebook excitation vector.
*
* @param vector decoded excitation vector
* @param subfrm current subframe
* @param cur_rate current bitrate
* @param pitch_lag closed loop pitch lag
* @param index current subframe index
*/
static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
enum Rate cur_rate, int pitch_lag, int index)
{
int temp, i, j;
memset(vector, 0, SUBFRAME_LEN * sizeof(*vector));
if (cur_rate == RATE_6300) {
if (subfrm->pulse_pos >= max_pos[index])
return;
/* Decode amplitudes and positions */
j = PULSE_MAX - pulses[index];
temp = subfrm->pulse_pos;
for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {
temp -= ff_g723_1_combinatorial_table[j][i];
if (temp >= 0)
continue;
temp += ff_g723_1_combinatorial_table[j++][i];
if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) {
vector[subfrm->grid_index + GRID_SIZE * i] =
-ff_g723_1_fixed_cb_gain[subfrm->amp_index];
} else {
vector[subfrm->grid_index + GRID_SIZE * i] =
ff_g723_1_fixed_cb_gain[subfrm->amp_index];
}
if (j == PULSE_MAX)
break;
}
if (subfrm->dirac_train == 1)
ff_g723_1_gen_dirac_train(vector, pitch_lag);
} else { /* 5300 bps */
int cb_gain = ff_g723_1_fixed_cb_gain[subfrm->amp_index];
int cb_shift = subfrm->grid_index;
int cb_sign = subfrm->pulse_sign;
int cb_pos = subfrm->pulse_pos;
int offset, beta, lag;
for (i = 0; i < 8; i += 2) {
offset = ((cb_pos & 7) << 3) + cb_shift + i;
vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain;
cb_pos >>= 3;
cb_sign >>= 1;
}
/* Enhance harmonic components */
lag = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag +
subfrm->ad_cb_lag - 1;
beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1];
if (lag < SUBFRAME_LEN - 2) {
for (i = lag; i < SUBFRAME_LEN; i++)
vector[i] += beta * vector[i - lag] >> 15;
}
}
}
/**
* Estimate maximum auto-correlation around pitch lag.
*
* @param buf buffer with offset applied
* @param offset offset of the excitation vector
* @param ccr_max pointer to the maximum auto-correlation
* @param pitch_lag decoded pitch lag
* @param length length of autocorrelation
* @param dir forward lag(1) / backward lag(-1)
*/
static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,
int pitch_lag, int length, int dir)
{
int limit, ccr, lag = 0;
int i;
pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);
if (dir > 0)
limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);
else
limit = pitch_lag + 3;
for (i = pitch_lag - 3; i <= limit; i++) {
ccr = ff_g723_1_dot_product(buf, buf + dir * i, length);
if (ccr > *ccr_max) {
*ccr_max = ccr;
lag = i;
}
}
return lag;
}
/**
* Calculate pitch postfilter optimal and scaling gains.
*
* @param lag pitch postfilter forward/backward lag
* @param ppf pitch postfilter parameters
* @param cur_rate current bitrate
* @param tgt_eng target energy
* @param ccr cross-correlation
* @param res_eng residual energy
*/
static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate,
int tgt_eng, int ccr, int res_eng)
{
int pf_residual; /* square of postfiltered residual */
int temp1, temp2;
ppf->index = lag;
temp1 = tgt_eng * res_eng >> 1;
temp2 = ccr * ccr << 1;
if (temp2 > temp1) {
if (ccr >= res_eng) {
ppf->opt_gain = ppf_gain_weight[cur_rate];
} else {
ppf->opt_gain = (ccr << 15) / res_eng *
ppf_gain_weight[cur_rate] >> 15;
}
/* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */
temp1 = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1);
temp2 = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng;
pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16;
if (tgt_eng >= pf_residual << 1) {
temp1 = 0x7fff;
} else {
temp1 = (tgt_eng << 14) / pf_residual;
}
/* scaling_gain = sqrt(tgt_eng/pf_res^2) */
ppf->sc_gain = square_root(temp1 << 16);
} else {
ppf->opt_gain = 0;
ppf->sc_gain = 0x7fff;
}
ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15);
}
/**
* Calculate pitch postfilter parameters.
*
* @param p the context
* @param offset offset of the excitation vector
* @param pitch_lag decoded pitch lag
* @param ppf pitch postfilter parameters
* @param cur_rate current bitrate
*/
static void comp_ppf_coeff(G723_1_ChannelContext *p, int offset, int pitch_lag,
PPFParam *ppf, enum Rate cur_rate)
{
int16_t scale;
int i;
int temp1, temp2;
/*
* 0 - target energy
* 1 - forward cross-correlation
* 2 - forward residual energy
* 3 - backward cross-correlation
* 4 - backward residual energy
*/
int energy[5] = {0, 0, 0, 0, 0};
int16_t *buf = p->audio + LPC_ORDER + offset;
int fwd_lag = autocorr_max(buf, offset, &energy[1], pitch_lag,
SUBFRAME_LEN, 1);
int back_lag = autocorr_max(buf, offset, &energy[3], pitch_lag,
SUBFRAME_LEN, -1);
ppf->index = 0;
ppf->opt_gain = 0;
ppf->sc_gain = 0x7fff;
/* Case 0, Section 3.6 */
if (!back_lag && !fwd_lag)
return;
/* Compute target energy */
energy[0] = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN);
/* Compute forward residual energy */
if (fwd_lag)
energy[2] = ff_g723_1_dot_product(buf + fwd_lag, buf + fwd_lag,
SUBFRAME_LEN);
/* Compute backward residual energy */
if (back_lag)
energy[4] = ff_g723_1_dot_product(buf - back_lag, buf - back_lag,
SUBFRAME_LEN);
/* Normalize and shorten */
temp1 = 0;
for (i = 0; i < 5; i++)
temp1 = FFMAX(energy[i], temp1);
scale = ff_g723_1_normalize_bits(temp1, 31);
for (i = 0; i < 5; i++)
energy[i] = (energy[i] << scale) >> 16;
if (fwd_lag && !back_lag) { /* Case 1 */
comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
energy[2]);
} else if (!fwd_lag) { /* Case 2 */
comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
energy[4]);
} else { /* Case 3 */
/*
* Select the largest of energy[1]^2/energy[2]
* and energy[3]^2/energy[4]
*/
temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15);
temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15);
if (temp1 >= temp2) {
comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
energy[2]);
} else {
comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
energy[4]);
}
}
}
/**
* Classify frames as voiced/unvoiced.
*
* @param p the context
* @param pitch_lag decoded pitch_lag
* @param exc_eng excitation energy estimation
* @param scale scaling factor of exc_eng
*
* @return residual interpolation index if voiced, 0 otherwise
*/
static int comp_interp_index(G723_1_ChannelContext *p, int pitch_lag,
int *exc_eng, int *scale)
{
int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
int16_t *buf = p->audio + LPC_ORDER;
int index, ccr, tgt_eng, best_eng, temp;
*scale = ff_g723_1_scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX);
buf += offset;
/* Compute maximum backward cross-correlation */
ccr = 0;
index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
ccr = av_sat_add32(ccr, 1 << 15) >> 16;
/* Compute target energy */
tgt_eng = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN * 2);
*exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16;
if (ccr <= 0)
return 0;
/* Compute best energy */
best_eng = ff_g723_1_dot_product(buf - index, buf - index,
SUBFRAME_LEN * 2);
best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;
temp = best_eng * *exc_eng >> 3;
if (temp < ccr * ccr) {
return index;
} else
return 0;
}
/**
* Perform residual interpolation based on frame classification.
*
* @param buf decoded excitation vector
* @param out output vector
* @param lag decoded pitch lag
* @param gain interpolated gain
* @param rseed seed for random number generator
*/
static void residual_interp(int16_t *buf, int16_t *out, int lag,
int gain, int *rseed)
{
int i;
if (lag) { /* Voiced */
int16_t *vector_ptr = buf + PITCH_MAX;
/* Attenuate */
for (i = 0; i < lag; i++)
out[i] = vector_ptr[i - lag] * 3 >> 2;
av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out),
(FRAME_LEN - lag) * sizeof(*out));
} else { /* Unvoiced */
for (i = 0; i < FRAME_LEN; i++) {
*rseed = (int16_t)(*rseed * 521 + 259);
out[i] = gain * *rseed >> 15;
}
memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(*buf));
}
}
/**
* Perform IIR filtering.
*
* @param fir_coef FIR coefficients
* @param iir_coef IIR coefficients
* @param src source vector
* @param dest destination vector
* @param width width of the output, 16 bits(0) / 32 bits(1)
*/
#define iir_filter(fir_coef, iir_coef, src, dest, width)\
{\
int m, n;\
int res_shift = 16 & ~-(width);\
int in_shift = 16 - res_shift;\
\
for (m = 0; m < SUBFRAME_LEN; m++) {\
int64_t filter = 0;\
for (n = 1; n <= LPC_ORDER; n++) {\
filter -= (fir_coef)[n - 1] * (src)[m - n] -\
(iir_coef)[n - 1] * ((dest)[m - n] >> in_shift);\
}\
\
(dest)[m] = av_clipl_int32(((src)[m] * 65536) + (filter * 8) +\
(1 << 15)) >> res_shift;\
}\
}
/**
* Adjust gain of postfiltered signal.
*
* @param p the context
* @param buf postfiltered output vector
* @param energy input energy coefficient
*/
static void gain_scale(G723_1_ChannelContext *p, int16_t * buf, int energy)
{
int num, denom, gain, bits1, bits2;
int i;
num = energy;
denom = 0;
for (i = 0; i < SUBFRAME_LEN; i++) {
int temp = buf[i] >> 2;
temp *= temp;
denom = av_sat_dadd32(denom, temp);
}
if (num && denom) {
bits1 = ff_g723_1_normalize_bits(num, 31);
bits2 = ff_g723_1_normalize_bits(denom, 31);
num = num << bits1 >> 1;
denom <<= bits2;
bits2 = 5 + bits1 - bits2;
bits2 = av_clip_uintp2(bits2, 5);
gain = (num >> 1) / (denom >> 16);
gain = square_root(gain << 16 >> bits2);
} else {
gain = 1 << 12;
}
for (i = 0; i < SUBFRAME_LEN; i++) {
p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4;
buf[i] = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) +
(1 << 10)) >> 11);
}
}
/**
* Perform formant filtering.
*
* @param p the context
* @param lpc quantized lpc coefficients
* @param buf input buffer
* @param dst output buffer
*/
static void formant_postfilter(G723_1_ChannelContext *p, int16_t *lpc,
int16_t *buf, int16_t *dst)
{
int16_t filter_coef[2][LPC_ORDER];
int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;
int i, j, k;
memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf));
memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal));
for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
for (k = 0; k < LPC_ORDER; k++) {
filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] +
(1 << 14)) >> 15;
filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] +
(1 << 14)) >> 15;
}
iir_filter(filter_coef[0], filter_coef[1], buf + i, filter_signal + i, 1);
lpc += LPC_ORDER;
}
memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));
memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int));
buf += LPC_ORDER;
signal_ptr = filter_signal + LPC_ORDER;
for (i = 0; i < SUBFRAMES; i++) {
int temp;
int auto_corr[2];
int scale, energy;
/* Normalize */
scale = ff_g723_1_scale_vector(dst, buf, SUBFRAME_LEN);
/* Compute auto correlation coefficients */
auto_corr[0] = ff_g723_1_dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
auto_corr[1] = ff_g723_1_dot_product(dst, dst, SUBFRAME_LEN);
/* Compute reflection coefficient */
temp = auto_corr[1] >> 16;
if (temp) {
temp = (auto_corr[0] >> 2) / temp;
}
p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2;
temp = -p->reflection_coef >> 1 & ~3;
/* Compensation filter */
for (j = 0; j < SUBFRAME_LEN; j++) {
dst[j] = av_sat_dadd32(signal_ptr[j],
(signal_ptr[j - 1] >> 16) * temp) >> 16;
}
/* Compute normalized signal energy */
temp = 2 * scale + 4;
if (temp < 0) {
energy = av_clipl_int32((int64_t)auto_corr[1] << -temp);
} else
energy = auto_corr[1] >> temp;
gain_scale(p, dst, energy);
buf += SUBFRAME_LEN;
signal_ptr += SUBFRAME_LEN;
dst += SUBFRAME_LEN;
}
}
static int sid_gain_to_lsp_index(int gain)
{
if (gain < 0x10)
return gain << 6;
else if (gain < 0x20)
return gain - 8 << 7;
else
return gain - 20 << 8;
}
static inline int cng_rand(int *state, int base)
{
*state = (*state * 521 + 259) & 0xFFFF;
return (*state & 0x7FFF) * base >> 15;
}
static int estimate_sid_gain(G723_1_ChannelContext *p)
{
int i, shift, seg, seg2, t, val, val_add, x, y;
shift = 16 - p->cur_gain * 2;
if (shift > 0) {
if (p->sid_gain == 0) {
t = 0;
} else if (shift >= 31 || (int32_t)((uint32_t)p->sid_gain << shift) >> shift != p->sid_gain) {
if (p->sid_gain < 0) t = INT32_MIN;
else t = INT32_MAX;
} else
t = p->sid_gain * (1 << shift);
} else if(shift < -31) {
t = (p->sid_gain < 0) ? -1 : 0;
}else
t = p->sid_gain >> -shift;
x = av_clipl_int32(t * (int64_t)cng_filt[0] >> 16);
if (x >= cng_bseg[2])
return 0x3F;
if (x >= cng_bseg[1]) {
shift = 4;
seg = 3;
} else {
shift = 3;
seg = (x >= cng_bseg[0]);
}
seg2 = FFMIN(seg, 3);
val = 1 << shift;
val_add = val >> 1;
for (i = 0; i < shift; i++) {
t = seg * 32 + (val << seg2);
t *= t;
if (x >= t)
val += val_add;
else
val -= val_add;
val_add >>= 1;
}
t = seg * 32 + (val << seg2);
y = t * t - x;
if (y <= 0) {
t = seg * 32 + (val + 1 << seg2);
t = t * t - x;
val = (seg2 - 1) * 16 + val;
if (t >= y)
val++;
} else {
t = seg * 32 + (val - 1 << seg2);
t = t * t - x;
val = (seg2 - 1) * 16 + val;
if (t >= y)
val--;
}
return val;
}
static void generate_noise(G723_1_ChannelContext *p)
{
int i, j, idx, t;
int off[SUBFRAMES];
int signs[SUBFRAMES / 2 * 11], pos[SUBFRAMES / 2 * 11];
int tmp[SUBFRAME_LEN * 2];
int16_t *vector_ptr;
int64_t sum;
int b0, c, delta, x, shift;
p->pitch_lag[0] = cng_rand(&p->cng_random_seed, 21) + 123;
p->pitch_lag[1] = cng_rand(&p->cng_random_seed, 19) + 123;
for (i = 0; i < SUBFRAMES; i++) {
p->subframe[i].ad_cb_gain = cng_rand(&p->cng_random_seed, 50) + 1;
p->subframe[i].ad_cb_lag = cng_adaptive_cb_lag[i];
}
for (i = 0; i < SUBFRAMES / 2; i++) {
t = cng_rand(&p->cng_random_seed, 1 << 13);
off[i * 2] = t & 1;
off[i * 2 + 1] = ((t >> 1) & 1) + SUBFRAME_LEN;
t >>= 2;
for (j = 0; j < 11; j++) {
signs[i * 11 + j] = ((t & 1) * 2 - 1) * (1 << 14);
t >>= 1;
}
}
idx = 0;
for (i = 0; i < SUBFRAMES; i++) {
for (j = 0; j < SUBFRAME_LEN / 2; j++)
tmp[j] = j;
t = SUBFRAME_LEN / 2;
for (j = 0; j < pulses[i]; j++, idx++) {
int idx2 = cng_rand(&p->cng_random_seed, t);
pos[idx] = tmp[idx2] * 2 + off[i];
tmp[idx2] = tmp[--t];
}
}
vector_ptr = p->audio + LPC_ORDER;
memcpy(vector_ptr, p->prev_excitation,
PITCH_MAX * sizeof(*p->excitation));
for (i = 0; i < SUBFRAMES; i += 2) {
ff_g723_1_gen_acb_excitation(vector_ptr, vector_ptr,
p->pitch_lag[i >> 1], &p->subframe[i],
p->cur_rate);
ff_g723_1_gen_acb_excitation(vector_ptr + SUBFRAME_LEN,
vector_ptr + SUBFRAME_LEN,
p->pitch_lag[i >> 1], &p->subframe[i + 1],
p->cur_rate);
t = 0;
for (j = 0; j < SUBFRAME_LEN * 2; j++)
t |= FFABS(vector_ptr[j]);
t = FFMIN(t, 0x7FFF);
if (!t) {
shift = 0;
} else {
shift = -10 + av_log2(t);
if (shift < -2)
shift = -2;
}
sum = 0;
if (shift < 0) {
for (j = 0; j < SUBFRAME_LEN * 2; j++) {
t = vector_ptr[j] * (1 << -shift);
sum += t * t;
tmp[j] = t;
}
} else {
for (j = 0; j < SUBFRAME_LEN * 2; j++) {
t = vector_ptr[j] >> shift;
sum += t * t;
tmp[j] = t;
}
}
b0 = 0;
for (j = 0; j < 11; j++)
b0 += tmp[pos[(i / 2) * 11 + j]] * signs[(i / 2) * 11 + j];
b0 = b0 * 2 * 2979LL + (1 << 29) >> 30; // approximated division by 11
c = p->cur_gain * (p->cur_gain * SUBFRAME_LEN >> 5);
if (shift * 2 + 3 >= 0)
c >>= shift * 2 + 3;
else
c <<= -(shift * 2 + 3);
c = (av_clipl_int32(sum << 1) - c) * 2979LL >> 15;
delta = b0 * b0 * 2 - c;
if (delta <= 0) {
x = -b0;
} else {
delta = square_root(delta);
x = delta - b0;
t = delta + b0;
if (FFABS(t) < FFABS(x))
x = -t;
}
shift++;
if (shift < 0)
x >>= -shift;
else
x *= 1 << shift;
x = av_clip(x, -10000, 10000);
for (j = 0; j < 11; j++) {
idx = (i / 2) * 11 + j;
vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] +
(x * signs[idx] >> 15));
}
/* copy decoded data to serve as a history for the next decoded subframes */
memcpy(vector_ptr + PITCH_MAX, vector_ptr,
sizeof(*vector_ptr) * SUBFRAME_LEN * 2);
vector_ptr += SUBFRAME_LEN * 2;
}
/* Save the excitation for the next frame */
memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN,
PITCH_MAX * sizeof(*p->excitation));
}
static int g723_1_decode_frame(AVCodecContext *avctx, AVFrame *frame,
int *got_frame_ptr, AVPacket *avpkt)
{
G723_1_Context *s = avctx->priv_data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int dec_mode = buf[0] & 3;
int channels = avctx->ch_layout.nb_channels;
PPFParam ppf[SUBFRAMES];
int16_t cur_lsp[LPC_ORDER];
int16_t lpc[SUBFRAMES * LPC_ORDER];
int16_t acb_vector[SUBFRAME_LEN];
int16_t *out;
int bad_frame = 0, i, j, ret;
if (buf_size < frame_size[dec_mode] * channels) {
if (buf_size)
av_log(avctx, AV_LOG_WARNING,
"Expected %d bytes, got %d - skipping packet\n",
frame_size[dec_mode], buf_size);
*got_frame_ptr = 0;
return buf_size;
}
frame->nb_samples = FRAME_LEN;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
for (int ch = 0; ch < channels; ch++) {
G723_1_ChannelContext *p = &s->ch[ch];
int16_t *audio = p->audio;
if (unpack_bitstream(p, buf + ch * (buf_size / channels),
buf_size / channels) < 0) {
bad_frame = 1;
if (p->past_frame_type == ACTIVE_FRAME)
p->cur_frame_type = ACTIVE_FRAME;
else
p->cur_frame_type = UNTRANSMITTED_FRAME;
}
out = (int16_t *)frame->extended_data[ch];
if (p->cur_frame_type == ACTIVE_FRAME) {
if (!bad_frame)
p->erased_frames = 0;
else if (p->erased_frames != 3)
p->erased_frames++;
ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
ff_g723_1_lsp_interpolate(lpc, cur_lsp, p->prev_lsp);
/* Save the lsp_vector for the next frame */
memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
/* Generate the excitation for the frame */
memcpy(p->excitation, p->prev_excitation,
PITCH_MAX * sizeof(*p->excitation));
if (!p->erased_frames) {
int16_t *vector_ptr = p->excitation + PITCH_MAX;
/* Update interpolation gain memory */
p->interp_gain = ff_g723_1_fixed_cb_gain[(p->subframe[2].amp_index +
p->subframe[3].amp_index) >> 1];
for (i = 0; i < SUBFRAMES; i++) {
gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,
p->pitch_lag[i >> 1], i);
ff_g723_1_gen_acb_excitation(acb_vector,
&p->excitation[SUBFRAME_LEN * i],
p->pitch_lag[i >> 1],
&p->subframe[i], p->cur_rate);
/* Get the total excitation */
for (j = 0; j < SUBFRAME_LEN; j++) {
int v = av_clip_int16(vector_ptr[j] * 2);
vector_ptr[j] = av_clip_int16(v + acb_vector[j]);
}
vector_ptr += SUBFRAME_LEN;
}
vector_ptr = p->excitation + PITCH_MAX;
p->interp_index = comp_interp_index(p, p->pitch_lag[1],
&p->sid_gain, &p->cur_gain);
/* Perform pitch postfiltering */
if (s->postfilter) {
i = PITCH_MAX;
for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
ppf + j, p->cur_rate);
for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
vector_ptr + i,
vector_ptr + i + ppf[j].index,
ppf[j].sc_gain,
ppf[j].opt_gain,
1 << 14, 15, SUBFRAME_LEN);
} else {
audio = vector_ptr - LPC_ORDER;
}
/* Save the excitation for the next frame */
memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
PITCH_MAX * sizeof(*p->excitation));
} else {
p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
if (p->erased_frames == 3) {
/* Mute output */
memset(p->excitation, 0,
(FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));
memset(p->prev_excitation, 0,
PITCH_MAX * sizeof(*p->excitation));
memset(frame->data[0], 0,
(FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
} else {
int16_t *buf = p->audio + LPC_ORDER;
/* Regenerate frame */
residual_interp(p->excitation, buf, p->interp_index,
p->interp_gain, &p->random_seed);
/* Save the excitation for the next frame */
memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
PITCH_MAX * sizeof(*p->excitation));
}
}
p->cng_random_seed = CNG_RANDOM_SEED;
} else {
if (p->cur_frame_type == SID_FRAME) {
p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index);
ff_g723_1_inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);
} else if (p->past_frame_type == ACTIVE_FRAME) {
p->sid_gain = estimate_sid_gain(p);
}
if (p->past_frame_type == ACTIVE_FRAME)
p->cur_gain = p->sid_gain;
else
p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
generate_noise(p);
ff_g723_1_lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);
/* Save the lsp_vector for the next frame */
memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
}
p->past_frame_type = p->cur_frame_type;
memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));
for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],
audio + i, SUBFRAME_LEN, LPC_ORDER,
0, 1, 1 << 12);
memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
if (s->postfilter) {
formant_postfilter(p, lpc, p->audio, out);
} else { // if output is not postfiltered it should be scaled by 2
for (i = 0; i < FRAME_LEN; i++)
out[i] = av_clip_int16(2 * p->audio[LPC_ORDER + i]);
}
}
*got_frame_ptr = 1;
return frame_size[dec_mode] * channels;
}
#define OFFSET(x) offsetof(G723_1_Context, x)
#define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
static const AVOption options[] = {
{ "postfilter", "enable postfilter", OFFSET(postfilter), AV_OPT_TYPE_BOOL,
{ .i64 = 1 }, 0, 1, AD },
{ NULL }
};
static const AVClass g723_1dec_class = {
.class_name = "G.723.1 decoder",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
const FFCodec ff_g723_1_decoder = {
.p.name = "g723_1",
.p.long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_G723_1,
.priv_data_size = sizeof(G723_1_Context),
.init = g723_1_decode_init,
FF_CODEC_DECODE_CB(g723_1_decode_frame),
.p.capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
.p.priv_class = &g723_1dec_class,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
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