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/*
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* AAC decoder 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 decoder 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_AACDEC_H
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#define AVCODEC_AAC_AACDEC_H
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#include <stdint.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 "libavcodec/aac.h"
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#include "libavcodec/avcodec.h"
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#include "libavcodec/mpeg4audio.h"
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#include "aacdec_ac.h"
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typedef struct AACDecContext AACDecContext;
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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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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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* 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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enum AACUsacElem {
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ID_USAC_SCE = 0,
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ID_USAC_CPE = 1,
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ID_USAC_LFE = 2,
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ID_USAC_EXT = 3,
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};
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enum ExtensionHeaderType {
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ID_CONFIG_EXT_FILL = 0,
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ID_CONFIG_EXT_LOUDNESS_INFO = 2,
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ID_CONFIG_EXT_STREAM_ID = 7,
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};
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enum AACUsacExtension {
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ID_EXT_ELE_FILL,
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ID_EXT_ELE_MPEGS,
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ID_EXT_ELE_SAOC,
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ID_EXT_ELE_AUDIOPREROLL,
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ID_EXT_ELE_UNI_DRC,
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};
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enum AACUSACLoudnessExt {
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UNIDRCLOUDEXT_TERM = 0x0,
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UNIDRCLOUDEXT_EQ = 0x1,
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};
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// Supposed to be equal to AAC_RENAME() in case of USE_FIXED.
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#define RENAME_FIXED(name) name ## _fixed
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#define INTFLOAT_UNION(name, elems) \
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union { \
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int RENAME_FIXED(name) elems; \
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float name elems; \
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}
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#define INTFLOAT_ALIGNED_UNION(alignment, name, nb_elems) \
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union { \
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DECLARE_ALIGNED(alignment, int, RENAME_FIXED(name))[nb_elems]; \
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DECLARE_ALIGNED(alignment, float, name)[nb_elems]; \
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}
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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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INTFLOAT_UNION(coef,);
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int8_t used[MAX_LTP_LONG_SFB];
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} LongTermPrediction;
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/* Per channel core mode */
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typedef struct AACUsacElemData {
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uint8_t core_mode;
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uint8_t scale_factor_grouping;
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uint8_t tns_data_present;
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/* Timewarping ratio */
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#define NUM_TW_NODES 16
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uint8_t tw_ratio[NUM_TW_NODES];
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struct {
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uint8_t acelp_core_mode : 3;
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uint8_t lpd_mode : 5;
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uint8_t bpf_control_info : 1;
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uint8_t core_mode_last : 1;
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uint8_t fac_data_present : 1;
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int last_lpd_mode;
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} ldp;
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struct {
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unsigned int seed;
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uint8_t level : 3;
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uint8_t offset : 5;
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} noise;
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struct {
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uint8_t gain;
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uint32_t kv[8 /* (1024 / 16) / 8 */][8];
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} fac;
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AACArithState ac;
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} AACUsacElemData;
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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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int prev_num_window_groups; ///< Previous frame's number of 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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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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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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} 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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INTFLOAT_UNION(coef, [8][4][TNS_MAX_ORDER]);
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} TemporalNoiseShaping;
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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_UNION(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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AACUsacElemData ue; ///< USAC element data
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TemporalNoiseShaping tns;
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enum BandType band_type[128]; ///< band types
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aacdec: move from scalefactor ranged arrays to flat arrays
AAC uses an unconventional system to send scalefactors
(the volume+quantization value for each band).
Each window is split into either 1 or 8 blocks (long vs short),
and transformed separately from one another, with the coefficients
for each being also completely independent. The scalefactors
slightly increase from 64 (long) to 128 (short) to accomodate
better per-block-per-band volume for each window.
To reduce overhead, the codec signals scalefactor sizes in an obtuse way,
where each group's scalefactor types are sent via a variable length decoding,
with a range.
But our decoder was written in a way where those ranges were carried through
the entire decoder, and to actually read them you had to use the range.
Instead of having a dedicated array with a range for each scalefactor,
just let the decoder directly index each scalefactor.
This also switches the form of quantized scalefactors to the format
the spec uses, where for intensity stereo and regular, scalefactors
are stored in a scalefactor - 100 form, rather than as-is.
USAC gets rid of the complex scalefactor handling. This commit permits
for code sharing between both.
8 months ago
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int sfo[128]; ///< scalefactor offsets
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INTFLOAT_UNION(sf, [128]); ///< scalefactors (8 windows * 16 sfb max)
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INTFLOAT_ALIGNED_UNION(32, coeffs, 1024); ///< coefficients for IMDCT, maybe processed
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INTFLOAT_ALIGNED_UNION(32, prev_coeffs, 1024); ///< unscaled previous contents of coeffs[] for USAC
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INTFLOAT_ALIGNED_UNION(32, saved, 1536); ///< overlap
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INTFLOAT_ALIGNED_UNION(32, ret_buf, 2048); ///< PCM output buffer
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INTFLOAT_ALIGNED_UNION(16, ltp_state, 3072); ///< time signal for LTP
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union {
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struct PredictorStateFixed *RENAME_FIXED(predictor_state);
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struct PredictorState *predictor_state;
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};
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union {
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float *output; ///< PCM output
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int *RENAME_FIXED(output); ///< PCM output
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};
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} SingleChannelElement;
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typedef struct AACUsacStereo {
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uint8_t common_window;
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uint8_t common_tw;
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uint8_t tns_on_lr; ///< Apply TNS before M/S and stereo prediction
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uint8_t ms_mask_mode;
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uint8_t config_idx;
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/* Complex prediction */
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uint8_t use_prev_frame;
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uint8_t pred_dir;
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uint8_t complex_coef;
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uint8_t pred_used[128];
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INTFLOAT_ALIGNED_UNION(32, alpha_q_re, 1024);
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INTFLOAT_ALIGNED_UNION(32, alpha_q_im, 1024);
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INTFLOAT_ALIGNED_UNION(32, prev_alpha_q_re, 1024);
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INTFLOAT_ALIGNED_UNION(32, prev_alpha_q_im, 1024);
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INTFLOAT_ALIGNED_UNION(32, dmix_re, 1024);
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INTFLOAT_ALIGNED_UNION(32, prev_dmix_re, 1024); /* Recalculated on every frame */
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INTFLOAT_ALIGNED_UNION(32, dmix_im, 1024); /* Final prediction data */
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} AACUsacStereo;
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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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uint8_t max_sfb_ste; ///< (USAC) Maximum of both max_sfb values
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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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// shared
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SingleChannelElement ch[2];
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// CCE specific
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ChannelCoupling coup;
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// USAC stereo coupling data
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AACUsacStereo us;
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} ChannelElement;
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typedef struct AACUSACLoudnessInfo {
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uint8_t drc_set_id : 6;
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uint8_t downmix_id : 7;
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struct {
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uint16_t lvl : 12;
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uint8_t present : 1;
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} sample_peak;
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struct {
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uint16_t lvl : 12;
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uint8_t measurement : 4;
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uint8_t reliability : 2;
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uint8_t present : 1;
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} true_peak;
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uint8_t nb_measurements : 4;
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struct {
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uint8_t method_def : 4;
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uint8_t method_val;
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uint8_t measurement : 4;
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uint8_t reliability : 2;
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} measurements[16];
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} AACUSACLoudnessInfo;
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typedef struct AACUsacElemConfig {
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enum AACUsacElem type;
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uint8_t tw_mdct : 1;
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uint8_t noise_fill : 1;
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uint8_t stereo_config_index;
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struct {
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int ratio;
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uint8_t harmonic_sbr : 1; /* harmonicSBR */
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uint8_t bs_intertes : 1; /* bs_interTes */
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uint8_t bs_pvc : 1; /* bs_pvc */
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struct {
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uint8_t start_freq; /* dflt_start_freq */
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uint8_t stop_freq; /* dflt_stop_freq */
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uint8_t freq_scale; /* dflt_freq_scale */
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uint8_t alter_scale : 1; /* dflt_alter_scale */
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uint8_t noise_bands; /* dflt_noise_bands */
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uint8_t limiter_bands; /* dflt_limiter_bands */
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uint8_t limiter_gains; /* dflt_limiter_gains */
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uint8_t interpol_freq : 1; /* dflt_interpol_freq */
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uint8_t smoothing_mode : 1; /* dflt_smoothing_mode */
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} dflt;
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} sbr;
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struct {
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uint8_t freq_res; /* bsFreqRes */
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uint8_t fixed_gain; /* bsFixedGainDMX */
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uint8_t temp_shape_config; /* bsTempShapeConfig */
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uint8_t decorr_config; /* bsDecorrConfig */
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uint8_t high_rate_mode : 1; /* bsHighRateMode */
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uint8_t phase_coding : 1; /* bsPhaseCoding */
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uint8_t otts_bands_phase; /* bsOttBandsPhase */
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uint8_t residual_coding; /* bsResidualCoding */
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uint8_t residual_bands; /* bsResidualBands */
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uint8_t pseudo_lr : 1; /* bsPseudoLr */
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uint8_t env_quant_mode : 1; /* bsEnvQuantMode */
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} mps;
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struct {
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enum AACUsacExtension type;
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uint8_t payload_frag;
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uint32_t default_len;
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uint32_t pl_data_offset;
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uint8_t *pl_data;
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} ext;
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} AACUsacElemConfig;
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typedef struct AACUSACConfig {
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uint8_t core_sbr_frame_len_idx; /* coreSbrFrameLengthIndex */
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uint16_t core_frame_len;
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|
uint16_t stream_identifier;
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AACUsacElemConfig elems[64];
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|
int nb_elems;
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|
struct {
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|
uint8_t nb_album;
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|
AACUSACLoudnessInfo album_info[64];
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|
uint8_t nb_info;
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|
AACUSACLoudnessInfo info[64];
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|
} loudness;
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|
} AACUSACConfig;
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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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|
|
AACUSACConfig usac;
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|
|
|
} OutputConfiguration;
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|
/**
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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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|
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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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|
|
|
|
|
/**
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|
|
|
* Decode-specific primitives
|
|
|
|
*/
|
|
|
|
typedef struct AACDecProc {
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|
int (*decode_spectrum_and_dequant)(AACDecContext *ac,
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|
|
GetBitContext *gb,
|
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|
|
const Pulse *pulse,
|
|
|
|
SingleChannelElement *sce);
|
|
|
|
|
|
|
|
int (*decode_cce)(AACDecContext *ac, GetBitContext *gb, ChannelElement *che);
|
|
|
|
|
|
|
|
int (*sbr_ctx_alloc_init)(AACDecContext *ac, ChannelElement **che, int id_aac);
|
|
|
|
int (*sbr_decode_extension)(AACDecContext *ac, ChannelElement *che,
|
|
|
|
GetBitContext *gb, int crc, int cnt, int id_aac);
|
|
|
|
void (*sbr_apply)(AACDecContext *ac, ChannelElement *che,
|
|
|
|
int id_aac, void /* INTFLOAT */ *L, void /* INTFLOAT */ *R);
|
|
|
|
void (*sbr_ctx_close)(ChannelElement *che);
|
|
|
|
} AACDecProc;
|
|
|
|
|
|
|
|
/**
|
|
|
|
* DSP-specific primitives
|
|
|
|
*/
|
|
|
|
typedef struct AACDecDSP {
|
|
|
|
void (*dequant_scalefactors)(SingleChannelElement *sce);
|
|
|
|
|
|
|
|
void (*apply_mid_side_stereo)(AACDecContext *ac, ChannelElement *cpe);
|
|
|
|
void (*apply_intensity_stereo)(AACDecContext *ac, ChannelElement *cpe,
|
|
|
|
int ms_present);
|
|
|
|
|
|
|
|
void (*apply_tns)(void *_coef_param, TemporalNoiseShaping *tns,
|
|
|
|
IndividualChannelStream *ics, int decode);
|
|
|
|
|
|
|
|
void (*apply_ltp)(AACDecContext *ac, SingleChannelElement *sce);
|
|
|
|
void (*update_ltp)(AACDecContext *ac, SingleChannelElement *sce);
|
|
|
|
|
|
|
|
void (*apply_prediction)(AACDecContext *ac, SingleChannelElement *sce);
|
|
|
|
|
|
|
|
void (*apply_dependent_coupling)(AACDecContext *ac,
|
|
|
|
SingleChannelElement *target,
|
|
|
|
ChannelElement *cce, int index);
|
|
|
|
void (*apply_independent_coupling)(AACDecContext *ac,
|
|
|
|
SingleChannelElement *target,
|
|
|
|
ChannelElement *cce, int index);
|
|
|
|
|
|
|
|
void (*imdct_and_windowing)(AACDecContext *ac, SingleChannelElement *sce);
|
|
|
|
void (*imdct_and_windowing_768)(AACDecContext *ac, SingleChannelElement *sce);
|
|
|
|
void (*imdct_and_windowing_960)(AACDecContext *ac, SingleChannelElement *sce);
|
|
|
|
void (*imdct_and_windowing_ld)(AACDecContext *ac, SingleChannelElement *sce);
|
|
|
|
void (*imdct_and_windowing_eld)(AACDecContext *ac, SingleChannelElement *sce);
|
|
|
|
|
|
|
|
void (*clip_output)(AACDecContext *ac, ChannelElement *che, int type, int samples);
|
|
|
|
} AACDecDSP;
|
|
|
|
|
|
|
|
/**
|
|
|
|
* main AAC decoding context
|
|
|
|
*/
|
|
|
|
struct AACDecContext {
|
|
|
|
const struct AVClass *class;
|
|
|
|
struct AVCodecContext *avctx;
|
|
|
|
|
|
|
|
AACDecDSP dsp;
|
|
|
|
AACDecProc proc;
|
|
|
|
|
|
|
|
struct AVFrame *frame;
|
|
|
|
|
|
|
|
int is_saved; ///< Set if elements have stored overlap from previous frame.
|
|
|
|
DynamicRangeControl che_drc;
|
|
|
|
|
|
|
|
/**
|
|
|
|
* @name Channel element related data
|
|
|
|
* @{
|
|
|
|
*/
|
|
|
|
ChannelElement *che[4][MAX_ELEM_ID];
|
|
|
|
ChannelElement *tag_che_map[4][MAX_ELEM_ID];
|
|
|
|
int tags_mapped;
|
|
|
|
int warned_remapping_once;
|
|
|
|
/** @} */
|
|
|
|
|
|
|
|
/**
|
|
|
|
* @name temporary aligned temporary buffers
|
|
|
|
* (We do not want to have these on the stack.)
|
|
|
|
* @{
|
|
|
|
*/
|
|
|
|
INTFLOAT_ALIGNED_UNION(32, buf_mdct, 1024);
|
|
|
|
INTFLOAT_ALIGNED_UNION(32, temp, 128);
|
|
|
|
/** @} */
|
|
|
|
|
|
|
|
/**
|
|
|
|
* @name Computed / set up during initialization
|
|
|
|
* @{
|
|
|
|
*/
|
|
|
|
AVTXContext *mdct96;
|
|
|
|
AVTXContext *mdct120;
|
|
|
|
AVTXContext *mdct128;
|
|
|
|
AVTXContext *mdct480;
|
|
|
|
AVTXContext *mdct512;
|
|
|
|
AVTXContext *mdct768;
|
|
|
|
AVTXContext *mdct960;
|
|
|
|
AVTXContext *mdct1024;
|
|
|
|
AVTXContext *mdct_ltp;
|
|
|
|
|
|
|
|
av_tx_fn mdct96_fn;
|
|
|
|
av_tx_fn mdct120_fn;
|
|
|
|
av_tx_fn mdct128_fn;
|
|
|
|
av_tx_fn mdct480_fn;
|
|
|
|
av_tx_fn mdct512_fn;
|
|
|
|
av_tx_fn mdct768_fn;
|
|
|
|
av_tx_fn mdct960_fn;
|
|
|
|
av_tx_fn mdct1024_fn;
|
|
|
|
av_tx_fn mdct_ltp_fn;
|
|
|
|
union {
|
|
|
|
AVFixedDSPContext *RENAME_FIXED(fdsp);
|
|
|
|
AVFloatDSPContext *fdsp;
|
|
|
|
};
|
|
|
|
int random_state;
|
|
|
|
/** @} */
|
|
|
|
|
|
|
|
/**
|
|
|
|
* @name Members used for output
|
|
|
|
* @{
|
|
|
|
*/
|
|
|
|
SingleChannelElement *output_element[MAX_CHANNELS]; ///< Points to each SingleChannelElement
|
|
|
|
/** @} */
|
|
|
|
|
|
|
|
|
|
|
|
/**
|
|
|
|
* @name Japanese DTV specific extension
|
|
|
|
* @{
|
|
|
|
*/
|
|
|
|
int force_dmono_mode;///< 0->not dmono, 1->use first channel, 2->use second channel
|
|
|
|
int dmono_mode; ///< 0->not dmono, 1->use first channel, 2->use second channel
|
|
|
|
/** @} */
|
|
|
|
|
|
|
|
enum AACOutputChannelOrder output_channel_order;
|
|
|
|
|
|
|
|
OutputConfiguration oc[2];
|
|
|
|
int warned_num_aac_frames;
|
|
|
|
int warned_960_sbr;
|
|
|
|
unsigned warned_71_wide;
|
|
|
|
int warned_gain_control;
|
|
|
|
int warned_he_aac_mono;
|
|
|
|
|
|
|
|
int is_fixed;
|
|
|
|
};
|
|
|
|
|
|
|
|
#if defined(USE_FIXED) && USE_FIXED
|
|
|
|
#define fdsp RENAME_FIXED(fdsp)
|
|
|
|
#endif
|
|
|
|
|
|
|
|
int ff_aac_decode_init(AVCodecContext *avctx);
|
|
|
|
int ff_aac_decode_init_float(AVCodecContext *avctx);
|
|
|
|
int ff_aac_decode_init_fixed(AVCodecContext *avctx);
|
|
|
|
|
|
|
|
int ff_aac_decode_ics(AACDecContext *ac, SingleChannelElement *sce,
|
|
|
|
GetBitContext *gb, int common_window, int scale_flag);
|
|
|
|
|
|
|
|
int ff_aac_decode_tns(AACDecContext *ac, TemporalNoiseShaping *tns,
|
|
|
|
GetBitContext *gb, const IndividualChannelStream *ics);
|
|
|
|
|
|
|
|
int ff_aac_set_default_channel_config(AACDecContext *ac, AVCodecContext *avctx,
|
|
|
|
uint8_t (*layout_map)[3],
|
|
|
|
int *tags,
|
|
|
|
int channel_config);
|
|
|
|
|
|
|
|
int ff_aac_output_configure(AACDecContext *ac,
|
|
|
|
uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
|
|
|
|
enum OCStatus oc_type, int get_new_frame);
|
|
|
|
|
|
|
|
ChannelElement *ff_aac_get_che(AACDecContext *ac, int type, int elem_id);
|
|
|
|
|
|
|
|
#endif /* AVCODEC_AAC_AACDEC_H */
|