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FFmpeg/libavcodec/g723_1dec.c
Anton Khirnov 08bebeb1be Revert "all: Don't set AVClass.item_name to its default value" 
Some callers assume that item_name is always set, so this may be
considered an API break.

This reverts commit 0c6203c97a.
2024-01-20 10:34:48 +01:00

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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 "codec_internal.h"
#include "decode.h"
#include "get_bits.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",
CODEC_LONG_NAME("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 =
#if FF_API_SUBFRAMES
AV_CODEC_CAP_SUBFRAMES |
#endif
AV_CODEC_CAP_DR1,
.p.priv_class = &g723_1dec_class,
};
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