/* * AAC decoder 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 decoder definitions and structures * @author Oded Shimon ( ods15 ods15 dyndns org ) * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) */ #ifndef AVCODEC_AAC_AACDEC_H #define AVCODEC_AAC_AACDEC_H #include #include "libavutil/channel_layout.h" #include "libavutil/float_dsp.h" #include "libavutil/fixed_dsp.h" #include "libavutil/mem_internal.h" #include "libavutil/tx.h" #include "libavcodec/aac.h" #include "libavcodec/avcodec.h" #include "libavcodec/mpeg4audio.h" #include "aacdec_ac.h" typedef struct AACDecContext AACDecContext; /** * 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 }; enum AACOutputChannelOrder { CHANNEL_ORDER_DEFAULT, CHANNEL_ORDER_CODED, }; /** * The point during decoding at which channel coupling is applied. */ enum CouplingPoint { BEFORE_TNS, BETWEEN_TNS_AND_IMDCT, AFTER_IMDCT = 3, }; enum AACUsacElem { ID_USAC_SCE = 0, ID_USAC_CPE = 1, ID_USAC_LFE = 2, ID_USAC_EXT = 3, }; enum ExtensionHeaderType { ID_CONFIG_EXT_FILL = 0, ID_CONFIG_EXT_LOUDNESS_INFO = 2, ID_CONFIG_EXT_STREAM_ID = 7, }; enum AACUsacExtension { ID_EXT_ELE_FILL, ID_EXT_ELE_MPEGS, ID_EXT_ELE_SAOC, ID_EXT_ELE_AUDIOPREROLL, ID_EXT_ELE_UNI_DRC, }; enum AACUSACLoudnessExt { UNIDRCLOUDEXT_TERM = 0x0, UNIDRCLOUDEXT_EQ = 0x1, }; // Supposed to be equal to AAC_RENAME() in case of USE_FIXED. #define RENAME_FIXED(name) name ## _fixed #define INTFLOAT_UNION(name, elems) \ union { \ int RENAME_FIXED(name) elems; \ float name elems; \ } #define INTFLOAT_ALIGNED_UNION(alignment, name, nb_elems) \ union { \ DECLARE_ALIGNED(alignment, int, RENAME_FIXED(name))[nb_elems]; \ DECLARE_ALIGNED(alignment, float, name)[nb_elems]; \ } /** * Long Term Prediction */ typedef struct LongTermPrediction { int8_t present; int16_t lag; INTFLOAT_UNION(coef,); int8_t used[MAX_LTP_LONG_SFB]; } LongTermPrediction; /* Per channel core mode */ typedef struct AACUsacElemData { uint8_t core_mode; uint8_t scale_factor_grouping; /* Timewarping ratio */ #define NUM_TW_NODES 16 uint8_t tw_ratio[NUM_TW_NODES]; struct { uint8_t acelp_core_mode : 3; uint8_t lpd_mode : 5; uint8_t bpf_control_info : 1; uint8_t core_mode_last : 1; uint8_t fac_data_present : 1; int last_lpd_mode; } ldp; struct { unsigned int seed; uint8_t level : 3; uint8_t offset : 5; } noise; struct { uint8_t gain; uint32_t kv[8 /* (1024 / 16) / 8 */][8]; } fac; AACArithState ac; } AACUsacElemData; /** * Individual Channel Stream */ typedef struct IndividualChannelStream { 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 sine 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 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]; uint8_t window_clipping[8]; ///< set if a certain window is near clipping } IndividualChannelStream; /** * Temporal Noise Shaping */ typedef struct TemporalNoiseShaping { int present; int n_filt[8]; int length[8][4]; int direction[8][4]; int order[8][4]; INTFLOAT_UNION(coef, [8][4][TNS_MAX_ORDER]); } TemporalNoiseShaping; /** * coupling parameters */ typedef struct ChannelCoupling { 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 */ INTFLOAT_UNION(gain, [16][120]); } ChannelCoupling; /** * Single Channel Element - used for both SCE and LFE elements. */ typedef struct SingleChannelElement { IndividualChannelStream ics; AACUsacElemData ue; ///< USAC element data TemporalNoiseShaping tns; enum BandType band_type[128]; ///< band types int sfo[128]; ///< scalefactor offsets INTFLOAT_UNION(sf, [128]); ///< scalefactors (8 windows * 16 sfb max) INTFLOAT_ALIGNED_UNION(32, coeffs, 1024); ///< coefficients for IMDCT, maybe processed INTFLOAT_ALIGNED_UNION(32, prev_coeffs, 1024); ///< unscaled previous contents of coeffs[] for USAC INTFLOAT_ALIGNED_UNION(32, saved, 1536); ///< overlap INTFLOAT_ALIGNED_UNION(32, ret_buf, 2048); ///< PCM output buffer INTFLOAT_ALIGNED_UNION(16, ltp_state, 3072); ///< time signal for LTP union { struct PredictorStateFixed *RENAME_FIXED(predictor_state); struct PredictorState *predictor_state; }; union { float *output; ///< PCM output int *RENAME_FIXED(output); ///< PCM output }; } SingleChannelElement; typedef struct AACUsacStereo { uint8_t common_window; uint8_t common_tw; uint8_t ms_mask_mode; uint8_t config_idx; /* Complex prediction */ uint8_t use_prev_frame; uint8_t pred_dir; uint8_t complex_coef; uint8_t pred_used[128]; INTFLOAT_ALIGNED_UNION(32, alpha_q_re, 1024); INTFLOAT_ALIGNED_UNION(32, alpha_q_im, 1024); INTFLOAT_ALIGNED_UNION(32, prev_alpha_q_re, 1024); INTFLOAT_ALIGNED_UNION(32, prev_alpha_q_im, 1024); INTFLOAT_ALIGNED_UNION(32, dmix_re, 1024); INTFLOAT_ALIGNED_UNION(32, prev_dmix_re, 1024); /* Recalculated on every frame */ INTFLOAT_ALIGNED_UNION(32, dmix_im, 1024); /* Final prediction data */ } AACUsacStereo; /** * channel element - generic struct for SCE/CPE/CCE/LFE */ typedef struct ChannelElement { int present; // CPE specific uint8_t max_sfb_ste; ///< (USAC) Maximum of both max_sfb values 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; // USAC stereo coupling data AACUsacStereo us; } ChannelElement; typedef struct AACUSACLoudnessInfo { uint8_t drc_set_id : 6; uint8_t downmix_id : 7; struct { uint16_t lvl : 12; uint8_t present : 1; } sample_peak; struct { uint16_t lvl : 12; uint8_t measurement : 4; uint8_t reliability : 2; uint8_t present : 1; } true_peak; uint8_t nb_measurements : 4; struct { uint8_t method_def : 4; uint8_t method_val; uint8_t measurement : 4; uint8_t reliability : 2; } measurements[16]; } AACUSACLoudnessInfo; typedef struct AACUsacElemConfig { enum AACUsacElem type; uint8_t tw_mdct : 1; uint8_t noise_fill : 1; uint8_t stereo_config_index; struct { int ratio; uint8_t harmonic_sbr : 1; /* harmonicSBR */ uint8_t bs_intertes : 1; /* bs_interTes */ uint8_t bs_pvc : 1; /* bs_pvc */ struct { uint8_t start_freq; /* dflt_start_freq */ uint8_t stop_freq; /* dflt_stop_freq */ uint8_t freq_scale; /* dflt_freq_scale */ uint8_t alter_scale : 1; /* dflt_alter_scale */ uint8_t noise_scale; /* dflt_noise_scale */ uint8_t limiter_bands; /* dflt_limiter_bands */ uint8_t limiter_gains; /* dflt_limiter_gains */ uint8_t interpol_freq : 1; /* dflt_interpol_freq */ uint8_t smoothing_mode : 1; /* dflt_smoothing_mode */ } dflt; } sbr; struct { uint8_t freq_res; /* bsFreqRes */ uint8_t fixed_gain; /* bsFixedGainDMX */ uint8_t temp_shape_config; /* bsTempShapeConfig */ uint8_t decorr_config; /* bsDecorrConfig */ uint8_t high_rate_mode : 1; /* bsHighRateMode */ uint8_t phase_coding : 1; /* bsPhaseCoding */ uint8_t otts_bands_phase; /* bsOttBandsPhase */ uint8_t residual_coding; /* bsResidualCoding */ uint8_t residual_bands; /* bsResidualBands */ uint8_t pseudo_lr : 1; /* bsPseudoLr */ uint8_t env_quant_mode : 1; /* bsEnvQuantMode */ } mps; struct { enum AACUsacExtension type; uint8_t payload_frag; uint32_t default_len; uint32_t pl_data_offset; uint8_t *pl_data; } ext; } AACUsacElemConfig; typedef struct AACUSACConfig { uint8_t core_sbr_frame_len_idx; /* coreSbrFrameLengthIndex */ uint8_t rate_idx; uint16_t core_frame_len; uint16_t stream_identifier; AACUsacElemConfig elems[64]; int nb_elems; struct { uint8_t nb_album; AACUSACLoudnessInfo album_info[64]; uint8_t nb_info; AACUSACLoudnessInfo info[64]; } loudness; } AACUSACConfig; typedef struct OutputConfiguration { MPEG4AudioConfig m4ac; uint8_t layout_map[MAX_ELEM_ID*4][3]; int layout_map_tags; AVChannelLayout ch_layout; enum OCStatus status; AACUSACConfig usac; } OutputConfiguration; /** * Dynamic Range Control - decoded from the bitstream but not processed further. */ typedef struct DynamicRangeControl { 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; /** * Decode-specific primitives */ typedef struct AACDecProc { int (*decode_spectrum_and_dequant)(AACDecContext *ac, GetBitContext *gb, 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 */