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FFmpeg/libavcodec/g723_1.c

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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
*/
#define BITSTREAM_READER_LE
#include "libavutil/audioconvert.h"
#include "libavutil/lzo.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "internal.h"
#include "get_bits.h"
#include "acelp_vectors.h"
#include "celp_filters.h"
#include "celp_math.h"
#include "g723_1_data.h"
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 + 4];
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; ///< formant postfilter
///< gain scaling unit memory
int postfilter;
int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX];
int16_t prev_data[HALF_FRAME_LEN];
int16_t prev_weight_sig[PITCH_MAX];
int16_t hpf_fir_mem; ///< highpass filter fir
int hpf_iir_mem; ///< and iir memories
int16_t perf_fir_mem[LPC_ORDER]; ///< perceptual filter fir
int16_t perf_iir_mem[LPC_ORDER]; ///< and iir memories
int16_t harmonic_mem[PITCH_MAX];
} 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;
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_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(int val)
{
return (ff_sqrt(val << 1) >> 1) & (~1);
}
/**
* Calculate the number of left-shifts required for normalizing the input.
*
* @param num input number
* @param width width of the input, 15 or 31 bits
*/
static int normalize_bits(int num, int width)
{
return width - av_log2(num) - 1;
}
#define normalize_bits_int16(num) normalize_bits(num, 15)
#define normalize_bits_int32(num) normalize_bits(num, 31)
/**
* Scale vector contents based on the largest of their absolutes.
*/
static int scale_vector(int16_t *dst, const int16_t *vector, int length)
{
int bits, max = 0;
int i;
for (i = 0; i < length; i++)
max |= FFABS(vector[i]);
bits= 14 - av_log2_16bit(max);
bits= FFMAX(bits, 0);
for (i = 0; i < length; i++)
dst[i] = vector[i] << bits >> 3;
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) \
MULL(a,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;
int temp1 = cos_tab[index] << 16;
int temp2 = (cos_tab[index + 1] - cos_tab[index]) *
((offset << 8) + 0x80) << 1;
lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 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 sum = ff_dot_product(a,b,length);
return av_sat_add32(sum, 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;
int 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 = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER);
vector[i] = av_sat_dadd32(1 << 15, av_sat_add32(sum, sum)) >> 16;
}
}
/**
* 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 = 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_Context *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] = dot_product(buf, buf, SUBFRAME_LEN);
/* Compute forward residual energy */
if (fwd_lag)
energy[2] = dot_product(buf + fwd_lag, buf + fwd_lag, SUBFRAME_LEN);
/* Compute backward residual energy */
if (back_lag)
energy[4] = 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 = normalize_bits(temp1, 31);
for (i = 0; i < 5; i++)
energy[i] = av_clipl_int32(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->audio + LPC_ORDER;
int index, ccr, tgt_eng, best_eng, temp;
*scale = 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 = 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 = 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;
}
/**
* 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++)
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 = *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] << 16) + (filter << 3) +\
(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_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++) {
int temp = buf[i] >> 2;
temp *= temp;
denom = av_sat_dadd32(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 = (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_Context *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 = scale_vector(dst, buf, SUBFRAME_LEN);
/* Compute auto correlation coefficients */
auto_corr[0] = dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
auto_corr[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 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 *out;
int bad_frame = 0, i, j, ret;
int16_t *audio = p->audio;
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;
}
out = (int16_t *)p->frame.data[0];
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));
if (!p->erased_frames) {
int16_t *vector_ptr = p->excitation + PITCH_MAX;
/* 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++) {
int v = av_clip_int16(vector_ptr[j] << 1);
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);
/* Peform pitch postfiltering */
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);
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(p->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));
}
}
} else {
memset(out, 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],
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, 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(p->audio[LPC_ORDER + i] << 1);
}
*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 = AV_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 | CODEC_CAP_DR1,
.priv_class = &g723_1dec_class,
};
#if CONFIG_G723_1_ENCODER
#define BITSTREAM_WRITER_LE
#include "put_bits.h"
static av_cold int g723_1_encode_init(AVCodecContext *avctx)
{
G723_1_Context *p = avctx->priv_data;
if (avctx->sample_rate != 8000) {
av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");
return -1;
}
if (avctx->channels != 1) {
av_log(avctx, AV_LOG_ERROR, "Only mono supported\n");
return AVERROR(EINVAL);
}
if (avctx->bit_rate == 6300) {
p->cur_rate = RATE_6300;
} else if (avctx->bit_rate == 5300) {
av_log(avctx, AV_LOG_ERROR, "Bitrate not supported yet, use 6.3k\n");
return AVERROR_PATCHWELCOME;
} else {
av_log(avctx, AV_LOG_ERROR,
"Bitrate not supported, use 6.3k\n");
return AVERROR(EINVAL);
}
avctx->frame_size = 240;
memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t));
return 0;
}
/**
* Remove DC component from the input signal.
*
* @param buf input signal
* @param fir zero memory
* @param iir pole memory
*/
static void highpass_filter(int16_t *buf, int16_t *fir, int *iir)
{
int i;
for (i = 0; i < FRAME_LEN; i++) {
*iir = (buf[i] << 15) + ((-*fir) << 15) + MULL2(*iir, 0x7f00);
*fir = buf[i];
buf[i] = av_clipl_int32((int64_t)*iir + (1 << 15)) >> 16;
}
}
/**
* Estimate autocorrelation of the input vector.
*
* @param buf input buffer
* @param autocorr autocorrelation coefficients vector
*/
static void comp_autocorr(int16_t *buf, int16_t *autocorr)
{
int i, scale, temp;
int16_t vector[LPC_FRAME];
scale_vector(vector, buf, LPC_FRAME);
/* Apply the Hamming window */
for (i = 0; i < LPC_FRAME; i++)
vector[i] = (vector[i] * hamming_window[i] + (1 << 14)) >> 15;
/* Compute the first autocorrelation coefficient */
temp = ff_dot_product(vector, vector, LPC_FRAME);
/* Apply a white noise correlation factor of (1025/1024) */
temp += temp >> 10;
/* Normalize */
scale = normalize_bits_int32(temp);
autocorr[0] = av_clipl_int32((int64_t)(temp << scale) +
(1 << 15)) >> 16;
/* Compute the remaining coefficients */
if (!autocorr[0]) {
memset(autocorr + 1, 0, LPC_ORDER * sizeof(int16_t));
} else {
for (i = 1; i <= LPC_ORDER; i++) {
temp = ff_dot_product(vector, vector + i, LPC_FRAME - i);
temp = MULL2((temp << scale), binomial_window[i - 1]);
autocorr[i] = av_clipl_int32((int64_t)temp + (1 << 15)) >> 16;
}
}
}
/**
* Use Levinson-Durbin recursion to compute LPC coefficients from
* autocorrelation values.
*
* @param lpc LPC coefficients vector
* @param autocorr autocorrelation coefficients vector
* @param error prediction error
*/
static void levinson_durbin(int16_t *lpc, int16_t *autocorr, int16_t error)
{
int16_t vector[LPC_ORDER];
int16_t partial_corr;
int i, j, temp;
memset(lpc, 0, LPC_ORDER * sizeof(int16_t));
for (i = 0; i < LPC_ORDER; i++) {
/* Compute the partial correlation coefficient */
temp = 0;
for (j = 0; j < i; j++)
temp -= lpc[j] * autocorr[i - j - 1];
temp = ((autocorr[i] << 13) + temp) << 3;
if (FFABS(temp) >= (error << 16))
break;
partial_corr = temp / (error << 1);
lpc[i] = av_clipl_int32((int64_t)(partial_corr << 14) +
(1 << 15)) >> 16;
/* Update the prediction error */
temp = MULL2(temp, partial_corr);
error = av_clipl_int32((int64_t)(error << 16) - temp +
(1 << 15)) >> 16;
memcpy(vector, lpc, i * sizeof(int16_t));
for (j = 0; j < i; j++) {
temp = partial_corr * vector[i - j - 1] << 1;
lpc[j] = av_clipl_int32((int64_t)(lpc[j] << 16) - temp +
(1 << 15)) >> 16;
}
}
}
/**
* Calculate LPC coefficients for the current frame.
*
* @param buf current frame
* @param prev_data 2 trailing subframes of the previous frame
* @param lpc LPC coefficients vector
*/
static void comp_lpc_coeff(int16_t *buf, int16_t *lpc)
{
int16_t autocorr[(LPC_ORDER + 1) * SUBFRAMES];
int16_t *autocorr_ptr = autocorr;
int16_t *lpc_ptr = lpc;
int i, j;
for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
comp_autocorr(buf + i, autocorr_ptr);
levinson_durbin(lpc_ptr, autocorr_ptr + 1, autocorr_ptr[0]);
lpc_ptr += LPC_ORDER;
autocorr_ptr += LPC_ORDER + 1;
}
}
static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp)
{
int f[LPC_ORDER + 2]; ///< coefficients of the sum and difference
///< polynomials (F1, F2) ordered as
///< f1[0], f2[0], ...., f1[5], f2[5]
int max, shift, cur_val, prev_val, count, p;
int i, j;
int64_t temp;
/* Initialize f1[0] and f2[0] to 1 in Q25 */
for (i = 0; i < LPC_ORDER; i++)
lsp[i] = (lpc[i] * bandwidth_expand[i] + (1 << 14)) >> 15;
/* Apply bandwidth expansion on the LPC coefficients */
f[0] = f[1] = 1 << 25;
/* Compute the remaining coefficients */
for (i = 0; i < LPC_ORDER / 2; i++) {
/* f1 */
f[2 * i + 2] = -f[2 * i] - ((lsp[i] + lsp[LPC_ORDER - 1 - i]) << 12);
/* f2 */
f[2 * i + 3] = f[2 * i + 1] - ((lsp[i] - lsp[LPC_ORDER - 1 - i]) << 12);
}
/* Divide f1[5] and f2[5] by 2 for use in polynomial evaluation */
f[LPC_ORDER] >>= 1;
f[LPC_ORDER + 1] >>= 1;
/* Normalize and shorten */
max = FFABS(f[0]);
for (i = 1; i < LPC_ORDER + 2; i++)
max = FFMAX(max, FFABS(f[i]));
shift = normalize_bits_int32(max);
for (i = 0; i < LPC_ORDER + 2; i++)
f[i] = av_clipl_int32((int64_t)(f[i] << shift) + (1 << 15)) >> 16;
/**
* Evaluate F1 and F2 at uniform intervals of pi/256 along the
* unit circle and check for zero crossings.
*/
p = 0;
temp = 0;
for (i = 0; i <= LPC_ORDER / 2; i++)
temp += f[2 * i] * cos_tab[0];
prev_val = av_clipl_int32(temp << 1);
count = 0;
for ( i = 1; i < COS_TBL_SIZE / 2; i++) {
/* Evaluate */
temp = 0;
for (j = 0; j <= LPC_ORDER / 2; j++)
temp += f[LPC_ORDER - 2 * j + p] * cos_tab[i * j % COS_TBL_SIZE];
cur_val = av_clipl_int32(temp << 1);
/* Check for sign change, indicating a zero crossing */
if ((cur_val ^ prev_val) < 0) {
int abs_cur = FFABS(cur_val);
int abs_prev = FFABS(prev_val);
int sum = abs_cur + abs_prev;
shift = normalize_bits_int32(sum);
sum <<= shift;
abs_prev = abs_prev << shift >> 8;
lsp[count++] = ((i - 1) << 7) + (abs_prev >> 1) / (sum >> 16);
if (count == LPC_ORDER)
break;
/* Switch between sum and difference polynomials */
p ^= 1;
/* Evaluate */
temp = 0;
for (j = 0; j <= LPC_ORDER / 2; j++){
temp += f[LPC_ORDER - 2 * j + p] *
cos_tab[i * j % COS_TBL_SIZE];
}
cur_val = av_clipl_int32(temp<<1);
}
prev_val = cur_val;
}
if (count != LPC_ORDER)
memcpy(lsp, prev_lsp, LPC_ORDER * sizeof(int16_t));
}
/**
* Quantize the current LSP subvector.
*
* @param num band number
* @param offset offset of the current subvector in an LPC_ORDER vector
* @param size size of the current subvector
*/
#define get_index(num, offset, size) \
{\
int error, max = -1;\
int16_t temp[4];\
int i, j;\
for (i = 0; i < LSP_CB_SIZE; i++) {\
for (j = 0; j < size; j++){\
temp[j] = (weight[j + (offset)] * lsp_band##num[i][j] +\
(1 << 14)) >> 15;\
}\
error = dot_product(lsp + (offset), temp, size) << 1;\
error -= dot_product(lsp_band##num[i], temp, size);\
if (error > max) {\
max = error;\
lsp_index[num] = i;\
}\
}\
}
/**
* Vector quantize the LSP frequencies.
*
* @param lsp the current lsp vector
* @param prev_lsp the previous lsp vector
*/
static void lsp_quantize(uint8_t *lsp_index, int16_t *lsp, int16_t *prev_lsp)
{
int16_t weight[LPC_ORDER];
int16_t min, max;
int shift, i;
/* Calculate the VQ weighting vector */
weight[0] = (1 << 20) / (lsp[1] - lsp[0]);
weight[LPC_ORDER - 1] = (1 << 20) /
(lsp[LPC_ORDER - 1] - lsp[LPC_ORDER - 2]);
for (i = 1; i < LPC_ORDER - 1; i++) {
min = FFMIN(lsp[i] - lsp[i - 1], lsp[i + 1] - lsp[i]);
if (min > 0x20)
weight[i] = (1 << 20) / min;
else
weight[i] = INT16_MAX;
}
/* Normalize */
max = 0;
for (i = 0; i < LPC_ORDER; i++)
max = FFMAX(weight[i], max);
shift = normalize_bits_int16(max);
for (i = 0; i < LPC_ORDER; i++) {
weight[i] <<= shift;
}
/* Compute the VQ target vector */
for (i = 0; i < LPC_ORDER; i++) {
lsp[i] -= dc_lsp[i] +
(((prev_lsp[i] - dc_lsp[i]) * 12288 + (1 << 14)) >> 15);
}
get_index(0, 0, 3);
get_index(1, 3, 3);
get_index(2, 6, 4);
}
/**
* Apply the formant perceptual weighting filter.
*
* @param flt_coef filter coefficients
* @param unq_lpc unquantized lpc vector
*/
static void perceptual_filter(G723_1_Context *p, int16_t *flt_coef,
int16_t *unq_lpc, int16_t *buf)
{
int16_t vector[FRAME_LEN + LPC_ORDER];
int i, j, k, l = 0;
memcpy(buf, p->iir_mem, sizeof(int16_t) * LPC_ORDER);
memcpy(vector, p->fir_mem, sizeof(int16_t) * LPC_ORDER);
memcpy(vector + LPC_ORDER, buf + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
for (k = 0; k < LPC_ORDER; k++) {
flt_coef[k + 2 * l] = (unq_lpc[k + l] * percept_flt_tbl[0][k] +
(1 << 14)) >> 15;
flt_coef[k + 2 * l + LPC_ORDER] = (unq_lpc[k + l] *
percept_flt_tbl[1][k] +
(1 << 14)) >> 15;
}
iir_filter(flt_coef + 2 * l, flt_coef + 2 * l + LPC_ORDER, vector + i,
buf + i, 0);
l += LPC_ORDER;
}
memcpy(p->iir_mem, buf + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
memcpy(p->fir_mem, vector + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
}
/**
* Estimate the open loop pitch period.
*
* @param buf perceptually weighted speech
* @param start estimation is carried out from this position
*/
static int estimate_pitch(int16_t *buf, int start)
{
int max_exp = 32;
int max_ccr = 0x4000;
int max_eng = 0x7fff;
int index = PITCH_MIN;
int offset = start - PITCH_MIN + 1;
int ccr, eng, orig_eng, ccr_eng, exp;
int diff, temp;
int i;
orig_eng = ff_dot_product(buf + offset, buf + offset, HALF_FRAME_LEN);
for (i = PITCH_MIN; i <= PITCH_MAX - 3; i++) {
offset--;
/* Update energy and compute correlation */
orig_eng += buf[offset] * buf[offset] -
buf[offset + HALF_FRAME_LEN] * buf[offset + HALF_FRAME_LEN];
ccr = ff_dot_product(buf + start, buf + offset, HALF_FRAME_LEN);
if (ccr <= 0)
continue;
/* Split into mantissa and exponent to maintain precision */
exp = normalize_bits_int32(ccr);
ccr = av_clipl_int32((int64_t)(ccr << exp) + (1 << 15)) >> 16;
exp <<= 1;
ccr *= ccr;
temp = normalize_bits_int32(ccr);
ccr = ccr << temp >> 16;
exp += temp;
temp = normalize_bits_int32(orig_eng);
eng = av_clipl_int32((int64_t)(orig_eng << temp) + (1 << 15)) >> 16;
exp -= temp;
if (ccr >= eng) {
exp--;
ccr >>= 1;
}
if (exp > max_exp)
continue;
if (exp + 1 < max_exp)
goto update;
/* Equalize exponents before comparison */
if (exp + 1 == max_exp)
temp = max_ccr >> 1;
else
temp = max_ccr;
ccr_eng = ccr * max_eng;
diff = ccr_eng - eng * temp;
if (diff > 0 && (i - index < PITCH_MIN || diff > ccr_eng >> 2)) {
update:
index = i;
max_exp = exp;
max_ccr = ccr;
max_eng = eng;
}
}
return index;
}
/**
* Compute harmonic noise filter parameters.
*
* @param buf perceptually weighted speech
* @param pitch_lag open loop pitch period
* @param hf harmonic filter parameters
*/
static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf)
{
int ccr, eng, max_ccr, max_eng;
int exp, max, diff;
int energy[15];
int i, j;
for (i = 0, j = pitch_lag - 3; j <= pitch_lag + 3; i++, j++) {
/* Compute residual energy */
energy[i << 1] = ff_dot_product(buf - j, buf - j, SUBFRAME_LEN);
/* Compute correlation */
energy[(i << 1) + 1] = ff_dot_product(buf, buf - j, SUBFRAME_LEN);
}
/* Compute target energy */
energy[14] = ff_dot_product(buf, buf, SUBFRAME_LEN);
/* Normalize */
max = 0;
for (i = 0; i < 15; i++)
max = FFMAX(max, FFABS(energy[i]));
exp = normalize_bits_int32(max);
for (i = 0; i < 15; i++) {
energy[i] = av_clipl_int32((int64_t)(energy[i] << exp) +
(1 << 15)) >> 16;
}
hf->index = -1;
hf->gain = 0;
max_ccr = 1;
max_eng = 0x7fff;
for (i = 0; i <= 6; i++) {
eng = energy[i << 1];
ccr = energy[(i << 1) + 1];
if (ccr <= 0)
continue;
ccr = (ccr * ccr + (1 << 14)) >> 15;
diff = ccr * max_eng - eng * max_ccr;
if (diff > 0) {
max_ccr = ccr;
max_eng = eng;
hf->index = i;
}
}
if (hf->index == -1) {
hf->index = pitch_lag;
return;
}
eng = energy[14] * max_eng;
eng = (eng >> 2) + (eng >> 3);
ccr = energy[(hf->index << 1) + 1] * energy[(hf->index << 1) + 1];
if (eng < ccr) {
eng = energy[(hf->index << 1) + 1];
if (eng >= max_eng)
hf->gain = 0x2800;
else
hf->gain = ((eng << 15) / max_eng * 0x2800 + (1 << 14)) >> 15;
}
hf->index += pitch_lag - 3;
}
/**
* Apply the harmonic noise shaping filter.
*
* @param hf filter parameters
*/
static void harmonic_filter(HFParam *hf, int16_t *src, int16_t *dest)
{
int i;
for (i = 0; i < SUBFRAME_LEN; i++) {
int64_t temp = hf->gain * src[i - hf->index] << 1;
dest[i] = av_clipl_int32((src[i] << 16) - temp + (1 << 15)) >> 16;
}
}
static void harmonic_noise_sub(HFParam *hf, int16_t *src, int16_t *dest)
{
int i;
for (i = 0; i < SUBFRAME_LEN; i++) {
int64_t temp = hf->gain * src[i - hf->index] << 1;
dest[i] = av_clipl_int32(((dest[i] - src[i]) << 16) + temp +
(1 << 15)) >> 16;
}
}
/**
* Combined synthesis and formant perceptual weighting filer.
*
* @param qnt_lpc quantized lpc coefficients
* @param perf_lpc perceptual filter coefficients
* @param perf_fir perceptual filter fir memory
* @param perf_iir perceptual filter iir memory
* @param scale the filter output will be scaled by 2^scale
*/
static void synth_percept_filter(int16_t *qnt_lpc, int16_t *perf_lpc,
int16_t *perf_fir, int16_t *perf_iir,
int16_t *src, int16_t *dest, int scale)
{
int i, j;
int16_t buf_16[SUBFRAME_LEN + LPC_ORDER];
int64_t buf[SUBFRAME_LEN];
int16_t *bptr_16 = buf_16 + LPC_ORDER;
memcpy(buf_16, perf_fir, sizeof(int16_t) * LPC_ORDER);
memcpy(dest - LPC_ORDER, perf_iir, sizeof(int16_t) * LPC_ORDER);
for (i = 0; i < SUBFRAME_LEN; i++) {
int64_t temp = 0;
for (j = 1; j <= LPC_ORDER; j++)
temp -= qnt_lpc[j - 1] * bptr_16[i - j];
buf[i] = (src[i] << 15) + (temp << 3);
bptr_16[i] = av_clipl_int32(buf[i] + (1 << 15)) >> 16;
}
for (i = 0; i < SUBFRAME_LEN; i++) {
int64_t fir = 0, iir = 0;
for (j = 1; j <= LPC_ORDER; j++) {
fir -= perf_lpc[j - 1] * bptr_16[i - j];
iir += perf_lpc[j + LPC_ORDER - 1] * dest[i - j];
}
dest[i] = av_clipl_int32(((buf[i] + (fir << 3)) << scale) + (iir << 3) +
(1 << 15)) >> 16;
}
memcpy(perf_fir, buf_16 + SUBFRAME_LEN, sizeof(int16_t) * LPC_ORDER);
memcpy(perf_iir, dest + SUBFRAME_LEN - LPC_ORDER,
sizeof(int16_t) * LPC_ORDER);
}
/**
* Compute the adaptive codebook contribution.
*
* @param buf input signal
* @param index the current subframe index
*/
static void acb_search(G723_1_Context *p, int16_t *residual,
int16_t *impulse_resp, int16_t *buf,
int index)
{
int16_t flt_buf[PITCH_ORDER][SUBFRAME_LEN];
const int16_t *cb_tbl = adaptive_cb_gain85;
int ccr_buf[PITCH_ORDER * SUBFRAMES << 2];
int pitch_lag = p->pitch_lag[index >> 1];
int acb_lag = 1;
int acb_gain = 0;
int odd_frame = index & 1;
int iter = 3 + odd_frame;
int count = 0;
int tbl_size = 85;
int i, j, k, l, max;
int64_t temp;
if (!odd_frame) {
if (pitch_lag == PITCH_MIN)
pitch_lag++;
else
pitch_lag = FFMIN(pitch_lag, PITCH_MAX - 5);
}
for (i = 0; i < iter; i++) {
get_residual(residual, p->prev_excitation, pitch_lag + i - 1);
for (j = 0; j < SUBFRAME_LEN; j++) {
temp = 0;
for (k = 0; k <= j; k++)
temp += residual[PITCH_ORDER - 1 + k] * impulse_resp[j - k];
flt_buf[PITCH_ORDER - 1][j] = av_clipl_int32((temp << 1) +
(1 << 15)) >> 16;
}
for (j = PITCH_ORDER - 2; j >= 0; j--) {
flt_buf[j][0] = ((residual[j] << 13) + (1 << 14)) >> 15;
for (k = 1; k < SUBFRAME_LEN; k++) {
temp = (flt_buf[j + 1][k - 1] << 15) +
residual[j] * impulse_resp[k];
flt_buf[j][k] = av_clipl_int32((temp << 1) + (1 << 15)) >> 16;
}
}
/* Compute crosscorrelation with the signal */
for (j = 0; j < PITCH_ORDER; j++) {
temp = ff_dot_product(buf, flt_buf[j], SUBFRAME_LEN);
ccr_buf[count++] = av_clipl_int32(temp << 1);
}
/* Compute energies */
for (j = 0; j < PITCH_ORDER; j++) {
ccr_buf[count++] = dot_product(flt_buf[j], flt_buf[j],
SUBFRAME_LEN);
}
for (j = 1; j < PITCH_ORDER; j++) {
for (k = 0; k < j; k++) {
temp = ff_dot_product(flt_buf[j], flt_buf[k], SUBFRAME_LEN);
ccr_buf[count++] = av_clipl_int32(temp<<2);
}
}
}
/* Normalize and shorten */
max = 0;
for (i = 0; i < 20 * iter; i++)
max = FFMAX(max, FFABS(ccr_buf[i]));
temp = normalize_bits_int32(max);
for (i = 0; i < 20 * iter; i++){
ccr_buf[i] = av_clipl_int32((int64_t)(ccr_buf[i] << temp) +
(1 << 15)) >> 16;
}
max = 0;
for (i = 0; i < iter; i++) {
/* Select quantization table */
if (!odd_frame && pitch_lag + i - 1 >= SUBFRAME_LEN - 2 ||
odd_frame && pitch_lag >= SUBFRAME_LEN - 2) {
cb_tbl = adaptive_cb_gain170;
tbl_size = 170;
}
for (j = 0, k = 0; j < tbl_size; j++, k += 20) {
temp = 0;
for (l = 0; l < 20; l++)
temp += ccr_buf[20 * i + l] * cb_tbl[k + l];
temp = av_clipl_int32(temp);
if (temp > max) {
max = temp;
acb_gain = j;
acb_lag = i;
}
}
}
if (!odd_frame) {
pitch_lag += acb_lag - 1;
acb_lag = 1;
}
p->pitch_lag[index >> 1] = pitch_lag;
p->subframe[index].ad_cb_lag = acb_lag;
p->subframe[index].ad_cb_gain = acb_gain;
}
/**
* Subtract the adaptive codebook contribution from the input
* to obtain the residual.
*
* @param buf target vector
*/
static void sub_acb_contrib(int16_t *residual, int16_t *impulse_resp,
int16_t *buf)
{
int i, j;
/* Subtract adaptive CB contribution to obtain the residual */
for (i = 0; i < SUBFRAME_LEN; i++) {
int64_t temp = buf[i] << 14;
for (j = 0; j <= i; j++)
temp -= residual[j] * impulse_resp[i - j];
buf[i] = av_clipl_int32((temp << 2) + (1 << 15)) >> 16;
}
}
/**
* Quantize the residual signal using the fixed codebook (MP-MLQ).
*
* @param optim optimized fixed codebook parameters
* @param buf excitation vector
*/
static void get_fcb_param(FCBParam *optim, int16_t *impulse_resp,
int16_t *buf, int pulse_cnt, int pitch_lag)
{
FCBParam param;
int16_t impulse_r[SUBFRAME_LEN];
int16_t temp_corr[SUBFRAME_LEN];
int16_t impulse_corr[SUBFRAME_LEN];
int ccr1[SUBFRAME_LEN];
int ccr2[SUBFRAME_LEN];
int amp, err, max, max_amp_index, min, scale, i, j, k, l;
int64_t temp;
/* Update impulse response */
memcpy(impulse_r, impulse_resp, sizeof(int16_t) * SUBFRAME_LEN);
param.dirac_train = 0;
if (pitch_lag < SUBFRAME_LEN - 2) {
param.dirac_train = 1;
gen_dirac_train(impulse_r, pitch_lag);
}
for (i = 0; i < SUBFRAME_LEN; i++)
temp_corr[i] = impulse_r[i] >> 1;
/* Compute impulse response autocorrelation */
temp = dot_product(temp_corr, temp_corr, SUBFRAME_LEN);
scale = normalize_bits_int32(temp);
impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
for (i = 1; i < SUBFRAME_LEN; i++) {
temp = dot_product(temp_corr + i, temp_corr, SUBFRAME_LEN - i);
impulse_corr[i] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
}
/* Compute crosscorrelation of impulse response with residual signal */
scale -= 4;
for (i = 0; i < SUBFRAME_LEN; i++){
temp = dot_product(buf + i, impulse_r, SUBFRAME_LEN - i);
if (scale < 0)
ccr1[i] = temp >> -scale;
else
ccr1[i] = av_clipl_int32(temp << scale);
}
/* Search loop */
for (i = 0; i < GRID_SIZE; i++) {
/* Maximize the crosscorrelation */
max = 0;
for (j = i; j < SUBFRAME_LEN; j += GRID_SIZE) {
temp = FFABS(ccr1[j]);
if (temp >= max) {
max = temp;
param.pulse_pos[0] = j;
}
}
/* Quantize the gain (max crosscorrelation/impulse_corr[0]) */
amp = max;
min = 1 << 30;
max_amp_index = GAIN_LEVELS - 2;
for (j = max_amp_index; j >= 2; j--) {
temp = av_clipl_int32((int64_t)fixed_cb_gain[j] *
impulse_corr[0] << 1);
temp = FFABS(temp - amp);
if (temp < min) {
min = temp;
max_amp_index = j;
}
}
max_amp_index--;
/* Select additional gain values */
for (j = 1; j < 5; j++) {
for (k = i; k < SUBFRAME_LEN; k += GRID_SIZE) {
temp_corr[k] = 0;
ccr2[k] = ccr1[k];
}
param.amp_index = max_amp_index + j - 2;
amp = fixed_cb_gain[param.amp_index];
param.pulse_sign[0] = (ccr2[param.pulse_pos[0]] < 0) ? -amp : amp;
temp_corr[param.pulse_pos[0]] = 1;
for (k = 1; k < pulse_cnt; k++) {
max = -1 << 30;
for (l = i; l < SUBFRAME_LEN; l += GRID_SIZE) {
if (temp_corr[l])
continue;
temp = impulse_corr[FFABS(l - param.pulse_pos[k - 1])];
temp = av_clipl_int32((int64_t)temp *
param.pulse_sign[k - 1] << 1);
ccr2[l] -= temp;
temp = FFABS(ccr2[l]);
if (temp > max) {
max = temp;
param.pulse_pos[k] = l;
}
}
param.pulse_sign[k] = (ccr2[param.pulse_pos[k]] < 0) ?
-amp : amp;
temp_corr[param.pulse_pos[k]] = 1;
}
/* Create the error vector */
memset(temp_corr, 0, sizeof(int16_t) * SUBFRAME_LEN);
for (k = 0; k < pulse_cnt; k++)
temp_corr[param.pulse_pos[k]] = param.pulse_sign[k];
for (k = SUBFRAME_LEN - 1; k >= 0; k--) {
temp = 0;
for (l = 0; l <= k; l++) {
int prod = av_clipl_int32((int64_t)temp_corr[l] *
impulse_r[k - l] << 1);
temp = av_clipl_int32(temp + prod);
}
temp_corr[k] = temp << 2 >> 16;
}
/* Compute square of error */
err = 0;
for (k = 0; k < SUBFRAME_LEN; k++) {
int64_t prod;
prod = av_clipl_int32((int64_t)buf[k] * temp_corr[k] << 1);
err = av_clipl_int32(err - prod);
prod = av_clipl_int32((int64_t)temp_corr[k] * temp_corr[k]);
err = av_clipl_int32(err + prod);
}
/* Minimize */
if (err < optim->min_err) {
optim->min_err = err;
optim->grid_index = i;
optim->amp_index = param.amp_index;
optim->dirac_train = param.dirac_train;
for (k = 0; k < pulse_cnt; k++) {
optim->pulse_sign[k] = param.pulse_sign[k];
optim->pulse_pos[k] = param.pulse_pos[k];
}
}
}
}
}
/**
* Encode the pulse position and gain of the current subframe.
*
* @param optim optimized fixed CB parameters
* @param buf excitation vector
*/
static void pack_fcb_param(G723_1_Subframe *subfrm, FCBParam *optim,
int16_t *buf, int pulse_cnt)
{
int i, j;
j = PULSE_MAX - pulse_cnt;
subfrm->pulse_sign = 0;
subfrm->pulse_pos = 0;
for (i = 0; i < SUBFRAME_LEN >> 1; i++) {
int val = buf[optim->grid_index + (i << 1)];
if (!val) {
subfrm->pulse_pos += combinatorial_table[j][i];
} else {
subfrm->pulse_sign <<= 1;
if (val < 0) subfrm->pulse_sign++;
j++;
if (j == PULSE_MAX) break;
}
}
subfrm->amp_index = optim->amp_index;
subfrm->grid_index = optim->grid_index;
subfrm->dirac_train = optim->dirac_train;
}
/**
* Compute the fixed codebook excitation.
*
* @param buf target vector
* @param impulse_resp impulse response of the combined filter
*/
static void fcb_search(G723_1_Context *p, int16_t *impulse_resp,
int16_t *buf, int index)
{
FCBParam optim;
int pulse_cnt = pulses[index];
int i;
optim.min_err = 1 << 30;
get_fcb_param(&optim, impulse_resp, buf, pulse_cnt, SUBFRAME_LEN);
if (p->pitch_lag[index >> 1] < SUBFRAME_LEN - 2) {
get_fcb_param(&optim, impulse_resp, buf, pulse_cnt,
p->pitch_lag[index >> 1]);
}
/* Reconstruct the excitation */
memset(buf, 0, sizeof(int16_t) * SUBFRAME_LEN);
for (i = 0; i < pulse_cnt; i++)
buf[optim.pulse_pos[i]] = optim.pulse_sign[i];
pack_fcb_param(&p->subframe[index], &optim, buf, pulse_cnt);
if (optim.dirac_train)
gen_dirac_train(buf, p->pitch_lag[index >> 1]);
}
/**
* Pack the frame parameters into output bitstream.
*
* @param frame output buffer
* @param size size of the buffer
*/
static int pack_bitstream(G723_1_Context *p, unsigned char *frame, int size)
{
PutBitContext pb;
int info_bits, i, temp;
init_put_bits(&pb, frame, size);
if (p->cur_rate == RATE_6300) {
info_bits = 0;
put_bits(&pb, 2, info_bits);
}
put_bits(&pb, 8, p->lsp_index[2]);
put_bits(&pb, 8, p->lsp_index[1]);
put_bits(&pb, 8, p->lsp_index[0]);
put_bits(&pb, 7, p->pitch_lag[0] - PITCH_MIN);
put_bits(&pb, 2, p->subframe[1].ad_cb_lag);
put_bits(&pb, 7, p->pitch_lag[1] - PITCH_MIN);
put_bits(&pb, 2, p->subframe[3].ad_cb_lag);
/* Write 12 bit combined gain */
for (i = 0; i < SUBFRAMES; i++) {
temp = p->subframe[i].ad_cb_gain * GAIN_LEVELS +
p->subframe[i].amp_index;
if (p->cur_rate == RATE_6300)
temp += p->subframe[i].dirac_train << 11;
put_bits(&pb, 12, temp);
}
put_bits(&pb, 1, p->subframe[0].grid_index);
put_bits(&pb, 1, p->subframe[1].grid_index);
put_bits(&pb, 1, p->subframe[2].grid_index);
put_bits(&pb, 1, p->subframe[3].grid_index);
if (p->cur_rate == RATE_6300) {
skip_put_bits(&pb, 1); /* reserved bit */
/* Write 13 bit combined position index */
temp = (p->subframe[0].pulse_pos >> 16) * 810 +
(p->subframe[1].pulse_pos >> 14) * 90 +
(p->subframe[2].pulse_pos >> 16) * 9 +
(p->subframe[3].pulse_pos >> 14);
put_bits(&pb, 13, temp);
put_bits(&pb, 16, p->subframe[0].pulse_pos & 0xffff);
put_bits(&pb, 14, p->subframe[1].pulse_pos & 0x3fff);
put_bits(&pb, 16, p->subframe[2].pulse_pos & 0xffff);
put_bits(&pb, 14, p->subframe[3].pulse_pos & 0x3fff);
put_bits(&pb, 6, p->subframe[0].pulse_sign);
put_bits(&pb, 5, p->subframe[1].pulse_sign);
put_bits(&pb, 6, p->subframe[2].pulse_sign);
put_bits(&pb, 5, p->subframe[3].pulse_sign);
}
flush_put_bits(&pb);
return frame_size[info_bits];
}
static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
G723_1_Context *p = avctx->priv_data;
int16_t unq_lpc[LPC_ORDER * SUBFRAMES];
int16_t qnt_lpc[LPC_ORDER * SUBFRAMES];
int16_t cur_lsp[LPC_ORDER];
int16_t weighted_lpc[LPC_ORDER * SUBFRAMES << 1];
int16_t vector[FRAME_LEN + PITCH_MAX];
int offset, ret;
int16_t *in = (const int16_t *)frame->data[0];
HFParam hf[4];
int i, j;
highpass_filter(in, &p->hpf_fir_mem, &p->hpf_iir_mem);
memcpy(vector, p->prev_data, HALF_FRAME_LEN * sizeof(int16_t));
memcpy(vector + HALF_FRAME_LEN, in, FRAME_LEN * sizeof(int16_t));
comp_lpc_coeff(vector, unq_lpc);
lpc2lsp(&unq_lpc[LPC_ORDER * 3], p->prev_lsp, cur_lsp);
lsp_quantize(p->lsp_index, cur_lsp, p->prev_lsp);
/* Update memory */
memcpy(vector + LPC_ORDER, p->prev_data + SUBFRAME_LEN,
sizeof(int16_t) * SUBFRAME_LEN);
memcpy(vector + LPC_ORDER + SUBFRAME_LEN, in,
sizeof(int16_t) * (HALF_FRAME_LEN + SUBFRAME_LEN));
memcpy(p->prev_data, in + HALF_FRAME_LEN,
sizeof(int16_t) * HALF_FRAME_LEN);
memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
perceptual_filter(p, weighted_lpc, unq_lpc, vector);
memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
scale_vector(vector, vector, FRAME_LEN + PITCH_MAX);
p->pitch_lag[0] = estimate_pitch(vector, PITCH_MAX);
p->pitch_lag[1] = estimate_pitch(vector, PITCH_MAX + HALF_FRAME_LEN);
for (i = PITCH_MAX, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
comp_harmonic_coeff(vector + i, p->pitch_lag[j >> 1], hf + j);
memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
memcpy(p->prev_weight_sig, vector + FRAME_LEN, sizeof(int16_t) * PITCH_MAX);
for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
harmonic_filter(hf + j, vector + PITCH_MAX + i, in + i);
inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);
lsp_interpolate(qnt_lpc, cur_lsp, p->prev_lsp);
memcpy(p->prev_lsp, cur_lsp, sizeof(int16_t) * LPC_ORDER);
offset = 0;
for (i = 0; i < SUBFRAMES; i++) {
int16_t impulse_resp[SUBFRAME_LEN];
int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
int16_t flt_in[SUBFRAME_LEN];
int16_t zero[LPC_ORDER], fir[LPC_ORDER], iir[LPC_ORDER];
/**
* Compute the combined impulse response of the synthesis filter,
* formant perceptual weighting filter and harmonic noise shaping filter
*/
memset(zero, 0, sizeof(int16_t) * LPC_ORDER);
memset(vector, 0, sizeof(int16_t) * PITCH_MAX);
memset(flt_in, 0, sizeof(int16_t) * SUBFRAME_LEN);
flt_in[0] = 1 << 13; /* Unit impulse */
synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
zero, zero, flt_in, vector + PITCH_MAX, 1);
harmonic_filter(hf + i, vector + PITCH_MAX, impulse_resp);
/* Compute the combined zero input response */
flt_in[0] = 0;
memcpy(fir, p->perf_fir_mem, sizeof(int16_t) * LPC_ORDER);
memcpy(iir, p->perf_iir_mem, sizeof(int16_t) * LPC_ORDER);
synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
fir, iir, flt_in, vector + PITCH_MAX, 0);
memcpy(vector, p->harmonic_mem, sizeof(int16_t) * PITCH_MAX);
harmonic_noise_sub(hf + i, vector + PITCH_MAX, in);
acb_search(p, residual, impulse_resp, in, i);
gen_acb_excitation(residual, p->prev_excitation,p->pitch_lag[i >> 1],
&p->subframe[i], p->cur_rate);
sub_acb_contrib(residual, impulse_resp, in);
fcb_search(p, impulse_resp, in, i);
/* Reconstruct the excitation */
gen_acb_excitation(impulse_resp, p->prev_excitation, p->pitch_lag[i >> 1],
&p->subframe[i], RATE_6300);
memmove(p->prev_excitation, p->prev_excitation + SUBFRAME_LEN,
sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
for (j = 0; j < SUBFRAME_LEN; j++)
in[j] = av_clip_int16((in[j] << 1) + impulse_resp[j]);
memcpy(p->prev_excitation + PITCH_MAX - SUBFRAME_LEN, in,
sizeof(int16_t) * SUBFRAME_LEN);
/* Update filter memories */
synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
p->perf_fir_mem, p->perf_iir_mem,
in, vector + PITCH_MAX, 0);
memmove(p->harmonic_mem, p->harmonic_mem + SUBFRAME_LEN,
sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
memcpy(p->harmonic_mem + PITCH_MAX - SUBFRAME_LEN, vector + PITCH_MAX,
sizeof(int16_t) * SUBFRAME_LEN);
in += SUBFRAME_LEN;
offset += LPC_ORDER;
}
if ((ret = ff_alloc_packet2(avctx, avpkt, 24)))
return ret;
*got_packet_ptr = 1;
avpkt->size = pack_bitstream(p, avpkt->data, avpkt->size);
return 0;
}
AVCodec ff_g723_1_encoder = {
.name = "g723_1",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_G723_1,
.priv_data_size = sizeof(G723_1_Context),
.init = g723_1_encode_init,
.encode2 = g723_1_encode_frame,
.long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE},
};
#endif