* Copyright (c) 2006 Benjamin Larsson
* Copyright (c) 2010 Mohamed Naufal Basheer
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/lzo.h"
#include "libavutil/opt.h"
#include "avcodec.h"
+#include "internal.h"
#include "get_bits.h"
#include "acelp_vectors.h"
#include "celp_filters.h"
+#include "celp_math.h"
#include "g723_1_data.h"
-/**
- * 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;
int sid_gain;
int cur_gain;
int reflection_coef;
- int pf_gain;
+ int pf_gain; ///< formant postfilter
+ ///< gain scaling unit memory
int postfilter;
+ int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX];
+ int16_t prev_data[HALF_FRAME_LEN];
+ int16_t prev_weight_sig[PITCH_MAX];
+
+
+ int16_t hpf_fir_mem; ///< highpass filter fir
+ int hpf_iir_mem; ///< and iir memories
+ int16_t perf_fir_mem[LPC_ORDER]; ///< perceptual filter fir
+ int16_t perf_iir_mem[LPC_ORDER]; ///< and iir memories
- int16_t audio[FRAME_LEN + LPC_ORDER];
+ int16_t harmonic_mem[PITCH_MAX];
} G723_1_Context;
static av_cold int g723_1_decode_init(AVCodecContext *avctx)
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);
}
}
- 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);
+ p->subframe[0].grid_index = get_bits1(&gb);
+ p->subframe[1].grid_index = get_bits1(&gb);
+ p->subframe[2].grid_index = get_bits1(&gb);
+ p->subframe[3].grid_index = get_bits1(&gb);
if (p->cur_rate == RATE_6300) {
- skip_bits(&gb, 1); /* skip reserved bit */
+ skip_bits1(&gb); /* skip reserved bit */
/* Compute pulse_pos index using the 13-bit combined position index */
temp = get_bits(&gb, 13);
*/
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;
+ return (ff_sqrt(val << 1) >> 1) & (~1);
}
/**
* Calculate the number of left-shifts required for normalizing the input.
*
* @param num input number
- * @param width width of the input, 16 bits(0) / 32 bits(1)
+ * @param width width of the input, 15 or 31 bits
*/
static int normalize_bits(int num, int width)
{
- if (!num)
- return 0;
- if (num == -1)
- return width;
- if (num < 0)
- num = ~num;
-
return width - av_log2(num) - 1;
}
+#define normalize_bits_int16(num) normalize_bits(num, 15)
+#define normalize_bits_int32(num) normalize_bits(num, 31)
+
/**
* Scale vector contents based on the largest of their absolutes.
*/
-static int scale_vector(int16_t *vector, int length)
+static int scale_vector(int16_t *dst, const int16_t *vector, int length)
{
int bits, max = 0;
- int64_t scale;
int i;
-
for (i = 0; i < length; i++)
- max = FFMAX(max, FFABS(vector[i]));
+ max |= FFABS(vector[i]);
- max = FFMIN(max, 0x7FFF);
- bits = normalize_bits(max, 15);
- scale = (bits == 15) ? 0x7FFF : (1 << bits);
+ bits= 14 - av_log2_16bit(max);
+ bits= FFMAX(bits, 0);
for (i = 0; i < length; i++)
- vector[i] = av_clipl_int32(vector[i] * scale << 1) >> 4;
+ dst[i] = vector[i] << bits >> 3;
return bits - 3;
}
* @param b 16 bit multiplier
*/
#define MULL2(a, b) \
- ((((a) >> 16) * (b) << 1) + (((a) & 0xffff) * (b) >> 15))
+ MULL(a,b,15)
/**
* Convert LSP frequencies to LPC coefficients.
for (j = 0; j < LPC_ORDER; j++) {
int index = lpc[j] >> 7;
int offset = lpc[j] & 0x7f;
- int64_t temp1 = cos_tab[index] << 16;
+ int temp1 = cos_tab[index] << 16;
int temp2 = (cos_tab[index + 1] - cos_tab[index]) *
((offset << 8) + 0x80) << 1;
- lpc[j] = -(av_clipl_int32(((temp1 + temp2) << 1) + (1 << 15)) >> 16);
+ lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 16);
}
/*
* @param pitch_lag closed loop pitch lag
* @param index current subframe index
*/
-static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe subfrm,
+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])
+ if (subfrm->pulse_pos >= max_pos[index])
return;
/* Decode amplitudes and positions */
j = PULSE_MAX - pulses[index];
- temp = subfrm.pulse_pos;
+ 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];
+ 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];
+ vector[subfrm->grid_index + GRID_SIZE * i] =
+ fixed_cb_gain[subfrm->amp_index];
}
if (j == PULSE_MAX)
break;
}
- if (subfrm.dirac_train == 1)
+ 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 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) {
}
/* 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];
+ 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++)
static int dot_product(const int16_t *a, const int16_t *b, int length)
{
- int i, sum = 0;
-
- for (i = 0; i < length; i++) {
- int64_t prod = av_clipl_int32((int64_t)(a[i] * b[i]) << 1);
- sum = av_clipl_int32(sum + prod);
- }
- return sum;
+ int sum = ff_dot_product(a,b,length);
+ return av_sat_add32(sum, sum);
}
/**
* Generate adaptive codebook excitation.
*/
static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
- int pitch_lag, G723_1_Subframe subfrm,
+ 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 lag = pitch_lag + subfrm->ad_cb_lag - 1;
int i;
- int64_t sum;
+ int sum;
get_residual(residual, prev_excitation, lag);
/* Select quantization table */
- if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2)
+ if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2) {
cb_ptr = adaptive_cb_gain85;
- else
+ } else
cb_ptr = adaptive_cb_gain170;
/* Calculate adaptive vector */
- cb_ptr += subfrm.ad_cb_gain * 20;
+ 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_clipl_int32((sum << 1) + (1 << 15)) >> 16;
+ sum = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER);
+ vector[i] = av_sat_dadd32(1 << 15, av_sat_add32(sum, sum)) >> 16;
}
}
/**
* Estimate maximum auto-correlation around pitch lag.
*
- * @param p the context
+ * @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(G723_1_Context *p, int offset, int *ccr_max,
+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;
- int16_t *buf = p->excitation + offset;
int i;
pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);
int tgt_eng, int ccr, int res_eng)
{
int pf_residual; /* square of postfiltered residual */
- int64_t temp1, temp2;
+ int temp1, temp2;
ppf->index = lag;
/* 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_clipl_int32(temp1 + temp2 + (1 << 15)) >> 16;
+ pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16;
if (tgt_eng >= pf_residual << 1) {
temp1 = 0x7fff;
int16_t scale;
int i;
- int64_t temp1, temp2;
+ int temp1, temp2;
/*
* 0 - target energy
* 4 - backward residual energy
*/
int energy[5] = {0, 0, 0, 0, 0};
- int16_t *buf = p->excitation + offset;
- int fwd_lag = autocorr_max(p, offset, &energy[1], pitch_lag,
+ 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(p, offset, &energy[3], pitch_lag,
+ int back_lag = autocorr_max(buf, offset, &energy[3], pitch_lag,
SUBFRAME_LEN, -1);
ppf->index = 0;
scale = normalize_bits(temp1, 31);
for (i = 0; i < 5; i++)
- energy[i] = (energy[i] << scale) >> 16;
+ energy[i] = av_clipl_int32(energy[i] << scale) >> 16;
if (fwd_lag && !back_lag) { /* Case 1 */
comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
int *exc_eng, int *scale)
{
int offset = PITCH_MAX + 2 * SUBFRAME_LEN;
- int16_t *buf = p->excitation + offset;
+ int16_t *buf = p->audio + LPC_ORDER;
int index, ccr, tgt_eng, best_eng, temp;
- *scale = scale_vector(p->excitation, FRAME_LEN + PITCH_MAX);
+ *scale = scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX);
+ buf += offset;
/* Compute maximum backward cross-correlation */
ccr = 0;
- index = autocorr_max(p, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);
- ccr = av_clipl_int32((int64_t)ccr + (1 << 15)) >> 16;
+ 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_clipl_int32((int64_t)tgt_eng + (1 << 15)) >> 16;
+ *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_clipl_int32((int64_t)best_eng + (1 << 15)) >> 16;
+ best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;
temp = best_eng * *exc_eng >> 3;
- if (temp < ccr * ccr)
+ if (temp < ccr * ccr) {
return index;
- else
+ } else
return 0;
}
int16_t *vector_ptr = buf + PITCH_MAX;
/* Attenuate */
for (i = 0; i < lag; i++)
- vector_ptr[i - lag] = vector_ptr[i - lag] * 3 >> 2;
- av_memcpy_backptr((uint8_t*)vector_ptr, lag * sizeof(*vector_ptr),
- FRAME_LEN * sizeof(*vector_ptr));
- memcpy(out, vector_ptr, FRAME_LEN * sizeof(*vector_ptr));
+ 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;
* @param iir_coef IIR coefficients
* @param src source vector
* @param dest destination vector
+ * @param width width of the output, 16 bits(0) / 32 bits(1)
*/
-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));
- }
+#define iir_filter(fir_coef, iir_coef, src, dest, width)\
+{\
+ int m, n;\
+ int res_shift = 16 & ~-(width);\
+ int in_shift = 16 - res_shift;\
+\
+ for (m = 0; m < SUBFRAME_LEN; m++) {\
+ int64_t filter = 0;\
+ for (n = 1; n <= LPC_ORDER; n++) {\
+ filter -= (fir_coef)[n - 1] * (src)[m - n] -\
+ (iir_coef)[n - 1] * ((dest)[m - n] >> in_shift);\
+ }\
+\
+ (dest)[m] = av_clipl_int32(((src)[m] << 16) + (filter << 3) +\
+ (1 << 15)) >> res_shift;\
+ }\
}
/**
for (i = 0; i < SUBFRAME_LEN; i++) {
int temp = buf[i] >> 2;
temp *= temp;
- denom = av_clipl_int32((int64_t)denom + (temp << 1));
+ denom = av_sat_dadd32(denom, temp);
}
if (num && denom) {
*
* @param p the context
* @param lpc quantized lpc coefficients
- * @param buf output buffer
+ * @param buf input buffer
+ * @param dst output buffer
*/
-static void formant_postfilter(G723_1_Context *p, int16_t *lpc, int16_t *buf)
+static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
+ int16_t *buf, int16_t *dst)
{
- int16_t filter_coef[2][LPC_ORDER], *buf_ptr;
+ int16_t filter_coef[2][LPC_ORDER];
int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;
int i, j, k;
(1 << 14)) >> 15;
}
iir_filter(filter_coef[0], filter_coef[1], buf + i,
- filter_signal + i);
+ filter_signal + i, 1);
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));
+ memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));
+ memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int));
- buf_ptr = buf + LPC_ORDER;
+ buf += LPC_ORDER;
signal_ptr = filter_signal + LPC_ORDER;
for (i = 0; i < SUBFRAMES; i++) {
- int16_t temp_vector[SUBFRAME_LEN];
int temp;
int auto_corr[2];
int scale, energy;
/* Normalize */
- memcpy(temp_vector, buf_ptr, SUBFRAME_LEN * sizeof(*temp_vector));
- scale = scale_vector(temp_vector, SUBFRAME_LEN);
+ scale = scale_vector(dst, buf, SUBFRAME_LEN);
/* Compute auto correlation coefficients */
- auto_corr[0] = dot_product(temp_vector, temp_vector + 1,
- SUBFRAME_LEN - 1);
- auto_corr[1] = dot_product(temp_vector, temp_vector, SUBFRAME_LEN);
+ 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;
/* Compensation filter */
for (j = 0; j < SUBFRAME_LEN; j++) {
- buf_ptr[j] = av_clipl_int32((int64_t)signal_ptr[j] +
- ((signal_ptr[j - 1] >> 16) *
- temp << 1)) >> 16;
+ dst[j] = av_sat_dadd32(signal_ptr[j],
+ (signal_ptr[j - 1] >> 16) * temp) >> 16;
}
/* Compute normalized signal energy */
} else
energy = auto_corr[1] >> temp;
- gain_scale(p, buf_ptr, energy);
+ gain_scale(p, dst, energy);
- buf_ptr += SUBFRAME_LEN;
+ buf += SUBFRAME_LEN;
signal_ptr += SUBFRAME_LEN;
+ dst += SUBFRAME_LEN;
}
}
int16_t cur_lsp[LPC_ORDER];
int16_t lpc[SUBFRAMES * LPC_ORDER];
int16_t acb_vector[SUBFRAME_LEN];
- int16_t *vector_ptr;
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)
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;
+ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
+ return ret;
}
out = (int16_t *)p->frame.data[0];
/* Generate the excitation for the frame */
memcpy(p->excitation, p->prev_excitation,
PITCH_MAX * sizeof(*p->excitation));
- vector_ptr = p->excitation + PITCH_MAX;
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,
+ 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->pitch_lag[i >> 1], &p->subframe[i],
p->cur_rate);
/* Get the total excitation */
for (j = 0; j < SUBFRAME_LEN; j++) {
- vector_ptr[j] = av_clip_int16(vector_ptr[j] << 1);
- vector_ptr[j] = av_clip_int16(vector_ptr[j] +
- acb_vector[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;
- /* Save the excitation */
- memcpy(p->audio + LPC_ORDER, vector_ptr, FRAME_LEN * sizeof(*p->audio));
-
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);
- }
-
- /* Restore the original excitation */
- memcpy(p->excitation, p->prev_excitation,
- PITCH_MAX * sizeof(*p->excitation));
- memcpy(vector_ptr, p->audio + LPC_ORDER, FRAME_LEN * sizeof(*vector_ptr));
- /* Peform pitch postfiltering */
- if (p->postfilter)
for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,
vector_ptr + i,
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, p->audio + LPC_ORDER, p->interp_index,
+ 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));
}
}
- /* Save the excitation for the next frame */
- memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
- PITCH_MAX * sizeof(*p->excitation));
} else {
memset(out, 0, FRAME_LEN * 2);
av_log(avctx, AV_LOG_WARNING,
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],
- p->audio + i, SUBFRAME_LEN, 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);
- memcpy(p->frame.data[0], p->audio + LPC_ORDER, FRAME_LEN * 2);
+ 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);
.init = g723_1_decode_init,
.decode = g723_1_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
- .capabilities = CODEC_CAP_SUBFRAMES,
+ .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
.priv_class = &g723_1dec_class,
};
+
+#if CONFIG_G723_1_ENCODER
+#define BITSTREAM_WRITER_LE
+#include "put_bits.h"
+
+static av_cold int g723_1_encode_init(AVCodecContext *avctx)
+{
+ G723_1_Context *p = avctx->priv_data;
+
+ if (avctx->sample_rate != 8000) {
+ av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");
+ return -1;
+ }
+
+ if (avctx->channels != 1) {
+ av_log(avctx, AV_LOG_ERROR, "Only mono supported\n");
+ return AVERROR(EINVAL);
+ }
+
+ if (avctx->bit_rate == 6300) {
+ p->cur_rate = RATE_6300;
+ } else if (avctx->bit_rate == 5300) {
+ av_log(avctx, AV_LOG_ERROR, "Bitrate not supported yet, use 6.3k\n");
+ return AVERROR_PATCHWELCOME;
+ } else {
+ av_log(avctx, AV_LOG_ERROR,
+ "Bitrate not supported, use 6.3k\n");
+ return AVERROR(EINVAL);
+ }
+ avctx->frame_size = 240;
+ memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t));
+
+ return 0;
+}
+
+/**
+ * Remove DC component from the input signal.
+ *
+ * @param buf input signal
+ * @param fir zero memory
+ * @param iir pole memory
+ */
+static void highpass_filter(int16_t *buf, int16_t *fir, int *iir)
+{
+ int i;
+ for (i = 0; i < FRAME_LEN; i++) {
+ *iir = (buf[i] << 15) + ((-*fir) << 15) + MULL2(*iir, 0x7f00);
+ *fir = buf[i];
+ buf[i] = av_clipl_int32((int64_t)*iir + (1 << 15)) >> 16;
+ }
+}
+
+/**
+ * Estimate autocorrelation of the input vector.
+ *
+ * @param buf input buffer
+ * @param autocorr autocorrelation coefficients vector
+ */
+static void comp_autocorr(int16_t *buf, int16_t *autocorr)
+{
+ int i, scale, temp;
+ int16_t vector[LPC_FRAME];
+
+ scale_vector(vector, buf, LPC_FRAME);
+
+ /* Apply the Hamming window */
+ for (i = 0; i < LPC_FRAME; i++)
+ vector[i] = (vector[i] * hamming_window[i] + (1 << 14)) >> 15;
+
+ /* Compute the first autocorrelation coefficient */
+ temp = ff_dot_product(vector, vector, LPC_FRAME);
+
+ /* Apply a white noise correlation factor of (1025/1024) */
+ temp += temp >> 10;
+
+ /* Normalize */
+ scale = normalize_bits_int32(temp);
+ autocorr[0] = av_clipl_int32((int64_t)(temp << scale) +
+ (1 << 15)) >> 16;
+
+ /* Compute the remaining coefficients */
+ if (!autocorr[0]) {
+ memset(autocorr + 1, 0, LPC_ORDER * sizeof(int16_t));
+ } else {
+ for (i = 1; i <= LPC_ORDER; i++) {
+ temp = ff_dot_product(vector, vector + i, LPC_FRAME - i);
+ temp = MULL2((temp << scale), binomial_window[i - 1]);
+ autocorr[i] = av_clipl_int32((int64_t)temp + (1 << 15)) >> 16;
+ }
+ }
+}
+
+/**
+ * Use Levinson-Durbin recursion to compute LPC coefficients from
+ * autocorrelation values.
+ *
+ * @param lpc LPC coefficients vector
+ * @param autocorr autocorrelation coefficients vector
+ * @param error prediction error
+ */
+static void levinson_durbin(int16_t *lpc, int16_t *autocorr, int16_t error)
+{
+ int16_t vector[LPC_ORDER];
+ int16_t partial_corr;
+ int i, j, temp;
+
+ memset(lpc, 0, LPC_ORDER * sizeof(int16_t));
+
+ for (i = 0; i < LPC_ORDER; i++) {
+ /* Compute the partial correlation coefficient */
+ temp = 0;
+ for (j = 0; j < i; j++)
+ temp -= lpc[j] * autocorr[i - j - 1];
+ temp = ((autocorr[i] << 13) + temp) << 3;
+
+ if (FFABS(temp) >= (error << 16))
+ break;
+
+ partial_corr = temp / (error << 1);
+
+ lpc[i] = av_clipl_int32((int64_t)(partial_corr << 14) +
+ (1 << 15)) >> 16;
+
+ /* Update the prediction error */
+ temp = MULL2(temp, partial_corr);
+ error = av_clipl_int32((int64_t)(error << 16) - temp +
+ (1 << 15)) >> 16;
+
+ memcpy(vector, lpc, i * sizeof(int16_t));
+ for (j = 0; j < i; j++) {
+ temp = partial_corr * vector[i - j - 1] << 1;
+ lpc[j] = av_clipl_int32((int64_t)(lpc[j] << 16) - temp +
+ (1 << 15)) >> 16;
+ }
+ }
+}
+
+/**
+ * Calculate LPC coefficients for the current frame.
+ *
+ * @param buf current frame
+ * @param prev_data 2 trailing subframes of the previous frame
+ * @param lpc LPC coefficients vector
+ */
+static void comp_lpc_coeff(int16_t *buf, int16_t *lpc)
+{
+ int16_t autocorr[(LPC_ORDER + 1) * SUBFRAMES];
+ int16_t *autocorr_ptr = autocorr;
+ int16_t *lpc_ptr = lpc;
+ int i, j;
+
+ for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
+ comp_autocorr(buf + i, autocorr_ptr);
+ levinson_durbin(lpc_ptr, autocorr_ptr + 1, autocorr_ptr[0]);
+
+ lpc_ptr += LPC_ORDER;
+ autocorr_ptr += LPC_ORDER + 1;
+ }
+}
+
+static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp)
+{
+ int f[LPC_ORDER + 2]; ///< coefficients of the sum and difference
+ ///< polynomials (F1, F2) ordered as
+ ///< f1[0], f2[0], ...., f1[5], f2[5]
+
+ int max, shift, cur_val, prev_val, count, p;
+ int i, j;
+ int64_t temp;
+
+ /* Initialize f1[0] and f2[0] to 1 in Q25 */
+ for (i = 0; i < LPC_ORDER; i++)
+ lsp[i] = (lpc[i] * bandwidth_expand[i] + (1 << 14)) >> 15;
+
+ /* Apply bandwidth expansion on the LPC coefficients */
+ f[0] = f[1] = 1 << 25;
+
+ /* Compute the remaining coefficients */
+ for (i = 0; i < LPC_ORDER / 2; i++) {
+ /* f1 */
+ f[2 * i + 2] = -f[2 * i] - ((lsp[i] + lsp[LPC_ORDER - 1 - i]) << 12);
+ /* f2 */
+ f[2 * i + 3] = f[2 * i + 1] - ((lsp[i] - lsp[LPC_ORDER - 1 - i]) << 12);
+ }
+
+ /* Divide f1[5] and f2[5] by 2 for use in polynomial evaluation */
+ f[LPC_ORDER] >>= 1;
+ f[LPC_ORDER + 1] >>= 1;
+
+ /* Normalize and shorten */
+ max = FFABS(f[0]);
+ for (i = 1; i < LPC_ORDER + 2; i++)
+ max = FFMAX(max, FFABS(f[i]));
+
+ shift = normalize_bits_int32(max);
+
+ for (i = 0; i < LPC_ORDER + 2; i++)
+ f[i] = av_clipl_int32((int64_t)(f[i] << shift) + (1 << 15)) >> 16;
+
+ /**
+ * Evaluate F1 and F2 at uniform intervals of pi/256 along the
+ * unit circle and check for zero crossings.
+ */
+ p = 0;
+ temp = 0;
+ for (i = 0; i <= LPC_ORDER / 2; i++)
+ temp += f[2 * i] * cos_tab[0];
+ prev_val = av_clipl_int32(temp << 1);
+ count = 0;
+ for ( i = 1; i < COS_TBL_SIZE / 2; i++) {
+ /* Evaluate */
+ temp = 0;
+ for (j = 0; j <= LPC_ORDER / 2; j++)
+ temp += f[LPC_ORDER - 2 * j + p] * cos_tab[i * j % COS_TBL_SIZE];
+ cur_val = av_clipl_int32(temp << 1);
+
+ /* Check for sign change, indicating a zero crossing */
+ if ((cur_val ^ prev_val) < 0) {
+ int abs_cur = FFABS(cur_val);
+ int abs_prev = FFABS(prev_val);
+ int sum = abs_cur + abs_prev;
+
+ shift = normalize_bits_int32(sum);
+ sum <<= shift;
+ abs_prev = abs_prev << shift >> 8;
+ lsp[count++] = ((i - 1) << 7) + (abs_prev >> 1) / (sum >> 16);
+
+ if (count == LPC_ORDER)
+ break;
+
+ /* Switch between sum and difference polynomials */
+ p ^= 1;
+
+ /* Evaluate */
+ temp = 0;
+ for (j = 0; j <= LPC_ORDER / 2; j++){
+ temp += f[LPC_ORDER - 2 * j + p] *
+ cos_tab[i * j % COS_TBL_SIZE];
+ }
+ cur_val = av_clipl_int32(temp<<1);
+ }
+ prev_val = cur_val;
+ }
+
+ if (count != LPC_ORDER)
+ memcpy(lsp, prev_lsp, LPC_ORDER * sizeof(int16_t));
+}
+
+/**
+ * Quantize the current LSP subvector.
+ *
+ * @param num band number
+ * @param offset offset of the current subvector in an LPC_ORDER vector
+ * @param size size of the current subvector
+ */
+#define get_index(num, offset, size) \
+{\
+ int error, max = -1;\
+ int16_t temp[4];\
+ int i, j;\
+ for (i = 0; i < LSP_CB_SIZE; i++) {\
+ for (j = 0; j < size; j++){\
+ temp[j] = (weight[j + (offset)] * lsp_band##num[i][j] +\
+ (1 << 14)) >> 15;\
+ }\
+ error = dot_product(lsp + (offset), temp, size) << 1;\
+ error -= dot_product(lsp_band##num[i], temp, size);\
+ if (error > max) {\
+ max = error;\
+ lsp_index[num] = i;\
+ }\
+ }\
+}
+
+/**
+ * Vector quantize the LSP frequencies.
+ *
+ * @param lsp the current lsp vector
+ * @param prev_lsp the previous lsp vector
+ */
+static void lsp_quantize(uint8_t *lsp_index, int16_t *lsp, int16_t *prev_lsp)
+{
+ int16_t weight[LPC_ORDER];
+ int16_t min, max;
+ int shift, i;
+
+ /* Calculate the VQ weighting vector */
+ weight[0] = (1 << 20) / (lsp[1] - lsp[0]);
+ weight[LPC_ORDER - 1] = (1 << 20) /
+ (lsp[LPC_ORDER - 1] - lsp[LPC_ORDER - 2]);
+
+ for (i = 1; i < LPC_ORDER - 1; i++) {
+ min = FFMIN(lsp[i] - lsp[i - 1], lsp[i + 1] - lsp[i]);
+ if (min > 0x20)
+ weight[i] = (1 << 20) / min;
+ else
+ weight[i] = INT16_MAX;
+ }
+
+ /* Normalize */
+ max = 0;
+ for (i = 0; i < LPC_ORDER; i++)
+ max = FFMAX(weight[i], max);
+
+ shift = normalize_bits_int16(max);
+ for (i = 0; i < LPC_ORDER; i++) {
+ weight[i] <<= shift;
+ }
+
+ /* Compute the VQ target vector */
+ for (i = 0; i < LPC_ORDER; i++) {
+ lsp[i] -= dc_lsp[i] +
+ (((prev_lsp[i] - dc_lsp[i]) * 12288 + (1 << 14)) >> 15);
+ }
+
+ get_index(0, 0, 3);
+ get_index(1, 3, 3);
+ get_index(2, 6, 4);
+}
+
+/**
+ * Apply the formant perceptual weighting filter.
+ *
+ * @param flt_coef filter coefficients
+ * @param unq_lpc unquantized lpc vector
+ */
+static void perceptual_filter(G723_1_Context *p, int16_t *flt_coef,
+ int16_t *unq_lpc, int16_t *buf)
+{
+ int16_t vector[FRAME_LEN + LPC_ORDER];
+ int i, j, k, l = 0;
+
+ memcpy(buf, p->iir_mem, sizeof(int16_t) * LPC_ORDER);
+ memcpy(vector, p->fir_mem, sizeof(int16_t) * LPC_ORDER);
+ memcpy(vector + LPC_ORDER, buf + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
+
+ for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
+ for (k = 0; k < LPC_ORDER; k++) {
+ flt_coef[k + 2 * l] = (unq_lpc[k + l] * percept_flt_tbl[0][k] +
+ (1 << 14)) >> 15;
+ flt_coef[k + 2 * l + LPC_ORDER] = (unq_lpc[k + l] *
+ percept_flt_tbl[1][k] +
+ (1 << 14)) >> 15;
+ }
+ iir_filter(flt_coef + 2 * l, flt_coef + 2 * l + LPC_ORDER, vector + i,
+ buf + i, 0);
+ l += LPC_ORDER;
+ }
+ memcpy(p->iir_mem, buf + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
+ memcpy(p->fir_mem, vector + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
+}
+
+/**
+ * Estimate the open loop pitch period.
+ *
+ * @param buf perceptually weighted speech
+ * @param start estimation is carried out from this position
+ */
+static int estimate_pitch(int16_t *buf, int start)
+{
+ int max_exp = 32;
+ int max_ccr = 0x4000;
+ int max_eng = 0x7fff;
+ int index = PITCH_MIN;
+ int offset = start - PITCH_MIN + 1;
+
+ int ccr, eng, orig_eng, ccr_eng, exp;
+ int diff, temp;
+
+ int i;
+
+ orig_eng = ff_dot_product(buf + offset, buf + offset, HALF_FRAME_LEN);
+
+ for (i = PITCH_MIN; i <= PITCH_MAX - 3; i++) {
+ offset--;
+
+ /* Update energy and compute correlation */
+ orig_eng += buf[offset] * buf[offset] -
+ buf[offset + HALF_FRAME_LEN] * buf[offset + HALF_FRAME_LEN];
+ ccr = ff_dot_product(buf + start, buf + offset, HALF_FRAME_LEN);
+ if (ccr <= 0)
+ continue;
+
+ /* Split into mantissa and exponent to maintain precision */
+ exp = normalize_bits_int32(ccr);
+ ccr = av_clipl_int32((int64_t)(ccr << exp) + (1 << 15)) >> 16;
+ exp <<= 1;
+ ccr *= ccr;
+ temp = normalize_bits_int32(ccr);
+ ccr = ccr << temp >> 16;
+ exp += temp;
+
+ temp = normalize_bits_int32(orig_eng);
+ eng = av_clipl_int32((int64_t)(orig_eng << temp) + (1 << 15)) >> 16;
+ exp -= temp;
+
+ if (ccr >= eng) {
+ exp--;
+ ccr >>= 1;
+ }
+ if (exp > max_exp)
+ continue;
+
+ if (exp + 1 < max_exp)
+ goto update;
+
+ /* Equalize exponents before comparison */
+ if (exp + 1 == max_exp)
+ temp = max_ccr >> 1;
+ else
+ temp = max_ccr;
+ ccr_eng = ccr * max_eng;
+ diff = ccr_eng - eng * temp;
+ if (diff > 0 && (i - index < PITCH_MIN || diff > ccr_eng >> 2)) {
+update:
+ index = i;
+ max_exp = exp;
+ max_ccr = ccr;
+ max_eng = eng;
+ }
+ }
+ return index;
+}
+
+/**
+ * Compute harmonic noise filter parameters.
+ *
+ * @param buf perceptually weighted speech
+ * @param pitch_lag open loop pitch period
+ * @param hf harmonic filter parameters
+ */
+static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf)
+{
+ int ccr, eng, max_ccr, max_eng;
+ int exp, max, diff;
+ int energy[15];
+ int i, j;
+
+ for (i = 0, j = pitch_lag - 3; j <= pitch_lag + 3; i++, j++) {
+ /* Compute residual energy */
+ energy[i << 1] = ff_dot_product(buf - j, buf - j, SUBFRAME_LEN);
+ /* Compute correlation */
+ energy[(i << 1) + 1] = ff_dot_product(buf, buf - j, SUBFRAME_LEN);
+ }
+
+ /* Compute target energy */
+ energy[14] = ff_dot_product(buf, buf, SUBFRAME_LEN);
+
+ /* Normalize */
+ max = 0;
+ for (i = 0; i < 15; i++)
+ max = FFMAX(max, FFABS(energy[i]));
+
+ exp = normalize_bits_int32(max);
+ for (i = 0; i < 15; i++) {
+ energy[i] = av_clipl_int32((int64_t)(energy[i] << exp) +
+ (1 << 15)) >> 16;
+ }
+
+ hf->index = -1;
+ hf->gain = 0;
+ max_ccr = 1;
+ max_eng = 0x7fff;
+
+ for (i = 0; i <= 6; i++) {
+ eng = energy[i << 1];
+ ccr = energy[(i << 1) + 1];
+
+ if (ccr <= 0)
+ continue;
+
+ ccr = (ccr * ccr + (1 << 14)) >> 15;
+ diff = ccr * max_eng - eng * max_ccr;
+ if (diff > 0) {
+ max_ccr = ccr;
+ max_eng = eng;
+ hf->index = i;
+ }
+ }
+
+ if (hf->index == -1) {
+ hf->index = pitch_lag;
+ return;
+ }
+
+ eng = energy[14] * max_eng;
+ eng = (eng >> 2) + (eng >> 3);
+ ccr = energy[(hf->index << 1) + 1] * energy[(hf->index << 1) + 1];
+ if (eng < ccr) {
+ eng = energy[(hf->index << 1) + 1];
+
+ if (eng >= max_eng)
+ hf->gain = 0x2800;
+ else
+ hf->gain = ((eng << 15) / max_eng * 0x2800 + (1 << 14)) >> 15;
+ }
+ hf->index += pitch_lag - 3;
+}
+
+/**
+ * Apply the harmonic noise shaping filter.
+ *
+ * @param hf filter parameters
+ */
+static void harmonic_filter(HFParam *hf, int16_t *src, int16_t *dest)
+{
+ int i;
+
+ for (i = 0; i < SUBFRAME_LEN; i++) {
+ int64_t temp = hf->gain * src[i - hf->index] << 1;
+ dest[i] = av_clipl_int32((src[i] << 16) - temp + (1 << 15)) >> 16;
+ }
+}
+
+static void harmonic_noise_sub(HFParam *hf, int16_t *src, int16_t *dest)
+{
+ int i;
+ for (i = 0; i < SUBFRAME_LEN; i++) {
+ int64_t temp = hf->gain * src[i - hf->index] << 1;
+ dest[i] = av_clipl_int32(((dest[i] - src[i]) << 16) + temp +
+ (1 << 15)) >> 16;
+
+ }
+}
+
+/**
+ * Combined synthesis and formant perceptual weighting filer.
+ *
+ * @param qnt_lpc quantized lpc coefficients
+ * @param perf_lpc perceptual filter coefficients
+ * @param perf_fir perceptual filter fir memory
+ * @param perf_iir perceptual filter iir memory
+ * @param scale the filter output will be scaled by 2^scale
+ */
+static void synth_percept_filter(int16_t *qnt_lpc, int16_t *perf_lpc,
+ int16_t *perf_fir, int16_t *perf_iir,
+ int16_t *src, int16_t *dest, int scale)
+{
+ int i, j;
+ int16_t buf_16[SUBFRAME_LEN + LPC_ORDER];
+ int64_t buf[SUBFRAME_LEN];
+
+ int16_t *bptr_16 = buf_16 + LPC_ORDER;
+
+ memcpy(buf_16, perf_fir, sizeof(int16_t) * LPC_ORDER);
+ memcpy(dest - LPC_ORDER, perf_iir, sizeof(int16_t) * LPC_ORDER);
+
+ for (i = 0; i < SUBFRAME_LEN; i++) {
+ int64_t temp = 0;
+ for (j = 1; j <= LPC_ORDER; j++)
+ temp -= qnt_lpc[j - 1] * bptr_16[i - j];
+
+ buf[i] = (src[i] << 15) + (temp << 3);
+ bptr_16[i] = av_clipl_int32(buf[i] + (1 << 15)) >> 16;
+ }
+
+ for (i = 0; i < SUBFRAME_LEN; i++) {
+ int64_t fir = 0, iir = 0;
+ for (j = 1; j <= LPC_ORDER; j++) {
+ fir -= perf_lpc[j - 1] * bptr_16[i - j];
+ iir += perf_lpc[j + LPC_ORDER - 1] * dest[i - j];
+ }
+ dest[i] = av_clipl_int32(((buf[i] + (fir << 3)) << scale) + (iir << 3) +
+ (1 << 15)) >> 16;
+ }
+ memcpy(perf_fir, buf_16 + SUBFRAME_LEN, sizeof(int16_t) * LPC_ORDER);
+ memcpy(perf_iir, dest + SUBFRAME_LEN - LPC_ORDER,
+ sizeof(int16_t) * LPC_ORDER);
+}
+
+/**
+ * Compute the adaptive codebook contribution.
+ *
+ * @param buf input signal
+ * @param index the current subframe index
+ */
+static void acb_search(G723_1_Context *p, int16_t *residual,
+ int16_t *impulse_resp, int16_t *buf,
+ int index)
+{
+
+ int16_t flt_buf[PITCH_ORDER][SUBFRAME_LEN];
+
+ const int16_t *cb_tbl = adaptive_cb_gain85;
+
+ int ccr_buf[PITCH_ORDER * SUBFRAMES << 2];
+
+ int pitch_lag = p->pitch_lag[index >> 1];
+ int acb_lag = 1;
+ int acb_gain = 0;
+ int odd_frame = index & 1;
+ int iter = 3 + odd_frame;
+ int count = 0;
+ int tbl_size = 85;
+
+ int i, j, k, l, max;
+ int64_t temp;
+
+ if (!odd_frame) {
+ if (pitch_lag == PITCH_MIN)
+ pitch_lag++;
+ else
+ pitch_lag = FFMIN(pitch_lag, PITCH_MAX - 5);
+ }
+
+ for (i = 0; i < iter; i++) {
+ get_residual(residual, p->prev_excitation, pitch_lag + i - 1);
+
+ for (j = 0; j < SUBFRAME_LEN; j++) {
+ temp = 0;
+ for (k = 0; k <= j; k++)
+ temp += residual[PITCH_ORDER - 1 + k] * impulse_resp[j - k];
+ flt_buf[PITCH_ORDER - 1][j] = av_clipl_int32((temp << 1) +
+ (1 << 15)) >> 16;
+ }
+
+ for (j = PITCH_ORDER - 2; j >= 0; j--) {
+ flt_buf[j][0] = ((residual[j] << 13) + (1 << 14)) >> 15;
+ for (k = 1; k < SUBFRAME_LEN; k++) {
+ temp = (flt_buf[j + 1][k - 1] << 15) +
+ residual[j] * impulse_resp[k];
+ flt_buf[j][k] = av_clipl_int32((temp << 1) + (1 << 15)) >> 16;
+ }
+ }
+
+ /* Compute crosscorrelation with the signal */
+ for (j = 0; j < PITCH_ORDER; j++) {
+ temp = ff_dot_product(buf, flt_buf[j], SUBFRAME_LEN);
+ ccr_buf[count++] = av_clipl_int32(temp << 1);
+ }
+
+ /* Compute energies */
+ for (j = 0; j < PITCH_ORDER; j++) {
+ ccr_buf[count++] = dot_product(flt_buf[j], flt_buf[j],
+ SUBFRAME_LEN);
+ }
+
+ for (j = 1; j < PITCH_ORDER; j++) {
+ for (k = 0; k < j; k++) {
+ temp = ff_dot_product(flt_buf[j], flt_buf[k], SUBFRAME_LEN);
+ ccr_buf[count++] = av_clipl_int32(temp<<2);
+ }
+ }
+ }
+
+ /* Normalize and shorten */
+ max = 0;
+ for (i = 0; i < 20 * iter; i++)
+ max = FFMAX(max, FFABS(ccr_buf[i]));
+
+ temp = normalize_bits_int32(max);
+
+ for (i = 0; i < 20 * iter; i++){
+ ccr_buf[i] = av_clipl_int32((int64_t)(ccr_buf[i] << temp) +
+ (1 << 15)) >> 16;
+ }
+
+ max = 0;
+ for (i = 0; i < iter; i++) {
+ /* Select quantization table */
+ if (!odd_frame && pitch_lag + i - 1 >= SUBFRAME_LEN - 2 ||
+ odd_frame && pitch_lag >= SUBFRAME_LEN - 2) {
+ cb_tbl = adaptive_cb_gain170;
+ tbl_size = 170;
+ }
+
+ for (j = 0, k = 0; j < tbl_size; j++, k += 20) {
+ temp = 0;
+ for (l = 0; l < 20; l++)
+ temp += ccr_buf[20 * i + l] * cb_tbl[k + l];
+ temp = av_clipl_int32(temp);
+
+ if (temp > max) {
+ max = temp;
+ acb_gain = j;
+ acb_lag = i;
+ }
+ }
+ }
+
+ if (!odd_frame) {
+ pitch_lag += acb_lag - 1;
+ acb_lag = 1;
+ }
+
+ p->pitch_lag[index >> 1] = pitch_lag;
+ p->subframe[index].ad_cb_lag = acb_lag;
+ p->subframe[index].ad_cb_gain = acb_gain;
+}
+
+/**
+ * Subtract the adaptive codebook contribution from the input
+ * to obtain the residual.
+ *
+ * @param buf target vector
+ */
+static void sub_acb_contrib(int16_t *residual, int16_t *impulse_resp,
+ int16_t *buf)
+{
+ int i, j;
+ /* Subtract adaptive CB contribution to obtain the residual */
+ for (i = 0; i < SUBFRAME_LEN; i++) {
+ int64_t temp = buf[i] << 14;
+ for (j = 0; j <= i; j++)
+ temp -= residual[j] * impulse_resp[i - j];
+
+ buf[i] = av_clipl_int32((temp << 2) + (1 << 15)) >> 16;
+ }
+}
+
+/**
+ * Quantize the residual signal using the fixed codebook (MP-MLQ).
+ *
+ * @param optim optimized fixed codebook parameters
+ * @param buf excitation vector
+ */
+static void get_fcb_param(FCBParam *optim, int16_t *impulse_resp,
+ int16_t *buf, int pulse_cnt, int pitch_lag)
+{
+ FCBParam param;
+ int16_t impulse_r[SUBFRAME_LEN];
+ int16_t temp_corr[SUBFRAME_LEN];
+ int16_t impulse_corr[SUBFRAME_LEN];
+
+ int ccr1[SUBFRAME_LEN];
+ int ccr2[SUBFRAME_LEN];
+ int amp, err, max, max_amp_index, min, scale, i, j, k, l;
+
+ int64_t temp;
+
+ /* Update impulse response */
+ memcpy(impulse_r, impulse_resp, sizeof(int16_t) * SUBFRAME_LEN);
+ param.dirac_train = 0;
+ if (pitch_lag < SUBFRAME_LEN - 2) {
+ param.dirac_train = 1;
+ gen_dirac_train(impulse_r, pitch_lag);
+ }
+
+ for (i = 0; i < SUBFRAME_LEN; i++)
+ temp_corr[i] = impulse_r[i] >> 1;
+
+ /* Compute impulse response autocorrelation */
+ temp = dot_product(temp_corr, temp_corr, SUBFRAME_LEN);
+
+ scale = normalize_bits_int32(temp);
+ impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
+
+ for (i = 1; i < SUBFRAME_LEN; i++) {
+ temp = dot_product(temp_corr + i, temp_corr, SUBFRAME_LEN - i);
+ impulse_corr[i] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
+ }
+
+ /* Compute crosscorrelation of impulse response with residual signal */
+ scale -= 4;
+ for (i = 0; i < SUBFRAME_LEN; i++){
+ temp = dot_product(buf + i, impulse_r, SUBFRAME_LEN - i);
+ if (scale < 0)
+ ccr1[i] = temp >> -scale;
+ else
+ ccr1[i] = av_clipl_int32(temp << scale);
+ }
+
+ /* Search loop */
+ for (i = 0; i < GRID_SIZE; i++) {
+ /* Maximize the crosscorrelation */
+ max = 0;
+ for (j = i; j < SUBFRAME_LEN; j += GRID_SIZE) {
+ temp = FFABS(ccr1[j]);
+ if (temp >= max) {
+ max = temp;
+ param.pulse_pos[0] = j;
+ }
+ }
+
+ /* Quantize the gain (max crosscorrelation/impulse_corr[0]) */
+ amp = max;
+ min = 1 << 30;
+ max_amp_index = GAIN_LEVELS - 2;
+ for (j = max_amp_index; j >= 2; j--) {
+ temp = av_clipl_int32((int64_t)fixed_cb_gain[j] *
+ impulse_corr[0] << 1);
+ temp = FFABS(temp - amp);
+ if (temp < min) {
+ min = temp;
+ max_amp_index = j;
+ }
+ }
+
+ max_amp_index--;
+ /* Select additional gain values */
+ for (j = 1; j < 5; j++) {
+ for (k = i; k < SUBFRAME_LEN; k += GRID_SIZE) {
+ temp_corr[k] = 0;
+ ccr2[k] = ccr1[k];
+ }
+ param.amp_index = max_amp_index + j - 2;
+ amp = fixed_cb_gain[param.amp_index];
+
+ param.pulse_sign[0] = (ccr2[param.pulse_pos[0]] < 0) ? -amp : amp;
+ temp_corr[param.pulse_pos[0]] = 1;
+
+ for (k = 1; k < pulse_cnt; k++) {
+ max = -1 << 30;
+ for (l = i; l < SUBFRAME_LEN; l += GRID_SIZE) {
+ if (temp_corr[l])
+ continue;
+ temp = impulse_corr[FFABS(l - param.pulse_pos[k - 1])];
+ temp = av_clipl_int32((int64_t)temp *
+ param.pulse_sign[k - 1] << 1);
+ ccr2[l] -= temp;
+ temp = FFABS(ccr2[l]);
+ if (temp > max) {
+ max = temp;
+ param.pulse_pos[k] = l;
+ }
+ }
+
+ param.pulse_sign[k] = (ccr2[param.pulse_pos[k]] < 0) ?
+ -amp : amp;
+ temp_corr[param.pulse_pos[k]] = 1;
+ }
+
+ /* Create the error vector */
+ memset(temp_corr, 0, sizeof(int16_t) * SUBFRAME_LEN);
+
+ for (k = 0; k < pulse_cnt; k++)
+ temp_corr[param.pulse_pos[k]] = param.pulse_sign[k];
+
+ for (k = SUBFRAME_LEN - 1; k >= 0; k--) {
+ temp = 0;
+ for (l = 0; l <= k; l++) {
+ int prod = av_clipl_int32((int64_t)temp_corr[l] *
+ impulse_r[k - l] << 1);
+ temp = av_clipl_int32(temp + prod);
+ }
+ temp_corr[k] = temp << 2 >> 16;
+ }
+
+ /* Compute square of error */
+ err = 0;
+ for (k = 0; k < SUBFRAME_LEN; k++) {
+ int64_t prod;
+ prod = av_clipl_int32((int64_t)buf[k] * temp_corr[k] << 1);
+ err = av_clipl_int32(err - prod);
+ prod = av_clipl_int32((int64_t)temp_corr[k] * temp_corr[k]);
+ err = av_clipl_int32(err + prod);
+ }
+
+ /* Minimize */
+ if (err < optim->min_err) {
+ optim->min_err = err;
+ optim->grid_index = i;
+ optim->amp_index = param.amp_index;
+ optim->dirac_train = param.dirac_train;
+
+ for (k = 0; k < pulse_cnt; k++) {
+ optim->pulse_sign[k] = param.pulse_sign[k];
+ optim->pulse_pos[k] = param.pulse_pos[k];
+ }
+ }
+ }
+ }
+}
+
+/**
+ * Encode the pulse position and gain of the current subframe.
+ *
+ * @param optim optimized fixed CB parameters
+ * @param buf excitation vector
+ */
+static void pack_fcb_param(G723_1_Subframe *subfrm, FCBParam *optim,
+ int16_t *buf, int pulse_cnt)
+{
+ int i, j;
+
+ j = PULSE_MAX - pulse_cnt;
+
+ subfrm->pulse_sign = 0;
+ subfrm->pulse_pos = 0;
+
+ for (i = 0; i < SUBFRAME_LEN >> 1; i++) {
+ int val = buf[optim->grid_index + (i << 1)];
+ if (!val) {
+ subfrm->pulse_pos += combinatorial_table[j][i];
+ } else {
+ subfrm->pulse_sign <<= 1;
+ if (val < 0) subfrm->pulse_sign++;
+ j++;
+
+ if (j == PULSE_MAX) break;
+ }
+ }
+ subfrm->amp_index = optim->amp_index;
+ subfrm->grid_index = optim->grid_index;
+ subfrm->dirac_train = optim->dirac_train;
+}
+
+/**
+ * Compute the fixed codebook excitation.
+ *
+ * @param buf target vector
+ * @param impulse_resp impulse response of the combined filter
+ */
+static void fcb_search(G723_1_Context *p, int16_t *impulse_resp,
+ int16_t *buf, int index)
+{
+ FCBParam optim;
+ int pulse_cnt = pulses[index];
+ int i;
+
+ optim.min_err = 1 << 30;
+ get_fcb_param(&optim, impulse_resp, buf, pulse_cnt, SUBFRAME_LEN);
+
+ if (p->pitch_lag[index >> 1] < SUBFRAME_LEN - 2) {
+ get_fcb_param(&optim, impulse_resp, buf, pulse_cnt,
+ p->pitch_lag[index >> 1]);
+ }
+
+ /* Reconstruct the excitation */
+ memset(buf, 0, sizeof(int16_t) * SUBFRAME_LEN);
+ for (i = 0; i < pulse_cnt; i++)
+ buf[optim.pulse_pos[i]] = optim.pulse_sign[i];
+
+ pack_fcb_param(&p->subframe[index], &optim, buf, pulse_cnt);
+
+ if (optim.dirac_train)
+ gen_dirac_train(buf, p->pitch_lag[index >> 1]);
+}
+
+/**
+ * Pack the frame parameters into output bitstream.
+ *
+ * @param frame output buffer
+ * @param size size of the buffer
+ */
+static int pack_bitstream(G723_1_Context *p, unsigned char *frame, int size)
+{
+ PutBitContext pb;
+ int info_bits, i, temp;
+
+ init_put_bits(&pb, frame, size);
+
+ if (p->cur_rate == RATE_6300) {
+ info_bits = 0;
+ put_bits(&pb, 2, info_bits);
+ }
+
+ put_bits(&pb, 8, p->lsp_index[2]);
+ put_bits(&pb, 8, p->lsp_index[1]);
+ put_bits(&pb, 8, p->lsp_index[0]);
+
+ put_bits(&pb, 7, p->pitch_lag[0] - PITCH_MIN);
+ put_bits(&pb, 2, p->subframe[1].ad_cb_lag);
+ put_bits(&pb, 7, p->pitch_lag[1] - PITCH_MIN);
+ put_bits(&pb, 2, p->subframe[3].ad_cb_lag);
+
+ /* Write 12 bit combined gain */
+ for (i = 0; i < SUBFRAMES; i++) {
+ temp = p->subframe[i].ad_cb_gain * GAIN_LEVELS +
+ p->subframe[i].amp_index;
+ if (p->cur_rate == RATE_6300)
+ temp += p->subframe[i].dirac_train << 11;
+ put_bits(&pb, 12, temp);
+ }
+
+ put_bits(&pb, 1, p->subframe[0].grid_index);
+ put_bits(&pb, 1, p->subframe[1].grid_index);
+ put_bits(&pb, 1, p->subframe[2].grid_index);
+ put_bits(&pb, 1, p->subframe[3].grid_index);
+
+ if (p->cur_rate == RATE_6300) {
+ skip_put_bits(&pb, 1); /* reserved bit */
+
+ /* Write 13 bit combined position index */
+ temp = (p->subframe[0].pulse_pos >> 16) * 810 +
+ (p->subframe[1].pulse_pos >> 14) * 90 +
+ (p->subframe[2].pulse_pos >> 16) * 9 +
+ (p->subframe[3].pulse_pos >> 14);
+ put_bits(&pb, 13, temp);
+
+ put_bits(&pb, 16, p->subframe[0].pulse_pos & 0xffff);
+ put_bits(&pb, 14, p->subframe[1].pulse_pos & 0x3fff);
+ put_bits(&pb, 16, p->subframe[2].pulse_pos & 0xffff);
+ put_bits(&pb, 14, p->subframe[3].pulse_pos & 0x3fff);
+
+ put_bits(&pb, 6, p->subframe[0].pulse_sign);
+ put_bits(&pb, 5, p->subframe[1].pulse_sign);
+ put_bits(&pb, 6, p->subframe[2].pulse_sign);
+ put_bits(&pb, 5, p->subframe[3].pulse_sign);
+ }
+
+ flush_put_bits(&pb);
+ return frame_size[info_bits];
+}
+
+static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+ const AVFrame *frame, int *got_packet_ptr)
+{
+ G723_1_Context *p = avctx->priv_data;
+ int16_t unq_lpc[LPC_ORDER * SUBFRAMES];
+ int16_t qnt_lpc[LPC_ORDER * SUBFRAMES];
+ int16_t cur_lsp[LPC_ORDER];
+ int16_t weighted_lpc[LPC_ORDER * SUBFRAMES << 1];
+ int16_t vector[FRAME_LEN + PITCH_MAX];
+ int offset, ret;
+ int16_t *in = (const int16_t *)frame->data[0];
+
+ HFParam hf[4];
+ int i, j;
+
+ highpass_filter(in, &p->hpf_fir_mem, &p->hpf_iir_mem);
+
+ memcpy(vector, p->prev_data, HALF_FRAME_LEN * sizeof(int16_t));
+ memcpy(vector + HALF_FRAME_LEN, in, FRAME_LEN * sizeof(int16_t));
+
+ comp_lpc_coeff(vector, unq_lpc);
+ lpc2lsp(&unq_lpc[LPC_ORDER * 3], p->prev_lsp, cur_lsp);
+ lsp_quantize(p->lsp_index, cur_lsp, p->prev_lsp);
+
+ /* Update memory */
+ memcpy(vector + LPC_ORDER, p->prev_data + SUBFRAME_LEN,
+ sizeof(int16_t) * SUBFRAME_LEN);
+ memcpy(vector + LPC_ORDER + SUBFRAME_LEN, in,
+ sizeof(int16_t) * (HALF_FRAME_LEN + SUBFRAME_LEN));
+ memcpy(p->prev_data, in + HALF_FRAME_LEN,
+ sizeof(int16_t) * HALF_FRAME_LEN);
+ memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
+
+ perceptual_filter(p, weighted_lpc, unq_lpc, vector);
+
+ memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
+ memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
+ memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
+
+ scale_vector(vector, vector, FRAME_LEN + PITCH_MAX);
+
+ p->pitch_lag[0] = estimate_pitch(vector, PITCH_MAX);
+ p->pitch_lag[1] = estimate_pitch(vector, PITCH_MAX + HALF_FRAME_LEN);
+
+ for (i = PITCH_MAX, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
+ comp_harmonic_coeff(vector + i, p->pitch_lag[j >> 1], hf + j);
+
+ memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
+ memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
+ memcpy(p->prev_weight_sig, vector + FRAME_LEN, sizeof(int16_t) * PITCH_MAX);
+
+ for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
+ harmonic_filter(hf + j, vector + PITCH_MAX + i, in + i);
+
+ inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);
+ lsp_interpolate(qnt_lpc, cur_lsp, p->prev_lsp);
+
+ memcpy(p->prev_lsp, cur_lsp, sizeof(int16_t) * LPC_ORDER);
+
+ offset = 0;
+ for (i = 0; i < SUBFRAMES; i++) {
+ int16_t impulse_resp[SUBFRAME_LEN];
+ int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
+ int16_t flt_in[SUBFRAME_LEN];
+ int16_t zero[LPC_ORDER], fir[LPC_ORDER], iir[LPC_ORDER];
+
+ /**
+ * Compute the combined impulse response of the synthesis filter,
+ * formant perceptual weighting filter and harmonic noise shaping filter
+ */
+ memset(zero, 0, sizeof(int16_t) * LPC_ORDER);
+ memset(vector, 0, sizeof(int16_t) * PITCH_MAX);
+ memset(flt_in, 0, sizeof(int16_t) * SUBFRAME_LEN);
+
+ flt_in[0] = 1 << 13; /* Unit impulse */
+ synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
+ zero, zero, flt_in, vector + PITCH_MAX, 1);
+ harmonic_filter(hf + i, vector + PITCH_MAX, impulse_resp);
+
+ /* Compute the combined zero input response */
+ flt_in[0] = 0;
+ memcpy(fir, p->perf_fir_mem, sizeof(int16_t) * LPC_ORDER);
+ memcpy(iir, p->perf_iir_mem, sizeof(int16_t) * LPC_ORDER);
+
+ synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
+ fir, iir, flt_in, vector + PITCH_MAX, 0);
+ memcpy(vector, p->harmonic_mem, sizeof(int16_t) * PITCH_MAX);
+ harmonic_noise_sub(hf + i, vector + PITCH_MAX, in);
+
+ acb_search(p, residual, impulse_resp, in, i);
+ gen_acb_excitation(residual, p->prev_excitation,p->pitch_lag[i >> 1],
+ &p->subframe[i], p->cur_rate);
+ sub_acb_contrib(residual, impulse_resp, in);
+
+ fcb_search(p, impulse_resp, in, i);
+
+ /* Reconstruct the excitation */
+ gen_acb_excitation(impulse_resp, p->prev_excitation, p->pitch_lag[i >> 1],
+ &p->subframe[i], RATE_6300);
+
+ memmove(p->prev_excitation, p->prev_excitation + SUBFRAME_LEN,
+ sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
+ for (j = 0; j < SUBFRAME_LEN; j++)
+ in[j] = av_clip_int16((in[j] << 1) + impulse_resp[j]);
+ memcpy(p->prev_excitation + PITCH_MAX - SUBFRAME_LEN, in,
+ sizeof(int16_t) * SUBFRAME_LEN);
+
+ /* Update filter memories */
+ synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
+ p->perf_fir_mem, p->perf_iir_mem,
+ in, vector + PITCH_MAX, 0);
+ memmove(p->harmonic_mem, p->harmonic_mem + SUBFRAME_LEN,
+ sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
+ memcpy(p->harmonic_mem + PITCH_MAX - SUBFRAME_LEN, vector + PITCH_MAX,
+ sizeof(int16_t) * SUBFRAME_LEN);
+
+ in += SUBFRAME_LEN;
+ offset += LPC_ORDER;
+ }
+
+ if ((ret = ff_alloc_packet2(avctx, avpkt, 24)))
+ return ret;
+
+ *got_packet_ptr = 1;
+ avpkt->size = pack_bitstream(p, avpkt->data, avpkt->size);
+ return 0;
+}
+
+AVCodec ff_g723_1_encoder = {
+ .name = "g723_1",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = AV_CODEC_ID_G723_1,
+ .priv_data_size = sizeof(G723_1_Context),
+ .init = g723_1_encode_init,
+ .encode2 = g723_1_encode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("G.723.1"),
+ .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,
+ AV_SAMPLE_FMT_NONE},
+};
+#endif
projects / ffmpeg / blobdiff