mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-21 10:55:51 +02:00
fa4d900c27
Changes: - strongly prefer dual filters to a single filter - less strict about using 2 filters w.r.t. energy - scrap the usage of threshold and spread, useless - use odd-shaped windows to set the filter direction - use 4 bits instead of 3 bits for short windows - simplify and reduce the main loop to a single level - add stricter regulations for short windows All of this now makes the TNS implementation operate as good as it can and it definitely shows. The frequency thresholds are now even better defined by looking at the spectrals and the overall sound has been improved at the price of just a few bits that are well worth it.
240 lines
8.9 KiB
C
240 lines
8.9 KiB
C
/*
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* AAC encoder TNS
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* Copyright (C) 2015 Rostislav Pehlivanov
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* AAC encoder temporal noise shaping
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* @author Rostislav Pehlivanov ( atomnuker gmail com )
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*/
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#include "aacenc.h"
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#include "aacenc_tns.h"
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#include "aactab.h"
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#include "aacenc_utils.h"
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#include "aacenc_quantization.h"
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/* Could be set to 3 to save an additional bit at the cost of little quality */
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#define TNS_Q_BITS 4
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/* Coefficient resolution in short windows */
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#define TNS_Q_BITS_IS8 4
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/* Define this to save a bit, be warned decoders can't deal with it
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* so it is not lossless despite what the specifications say */
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// #define TNS_ENABLE_COEF_COMPRESSION
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/* TNS will only be used if the LPC gain is within these margins */
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#define TNS_GAIN_THRESHOLD_LOW 1.477f
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#define TNS_GAIN_THRESHOLD_HIGH 7.0f
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#define TNS_GAIN_THRESHOLD_LOW_IS8 0.16f*TNS_GAIN_THRESHOLD_LOW
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#define TNS_GAIN_THRESHOLD_HIGH_IS8 0.26f*TNS_GAIN_THRESHOLD_HIGH
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static inline int compress_coeffs(int *coef, int order, int c_bits)
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{
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int i;
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const int low_idx = c_bits ? 4 : 2;
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const int shift_val = c_bits ? 8 : 4;
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const int high_idx = c_bits ? 11 : 5;
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#ifndef TNS_ENABLE_COEF_COMPRESSION
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return 0;
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#endif /* TNS_ENABLE_COEF_COMPRESSION */
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for (i = 0; i < order; i++)
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if (coef[i] >= low_idx && coef[i] <= high_idx)
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return 0;
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for (i = 0; i < order; i++)
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coef[i] -= (coef[i] > high_idx) ? shift_val : 0;
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return 1;
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}
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/**
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* Encode TNS data.
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* Coefficient compression is simply not lossless as it should be
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* on any decoder tested and as such is not active.
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*/
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void ff_aac_encode_tns_info(AACEncContext *s, SingleChannelElement *sce)
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{
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int i, w, filt, coef_compress = 0, coef_len;
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TemporalNoiseShaping *tns = &sce->tns;
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const int is8 = sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
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const int c_bits = is8 ? TNS_Q_BITS_IS8 == 4 : TNS_Q_BITS == 4;
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if (!sce->tns.present)
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return;
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for (i = 0; i < sce->ics.num_windows; i++) {
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put_bits(&s->pb, 2 - is8, sce->tns.n_filt[i]);
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if (!tns->n_filt[i])
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continue;
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put_bits(&s->pb, 1, c_bits);
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for (filt = 0; filt < tns->n_filt[i]; filt++) {
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put_bits(&s->pb, 6 - 2 * is8, tns->length[i][filt]);
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put_bits(&s->pb, 5 - 2 * is8, tns->order[i][filt]);
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if (!tns->order[i][filt])
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continue;
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put_bits(&s->pb, 1, tns->direction[i][filt]);
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coef_compress = compress_coeffs(tns->coef_idx[i][filt],
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tns->order[i][filt], c_bits);
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put_bits(&s->pb, 1, coef_compress);
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coef_len = c_bits + 3 - coef_compress;
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for (w = 0; w < tns->order[i][filt]; w++)
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put_bits(&s->pb, coef_len, tns->coef_idx[i][filt][w]);
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}
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}
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}
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/* Apply TNS filter */
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void ff_aac_apply_tns(AACEncContext *s, SingleChannelElement *sce)
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{
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TemporalNoiseShaping *tns = &sce->tns;
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IndividualChannelStream *ics = &sce->ics;
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int w, filt, m, i, top, order, bottom, start, end, size, inc;
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const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
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float lpc[TNS_MAX_ORDER], tmp[TNS_MAX_ORDER+1];
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for (w = 0; w < ics->num_windows; w++) {
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bottom = ics->num_swb;
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for (filt = 0; filt < tns->n_filt[w]; filt++) {
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top = bottom;
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bottom = FFMAX(0, top - tns->length[w][filt]);
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order = tns->order[w][filt];
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if (order == 0)
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continue;
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// tns_decode_coef
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compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
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start = ics->swb_offset[FFMIN(bottom, mmm)];
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end = ics->swb_offset[FFMIN( top, mmm)];
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if ((size = end - start) <= 0)
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continue;
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if (tns->direction[w][filt]) {
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inc = -1;
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start = end - 1;
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} else {
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inc = 1;
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}
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start += w * 128;
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if (!s->options.ltp) { // ar filter
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for (m = 0; m < size; m++, start += inc) {
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for (i = 1; i <= FFMIN(m, order); i++) {
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sce->coeffs[start] += lpc[i-1]*sce->pcoeffs[start - i*inc];
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}
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}
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} else { // ma filter
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for (m = 0; m < size; m++, start += inc) {
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tmp[0] = sce->pcoeffs[start];
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for (i = 1; i <= FFMIN(m, order); i++)
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sce->coeffs[start] += lpc[i-1]*tmp[i];
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for (i = order; i > 0; i--)
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tmp[i] = tmp[i - 1];
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}
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}
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}
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}
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}
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/*
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* c_bits - 1 if 4 bit coefficients, 0 if 3 bit coefficients
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*/
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static inline void quantize_coefs(double *coef, int *idx, float *lpc, int order,
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int c_bits)
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{
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int i;
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const float *quant_arr = tns_tmp2_map[c_bits];
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for (i = 0; i < order; i++) {
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idx[i] = quant_array_idx((float)coef[i], quant_arr, c_bits ? 16 : 8);
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lpc[i] = quant_arr[idx[i]];
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}
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}
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/*
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* 3 bits per coefficient with 8 short windows
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*/
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void ff_aac_search_for_tns(AACEncContext *s, SingleChannelElement *sce)
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{
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TemporalNoiseShaping *tns = &sce->tns;
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double gain, coefs[MAX_LPC_ORDER];
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int w, w2, g, count = 0;
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const int mmm = FFMIN(sce->ics.tns_max_bands, sce->ics.max_sfb);
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const int is8 = sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
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const int c_bits = is8 ? TNS_Q_BITS_IS8 == 4 : TNS_Q_BITS == 4;
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const int slant = sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE ? 1 :
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sce->ics.window_sequence[0] == LONG_START_SEQUENCE ? 0 : 2;
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int sfb_start = av_clip(tns_min_sfb[is8][s->samplerate_index], 0, mmm);
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int sfb_end = av_clip(sce->ics.num_swb, 0, mmm);
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int order = is8 ? 5 : s->profile == FF_PROFILE_AAC_LOW ? 12 : TNS_MAX_ORDER;
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for (w = 0; w < sce->ics.num_windows; w++) {
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float en[2] = {0.0f, 0.0f};
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int coef_start = w*sce->ics.num_swb + sce->ics.swb_offset[sfb_start];
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int coef_len = sce->ics.swb_offset[sfb_end] - sce->ics.swb_offset[sfb_start];
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for (g = 0; g < sce->ics.num_swb; g++) {
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if (w*16+g < sfb_start || w*16+g > sfb_end)
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continue;
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for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
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FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
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if ((w+w2)*16+g > sfb_start + ((sfb_end - sfb_start)/2))
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en[1] += band->energy;
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else
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en[0] += band->energy;
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}
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}
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if (coef_len <= 0 || (sfb_end - sfb_start) <= 0)
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continue;
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/* LPC */
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gain = ff_lpc_calc_ref_coefs_f(&s->lpc, &sce->coeffs[coef_start],
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coef_len, order, coefs);
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if (!order || gain < TNS_GAIN_THRESHOLD_LOW || gain > TNS_GAIN_THRESHOLD_HIGH)
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continue;
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if (is8 && (gain < TNS_GAIN_THRESHOLD_LOW_IS8 || gain > TNS_GAIN_THRESHOLD_HIGH_IS8))
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continue;
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if (is8 || order < 2) {
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tns->n_filt[w] = 1;
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for (g = 0; g < tns->n_filt[w]; g++) {
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tns->length[w][g] = sfb_end - sfb_start;
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tns->direction[w][g] = slant != 2 ? slant : en[0] < en[1];
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tns->order[w][g] = order;
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quantize_coefs(coefs, tns->coef_idx[w][g], tns->coef[w][g],
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order, c_bits);
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}
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} else { /* 2 filters due to energy disbalance */
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tns->n_filt[w] = 2;
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for (g = 0; g < tns->n_filt[w]; g++) {
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tns->direction[w][g] = slant != 2 ? slant : en[g] < en[!g];
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tns->order[w][g] = !g ? order/2 : order - tns->order[w][g-1];
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tns->length[w][g] = !g ? (sfb_end - sfb_start)/2 : \
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(sfb_end - sfb_start) - tns->length[w][g-1];
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quantize_coefs(&coefs[!g ? 0 : order - tns->order[w][g-1]],
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tns->coef_idx[w][g], tns->coef[w][g],
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tns->order[w][g], c_bits);
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}
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}
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count += tns->n_filt[w];
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}
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sce->tns.present = !!count;
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}
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