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d552f616a2
* qatar/master: (28 commits) Remove some non-compiling debug messages. ffplay: Fix non-compiling debug printf and replace it by av_dlog. H264: x86 predict init cosmetics. ac3enc: Fix linking of AC-3 encoder without the E-AC-3 encoder. Move E-AC-3 encoder functions to a separate eac3enc.c file. ac3enc: remove convenience macro, #define DEBUG ac3enc: remove unused #define vc1: re-initialize tables after width/height change. APIchanges: fill-in git commit hash for av_get_bytes_per_sample() addition samplefmt: add av_get_bytes_per_sample() iirfilter: fix biquad filter coefficients. swscale: remove duplicate conversion routine in swScale(). swscale: add yuv2planar/packed function typedefs. swscale: integrate yuv2nv12X_C into yuv2yuvX() function pointers. swscale: reindent x86 init code. swscale: extract SWS_FULL_CHR_H_INT conditional into init code. swscale: cosmetics. swscale: remove alp/chr/lumSrcOffset. swscale: un-special-case yuv2yuvX16_c(). shorten: Remove stray DEBUG #define and corresponding av_dlog statement. ... Conflicts: doc/APIchanges libavcodec/ac3enc.c libavutil/avutil.h libavutil/samplefmt.c libswscale/swscale.c libswscale/swscale_internal.h libswscale/x86/swscale_template.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
221 lines
7.1 KiB
C
221 lines
7.1 KiB
C
/*
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* Common code between the AC-3 encoder and decoder
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* Copyright (c) 2000, 2001, 2002 Fabrice Bellard
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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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* Common code between the AC-3 encoder and decoder.
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*/
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#ifndef AVCODEC_AC3_H
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#define AVCODEC_AC3_H
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#define AC3_MAX_CODED_FRAME_SIZE 3840 /* in bytes */
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#define AC3_MAX_CHANNELS 7 /**< maximum number of channels, including coupling channel */
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#define CPL_CH 0 /**< coupling channel index */
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#define AC3_MAX_COEFS 256
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#define AC3_BLOCK_SIZE 256
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#define AC3_MAX_BLOCKS 6
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#define AC3_FRAME_SIZE (AC3_MAX_BLOCKS * 256)
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#define AC3_WINDOW_SIZE (AC3_BLOCK_SIZE * 2)
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#define AC3_CRITICAL_BANDS 50
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#define AC3_MAX_CPL_BANDS 18
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "ac3tab.h"
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/* exponent encoding strategy */
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#define EXP_REUSE 0
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#define EXP_NEW 1
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#define EXP_D15 1
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#define EXP_D25 2
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#define EXP_D45 3
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/* pre-defined gain values */
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#define LEVEL_PLUS_3DB 1.4142135623730950
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#define LEVEL_PLUS_1POINT5DB 1.1892071150027209
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#define LEVEL_MINUS_1POINT5DB 0.8408964152537145
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#define LEVEL_MINUS_3DB 0.7071067811865476
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#define LEVEL_MINUS_4POINT5DB 0.5946035575013605
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#define LEVEL_MINUS_6DB 0.5000000000000000
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#define LEVEL_MINUS_9DB 0.3535533905932738
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#define LEVEL_ZERO 0.0000000000000000
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#define LEVEL_ONE 1.0000000000000000
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/** Delta bit allocation strategy */
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typedef enum {
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DBA_REUSE = 0,
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DBA_NEW,
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DBA_NONE,
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DBA_RESERVED
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} AC3DeltaStrategy;
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/** Channel mode (audio coding mode) */
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typedef enum {
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AC3_CHMODE_DUALMONO = 0,
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AC3_CHMODE_MONO,
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AC3_CHMODE_STEREO,
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AC3_CHMODE_3F,
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AC3_CHMODE_2F1R,
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AC3_CHMODE_3F1R,
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AC3_CHMODE_2F2R,
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AC3_CHMODE_3F2R
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} AC3ChannelMode;
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typedef struct AC3BitAllocParameters {
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int sr_code;
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int sr_shift;
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int slow_gain, slow_decay, fast_decay, db_per_bit, floor;
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int cpl_fast_leak, cpl_slow_leak;
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} AC3BitAllocParameters;
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/**
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* @struct AC3HeaderInfo
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* Coded AC-3 header values up to the lfeon element, plus derived values.
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*/
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typedef struct {
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/** @defgroup coded Coded elements
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* @{
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*/
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uint16_t sync_word;
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uint16_t crc1;
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uint8_t sr_code;
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uint8_t bitstream_id;
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uint8_t bitstream_mode;
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uint8_t channel_mode;
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uint8_t lfe_on;
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uint8_t frame_type;
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int substreamid; ///< substream identification
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int center_mix_level; ///< Center mix level index
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int surround_mix_level; ///< Surround mix level index
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uint16_t channel_map;
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int num_blocks; ///< number of audio blocks
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/** @} */
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/** @defgroup derived Derived values
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* @{
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*/
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uint8_t sr_shift;
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uint16_t sample_rate;
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uint32_t bit_rate;
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uint8_t channels;
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uint16_t frame_size;
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int64_t channel_layout;
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/** @} */
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} AC3HeaderInfo;
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typedef enum {
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EAC3_FRAME_TYPE_INDEPENDENT = 0,
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EAC3_FRAME_TYPE_DEPENDENT,
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EAC3_FRAME_TYPE_AC3_CONVERT,
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EAC3_FRAME_TYPE_RESERVED
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} EAC3FrameType;
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/**
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* Encoding Options used by AVOption.
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*/
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typedef struct AC3EncOptions {
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/* AC-3 metadata options*/
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int dialogue_level;
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int bitstream_mode;
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float center_mix_level;
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float surround_mix_level;
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int dolby_surround_mode;
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int audio_production_info;
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int mixing_level;
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int room_type;
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int copyright;
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int original;
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int extended_bsi_1;
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int preferred_stereo_downmix;
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float ltrt_center_mix_level;
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float ltrt_surround_mix_level;
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float loro_center_mix_level;
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float loro_surround_mix_level;
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int extended_bsi_2;
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int dolby_surround_ex_mode;
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int dolby_headphone_mode;
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int ad_converter_type;
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/* other encoding options */
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int allow_per_frame_metadata;
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int stereo_rematrixing;
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int channel_coupling;
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int cpl_start;
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} AC3EncOptions;
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void ff_ac3_common_init(void);
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extern const int64_t ff_ac3_channel_layouts[];
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extern const AVOption ff_ac3_options[];
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extern AVCodec ff_ac3_float_encoder;
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extern AVCodec ff_ac3_fixed_encoder;
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/**
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* Calculate the log power-spectral density of the input signal.
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* This gives a rough estimate of signal power in the frequency domain by using
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* the spectral envelope (exponents). The psd is also separately grouped
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* into critical bands for use in the calculating the masking curve.
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* 128 units in psd = -6 dB. The dbknee parameter in AC3BitAllocParameters
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* determines the reference level.
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*
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* @param[in] exp frequency coefficient exponents
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* @param[in] start starting bin location
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* @param[in] end ending bin location
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* @param[out] psd signal power for each frequency bin
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* @param[out] band_psd signal power for each critical band
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*/
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void ff_ac3_bit_alloc_calc_psd(int8_t *exp, int start, int end, int16_t *psd,
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int16_t *band_psd);
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/**
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* Calculate the masking curve.
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* First, the excitation is calculated using parameters in s and the signal
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* power in each critical band. The excitation is compared with a predefined
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* hearing threshold table to produce the masking curve. If delta bit
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* allocation information is provided, it is used for adjusting the masking
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* curve, usually to give a closer match to a better psychoacoustic model.
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*
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* @param[in] s adjustable bit allocation parameters
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* @param[in] band_psd signal power for each critical band
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* @param[in] start starting bin location
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* @param[in] end ending bin location
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* @param[in] fast_gain fast gain (estimated signal-to-mask ratio)
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* @param[in] is_lfe whether or not the channel being processed is the LFE
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* @param[in] dba_mode delta bit allocation mode (none, reuse, or new)
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* @param[in] dba_nsegs number of delta segments
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* @param[in] dba_offsets location offsets for each segment
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* @param[in] dba_lengths length of each segment
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* @param[in] dba_values delta bit allocation for each segment
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* @param[out] mask calculated masking curve
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* @return returns 0 for success, non-zero for error
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*/
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int ff_ac3_bit_alloc_calc_mask(AC3BitAllocParameters *s, int16_t *band_psd,
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int start, int end, int fast_gain, int is_lfe,
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int dba_mode, int dba_nsegs, uint8_t *dba_offsets,
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uint8_t *dba_lengths, uint8_t *dba_values,
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int16_t *mask);
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#endif /* AVCODEC_AC3_H */
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