Mercurial > libavcodec.hg
view aac.h @ 10345:294c444866f7 libavcodec
Make avcodec_open set codec_id and codec_type if they haven't been set.
This fixes the API breakage introduced by the check that avctx codec id and type
match the opened codec and should make (almost?) all applications work again.
author | reimar |
---|---|
date | Thu, 01 Oct 2009 15:54:55 +0000 |
parents | 38ab367d4231 |
children | 74e51b86b934 |
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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 libavcodec/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 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 #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 /** * 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 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_16(float, coeffs[1024]); ///< coefficients for IMDCT DECLARE_ALIGNED_16(float, saved[1024]); ///< 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 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; } 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 Channel element related data. * @{ */ 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]; ChannelElement * tag_che_map[4][MAX_ELEM_ID]; int tags_mapped; /** @} */ /** * @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. * @{ */ FFTContext mdct; FFTContext 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]); int output_configured; } AACContext; #endif /* AVCODEC_AAC_H */