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FFmpeg/libavcodec/aacenc_tns.c

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/*
* AAC encoder TNS
* Copyright (C) 2015 Rostislav Pehlivanov
*
* 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
* AAC encoder temporal noise shaping
* @author Rostislav Pehlivanov ( atomnuker gmail com )
*/
#include "aacenc.h"
#include "aacenc_tns.h"
#include "aactab.h"
#include "aacenc_utils.h"
#include "aacenc_quantization.h"
/* Could be set to 3 to save an additional bit at the cost of little quality */
#define TNS_Q_BITS 4
/* Coefficient resolution in short windows */
#define TNS_Q_BITS_IS8 4
/* Define this to save a bit, be warned decoders can't deal with it
* so it is not lossless despite what the specifications say */
// #define TNS_ENABLE_COEF_COMPRESSION
/* TNS will only be used if the LPC gain is within these margins */
#define TNS_GAIN_THRESHOLD_LOW 1.477f
#define TNS_GAIN_THRESHOLD_HIGH 7.0f
#define TNS_GAIN_THRESHOLD_LOW_IS8 0.16f*TNS_GAIN_THRESHOLD_LOW
#define TNS_GAIN_THRESHOLD_HIGH_IS8 0.26f*TNS_GAIN_THRESHOLD_HIGH
static inline int compress_coeffs(int *coef, int order, int c_bits)
{
int i;
const int low_idx = c_bits ? 4 : 2;
const int shift_val = c_bits ? 8 : 4;
const int high_idx = c_bits ? 11 : 5;
#ifndef TNS_ENABLE_COEF_COMPRESSION
return 0;
#endif /* TNS_ENABLE_COEF_COMPRESSION */
for (i = 0; i < order; i++)
if (coef[i] >= low_idx && coef[i] <= high_idx)
return 0;
for (i = 0; i < order; i++)
coef[i] -= (coef[i] > high_idx) ? shift_val : 0;
return 1;
}
/**
* Encode TNS data.
* Coefficient compression is simply not lossless as it should be
* on any decoder tested and as such is not active.
*/
void ff_aac_encode_tns_info(AACEncContext *s, SingleChannelElement *sce)
{
int i, w, filt, coef_compress = 0, coef_len;
TemporalNoiseShaping *tns = &sce->tns;
const int is8 = sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
const int c_bits = is8 ? TNS_Q_BITS_IS8 == 4 : TNS_Q_BITS == 4;
if (!sce->tns.present)
return;
for (i = 0; i < sce->ics.num_windows; i++) {
put_bits(&s->pb, 2 - is8, sce->tns.n_filt[i]);
if (!tns->n_filt[i])
continue;
put_bits(&s->pb, 1, c_bits);
for (filt = 0; filt < tns->n_filt[i]; filt++) {
put_bits(&s->pb, 6 - 2 * is8, tns->length[i][filt]);
put_bits(&s->pb, 5 - 2 * is8, tns->order[i][filt]);
if (!tns->order[i][filt])
continue;
put_bits(&s->pb, 1, tns->direction[i][filt]);
coef_compress = compress_coeffs(tns->coef_idx[i][filt],
tns->order[i][filt], c_bits);
put_bits(&s->pb, 1, coef_compress);
coef_len = c_bits + 3 - coef_compress;
for (w = 0; w < tns->order[i][filt]; w++)
put_bits(&s->pb, coef_len, tns->coef_idx[i][filt][w]);
}
}
}
/* Apply TNS filter */
void ff_aac_apply_tns(AACEncContext *s, SingleChannelElement *sce)
{
TemporalNoiseShaping *tns = &sce->tns;
IndividualChannelStream *ics = &sce->ics;
int w, filt, m, i, top, order, bottom, start, end, size, inc;
const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
float lpc[TNS_MAX_ORDER], tmp[TNS_MAX_ORDER+1];
for (w = 0; w < ics->num_windows; w++) {
bottom = ics->num_swb;
for (filt = 0; filt < tns->n_filt[w]; filt++) {
top = bottom;
bottom = FFMAX(0, top - tns->length[w][filt]);
order = tns->order[w][filt];
if (order == 0)
continue;
// tns_decode_coef
compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
start = ics->swb_offset[FFMIN(bottom, mmm)];
end = ics->swb_offset[FFMIN( top, mmm)];
if ((size = end - start) <= 0)
continue;
if (tns->direction[w][filt]) {
inc = -1;
start = end - 1;
} else {
inc = 1;
}
start += w * 128;
if (!s->options.ltp) { // ar filter
for (m = 0; m < size; m++, start += inc) {
for (i = 1; i <= FFMIN(m, order); i++) {
sce->coeffs[start] += lpc[i-1]*sce->pcoeffs[start - i*inc];
}
}
} else { // ma filter
for (m = 0; m < size; m++, start += inc) {
tmp[0] = sce->pcoeffs[start];
for (i = 1; i <= FFMIN(m, order); i++)
sce->coeffs[start] += lpc[i-1]*tmp[i];
for (i = order; i > 0; i--)
tmp[i] = tmp[i - 1];
}
}
}
}
}
/*
* c_bits - 1 if 4 bit coefficients, 0 if 3 bit coefficients
*/
static inline void quantize_coefs(double *coef, int *idx, float *lpc, int order,
int c_bits)
{
int i;
const float *quant_arr = tns_tmp2_map[c_bits];
for (i = 0; i < order; i++) {
idx[i] = quant_array_idx((float)coef[i], quant_arr, c_bits ? 16 : 8);
lpc[i] = quant_arr[idx[i]];
}
}
/*
* 3 bits per coefficient with 8 short windows
*/
void ff_aac_search_for_tns(AACEncContext *s, SingleChannelElement *sce)
{
TemporalNoiseShaping *tns = &sce->tns;
double gain, coefs[MAX_LPC_ORDER];
int w, w2, g, count = 0;
const int mmm = FFMIN(sce->ics.tns_max_bands, sce->ics.max_sfb);
const int is8 = sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
const int c_bits = is8 ? TNS_Q_BITS_IS8 == 4 : TNS_Q_BITS == 4;
const int slant = sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE ? 1 :
sce->ics.window_sequence[0] == LONG_START_SEQUENCE ? 0 : 2;
int sfb_start = av_clip(tns_min_sfb[is8][s->samplerate_index], 0, mmm);
int sfb_end = av_clip(sce->ics.num_swb, 0, mmm);
int order = is8 ? 5 : s->profile == FF_PROFILE_AAC_LOW ? 12 : TNS_MAX_ORDER;
for (w = 0; w < sce->ics.num_windows; w++) {
float en[2] = {0.0f, 0.0f};
int coef_start = w*sce->ics.num_swb + sce->ics.swb_offset[sfb_start];
int coef_len = sce->ics.swb_offset[sfb_end] - sce->ics.swb_offset[sfb_start];
for (g = 0; g < sce->ics.num_swb; g++) {
if (w*16+g < sfb_start || w*16+g > sfb_end)
continue;
for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
if ((w+w2)*16+g > sfb_start + ((sfb_end - sfb_start)/2))
en[1] += band->energy;
else
en[0] += band->energy;
}
}
if (coef_len <= 0 || (sfb_end - sfb_start) <= 0)
continue;
/* LPC */
gain = ff_lpc_calc_ref_coefs_f(&s->lpc, &sce->coeffs[coef_start],
coef_len, order, coefs);
if (!order || gain < TNS_GAIN_THRESHOLD_LOW || gain > TNS_GAIN_THRESHOLD_HIGH)
continue;
if (is8 && (gain < TNS_GAIN_THRESHOLD_LOW_IS8 || gain > TNS_GAIN_THRESHOLD_HIGH_IS8))
continue;
if (is8 || order < 2) {
tns->n_filt[w] = 1;
for (g = 0; g < tns->n_filt[w]; g++) {
tns->length[w][g] = sfb_end - sfb_start;
tns->direction[w][g] = slant != 2 ? slant : en[0] < en[1];
tns->order[w][g] = order;
quantize_coefs(coefs, tns->coef_idx[w][g], tns->coef[w][g],
order, c_bits);
}
} else { /* 2 filters due to energy disbalance */
tns->n_filt[w] = 2;
for (g = 0; g < tns->n_filt[w]; g++) {
tns->direction[w][g] = slant != 2 ? slant : en[g] < en[!g];
tns->order[w][g] = !g ? order/2 : order - tns->order[w][g-1];
tns->length[w][g] = !g ? (sfb_end - sfb_start)/2 : \
(sfb_end - sfb_start) - tns->length[w][g-1];
quantize_coefs(&coefs[!g ? 0 : order - tns->order[w][g-1]],
tns->coef_idx[w][g], tns->coef[w][g],
tns->order[w][g], c_bits);
}
}
count += tns->n_filt[w];
}
sce->tns.present = !!count;
}