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FFmpeg/libavcodec/aac.h
Claudio Freire 6394acaf36 AAC: Fix M/S stereo encoding
This patch fixes a pointer arithmetic bug in adjust_frame_information that resulted in heavily corrupted audio when using M/S encoding. Also, a backup copy of untransformed coefficients has to be kept around or attempts at re-processing the frame (which happens when hevavily overspending bits during transients) will result in re-encoding of the coefficients and subsequent corruption of the resulting stream.

A/B testing shows the bug as corrected, but still cannot prove that M/S coding is a win at least in numbers. Limited listening tests do show improvement on M/S encoded samples in lower bitrates, but they're hidden among the other artifacts that remain to be corrected in the encoder.

Some of the regressions flagged in the report do show poor stereo image (but not buggy), so M/S encoding is clearly not good enough yet to be defaulted to auto.

In numbers, Patched against Unpatched, stereo_mode auto:

  Files: 114
  Bitrates: 6
  Tests: 683

  Serious Regressions: 0 (0%)
  Regressions: 0 (0%)
  Improvements: 227 (33%)
  Big improvements: 92 (13%)
  Worst regression - mybloodrusts.wv - 256k
    - StdDev: 28.61       pSNR: -0.43     maxdiff: 1372.00
  Best improvement - 60.wv - 384k
    - StdDev: -369.57     pSNR: 45.02     maxdiff: -13322.00
  Average          - StdDev: -80.56       pSNR: 2.49      maxdiff: -8858.00

Patched against Unpatched stereo_mode ms_off shows no difference.

Patched stereo_mode auto vs Unpatched stereo_mode ms_off shows a small average improvement, just not too significant:

  Serious Regressions: 0 (0%)
  Regressions: 10 (1%)
  Improvements: 45 (6%)
  Big improvements: 2 (0%)
  Worst regression - Illinois.wv - 256k
    - StdDev: 33.20       pSNR: -2.03     maxdiff: 477.00
  Best improvement - song_of_circomstances.flac - 384k
    - StdDev: -3.97       pSNR: 7.61      maxdiff: -826.00
  Average          - StdDev: -10.25       pSNR: 0.20      maxdiff: -281.00

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2015-03-03 13:57:42 +01:00

338 lines
11 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 "libavutil/float_dsp.h"
#include "avcodec.h"
#include "imdct15.h"
#include "fft.h"
#include "mpeg4audio.h"
#include "sbr.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 OutputConfiguration {
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 PredictorState {
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
/**
* Long Term Prediction
*/
typedef struct LongTermPrediction {
int8_t present;
int16_t lag;
float coef;
int8_t used[MAX_LTP_LONG_SFB];
} LongTermPrediction;
/**
* 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
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 TemporalNoiseShaping {
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 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;
typedef struct Pulse {
int num_pulse;
int start;
int pos[4];
int amp[4];
} Pulse;
/**
* 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
*/
float gain[16][120];
} ChannelCoupling;
/**
* Single Channel Element - used for both SCE and LFE elements.
*/
typedef struct SingleChannelElement {
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, pcoeffs)[1024]; ///< coefficients for IMDCT, pristine
DECLARE_ALIGNED(32, float, coeffs)[1024]; ///< coefficients for IMDCT, maybe processed
DECLARE_ALIGNED(32, float, saved)[1536]; ///< overlap
DECLARE_ALIGNED(32, float, ret_buf)[2048]; ///< PCM output buffer
DECLARE_ALIGNED(16, float, ltp_state)[3072]; ///< time signal for LTP
PredictorState predictor_state[MAX_PREDICTORS];
float *ret; ///< PCM output
} SingleChannelElement;
/**
* channel element - generic struct for SCE/CPE/CCE/LFE
*/
typedef struct ChannelElement {
int present;
// 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
*/
struct AACContext {
AVClass *class;
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;
int warned_remapping_once;
/** @} */
/**
* @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_ld;
FFTContext mdct_ltp;
IMDCT15Context *mdct480;
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
/** @} */
DECLARE_ALIGNED(32, float, temp)[128];
OutputConfiguration oc[2];
int warned_num_aac_frames;
/* aacdec functions pointers */
void (*imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce);
void (*apply_ltp)(AACContext *ac, SingleChannelElement *sce);
void (*apply_tns)(float coef[1024], TemporalNoiseShaping *tns,
IndividualChannelStream *ics, int decode);
void (*windowing_and_mdct_ltp)(AACContext *ac, float *out,
float *in, IndividualChannelStream *ics);
void (*update_ltp)(AACContext *ac, SingleChannelElement *sce);
};
void ff_aacdec_init_mips(AACContext *c);
#endif /* AVCODEC_AAC_H */