mirror of https://github.com/mpv-player/mpv
358 lines
10 KiB
C
358 lines
10 KiB
C
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/*
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** FAAD2 - Freeware Advanced Audio (AAC) Decoder including SBR decoding
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** Copyright (C) 2003 M. Bakker, Ahead Software AG, http://www.nero.com
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**
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** This program is free software; you can redistribute it and/or modify
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** it under the terms of the GNU General Public License as published by
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** the Free Software Foundation; either version 2 of the License, or
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** (at your option) any later version.
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**
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** This program 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
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** GNU General Public License for more details.
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**
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** You should have received a copy of the GNU General Public License
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** along with this program; if not, write to the Free Software
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** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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**
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** Any non-GPL usage of this software or parts of this software is strictly
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** forbidden.
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**
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** Commercial non-GPL licensing of this software is possible.
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** For more info contact Ahead Software through Mpeg4AAClicense@nero.com.
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**
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** $Id: sbr_dec.c,v 1.5 2003/07/29 08:20:13 menno Exp $
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**/
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/*
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SBR Decoder overview:
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To achieve a synchronized output signal, the following steps have to be
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acknowledged in the decoder:
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- The bitstream parser divides the bitstream into two parts; the AAC
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core coder part and the SBR part.
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- The SBR bitstream part is fed to the bitstream de-multiplexer followed
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by de-quantization The raw data is Huffman decoded.
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- The AAC bitstream part is fed to the AAC core decoder, where the
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bitstream data of the current frame is decoded, yielding a time domain
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audio signal block of 1024 samples. The block length could easily be
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adapted to other sizes e.g. 960.
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- The core coder audio block is fed to the analysis QMF bank using a
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delay of 1312 samples.
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- The analysis QMF bank performs the filtering of the delayed core coder
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audio signal. The output from the filtering is stored in the matrix
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Xlow. The output from the analysis QMF bank is delayed tHFGen subband
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samples, before being fed to the synthesis QMF bank. To achieve
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synchronization tHFGen = 32, i.e. the value must equal the number of
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subband samples corresponding to one frame.
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- The HF generator calculates XHigh given the matrix XLow. The process
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is guided by the SBR data contained in the current frame.
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- The envelope adjuster calculates the matrix Y given the matrix XHigh
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and the SBR envelope data, extracted from the SBR bitstream. To
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achieve synchronization, tHFAdj has to be set to tHFAdj = 0, i.e. the
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envelope adjuster operates on data delayed tHFGen subband samples.
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- The synthesis QMF bank operates on the delayed output from the analysis
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QMF bank and the output from the envelope adjuster.
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*/
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#include "common.h"
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#include "structs.h"
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#ifdef SBR_DEC
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#include <stdlib.h>
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#include "syntax.h"
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#include "bits.h"
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#include "sbr_syntax.h"
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#include "sbr_qmf.h"
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#include "sbr_hfgen.h"
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#include "sbr_hfadj.h"
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sbr_info *sbrDecodeInit()
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{
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sbr_info *sbr = malloc(sizeof(sbr_info));
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memset(sbr, 0, sizeof(sbr_info));
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sbr->bs_freq_scale = 2;
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sbr->bs_alter_scale = 1;
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sbr->bs_noise_bands = 2;
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sbr->bs_limiter_bands = 2;
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sbr->bs_limiter_gains = 2;
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sbr->bs_interpol_freq = 1;
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sbr->bs_smoothing_mode = 1;
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sbr->bs_start_freq = 5;
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sbr->bs_amp_res = 1;
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sbr->bs_samplerate_mode = 1;
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sbr->prevEnvIsShort[0] = -1;
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sbr->prevEnvIsShort[1] = -1;
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sbr->header_count = 0;
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return sbr;
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}
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void sbrDecodeEnd(sbr_info *sbr)
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{
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uint8_t j;
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if (sbr)
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{
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qmfa_end(sbr->qmfa[0]);
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qmfs_end(sbr->qmfs[0]);
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if (sbr->id_aac == ID_CPE)
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{
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qmfa_end(sbr->qmfa[1]);
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qmfs_end(sbr->qmfs[1]);
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}
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if (sbr->Xcodec[0]) free(sbr->Xcodec[0]);
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if (sbr->Xsbr[0]) free(sbr->Xsbr[0]);
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if (sbr->Xcodec[1]) free(sbr->Xcodec[1]);
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if (sbr->Xsbr[1]) free(sbr->Xsbr[1]);
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for (j = 0; j < 5; j++)
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{
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if (sbr->G_temp_prev[0][j]) free(sbr->G_temp_prev[0][j]);
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if (sbr->Q_temp_prev[0][j]) free(sbr->Q_temp_prev[0][j]);
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if (sbr->G_temp_prev[1][j]) free(sbr->G_temp_prev[1][j]);
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if (sbr->Q_temp_prev[1][j]) free(sbr->Q_temp_prev[1][j]);
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}
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free(sbr);
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}
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}
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void sbr_save_prev_data(sbr_info *sbr, uint8_t ch)
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{
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uint8_t i;
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/* save data for next frame */
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sbr->kx_prev = sbr->kx;
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sbr->L_E_prev[ch] = sbr->L_E[ch];
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sbr->f_prev[ch] = sbr->f[ch][sbr->L_E[ch] - 1];
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for (i = 0; i < 64; i++)
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{
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sbr->E_prev[ch][i] = sbr->E[ch][i][sbr->L_E[ch] - 1];
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sbr->Q_prev[ch][i] = sbr->Q[ch][i][sbr->L_Q[ch] - 1];
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}
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for (i = 0; i < 64; i++)
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{
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sbr->bs_add_harmonic_prev[ch][i] = sbr->bs_add_harmonic[ch][i];
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}
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sbr->bs_add_harmonic_flag_prev[ch] = sbr->bs_add_harmonic_flag[ch];
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if (sbr->l_A[ch] == sbr->L_E[ch])
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sbr->prevEnvIsShort[ch] = 0;
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else
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sbr->prevEnvIsShort[ch] = -1;
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}
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void sbrDecodeFrame(sbr_info *sbr, real_t *left_channel,
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real_t *right_channel, uint8_t id_aac,
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uint8_t just_seeked)
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{
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int16_t i, k, l;
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uint8_t dont_process = 0;
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uint8_t ch, channels, ret;
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real_t *ch_buf;
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qmf_t X[32*64];
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#ifdef SBR_LOW_POWER
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real_t deg[64];
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#endif
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bitfile *ld = (bitfile*)malloc(sizeof(bitfile));
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sbr->id_aac = id_aac;
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channels = (id_aac == ID_SCE) ? 1 : 2;
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/* initialise and read the bitstream */
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faad_initbits(ld, sbr->data, sbr->data_size);
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ret = sbr_extension_data(ld, sbr, id_aac);
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if (sbr->data) free(sbr->data);
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sbr->data = NULL;
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ret = ld->error ? ld->error : ret;
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faad_endbits(ld);
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if (ld) free(ld);
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ld = NULL;
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if (ret || (sbr->header_count == 0))
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{
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/* don't process just upsample */
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dont_process = 1;
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}
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if (just_seeked)
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sbr->just_seeked = 1;
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else
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sbr->just_seeked = 0;
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for (ch = 0; ch < channels; ch++)
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{
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if (sbr->frame == 0)
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{
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uint8_t j;
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sbr->qmfa[ch] = qmfa_init(32);
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sbr->qmfs[ch] = qmfs_init(64);
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for (j = 0; j < 5; j++)
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{
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sbr->G_temp_prev[ch][j] = malloc(64*sizeof(real_t));
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sbr->Q_temp_prev[ch][j] = malloc(64*sizeof(real_t));
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}
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sbr->Xsbr[ch] = malloc((32+tHFGen)*64 * sizeof(qmf_t));
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sbr->Xcodec[ch] = malloc((32+tHFGen)*32 * sizeof(qmf_t));
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memset(sbr->Xsbr[ch], 0, (32+tHFGen)*64 * sizeof(qmf_t));
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memset(sbr->Xcodec[ch], 0, (32+tHFGen)*32 * sizeof(qmf_t));
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}
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if (ch == 0)
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ch_buf = left_channel;
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else
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ch_buf = right_channel;
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for (i = 0; i < tHFAdj; i++)
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{
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int8_t j;
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for (j = sbr->kx_prev; j < sbr->kx; j++)
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{
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QMF_RE(sbr->Xcodec[ch][i*32 + j]) = 0;
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#ifndef SBR_LOW_POWER
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QMF_IM(sbr->Xcodec[ch][i*32 + j]) = 0;
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#endif
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}
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}
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/* subband analysis */
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sbr_qmf_analysis_32(sbr->qmfa[ch], ch_buf, sbr->Xcodec[ch], tHFGen);
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if (!dont_process)
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{
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/* insert high frequencies here */
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/* hf generation using patching */
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hf_generation(sbr, sbr->Xcodec[ch], sbr->Xsbr[ch]
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#ifdef SBR_LOW_POWER
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,deg
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#endif
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,ch);
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#ifdef SBR_LOW_POWER
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for (l = sbr->t_E[ch][0]; l < sbr->t_E[ch][sbr->L_E[ch]]; l++)
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{
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for (k = 0; k < sbr->kx; k++)
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{
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QMF_RE(sbr->Xsbr[ch][(tHFAdj + l)*64 + k]) = 0;
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}
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}
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#endif
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/* hf adjustment */
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hf_adjustment(sbr, sbr->Xsbr[ch]
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#ifdef SBR_LOW_POWER
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,deg
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#endif
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,ch);
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}
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if ((sbr->just_seeked != 0) || dont_process)
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{
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for (l = 0; l < 32; l++)
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{
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for (k = 0; k < 32; k++)
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{
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QMF_RE(X[l * 64 + k]) = QMF_RE(sbr->Xcodec[ch][(l + tHFAdj)*32 + k]);
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#ifndef SBR_LOW_POWER
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QMF_IM(X[l * 64 + k]) = QMF_IM(sbr->Xcodec[ch][(l + tHFAdj)*32 + k]);
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#endif
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}
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for (k = 32; k < 64; k++)
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{
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QMF_RE(X[l * 64 + k]) = 0;
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#ifndef SBR_LOW_POWER
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QMF_IM(X[l * 64 + k]) = 0;
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#endif
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}
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}
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} else {
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for (l = 0; l < 32; l++)
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{
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uint8_t xover_band;
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if (l < sbr->t_E[ch][0])
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xover_band = sbr->kx_prev;
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else
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xover_band = sbr->kx;
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for (k = 0; k < xover_band; k++)
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{
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QMF_RE(X[l * 64 + k]) = QMF_RE(sbr->Xcodec[ch][(l + tHFAdj)*32 + k]);
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#ifndef SBR_LOW_POWER
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QMF_IM(X[l * 64 + k]) = QMF_IM(sbr->Xcodec[ch][(l + tHFAdj)*32 + k]);
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#endif
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}
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for (k = xover_band; k < 64; k++)
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{
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QMF_RE(X[l * 64 + k]) = QMF_RE(sbr->Xsbr[ch][(l + tHFAdj)*64 + k]);
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#ifndef SBR_LOW_POWER
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QMF_IM(X[l * 64 + k]) = QMF_IM(sbr->Xsbr[ch][(l + tHFAdj)*64 + k]);
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#endif
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}
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#ifdef SBR_LOW_POWER
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QMF_RE(X[l * 64 + xover_band - 1]) += QMF_RE(sbr->Xsbr[ch][(l + tHFAdj)*64 + xover_band - 1]);
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#endif
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}
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}
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/* subband synthesis */
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sbr_qmf_synthesis_64(sbr->qmfs[ch], (const complex_t*)X, ch_buf);
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for (i = 0; i < 32; i++)
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{
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int8_t j;
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for (j = 0; j < tHFGen; j++)
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{
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QMF_RE(sbr->Xcodec[ch][j*32 + i]) = QMF_RE(sbr->Xcodec[ch][(j+32)*32 + i]);
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#ifndef SBR_LOW_POWER
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QMF_IM(sbr->Xcodec[ch][j*32 + i]) = QMF_IM(sbr->Xcodec[ch][(j+32)*32 + i]);
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#endif
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}
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}
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for (i = 0; i < 64; i++)
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{
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int8_t j;
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for (j = 0; j < tHFGen; j++)
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{
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QMF_RE(sbr->Xsbr[ch][j*64 + i]) = QMF_RE(sbr->Xsbr[ch][(j+32)*64 + i]);
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#ifndef SBR_LOW_POWER
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QMF_IM(sbr->Xsbr[ch][j*64 + i]) = QMF_IM(sbr->Xsbr[ch][(j+32)*64 + i]);
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#endif
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}
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}
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}
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if (sbr->bs_header_flag)
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sbr->just_seeked = 0;
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if (sbr->header_count != 0)
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{
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for (ch = 0; ch < channels; ch++)
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sbr_save_prev_data(sbr, ch);
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}
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sbr->frame++;
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}
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#endif
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