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FFmpeg/libavcodec/g723_1enc.c
Andreas Rheinhardt 4243da4ff4 avcodec/codec_internal: Use union for FFCodec decode/encode callbacks 
This is possible, because every given FFCodec has to implement
exactly one of these. Doing so decreases sizeof(FFCodec) and
therefore decreases the size of the binary.
Notice that in case of position-independent code the decrease
is in .data.rel.ro, so that this translates to decreased
memory consumption.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-04-05 20:02:37 +02:00

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/*
* G.723.1 compatible encoder
* Copyright (c) Mohamed Naufal <naufal22@gmail.com>
*
* 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 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 "codec_internal.h"
#include "encode.h"
#include "g723_1.h"
#define BITSTREAM_WRITER_LE
#include "put_bits.h"
/**
* Hamming window coefficients scaled by 2^15
*/
static const int16_t hamming_window[LPC_FRAME] = {
2621, 2631, 2659, 2705, 2770, 2853, 2955, 3074, 3212, 3367,
3541, 3731, 3939, 4164, 4405, 4663, 4937, 5226, 5531, 5851,
6186, 6534, 6897, 7273, 7661, 8062, 8475, 8899, 9334, 9780,
10235, 10699, 11172, 11653, 12141, 12636, 13138, 13645, 14157, 14673,
15193, 15716, 16242, 16769, 17298, 17827, 18356, 18884, 19411, 19935,
20457, 20975, 21489, 21999, 22503, 23002, 23494, 23978, 24455, 24924,
25384, 25834, 26274, 26704, 27122, 27529, 27924, 28306, 28675, 29031,
29373, 29700, 30012, 30310, 30592, 30857, 31107, 31340, 31557, 31756,
31938, 32102, 32249, 32377, 32488, 32580, 32654, 32710, 32747, 32766,
32766, 32747, 32710, 32654, 32580, 32488, 32377, 32249, 32102, 31938,
31756, 31557, 31340, 31107, 30857, 30592, 30310, 30012, 29700, 29373,
29031, 28675, 28306, 27924, 27529, 27122, 26704, 26274, 25834, 25384,
24924, 24455, 23978, 23494, 23002, 22503, 21999, 21489, 20975, 20457,
19935, 19411, 18884, 18356, 17827, 17298, 16769, 16242, 15716, 15193,
14673, 14157, 13645, 13138, 12636, 12141, 11653, 11172, 10699, 10235,
9780, 9334, 8899, 8475, 8062, 7661, 7273, 6897, 6534, 6186,
5851, 5531, 5226, 4937, 4663, 4405, 4164, 3939, 3731, 3541,
3367, 3212, 3074, 2955, 2853, 2770, 2705, 2659, 2631, 2621
};
/**
* Binomial window coefficients scaled by 2^15
*/
static const int16_t binomial_window[LPC_ORDER] = {
32749, 32695, 32604, 32477, 32315, 32118, 31887, 31622, 31324, 30995
};
/**
* 0.994^i scaled by 2^15
*/
static const int16_t bandwidth_expand[LPC_ORDER] = {
32571, 32376, 32182, 31989, 31797, 31606, 31416, 31228, 31040, 30854
};
/**
* 0.5^i scaled by 2^15
*/
static const int16_t percept_flt_tbl[2][LPC_ORDER] = {
/* Zero part */
{29491, 26542, 23888, 21499, 19349, 17414, 15673, 14106, 12695, 11425},
/* Pole part */
{16384, 8192, 4096, 2048, 1024, 512, 256, 128, 64, 32}
};
static av_cold int g723_1_encode_init(AVCodecContext *avctx)
{
G723_1_Context *s = avctx->priv_data;
G723_1_ChannelContext *p = &s->ch[0];
if (avctx->sample_rate != 8000) {
av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate 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, "Use bitrate 6300 instead of 5300.\n");
avpriv_report_missing_feature(avctx, "Bitrate 5300");
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] * G723_1_COS_TAB_FIRST_ELEMENT;
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] * ff_g723_1_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] *
ff_g723_1_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)] * ff_g723_1_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(ff_g723_1_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_ChannelContext *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_ChannelContext *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 = ff_g723_1_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 = ff_g723_1_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) ff_g723_1_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 = ff_g723_1_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 += ff_g723_1_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_ChannelContext *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 void pack_bitstream(G723_1_ChannelContext *p, AVPacket *avpkt, int info_bits)
{
PutBitContext pb;
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) {
put_bits(&pb, 1, 0); /* 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);
}
static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
G723_1_Context *s = avctx->priv_data;
G723_1_ChannelContext *p = &s->ch[0];
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, info_bits = 0;
int16_t *in, *start;
HFParam hf[4];
/* duplicate input */
start = in = av_memdup(frame->data[0], frame->nb_samples * sizeof(int16_t));
if (!in)
return AVERROR(ENOMEM);
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],
p->cur_rate);
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_get_encode_buffer(avctx, avpkt, frame_size[info_bits], 0);
if (ret < 0)
return ret;
*got_packet_ptr = 1;
pack_bitstream(p, avpkt, info_bits);
return 0;
}
static const FFCodecDefault defaults[] = {
{ "b", "6300" },
{ NULL },
};
const FFCodec ff_g723_1_encoder = {
.p.name = "g723_1",
.p.long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_G723_1,
.p.capabilities = AV_CODEC_CAP_DR1,
.priv_data_size = sizeof(G723_1_Context),
.init = g723_1_encode_init,
FF_CODEC_ENCODE_CB(g723_1_encode_frame),
.defaults = defaults,
.p.sample_fmts = (const enum AVSampleFormat[]) {
AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE
},
.p.ch_layouts = (const AVChannelLayout[]){
AV_CHANNEL_LAYOUT_MONO, { 0 }
},
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
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