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* newdev/master: mpegts: propagate avio EOF in read_packet() configure: Initial support for --target-os=symbian Fixed-point FFT and MDCT Include dependencies for test programs ac3enc: simplify sym_quant() flvdec: read index stored in the 'keyframes' tag. mov: Add support for zero-sized stsc runs. Merged-by: Michael Niedermayer <michaelni@gmx.at>
2256 lines
77 KiB
C
2256 lines
77 KiB
C
/*
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* The simplest AC-3 encoder
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* Copyright (c) 2000 Fabrice Bellard
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* Copyright (c) 2006-2010 Justin Ruggles <justin.ruggles@gmail.com>
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* Copyright (c) 2006-2010 Prakash Punnoor <prakash@punnoor.de>
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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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* The simplest AC-3 encoder.
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*/
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//#define DEBUG
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//#define ASSERT_LEVEL 2
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#include "libavutil/audioconvert.h"
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#include "libavutil/avassert.h"
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#include "libavutil/crc.h"
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "put_bits.h"
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#include "dsputil.h"
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#include "ac3dsp.h"
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#include "ac3.h"
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#include "audioconvert.h"
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#ifndef CONFIG_AC3ENC_FLOAT
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#define CONFIG_AC3ENC_FLOAT 0
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#endif
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/** Maximum number of exponent groups. +1 for separate DC exponent. */
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#define AC3_MAX_EXP_GROUPS 85
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/* stereo rematrixing algorithms */
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#define AC3_REMATRIXING_IS_STATIC 0x1
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#define AC3_REMATRIXING_SUMS 0
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#define AC3_REMATRIXING_NONE 1
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#define AC3_REMATRIXING_ALWAYS 3
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/** Scale a float value by 2^bits and convert to an integer. */
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#define SCALE_FLOAT(a, bits) lrintf((a) * (float)(1 << (bits)))
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#if CONFIG_AC3ENC_FLOAT
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#include "ac3enc_float.h"
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#else
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#include "ac3enc_fixed.h"
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#endif
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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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} AC3EncOptions;
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/**
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* Data for a single audio block.
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*/
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typedef struct AC3Block {
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uint8_t **bap; ///< bit allocation pointers (bap)
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CoefType **mdct_coef; ///< MDCT coefficients
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int32_t **fixed_coef; ///< fixed-point MDCT coefficients
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uint8_t **exp; ///< original exponents
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uint8_t **grouped_exp; ///< grouped exponents
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int16_t **psd; ///< psd per frequency bin
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int16_t **band_psd; ///< psd per critical band
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int16_t **mask; ///< masking curve
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uint16_t **qmant; ///< quantized mantissas
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uint8_t coeff_shift[AC3_MAX_CHANNELS]; ///< fixed-point coefficient shift values
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uint8_t new_rematrixing_strategy; ///< send new rematrixing flags in this block
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uint8_t rematrixing_flags[4]; ///< rematrixing flags
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struct AC3Block *exp_ref_block[AC3_MAX_CHANNELS]; ///< reference blocks for EXP_REUSE
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} AC3Block;
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/**
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* AC-3 encoder private context.
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*/
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typedef struct AC3EncodeContext {
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AVClass *av_class; ///< AVClass used for AVOption
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AC3EncOptions options; ///< encoding options
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PutBitContext pb; ///< bitstream writer context
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DSPContext dsp;
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AC3DSPContext ac3dsp; ///< AC-3 optimized functions
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AC3MDCTContext mdct; ///< MDCT context
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AC3Block blocks[AC3_MAX_BLOCKS]; ///< per-block info
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int bitstream_id; ///< bitstream id (bsid)
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int bitstream_mode; ///< bitstream mode (bsmod)
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int bit_rate; ///< target bit rate, in bits-per-second
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int sample_rate; ///< sampling frequency, in Hz
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int frame_size_min; ///< minimum frame size in case rounding is necessary
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int frame_size; ///< current frame size in bytes
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int frame_size_code; ///< frame size code (frmsizecod)
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uint16_t crc_inv[2];
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int bits_written; ///< bit count (used to avg. bitrate)
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int samples_written; ///< sample count (used to avg. bitrate)
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int fbw_channels; ///< number of full-bandwidth channels (nfchans)
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int channels; ///< total number of channels (nchans)
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int lfe_on; ///< indicates if there is an LFE channel (lfeon)
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int lfe_channel; ///< channel index of the LFE channel
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int has_center; ///< indicates if there is a center channel
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int has_surround; ///< indicates if there are one or more surround channels
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int channel_mode; ///< channel mode (acmod)
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const uint8_t *channel_map; ///< channel map used to reorder channels
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int center_mix_level; ///< center mix level code
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int surround_mix_level; ///< surround mix level code
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int ltrt_center_mix_level; ///< Lt/Rt center mix level code
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int ltrt_surround_mix_level; ///< Lt/Rt surround mix level code
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int loro_center_mix_level; ///< Lo/Ro center mix level code
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int loro_surround_mix_level; ///< Lo/Ro surround mix level code
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int cutoff; ///< user-specified cutoff frequency, in Hz
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int bandwidth_code[AC3_MAX_CHANNELS]; ///< bandwidth code (0 to 60) (chbwcod)
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int nb_coefs[AC3_MAX_CHANNELS];
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int rematrixing; ///< determines how rematrixing strategy is calculated
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int num_rematrixing_bands; ///< number of rematrixing bands
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/* bitrate allocation control */
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int slow_gain_code; ///< slow gain code (sgaincod)
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int slow_decay_code; ///< slow decay code (sdcycod)
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int fast_decay_code; ///< fast decay code (fdcycod)
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int db_per_bit_code; ///< dB/bit code (dbpbcod)
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int floor_code; ///< floor code (floorcod)
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AC3BitAllocParameters bit_alloc; ///< bit allocation parameters
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int coarse_snr_offset; ///< coarse SNR offsets (csnroffst)
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int fast_gain_code[AC3_MAX_CHANNELS]; ///< fast gain codes (signal-to-mask ratio) (fgaincod)
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int fine_snr_offset[AC3_MAX_CHANNELS]; ///< fine SNR offsets (fsnroffst)
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int frame_bits_fixed; ///< number of non-coefficient bits for fixed parameters
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int frame_bits; ///< all frame bits except exponents and mantissas
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int exponent_bits; ///< number of bits used for exponents
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SampleType **planar_samples;
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uint8_t *bap_buffer;
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uint8_t *bap1_buffer;
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CoefType *mdct_coef_buffer;
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int32_t *fixed_coef_buffer;
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uint8_t *exp_buffer;
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uint8_t *grouped_exp_buffer;
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int16_t *psd_buffer;
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int16_t *band_psd_buffer;
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int16_t *mask_buffer;
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uint16_t *qmant_buffer;
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uint8_t exp_strategy[AC3_MAX_CHANNELS][AC3_MAX_BLOCKS]; ///< exponent strategies
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DECLARE_ALIGNED(16, SampleType, windowed_samples)[AC3_WINDOW_SIZE];
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} AC3EncodeContext;
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typedef struct AC3Mant {
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uint16_t *qmant1_ptr, *qmant2_ptr, *qmant4_ptr; ///< mantissa pointers for bap=1,2,4
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int mant1_cnt, mant2_cnt, mant4_cnt; ///< mantissa counts for bap=1,2,4
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} AC3Mant;
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#define CMIXLEV_NUM_OPTIONS 3
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static const float cmixlev_options[CMIXLEV_NUM_OPTIONS] = {
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LEVEL_MINUS_3DB, LEVEL_MINUS_4POINT5DB, LEVEL_MINUS_6DB
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};
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#define SURMIXLEV_NUM_OPTIONS 3
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static const float surmixlev_options[SURMIXLEV_NUM_OPTIONS] = {
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LEVEL_MINUS_3DB, LEVEL_MINUS_6DB, LEVEL_ZERO
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};
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#define EXTMIXLEV_NUM_OPTIONS 8
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static const float extmixlev_options[EXTMIXLEV_NUM_OPTIONS] = {
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LEVEL_PLUS_3DB, LEVEL_PLUS_1POINT5DB, LEVEL_ONE, LEVEL_MINUS_4POINT5DB,
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LEVEL_MINUS_3DB, LEVEL_MINUS_4POINT5DB, LEVEL_MINUS_6DB, LEVEL_ZERO
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};
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#define OFFSET(param) offsetof(AC3EncodeContext, options.param)
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#define AC3ENC_PARAM (AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM)
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static const AVOption options[] = {
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/* Metadata Options */
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{"per_frame_metadata", "Allow Changing Metadata Per-Frame", OFFSET(allow_per_frame_metadata), FF_OPT_TYPE_INT, 0, 0, 1, AC3ENC_PARAM},
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/* downmix levels */
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{"center_mixlev", "Center Mix Level", OFFSET(center_mix_level), FF_OPT_TYPE_FLOAT, LEVEL_MINUS_4POINT5DB, 0.0, 1.0, AC3ENC_PARAM},
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{"surround_mixlev", "Surround Mix Level", OFFSET(surround_mix_level), FF_OPT_TYPE_FLOAT, LEVEL_MINUS_6DB, 0.0, 1.0, AC3ENC_PARAM},
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/* audio production information */
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{"mixing_level", "Mixing Level", OFFSET(mixing_level), FF_OPT_TYPE_INT, -1, -1, 111, AC3ENC_PARAM},
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{"room_type", "Room Type", OFFSET(room_type), FF_OPT_TYPE_INT, -1, -1, 2, AC3ENC_PARAM, "room_type"},
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{"notindicated", "Not Indicated (default)", 0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "room_type"},
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{"large", "Large Room", 0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "room_type"},
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{"small", "Small Room", 0, FF_OPT_TYPE_CONST, 2, INT_MIN, INT_MAX, AC3ENC_PARAM, "room_type"},
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/* other metadata options */
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{"copyright", "Copyright Bit", OFFSET(copyright), FF_OPT_TYPE_INT, 0, 0, 1, AC3ENC_PARAM},
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{"dialnorm", "Dialogue Level (dB)", OFFSET(dialogue_level), FF_OPT_TYPE_INT, -31, -31, -1, AC3ENC_PARAM},
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{"dsur_mode", "Dolby Surround Mode", OFFSET(dolby_surround_mode), FF_OPT_TYPE_INT, 0, 0, 2, AC3ENC_PARAM, "dsur_mode"},
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{"notindicated", "Not Indicated (default)", 0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsur_mode"},
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{"on", "Dolby Surround Encoded", 0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsur_mode"},
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{"off", "Not Dolby Surround Encoded", 0, FF_OPT_TYPE_CONST, 2, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsur_mode"},
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{"original", "Original Bit Stream", OFFSET(original), FF_OPT_TYPE_INT, 1, 0, 1, AC3ENC_PARAM},
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/* extended bitstream information */
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{"dmix_mode", "Preferred Stereo Downmix Mode", OFFSET(preferred_stereo_downmix), FF_OPT_TYPE_INT, -1, -1, 2, AC3ENC_PARAM, "dmix_mode"},
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{"notindicated", "Not Indicated (default)", 0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "dmix_mode"},
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{"ltrt", "Lt/Rt Downmix Preferred", 0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "dmix_mode"},
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{"loro", "Lo/Ro Downmix Preferred", 0, FF_OPT_TYPE_CONST, 2, INT_MIN, INT_MAX, AC3ENC_PARAM, "dmix_mode"},
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{"ltrt_cmixlev", "Lt/Rt Center Mix Level", OFFSET(ltrt_center_mix_level), FF_OPT_TYPE_FLOAT, -1.0, -1.0, 2.0, AC3ENC_PARAM},
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{"ltrt_surmixlev", "Lt/Rt Surround Mix Level", OFFSET(ltrt_surround_mix_level), FF_OPT_TYPE_FLOAT, -1.0, -1.0, 2.0, AC3ENC_PARAM},
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{"loro_cmixlev", "Lo/Ro Center Mix Level", OFFSET(loro_center_mix_level), FF_OPT_TYPE_FLOAT, -1.0, -1.0, 2.0, AC3ENC_PARAM},
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{"loro_surmixlev", "Lo/Ro Surround Mix Level", OFFSET(loro_surround_mix_level), FF_OPT_TYPE_FLOAT, -1.0, -1.0, 2.0, AC3ENC_PARAM},
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{"dsurex_mode", "Dolby Surround EX Mode", OFFSET(dolby_surround_ex_mode), FF_OPT_TYPE_INT, -1, -1, 2, AC3ENC_PARAM, "dsurex_mode"},
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{"notindicated", "Not Indicated (default)", 0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsurex_mode"},
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{"on", "Dolby Surround EX Encoded", 0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsurex_mode"},
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{"off", "Not Dolby Surround EX Encoded", 0, FF_OPT_TYPE_CONST, 2, INT_MIN, INT_MAX, AC3ENC_PARAM, "dsurex_mode"},
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{"dheadphone_mode", "Dolby Headphone Mode", OFFSET(dolby_headphone_mode), FF_OPT_TYPE_INT, -1, -1, 2, AC3ENC_PARAM, "dheadphone_mode"},
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{"notindicated", "Not Indicated (default)", 0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "dheadphone_mode"},
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{"on", "Dolby Headphone Encoded", 0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "dheadphone_mode"},
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{"off", "Not Dolby Headphone Encoded", 0, FF_OPT_TYPE_CONST, 2, INT_MIN, INT_MAX, AC3ENC_PARAM, "dheadphone_mode"},
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{"ad_conv_type", "A/D Converter Type", OFFSET(ad_converter_type), FF_OPT_TYPE_INT, -1, -1, 1, AC3ENC_PARAM, "ad_conv_type"},
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{"standard", "Standard (default)", 0, FF_OPT_TYPE_CONST, 0, INT_MIN, INT_MAX, AC3ENC_PARAM, "ad_conv_type"},
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{"hdcd", "HDCD", 0, FF_OPT_TYPE_CONST, 1, INT_MIN, INT_MAX, AC3ENC_PARAM, "ad_conv_type"},
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{NULL}
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};
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#if CONFIG_AC3ENC_FLOAT
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static AVClass ac3enc_class = { "AC-3 Encoder", av_default_item_name,
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options, LIBAVUTIL_VERSION_INT };
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#else
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static AVClass ac3enc_class = { "Fixed-Point AC-3 Encoder", av_default_item_name,
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options, LIBAVUTIL_VERSION_INT };
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#endif
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/* prototypes for functions in ac3enc_fixed.c and ac3enc_float.c */
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static av_cold void mdct_end(AC3MDCTContext *mdct);
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static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
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int nbits);
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static void mdct512(AC3MDCTContext *mdct, CoefType *out, SampleType *in);
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static void apply_window(DSPContext *dsp, SampleType *output, const SampleType *input,
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const SampleType *window, unsigned int len);
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static int normalize_samples(AC3EncodeContext *s);
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static void scale_coefficients(AC3EncodeContext *s);
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/**
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* LUT for number of exponent groups.
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* exponent_group_tab[exponent strategy-1][number of coefficients]
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*/
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static uint8_t exponent_group_tab[3][256];
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/**
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* List of supported channel layouts.
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*/
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static const int64_t ac3_channel_layouts[] = {
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AV_CH_LAYOUT_MONO,
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AV_CH_LAYOUT_STEREO,
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AV_CH_LAYOUT_2_1,
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AV_CH_LAYOUT_SURROUND,
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AV_CH_LAYOUT_2_2,
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AV_CH_LAYOUT_QUAD,
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AV_CH_LAYOUT_4POINT0,
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AV_CH_LAYOUT_5POINT0,
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AV_CH_LAYOUT_5POINT0_BACK,
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(AV_CH_LAYOUT_MONO | AV_CH_LOW_FREQUENCY),
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(AV_CH_LAYOUT_STEREO | AV_CH_LOW_FREQUENCY),
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(AV_CH_LAYOUT_2_1 | AV_CH_LOW_FREQUENCY),
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(AV_CH_LAYOUT_SURROUND | AV_CH_LOW_FREQUENCY),
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(AV_CH_LAYOUT_2_2 | AV_CH_LOW_FREQUENCY),
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(AV_CH_LAYOUT_QUAD | AV_CH_LOW_FREQUENCY),
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(AV_CH_LAYOUT_4POINT0 | AV_CH_LOW_FREQUENCY),
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AV_CH_LAYOUT_5POINT1,
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AV_CH_LAYOUT_5POINT1_BACK,
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0
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};
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/**
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* Adjust the frame size to make the average bit rate match the target bit rate.
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* This is only needed for 11025, 22050, and 44100 sample rates.
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*/
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static void adjust_frame_size(AC3EncodeContext *s)
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{
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while (s->bits_written >= s->bit_rate && s->samples_written >= s->sample_rate) {
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s->bits_written -= s->bit_rate;
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s->samples_written -= s->sample_rate;
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}
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s->frame_size = s->frame_size_min +
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2 * (s->bits_written * s->sample_rate < s->samples_written * s->bit_rate);
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s->bits_written += s->frame_size * 8;
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s->samples_written += AC3_FRAME_SIZE;
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}
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/**
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* Deinterleave input samples.
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* Channels are reordered from FFmpeg's default order to AC-3 order.
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*/
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static void deinterleave_input_samples(AC3EncodeContext *s,
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const SampleType *samples)
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{
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int ch, i;
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/* deinterleave and remap input samples */
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for (ch = 0; ch < s->channels; ch++) {
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const SampleType *sptr;
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int sinc;
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/* copy last 256 samples of previous frame to the start of the current frame */
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memcpy(&s->planar_samples[ch][0], &s->planar_samples[ch][AC3_FRAME_SIZE],
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AC3_BLOCK_SIZE * sizeof(s->planar_samples[0][0]));
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/* deinterleave */
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sinc = s->channels;
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sptr = samples + s->channel_map[ch];
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for (i = AC3_BLOCK_SIZE; i < AC3_FRAME_SIZE+AC3_BLOCK_SIZE; i++) {
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s->planar_samples[ch][i] = *sptr;
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sptr += sinc;
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}
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}
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}
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/**
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* Apply the MDCT to input samples to generate frequency coefficients.
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* This applies the KBD window and normalizes the input to reduce precision
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* loss due to fixed-point calculations.
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*/
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static void apply_mdct(AC3EncodeContext *s)
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{
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int blk, ch;
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for (ch = 0; ch < s->channels; ch++) {
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for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
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AC3Block *block = &s->blocks[blk];
|
|
const SampleType *input_samples = &s->planar_samples[ch][blk * AC3_BLOCK_SIZE];
|
|
|
|
apply_window(&s->dsp, s->windowed_samples, input_samples, s->mdct.window, AC3_WINDOW_SIZE);
|
|
|
|
block->coeff_shift[ch] = normalize_samples(s);
|
|
|
|
mdct512(&s->mdct, block->mdct_coef[ch], s->windowed_samples);
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* Initialize stereo rematrixing.
|
|
* If the strategy does not change for each frame, set the rematrixing flags.
|
|
*/
|
|
static void rematrixing_init(AC3EncodeContext *s)
|
|
{
|
|
if (s->channel_mode == AC3_CHMODE_STEREO)
|
|
s->rematrixing = AC3_REMATRIXING_SUMS;
|
|
else
|
|
s->rematrixing = AC3_REMATRIXING_NONE;
|
|
/* NOTE: AC3_REMATRIXING_ALWAYS might be used in
|
|
the future in conjunction with channel coupling. */
|
|
|
|
if (s->rematrixing & AC3_REMATRIXING_IS_STATIC) {
|
|
int flag = (s->rematrixing == AC3_REMATRIXING_ALWAYS);
|
|
s->blocks[0].new_rematrixing_strategy = 1;
|
|
memset(s->blocks[0].rematrixing_flags, flag,
|
|
sizeof(s->blocks[0].rematrixing_flags));
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* Determine rematrixing flags for each block and band.
|
|
*/
|
|
static void compute_rematrixing_strategy(AC3EncodeContext *s)
|
|
{
|
|
int nb_coefs;
|
|
int blk, bnd, i;
|
|
AC3Block *block, *block0;
|
|
|
|
s->num_rematrixing_bands = 4;
|
|
|
|
if (s->rematrixing & AC3_REMATRIXING_IS_STATIC)
|
|
return;
|
|
|
|
nb_coefs = FFMIN(s->nb_coefs[0], s->nb_coefs[1]);
|
|
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
block = &s->blocks[blk];
|
|
block->new_rematrixing_strategy = !blk;
|
|
for (bnd = 0; bnd < s->num_rematrixing_bands; bnd++) {
|
|
/* calculate calculate sum of squared coeffs for one band in one block */
|
|
int start = ff_ac3_rematrix_band_tab[bnd];
|
|
int end = FFMIN(nb_coefs, ff_ac3_rematrix_band_tab[bnd+1]);
|
|
CoefSumType sum[4] = {0,};
|
|
for (i = start; i < end; i++) {
|
|
CoefType lt = block->mdct_coef[0][i];
|
|
CoefType rt = block->mdct_coef[1][i];
|
|
CoefType md = lt + rt;
|
|
CoefType sd = lt - rt;
|
|
MAC_COEF(sum[0], lt, lt);
|
|
MAC_COEF(sum[1], rt, rt);
|
|
MAC_COEF(sum[2], md, md);
|
|
MAC_COEF(sum[3], sd, sd);
|
|
}
|
|
|
|
/* compare sums to determine if rematrixing will be used for this band */
|
|
if (FFMIN(sum[2], sum[3]) < FFMIN(sum[0], sum[1]))
|
|
block->rematrixing_flags[bnd] = 1;
|
|
else
|
|
block->rematrixing_flags[bnd] = 0;
|
|
|
|
/* determine if new rematrixing flags will be sent */
|
|
if (blk &&
|
|
block->rematrixing_flags[bnd] != block0->rematrixing_flags[bnd]) {
|
|
block->new_rematrixing_strategy = 1;
|
|
}
|
|
}
|
|
block0 = block;
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* Apply stereo rematrixing to coefficients based on rematrixing flags.
|
|
*/
|
|
static void apply_rematrixing(AC3EncodeContext *s)
|
|
{
|
|
int nb_coefs;
|
|
int blk, bnd, i;
|
|
int start, end;
|
|
uint8_t *flags;
|
|
|
|
if (s->rematrixing == AC3_REMATRIXING_NONE)
|
|
return;
|
|
|
|
nb_coefs = FFMIN(s->nb_coefs[0], s->nb_coefs[1]);
|
|
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
AC3Block *block = &s->blocks[blk];
|
|
if (block->new_rematrixing_strategy)
|
|
flags = block->rematrixing_flags;
|
|
for (bnd = 0; bnd < s->num_rematrixing_bands; bnd++) {
|
|
if (flags[bnd]) {
|
|
start = ff_ac3_rematrix_band_tab[bnd];
|
|
end = FFMIN(nb_coefs, ff_ac3_rematrix_band_tab[bnd+1]);
|
|
for (i = start; i < end; i++) {
|
|
int32_t lt = block->fixed_coef[0][i];
|
|
int32_t rt = block->fixed_coef[1][i];
|
|
block->fixed_coef[0][i] = (lt + rt) >> 1;
|
|
block->fixed_coef[1][i] = (lt - rt) >> 1;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* Initialize exponent tables.
|
|
*/
|
|
static av_cold void exponent_init(AC3EncodeContext *s)
|
|
{
|
|
int i;
|
|
for (i = 73; i < 256; i++) {
|
|
exponent_group_tab[0][i] = (i - 1) / 3;
|
|
exponent_group_tab[1][i] = (i + 2) / 6;
|
|
exponent_group_tab[2][i] = (i + 8) / 12;
|
|
}
|
|
/* LFE */
|
|
exponent_group_tab[0][7] = 2;
|
|
}
|
|
|
|
|
|
/**
|
|
* Extract exponents from the MDCT coefficients.
|
|
* This takes into account the normalization that was done to the input samples
|
|
* by adjusting the exponents by the exponent shift values.
|
|
*/
|
|
static void extract_exponents(AC3EncodeContext *s)
|
|
{
|
|
int blk, ch, i;
|
|
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
AC3Block *block = &s->blocks[blk];
|
|
uint8_t *exp = block->exp[ch];
|
|
int32_t *coef = block->fixed_coef[ch];
|
|
for (i = 0; i < AC3_MAX_COEFS; i++) {
|
|
int e;
|
|
int v = abs(coef[i]);
|
|
if (v == 0)
|
|
e = 24;
|
|
else {
|
|
e = 23 - av_log2(v);
|
|
if (e >= 24) {
|
|
e = 24;
|
|
coef[i] = 0;
|
|
}
|
|
av_assert2(e >= 0);
|
|
}
|
|
exp[i] = e;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* Exponent Difference Threshold.
|
|
* New exponents are sent if their SAD exceed this number.
|
|
*/
|
|
#define EXP_DIFF_THRESHOLD 500
|
|
|
|
|
|
/**
|
|
* Calculate exponent strategies for all blocks in a single channel.
|
|
*/
|
|
static void compute_exp_strategy_ch(AC3EncodeContext *s, uint8_t *exp_strategy,
|
|
uint8_t *exp)
|
|
{
|
|
int blk, blk1;
|
|
int exp_diff;
|
|
|
|
/* estimate if the exponent variation & decide if they should be
|
|
reused in the next frame */
|
|
exp_strategy[0] = EXP_NEW;
|
|
exp += AC3_MAX_COEFS;
|
|
for (blk = 1; blk < AC3_MAX_BLOCKS; blk++) {
|
|
exp_diff = s->dsp.sad[0](NULL, exp, exp - AC3_MAX_COEFS, 16, 16);
|
|
if (exp_diff > EXP_DIFF_THRESHOLD)
|
|
exp_strategy[blk] = EXP_NEW;
|
|
else
|
|
exp_strategy[blk] = EXP_REUSE;
|
|
exp += AC3_MAX_COEFS;
|
|
}
|
|
|
|
/* now select the encoding strategy type : if exponents are often
|
|
recoded, we use a coarse encoding */
|
|
blk = 0;
|
|
while (blk < AC3_MAX_BLOCKS) {
|
|
blk1 = blk + 1;
|
|
while (blk1 < AC3_MAX_BLOCKS && exp_strategy[blk1] == EXP_REUSE)
|
|
blk1++;
|
|
switch (blk1 - blk) {
|
|
case 1: exp_strategy[blk] = EXP_D45; break;
|
|
case 2:
|
|
case 3: exp_strategy[blk] = EXP_D25; break;
|
|
default: exp_strategy[blk] = EXP_D15; break;
|
|
}
|
|
blk = blk1;
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* Calculate exponent strategies for all channels.
|
|
* Array arrangement is reversed to simplify the per-channel calculation.
|
|
*/
|
|
static void compute_exp_strategy(AC3EncodeContext *s)
|
|
{
|
|
int ch, blk;
|
|
|
|
for (ch = 0; ch < s->fbw_channels; ch++) {
|
|
compute_exp_strategy_ch(s, s->exp_strategy[ch], s->blocks[0].exp[ch]);
|
|
}
|
|
if (s->lfe_on) {
|
|
ch = s->lfe_channel;
|
|
s->exp_strategy[ch][0] = EXP_D15;
|
|
for (blk = 1; blk < AC3_MAX_BLOCKS; blk++)
|
|
s->exp_strategy[ch][blk] = EXP_REUSE;
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* Update the exponents so that they are the ones the decoder will decode.
|
|
*/
|
|
static void encode_exponents_blk_ch(uint8_t *exp, int nb_exps, int exp_strategy)
|
|
{
|
|
int nb_groups, i, k;
|
|
|
|
nb_groups = exponent_group_tab[exp_strategy-1][nb_exps] * 3;
|
|
|
|
/* for each group, compute the minimum exponent */
|
|
switch(exp_strategy) {
|
|
case EXP_D25:
|
|
for (i = 1, k = 1; i <= nb_groups; i++) {
|
|
uint8_t exp_min = exp[k];
|
|
if (exp[k+1] < exp_min)
|
|
exp_min = exp[k+1];
|
|
exp[i] = exp_min;
|
|
k += 2;
|
|
}
|
|
break;
|
|
case EXP_D45:
|
|
for (i = 1, k = 1; i <= nb_groups; i++) {
|
|
uint8_t exp_min = exp[k];
|
|
if (exp[k+1] < exp_min)
|
|
exp_min = exp[k+1];
|
|
if (exp[k+2] < exp_min)
|
|
exp_min = exp[k+2];
|
|
if (exp[k+3] < exp_min)
|
|
exp_min = exp[k+3];
|
|
exp[i] = exp_min;
|
|
k += 4;
|
|
}
|
|
break;
|
|
}
|
|
|
|
/* constraint for DC exponent */
|
|
if (exp[0] > 15)
|
|
exp[0] = 15;
|
|
|
|
/* decrease the delta between each groups to within 2 so that they can be
|
|
differentially encoded */
|
|
for (i = 1; i <= nb_groups; i++)
|
|
exp[i] = FFMIN(exp[i], exp[i-1] + 2);
|
|
i--;
|
|
while (--i >= 0)
|
|
exp[i] = FFMIN(exp[i], exp[i+1] + 2);
|
|
|
|
/* now we have the exponent values the decoder will see */
|
|
switch (exp_strategy) {
|
|
case EXP_D25:
|
|
for (i = nb_groups, k = nb_groups * 2; i > 0; i--) {
|
|
uint8_t exp1 = exp[i];
|
|
exp[k--] = exp1;
|
|
exp[k--] = exp1;
|
|
}
|
|
break;
|
|
case EXP_D45:
|
|
for (i = nb_groups, k = nb_groups * 4; i > 0; i--) {
|
|
exp[k] = exp[k-1] = exp[k-2] = exp[k-3] = exp[i];
|
|
k -= 4;
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* Encode exponents from original extracted form to what the decoder will see.
|
|
* This copies and groups exponents based on exponent strategy and reduces
|
|
* deltas between adjacent exponent groups so that they can be differentially
|
|
* encoded.
|
|
*/
|
|
static void encode_exponents(AC3EncodeContext *s)
|
|
{
|
|
int blk, blk1, ch;
|
|
uint8_t *exp, *exp_strategy;
|
|
int nb_coefs, num_reuse_blocks;
|
|
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
exp = s->blocks[0].exp[ch];
|
|
exp_strategy = s->exp_strategy[ch];
|
|
nb_coefs = s->nb_coefs[ch];
|
|
|
|
blk = 0;
|
|
while (blk < AC3_MAX_BLOCKS) {
|
|
blk1 = blk + 1;
|
|
|
|
/* count the number of EXP_REUSE blocks after the current block
|
|
and set exponent reference block pointers */
|
|
s->blocks[blk].exp_ref_block[ch] = &s->blocks[blk];
|
|
while (blk1 < AC3_MAX_BLOCKS && exp_strategy[blk1] == EXP_REUSE) {
|
|
s->blocks[blk1].exp_ref_block[ch] = &s->blocks[blk];
|
|
blk1++;
|
|
}
|
|
num_reuse_blocks = blk1 - blk - 1;
|
|
|
|
/* for the EXP_REUSE case we select the min of the exponents */
|
|
s->ac3dsp.ac3_exponent_min(exp, num_reuse_blocks, nb_coefs);
|
|
|
|
encode_exponents_blk_ch(exp, nb_coefs, exp_strategy[blk]);
|
|
|
|
exp += AC3_MAX_COEFS * (num_reuse_blocks + 1);
|
|
blk = blk1;
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* Group exponents.
|
|
* 3 delta-encoded exponents are in each 7-bit group. The number of groups
|
|
* varies depending on exponent strategy and bandwidth.
|
|
*/
|
|
static void group_exponents(AC3EncodeContext *s)
|
|
{
|
|
int blk, ch, i;
|
|
int group_size, nb_groups, bit_count;
|
|
uint8_t *p;
|
|
int delta0, delta1, delta2;
|
|
int exp0, exp1;
|
|
|
|
bit_count = 0;
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
AC3Block *block = &s->blocks[blk];
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
int exp_strategy = s->exp_strategy[ch][blk];
|
|
if (exp_strategy == EXP_REUSE)
|
|
continue;
|
|
group_size = exp_strategy + (exp_strategy == EXP_D45);
|
|
nb_groups = exponent_group_tab[exp_strategy-1][s->nb_coefs[ch]];
|
|
bit_count += 4 + (nb_groups * 7);
|
|
p = block->exp[ch];
|
|
|
|
/* DC exponent */
|
|
exp1 = *p++;
|
|
block->grouped_exp[ch][0] = exp1;
|
|
|
|
/* remaining exponents are delta encoded */
|
|
for (i = 1; i <= nb_groups; i++) {
|
|
/* merge three delta in one code */
|
|
exp0 = exp1;
|
|
exp1 = p[0];
|
|
p += group_size;
|
|
delta0 = exp1 - exp0 + 2;
|
|
av_assert2(delta0 >= 0 && delta0 <= 4);
|
|
|
|
exp0 = exp1;
|
|
exp1 = p[0];
|
|
p += group_size;
|
|
delta1 = exp1 - exp0 + 2;
|
|
av_assert2(delta1 >= 0 && delta1 <= 4);
|
|
|
|
exp0 = exp1;
|
|
exp1 = p[0];
|
|
p += group_size;
|
|
delta2 = exp1 - exp0 + 2;
|
|
av_assert2(delta2 >= 0 && delta2 <= 4);
|
|
|
|
block->grouped_exp[ch][i] = ((delta0 * 5 + delta1) * 5) + delta2;
|
|
}
|
|
}
|
|
}
|
|
|
|
s->exponent_bits = bit_count;
|
|
}
|
|
|
|
|
|
/**
|
|
* Calculate final exponents from the supplied MDCT coefficients and exponent shift.
|
|
* Extract exponents from MDCT coefficients, calculate exponent strategies,
|
|
* and encode final exponents.
|
|
*/
|
|
static void process_exponents(AC3EncodeContext *s)
|
|
{
|
|
extract_exponents(s);
|
|
|
|
compute_exp_strategy(s);
|
|
|
|
encode_exponents(s);
|
|
|
|
group_exponents(s);
|
|
|
|
emms_c();
|
|
}
|
|
|
|
|
|
/**
|
|
* Count frame bits that are based solely on fixed parameters.
|
|
* This only has to be run once when the encoder is initialized.
|
|
*/
|
|
static void count_frame_bits_fixed(AC3EncodeContext *s)
|
|
{
|
|
static const int frame_bits_inc[8] = { 0, 0, 2, 2, 2, 4, 2, 4 };
|
|
int blk;
|
|
int frame_bits;
|
|
|
|
/* assumptions:
|
|
* no dynamic range codes
|
|
* no channel coupling
|
|
* bit allocation parameters do not change between blocks
|
|
* SNR offsets do not change between blocks
|
|
* no delta bit allocation
|
|
* no skipped data
|
|
* no auxilliary data
|
|
*/
|
|
|
|
/* header size */
|
|
frame_bits = 65;
|
|
frame_bits += frame_bits_inc[s->channel_mode];
|
|
|
|
/* audio blocks */
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
frame_bits += s->fbw_channels * 2 + 2; /* blksw * c, dithflag * c, dynrnge, cplstre */
|
|
if (s->channel_mode == AC3_CHMODE_STEREO) {
|
|
frame_bits++; /* rematstr */
|
|
}
|
|
frame_bits += 2 * s->fbw_channels; /* chexpstr[2] * c */
|
|
if (s->lfe_on)
|
|
frame_bits++; /* lfeexpstr */
|
|
frame_bits++; /* baie */
|
|
frame_bits++; /* snr */
|
|
frame_bits += 2; /* delta / skip */
|
|
}
|
|
frame_bits++; /* cplinu for block 0 */
|
|
/* bit alloc info */
|
|
/* sdcycod[2], fdcycod[2], sgaincod[2], dbpbcod[2], floorcod[3] */
|
|
/* csnroffset[6] */
|
|
/* (fsnoffset[4] + fgaincod[4]) * c */
|
|
frame_bits += 2*4 + 3 + 6 + s->channels * (4 + 3);
|
|
|
|
/* auxdatae, crcrsv */
|
|
frame_bits += 2;
|
|
|
|
/* CRC */
|
|
frame_bits += 16;
|
|
|
|
s->frame_bits_fixed = frame_bits;
|
|
}
|
|
|
|
|
|
/**
|
|
* Initialize bit allocation.
|
|
* Set default parameter codes and calculate parameter values.
|
|
*/
|
|
static void bit_alloc_init(AC3EncodeContext *s)
|
|
{
|
|
int ch;
|
|
|
|
/* init default parameters */
|
|
s->slow_decay_code = 2;
|
|
s->fast_decay_code = 1;
|
|
s->slow_gain_code = 1;
|
|
s->db_per_bit_code = 3;
|
|
s->floor_code = 7;
|
|
for (ch = 0; ch < s->channels; ch++)
|
|
s->fast_gain_code[ch] = 4;
|
|
|
|
/* initial snr offset */
|
|
s->coarse_snr_offset = 40;
|
|
|
|
/* compute real values */
|
|
/* currently none of these values change during encoding, so we can just
|
|
set them once at initialization */
|
|
s->bit_alloc.slow_decay = ff_ac3_slow_decay_tab[s->slow_decay_code] >> s->bit_alloc.sr_shift;
|
|
s->bit_alloc.fast_decay = ff_ac3_fast_decay_tab[s->fast_decay_code] >> s->bit_alloc.sr_shift;
|
|
s->bit_alloc.slow_gain = ff_ac3_slow_gain_tab[s->slow_gain_code];
|
|
s->bit_alloc.db_per_bit = ff_ac3_db_per_bit_tab[s->db_per_bit_code];
|
|
s->bit_alloc.floor = ff_ac3_floor_tab[s->floor_code];
|
|
|
|
count_frame_bits_fixed(s);
|
|
}
|
|
|
|
|
|
/**
|
|
* Count the bits used to encode the frame, minus exponents and mantissas.
|
|
* Bits based on fixed parameters have already been counted, so now we just
|
|
* have to add the bits based on parameters that change during encoding.
|
|
*/
|
|
static void count_frame_bits(AC3EncodeContext *s)
|
|
{
|
|
AC3EncOptions *opt = &s->options;
|
|
int blk, ch;
|
|
int frame_bits = 0;
|
|
|
|
if (opt->audio_production_info)
|
|
frame_bits += 7;
|
|
if (s->bitstream_id == 6) {
|
|
if (opt->extended_bsi_1)
|
|
frame_bits += 14;
|
|
if (opt->extended_bsi_2)
|
|
frame_bits += 14;
|
|
}
|
|
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
/* stereo rematrixing */
|
|
if (s->channel_mode == AC3_CHMODE_STEREO &&
|
|
s->blocks[blk].new_rematrixing_strategy) {
|
|
frame_bits += s->num_rematrixing_bands;
|
|
}
|
|
|
|
for (ch = 0; ch < s->fbw_channels; ch++) {
|
|
if (s->exp_strategy[ch][blk] != EXP_REUSE)
|
|
frame_bits += 6 + 2; /* chbwcod[6], gainrng[2] */
|
|
}
|
|
}
|
|
s->frame_bits = s->frame_bits_fixed + frame_bits;
|
|
}
|
|
|
|
|
|
/**
|
|
* Finalize the mantissa bit count by adding in the grouped mantissas.
|
|
*/
|
|
static int compute_mantissa_size_final(int mant_cnt[5])
|
|
{
|
|
// bap=1 : 3 mantissas in 5 bits
|
|
int bits = (mant_cnt[1] / 3) * 5;
|
|
// bap=2 : 3 mantissas in 7 bits
|
|
// bap=4 : 2 mantissas in 7 bits
|
|
bits += ((mant_cnt[2] / 3) + (mant_cnt[4] >> 1)) * 7;
|
|
// bap=3 : each mantissa is 3 bits
|
|
bits += mant_cnt[3] * 3;
|
|
return bits;
|
|
}
|
|
|
|
|
|
/**
|
|
* Calculate masking curve based on the final exponents.
|
|
* Also calculate the power spectral densities to use in future calculations.
|
|
*/
|
|
static void bit_alloc_masking(AC3EncodeContext *s)
|
|
{
|
|
int blk, ch;
|
|
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
AC3Block *block = &s->blocks[blk];
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
/* We only need psd and mask for calculating bap.
|
|
Since we currently do not calculate bap when exponent
|
|
strategy is EXP_REUSE we do not need to calculate psd or mask. */
|
|
if (s->exp_strategy[ch][blk] != EXP_REUSE) {
|
|
ff_ac3_bit_alloc_calc_psd(block->exp[ch], 0,
|
|
s->nb_coefs[ch],
|
|
block->psd[ch], block->band_psd[ch]);
|
|
ff_ac3_bit_alloc_calc_mask(&s->bit_alloc, block->band_psd[ch],
|
|
0, s->nb_coefs[ch],
|
|
ff_ac3_fast_gain_tab[s->fast_gain_code[ch]],
|
|
ch == s->lfe_channel,
|
|
DBA_NONE, 0, NULL, NULL, NULL,
|
|
block->mask[ch]);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* Ensure that bap for each block and channel point to the current bap_buffer.
|
|
* They may have been switched during the bit allocation search.
|
|
*/
|
|
static void reset_block_bap(AC3EncodeContext *s)
|
|
{
|
|
int blk, ch;
|
|
if (s->blocks[0].bap[0] == s->bap_buffer)
|
|
return;
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
s->blocks[blk].bap[ch] = &s->bap_buffer[AC3_MAX_COEFS * (blk * s->channels + ch)];
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* Run the bit allocation with a given SNR offset.
|
|
* This calculates the bit allocation pointers that will be used to determine
|
|
* the quantization of each mantissa.
|
|
* @return the number of bits needed for mantissas if the given SNR offset is
|
|
* is used.
|
|
*/
|
|
static int bit_alloc(AC3EncodeContext *s, int snr_offset)
|
|
{
|
|
int blk, ch;
|
|
int mantissa_bits;
|
|
int mant_cnt[5];
|
|
|
|
snr_offset = (snr_offset - 240) << 2;
|
|
|
|
reset_block_bap(s);
|
|
mantissa_bits = 0;
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
AC3Block *block;
|
|
// initialize grouped mantissa counts. these are set so that they are
|
|
// padded to the next whole group size when bits are counted in
|
|
// compute_mantissa_size_final
|
|
mant_cnt[0] = mant_cnt[3] = 0;
|
|
mant_cnt[1] = mant_cnt[2] = 2;
|
|
mant_cnt[4] = 1;
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
/* Currently the only bit allocation parameters which vary across
|
|
blocks within a frame are the exponent values. We can take
|
|
advantage of that by reusing the bit allocation pointers
|
|
whenever we reuse exponents. */
|
|
block = s->blocks[blk].exp_ref_block[ch];
|
|
if (s->exp_strategy[ch][blk] != EXP_REUSE) {
|
|
s->ac3dsp.bit_alloc_calc_bap(block->mask[ch], block->psd[ch], 0,
|
|
s->nb_coefs[ch], snr_offset,
|
|
s->bit_alloc.floor, ff_ac3_bap_tab,
|
|
block->bap[ch]);
|
|
}
|
|
mantissa_bits += s->ac3dsp.compute_mantissa_size(mant_cnt, block->bap[ch], s->nb_coefs[ch]);
|
|
}
|
|
mantissa_bits += compute_mantissa_size_final(mant_cnt);
|
|
}
|
|
return mantissa_bits;
|
|
}
|
|
|
|
|
|
/**
|
|
* Constant bitrate bit allocation search.
|
|
* Find the largest SNR offset that will allow data to fit in the frame.
|
|
*/
|
|
static int cbr_bit_allocation(AC3EncodeContext *s)
|
|
{
|
|
int ch;
|
|
int bits_left;
|
|
int snr_offset, snr_incr;
|
|
|
|
bits_left = 8 * s->frame_size - (s->frame_bits + s->exponent_bits);
|
|
av_assert2(bits_left >= 0);
|
|
|
|
snr_offset = s->coarse_snr_offset << 4;
|
|
|
|
/* if previous frame SNR offset was 1023, check if current frame can also
|
|
use SNR offset of 1023. if so, skip the search. */
|
|
if ((snr_offset | s->fine_snr_offset[0]) == 1023) {
|
|
if (bit_alloc(s, 1023) <= bits_left)
|
|
return 0;
|
|
}
|
|
|
|
while (snr_offset >= 0 &&
|
|
bit_alloc(s, snr_offset) > bits_left) {
|
|
snr_offset -= 64;
|
|
}
|
|
if (snr_offset < 0)
|
|
return AVERROR(EINVAL);
|
|
|
|
FFSWAP(uint8_t *, s->bap_buffer, s->bap1_buffer);
|
|
for (snr_incr = 64; snr_incr > 0; snr_incr >>= 2) {
|
|
while (snr_offset + snr_incr <= 1023 &&
|
|
bit_alloc(s, snr_offset + snr_incr) <= bits_left) {
|
|
snr_offset += snr_incr;
|
|
FFSWAP(uint8_t *, s->bap_buffer, s->bap1_buffer);
|
|
}
|
|
}
|
|
FFSWAP(uint8_t *, s->bap_buffer, s->bap1_buffer);
|
|
reset_block_bap(s);
|
|
|
|
s->coarse_snr_offset = snr_offset >> 4;
|
|
for (ch = 0; ch < s->channels; ch++)
|
|
s->fine_snr_offset[ch] = snr_offset & 0xF;
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
/**
|
|
* Downgrade exponent strategies to reduce the bits used by the exponents.
|
|
* This is a fallback for when bit allocation fails with the normal exponent
|
|
* strategies. Each time this function is run it only downgrades the
|
|
* strategy in 1 channel of 1 block.
|
|
* @return non-zero if downgrade was unsuccessful
|
|
*/
|
|
static int downgrade_exponents(AC3EncodeContext *s)
|
|
{
|
|
int ch, blk;
|
|
|
|
for (ch = 0; ch < s->fbw_channels; ch++) {
|
|
for (blk = AC3_MAX_BLOCKS-1; blk >= 0; blk--) {
|
|
if (s->exp_strategy[ch][blk] == EXP_D15) {
|
|
s->exp_strategy[ch][blk] = EXP_D25;
|
|
return 0;
|
|
}
|
|
}
|
|
}
|
|
for (ch = 0; ch < s->fbw_channels; ch++) {
|
|
for (blk = AC3_MAX_BLOCKS-1; blk >= 0; blk--) {
|
|
if (s->exp_strategy[ch][blk] == EXP_D25) {
|
|
s->exp_strategy[ch][blk] = EXP_D45;
|
|
return 0;
|
|
}
|
|
}
|
|
}
|
|
for (ch = 0; ch < s->fbw_channels; ch++) {
|
|
/* block 0 cannot reuse exponents, so only downgrade D45 to REUSE if
|
|
the block number > 0 */
|
|
for (blk = AC3_MAX_BLOCKS-1; blk > 0; blk--) {
|
|
if (s->exp_strategy[ch][blk] > EXP_REUSE) {
|
|
s->exp_strategy[ch][blk] = EXP_REUSE;
|
|
return 0;
|
|
}
|
|
}
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
|
|
/**
|
|
* Reduce the bandwidth to reduce the number of bits used for a given SNR offset.
|
|
* This is a second fallback for when bit allocation still fails after exponents
|
|
* have been downgraded.
|
|
* @return non-zero if bandwidth reduction was unsuccessful
|
|
*/
|
|
static int reduce_bandwidth(AC3EncodeContext *s, int min_bw_code)
|
|
{
|
|
int ch;
|
|
|
|
if (s->bandwidth_code[0] > min_bw_code) {
|
|
for (ch = 0; ch < s->fbw_channels; ch++) {
|
|
s->bandwidth_code[ch]--;
|
|
s->nb_coefs[ch] = s->bandwidth_code[ch] * 3 + 73;
|
|
}
|
|
return 0;
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
|
|
/**
|
|
* Perform bit allocation search.
|
|
* Finds the SNR offset value that maximizes quality and fits in the specified
|
|
* frame size. Output is the SNR offset and a set of bit allocation pointers
|
|
* used to quantize the mantissas.
|
|
*/
|
|
static int compute_bit_allocation(AC3EncodeContext *s)
|
|
{
|
|
int ret;
|
|
|
|
count_frame_bits(s);
|
|
|
|
bit_alloc_masking(s);
|
|
|
|
ret = cbr_bit_allocation(s);
|
|
while (ret) {
|
|
/* fallback 1: downgrade exponents */
|
|
if (!downgrade_exponents(s)) {
|
|
extract_exponents(s);
|
|
encode_exponents(s);
|
|
group_exponents(s);
|
|
ret = compute_bit_allocation(s);
|
|
continue;
|
|
}
|
|
|
|
/* fallback 2: reduce bandwidth */
|
|
/* only do this if the user has not specified a specific cutoff
|
|
frequency */
|
|
if (!s->cutoff && !reduce_bandwidth(s, 0)) {
|
|
process_exponents(s);
|
|
ret = compute_bit_allocation(s);
|
|
continue;
|
|
}
|
|
|
|
/* fallbacks were not enough... */
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
|
|
/**
|
|
* Symmetric quantization on 'levels' levels.
|
|
*/
|
|
static inline int sym_quant(int c, int e, int levels)
|
|
{
|
|
int v = (((levels * c) >> (24 - e)) + levels) >> 1;
|
|
av_assert2(v >= 0 && v < levels);
|
|
return v;
|
|
}
|
|
|
|
|
|
/**
|
|
* Asymmetric quantization on 2^qbits levels.
|
|
*/
|
|
static inline int asym_quant(int c, int e, int qbits)
|
|
{
|
|
int lshift, m, v;
|
|
|
|
lshift = e + qbits - 24;
|
|
if (lshift >= 0)
|
|
v = c << lshift;
|
|
else
|
|
v = c >> (-lshift);
|
|
/* rounding */
|
|
v = (v + 1) >> 1;
|
|
m = (1 << (qbits-1));
|
|
if (v >= m)
|
|
v = m - 1;
|
|
av_assert2(v >= -m);
|
|
return v & ((1 << qbits)-1);
|
|
}
|
|
|
|
|
|
/**
|
|
* Quantize a set of mantissas for a single channel in a single block.
|
|
*/
|
|
static void quantize_mantissas_blk_ch(AC3Mant *s, int32_t *fixed_coef,
|
|
uint8_t *exp,
|
|
uint8_t *bap, uint16_t *qmant, int n)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < n; i++) {
|
|
int v;
|
|
int c = fixed_coef[i];
|
|
int e = exp[i];
|
|
int b = bap[i];
|
|
switch (b) {
|
|
case 0:
|
|
v = 0;
|
|
break;
|
|
case 1:
|
|
v = sym_quant(c, e, 3);
|
|
switch (s->mant1_cnt) {
|
|
case 0:
|
|
s->qmant1_ptr = &qmant[i];
|
|
v = 9 * v;
|
|
s->mant1_cnt = 1;
|
|
break;
|
|
case 1:
|
|
*s->qmant1_ptr += 3 * v;
|
|
s->mant1_cnt = 2;
|
|
v = 128;
|
|
break;
|
|
default:
|
|
*s->qmant1_ptr += v;
|
|
s->mant1_cnt = 0;
|
|
v = 128;
|
|
break;
|
|
}
|
|
break;
|
|
case 2:
|
|
v = sym_quant(c, e, 5);
|
|
switch (s->mant2_cnt) {
|
|
case 0:
|
|
s->qmant2_ptr = &qmant[i];
|
|
v = 25 * v;
|
|
s->mant2_cnt = 1;
|
|
break;
|
|
case 1:
|
|
*s->qmant2_ptr += 5 * v;
|
|
s->mant2_cnt = 2;
|
|
v = 128;
|
|
break;
|
|
default:
|
|
*s->qmant2_ptr += v;
|
|
s->mant2_cnt = 0;
|
|
v = 128;
|
|
break;
|
|
}
|
|
break;
|
|
case 3:
|
|
v = sym_quant(c, e, 7);
|
|
break;
|
|
case 4:
|
|
v = sym_quant(c, e, 11);
|
|
switch (s->mant4_cnt) {
|
|
case 0:
|
|
s->qmant4_ptr = &qmant[i];
|
|
v = 11 * v;
|
|
s->mant4_cnt = 1;
|
|
break;
|
|
default:
|
|
*s->qmant4_ptr += v;
|
|
s->mant4_cnt = 0;
|
|
v = 128;
|
|
break;
|
|
}
|
|
break;
|
|
case 5:
|
|
v = sym_quant(c, e, 15);
|
|
break;
|
|
case 14:
|
|
v = asym_quant(c, e, 14);
|
|
break;
|
|
case 15:
|
|
v = asym_quant(c, e, 16);
|
|
break;
|
|
default:
|
|
v = asym_quant(c, e, b - 1);
|
|
break;
|
|
}
|
|
qmant[i] = v;
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* Quantize mantissas using coefficients, exponents, and bit allocation pointers.
|
|
*/
|
|
static void quantize_mantissas(AC3EncodeContext *s)
|
|
{
|
|
int blk, ch;
|
|
|
|
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
AC3Block *block = &s->blocks[blk];
|
|
AC3Block *ref_block;
|
|
AC3Mant m = { 0 };
|
|
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
ref_block = block->exp_ref_block[ch];
|
|
quantize_mantissas_blk_ch(&m, block->fixed_coef[ch],
|
|
ref_block->exp[ch], ref_block->bap[ch],
|
|
block->qmant[ch], s->nb_coefs[ch]);
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* Write the AC-3 frame header to the output bitstream.
|
|
*/
|
|
static void output_frame_header(AC3EncodeContext *s)
|
|
{
|
|
AC3EncOptions *opt = &s->options;
|
|
|
|
put_bits(&s->pb, 16, 0x0b77); /* frame header */
|
|
put_bits(&s->pb, 16, 0); /* crc1: will be filled later */
|
|
put_bits(&s->pb, 2, s->bit_alloc.sr_code);
|
|
put_bits(&s->pb, 6, s->frame_size_code + (s->frame_size - s->frame_size_min) / 2);
|
|
put_bits(&s->pb, 5, s->bitstream_id);
|
|
put_bits(&s->pb, 3, s->bitstream_mode);
|
|
put_bits(&s->pb, 3, s->channel_mode);
|
|
if ((s->channel_mode & 0x01) && s->channel_mode != AC3_CHMODE_MONO)
|
|
put_bits(&s->pb, 2, s->center_mix_level);
|
|
if (s->channel_mode & 0x04)
|
|
put_bits(&s->pb, 2, s->surround_mix_level);
|
|
if (s->channel_mode == AC3_CHMODE_STEREO)
|
|
put_bits(&s->pb, 2, opt->dolby_surround_mode);
|
|
put_bits(&s->pb, 1, s->lfe_on); /* LFE */
|
|
put_bits(&s->pb, 5, -opt->dialogue_level);
|
|
put_bits(&s->pb, 1, 0); /* no compression control word */
|
|
put_bits(&s->pb, 1, 0); /* no lang code */
|
|
put_bits(&s->pb, 1, opt->audio_production_info);
|
|
if (opt->audio_production_info) {
|
|
put_bits(&s->pb, 5, opt->mixing_level - 80);
|
|
put_bits(&s->pb, 2, opt->room_type);
|
|
}
|
|
put_bits(&s->pb, 1, opt->copyright);
|
|
put_bits(&s->pb, 1, opt->original);
|
|
if (s->bitstream_id == 6) {
|
|
/* alternate bit stream syntax */
|
|
put_bits(&s->pb, 1, opt->extended_bsi_1);
|
|
if (opt->extended_bsi_1) {
|
|
put_bits(&s->pb, 2, opt->preferred_stereo_downmix);
|
|
put_bits(&s->pb, 3, s->ltrt_center_mix_level);
|
|
put_bits(&s->pb, 3, s->ltrt_surround_mix_level);
|
|
put_bits(&s->pb, 3, s->loro_center_mix_level);
|
|
put_bits(&s->pb, 3, s->loro_surround_mix_level);
|
|
}
|
|
put_bits(&s->pb, 1, opt->extended_bsi_2);
|
|
if (opt->extended_bsi_2) {
|
|
put_bits(&s->pb, 2, opt->dolby_surround_ex_mode);
|
|
put_bits(&s->pb, 2, opt->dolby_headphone_mode);
|
|
put_bits(&s->pb, 1, opt->ad_converter_type);
|
|
put_bits(&s->pb, 9, 0); /* xbsi2 and encinfo : reserved */
|
|
}
|
|
} else {
|
|
put_bits(&s->pb, 1, 0); /* no time code 1 */
|
|
put_bits(&s->pb, 1, 0); /* no time code 2 */
|
|
}
|
|
put_bits(&s->pb, 1, 0); /* no additional bit stream info */
|
|
}
|
|
|
|
|
|
/**
|
|
* Write one audio block to the output bitstream.
|
|
*/
|
|
static void output_audio_block(AC3EncodeContext *s, int blk)
|
|
{
|
|
int ch, i, baie, rbnd;
|
|
AC3Block *block = &s->blocks[blk];
|
|
|
|
/* block switching */
|
|
for (ch = 0; ch < s->fbw_channels; ch++)
|
|
put_bits(&s->pb, 1, 0);
|
|
|
|
/* dither flags */
|
|
for (ch = 0; ch < s->fbw_channels; ch++)
|
|
put_bits(&s->pb, 1, 1);
|
|
|
|
/* dynamic range codes */
|
|
put_bits(&s->pb, 1, 0);
|
|
|
|
/* channel coupling */
|
|
if (!blk) {
|
|
put_bits(&s->pb, 1, 1); /* coupling strategy present */
|
|
put_bits(&s->pb, 1, 0); /* no coupling strategy */
|
|
} else {
|
|
put_bits(&s->pb, 1, 0); /* no new coupling strategy */
|
|
}
|
|
|
|
/* stereo rematrixing */
|
|
if (s->channel_mode == AC3_CHMODE_STEREO) {
|
|
put_bits(&s->pb, 1, block->new_rematrixing_strategy);
|
|
if (block->new_rematrixing_strategy) {
|
|
/* rematrixing flags */
|
|
for (rbnd = 0; rbnd < s->num_rematrixing_bands; rbnd++)
|
|
put_bits(&s->pb, 1, block->rematrixing_flags[rbnd]);
|
|
}
|
|
}
|
|
|
|
/* exponent strategy */
|
|
for (ch = 0; ch < s->fbw_channels; ch++)
|
|
put_bits(&s->pb, 2, s->exp_strategy[ch][blk]);
|
|
if (s->lfe_on)
|
|
put_bits(&s->pb, 1, s->exp_strategy[s->lfe_channel][blk]);
|
|
|
|
/* bandwidth */
|
|
for (ch = 0; ch < s->fbw_channels; ch++) {
|
|
if (s->exp_strategy[ch][blk] != EXP_REUSE)
|
|
put_bits(&s->pb, 6, s->bandwidth_code[ch]);
|
|
}
|
|
|
|
/* exponents */
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
int nb_groups;
|
|
|
|
if (s->exp_strategy[ch][blk] == EXP_REUSE)
|
|
continue;
|
|
|
|
/* DC exponent */
|
|
put_bits(&s->pb, 4, block->grouped_exp[ch][0]);
|
|
|
|
/* exponent groups */
|
|
nb_groups = exponent_group_tab[s->exp_strategy[ch][blk]-1][s->nb_coefs[ch]];
|
|
for (i = 1; i <= nb_groups; i++)
|
|
put_bits(&s->pb, 7, block->grouped_exp[ch][i]);
|
|
|
|
/* gain range info */
|
|
if (ch != s->lfe_channel)
|
|
put_bits(&s->pb, 2, 0);
|
|
}
|
|
|
|
/* bit allocation info */
|
|
baie = (blk == 0);
|
|
put_bits(&s->pb, 1, baie);
|
|
if (baie) {
|
|
put_bits(&s->pb, 2, s->slow_decay_code);
|
|
put_bits(&s->pb, 2, s->fast_decay_code);
|
|
put_bits(&s->pb, 2, s->slow_gain_code);
|
|
put_bits(&s->pb, 2, s->db_per_bit_code);
|
|
put_bits(&s->pb, 3, s->floor_code);
|
|
}
|
|
|
|
/* snr offset */
|
|
put_bits(&s->pb, 1, baie);
|
|
if (baie) {
|
|
put_bits(&s->pb, 6, s->coarse_snr_offset);
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
put_bits(&s->pb, 4, s->fine_snr_offset[ch]);
|
|
put_bits(&s->pb, 3, s->fast_gain_code[ch]);
|
|
}
|
|
}
|
|
|
|
put_bits(&s->pb, 1, 0); /* no delta bit allocation */
|
|
put_bits(&s->pb, 1, 0); /* no data to skip */
|
|
|
|
/* mantissas */
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
int b, q;
|
|
AC3Block *ref_block = block->exp_ref_block[ch];
|
|
for (i = 0; i < s->nb_coefs[ch]; i++) {
|
|
q = block->qmant[ch][i];
|
|
b = ref_block->bap[ch][i];
|
|
switch (b) {
|
|
case 0: break;
|
|
case 1: if (q != 128) put_bits(&s->pb, 5, q); break;
|
|
case 2: if (q != 128) put_bits(&s->pb, 7, q); break;
|
|
case 3: put_bits(&s->pb, 3, q); break;
|
|
case 4: if (q != 128) put_bits(&s->pb, 7, q); break;
|
|
case 14: put_bits(&s->pb, 14, q); break;
|
|
case 15: put_bits(&s->pb, 16, q); break;
|
|
default: put_bits(&s->pb, b-1, q); break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
/** CRC-16 Polynomial */
|
|
#define CRC16_POLY ((1 << 0) | (1 << 2) | (1 << 15) | (1 << 16))
|
|
|
|
|
|
static unsigned int mul_poly(unsigned int a, unsigned int b, unsigned int poly)
|
|
{
|
|
unsigned int c;
|
|
|
|
c = 0;
|
|
while (a) {
|
|
if (a & 1)
|
|
c ^= b;
|
|
a = a >> 1;
|
|
b = b << 1;
|
|
if (b & (1 << 16))
|
|
b ^= poly;
|
|
}
|
|
return c;
|
|
}
|
|
|
|
|
|
static unsigned int pow_poly(unsigned int a, unsigned int n, unsigned int poly)
|
|
{
|
|
unsigned int r;
|
|
r = 1;
|
|
while (n) {
|
|
if (n & 1)
|
|
r = mul_poly(r, a, poly);
|
|
a = mul_poly(a, a, poly);
|
|
n >>= 1;
|
|
}
|
|
return r;
|
|
}
|
|
|
|
|
|
/**
|
|
* Fill the end of the frame with 0's and compute the two CRCs.
|
|
*/
|
|
static void output_frame_end(AC3EncodeContext *s)
|
|
{
|
|
const AVCRC *crc_ctx = av_crc_get_table(AV_CRC_16_ANSI);
|
|
int frame_size_58, pad_bytes, crc1, crc2_partial, crc2, crc_inv;
|
|
uint8_t *frame;
|
|
|
|
frame_size_58 = ((s->frame_size >> 2) + (s->frame_size >> 4)) << 1;
|
|
|
|
/* pad the remainder of the frame with zeros */
|
|
av_assert2(s->frame_size * 8 - put_bits_count(&s->pb) >= 18);
|
|
flush_put_bits(&s->pb);
|
|
frame = s->pb.buf;
|
|
pad_bytes = s->frame_size - (put_bits_ptr(&s->pb) - frame) - 2;
|
|
av_assert2(pad_bytes >= 0);
|
|
if (pad_bytes > 0)
|
|
memset(put_bits_ptr(&s->pb), 0, pad_bytes);
|
|
|
|
/* compute crc1 */
|
|
/* this is not so easy because it is at the beginning of the data... */
|
|
crc1 = av_bswap16(av_crc(crc_ctx, 0, frame + 4, frame_size_58 - 4));
|
|
crc_inv = s->crc_inv[s->frame_size > s->frame_size_min];
|
|
crc1 = mul_poly(crc_inv, crc1, CRC16_POLY);
|
|
AV_WB16(frame + 2, crc1);
|
|
|
|
/* compute crc2 */
|
|
crc2_partial = av_crc(crc_ctx, 0, frame + frame_size_58,
|
|
s->frame_size - frame_size_58 - 3);
|
|
crc2 = av_crc(crc_ctx, crc2_partial, frame + s->frame_size - 3, 1);
|
|
/* ensure crc2 does not match sync word by flipping crcrsv bit if needed */
|
|
if (crc2 == 0x770B) {
|
|
frame[s->frame_size - 3] ^= 0x1;
|
|
crc2 = av_crc(crc_ctx, crc2_partial, frame + s->frame_size - 3, 1);
|
|
}
|
|
crc2 = av_bswap16(crc2);
|
|
AV_WB16(frame + s->frame_size - 2, crc2);
|
|
}
|
|
|
|
|
|
/**
|
|
* Write the frame to the output bitstream.
|
|
*/
|
|
static void output_frame(AC3EncodeContext *s, unsigned char *frame)
|
|
{
|
|
int blk;
|
|
|
|
init_put_bits(&s->pb, frame, AC3_MAX_CODED_FRAME_SIZE);
|
|
|
|
output_frame_header(s);
|
|
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++)
|
|
output_audio_block(s, blk);
|
|
|
|
output_frame_end(s);
|
|
}
|
|
|
|
|
|
static void dprint_options(AVCodecContext *avctx)
|
|
{
|
|
#ifdef DEBUG
|
|
AC3EncodeContext *s = avctx->priv_data;
|
|
AC3EncOptions *opt = &s->options;
|
|
char strbuf[32];
|
|
|
|
switch (s->bitstream_id) {
|
|
case 6: strncpy(strbuf, "AC-3 (alt syntax)", 32); break;
|
|
case 8: strncpy(strbuf, "AC-3 (standard)", 32); break;
|
|
case 9: strncpy(strbuf, "AC-3 (dnet half-rate)", 32); break;
|
|
case 10: strncpy(strbuf, "AC-3 (dnet quater-rate", 32); break;
|
|
default: snprintf(strbuf, 32, "ERROR");
|
|
}
|
|
av_dlog(avctx, "bitstream_id: %s (%d)\n", strbuf, s->bitstream_id);
|
|
av_dlog(avctx, "sample_fmt: %s\n", av_get_sample_fmt_name(avctx->sample_fmt));
|
|
av_get_channel_layout_string(strbuf, 32, s->channels, avctx->channel_layout);
|
|
av_dlog(avctx, "channel_layout: %s\n", strbuf);
|
|
av_dlog(avctx, "sample_rate: %d\n", s->sample_rate);
|
|
av_dlog(avctx, "bit_rate: %d\n", s->bit_rate);
|
|
if (s->cutoff)
|
|
av_dlog(avctx, "cutoff: %d\n", s->cutoff);
|
|
|
|
av_dlog(avctx, "per_frame_metadata: %s\n",
|
|
opt->allow_per_frame_metadata?"on":"off");
|
|
if (s->has_center)
|
|
av_dlog(avctx, "center_mixlev: %0.3f (%d)\n", opt->center_mix_level,
|
|
s->center_mix_level);
|
|
else
|
|
av_dlog(avctx, "center_mixlev: {not written}\n");
|
|
if (s->has_surround)
|
|
av_dlog(avctx, "surround_mixlev: %0.3f (%d)\n", opt->surround_mix_level,
|
|
s->surround_mix_level);
|
|
else
|
|
av_dlog(avctx, "surround_mixlev: {not written}\n");
|
|
if (opt->audio_production_info) {
|
|
av_dlog(avctx, "mixing_level: %ddB\n", opt->mixing_level);
|
|
switch (opt->room_type) {
|
|
case 0: strncpy(strbuf, "notindicated", 32); break;
|
|
case 1: strncpy(strbuf, "large", 32); break;
|
|
case 2: strncpy(strbuf, "small", 32); break;
|
|
default: snprintf(strbuf, 32, "ERROR (%d)", opt->room_type);
|
|
}
|
|
av_dlog(avctx, "room_type: %s\n", strbuf);
|
|
} else {
|
|
av_dlog(avctx, "mixing_level: {not written}\n");
|
|
av_dlog(avctx, "room_type: {not written}\n");
|
|
}
|
|
av_dlog(avctx, "copyright: %s\n", opt->copyright?"on":"off");
|
|
av_dlog(avctx, "dialnorm: %ddB\n", opt->dialogue_level);
|
|
if (s->channel_mode == AC3_CHMODE_STEREO) {
|
|
switch (opt->dolby_surround_mode) {
|
|
case 0: strncpy(strbuf, "notindicated", 32); break;
|
|
case 1: strncpy(strbuf, "on", 32); break;
|
|
case 2: strncpy(strbuf, "off", 32); break;
|
|
default: snprintf(strbuf, 32, "ERROR (%d)", opt->dolby_surround_mode);
|
|
}
|
|
av_dlog(avctx, "dsur_mode: %s\n", strbuf);
|
|
} else {
|
|
av_dlog(avctx, "dsur_mode: {not written}\n");
|
|
}
|
|
av_dlog(avctx, "original: %s\n", opt->original?"on":"off");
|
|
|
|
if (s->bitstream_id == 6) {
|
|
if (opt->extended_bsi_1) {
|
|
switch (opt->preferred_stereo_downmix) {
|
|
case 0: strncpy(strbuf, "notindicated", 32); break;
|
|
case 1: strncpy(strbuf, "ltrt", 32); break;
|
|
case 2: strncpy(strbuf, "loro", 32); break;
|
|
default: snprintf(strbuf, 32, "ERROR (%d)", opt->preferred_stereo_downmix);
|
|
}
|
|
av_dlog(avctx, "dmix_mode: %s\n", strbuf);
|
|
av_dlog(avctx, "ltrt_cmixlev: %0.3f (%d)\n",
|
|
opt->ltrt_center_mix_level, s->ltrt_center_mix_level);
|
|
av_dlog(avctx, "ltrt_surmixlev: %0.3f (%d)\n",
|
|
opt->ltrt_surround_mix_level, s->ltrt_surround_mix_level);
|
|
av_dlog(avctx, "loro_cmixlev: %0.3f (%d)\n",
|
|
opt->loro_center_mix_level, s->loro_center_mix_level);
|
|
av_dlog(avctx, "loro_surmixlev: %0.3f (%d)\n",
|
|
opt->loro_surround_mix_level, s->loro_surround_mix_level);
|
|
} else {
|
|
av_dlog(avctx, "extended bitstream info 1: {not written}\n");
|
|
}
|
|
if (opt->extended_bsi_2) {
|
|
switch (opt->dolby_surround_ex_mode) {
|
|
case 0: strncpy(strbuf, "notindicated", 32); break;
|
|
case 1: strncpy(strbuf, "on", 32); break;
|
|
case 2: strncpy(strbuf, "off", 32); break;
|
|
default: snprintf(strbuf, 32, "ERROR (%d)", opt->dolby_surround_ex_mode);
|
|
}
|
|
av_dlog(avctx, "dsurex_mode: %s\n", strbuf);
|
|
switch (opt->dolby_headphone_mode) {
|
|
case 0: strncpy(strbuf, "notindicated", 32); break;
|
|
case 1: strncpy(strbuf, "on", 32); break;
|
|
case 2: strncpy(strbuf, "off", 32); break;
|
|
default: snprintf(strbuf, 32, "ERROR (%d)", opt->dolby_headphone_mode);
|
|
}
|
|
av_dlog(avctx, "dheadphone_mode: %s\n", strbuf);
|
|
|
|
switch (opt->ad_converter_type) {
|
|
case 0: strncpy(strbuf, "standard", 32); break;
|
|
case 1: strncpy(strbuf, "hdcd", 32); break;
|
|
default: snprintf(strbuf, 32, "ERROR (%d)", opt->ad_converter_type);
|
|
}
|
|
av_dlog(avctx, "ad_conv_type: %s\n", strbuf);
|
|
} else {
|
|
av_dlog(avctx, "extended bitstream info 2: {not written}\n");
|
|
}
|
|
}
|
|
#endif
|
|
}
|
|
|
|
|
|
#define FLT_OPTION_THRESHOLD 0.01
|
|
|
|
static int validate_float_option(float v, const float *v_list, int v_list_size)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < v_list_size; i++) {
|
|
if (v < (v_list[i] + FLT_OPTION_THRESHOLD) &&
|
|
v > (v_list[i] - FLT_OPTION_THRESHOLD))
|
|
break;
|
|
}
|
|
if (i == v_list_size)
|
|
return -1;
|
|
|
|
return i;
|
|
}
|
|
|
|
|
|
static void validate_mix_level(void *log_ctx, const char *opt_name,
|
|
float *opt_param, const float *list,
|
|
int list_size, int default_value, int min_value,
|
|
int *ctx_param)
|
|
{
|
|
int mixlev = validate_float_option(*opt_param, list, list_size);
|
|
if (mixlev < min_value) {
|
|
mixlev = default_value;
|
|
if (*opt_param >= 0.0) {
|
|
av_log(log_ctx, AV_LOG_WARNING, "requested %s is not valid. using "
|
|
"default value: %0.3f\n", opt_name, list[mixlev]);
|
|
}
|
|
}
|
|
*opt_param = list[mixlev];
|
|
*ctx_param = mixlev;
|
|
}
|
|
|
|
|
|
/**
|
|
* Validate metadata options as set by AVOption system.
|
|
* These values can optionally be changed per-frame.
|
|
*/
|
|
static int validate_metadata(AVCodecContext *avctx)
|
|
{
|
|
AC3EncodeContext *s = avctx->priv_data;
|
|
AC3EncOptions *opt = &s->options;
|
|
|
|
/* validate mixing levels */
|
|
if (s->has_center) {
|
|
validate_mix_level(avctx, "center_mix_level", &opt->center_mix_level,
|
|
cmixlev_options, CMIXLEV_NUM_OPTIONS, 1, 0,
|
|
&s->center_mix_level);
|
|
}
|
|
if (s->has_surround) {
|
|
validate_mix_level(avctx, "surround_mix_level", &opt->surround_mix_level,
|
|
surmixlev_options, SURMIXLEV_NUM_OPTIONS, 1, 0,
|
|
&s->surround_mix_level);
|
|
}
|
|
|
|
/* set audio production info flag */
|
|
if (opt->mixing_level >= 0 || opt->room_type >= 0) {
|
|
if (opt->mixing_level < 0) {
|
|
av_log(avctx, AV_LOG_ERROR, "mixing_level must be set if "
|
|
"room_type is set\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
if (opt->mixing_level < 80) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid mixing level. must be between "
|
|
"80dB and 111dB\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
/* default room type */
|
|
if (opt->room_type < 0)
|
|
opt->room_type = 0;
|
|
opt->audio_production_info = 1;
|
|
} else {
|
|
opt->audio_production_info = 0;
|
|
}
|
|
|
|
/* set extended bsi 1 flag */
|
|
if ((s->has_center || s->has_surround) &&
|
|
(opt->preferred_stereo_downmix >= 0 ||
|
|
opt->ltrt_center_mix_level >= 0 ||
|
|
opt->ltrt_surround_mix_level >= 0 ||
|
|
opt->loro_center_mix_level >= 0 ||
|
|
opt->loro_surround_mix_level >= 0)) {
|
|
/* default preferred stereo downmix */
|
|
if (opt->preferred_stereo_downmix < 0)
|
|
opt->preferred_stereo_downmix = 0;
|
|
/* validate Lt/Rt center mix level */
|
|
validate_mix_level(avctx, "ltrt_center_mix_level",
|
|
&opt->ltrt_center_mix_level, extmixlev_options,
|
|
EXTMIXLEV_NUM_OPTIONS, 5, 0,
|
|
&s->ltrt_center_mix_level);
|
|
/* validate Lt/Rt surround mix level */
|
|
validate_mix_level(avctx, "ltrt_surround_mix_level",
|
|
&opt->ltrt_surround_mix_level, extmixlev_options,
|
|
EXTMIXLEV_NUM_OPTIONS, 6, 3,
|
|
&s->ltrt_surround_mix_level);
|
|
/* validate Lo/Ro center mix level */
|
|
validate_mix_level(avctx, "loro_center_mix_level",
|
|
&opt->loro_center_mix_level, extmixlev_options,
|
|
EXTMIXLEV_NUM_OPTIONS, 5, 0,
|
|
&s->loro_center_mix_level);
|
|
/* validate Lo/Ro surround mix level */
|
|
validate_mix_level(avctx, "loro_surround_mix_level",
|
|
&opt->loro_surround_mix_level, extmixlev_options,
|
|
EXTMIXLEV_NUM_OPTIONS, 6, 3,
|
|
&s->loro_surround_mix_level);
|
|
opt->extended_bsi_1 = 1;
|
|
} else {
|
|
opt->extended_bsi_1 = 0;
|
|
}
|
|
|
|
/* set extended bsi 2 flag */
|
|
if (opt->dolby_surround_ex_mode >= 0 ||
|
|
opt->dolby_headphone_mode >= 0 ||
|
|
opt->ad_converter_type >= 0) {
|
|
/* default dolby surround ex mode */
|
|
if (opt->dolby_surround_ex_mode < 0)
|
|
opt->dolby_surround_ex_mode = 0;
|
|
/* default dolby headphone mode */
|
|
if (opt->dolby_headphone_mode < 0)
|
|
opt->dolby_headphone_mode = 0;
|
|
/* default A/D converter type */
|
|
if (opt->ad_converter_type < 0)
|
|
opt->ad_converter_type = 0;
|
|
opt->extended_bsi_2 = 1;
|
|
} else {
|
|
opt->extended_bsi_2 = 0;
|
|
}
|
|
|
|
/* set bitstream id for alternate bitstream syntax */
|
|
if (opt->extended_bsi_1 || opt->extended_bsi_2) {
|
|
if (s->bitstream_id > 8 && s->bitstream_id < 11) {
|
|
static int warn_once = 1;
|
|
if (warn_once) {
|
|
av_log(avctx, AV_LOG_WARNING, "alternate bitstream syntax is "
|
|
"not compatible with reduced samplerates. writing of "
|
|
"extended bitstream information will be disabled.\n");
|
|
warn_once = 0;
|
|
}
|
|
} else {
|
|
s->bitstream_id = 6;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
/**
|
|
* Encode a single AC-3 frame.
|
|
*/
|
|
static int ac3_encode_frame(AVCodecContext *avctx, unsigned char *frame,
|
|
int buf_size, void *data)
|
|
{
|
|
AC3EncodeContext *s = avctx->priv_data;
|
|
const SampleType *samples = data;
|
|
int ret;
|
|
|
|
if (s->options.allow_per_frame_metadata) {
|
|
ret = validate_metadata(avctx);
|
|
if (ret)
|
|
return ret;
|
|
}
|
|
|
|
if (s->bit_alloc.sr_code == 1)
|
|
adjust_frame_size(s);
|
|
|
|
deinterleave_input_samples(s, samples);
|
|
|
|
apply_mdct(s);
|
|
|
|
scale_coefficients(s);
|
|
|
|
compute_rematrixing_strategy(s);
|
|
|
|
apply_rematrixing(s);
|
|
|
|
process_exponents(s);
|
|
|
|
ret = compute_bit_allocation(s);
|
|
if (ret) {
|
|
av_log(avctx, AV_LOG_ERROR, "Bit allocation failed. Try increasing the bitrate.\n");
|
|
return ret;
|
|
}
|
|
|
|
quantize_mantissas(s);
|
|
|
|
output_frame(s, frame);
|
|
|
|
return s->frame_size;
|
|
}
|
|
|
|
|
|
/**
|
|
* Finalize encoding and free any memory allocated by the encoder.
|
|
*/
|
|
static av_cold int ac3_encode_close(AVCodecContext *avctx)
|
|
{
|
|
int blk, ch;
|
|
AC3EncodeContext *s = avctx->priv_data;
|
|
|
|
for (ch = 0; ch < s->channels; ch++)
|
|
av_freep(&s->planar_samples[ch]);
|
|
av_freep(&s->planar_samples);
|
|
av_freep(&s->bap_buffer);
|
|
av_freep(&s->bap1_buffer);
|
|
av_freep(&s->mdct_coef_buffer);
|
|
av_freep(&s->fixed_coef_buffer);
|
|
av_freep(&s->exp_buffer);
|
|
av_freep(&s->grouped_exp_buffer);
|
|
av_freep(&s->psd_buffer);
|
|
av_freep(&s->band_psd_buffer);
|
|
av_freep(&s->mask_buffer);
|
|
av_freep(&s->qmant_buffer);
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
AC3Block *block = &s->blocks[blk];
|
|
av_freep(&block->bap);
|
|
av_freep(&block->mdct_coef);
|
|
av_freep(&block->fixed_coef);
|
|
av_freep(&block->exp);
|
|
av_freep(&block->grouped_exp);
|
|
av_freep(&block->psd);
|
|
av_freep(&block->band_psd);
|
|
av_freep(&block->mask);
|
|
av_freep(&block->qmant);
|
|
}
|
|
|
|
mdct_end(&s->mdct);
|
|
|
|
av_freep(&avctx->coded_frame);
|
|
return 0;
|
|
}
|
|
|
|
|
|
/**
|
|
* Set channel information during initialization.
|
|
*/
|
|
static av_cold int set_channel_info(AC3EncodeContext *s, int channels,
|
|
int64_t *channel_layout)
|
|
{
|
|
int ch_layout;
|
|
|
|
if (channels < 1 || channels > AC3_MAX_CHANNELS)
|
|
return AVERROR(EINVAL);
|
|
if ((uint64_t)*channel_layout > 0x7FF)
|
|
return AVERROR(EINVAL);
|
|
ch_layout = *channel_layout;
|
|
if (!ch_layout)
|
|
ch_layout = avcodec_guess_channel_layout(channels, CODEC_ID_AC3, NULL);
|
|
if (av_get_channel_layout_nb_channels(ch_layout) != channels)
|
|
return AVERROR(EINVAL);
|
|
|
|
s->lfe_on = !!(ch_layout & AV_CH_LOW_FREQUENCY);
|
|
s->channels = channels;
|
|
s->fbw_channels = channels - s->lfe_on;
|
|
s->lfe_channel = s->lfe_on ? s->fbw_channels : -1;
|
|
if (s->lfe_on)
|
|
ch_layout -= AV_CH_LOW_FREQUENCY;
|
|
|
|
switch (ch_layout) {
|
|
case AV_CH_LAYOUT_MONO: s->channel_mode = AC3_CHMODE_MONO; break;
|
|
case AV_CH_LAYOUT_STEREO: s->channel_mode = AC3_CHMODE_STEREO; break;
|
|
case AV_CH_LAYOUT_SURROUND: s->channel_mode = AC3_CHMODE_3F; break;
|
|
case AV_CH_LAYOUT_2_1: s->channel_mode = AC3_CHMODE_2F1R; break;
|
|
case AV_CH_LAYOUT_4POINT0: s->channel_mode = AC3_CHMODE_3F1R; break;
|
|
case AV_CH_LAYOUT_QUAD:
|
|
case AV_CH_LAYOUT_2_2: s->channel_mode = AC3_CHMODE_2F2R; break;
|
|
case AV_CH_LAYOUT_5POINT0:
|
|
case AV_CH_LAYOUT_5POINT0_BACK: s->channel_mode = AC3_CHMODE_3F2R; break;
|
|
default:
|
|
return AVERROR(EINVAL);
|
|
}
|
|
s->has_center = (s->channel_mode & 0x01) && s->channel_mode != AC3_CHMODE_MONO;
|
|
s->has_surround = s->channel_mode & 0x04;
|
|
|
|
s->channel_map = ff_ac3_enc_channel_map[s->channel_mode][s->lfe_on];
|
|
*channel_layout = ch_layout;
|
|
if (s->lfe_on)
|
|
*channel_layout |= AV_CH_LOW_FREQUENCY;
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
static av_cold int validate_options(AVCodecContext *avctx, AC3EncodeContext *s)
|
|
{
|
|
int i, ret;
|
|
|
|
/* validate channel layout */
|
|
if (!avctx->channel_layout) {
|
|
av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
|
|
"encoder will guess the layout, but it "
|
|
"might be incorrect.\n");
|
|
}
|
|
ret = set_channel_info(s, avctx->channels, &avctx->channel_layout);
|
|
if (ret) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid channel layout\n");
|
|
return ret;
|
|
}
|
|
|
|
/* validate sample rate */
|
|
for (i = 0; i < 9; i++) {
|
|
if ((ff_ac3_sample_rate_tab[i / 3] >> (i % 3)) == avctx->sample_rate)
|
|
break;
|
|
}
|
|
if (i == 9) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid sample rate\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
s->sample_rate = avctx->sample_rate;
|
|
s->bit_alloc.sr_shift = i % 3;
|
|
s->bit_alloc.sr_code = i / 3;
|
|
s->bitstream_id = 8 + s->bit_alloc.sr_shift;
|
|
|
|
/* validate bit rate */
|
|
for (i = 0; i < 19; i++) {
|
|
if ((ff_ac3_bitrate_tab[i] >> s->bit_alloc.sr_shift)*1000 == avctx->bit_rate)
|
|
break;
|
|
}
|
|
if (i == 19) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid bit rate\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
s->bit_rate = avctx->bit_rate;
|
|
s->frame_size_code = i << 1;
|
|
|
|
/* validate cutoff */
|
|
if (avctx->cutoff < 0) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid cutoff frequency\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
s->cutoff = avctx->cutoff;
|
|
if (s->cutoff > (s->sample_rate >> 1))
|
|
s->cutoff = s->sample_rate >> 1;
|
|
|
|
/* validate audio service type / channels combination */
|
|
if ((avctx->audio_service_type == AV_AUDIO_SERVICE_TYPE_KARAOKE &&
|
|
avctx->channels == 1) ||
|
|
((avctx->audio_service_type == AV_AUDIO_SERVICE_TYPE_COMMENTARY ||
|
|
avctx->audio_service_type == AV_AUDIO_SERVICE_TYPE_EMERGENCY ||
|
|
avctx->audio_service_type == AV_AUDIO_SERVICE_TYPE_VOICE_OVER)
|
|
&& avctx->channels > 1)) {
|
|
av_log(avctx, AV_LOG_ERROR, "invalid audio service type for the "
|
|
"specified number of channels\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
|
|
ret = validate_metadata(avctx);
|
|
if (ret)
|
|
return ret;
|
|
|
|
return 0;
|
|
}
|
|
|
|
|
|
/**
|
|
* Set bandwidth for all channels.
|
|
* The user can optionally supply a cutoff frequency. Otherwise an appropriate
|
|
* default value will be used.
|
|
*/
|
|
static av_cold void set_bandwidth(AC3EncodeContext *s)
|
|
{
|
|
int ch, bw_code;
|
|
|
|
if (s->cutoff) {
|
|
/* calculate bandwidth based on user-specified cutoff frequency */
|
|
int fbw_coeffs;
|
|
fbw_coeffs = s->cutoff * 2 * AC3_MAX_COEFS / s->sample_rate;
|
|
bw_code = av_clip((fbw_coeffs - 73) / 3, 0, 60);
|
|
} else {
|
|
/* use default bandwidth setting */
|
|
/* XXX: should compute the bandwidth according to the frame
|
|
size, so that we avoid annoying high frequency artifacts */
|
|
bw_code = 50;
|
|
}
|
|
|
|
/* set number of coefficients for each channel */
|
|
for (ch = 0; ch < s->fbw_channels; ch++) {
|
|
s->bandwidth_code[ch] = bw_code;
|
|
s->nb_coefs[ch] = bw_code * 3 + 73;
|
|
}
|
|
if (s->lfe_on)
|
|
s->nb_coefs[s->lfe_channel] = 7; /* LFE channel always has 7 coefs */
|
|
}
|
|
|
|
|
|
static av_cold int allocate_buffers(AVCodecContext *avctx)
|
|
{
|
|
int blk, ch;
|
|
AC3EncodeContext *s = avctx->priv_data;
|
|
|
|
FF_ALLOC_OR_GOTO(avctx, s->planar_samples, s->channels * sizeof(*s->planar_samples),
|
|
alloc_fail);
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
FF_ALLOCZ_OR_GOTO(avctx, s->planar_samples[ch],
|
|
(AC3_FRAME_SIZE+AC3_BLOCK_SIZE) * sizeof(**s->planar_samples),
|
|
alloc_fail);
|
|
}
|
|
FF_ALLOC_OR_GOTO(avctx, s->bap_buffer, AC3_MAX_BLOCKS * s->channels *
|
|
AC3_MAX_COEFS * sizeof(*s->bap_buffer), alloc_fail);
|
|
FF_ALLOC_OR_GOTO(avctx, s->bap1_buffer, AC3_MAX_BLOCKS * s->channels *
|
|
AC3_MAX_COEFS * sizeof(*s->bap1_buffer), alloc_fail);
|
|
FF_ALLOC_OR_GOTO(avctx, s->mdct_coef_buffer, AC3_MAX_BLOCKS * s->channels *
|
|
AC3_MAX_COEFS * sizeof(*s->mdct_coef_buffer), alloc_fail);
|
|
FF_ALLOC_OR_GOTO(avctx, s->exp_buffer, AC3_MAX_BLOCKS * s->channels *
|
|
AC3_MAX_COEFS * sizeof(*s->exp_buffer), alloc_fail);
|
|
FF_ALLOC_OR_GOTO(avctx, s->grouped_exp_buffer, AC3_MAX_BLOCKS * s->channels *
|
|
128 * sizeof(*s->grouped_exp_buffer), alloc_fail);
|
|
FF_ALLOC_OR_GOTO(avctx, s->psd_buffer, AC3_MAX_BLOCKS * s->channels *
|
|
AC3_MAX_COEFS * sizeof(*s->psd_buffer), alloc_fail);
|
|
FF_ALLOC_OR_GOTO(avctx, s->band_psd_buffer, AC3_MAX_BLOCKS * s->channels *
|
|
64 * sizeof(*s->band_psd_buffer), alloc_fail);
|
|
FF_ALLOC_OR_GOTO(avctx, s->mask_buffer, AC3_MAX_BLOCKS * s->channels *
|
|
64 * sizeof(*s->mask_buffer), alloc_fail);
|
|
FF_ALLOC_OR_GOTO(avctx, s->qmant_buffer, AC3_MAX_BLOCKS * s->channels *
|
|
AC3_MAX_COEFS * sizeof(*s->qmant_buffer), alloc_fail);
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
AC3Block *block = &s->blocks[blk];
|
|
FF_ALLOC_OR_GOTO(avctx, block->bap, s->channels * sizeof(*block->bap),
|
|
alloc_fail);
|
|
FF_ALLOCZ_OR_GOTO(avctx, block->mdct_coef, s->channels * sizeof(*block->mdct_coef),
|
|
alloc_fail);
|
|
FF_ALLOCZ_OR_GOTO(avctx, block->exp, s->channels * sizeof(*block->exp),
|
|
alloc_fail);
|
|
FF_ALLOCZ_OR_GOTO(avctx, block->grouped_exp, s->channels * sizeof(*block->grouped_exp),
|
|
alloc_fail);
|
|
FF_ALLOCZ_OR_GOTO(avctx, block->psd, s->channels * sizeof(*block->psd),
|
|
alloc_fail);
|
|
FF_ALLOCZ_OR_GOTO(avctx, block->band_psd, s->channels * sizeof(*block->band_psd),
|
|
alloc_fail);
|
|
FF_ALLOCZ_OR_GOTO(avctx, block->mask, s->channels * sizeof(*block->mask),
|
|
alloc_fail);
|
|
FF_ALLOCZ_OR_GOTO(avctx, block->qmant, s->channels * sizeof(*block->qmant),
|
|
alloc_fail);
|
|
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
/* arrangement: block, channel, coeff */
|
|
block->bap[ch] = &s->bap_buffer [AC3_MAX_COEFS * (blk * s->channels + ch)];
|
|
block->mdct_coef[ch] = &s->mdct_coef_buffer [AC3_MAX_COEFS * (blk * s->channels + ch)];
|
|
block->grouped_exp[ch] = &s->grouped_exp_buffer[128 * (blk * s->channels + ch)];
|
|
block->psd[ch] = &s->psd_buffer [AC3_MAX_COEFS * (blk * s->channels + ch)];
|
|
block->band_psd[ch] = &s->band_psd_buffer [64 * (blk * s->channels + ch)];
|
|
block->mask[ch] = &s->mask_buffer [64 * (blk * s->channels + ch)];
|
|
block->qmant[ch] = &s->qmant_buffer [AC3_MAX_COEFS * (blk * s->channels + ch)];
|
|
|
|
/* arrangement: channel, block, coeff */
|
|
block->exp[ch] = &s->exp_buffer [AC3_MAX_COEFS * (AC3_MAX_BLOCKS * ch + blk)];
|
|
}
|
|
}
|
|
|
|
if (CONFIG_AC3ENC_FLOAT) {
|
|
FF_ALLOC_OR_GOTO(avctx, s->fixed_coef_buffer, AC3_MAX_BLOCKS * s->channels *
|
|
AC3_MAX_COEFS * sizeof(*s->fixed_coef_buffer), alloc_fail);
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
AC3Block *block = &s->blocks[blk];
|
|
FF_ALLOCZ_OR_GOTO(avctx, block->fixed_coef, s->channels *
|
|
sizeof(*block->fixed_coef), alloc_fail);
|
|
for (ch = 0; ch < s->channels; ch++)
|
|
block->fixed_coef[ch] = &s->fixed_coef_buffer[AC3_MAX_COEFS * (blk * s->channels + ch)];
|
|
}
|
|
} else {
|
|
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
|
|
AC3Block *block = &s->blocks[blk];
|
|
FF_ALLOCZ_OR_GOTO(avctx, block->fixed_coef, s->channels *
|
|
sizeof(*block->fixed_coef), alloc_fail);
|
|
for (ch = 0; ch < s->channels; ch++)
|
|
block->fixed_coef[ch] = (int32_t *)block->mdct_coef[ch];
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
alloc_fail:
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
|
|
/**
|
|
* Initialize the encoder.
|
|
*/
|
|
static av_cold int ac3_encode_init(AVCodecContext *avctx)
|
|
{
|
|
AC3EncodeContext *s = avctx->priv_data;
|
|
int ret, frame_size_58;
|
|
|
|
avctx->frame_size = AC3_FRAME_SIZE;
|
|
|
|
ff_ac3_common_init();
|
|
|
|
ret = validate_options(avctx, s);
|
|
if (ret)
|
|
return ret;
|
|
|
|
s->bitstream_mode = avctx->audio_service_type;
|
|
if (s->bitstream_mode == AV_AUDIO_SERVICE_TYPE_KARAOKE)
|
|
s->bitstream_mode = 0x7;
|
|
|
|
s->frame_size_min = 2 * ff_ac3_frame_size_tab[s->frame_size_code][s->bit_alloc.sr_code];
|
|
s->bits_written = 0;
|
|
s->samples_written = 0;
|
|
s->frame_size = s->frame_size_min;
|
|
|
|
/* calculate crc_inv for both possible frame sizes */
|
|
frame_size_58 = (( s->frame_size >> 2) + ( s->frame_size >> 4)) << 1;
|
|
s->crc_inv[0] = pow_poly((CRC16_POLY >> 1), (8 * frame_size_58) - 16, CRC16_POLY);
|
|
if (s->bit_alloc.sr_code == 1) {
|
|
frame_size_58 = (((s->frame_size+2) >> 2) + ((s->frame_size+2) >> 4)) << 1;
|
|
s->crc_inv[1] = pow_poly((CRC16_POLY >> 1), (8 * frame_size_58) - 16, CRC16_POLY);
|
|
}
|
|
|
|
set_bandwidth(s);
|
|
|
|
rematrixing_init(s);
|
|
|
|
exponent_init(s);
|
|
|
|
bit_alloc_init(s);
|
|
|
|
ret = mdct_init(avctx, &s->mdct, 9);
|
|
if (ret)
|
|
goto init_fail;
|
|
|
|
ret = allocate_buffers(avctx);
|
|
if (ret)
|
|
goto init_fail;
|
|
|
|
avctx->coded_frame= avcodec_alloc_frame();
|
|
|
|
dsputil_init(&s->dsp, avctx);
|
|
ff_ac3dsp_init(&s->ac3dsp, avctx->flags & CODEC_FLAG_BITEXACT);
|
|
|
|
dprint_options(avctx);
|
|
|
|
return 0;
|
|
init_fail:
|
|
ac3_encode_close(avctx);
|
|
return ret;
|
|
}
|