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https://github.com/FFmpeg/FFmpeg.git
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f3f6c6b928
This commit reworks the TNS implementation to a hybrid between what the specifications say, what the decoder does and what's the best thing to do. The filter application function was copied from the decoder and modified such that it applies the inverse AR filter to the coefficients. The LPC coefficients themselves are fed into the same quantization expression that the specifications say should be used however further processing is not done, instead they're converted to the form that the decoder expects them to be in and are sent off to the compute_lpc_coeffs function exactly the way the decoder does. This function does all conversions and will return the exact coefficients that the decoder will generate, which are then applied to the coefficients. Having the exact same coefficients on both the encoder and decoder is a must since otherwise the entire sfb's over which the filter is applied will be attenuated. Despite this major rework, TNS might not work fine on some audio types at very low bitrates (e.g. sub 90kbps) as it can attenuate some coefficients too much. Users are advised to experiment with TNS at higher bitrates if they wish to use this tool or simply wait for the implementation to be improved. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
183 lines
6.5 KiB
C
183 lines
6.5 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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/**
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* Encode TNS data.
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* Coefficient compression saves a single bit per coefficient.
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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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uint8_t u_coef;
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const uint8_t coef_res = TNS_Q_BITS == 4;
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int i, w, filt, coef_len, coef_compress = 0;
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const int is8 = sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
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TemporalNoiseShaping *tns = &sce->tns;
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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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put_bits(&s->pb, 1, coef_res);
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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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put_bits(&s->pb, 1, !!tns->direction[i][filt]);
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put_bits(&s->pb, 1, !!coef_compress);
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coef_len = coef_res + 3 - coef_compress;
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for (w = 0; w < tns->order[i][filt]; w++) {
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u_coef = (tns->coef_idx[i][filt][w])&(~(~0<<coef_len));
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put_bits(&s->pb, coef_len, u_coef);
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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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static int quantize_coefs(double *coef, int *idx, float *lpc, int order)
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{
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int i;
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uint8_t u_coef;
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const float *quant_arr = tns_tmp2_map[TNS_Q_BITS == 4];
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const double iqfac_p = ((1 << (TNS_Q_BITS-1)) - 0.5)/(M_PI/2.0);
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const double iqfac_m = ((1 << (TNS_Q_BITS-1)) + 0.5)/(M_PI/2.0);
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for (i = 0; i < order; i++) {
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idx[i] = ceilf(asin(coef[i])*((coef[i] >= 0) ? iqfac_p : iqfac_m));
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u_coef = (idx[i])&(~(~0<<TNS_Q_BITS));
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lpc[i] = quant_arr[u_coef];
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}
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return order;
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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];
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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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// 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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}
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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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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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int order = is8 ? 7 : s->profile == FF_PROFILE_AAC_LOW ? 12 : TNS_MAX_ORDER;
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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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for (w = 0; w < sce->ics.num_windows; w++) {
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float en_low = 0.0f, en_high = 0.0f, threshold = 0.0f, spread = 0.0f;
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double gain = 0.0f, coefs[MAX_LPC_ORDER] = {0};
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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_high += band->energy;
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else
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en_low += band->energy;
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threshold += band->threshold;
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spread += band->spread;
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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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gain *= s->lambda/110.0f;
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if (gain > TNS_GAIN_THRESHOLD_LOW && gain*0 < TNS_GAIN_THRESHOLD_HIGH &&
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(en_low+en_high) > TNS_GAIN_THRESHOLD_LOW*threshold &&
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spread > TNS_SPREAD_THRESHOLD) {
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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] = en_low < en_high && TNS_DIRECTION_VARY;
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tns->order[w][g] = quantize_coefs(coefs, tns->coef_idx[w][g],
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tns->coef[w][g], order);
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}
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count++;
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}
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}
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sce->tns.present = !!count;
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}
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