* qatar/master: fate: Add tests for more AAC features. aacps: Add missing newline in error message. fate: Add tests for vc1/wmapro in ism. aacdec: Add a fate test for 5.1 channel SBR. aacdec: Turn off PS for multichannel files that use PCE based configs. cabac: remove put_cabac_u/ueg from cabac-test. swscale: RGB4444 and BGR444 input FATE: add test for xWMA demuxer. FATE: add test for SMJPEG demuxer and associated IMA ADPCM audio decoder. mpegaudiodec: optimized iMDCT transform mpegaudiodec: change imdct window arrangment for better pointer alignment mpegaudiodec: move imdct and windowing function to mpegaudiodsp mpegaudiodec: interleave iMDCT buffer to simplify future SIMD implementations swscale: convert yuy2/uyvy/nv12/nv21ToY/UV from inline asm to yasm. FATE: test to exercise WTV demuxer. mjpegdec: K&R formatting cosmetics swscale: K&R formatting cosmetics for code examples swscale: K&R reformatting cosmetics for header files FATE test: cvid-grayscale; ensures that the grayscale Cinepak variant is exercised. Conflicts: libavcodec/cabac.c libavcodec/mjpegdec.c libavcodec/mpegaudiodec.c libavcodec/mpegaudiodsp.c libavcodec/mpegaudiodsp.h libavcodec/mpegaudiodsp_template.c libavcodec/x86/Makefile libavcodec/x86/imdct36_sse.asm libavcodec/x86/mpegaudiodec_mmx.c libswscale/swscale-test.c libswscale/swscale.c libswscale/swscale_internal.h libswscale/x86/swscale_template.c tests/fate/demux.mak tests/fate/microsoft.mak tests/fate/video.mak tests/fate/wma.mak tests/ref/lavfi/pixfmts_scale Merged-by: Michael Niedermayer <michaelni@gmx.at>
		
			
				
	
	
		
			308 lines
		
	
	
		
			9.8 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			308 lines
		
	
	
		
			9.8 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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| /**
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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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| 
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| #ifndef AVCODEC_AAC_H
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| #define AVCODEC_AAC_H
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| 
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| #include "avcodec.h"
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| #include "dsputil.h"
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| #include "fft.h"
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| #include "mpeg4audio.h"
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| #include "sbr.h"
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| #include "fmtconvert.h"
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| 
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| #include <stdint.h>
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| 
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| #define MAX_CHANNELS 64
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| #define MAX_ELEM_ID 16
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| 
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| #define TNS_MAX_ORDER 20
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| #define MAX_LTP_LONG_SFB 40
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| 
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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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| 
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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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| 
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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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| 
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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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|     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.
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|     INTENSITY_BT   = 15,    ///< Scalefactor data are intensity stereo positions.
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| };
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| 
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| #define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)
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| 
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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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| /**
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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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| /**
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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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| 
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| /**
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|  * Predictor State
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|  */
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| typedef struct {
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|     float cor0;
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|     float cor1;
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|     float var0;
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|     float var1;
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|     float r0;
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|     float r1;
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| } PredictorState;
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| 
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| #define MAX_PREDICTORS 672
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| 
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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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| 
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| /**
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|  * Long Term Prediction
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|  */
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| typedef struct {
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|     int8_t present;
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|     int16_t lag;
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|     float 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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| /**
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|  * Individual Channel Stream
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|  */
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| typedef struct {
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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 sinus 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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|     uint8_t prediction_used[41];
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| } IndividualChannelStream;
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| 
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| /**
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|  * Temporal Noise Shaping
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|  */
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| typedef struct {
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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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|     float coef[8][4][TNS_MAX_ORDER];
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| } TemporalNoiseShaping;
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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 {
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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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| typedef struct {
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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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| /**
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|  * coupling parameters
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|  */
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| typedef struct {
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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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|     float gain[16][120];
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| } ChannelCoupling;
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| 
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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 {
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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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|     int band_type_run_end[120];                     ///< band type run end points
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|     float 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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|     DECLARE_ALIGNED(32, float,   coeffs)[1024];     ///< coefficients for IMDCT
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|     DECLARE_ALIGNED(32, float,   saved)[1024];      ///< overlap
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|     DECLARE_ALIGNED(32, float,   ret)[2048];        ///< PCM output
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|     DECLARE_ALIGNED(16, float,   ltp_state)[3072];  ///< time signal for LTP
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|     PredictorState predictor_state[MAX_PREDICTORS];
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| } SingleChannelElement;
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| 
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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 {
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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 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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|     SpectralBandReplication sbr;
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| } ChannelElement;
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| 
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| /**
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|  * main AAC context
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|  */
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| typedef struct {
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|     AVCodecContext *avctx;
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|     AVFrame frame;
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| 
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|     MPEG4AudioConfig m4ac;
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| 
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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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|     /**
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|      * @name Channel element related data
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|      * @{
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|      */
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|     enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
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|                                                    *   first index as the first 4 raw data block types
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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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|     /** @} */
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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, float, buf_mdct)[1024];
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|     /** @} */
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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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|     FFTContext mdct;
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|     FFTContext mdct_small;
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|     FFTContext mdct_ltp;
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|     DSPContext dsp;
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|     FmtConvertContext fmt_conv;
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|     int random_state;
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|     /** @} */
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| 
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|     /**
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|      * @name Members used for output interleaving
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|      * @{
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|      */
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|     float *output_data[MAX_CHANNELS];                 ///< Points to each element's 'ret' buffer (PCM output).
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|     /** @} */
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| 
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|     DECLARE_ALIGNED(32, float, temp)[128];
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| 
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|     enum OCStatus output_configured;
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|     int warned_num_aac_frames;
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| } AACContext;
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| 
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| #endif /* AVCODEC_AAC_H */
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