/* * 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 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 "libavutil/float_dsp.h" #include "avcodec.h" #include "dsputil.h" #include "fft.h" #include "mpeg4audio.h" #include "sbr.h" #include "fmtconvert.h" #include <stdint.h> #define MAX_CHANNELS 64 #define MAX_ELEM_ID 16 #define TNS_MAX_ORDER 20 #define MAX_LTP_LONG_SFB 40 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_OFF = 0, 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, }; /** * Output configuration status */ enum OCStatus { OC_NONE, ///< Output unconfigured OC_TRIAL_PCE, ///< Output configuration under trial specified by an inband PCE OC_TRIAL_FRAME, ///< Output configuration under trial specified by a frame header OC_GLOBAL_HDR, ///< Output configuration set in a global header but not yet locked OC_LOCKED, ///< Output configuration locked in place }; typedef struct { MPEG4AudioConfig m4ac; uint8_t layout_map[MAX_ELEM_ID*4][3]; int layout_map_tags; int channels; uint64_t channel_layout; enum OCStatus status; } OutputConfiguration; /** * Predictor State */ typedef struct { float cor0; float cor1; float var0; float var1; float r0; float r1; } PredictorState; #define MAX_PREDICTORS 672 #define SCALE_DIV_512 36 ///< scalefactor difference that corresponds to scale difference in 512 times #define SCALE_ONE_POS 140 ///< scalefactor index that corresponds to scale=1.0 #define SCALE_MAX_POS 255 ///< scalefactor index maximum value #define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard #define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference #define POW_SF2_ZERO 200 ///< ff_aac_pow2sf_tab index corresponding to pow(2, 0); /** * Long Term Prediction */ typedef struct { int8_t present; int16_t lag; float coef; int8_t used[MAX_LTP_LONG_SFB]; } LongTermPrediction; /** * 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]; LongTermPrediction ltp; const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window const uint8_t *swb_sizes; ///< table of scalefactor band sizes 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 start; 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; Pulse pulse; enum BandType band_type[128]; ///< band types int band_type_run_end[120]; ///< band type run end points float sf[120]; ///< scalefactors int sf_idx[128]; ///< scalefactor indices (used by encoder) uint8_t zeroes[128]; ///< band is not coded (used by encoder) DECLARE_ALIGNED(32, float, coeffs)[1024]; ///< coefficients for IMDCT DECLARE_ALIGNED(32, float, saved)[1024]; ///< overlap DECLARE_ALIGNED(32, float, ret)[2048]; ///< PCM output DECLARE_ALIGNED(16, float, ltp_state)[3072]; ///< time signal for LTP PredictorState predictor_state[MAX_PREDICTORS]; } SingleChannelElement; /** * channel element - generic struct for SCE/CPE/CCE/LFE */ typedef struct { // CPE specific int common_window; ///< Set if channels share a common 'IndividualChannelStream' in bitstream. int ms_mode; ///< Signals mid/side stereo flags coding mode (used by encoder) uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band // shared SingleChannelElement ch[2]; // CCE specific ChannelCoupling coup; SpectralBandReplication sbr; } ChannelElement; /** * main AAC context */ typedef struct { AVCodecContext *avctx; 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; /** @} */ /** * @name temporary aligned temporary buffers * (We do not want to have these on the stack.) * @{ */ DECLARE_ALIGNED(32, float, buf_mdct)[1024]; /** @} */ /** * @name Computed / set up during initialization * @{ */ FFTContext mdct; FFTContext mdct_small; FFTContext mdct_ltp; DSPContext dsp; FmtConvertContext fmt_conv; AVFloatDSPContext fdsp; int random_state; /** @} */ /** * @name Members used for output interleaving * @{ */ float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output). /** @} */ DECLARE_ALIGNED(32, float, temp)[128]; OutputConfiguration oc[2]; int warned_num_aac_frames; } AACContext; #endif /* AVCODEC_AAC_H */