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210 lines
7.9 KiB
210 lines
7.9 KiB
/* |
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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 aac.h |
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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 FFMPEG_AAC_H |
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#define FFMPEG_AAC_H |
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#include "avcodec.h" |
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#include "dsputil.h" |
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#include "mpeg4audio.h" |
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#include <stdint.h> |
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#define AAC_INIT_VLC_STATIC(num, size) \ |
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INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \ |
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ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \ |
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ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \ |
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size); |
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#define MAX_CHANNELS 64 |
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#define MAX_ELEM_ID 16 |
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#define IVQUANT_SIZE 1024 |
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enum AudioObjectType { |
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AOT_NULL, |
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// Support? Name |
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AOT_AAC_MAIN, ///< Y Main |
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AOT_AAC_LC, ///< Y Low Complexity |
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AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate |
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AOT_AAC_LTP, ///< N (code in SoC repo) Long Term Prediction |
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AOT_SBR, ///< N (in progress) Spectral Band Replication |
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AOT_AAC_SCALABLE, ///< N Scalable |
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AOT_TWINVQ, ///< N Twin Vector Quantizer |
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AOT_CELP, ///< N Code Excited Linear Prediction |
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AOT_HVXC, ///< N Harmonic Vector eXcitation Coding |
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AOT_TTSI = 12, ///< N Text-To-Speech Interface |
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AOT_MAINSYNTH, ///< N Main Synthesis |
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AOT_WAVESYNTH, ///< N Wavetable Synthesis |
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AOT_MIDI, ///< N General MIDI |
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AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects |
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AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity |
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AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction |
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AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable |
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AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer |
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AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding |
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AOT_ER_AAC_LD, ///< N Error Resilient Low Delay |
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AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction |
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AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding |
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AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise |
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AOT_ER_PARAM, ///< N Error Resilient Parametric |
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AOT_SSC, ///< N SinuSoidal Coding |
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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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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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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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#define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10) |
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enum ChannelPosition { |
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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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* Individual Channel Stream |
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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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typedef struct { |
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int num_pulse; |
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int start; |
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int offset[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 { |
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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 * avccontext; |
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MPEG4AudioConfig m4ac; |
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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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* @defgroup elements |
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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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/** @} */ |
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/** |
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* @defgroup tables Computed / set up during initialization. |
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* @{ |
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*/ |
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MDCTContext mdct; |
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MDCTContext mdct_small; |
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DSPContext dsp; |
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int random_state; |
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/** @} */ |
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/** |
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* @defgroup output 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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float add_bias; ///< offset for dsp.float_to_int16 |
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float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16. |
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int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16 |
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/** @} */ |
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} AACContext; |
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#endif /* FFMPEG_AAC_H */
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