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/*
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* AAC definitions and structures
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* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
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* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* AAC definitions and structures
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* @author Oded Shimon ( ods15 ods15 dyndns org )
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* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
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*/
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#ifndef AVCODEC_AAC_H
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#define AVCODEC_AAC_H
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#include "aac_defines.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/fixed_dsp.h"
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#include "avcodec.h"
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#if !USE_FIXED
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#include "mdct15.h"
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#endif
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#include "fft.h"
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#include "mpeg4audio.h"
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#include "sbr.h"
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#include <stdint.h>
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#define MAX_CHANNELS 64
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#define MAX_ELEM_ID 16
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#define TNS_MAX_ORDER 20
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#define MAX_LTP_LONG_SFB 40
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#define CLIP_AVOIDANCE_FACTOR 0.95f
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enum RawDataBlockType {
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TYPE_SCE,
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TYPE_CPE,
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TYPE_CCE,
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TYPE_LFE,
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TYPE_DSE,
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TYPE_PCE,
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TYPE_FIL,
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TYPE_END,
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};
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enum ExtensionPayloadID {
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EXT_FILL,
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EXT_FILL_DATA,
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EXT_DATA_ELEMENT,
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EXT_DYNAMIC_RANGE = 0xb,
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EXT_SBR_DATA = 0xd,
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EXT_SBR_DATA_CRC = 0xe,
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};
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enum WindowSequence {
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ONLY_LONG_SEQUENCE,
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LONG_START_SEQUENCE,
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EIGHT_SHORT_SEQUENCE,
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LONG_STOP_SEQUENCE,
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};
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enum BandType {
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ZERO_BT = 0, ///< Scalefactors and spectral data are all zero.
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FIRST_PAIR_BT = 5, ///< This and later band types encode two values (rather than four) with one code word.
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ESC_BT = 11, ///< Spectral data are coded with an escape sequence.
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RESERVED_BT = 12, ///< Band types following are encoded differently from others.
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NOISE_BT = 13, ///< Spectral data are scaled white noise not coded in the bitstream.
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INTENSITY_BT2 = 14, ///< Scalefactor data are intensity stereo positions (out of phase).
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INTENSITY_BT = 15, ///< Scalefactor data are intensity stereo positions (in phase).
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};
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#define IS_CODEBOOK_UNSIGNED(x) (((x) - 1) & 10)
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enum ChannelPosition {
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AAC_CHANNEL_OFF = 0,
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AAC_CHANNEL_FRONT = 1,
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AAC_CHANNEL_SIDE = 2,
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AAC_CHANNEL_BACK = 3,
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AAC_CHANNEL_LFE = 4,
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AAC_CHANNEL_CC = 5,
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};
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/**
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* The point during decoding at which channel coupling is applied.
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*/
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enum CouplingPoint {
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BEFORE_TNS,
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BETWEEN_TNS_AND_IMDCT,
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AFTER_IMDCT = 3,
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};
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/**
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* Output configuration status
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*/
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enum OCStatus {
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OC_NONE, ///< Output unconfigured
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OC_TRIAL_PCE, ///< Output configuration under trial specified by an inband PCE
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OC_TRIAL_FRAME, ///< Output configuration under trial specified by a frame header
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OC_GLOBAL_HDR, ///< Output configuration set in a global header but not yet locked
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OC_LOCKED, ///< Output configuration locked in place
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};
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typedef struct OutputConfiguration {
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MPEG4AudioConfig m4ac;
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uint8_t layout_map[MAX_ELEM_ID*4][3];
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int layout_map_tags;
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int channels;
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uint64_t channel_layout;
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enum OCStatus status;
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} OutputConfiguration;
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/**
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* Predictor State
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*/
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typedef struct PredictorState {
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AAC_FLOAT cor0;
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AAC_FLOAT cor1;
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AAC_FLOAT var0;
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AAC_FLOAT var1;
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AAC_FLOAT r0;
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AAC_FLOAT r1;
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AAC_FLOAT k1;
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AAC_FLOAT x_est;
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} PredictorState;
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#define MAX_PREDICTORS 672
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#define SCALE_DIV_512 36 ///< scalefactor difference that corresponds to scale difference in 512 times
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#define SCALE_ONE_POS 140 ///< scalefactor index that corresponds to scale=1.0
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#define SCALE_MAX_POS 255 ///< scalefactor index maximum value
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#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard
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#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference
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#define POW_SF2_ZERO 200 ///< ff_aac_pow2sf_tab index corresponding to pow(2, 0);
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#define NOISE_PRE 256 ///< preamble for NOISE_BT, put in bitstream with the first noise band
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#define NOISE_PRE_BITS 9 ///< length of preamble
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#define NOISE_OFFSET 90 ///< subtracted from global gain, used as offset for the preamble
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/**
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* Long Term Prediction
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*/
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typedef struct LongTermPrediction {
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int8_t present;
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int16_t lag;
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int coef_idx;
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INTFLOAT coef;
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int8_t used[MAX_LTP_LONG_SFB];
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} LongTermPrediction;
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/**
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* Individual Channel Stream
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*/
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typedef struct IndividualChannelStream {
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uint8_t max_sfb; ///< number of scalefactor bands per group
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enum WindowSequence window_sequence[2];
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uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sine window.
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int num_window_groups;
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uint8_t group_len[8];
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LongTermPrediction ltp;
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const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
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const uint8_t *swb_sizes; ///< table of scalefactor band sizes for a particular window
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int num_swb; ///< number of scalefactor window bands
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int num_windows;
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int tns_max_bands;
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int predictor_present;
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int predictor_initialized;
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int predictor_reset_group;
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int predictor_reset_count[31]; ///< used by encoder to count prediction resets
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uint8_t prediction_used[41];
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uint8_t window_clipping[8]; ///< set if a certain window is near clipping
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float clip_avoidance_factor; ///< set if any window is near clipping to the necessary atennuation factor to avoid it
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} IndividualChannelStream;
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/**
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* Temporal Noise Shaping
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*/
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typedef struct TemporalNoiseShaping {
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int present;
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int n_filt[8];
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int length[8][4];
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int direction[8][4];
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int order[8][4];
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int coef_idx[8][4][TNS_MAX_ORDER];
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INTFLOAT coef[8][4][TNS_MAX_ORDER];
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} TemporalNoiseShaping;
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/**
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* Dynamic Range Control - decoded from the bitstream but not processed further.
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*/
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typedef struct DynamicRangeControl {
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int pce_instance_tag; ///< Indicates with which program the DRC info is associated.
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int dyn_rng_sgn[17]; ///< DRC sign information; 0 - positive, 1 - negative
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int dyn_rng_ctl[17]; ///< DRC magnitude information
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int exclude_mask[MAX_CHANNELS]; ///< Channels to be excluded from DRC processing.
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int band_incr; ///< Number of DRC bands greater than 1 having DRC info.
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int interpolation_scheme; ///< Indicates the interpolation scheme used in the SBR QMF domain.
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int band_top[17]; ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.
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int prog_ref_level; /**< A reference level for the long-term program audio level for all
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* channels combined.
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*/
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} DynamicRangeControl;
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typedef struct Pulse {
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int num_pulse;
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int start;
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int pos[4];
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int amp[4];
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} Pulse;
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/**
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* coupling parameters
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*/
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typedef struct ChannelCoupling {
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enum CouplingPoint coupling_point; ///< The point during decoding at which coupling is applied.
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int num_coupled; ///< number of target elements
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enum RawDataBlockType type[8]; ///< Type of channel element to be coupled - SCE or CPE.
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int id_select[8]; ///< element id
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int ch_select[8]; /**< [0] shared list of gains; [1] list of gains for right channel;
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* [2] list of gains for left channel; [3] lists of gains for both channels
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*/
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INTFLOAT gain[16][120];
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} ChannelCoupling;
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/**
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* Single Channel Element - used for both SCE and LFE elements.
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*/
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typedef struct SingleChannelElement {
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IndividualChannelStream ics;
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TemporalNoiseShaping tns;
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Pulse pulse;
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enum BandType band_type[128]; ///< band types
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enum BandType band_alt[128]; ///< alternative band type (used by encoder)
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int band_type_run_end[120]; ///< band type run end points
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INTFLOAT sf[120]; ///< scalefactors
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int sf_idx[128]; ///< scalefactor indices (used by encoder)
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uint8_t zeroes[128]; ///< band is not coded (used by encoder)
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AAC encoder: Extensive improvements
This finalizes merging of the work in the patches in ticket #2686.
Improvements to twoloop and RC logic are extensive.
The non-exhaustive list of twoloop improvments includes:
- Tweaks to distortion limits on the RD optimization phase of twoloop
- Deeper search in twoloop
- PNS information marking to let twoloop decide when to use it
(turned out having the decision made separately wasn't working)
- Tonal band detection and priorization
- Better band energy conservation rules
- Strict hole avoidance
For rate control:
- Use psymodel's bit allocation to allow proper use of the bit
reservoir. Don't work against the bit reservoir by moving lambda
in the opposite direction when psymodel decides to allocate more/less
bits to a frame.
- Retry the encode if the effective rate lies outside a reasonable
margin of psymodel's allocation or the selected ABR.
- Log average lambda at the end. Useful info for everyone, but especially
for tuning of the various encoder constants that relate to lambda
feedback.
Psy:
- Do not apply lowpass with a FIR filter, instead just let the coder
zero bands above the cutoff. The FIR filter induces group delay,
and while zeroing bands causes ripple, it's lost in the quantization
noise.
- Experimental VBR bit allocation code
- Tweak automatic lowpass filter threshold to maximize audio bandwidth
at all bitrates while still providing acceptable, stable quality.
I/S:
- Phase decision fixes. Unrelated to #2686, but the bugs only surfaced
when the merge was finalized. Measure I/S band energy accounting for
phase, and prevent I/S and M/S from being applied both.
PNS:
- Avoid marking short bands with PNS when they're part of a window
group in which there's a large variation of energy from one window
to the next. PNS can't preserve those and the effect is extremely
noticeable.
M/S:
- Implement BMLD protection similar to the specified in
ISO-IEC/13818:7-2003, Appendix C Section 6.1. Since M/S decision
doesn't conform to section 6.1, a different method had to be
implemented, but should provide equivalent protection.
- Move the decision logic closer to the method specified in
ISO-IEC/13818:7-2003, Appendix C Section 6.1. Specifically,
make sure M/S needs less bits than dual stereo.
- Don't apply M/S in bands that are using I/S
Now, this of course needed adjustments in the compare targets and
fuzz factors of the AAC encoder's fate tests, but if wondering why
the targets go up (more distortion), consider the previous coder
was using too many bits on LF content (far more than required by
psy), and thus those signals will now be more distorted, not less.
The extra distortion isn't audible though, I carried extensive
ABX testing to make sure.
A very similar patch was also extensively tested by Kamendo2 in
the context of #2686.
9 years ago
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uint8_t can_pns[128]; ///< band is allowed to PNS (informative)
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float is_ener[128]; ///< Intensity stereo pos (used by encoder)
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float pns_ener[128]; ///< Noise energy values (used by encoder)
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DECLARE_ALIGNED(32, INTFLOAT, pcoeffs)[1024]; ///< coefficients for IMDCT, pristine
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DECLARE_ALIGNED(32, INTFLOAT, coeffs)[1024]; ///< coefficients for IMDCT, maybe processed
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DECLARE_ALIGNED(32, INTFLOAT, saved)[1536]; ///< overlap
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DECLARE_ALIGNED(32, INTFLOAT, ret_buf)[2048]; ///< PCM output buffer
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DECLARE_ALIGNED(16, INTFLOAT, ltp_state)[3072]; ///< time signal for LTP
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DECLARE_ALIGNED(32, AAC_FLOAT, lcoeffs)[1024]; ///< MDCT of LTP coefficients (used by encoder)
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DECLARE_ALIGNED(32, AAC_FLOAT, prcoeffs)[1024]; ///< Main prediction coefs (used by encoder)
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PredictorState predictor_state[MAX_PREDICTORS];
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INTFLOAT *ret; ///< PCM output
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} SingleChannelElement;
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/**
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* channel element - generic struct for SCE/CPE/CCE/LFE
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*/
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typedef struct ChannelElement {
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int present;
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// CPE specific
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int common_window; ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
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int ms_mode; ///< Signals mid/side stereo flags coding mode (used by encoder)
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uint8_t is_mode; ///< Set if any bands have been encoded using intensity stereo (used by encoder)
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uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band
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uint8_t is_mask[128]; ///< Set if intensity stereo is used (used by encoder)
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// shared
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SingleChannelElement ch[2];
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// CCE specific
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ChannelCoupling coup;
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SpectralBandReplication sbr;
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} ChannelElement;
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/**
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* main AAC context
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*/
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struct AACContext {
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AVClass *class;
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AVCodecContext *avctx;
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AVFrame *frame;
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int is_saved; ///< Set if elements have stored overlap from previous frame.
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DynamicRangeControl che_drc;
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/**
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* @name Channel element related data
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* @{
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*/
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ChannelElement *che[4][MAX_ELEM_ID];
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ChannelElement *tag_che_map[4][MAX_ELEM_ID];
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int tags_mapped;
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int warned_remapping_once;
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/** @} */
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/**
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* @name temporary aligned temporary buffers
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* (We do not want to have these on the stack.)
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* @{
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*/
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DECLARE_ALIGNED(32, INTFLOAT, buf_mdct)[1024];
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/** @} */
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/**
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* @name Computed / set up during initialization
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* @{
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*/
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FFTContext mdct;
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FFTContext mdct_small;
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FFTContext mdct_ld;
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FFTContext mdct_ltp;
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#if USE_FIXED
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AVFixedDSPContext *fdsp;
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#else
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MDCT15Context *mdct120;
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MDCT15Context *mdct480;
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MDCT15Context *mdct960;
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AVFloatDSPContext *fdsp;
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#endif /* USE_FIXED */
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int random_state;
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/** @} */
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/**
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* @name Members used for output
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* @{
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*/
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SingleChannelElement *output_element[MAX_CHANNELS]; ///< Points to each SingleChannelElement
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/** @} */
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aacdec: add support for dual mono in Japanese DTV
Japanese DTV uses some non standard extensions in AAC audio.
One example is 'dual mono', which combines two independent
audio into one stereo stream, storing them in left and right channels
respectively. Historically, dual mono audio has been used for
multi-lingual audio, one for local/native language, and another for english,
and usually the "main" (local language) channel should be output without
any user interactions.
The frames of those dual mono audio are allowed to set
ADTS channel_config field to 0, and just contain two SCE's *WITHOUT* PCE,
which is a non standard extension by Japanese DTV standard.
(ref. ARIB STD-B32 PartII 5.2.3)
This patch adds an AVPacket side data, AV_PKT_DATA_JP_DUALMONO,
which indicates that the AVPacket is likely to contain an audio frame
with the above dual mono extension, and has the parameter to specify
the desired channel selection in that case.
It also makes aacdec to detect dual mono and output just the desired
channel when this side data is attached.
Signed-off-by: Akihiro Tsukada <atsukada@users.sourceforge.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
13 years ago
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/**
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* @name Japanese DTV specific extension
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* @{
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*/
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int force_dmono_mode;///< 0->not dmono, 1->use first channel, 2->use second channel
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int dmono_mode; ///< 0->not dmono, 1->use first channel, 2->use second channel
|
aacdec: add support for dual mono in Japanese DTV
Japanese DTV uses some non standard extensions in AAC audio.
One example is 'dual mono', which combines two independent
audio into one stereo stream, storing them in left and right channels
respectively. Historically, dual mono audio has been used for
multi-lingual audio, one for local/native language, and another for english,
and usually the "main" (local language) channel should be output without
any user interactions.
The frames of those dual mono audio are allowed to set
ADTS channel_config field to 0, and just contain two SCE's *WITHOUT* PCE,
which is a non standard extension by Japanese DTV standard.
(ref. ARIB STD-B32 PartII 5.2.3)
This patch adds an AVPacket side data, AV_PKT_DATA_JP_DUALMONO,
which indicates that the AVPacket is likely to contain an audio frame
with the above dual mono extension, and has the parameter to specify
the desired channel selection in that case.
It also makes aacdec to detect dual mono and output just the desired
channel when this side data is attached.
Signed-off-by: Akihiro Tsukada <atsukada@users.sourceforge.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
13 years ago
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/** @} */
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DECLARE_ALIGNED(32, INTFLOAT, temp)[128];
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OutputConfiguration oc[2];
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int warned_num_aac_frames;
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int warned_960_sbr;
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/* aacdec functions pointers */
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void (*imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce);
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void (*apply_ltp)(AACContext *ac, SingleChannelElement *sce);
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void (*apply_tns)(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
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IndividualChannelStream *ics, int decode);
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void (*windowing_and_mdct_ltp)(AACContext *ac, INTFLOAT *out,
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INTFLOAT *in, IndividualChannelStream *ics);
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void (*update_ltp)(AACContext *ac, SingleChannelElement *sce);
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void (*vector_pow43)(int *coefs, int len);
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void (*subband_scale)(int *dst, int *src, int scale, int offset, int len);
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};
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void ff_aacdec_init_mips(AACContext *c);
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#endif /* AVCODEC_AAC_H */
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