Mercurial > libavcodec.hg
view aac.h @ 8650:035c970a12ac libavcodec
flacdec: remove unneeded debugging statements
author | jbr |
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date | Sat, 24 Jan 2009 16:01:20 +0000 |
parents | 6f40f7d77263 |
children | e9d9d946f213 |
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/* * AAC definitions and structures * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file aac.h * AAC definitions and structures * @author Oded Shimon ( ods15 ods15 dyndns org ) * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) */ #ifndef AVCODEC_AAC_H #define AVCODEC_AAC_H #include "avcodec.h" #include "dsputil.h" #include "mpeg4audio.h" #include <stdint.h> #define AAC_INIT_VLC_STATIC(num, size) \ INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \ ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \ ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \ size); #define MAX_CHANNELS 64 #define MAX_ELEM_ID 16 #define TNS_MAX_ORDER 20 enum AudioObjectType { AOT_NULL, // Support? Name AOT_AAC_MAIN, ///< Y Main AOT_AAC_LC, ///< Y Low Complexity AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate AOT_AAC_LTP, ///< N (code in SoC repo) Long Term Prediction AOT_SBR, ///< N (in progress) Spectral Band Replication AOT_AAC_SCALABLE, ///< N Scalable AOT_TWINVQ, ///< N Twin Vector Quantizer AOT_CELP, ///< N Code Excited Linear Prediction AOT_HVXC, ///< N Harmonic Vector eXcitation Coding AOT_TTSI = 12, ///< N Text-To-Speech Interface AOT_MAINSYNTH, ///< N Main Synthesis AOT_WAVESYNTH, ///< N Wavetable Synthesis AOT_MIDI, ///< N General MIDI AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding AOT_ER_AAC_LD, ///< N Error Resilient Low Delay AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise AOT_ER_PARAM, ///< N Error Resilient Parametric AOT_SSC, ///< N SinuSoidal Coding }; enum RawDataBlockType { TYPE_SCE, TYPE_CPE, TYPE_CCE, TYPE_LFE, TYPE_DSE, TYPE_PCE, TYPE_FIL, TYPE_END, }; enum ExtensionPayloadID { EXT_FILL, EXT_FILL_DATA, EXT_DATA_ELEMENT, EXT_DYNAMIC_RANGE = 0xb, EXT_SBR_DATA = 0xd, EXT_SBR_DATA_CRC = 0xe, }; enum WindowSequence { ONLY_LONG_SEQUENCE, LONG_START_SEQUENCE, EIGHT_SHORT_SEQUENCE, LONG_STOP_SEQUENCE, }; enum BandType { ZERO_BT = 0, ///< Scalefactors and spectral data are all zero. FIRST_PAIR_BT = 5, ///< This and later band types encode two values (rather than four) with one code word. ESC_BT = 11, ///< Spectral data are coded with an escape sequence. NOISE_BT = 13, ///< Spectral data are scaled white noise not coded in the bitstream. INTENSITY_BT2 = 14, ///< Scalefactor data are intensity stereo positions. INTENSITY_BT = 15, ///< Scalefactor data are intensity stereo positions. }; #define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10) enum ChannelPosition { AAC_CHANNEL_FRONT = 1, AAC_CHANNEL_SIDE = 2, AAC_CHANNEL_BACK = 3, AAC_CHANNEL_LFE = 4, AAC_CHANNEL_CC = 5, }; /** * The point during decoding at which channel coupling is applied. */ enum CouplingPoint { BEFORE_TNS, BETWEEN_TNS_AND_IMDCT, AFTER_IMDCT = 3, }; /** * Predictor State */ typedef struct { float cor0; float cor1; float var0; float var1; float r0; float r1; } PredictorState; #define MAX_PREDICTORS 672 /** * Individual Channel Stream */ typedef struct { uint8_t max_sfb; ///< number of scalefactor bands per group enum WindowSequence window_sequence[2]; uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sinus window. int num_window_groups; uint8_t group_len[8]; const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window int num_swb; ///< number of scalefactor window bands int num_windows; int tns_max_bands; int predictor_present; int predictor_initialized; int predictor_reset_group; uint8_t prediction_used[41]; } IndividualChannelStream; /** * Temporal Noise Shaping */ typedef struct { int present; int n_filt[8]; int length[8][4]; int direction[8][4]; int order[8][4]; float coef[8][4][TNS_MAX_ORDER]; } TemporalNoiseShaping; /** * Dynamic Range Control - decoded from the bitstream but not processed further. */ typedef struct { int pce_instance_tag; ///< Indicates with which program the DRC info is associated. int dyn_rng_sgn[17]; ///< DRC sign information; 0 - positive, 1 - negative int dyn_rng_ctl[17]; ///< DRC magnitude information int exclude_mask[MAX_CHANNELS]; ///< Channels to be excluded from DRC processing. int band_incr; ///< Number of DRC bands greater than 1 having DRC info. int interpolation_scheme; ///< Indicates the interpolation scheme used in the SBR QMF domain. int band_top[17]; ///< Indicates the top of the i-th DRC band in units of 4 spectral lines. int prog_ref_level; /**< A reference level for the long-term program audio level for all * channels combined. */ } DynamicRangeControl; typedef struct { int num_pulse; int pos[4]; int amp[4]; } Pulse; /** * coupling parameters */ typedef struct { enum CouplingPoint coupling_point; ///< The point during decoding at which coupling is applied. int num_coupled; ///< number of target elements enum RawDataBlockType type[8]; ///< Type of channel element to be coupled - SCE or CPE. int id_select[8]; ///< element id int ch_select[8]; /**< [0] shared list of gains; [1] list of gains for right channel; * [2] list of gains for left channel; [3] lists of gains for both channels */ float gain[16][120]; } ChannelCoupling; /** * Single Channel Element - used for both SCE and LFE elements. */ typedef struct { IndividualChannelStream ics; TemporalNoiseShaping tns; enum BandType band_type[120]; ///< band types int band_type_run_end[120]; ///< band type run end points float sf[120]; ///< scalefactors DECLARE_ALIGNED_16(float, coeffs[1024]); ///< coefficients for IMDCT DECLARE_ALIGNED_16(float, saved[512]); ///< overlap DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output PredictorState predictor_state[MAX_PREDICTORS]; } SingleChannelElement; /** * channel element - generic struct for SCE/CPE/CCE/LFE */ typedef struct { // CPE specific uint8_t ms_mask[120]; ///< Set if mid/side stereo is used for each scalefactor window band // shared SingleChannelElement ch[2]; // CCE specific ChannelCoupling coup; } ChannelElement; /** * main AAC context */ typedef struct { AVCodecContext * avccontext; MPEG4AudioConfig m4ac; int is_saved; ///< Set if elements have stored overlap from previous frame. DynamicRangeControl che_drc; /** * @defgroup elements * @{ */ enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the * first index as the first 4 raw data block types */ ChannelElement * che[4][MAX_ELEM_ID]; /** @} */ /** * @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.) * @{ */ DECLARE_ALIGNED_16(float, buf_mdct[1024]); /** @} */ /** * @defgroup tables Computed / set up during initialization. * @{ */ MDCTContext mdct; MDCTContext mdct_small; DSPContext dsp; int random_state; /** @} */ /** * @defgroup output Members used for output interleaving. * @{ */ float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output). float add_bias; ///< offset for dsp.float_to_int16 float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16. int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16 /** @} */ DECLARE_ALIGNED(16, float, temp[128]); } AACContext; #endif /* AVCODEC_AAC_H */