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7f92db14f9
FFmpeg/libavcodec/g723_1.c
Kostya Shishkov 7f92db14f9 g723_1: save/restore excitation with offset to store LPC history 
The same buffer with saved data is used later in LPC reconstruction, so
it should have some head space for LPC history.
2012-08-03 07:07:06 +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 Libav.
*
* Libav 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.
*
* Libav 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 Libav; 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
*/
#define BITSTREAM_READER_LE
#include "libavutil/audioconvert.h"
#include "libavutil/lzo.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "get_bits.h"
#include "acelp_vectors.h"
#include "celp_filters.h"
#include "celp_math.h"
#include "lsp.h"
#include "g723_1_data.h"
/**
* G723.1 frame types
*/
enum FrameType {
ACTIVE_FRAME, ///< Active speech
SID_FRAME, ///< Silence Insertion Descriptor frame
UNTRANSMITTED_FRAME
};
enum Rate {
RATE_6300,
RATE_5300
};
/**
* G723.1 unpacked data subframe
*/
typedef struct {
int ad_cb_lag; ///< adaptive codebook lag
int ad_cb_gain;
int dirac_train;
int pulse_sign;
int grid_index;
int amp_index;
int pulse_pos;
} G723_1_Subframe;
/**
* Pitch postfilter parameters
*/
typedef struct {
int index; ///< postfilter backward/forward lag
int16_t opt_gain; ///< optimal gain
int16_t sc_gain; ///< scaling gain
} PPFParam;
typedef struct g723_1_context {
AVClass *class;
AVFrame frame;
G723_1_Subframe subframe[4];
enum FrameType cur_frame_type;
enum FrameType past_frame_type;
enum Rate cur_rate;
uint8_t lsp_index[LSP_BANDS];
int pitch_lag[2];
int erased_frames;
int16_t prev_lsp[LPC_ORDER];
int16_t prev_excitation[PITCH_MAX];
int16_t excitation[PITCH_MAX + FRAME_LEN];
int16_t synth_mem[LPC_ORDER];
int16_t fir_mem[LPC_ORDER];
int iir_mem[LPC_ORDER];
int random_seed;
int interp_index;
int interp_gain;
int sid_gain;
int cur_gain;
int reflection_coef;
int pf_gain;
int postfilter;
int16_t audio[FRAME_LEN + LPC_ORDER];
} G723_1_Context;
static av_cold int g723_1_decode_init(AVCodecContext *avctx)
{
G723_1_Context *p = avctx->priv_data;
avctx->channel_layout = AV_CH_LAYOUT_MONO;
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avctx->channels = 1;
avctx->sample_rate = 8000;
p->pf_gain = 1 << 12;
avcodec_get_frame_defaults(&p->frame);
avctx->coded_frame = &p->frame;
memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));
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_Context *p, const uint8_t *buf,
int buf_size)
{
GetBitContext gb;
int ad_cb_len;
int temp, info_bits, i;
init_get_bits(&gb, buf, buf_size * 8);
/* 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_bits(&gb, 1);
p->subframe[1].grid_index = get_bits(&gb, 1);
p->subframe[2].grid_index = get_bits(&gb, 1);
p->subframe[3].grid_index = get_bits(&gb, 1);
if (p->cur_rate == RATE_6300) {
skip_bits(&gb, 1); /* 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(int val)
{
int16_t res = 0;
int16_t exp = 0x4000;
int i;
for (i = 0; i < 14; i ++) {
int res_exp = res + exp;
if (val >= res_exp * res_exp << 1)
res += exp;
exp >>= 1;
}
return res;
}
/**
* Calculate the number of left-shifts required for normalizing the input.
*
* @param num input number
* @param width width of the input, 16 bits(0) / 32 bits(1)
*/
static int normalize_bits(int num, int width)
{
if (!num)
return 0;
if (num == -1)
return width;
if (num < 0)
num = ~num;
return width - av_log2(num);
}
/**
* Scale vector contents based on the largest of their absolutes.
*/
static int scale_vector(int16_t *vector, int length)
{
int bits, scale, max = 0;
int i;
for (i = 0; i < length; i++)
max = FFMAX(max, FFABS(vector[i]));
bits = normalize_bits(max, 15);
scale = (bits == 15) ? 0x7FFF : (1 << bits);
for (i = 0; i < length; i++)
vector[i] = (vector[i] * scale) >> 4;
return bits - 3;
}
/**
* Perform inverse quantization of LSP frequencies.
*
* @param cur_lsp the current LSP vector
* @param prev_lsp the previous LSP vector
* @param lsp_index VQ indices
* @param bad_frame bad frame flag
*/
static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp,
uint8_t *lsp_index, int bad_frame)
{
int min_dist, pred;
int i, j, temp, stable;
/* Check for frame erasure */
if (!bad_frame) {
min_dist = 0x100;
pred = 12288;
} else {
min_dist = 0x200;
pred = 23552;
lsp_index[0] = lsp_index[1] = lsp_index[2] = 0;
}
/* Get the VQ table entry corresponding to the transmitted index */
cur_lsp[0] = lsp_band0[lsp_index[0]][0];
cur_lsp[1] = lsp_band0[lsp_index[0]][1];
cur_lsp[2] = lsp_band0[lsp_index[0]][2];
cur_lsp[3] = lsp_band1[lsp_index[1]][0];
cur_lsp[4] = lsp_band1[lsp_index[1]][1];
cur_lsp[5] = lsp_band1[lsp_index[1]][2];
cur_lsp[6] = lsp_band2[lsp_index[2]][0];
cur_lsp[7] = lsp_band2[lsp_index[2]][1];
cur_lsp[8] = lsp_band2[lsp_index[2]][2];
cur_lsp[9] = lsp_band2[lsp_index[2]][3];
/* Add predicted vector & DC component to the previously quantized vector */
for (i = 0; i < LPC_ORDER; i++) {
temp = ((prev_lsp[i] - dc_lsp[i]) * pred + (1 << 14)) >> 15;
cur_lsp[i] += dc_lsp[i] + temp;
}
for (i = 0; i < LPC_ORDER; i++) {
cur_lsp[0] = FFMAX(cur_lsp[0], 0x180);
cur_lsp[LPC_ORDER - 1] = FFMIN(cur_lsp[LPC_ORDER - 1], 0x7e00);
/* Stability check */
for (j = 1; j < LPC_ORDER; j++) {
temp = min_dist + cur_lsp[j - 1] - cur_lsp[j];
if (temp > 0) {
temp >>= 1;
cur_lsp[j - 1] -= temp;
cur_lsp[j] += temp;
}
}
stable = 1;
for (j = 1; j < LPC_ORDER; j++) {
temp = cur_lsp[j - 1] + min_dist - cur_lsp[j] - 4;
if (temp > 0) {
stable = 0;
break;
}
}
if (stable)
break;
}
if (!stable)
memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(*cur_lsp));
}
/**
* Bitexact implementation of 2ab scaled by 1/2^16.
*
* @param a 32 bit multiplicand
* @param b 16 bit multiplier
*/
#define MULL2(a, b) \
((((a) >> 16) * (b) << 1) + (((a) & 0xffff) * (b) >> 15))
/**
* Convert LSP frequencies to LPC coefficients.
*
* @param lpc buffer for LPC coefficients
*/
static void lsp2lpc(int16_t *lpc)
{
int f1[LPC_ORDER / 2 + 1];
int f2[LPC_ORDER / 2 + 1];
int i, j;
/* Calculate negative cosine */
for (j = 0; j < LPC_ORDER; j++) {
int index = lpc[j] >> 7;
int offset = lpc[j] & 0x7f;
int64_t temp1 = cos_tab[index] << 16;
int temp2 = (cos_tab[index + 1] - cos_tab[index]) *
((offset << 8) + 0x80) << 1;
lpc[j] = -(av_clipl_int32(((temp1 + temp2) << 1) + (1 << 15)) >> 16);
}
/*
* Compute sum and difference polynomial coefficients
* (bitexact alternative to lsp2poly() in lsp.c)
*/
/* Initialize with values in Q28 */
f1[0] = 1 << 28;
f1[1] = (lpc[0] << 14) + (lpc[2] << 14);
f1[2] = lpc[0] * lpc[2] + (2 << 28);
f2[0] = 1 << 28;
f2[1] = (lpc[1] << 14) + (lpc[3] << 14);
f2[2] = lpc[1] * lpc[3] + (2 << 28);
/*
* Calculate and scale the coefficients by 1/2 in
* each iteration for a final scaling factor of Q25
*/
for (i = 2; i < LPC_ORDER / 2; i++) {
f1[i + 1] = f1[i - 1] + MULL2(f1[i], lpc[2 * i]);
f2[i + 1] = f2[i - 1] + MULL2(f2[i], lpc[2 * i + 1]);
for (j = i; j >= 2; j--) {
f1[j] = MULL2(f1[j - 1], lpc[2 * i]) +
(f1[j] >> 1) + (f1[j - 2] >> 1);
f2[j] = MULL2(f2[j - 1], lpc[2 * i + 1]) +
(f2[j] >> 1) + (f2[j - 2] >> 1);
}
f1[0] >>= 1;
f2[0] >>= 1;
f1[1] = ((lpc[2 * i] << 16 >> i) + f1[1]) >> 1;
f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1;
}
/* Convert polynomial coefficients to LPC coefficients */
for (i = 0; i < LPC_ORDER / 2; i++) {
int64_t ff1 = f1[i + 1] + f1[i];
int64_t ff2 = f2[i + 1] - f2[i];
lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) + (1 << 15)) >> 16;
lpc[LPC_ORDER - i - 1] = av_clipl_int32(((ff1 - ff2) << 3) +
(1 << 15)) >> 16;
}
}
/**
* Quantize LSP frequencies by interpolation and convert them to
* the corresponding LPC coefficients.
*
* @param lpc buffer for LPC coefficients
* @param cur_lsp the current LSP vector
* @param prev_lsp the previous LSP vector
*/
static void lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp)
{
int i;
int16_t *lpc_ptr = lpc;
/* cur_lsp * 0.25 + prev_lsp * 0.75 */
ff_acelp_weighted_vector_sum(lpc, cur_lsp, prev_lsp,
4096, 12288, 1 << 13, 14, LPC_ORDER);
ff_acelp_weighted_vector_sum(lpc + LPC_ORDER, cur_lsp, prev_lsp,
8192, 8192, 1 << 13, 14, LPC_ORDER);
ff_acelp_weighted_vector_sum(lpc + 2 * LPC_ORDER, cur_lsp, prev_lsp,
12288, 4096, 1 << 13, 14, LPC_ORDER);
memcpy(lpc + 3 * LPC_ORDER, cur_lsp, LPC_ORDER * sizeof(*lpc));
for (i = 0; i < SUBFRAMES; i++) {
lsp2lpc(lpc_ptr);
lpc_ptr += LPC_ORDER;
}
}
/**
* Generate a train of dirac functions with period as pitch lag.
*/
static void gen_dirac_train(int16_t *buf, int pitch_lag)
{
int16_t vector[SUBFRAME_LEN];
int i, j;
memcpy(vector, buf, SUBFRAME_LEN * sizeof(*vector));
for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) {
for (j = 0; j < SUBFRAME_LEN - i; j++)
buf[i + j] += vector[j];
}
}
/**
* 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 -= combinatorial_table[j][i];
if (temp >= 0)
continue;
temp += combinatorial_table[j++][i];
if (subfrm.pulse_sign & (1 << (PULSE_MAX - j))) {
vector[subfrm.grid_index + GRID_SIZE * i] =
-fixed_cb_gain[subfrm.amp_index];
} else {
vector[subfrm.grid_index + GRID_SIZE * i] =
fixed_cb_gain[subfrm.amp_index];
}
if (j == PULSE_MAX)
break;
}
if (subfrm.dirac_train == 1)
gen_dirac_train(vector, pitch_lag);
} else { /* 5300 bps */
int cb_gain = 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;
}
}
}
/**
* Get delayed contribution from the previous excitation vector.
*/
static void get_residual(int16_t *residual, int16_t *prev_excitation, int lag)
{
int offset = PITCH_MAX - PITCH_ORDER / 2 - lag;
int i;
residual[0] = prev_excitation[offset];
residual[1] = prev_excitation[offset + 1];
offset += 2;
for (i = 2; i < SUBFRAME_LEN + PITCH_ORDER - 1; i++)
residual[i] = prev_excitation[offset + (i - 2) % lag];
}
static int dot_product(const int16_t *a, const int16_t *b, int length,
int shift)
{
int i, sum = 0;
for (i = 0; i < length; i++) {
int64_t prod = av_clipl_int32(MUL64(a[i], b[i]) << shift);
sum = av_clipl_int32(sum + prod);
}
return sum;
}
/**
* Generate adaptive codebook excitation.
*/
static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
int pitch_lag, G723_1_Subframe subfrm,
enum Rate cur_rate)
{
int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
const int16_t *cb_ptr;
int lag = pitch_lag + subfrm.ad_cb_lag - 1;
int i;
int64_t sum;
get_residual(residual, prev_excitation, lag);
/* Select quantization table */
if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2)
cb_ptr = adaptive_cb_gain85;
else
cb_ptr = adaptive_cb_gain170;
/* Calculate adaptive vector */
cb_ptr += subfrm.ad_cb_gain * 20;
for (i = 0; i < SUBFRAME_LEN; i++) {
sum = dot_product(residual + i, cb_ptr, PITCH_ORDER, 1);
vector[i] = av_clipl_int32((sum << 1) + (1 << 15)) >> 16;
}
}
/**
* Estimate maximum auto-correlation around pitch lag.
*
* @param p the context
* @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(G723_1_Context *p, int offset, int *ccr_max,
int pitch_lag, int length, int dir)
{
int limit, ccr, lag = 0;
int16_t *buf = p->excitation + offset;
int i;
pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);
limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);
for (i = pitch_lag - 3; i <= limit; i++) {
ccr = dot_product(buf, buf + dir * i, length, 1);
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 */
int64_t 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_clipl_int32(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_Context *p, int offset, int pitch_lag,
PPFParam *ppf, enum Rate cur_rate)
{
int16_t scale;
int i;
int64_t 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->excitation + offset;
int fwd_lag = autocorr_max(p, offset, &energy[1], pitch_lag,
SUBFRAME_LEN, 1);
int back_lag = autocorr_max(p, 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] = dot_product(buf, buf, SUBFRAME_LEN, 1);
/* Compute forward residual energy */
if (fwd_lag)
energy[2] = dot_product(buf + fwd_lag, buf + fwd_lag,
SUBFRAME_LEN, 1);
/* Compute backward residual energy */
if (back_lag)
energy[4] = dot_product(buf - back_lag, buf - back_lag,
SUBFRAME_LEN, 1);
/* Normalize and shorten */
temp1 = 0;
for (i = 0; i < 5; i++)
temp1 = FFMAX(energy[i], temp1);
scale = 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_Context *p, int pitch_lag,
int *exc_eng, int *scale)
{
int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
int16_t *buf = p->excitation + offset;
int index, ccr, tgt_eng, best_eng, temp;
*scale = scale_vector(p->excitation, FRAME_LEN + PITCH_MAX);
/* Compute maximum backward cross-correlation */
ccr = 0;
index = autocorr_max(p, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
ccr = av_clipl_int32((int64_t)ccr + (1 << 15)) >> 16;
/* Compute target energy */
tgt_eng = dot_product(buf, buf, SUBFRAME_LEN * 2, 1);
*exc_eng = av_clipl_int32((int64_t)tgt_eng + (1 << 15)) >> 16;
if (ccr <= 0)
return 0;
/* Compute best energy */
best_eng = dot_product(buf - index, buf - index,
SUBFRAME_LEN * 2, 1);
best_eng = av_clipl_int32((int64_t)best_eng + (1 << 15)) >> 16;
temp = best_eng * *exc_eng >> 3;
if (temp < ccr * ccr)
return index;
else
return 0;
}
/**
* Peform 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++)
vector_ptr[i - lag] = vector_ptr[i - lag] * 3 >> 2;
av_memcpy_backptr((uint8_t*)vector_ptr, lag * sizeof(*vector_ptr),
FRAME_LEN * sizeof(*vector_ptr));
memcpy(out, vector_ptr, FRAME_LEN * sizeof(*vector_ptr));
} else { /* Unvoiced */
for (i = 0; i < FRAME_LEN; i++) {
*rseed = *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
*/
static inline void iir_filter(int16_t *fir_coef, int16_t *iir_coef,
int16_t *src, int *dest)
{
int m, n;
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] >> 16);
}
dest[m] = av_clipl_int32((src[m] << 16) + (filter << 3) + (1 << 15));
}
}
/**
* 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_Context *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++) {
int64_t temp = buf[i] >> 2;
temp = av_clipl_int32(MUL64(temp, temp) << 1);
denom = av_clipl_int32(denom + temp);
}
if (num && denom) {
bits1 = normalize_bits(num, 31);
bits2 = normalize_bits(denom, 31);
num = num << bits1 >> 1;
denom <<= bits2;
bits2 = 5 + bits1 - bits2;
bits2 = FFMAX(0, bits2);
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 = ((p->pf_gain << 4) - 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 output buffer
*/
static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int16_t *buf)
{
int16_t filter_coef[2][LPC_ORDER], *buf_ptr;
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);
}
memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(*p->fir_mem));
memcpy(p->iir_mem, filter_signal + FRAME_LEN,
LPC_ORDER * sizeof(*p->iir_mem));
buf_ptr = buf + LPC_ORDER;
signal_ptr = filter_signal + LPC_ORDER;
for (i = 0; i < SUBFRAMES; i++) {
int16_t temp_vector[SUBFRAME_LEN];
int16_t temp;
int auto_corr[2];
int scale, energy;
/* Normalize */
memcpy(temp_vector, buf_ptr, SUBFRAME_LEN * sizeof(*temp_vector));
scale = scale_vector(temp_vector, SUBFRAME_LEN);
/* Compute auto correlation coefficients */
auto_corr[0] = dot_product(temp_vector, temp_vector + 1,
SUBFRAME_LEN - 1, 1);
auto_corr[1] = dot_product(temp_vector, temp_vector, SUBFRAME_LEN, 1);
/* Compute reflection coefficient */
temp = auto_corr[1] >> 16;
if (temp) {
temp = (auto_corr[0] >> 2) / temp;
}
p->reflection_coef = ((p->reflection_coef << 2) - p->reflection_coef +
temp + 2) >> 2;
temp = (p->reflection_coef * 0xffffc >> 3) & 0xfffc;
/* Compensation filter */
for (j = 0; j < SUBFRAME_LEN; j++) {
buf_ptr[j] = av_clipl_int32(signal_ptr[j] +
((signal_ptr[j - 1] >> 16) *
temp << 1)) >> 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, buf_ptr, energy);
buf_ptr += SUBFRAME_LEN;
signal_ptr += SUBFRAME_LEN;
}
}
static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
G723_1_Context *p = avctx->priv_data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int dec_mode = buf[0] & 3;
PPFParam ppf[SUBFRAMES];
int16_t cur_lsp[LPC_ORDER];
int16_t lpc[SUBFRAMES * LPC_ORDER];
int16_t acb_vector[SUBFRAME_LEN];
int16_t *vector_ptr;
int bad_frame = 0, i, j, ret;
if (buf_size < frame_size[dec_mode]) {
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;
}
if (unpack_bitstream(p, buf, buf_size) < 0) {
bad_frame = 1;
if (p->past_frame_type == ACTIVE_FRAME)
p->cur_frame_type = ACTIVE_FRAME;
else
p->cur_frame_type = UNTRANSMITTED_FRAME;
}
p->frame.nb_samples = FRAME_LEN;
if ((ret = avctx->get_buffer(avctx, &p->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
if (p->cur_frame_type == ACTIVE_FRAME) {
if (!bad_frame)
p->erased_frames = 0;
else if (p->erased_frames != 3)
p->erased_frames++;
inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
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));
vector_ptr = p->excitation + PITCH_MAX;
if (!p->erased_frames) {
/* Update interpolation gain memory */
p->interp_gain = 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);
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++) {
vector_ptr[j] = av_clip_int16(vector_ptr[j] << 1);
vector_ptr[j] = av_clip_int16(vector_ptr[j] +
acb_vector[j]);
}
vector_ptr += SUBFRAME_LEN;
}
vector_ptr = p->excitation + PITCH_MAX;
/* Save the excitation */
memcpy(p->audio + LPC_ORDER, vector_ptr, FRAME_LEN * sizeof(*p->audio));
p->interp_index = comp_interp_index(p, p->pitch_lag[1],
&p->sid_gain, &p->cur_gain);
if (p->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);
}
/* Restore the original excitation */
memcpy(p->excitation, p->prev_excitation,
PITCH_MAX * sizeof(*p->excitation));
memcpy(vector_ptr, p->audio + LPC_ORDER, FRAME_LEN * sizeof(*vector_ptr));
/* Peform pitch postfiltering */
if (p->postfilter)
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 {
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->frame.data[0], 0,
(FRAME_LEN + LPC_ORDER) * sizeof(int16_t));
} else {
/* Regenerate frame */
residual_interp(p->excitation, p->audio + LPC_ORDER, p->interp_index,
p->interp_gain, &p->random_seed);
}
}
/* Save the excitation for the next frame */
memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
PITCH_MAX * sizeof(*p->excitation));
} else {
memset(p->frame.data[0], 0, FRAME_LEN * 2);
av_log(avctx, AV_LOG_WARNING,
"G.723.1: Comfort noise generation not supported yet\n");
*got_frame_ptr = 1;
*(AVFrame *)data = p->frame;
return frame_size[dec_mode];
}
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],
p->audio + i, SUBFRAME_LEN, LPC_ORDER,
0, 1, 1 << 12);
memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));
if (p->postfilter)
formant_postfilter(p, lpc, p->audio);
memcpy(p->frame.data[0], p->audio + LPC_ORDER, FRAME_LEN * 2);
*got_frame_ptr = 1;
*(AVFrame *)data = p->frame;
return frame_size[dec_mode];
}
#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", "postfilter on/off", OFFSET(postfilter), AV_OPT_TYPE_INT,
{ 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,
};
AVCodec ff_g723_1_decoder = {
.name = "g723_1",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_G723_1,
.priv_data_size = sizeof(G723_1_Context),
.init = g723_1_decode_init,
.decode = g723_1_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
.capabilities = CODEC_CAP_SUBFRAMES,
.priv_class = &g723_1dec_class,
};
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