mirror of
https://github.com/xenia-project/FFmpeg.git
synced 2024-12-06 10:37:41 +00:00
0665199e43
* qatar/master: vorbisdec: Rename silly "class_" variable to plain "class". simple_idct_alpha: Drop some useless casts. Simplify av_log_missing_feature(). ac3enc: remove check for mismatching channels and channel_layout If AVCodecContext.channels is 0 and AVCodecContext.channel_layout is non-zero, set channels based on channel_layout. If AVCodecContext.channel_layout and AVCodecContext.channels are both non-zero, check to make sure they do not contradict eachother. cosmetics: indentation Check AVCodec.supported_samplerates and AVCodec.channel_layouts in avcodec_open(). aacdec: remove sf_scale and sf_offset. aacdec: use a scale of 2 in the LTP MDCT rather than doubling the coefficient table values from the spec. Define POW_SF2_ZERO in aac.h and use for ff_aac_pow2sf_tabp[] offsets instead of hardcoding 200 everywhere. Large intensity stereo and PNS indices are legal. Clip them instead of erroring out. A magnitude of 100 corresponds to 2^25 so the will most likely result in clipped output anyway. qpeg: use reget_buffer() in decode_frame() ultimotion: use reget_buffer() in ulti_decode_frame() smacker: remove unnecessary call to avctx->release_buffer in decode_frame() avparser: don't av_malloc(0). Merged-by: Michael Niedermayer <michaelni@gmx.at>
304 lines
9.7 KiB
C
304 lines
9.7 KiB
C
/*
|
|
* 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 "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_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
|
|
};
|
|
|
|
/**
|
|
* 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, int16_t, 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;
|
|
|
|
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(32, float, buf_mdct)[1024];
|
|
/** @} */
|
|
|
|
/**
|
|
* @defgroup tables Computed / set up during initialization.
|
|
* @{
|
|
*/
|
|
FFTContext mdct;
|
|
FFTContext mdct_small;
|
|
FFTContext mdct_ltp;
|
|
DSPContext dsp;
|
|
FmtConvertContext fmt_conv;
|
|
int random_state;
|
|
/** @} */
|
|
|
|
/**
|
|
* @defgroup output 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];
|
|
|
|
enum OCStatus output_configured;
|
|
} AACContext;
|
|
|
|
#endif /* AVCODEC_AAC_H */
|