mirror of
https://gitee.com/openharmony/third_party_ffmpeg
synced 2024-11-23 11:19:55 +00:00
44ddee945a
This commit completely alters the algorithm of prediction. The original commit which introduced prediction was completely incorrect to even remotely care about what the actual coefficients contain or whether any options were enabled. Not my actual fault. This commit treats prediction the way the decoder does and expects to do: like lossy encryption. Everything related to prediction now happens at the very end but just before quantization and encoding of coefficients. On the decoder side, prediction happens before anything has had a chance to even access the coefficients. Also the original implementation had problems because it actually touched the band_type of special bands which already had their scalefactor indices marked and it's a wonder the asserion wasn't triggered when transmitting those. Overall, this now drastically increases audio quality and you should think about enabling it if you don't plan on playing anything encoded on really old low power ultra-embedded devices since they might not support decoding of prediction or AAC-Main. Though the specifications were written ages ago and as times change so do the FLOPS. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
343 lines
12 KiB
C
343 lines
12 KiB
C
/*
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* AAC encoder main-type prediction
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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 Intensity Stereo
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* @author Rostislav Pehlivanov ( atomnuker gmail com )
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*/
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#include "aactab.h"
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#include "aacenc_pred.h"
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#include "aacenc_utils.h"
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#include "aacenc_is.h" /* <- Needed for common window distortions */
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#include "aacenc_quantization.h"
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#define RESTORE_PRED(sce, sfb) \
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if (sce->ics.prediction_used[sfb]) {\
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sce->ics.prediction_used[sfb] = 0;\
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sce->band_type[sfb] = sce->band_alt[sfb];\
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}
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static inline float flt16_round(float pf)
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{
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union av_intfloat32 tmp;
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tmp.f = pf;
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tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
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return tmp.f;
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}
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static inline float flt16_even(float pf)
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{
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union av_intfloat32 tmp;
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tmp.f = pf;
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tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
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return tmp.f;
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}
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static inline float flt16_trunc(float pf)
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{
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union av_intfloat32 pun;
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pun.f = pf;
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pun.i &= 0xFFFF0000U;
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return pun.f;
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}
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static inline void predict(PredictorState *ps, float *coef, float *rcoef, int set)
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{
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float k2;
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const float a = 0.953125; // 61.0 / 64
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const float alpha = 0.90625; // 29.0 / 32
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const float k1 = ps->k1;
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const float r0 = ps->r0, r1 = ps->r1;
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const float cor0 = ps->cor0, cor1 = ps->cor1;
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const float var0 = ps->var0, var1 = ps->var1;
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const float e0 = *coef - ps->x_est;
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const float e1 = e0 - k1 * r0;
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if (set)
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*coef = e0;
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ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
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ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
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ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
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ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
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ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
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ps->r0 = flt16_trunc(a * e0);
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/* Prediction for next frame */
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ps->k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0;
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k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0;
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*rcoef = ps->x_est = flt16_round(ps->k1*ps->r0 + k2*ps->r1);
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}
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static inline void reset_predict_state(PredictorState *ps)
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{
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ps->r0 = 0.0f;
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ps->r1 = 0.0f;
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ps->k1 = 0.0f;
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ps->cor0 = 0.0f;
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ps->cor1 = 0.0f;
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ps->var0 = 1.0f;
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ps->var1 = 1.0f;
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ps->x_est = 0.0f;
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}
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static inline void reset_all_predictors(PredictorState *ps)
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{
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int i;
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for (i = 0; i < MAX_PREDICTORS; i++)
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reset_predict_state(&ps[i]);
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}
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static inline void reset_predictor_group(SingleChannelElement *sce, int group_num)
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{
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int i;
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PredictorState *ps = sce->predictor_state;
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for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
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reset_predict_state(&ps[i]);
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}
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void ff_aac_apply_main_pred(AACEncContext *s, SingleChannelElement *sce)
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{
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int sfb, k;
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const int pmax = FFMIN(sce->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
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if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
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for (sfb = 0; sfb < pmax; sfb++) {
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for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
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predict(&sce->predictor_state[k], &sce->coeffs[k], &sce->prcoeffs[k],
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sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
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}
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}
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if (sce->ics.predictor_reset_group) {
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reset_predictor_group(sce, sce->ics.predictor_reset_group);
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}
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} else {
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reset_all_predictors(sce->predictor_state);
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}
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}
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/* If inc = 0 you can check if this returns 0 to see if you can reset freely */
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static inline int update_counters(IndividualChannelStream *ics, int inc)
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{
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int i;
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for (i = 1; i < 31; i++) {
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ics->predictor_reset_count[i] += inc;
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if (ics->predictor_reset_count[i] > PRED_RESET_FRAME_MIN)
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return i; /* Reset this immediately */
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}
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return 0;
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}
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void ff_aac_adjust_common_prediction(AACEncContext *s, ChannelElement *cpe)
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{
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int start, w, w2, g, i, count = 0;
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SingleChannelElement *sce0 = &cpe->ch[0];
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SingleChannelElement *sce1 = &cpe->ch[1];
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const int pmax0 = FFMIN(sce0->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
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const int pmax1 = FFMIN(sce1->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
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const int pmax = FFMIN(pmax0, pmax1);
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if (!cpe->common_window ||
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sce0->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE ||
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sce1->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE)
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return;
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for (w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]) {
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start = 0;
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for (g = 0; g < sce0->ics.num_swb; g++) {
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int sfb = w*16+g;
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int sum = sce0->ics.prediction_used[sfb] + sce1->ics.prediction_used[sfb];
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float ener0 = 0.0f, ener1 = 0.0f, ener01 = 0.0f;
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struct AACISError ph_err1, ph_err2, *erf;
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if (sfb < PRED_SFB_START || sfb > pmax || sum != 2) {
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RESTORE_PRED(sce0, sfb);
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RESTORE_PRED(sce1, sfb);
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start += sce0->ics.swb_sizes[g];
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continue;
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}
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for (w2 = 0; w2 < sce0->ics.group_len[w]; w2++) {
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for (i = 0; i < sce0->ics.swb_sizes[g]; i++) {
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float coef0 = sce0->pcoeffs[start+(w+w2)*128+i];
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float coef1 = sce1->pcoeffs[start+(w+w2)*128+i];
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ener0 += coef0*coef0;
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ener1 += coef1*coef1;
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ener01 += (coef0 + coef1)*(coef0 + coef1);
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}
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}
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ph_err1 = ff_aac_is_encoding_err(s, cpe, start, w, g,
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ener0, ener1, ener01, -1);
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ph_err2 = ff_aac_is_encoding_err(s, cpe, start, w, g,
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ener0, ener1, ener01, +1);
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erf = ph_err1.error < ph_err2.error ? &ph_err1 : &ph_err2;
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if (erf->pass) {
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sce0->ics.prediction_used[sfb] = 1;
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sce1->ics.prediction_used[sfb] = 1;
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count++;
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} else {
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RESTORE_PRED(sce0, sfb);
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RESTORE_PRED(sce1, sfb);
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}
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start += sce0->ics.swb_sizes[g];
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}
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}
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sce1->ics.predictor_present = sce0->ics.predictor_present = !!count;
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}
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static void update_pred_resets(SingleChannelElement *sce)
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{
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int i, max_group_id_c, max_frame = 0;
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float avg_frame = 0.0f;
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IndividualChannelStream *ics = &sce->ics;
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/* Update the counters and immediately update any frame behind schedule */
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if ((ics->predictor_reset_group = update_counters(&sce->ics, 1)))
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return;
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for (i = 1; i < 31; i++) {
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/* Count-based */
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if (ics->predictor_reset_count[i] > max_frame) {
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max_group_id_c = i;
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max_frame = ics->predictor_reset_count[i];
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}
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avg_frame = (ics->predictor_reset_count[i] + avg_frame)/2;
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}
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if (max_frame > PRED_RESET_MIN) {
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ics->predictor_reset_group = max_group_id_c;
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} else {
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ics->predictor_reset_group = 0;
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}
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}
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void ff_aac_search_for_pred(AACEncContext *s, SingleChannelElement *sce)
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{
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int sfb, i, count = 0, cost_coeffs = 0, cost_pred = 0;
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const int pmax = FFMIN(sce->ics.max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
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float *O34 = &s->scoefs[128*0], *P34 = &s->scoefs[128*1];
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float *SENT = &s->scoefs[128*2], *S34 = &s->scoefs[128*3];
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float *QERR = &s->scoefs[128*4];
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if (sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
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sce->ics.predictor_present = 0;
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return;
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}
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if (!sce->ics.predictor_initialized) {
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reset_all_predictors(sce->predictor_state);
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sce->ics.predictor_initialized = 1;
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memcpy(sce->prcoeffs, sce->coeffs, 1024*sizeof(float));
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for (i = 1; i < 31; i++)
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sce->ics.predictor_reset_count[i] = i;
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}
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update_pred_resets(sce);
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memcpy(sce->band_alt, sce->band_type, sizeof(sce->band_type));
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for (sfb = PRED_SFB_START; sfb < pmax; sfb++) {
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int cost1, cost2, cb_p;
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float dist1, dist2, dist_spec_err = 0.0f;
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const int cb_n = sce->band_type[sfb];
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const int start_coef = sce->ics.swb_offset[sfb];
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const int num_coeffs = sce->ics.swb_offset[sfb + 1] - start_coef;
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const FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[sfb];
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if (start_coef + num_coeffs > MAX_PREDICTORS)
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continue;
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/* Normal coefficients */
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abs_pow34_v(O34, &sce->coeffs[start_coef], num_coeffs);
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dist1 = quantize_and_encode_band_cost(s, NULL, &sce->coeffs[start_coef], NULL,
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O34, num_coeffs, sce->sf_idx[sfb],
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cb_n, s->lambda / band->threshold, INFINITY, &cost1, 0);
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cost_coeffs += cost1;
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/* Encoded coefficients - needed for #bits, band type and quant. error */
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for (i = 0; i < num_coeffs; i++)
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SENT[i] = sce->coeffs[start_coef + i] - sce->prcoeffs[start_coef + i];
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abs_pow34_v(S34, SENT, num_coeffs);
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if (cb_n < RESERVED_BT)
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cb_p = find_min_book(find_max_val(1, num_coeffs, S34), sce->sf_idx[sfb]);
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else
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cb_p = cb_n;
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quantize_and_encode_band_cost(s, NULL, SENT, QERR, S34, num_coeffs,
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sce->sf_idx[sfb], cb_p, s->lambda / band->threshold, INFINITY,
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&cost2, 0);
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/* Reconstructed coefficients - needed for distortion measurements */
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for (i = 0; i < num_coeffs; i++)
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sce->prcoeffs[start_coef + i] += QERR[i] != 0.0f ? (sce->prcoeffs[start_coef + i] - QERR[i]) : 0.0f;
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abs_pow34_v(P34, &sce->prcoeffs[start_coef], num_coeffs);
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if (cb_n < RESERVED_BT)
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cb_p = find_min_book(find_max_val(1, num_coeffs, P34), sce->sf_idx[sfb]);
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else
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cb_p = cb_n;
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dist2 = quantize_and_encode_band_cost(s, NULL, &sce->prcoeffs[start_coef], NULL,
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P34, num_coeffs, sce->sf_idx[sfb],
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cb_p, s->lambda / band->threshold, INFINITY, NULL, 0);
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for (i = 0; i < num_coeffs; i++)
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dist_spec_err += (O34[i] - P34[i])*(O34[i] - P34[i]);
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dist_spec_err *= s->lambda / band->threshold;
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dist2 += dist_spec_err;
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if (dist2 <= dist1 && cb_p <= cb_n) {
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cost_pred += cost2;
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sce->ics.prediction_used[sfb] = 1;
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sce->band_alt[sfb] = cb_n;
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sce->band_type[sfb] = cb_p;
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count++;
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} else {
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cost_pred += cost1;
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sce->band_alt[sfb] = cb_p;
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}
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}
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if (count && cost_coeffs < cost_pred) {
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count = 0;
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for (sfb = PRED_SFB_START; sfb < pmax; sfb++)
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RESTORE_PRED(sce, sfb);
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memset(&sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
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}
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sce->ics.predictor_present = !!count;
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}
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/**
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* Encoder predictors data.
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*/
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void ff_aac_encode_main_pred(AACEncContext *s, SingleChannelElement *sce)
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{
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int sfb;
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IndividualChannelStream *ics = &sce->ics;
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const int pmax = FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[s->samplerate_index]);
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if (!ics->predictor_present)
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return;
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put_bits(&s->pb, 1, !!ics->predictor_reset_group);
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if (ics->predictor_reset_group)
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put_bits(&s->pb, 5, ics->predictor_reset_group);
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for (sfb = 0; sfb < pmax; sfb++)
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put_bits(&s->pb, 1, ics->prediction_used[sfb]);
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}
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