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ffmpeg/libavcodec/g723_1.c
Justin Ruggles a903f8f087 Include libavutil/channel_layout.h instead of libavutil/audioconvert.h 
Also reorder some other #include when applicable.
2012-11-11 13:35:12 -05:00

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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/channel_layout.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "get_bits.h"
#include "acelp_vectors.h"
#include "celp_filters.h"
#include "g723_1_data.h"
#define CNG_RANDOM_SEED 12345
/**
* 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 sid_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 cng_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 + PITCH_MAX + 4];
} 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));
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_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)
{
return width - av_log2(num) - 1;
}
/**
* 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]);
max = FFMIN(max, 0x7FFF);
bits = normalize_bits(max, 15);
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) \
((((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;
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 i, sum = 0;
for (i = 0; i < length; i++) {
int prod = a[i] * b[i];
sum = av_sat_dadd32(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;
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 = dot_product(residual + i, cb_ptr, PITCH_ORDER);
vector[i] = av_sat_dadd32(1 << 15, 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] = (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
*/
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++) {
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);
lpc += LPC_ORDER;
}
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 += 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 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_Context *p)
{
int i, shift, seg, seg2, t, val, val_add, x, y;
shift = 16 - p->cur_gain * 2;
if (shift > 0)
t = p->sid_gain << shift;
else
t = p->sid_gain >> -shift;
x = 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 << 4) + val;
if (t >= y)
val++;
} else {
t = seg * 32 + (val - 1 << seg2);
t = t * t - x;
val = (seg2 - 1 << 4) + val;
if (t >= y)
val--;
}
return val;
}
static void generate_noise(G723_1_Context *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 << 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) {
gen_acb_excitation(vector_ptr, vector_ptr,
p->pitch_lag[i >> 1], &p->subframe[i],
p->cur_rate);
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] << -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 <<= 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, 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));
}
}
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);
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);
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 (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,
{ .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,
};
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,
};
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