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ffmpeg/libavcodec/g723_1enc.c
Mohamed Naufal f023d57d35 lavc: G.723.1 encoder 
Additional improvements by Michael Niedermayer <michaelni@gmx.at>.

Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
2015-11-30 10:58:46 -05:00

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/*
* G.723.1 compatible encoder
* Copyright (c) Mohamed Naufal <naufal22@gmail.com>
*
* 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 encoder
*/
#include <stdint.h>
#include <string.h>
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "celp_math.h"
#include "g723_1.h"
#include "internal.h"
#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 AVERROR(EINVAL);
}
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 6300\n");
return AVERROR_PATCHWELCOME;
} else {
av_log(avctx, AV_LOG_ERROR, "Bitrate not supported, use 6300\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];
ff_g723_1_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 = ff_g723_1_normalize_bits(temp, 31);
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 = ff_g723_1_normalize_bits(max, 31);
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 = ff_g723_1_normalize_bits(sum, 31);
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 = ff_g723_1_dot_product(lsp + (offset), temp, size) << 1; \
error -= ff_g723_1_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 = ff_g723_1_normalize_bits(max, 15);
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);
}
/**
* Perform IIR filtering.
*
* @param fir_coef FIR coefficients
* @param iir_coef IIR coefficients
* @param src source vector
* @param dest destination vector
*/
static void iir_filter(int16_t *fir_coef, int16_t *iir_coef,
int16_t *src, int16_t *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];
}
dest[m] = av_clipl_int32((src[m] << 16) + (filter << 3) +
(1 << 15)) >> 16;
}
}
/**
* 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);
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 = ff_g723_1_normalize_bits(ccr, 31);
ccr = av_clipl_int32((int64_t) (ccr << exp) + (1 << 15)) >> 16;
exp <<= 1;
ccr *= ccr;
temp = ff_g723_1_normalize_bits(ccr, 31);
ccr = ccr << temp >> 16;
exp += temp;
temp = ff_g723_1_normalize_bits(orig_eng, 31);
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 = ff_g723_1_normalize_bits(max, 31);
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, const 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, const 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,
const 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, const 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++) {
ff_g723_1_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++] = ff_g723_1_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 = ff_g723_1_normalize_bits(max, 31);
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(const int16_t *residual, const 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;
ff_g723_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 = ff_g723_1_dot_product(temp_corr, temp_corr, SUBFRAME_LEN);
scale = ff_g723_1_normalize_bits(temp, 31);
impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
for (i = 1; i < SUBFRAME_LEN; i++) {
temp = ff_g723_1_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 = ff_g723_1_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 = INT_MIN;
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)
ff_g723_1_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, AVPacket *avpkt)
{
PutBitContext pb;
int info_bits = 0;
int i, temp;
init_put_bits(&pb, avpkt->data, avpkt->size);
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, i, j;
int16_t *in, *start;
HFParam hf[4];
/* duplicate input */
start = in = av_malloc(frame->nb_samples * sizeof(int16_t));
if (!in)
return AVERROR(ENOMEM);
memcpy(in, frame->data[0], frame->nb_samples * sizeof(int16_t));
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);
ff_g723_1_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);
ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);
ff_g723_1_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);
ff_g723_1_gen_acb_excitation(residual, p->prev_excitation,
p->pitch_lag[i >> 1], &p->subframe[i],
RATE_6300);
sub_acb_contrib(residual, impulse_resp, in);
fcb_search(p, impulse_resp, in, i);
/* Reconstruct the excitation */
ff_g723_1_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;
}
av_free(start);
ret = ff_alloc_packet(avpkt, 24);
if (ret < 0)
return ret;
*got_packet_ptr = 1;
return pack_bitstream(p, avpkt);
}
AVCodec ff_g723_1_encoder = {
.name = "g723_1",
.long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
.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,
.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
};
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