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g723_1: Move sharable functions to a separate file

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Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
pull/164/head
Vittorio Giovara 9 years ago
parent aac996cc01
commit 165cc6fb9d
4 changed files with 443 additions and 373 deletions
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  1. 4
      libavcodec/Makefile
  2. 267
      libavcodec/g723_1.c
  3. 141
      libavcodec/g723_1.h
  4. 404
      libavcodec/g723_1dec.c

4
libavcodec/Makefile
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@ -225,8 +225,8 @@ OBJS-$(CONFIG_FOURXM_DECODER) += 4xm.o
OBJS-$(CONFIG_FRAPS_DECODER) += fraps.o
OBJS-$(CONFIG_FRWU_DECODER) += frwu.o
OBJS-$(CONFIG_G2M_DECODER) += g2meet.o elsdec.o
OBJS-$(CONFIG_G723_1_DECODER) += g723_1dec.o acelp_vectors.o \
celp_filters.o
OBJS-$(CONFIG_G723_1_DECODER) += g723_1dec.o g723_1.o \
acelp_vectors.o celp_filters.o
OBJS-$(CONFIG_GIF_DECODER) += gifdec.o lzw.o
OBJS-$(CONFIG_GIF_ENCODER) += gif.o lzwenc.o
OBJS-$(CONFIG_GSM_DECODER) += gsmdec.o gsmdec_data.o msgsmdec.o

267
libavcodec/g723_1.c
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@ -0,0 +1,267 @@
/*
* 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
*/
#include <stdint.h>
#include "libavutil/common.h"
#include "acelp_vectors.h"
#include "avcodec.h"
#include "celp_math.h"
#include "g723_1.h"
int ff_g723_1_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 = ff_g723_1_normalize_bits(max, 15);
for (i = 0; i < length; i++)
dst[i] = vector[i] << bits >> 3;
return bits - 3;
}
int ff_g723_1_normalize_bits(int num, int width)
{
return width - av_log2(num) - 1;
}
int ff_g723_1_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;
}
void ff_g723_1_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];
}
void ff_g723_1_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];
}
}
void ff_g723_1_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;
ff_g723_1_get_residual(residual, prev_excitation, lag);
/* Select quantization table */
if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2)
cb_ptr = adaptive_cb_gain85;
else
cb_ptr = adaptive_cb_gain170;
/* Calculate adaptive vector */
cb_ptr += subfrm->ad_cb_gain * 20;
for (i = 0; i < SUBFRAME_LEN; i++) {
sum = ff_g723_1_dot_product(residual + i, cb_ptr, PITCH_ORDER);
vector[i] = av_sat_dadd32(1 << 15, sum) >> 16;
}
}
/**
* 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) & 0x1FF;
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;
}
}
void ff_g723_1_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;
}
}
void ff_g723_1_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));
}

141
libavcodec/g723_1.h
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@ -1,5 +1,5 @@
/*
* G.723.1 compatible decoder data tables.
* G.723.1 common header and data tables
* Copyright (c) 2006 Benjamin Larsson
* Copyright (c) 2010 Mohamed Naufal Basheer
*
@ -22,7 +22,7 @@
/**
* @file
* G.723.1 compatible decoder data tables
* G.723.1 types, functions and data tables
*/
#ifndef AVCODEC_G723_1_H
@ -44,6 +44,143 @@
#define GAIN_LEVELS 24
#define COS_TBL_SIZE 512
/**
* 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))
/**
* G723.1 frame types
*/
enum FrameType {
ACTIVE_FRAME, ///< Active speech
SID_FRAME, ///< Silence Insertion Descriptor frame
UNTRANSMITTED_FRAME
};
/**
* G723.1 rate values
*/
enum Rate {
RATE_6300,
RATE_5300
};
/**
* G723.1 unpacked data subframe
*/
typedef struct G723_1_Subframe {
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 PPFParam {
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;
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;
/**
* Scale vector contents based on the largest of their absolutes.
*/
int ff_g723_1_scale_vector(int16_t *dst, const int16_t *vector, int length);
/**
* 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)
*/
int ff_g723_1_normalize_bits(int num, int width);
int ff_g723_1_dot_product(const int16_t *a, const int16_t *b, int length);
/**
* Get delayed contribution from the previous excitation vector.
*/
void ff_g723_1_get_residual(int16_t *residual, int16_t *prev_excitation,
int lag);
/**
* Generate a train of dirac functions with period as pitch lag.
*/
void ff_g723_1_gen_dirac_train(int16_t *buf, int pitch_lag);
/**
* Generate adaptive codebook excitation.
*/
void ff_g723_1_gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
int pitch_lag, G723_1_Subframe *subfrm,
enum Rate cur_rate);
/**
* 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
*/
void ff_g723_1_lsp_interpolate(int16_t *lpc, int16_t *cur_lsp,
int16_t *prev_lsp);
/**
* 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
*/
void ff_g723_1_inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp,
uint8_t *lsp_index, int bad_frame);
static const uint8_t frame_size[4] = { 24, 20, 4, 1 };
/* Postfilter gain weighting factors scaled by 2^15 */

404
libavcodec/g723_1dec.c
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@ -38,74 +38,6 @@
#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 G723_1_Subframe {
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 PPFParam {
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;
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;
@ -262,108 +194,6 @@ static int16_t square_root(int val)
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.
*
@ -373,116 +203,6 @@ static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp,
#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) & 0x1FF;
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.
*
@ -522,7 +242,7 @@ static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
break;
}
if (subfrm->dirac_train == 1)
gen_dirac_train(vector, pitch_lag);
ff_g723_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;
@ -549,63 +269,6 @@ static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,
}
}
/**
* 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.
*
@ -629,7 +292,7 @@ static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,
limit = pitch_lag + 3;
for (i = pitch_lag - 3; i <= limit; i++) {
ccr = dot_product(buf, buf + dir * i, length);
ccr = ff_g723_1_dot_product(buf, buf + dir * i, length);
if (ccr > *ccr_max) {
*ccr_max = ccr;
@ -728,22 +391,24 @@ static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,
return;
/* Compute target energy */
energy[0] = dot_product(buf, buf, SUBFRAME_LEN);
energy[0] = ff_g723_1_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);
energy[2] = ff_g723_1_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);
energy[4] = ff_g723_1_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);
scale = ff_g723_1_normalize_bits(temp1, 31);
for (i = 0; i < 5; i++)
energy[i] = (energy[i] << scale) >> 16;
@ -789,7 +454,7 @@ static int comp_interp_index(G723_1_Context *p, int pitch_lag,
int index, ccr, tgt_eng, best_eng, temp;
*scale = scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX);
*scale = ff_g723_1_scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX);
buf += offset;
/* Compute maximum backward cross-correlation */
@ -798,14 +463,15 @@ static int comp_interp_index(G723_1_Context *p, int pitch_lag,
ccr = av_sat_add32(ccr, 1 << 15) >> 16;
/* Compute target energy */
tgt_eng = dot_product(buf, buf, SUBFRAME_LEN * 2);
tgt_eng = ff_g723_1_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 = ff_g723_1_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;
@ -853,8 +519,8 @@ static void residual_interp(int16_t *buf, int16_t *out, int lag,
* @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)
static void iir_filter(int16_t *fir_coef, int16_t *iir_coef,
int16_t *src, int *dest)
{
int m, n;
@ -890,8 +556,8 @@ static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
}
if (num && denom) {
bits1 = normalize_bits(num, 31);
bits2 = normalize_bits(denom, 31);
bits1 = ff_g723_1_normalize_bits(num, 31);
bits2 = ff_g723_1_normalize_bits(denom, 31);
num = num << bits1 >> 1;
denom <<= bits2;
@ -936,8 +602,7 @@ static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
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);
iir_filter(filter_coef[0], filter_coef[1], buf + i, filter_signal + i);
lpc += LPC_ORDER;
}
@ -953,11 +618,11 @@ static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
int scale, energy;
/* Normalize */
scale = scale_vector(dst, buf, SUBFRAME_LEN);
scale = ff_g723_1_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);
auto_corr[0] = ff_g723_1_dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
auto_corr[1] = ff_g723_1_dot_product(dst, dst, SUBFRAME_LEN);
/* Compute reflection coefficient */
temp = auto_corr[1] >> 16;
@ -1104,13 +769,13 @@ static void generate_noise(G723_1_Context *p)
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);
ff_g723_1_gen_acb_excitation(vector_ptr, vector_ptr,
p->pitch_lag[i >> 1], &p->subframe[i],
p->cur_rate);
ff_g723_1_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++)
@ -1231,8 +896,8 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
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);
ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
ff_g723_1_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));
@ -1249,9 +914,10 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
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);
ff_g723_1_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);
@ -1312,7 +978,7 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
} 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);
ff_g723_1_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);
}
@ -1322,7 +988,7 @@ static int g723_1_decode_frame(AVCodecContext *avctx, void *data,
else
p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
generate_noise(p);
lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);
ff_g723_1_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));
}

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