FFmpeg/libavcodec/ac3.h
Michael Niedermayer c3f5b81125 Merge remote branch 'qatar/master'
* qatar/master:
  ALPHA: Replace sized int_fast integer types with plain int/unsigned.
Duplicate  DPX image encoder
Duplicate  DPX decoder: read sample aspect ratio
Duplciate  DPX decoder: add buffer size checks.
  ac3enc: clip large coefficient values and negative exponents rather than using av_assert2().
  ac3enc: do not store a bandwidth code for each channel.
  ac3enc: remove bandwidth reduction as fallback for bit allocation failure.
  ac3enc: merge compute_exp_strategy_ch() into compute_exp_strategy()
  ac3enc: return error if frame+exponent bits are too large instead of using av_assert2().
  ac3enc: differentiate between current block and reference block in bit_alloc()
  ac3enc: simplify exponent_init() by calculating exponent_group_tab[] based on exponent group sizes.
  ac3enc: simplify stereo rematrixing decision options
Include both URLs:  Update URL to fate samples

Conflicts:
	Changelog
	doc/fate.txt
	libavcodec/ac3enc.c
	libavcodec/dpxenc.c
	libavcodec/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-05-02 03:45:58 +02:00

219 lines
6.9 KiB
C

/*
* Common code between the AC-3 encoder and decoder
* Copyright (c) 2000, 2001, 2002 Fabrice Bellard
*
* 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
* Common code between the AC-3 encoder and decoder.
*/
#ifndef AVCODEC_AC3_H
#define AVCODEC_AC3_H
#define AC3_MAX_CODED_FRAME_SIZE 3840 /* in bytes */
#define AC3_MAX_CHANNELS 6 /* including LFE channel */
#define AC3_MAX_COEFS 256
#define AC3_BLOCK_SIZE 256
#define AC3_MAX_BLOCKS 6
#define AC3_FRAME_SIZE (AC3_MAX_BLOCKS * 256)
#define AC3_WINDOW_SIZE (AC3_BLOCK_SIZE * 2)
#define AC3_CRITICAL_BANDS 50
#define AC3_MAX_CPL_BANDS 18
#include "libavutil/opt.h"
#include "avcodec.h"
#include "ac3tab.h"
/* exponent encoding strategy */
#define EXP_REUSE 0
#define EXP_NEW 1
#define EXP_D15 1
#define EXP_D25 2
#define EXP_D45 3
/* pre-defined gain values */
#define LEVEL_PLUS_3DB 1.4142135623730950
#define LEVEL_PLUS_1POINT5DB 1.1892071150027209
#define LEVEL_MINUS_1POINT5DB 0.8408964152537145
#define LEVEL_MINUS_3DB 0.7071067811865476
#define LEVEL_MINUS_4POINT5DB 0.5946035575013605
#define LEVEL_MINUS_6DB 0.5000000000000000
#define LEVEL_MINUS_9DB 0.3535533905932738
#define LEVEL_ZERO 0.0000000000000000
#define LEVEL_ONE 1.0000000000000000
/** Delta bit allocation strategy */
typedef enum {
DBA_REUSE = 0,
DBA_NEW,
DBA_NONE,
DBA_RESERVED
} AC3DeltaStrategy;
/** Channel mode (audio coding mode) */
typedef enum {
AC3_CHMODE_DUALMONO = 0,
AC3_CHMODE_MONO,
AC3_CHMODE_STEREO,
AC3_CHMODE_3F,
AC3_CHMODE_2F1R,
AC3_CHMODE_3F1R,
AC3_CHMODE_2F2R,
AC3_CHMODE_3F2R
} AC3ChannelMode;
typedef struct AC3BitAllocParameters {
int sr_code;
int sr_shift;
int slow_gain, slow_decay, fast_decay, db_per_bit, floor;
int cpl_fast_leak, cpl_slow_leak;
} AC3BitAllocParameters;
/**
* @struct AC3HeaderInfo
* Coded AC-3 header values up to the lfeon element, plus derived values.
*/
typedef struct {
/** @defgroup coded Coded elements
* @{
*/
uint16_t sync_word;
uint16_t crc1;
uint8_t sr_code;
uint8_t bitstream_id;
uint8_t bitstream_mode;
uint8_t channel_mode;
uint8_t lfe_on;
uint8_t frame_type;
int substreamid; ///< substream identification
int center_mix_level; ///< Center mix level index
int surround_mix_level; ///< Surround mix level index
uint16_t channel_map;
int num_blocks; ///< number of audio blocks
/** @} */
/** @defgroup derived Derived values
* @{
*/
uint8_t sr_shift;
uint16_t sample_rate;
uint32_t bit_rate;
uint8_t channels;
uint16_t frame_size;
int64_t channel_layout;
/** @} */
} AC3HeaderInfo;
typedef enum {
EAC3_FRAME_TYPE_INDEPENDENT = 0,
EAC3_FRAME_TYPE_DEPENDENT,
EAC3_FRAME_TYPE_AC3_CONVERT,
EAC3_FRAME_TYPE_RESERVED
} EAC3FrameType;
/**
* Encoding Options used by AVOption.
*/
typedef struct AC3EncOptions {
/* AC-3 metadata options*/
int dialogue_level;
int bitstream_mode;
float center_mix_level;
float surround_mix_level;
int dolby_surround_mode;
int audio_production_info;
int mixing_level;
int room_type;
int copyright;
int original;
int extended_bsi_1;
int preferred_stereo_downmix;
float ltrt_center_mix_level;
float ltrt_surround_mix_level;
float loro_center_mix_level;
float loro_surround_mix_level;
int extended_bsi_2;
int dolby_surround_ex_mode;
int dolby_headphone_mode;
int ad_converter_type;
/* other encoding options */
int allow_per_frame_metadata;
int stereo_rematrixing;
} AC3EncOptions;
void ff_ac3_common_init(void);
extern const int64_t ff_ac3_channel_layouts[];
extern const AVOption ff_ac3_options[];
extern AVCodec ff_ac3_float_encoder;
extern AVCodec ff_ac3_fixed_encoder;
/**
* Calculate the log power-spectral density of the input signal.
* This gives a rough estimate of signal power in the frequency domain by using
* the spectral envelope (exponents). The psd is also separately grouped
* into critical bands for use in the calculating the masking curve.
* 128 units in psd = -6 dB. The dbknee parameter in AC3BitAllocParameters
* determines the reference level.
*
* @param[in] exp frequency coefficient exponents
* @param[in] start starting bin location
* @param[in] end ending bin location
* @param[out] psd signal power for each frequency bin
* @param[out] band_psd signal power for each critical band
*/
void ff_ac3_bit_alloc_calc_psd(int8_t *exp, int start, int end, int16_t *psd,
int16_t *band_psd);
/**
* Calculate the masking curve.
* First, the excitation is calculated using parameters in s and the signal
* power in each critical band. The excitation is compared with a predefined
* hearing threshold table to produce the masking curve. If delta bit
* allocation information is provided, it is used for adjusting the masking
* curve, usually to give a closer match to a better psychoacoustic model.
*
* @param[in] s adjustable bit allocation parameters
* @param[in] band_psd signal power for each critical band
* @param[in] start starting bin location
* @param[in] end ending bin location
* @param[in] fast_gain fast gain (estimated signal-to-mask ratio)
* @param[in] is_lfe whether or not the channel being processed is the LFE
* @param[in] dba_mode delta bit allocation mode (none, reuse, or new)
* @param[in] dba_nsegs number of delta segments
* @param[in] dba_offsets location offsets for each segment
* @param[in] dba_lengths length of each segment
* @param[in] dba_values delta bit allocation for each segment
* @param[out] mask calculated masking curve
* @return returns 0 for success, non-zero for error
*/
int ff_ac3_bit_alloc_calc_mask(AC3BitAllocParameters *s, int16_t *band_psd,
int start, int end, int fast_gain, int is_lfe,
int dba_mode, int dba_nsegs, uint8_t *dba_offsets,
uint8_t *dba_lengths, uint8_t *dba_values,
int16_t *mask);
#endif /* AVCODEC_AC3_H */