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https://github.com/FFmpeg/FFmpeg.git
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469cd8d7fa
This patch replaces the transform used in AAC with lavu/tx and removes the limitation on only being able to decode 960-sample files with the float decoder. This commit also removes a whole bunch of unnecessary and slow lifting steps the decoder did to compensate for the poor accuracy of the old integer transformation code. Overall float decoder speedup on Zen 3 for 64kbps: 32%
392 lines
13 KiB
C
392 lines
13 KiB
C
/*
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* AAC definitions and structures
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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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*
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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 definitions and structures
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* @author Oded Shimon ( ods15 ods15 dyndns org )
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* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
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*/
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#ifndef AVCODEC_AAC_H
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#define AVCODEC_AAC_H
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#include "aac_defines.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/fixed_dsp.h"
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#include "libavutil/mem_internal.h"
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#include "libavutil/tx.h"
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#include "avcodec.h"
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#include "mpeg4audio.h"
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#include "sbr.h"
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#include <stdint.h>
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#define MAX_CHANNELS 64
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#define MAX_ELEM_ID 16
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#define TNS_MAX_ORDER 20
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#define MAX_LTP_LONG_SFB 40
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#define CLIP_AVOIDANCE_FACTOR 0.95f
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enum RawDataBlockType {
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TYPE_SCE,
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TYPE_CPE,
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TYPE_CCE,
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TYPE_LFE,
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TYPE_DSE,
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TYPE_PCE,
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TYPE_FIL,
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TYPE_END,
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};
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enum ExtensionPayloadID {
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EXT_FILL,
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EXT_FILL_DATA,
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EXT_DATA_ELEMENT,
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EXT_DYNAMIC_RANGE = 0xb,
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EXT_SBR_DATA = 0xd,
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EXT_SBR_DATA_CRC = 0xe,
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};
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enum WindowSequence {
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ONLY_LONG_SEQUENCE,
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LONG_START_SEQUENCE,
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EIGHT_SHORT_SEQUENCE,
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LONG_STOP_SEQUENCE,
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};
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enum BandType {
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ZERO_BT = 0, ///< Scalefactors and spectral data are all zero.
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FIRST_PAIR_BT = 5, ///< This and later band types encode two values (rather than four) with one code word.
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ESC_BT = 11, ///< Spectral data are coded with an escape sequence.
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RESERVED_BT = 12, ///< Band types following are encoded differently from others.
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NOISE_BT = 13, ///< Spectral data are scaled white noise not coded in the bitstream.
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INTENSITY_BT2 = 14, ///< Scalefactor data are intensity stereo positions (out of phase).
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INTENSITY_BT = 15, ///< Scalefactor data are intensity stereo positions (in phase).
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};
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#define IS_CODEBOOK_UNSIGNED(x) (((x) - 1) & 10)
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enum ChannelPosition {
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AAC_CHANNEL_OFF = 0,
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AAC_CHANNEL_FRONT = 1,
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AAC_CHANNEL_SIDE = 2,
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AAC_CHANNEL_BACK = 3,
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AAC_CHANNEL_LFE = 4,
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AAC_CHANNEL_CC = 5,
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};
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/**
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* The point during decoding at which channel coupling is applied.
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*/
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enum CouplingPoint {
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BEFORE_TNS,
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BETWEEN_TNS_AND_IMDCT,
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AFTER_IMDCT = 3,
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};
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/**
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* Output configuration status
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*/
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enum OCStatus {
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OC_NONE, ///< Output unconfigured
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OC_TRIAL_PCE, ///< Output configuration under trial specified by an inband PCE
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OC_TRIAL_FRAME, ///< Output configuration under trial specified by a frame header
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OC_GLOBAL_HDR, ///< Output configuration set in a global header but not yet locked
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OC_LOCKED, ///< Output configuration locked in place
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};
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typedef struct OutputConfiguration {
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MPEG4AudioConfig m4ac;
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uint8_t layout_map[MAX_ELEM_ID*4][3];
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int layout_map_tags;
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AVChannelLayout ch_layout;
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enum OCStatus status;
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} OutputConfiguration;
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/**
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* Predictor State
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*/
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typedef struct PredictorState {
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AAC_FLOAT cor0;
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AAC_FLOAT cor1;
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AAC_FLOAT var0;
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AAC_FLOAT var1;
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AAC_FLOAT r0;
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AAC_FLOAT r1;
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AAC_FLOAT k1;
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AAC_FLOAT x_est;
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} PredictorState;
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#define MAX_PREDICTORS 672
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#define SCALE_DIV_512 36 ///< scalefactor difference that corresponds to scale difference in 512 times
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#define SCALE_ONE_POS 140 ///< scalefactor index that corresponds to scale=1.0
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#define SCALE_MAX_POS 255 ///< scalefactor index maximum value
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#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard
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#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference
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#define POW_SF2_ZERO 200 ///< ff_aac_pow2sf_tab index corresponding to pow(2, 0);
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#define NOISE_PRE 256 ///< preamble for NOISE_BT, put in bitstream with the first noise band
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#define NOISE_PRE_BITS 9 ///< length of preamble
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#define NOISE_OFFSET 90 ///< subtracted from global gain, used as offset for the preamble
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/**
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* Long Term Prediction
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*/
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typedef struct LongTermPrediction {
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int8_t present;
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int16_t lag;
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int coef_idx;
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INTFLOAT coef;
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int8_t used[MAX_LTP_LONG_SFB];
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} LongTermPrediction;
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/**
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* Individual Channel Stream
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*/
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typedef struct IndividualChannelStream {
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uint8_t max_sfb; ///< number of scalefactor bands per group
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enum WindowSequence window_sequence[2];
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uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sine window.
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int num_window_groups;
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uint8_t group_len[8];
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LongTermPrediction ltp;
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const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
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const uint8_t *swb_sizes; ///< table of scalefactor band sizes for a particular window
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int num_swb; ///< number of scalefactor window bands
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int num_windows;
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int tns_max_bands;
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int predictor_present;
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int predictor_initialized;
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int predictor_reset_group;
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int predictor_reset_count[31]; ///< used by encoder to count prediction resets
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uint8_t prediction_used[41];
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uint8_t window_clipping[8]; ///< set if a certain window is near clipping
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float clip_avoidance_factor; ///< set if any window is near clipping to the necessary atennuation factor to avoid it
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} IndividualChannelStream;
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/**
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* Temporal Noise Shaping
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*/
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typedef struct TemporalNoiseShaping {
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int present;
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int n_filt[8];
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int length[8][4];
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int direction[8][4];
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int order[8][4];
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int coef_idx[8][4][TNS_MAX_ORDER];
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INTFLOAT coef[8][4][TNS_MAX_ORDER];
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} TemporalNoiseShaping;
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/**
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* Dynamic Range Control - decoded from the bitstream but not processed further.
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*/
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typedef struct DynamicRangeControl {
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int pce_instance_tag; ///< Indicates with which program the DRC info is associated.
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int dyn_rng_sgn[17]; ///< DRC sign information; 0 - positive, 1 - negative
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int dyn_rng_ctl[17]; ///< DRC magnitude information
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int exclude_mask[MAX_CHANNELS]; ///< Channels to be excluded from DRC processing.
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int band_incr; ///< Number of DRC bands greater than 1 having DRC info.
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int interpolation_scheme; ///< Indicates the interpolation scheme used in the SBR QMF domain.
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int band_top[17]; ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.
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int prog_ref_level; /**< A reference level for the long-term program audio level for all
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* channels combined.
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*/
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} DynamicRangeControl;
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typedef struct Pulse {
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int num_pulse;
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int start;
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int pos[4];
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int amp[4];
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} Pulse;
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/**
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* coupling parameters
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*/
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typedef struct ChannelCoupling {
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enum CouplingPoint coupling_point; ///< The point during decoding at which coupling is applied.
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int num_coupled; ///< number of target elements
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enum RawDataBlockType type[8]; ///< Type of channel element to be coupled - SCE or CPE.
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int id_select[8]; ///< element id
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int ch_select[8]; /**< [0] shared list of gains; [1] list of gains for right channel;
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* [2] list of gains for left channel; [3] lists of gains for both channels
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*/
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INTFLOAT gain[16][120];
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} ChannelCoupling;
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/**
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* Single Channel Element - used for both SCE and LFE elements.
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*/
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typedef struct SingleChannelElement {
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IndividualChannelStream ics;
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TemporalNoiseShaping tns;
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Pulse pulse;
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enum BandType band_type[128]; ///< band types
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enum BandType band_alt[128]; ///< alternative band type (used by encoder)
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int band_type_run_end[120]; ///< band type run end points
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INTFLOAT sf[120]; ///< scalefactors
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int sf_idx[128]; ///< scalefactor indices (used by encoder)
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uint8_t zeroes[128]; ///< band is not coded (used by encoder)
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uint8_t can_pns[128]; ///< band is allowed to PNS (informative)
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float is_ener[128]; ///< Intensity stereo pos (used by encoder)
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float pns_ener[128]; ///< Noise energy values (used by encoder)
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DECLARE_ALIGNED(32, INTFLOAT, pcoeffs)[1024]; ///< coefficients for IMDCT, pristine
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DECLARE_ALIGNED(32, INTFLOAT, coeffs)[1024]; ///< coefficients for IMDCT, maybe processed
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DECLARE_ALIGNED(32, INTFLOAT, saved)[1536]; ///< overlap
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DECLARE_ALIGNED(32, INTFLOAT, ret_buf)[2048]; ///< PCM output buffer
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DECLARE_ALIGNED(16, INTFLOAT, ltp_state)[3072]; ///< time signal for LTP
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DECLARE_ALIGNED(32, AAC_FLOAT, lcoeffs)[1024]; ///< MDCT of LTP coefficients (used by encoder)
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DECLARE_ALIGNED(32, AAC_FLOAT, prcoeffs)[1024]; ///< Main prediction coefs (used by encoder)
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PredictorState predictor_state[MAX_PREDICTORS];
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INTFLOAT *ret; ///< PCM output
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} SingleChannelElement;
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/**
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* channel element - generic struct for SCE/CPE/CCE/LFE
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*/
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typedef struct ChannelElement {
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int present;
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// CPE specific
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int common_window; ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
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int ms_mode; ///< Signals mid/side stereo flags coding mode (used by encoder)
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uint8_t is_mode; ///< Set if any bands have been encoded using intensity stereo (used by encoder)
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uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band
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uint8_t is_mask[128]; ///< Set if intensity stereo is used (used by encoder)
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// shared
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SingleChannelElement ch[2];
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// CCE specific
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ChannelCoupling coup;
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SpectralBandReplication sbr;
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} ChannelElement;
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enum AACOutputChannelOrder {
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CHANNEL_ORDER_DEFAULT,
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CHANNEL_ORDER_CODED,
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};
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/**
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* main AAC context
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*/
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struct AACContext {
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AVClass *class;
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AVCodecContext *avctx;
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AVFrame *frame;
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int is_saved; ///< Set if elements have stored overlap from previous frame.
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DynamicRangeControl che_drc;
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/**
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* @name Channel element related data
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* @{
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*/
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ChannelElement *che[4][MAX_ELEM_ID];
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ChannelElement *tag_che_map[4][MAX_ELEM_ID];
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int tags_mapped;
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int warned_remapping_once;
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/** @} */
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/**
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* @name temporary aligned temporary buffers
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* (We do not want to have these on the stack.)
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* @{
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*/
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DECLARE_ALIGNED(32, INTFLOAT, buf_mdct)[1024];
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/** @} */
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/**
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* @name Computed / set up during initialization
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* @{
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*/
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AVTXContext *mdct120;
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AVTXContext *mdct128;
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AVTXContext *mdct480;
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AVTXContext *mdct512;
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AVTXContext *mdct960;
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AVTXContext *mdct1024;
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AVTXContext *mdct_ltp;
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av_tx_fn mdct120_fn;
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av_tx_fn mdct128_fn;
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av_tx_fn mdct480_fn;
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av_tx_fn mdct512_fn;
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av_tx_fn mdct960_fn;
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av_tx_fn mdct1024_fn;
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av_tx_fn mdct_ltp_fn;
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#if USE_FIXED
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AVFixedDSPContext *fdsp;
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#else
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AVFloatDSPContext *fdsp;
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#endif /* USE_FIXED */
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int random_state;
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/** @} */
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/**
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* @name Members used for output
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* @{
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*/
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SingleChannelElement *output_element[MAX_CHANNELS]; ///< Points to each SingleChannelElement
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/** @} */
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/**
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* @name Japanese DTV specific extension
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* @{
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*/
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int force_dmono_mode;///< 0->not dmono, 1->use first channel, 2->use second channel
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int dmono_mode; ///< 0->not dmono, 1->use first channel, 2->use second channel
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/** @} */
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enum AACOutputChannelOrder output_channel_order;
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DECLARE_ALIGNED(32, INTFLOAT, temp)[128];
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OutputConfiguration oc[2];
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int warned_num_aac_frames;
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int warned_960_sbr;
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unsigned warned_71_wide;
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int warned_gain_control;
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int warned_he_aac_mono;
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/* aacdec functions pointers */
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void (*imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce);
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void (*apply_ltp)(AACContext *ac, SingleChannelElement *sce);
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void (*apply_tns)(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
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IndividualChannelStream *ics, int decode);
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void (*windowing_and_mdct_ltp)(AACContext *ac, INTFLOAT *out,
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INTFLOAT *in, IndividualChannelStream *ics);
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void (*update_ltp)(AACContext *ac, SingleChannelElement *sce);
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void (*vector_pow43)(int *coefs, int len);
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void (*subband_scale)(int *dst, int *src, int scale, int offset, int len, void *log_context);
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};
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void ff_aacdec_init_mips(AACContext *c);
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#endif /* AVCODEC_AAC_H */
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