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https://github.com/FFmpeg/FFmpeg.git
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aacdec: move prediction to separate files
This commit is contained in:
Lynne
parent
b1718ce0f9
commit
49e7be1e37
@ -569,6 +569,52 @@ static void AAC_RENAME(clip_output)(AACDecContext *ac, ChannelElement *che,
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#endif
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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(PredictorState *ps, int group_num)
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{
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int i;
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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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/**
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* Apply AAC-Main style frequency domain prediction.
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*/
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static void AAC_RENAME(apply_prediction)(AACDecContext *ac, SingleChannelElement *sce)
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{
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int sfb, k;
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if (!sce->ics.predictor_initialized) {
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reset_all_predictors(sce->AAC_RENAME(predictor_state));
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sce->ics.predictor_initialized = 1;
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}
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if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
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for (sfb = 0;
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sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
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sfb++) {
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for (k = sce->ics.swb_offset[sfb];
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k < sce->ics.swb_offset[sfb + 1];
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k++) {
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predict(&sce->AAC_RENAME(predictor_state)[k],
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&sce->AAC_RENAME(coeffs)[k],
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sce->ics.predictor_present &&
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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->AAC_RENAME(predictor_state),
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sce->ics.predictor_reset_group);
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} else
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reset_all_predictors(sce->AAC_RENAME(predictor_state));
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}
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const AACDecDSP AAC_RENAME(aac_dsp) = {
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.init_tables = &AAC_RENAME(init_tables),
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@ -579,6 +625,8 @@ const AACDecDSP AAC_RENAME(aac_dsp) = {
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.apply_ltp = &AAC_RENAME(apply_ltp),
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.update_ltp = &AAC_RENAME(update_ltp),
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.apply_prediction = AAC_RENAME(apply_prediction),
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.imdct_and_windowing = AAC_RENAME(imdct_and_windowing),
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.imdct_and_windowing_960 = AAC_RENAME(imdct_and_windowing_960),
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.imdct_and_windowing_ld = AAC_RENAME(imdct_and_windowing_ld),
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@ -79,5 +79,6 @@ static const int cce_scale_fixed[8] = {
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#include "aacdec_fixed_dequant.h"
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#include "aacdec_fixed_coupling.h"
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#include "aacdec_fixed_prediction.h"
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#include "aacdec_dsp_template.c"
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#include "aacdec_proc_template.c"
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151
libavcodec/aac/aacdec_fixed_prediction.h
Normal file
151
libavcodec/aac/aacdec_fixed_prediction.h
Normal file
@ -0,0 +1,151 @@
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/*
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* AAC decoder
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* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
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* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
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* Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
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*
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* AAC LATM decoder
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* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
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* Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
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*
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* AAC decoder fixed-point implementation
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* Copyright (c) 2013
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* MIPS Technologies, Inc., California.
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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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#ifndef AVCODEC_AAC_AACDEC_FIXED_PREDICTION_H
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#define AVCODEC_AAC_AACDEC_FIXED_PREDICTION_H
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static av_always_inline SoftFloat flt16_round(SoftFloat pf)
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{
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SoftFloat tmp;
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int s;
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tmp.exp = pf.exp;
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s = pf.mant >> 31;
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tmp.mant = (pf.mant ^ s) - s;
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tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U;
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tmp.mant = (tmp.mant ^ s) - s;
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return tmp;
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}
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static av_always_inline SoftFloat flt16_even(SoftFloat pf)
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{
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SoftFloat tmp;
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int s;
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tmp.exp = pf.exp;
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s = pf.mant >> 31;
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tmp.mant = (pf.mant ^ s) - s;
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tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U;
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tmp.mant = (tmp.mant ^ s) - s;
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return tmp;
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}
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static av_always_inline SoftFloat flt16_trunc(SoftFloat pf)
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{
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SoftFloat pun;
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int s;
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pun.exp = pf.exp;
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s = pf.mant >> 31;
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pun.mant = (pf.mant ^ s) - s;
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pun.mant = pun.mant & 0xFFC00000U;
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pun.mant = (pun.mant ^ s) - s;
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return pun;
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}
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static av_always_inline void predict(PredictorState *ps, int *coef,
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int output_enable)
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{
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const SoftFloat a = { 1023410176, 0 }; // 61.0 / 64
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const SoftFloat alpha = { 973078528, 0 }; // 29.0 / 32
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SoftFloat e0, e1;
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SoftFloat pv;
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SoftFloat k1, k2;
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SoftFloat r0 = ps->r0, r1 = ps->r1;
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SoftFloat cor0 = ps->cor0, cor1 = ps->cor1;
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SoftFloat var0 = ps->var0, var1 = ps->var1;
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SoftFloat tmp;
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if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) {
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k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0)));
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}
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else {
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k1.mant = 0;
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k1.exp = 0;
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}
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if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) {
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k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1)));
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}
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else {
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k2.mant = 0;
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k2.exp = 0;
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}
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tmp = av_mul_sf(k1, r0);
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pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1)));
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if (output_enable) {
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int shift = 28 - pv.exp;
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if (shift < 31) {
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if (shift > 0) {
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*coef += (unsigned)((pv.mant + (1 << (shift - 1))) >> shift);
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} else
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*coef += (unsigned)pv.mant << -shift;
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}
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}
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e0 = av_int2sf(*coef, 2);
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e1 = av_sub_sf(e0, tmp);
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ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1)));
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tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1));
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tmp.exp--;
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ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp));
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ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0)));
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tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0));
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tmp.exp--;
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ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp));
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ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0))));
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ps->r0 = flt16_trunc(av_mul_sf(a, e0));
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}
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static av_always_inline void reset_predict_state(PredictorState *ps)
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{
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ps->r0.mant = 0;
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ps->r0.exp = 0;
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ps->r1.mant = 0;
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ps->r1.exp = 0;
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ps->cor0.mant = 0;
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ps->cor0.exp = 0;
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ps->cor1.mant = 0;
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ps->cor1.exp = 0;
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ps->var0.mant = 0x20000000;
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ps->var0.exp = 1;
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ps->var1.mant = 0x20000000;
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ps->var1.exp = 1;
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}
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#endif /* AVCODEC_AAC_AACDEC_FIXED_PREDICTION_H */
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@ -147,5 +147,6 @@ static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
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#endif
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#include "aacdec_float_coupling.h"
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#include "aacdec_float_prediction.h"
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#include "aacdec_dsp_template.c"
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#include "aacdec_proc_template.c"
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100
libavcodec/aac/aacdec_float_prediction.h
Normal file
100
libavcodec/aac/aacdec_float_prediction.h
Normal file
@ -0,0 +1,100 @@
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/*
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* AAC decoder
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* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
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* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
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* Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
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*
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* AAC LATM decoder
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* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
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* Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
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*
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* AAC decoder fixed-point implementation
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* Copyright (c) 2013
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* MIPS Technologies, Inc., California.
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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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#ifndef AVCODEC_AAC_AACDEC_FLOAT_PREDICTION_H
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#define AVCODEC_AAC_AACDEC_FLOAT_PREDICTION_H
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static av_always_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 av_always_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 av_always_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 av_always_inline void predict(PredictorState *ps, float *coef,
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int output_enable)
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{
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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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float e0, e1;
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float pv;
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float k1, k2;
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float r0 = ps->r0, r1 = ps->r1;
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float cor0 = ps->cor0, cor1 = ps->cor1;
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float var0 = ps->var0, var1 = ps->var1;
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k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
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k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
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pv = flt16_round(k1 * r0 + k2 * r1);
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if (output_enable)
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*coef += pv;
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e0 = *coef;
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e1 = e0 - k1 * r0;
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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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}
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static av_always_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->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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}
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#endif /* AVCODEC_AAC_AACDEC_FLOAT_PREDICTION_H */
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@ -63,71 +63,6 @@
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# include "mips/aacdec_mips.h"
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#endif
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static av_always_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->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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}
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static av_always_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 av_always_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 av_always_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 av_always_inline void predict(PredictorState *ps, float *coef,
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int output_enable)
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{
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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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float e0, e1;
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float pv;
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float k1, k2;
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float r0 = ps->r0, r1 = ps->r1;
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float cor0 = ps->cor0, cor1 = ps->cor1;
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float var0 = ps->var0, var1 = ps->var1;
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k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
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k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
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pv = flt16_round(k1 * r0 + k2 * r1);
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if (output_enable)
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*coef += pv;
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e0 = *coef;
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e1 = e0 - k1 * r0;
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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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}
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#include "aacdec_template.c"
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#define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word
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@ -230,6 +230,8 @@ typedef struct AACDecDSP {
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void (*apply_ltp)(AACDecContext *ac, SingleChannelElement *sce);
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void (*update_ltp)(AACDecContext *ac, SingleChannelElement *sce);
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void (*apply_prediction)(AACDecContext *ac, SingleChannelElement *sce);
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void (*apply_dependent_coupling)(AACDecContext *ac,
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SingleChannelElement *target,
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ChannelElement *cce, int index);
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@ -127,124 +127,8 @@ static const int * const tns_tmp2_map_fixed[4] = {
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};
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// @}
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static av_always_inline void reset_predict_state(PredictorState *ps)
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{
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ps->r0.mant = 0;
|
||||
ps->r0.exp = 0;
|
||||
ps->r1.mant = 0;
|
||||
ps->r1.exp = 0;
|
||||
ps->cor0.mant = 0;
|
||||
ps->cor0.exp = 0;
|
||||
ps->cor1.mant = 0;
|
||||
ps->cor1.exp = 0;
|
||||
ps->var0.mant = 0x20000000;
|
||||
ps->var0.exp = 1;
|
||||
ps->var1.mant = 0x20000000;
|
||||
ps->var1.exp = 1;
|
||||
}
|
||||
|
||||
static const int exp2tab[4] = { Q31(1.0000000000/2), Q31(1.1892071150/2), Q31(1.4142135624/2), Q31(1.6817928305/2) }; // 2^0, 2^0.25, 2^0.5, 2^0.75
|
||||
|
||||
static av_always_inline SoftFloat flt16_round(SoftFloat pf)
|
||||
{
|
||||
SoftFloat tmp;
|
||||
int s;
|
||||
|
||||
tmp.exp = pf.exp;
|
||||
s = pf.mant >> 31;
|
||||
tmp.mant = (pf.mant ^ s) - s;
|
||||
tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U;
|
||||
tmp.mant = (tmp.mant ^ s) - s;
|
||||
|
||||
return tmp;
|
||||
}
|
||||
|
||||
static av_always_inline SoftFloat flt16_even(SoftFloat pf)
|
||||
{
|
||||
SoftFloat tmp;
|
||||
int s;
|
||||
|
||||
tmp.exp = pf.exp;
|
||||
s = pf.mant >> 31;
|
||||
tmp.mant = (pf.mant ^ s) - s;
|
||||
tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U;
|
||||
tmp.mant = (tmp.mant ^ s) - s;
|
||||
|
||||
return tmp;
|
||||
}
|
||||
|
||||
static av_always_inline SoftFloat flt16_trunc(SoftFloat pf)
|
||||
{
|
||||
SoftFloat pun;
|
||||
int s;
|
||||
|
||||
pun.exp = pf.exp;
|
||||
s = pf.mant >> 31;
|
||||
pun.mant = (pf.mant ^ s) - s;
|
||||
pun.mant = pun.mant & 0xFFC00000U;
|
||||
pun.mant = (pun.mant ^ s) - s;
|
||||
|
||||
return pun;
|
||||
}
|
||||
|
||||
static av_always_inline void predict(PredictorState *ps, int *coef,
|
||||
int output_enable)
|
||||
{
|
||||
const SoftFloat a = { 1023410176, 0 }; // 61.0 / 64
|
||||
const SoftFloat alpha = { 973078528, 0 }; // 29.0 / 32
|
||||
SoftFloat e0, e1;
|
||||
SoftFloat pv;
|
||||
SoftFloat k1, k2;
|
||||
SoftFloat r0 = ps->r0, r1 = ps->r1;
|
||||
SoftFloat cor0 = ps->cor0, cor1 = ps->cor1;
|
||||
SoftFloat var0 = ps->var0, var1 = ps->var1;
|
||||
SoftFloat tmp;
|
||||
|
||||
if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) {
|
||||
k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0)));
|
||||
}
|
||||
else {
|
||||
k1.mant = 0;
|
||||
k1.exp = 0;
|
||||
}
|
||||
|
||||
if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) {
|
||||
k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1)));
|
||||
}
|
||||
else {
|
||||
k2.mant = 0;
|
||||
k2.exp = 0;
|
||||
}
|
||||
|
||||
tmp = av_mul_sf(k1, r0);
|
||||
pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1)));
|
||||
if (output_enable) {
|
||||
int shift = 28 - pv.exp;
|
||||
|
||||
if (shift < 31) {
|
||||
if (shift > 0) {
|
||||
*coef += (unsigned)((pv.mant + (1 << (shift - 1))) >> shift);
|
||||
} else
|
||||
*coef += (unsigned)pv.mant << -shift;
|
||||
}
|
||||
}
|
||||
|
||||
e0 = av_int2sf(*coef, 2);
|
||||
e1 = av_sub_sf(e0, tmp);
|
||||
|
||||
ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1)));
|
||||
tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1));
|
||||
tmp.exp--;
|
||||
ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp));
|
||||
ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0)));
|
||||
tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0));
|
||||
tmp.exp--;
|
||||
ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp));
|
||||
|
||||
ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0))));
|
||||
ps->r0 = flt16_trunc(av_mul_sf(a, e0));
|
||||
}
|
||||
|
||||
#include "aacdec_template.c"
|
||||
|
||||
const FFCodec ff_aac_fixed_decoder = {
|
||||
|
||||
@ -1072,13 +1072,6 @@ static int decode_audio_specific_config(AACDecContext *ac,
|
||||
sync_extension);
|
||||
}
|
||||
|
||||
static void reset_all_predictors(PredictorState *ps)
|
||||
{
|
||||
int i;
|
||||
for (i = 0; i < MAX_PREDICTORS; i++)
|
||||
reset_predict_state(&ps[i]);
|
||||
}
|
||||
|
||||
static int sample_rate_idx (int rate)
|
||||
{
|
||||
if (92017 <= rate) return 0;
|
||||
@ -1095,13 +1088,6 @@ static int sample_rate_idx (int rate)
|
||||
else return 11;
|
||||
}
|
||||
|
||||
static void reset_predictor_group(PredictorState *ps, int group_num)
|
||||
{
|
||||
int i;
|
||||
for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
|
||||
reset_predict_state(&ps[i]);
|
||||
}
|
||||
|
||||
static void aacdec_init(AACDecContext *ac);
|
||||
|
||||
static av_cold void aac_static_table_init(void)
|
||||
@ -1573,38 +1559,6 @@ static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
|
||||
}
|
||||
}
|
||||
|
||||
/**
|
||||
* Apply AAC-Main style frequency domain prediction.
|
||||
*/
|
||||
static void apply_prediction(AACDecContext *ac, SingleChannelElement *sce)
|
||||
{
|
||||
int sfb, k;
|
||||
|
||||
if (!sce->ics.predictor_initialized) {
|
||||
reset_all_predictors(sce->AAC_RENAME(predictor_state));
|
||||
sce->ics.predictor_initialized = 1;
|
||||
}
|
||||
|
||||
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
|
||||
for (sfb = 0;
|
||||
sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
|
||||
sfb++) {
|
||||
for (k = sce->ics.swb_offset[sfb];
|
||||
k < sce->ics.swb_offset[sfb + 1];
|
||||
k++) {
|
||||
predict(&sce->AAC_RENAME(predictor_state)[k],
|
||||
&sce->AAC_RENAME(coeffs)[k],
|
||||
sce->ics.predictor_present &&
|
||||
sce->ics.prediction_used[sfb]);
|
||||
}
|
||||
}
|
||||
if (sce->ics.predictor_reset_group)
|
||||
reset_predictor_group(sce->AAC_RENAME(predictor_state),
|
||||
sce->ics.predictor_reset_group);
|
||||
} else
|
||||
reset_all_predictors(sce->AAC_RENAME(predictor_state));
|
||||
}
|
||||
|
||||
static void decode_gain_control(SingleChannelElement * sce, GetBitContext * gb)
|
||||
{
|
||||
// wd_num, wd_test, aloc_size
|
||||
@ -1722,7 +1676,7 @@ int AAC_RENAME(ff_aac_decode_ics)(AACDecContext *ac, SingleChannelElement *sce,
|
||||
goto fail;
|
||||
|
||||
if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
|
||||
apply_prediction(ac, sce);
|
||||
ac->dsp.apply_prediction(ac, sce);
|
||||
|
||||
return 0;
|
||||
fail:
|
||||
@ -1767,8 +1721,8 @@ static int decode_cpe(AACDecContext *ac, GetBitContext *gb, ChannelElement *cpe)
|
||||
if (ms_present)
|
||||
ac->dsp.apply_mid_side_stereo(ac, cpe);
|
||||
if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
|
||||
apply_prediction(ac, &cpe->ch[0]);
|
||||
apply_prediction(ac, &cpe->ch[1]);
|
||||
ac->dsp.apply_prediction(ac, &cpe->ch[0]);
|
||||
ac->dsp.apply_prediction(ac, &cpe->ch[1]);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user