This file is raw output from pdftotext and may not be ideal for distribution. If you are a maintainer for Hackipedia, please sit down when you have time and clean this text version up. Source PDF: /mnt/main/jmc-storage/docs/ATSC/A-52 Digital Audio Compression Standard (AC-3, E-AC-3) (November 2010).pdf Like all conversions the text below should be fully readable as UTF-8 unicode text. --------------------------------------------------------------- Digital Audio Compression Standard (AC-3, E-AC-3) Document A/52:2010, 22 November 2010 Advanced Television Systems Committee, Inc. 1776 K Street, N.W., Suite 200 Washington, D.C. 20006 Advanced Television Systems Committee Document A/52:2010 The Advanced Television Systems Committee, Inc., is an international, non-profit organization developing voluntary standards for digital television. The ATSC member organizations represent the broadcast, broadcast equipment, motion picture, consumer electronics, computer, cable, satellite, and semiconductor industries. Specifically, ATSC is working to coordinate television standards among different communications media focusing on digital television, interactive systems, and broadband multimedia communications. ATSC is also developing digital television implementation strategies and presenting educational seminars on the ATSC standards. ATSC was formed in 1982 by the member organizations of the Joint Committee on InterSociety Coordination (JCIC): the Electronic Industries Association (EIA), the Institute of Electrical and Electronic Engineers (IEEE), the National Association of Broadcasters (NAB), the National Cable Telecommunications Association (NCTA), and the Society of Motion Picture and Television Engineers (SMPTE). Currently, there are approximately 170 members representing the broadcast, broadcast equipment, motion picture, consumer electronics, computer, cable, satellite, and semiconductor industries. ATSC Digital TV Standards include digital high definition television (HDTV), standard definition television (SDTV), data broadcasting, multichannel surround-sound audio, and satellite direct-to-home broadcasting. Contact information is given below.. NOTE: The user's attention is called to the possibility that compliance with this standard may require use of an invention covered by patent rights. By publication of this standard, no position is taken with respect to the validity of this claim or of any patent rights in connection therewith. One or more patent holders have, however, filed a statement regarding the terms on which such patent holder(s) may be willing to grant a license under these rights to individuals or entities desiring to obtain such a license. Details may be obtained from the ATSC Secretary and the patent holder. The revision history of this document is given below. A/52 Revision History A/52 approved 10 November 1994 Annex A approved 12 April 1995, Annex B and Annex C approved 20 December 1995 A/52A revision approved 20 August 2001 Revision A corrected some errata in the detailed specifications, revised Annex A to include additional information about the DVB standard, removed Annex B that described an interface specification (superseeded by IEC and SMPTE standards), and added a new annex, “Alternate Bit Stream Syntax,” which contributes (in a compatible fashion) some new features to the AC-3 bit stream. A/52B revision approved 14 June 2005 Revision B corrected some errata in the detailed specifications, and added a new annex, “Enhanced AC-3 Bit Stream Syntax” which specifies a non-backwards compatible syntax that offers additional coding tools and features. Informative references were removed from the body of the document and placed in a new Annex B. A/52:2010 approved 22 November 2010 A/52:2010 clarified several areas of ambiguity in A/52B identified by CEA working group R4.3 WG12. Additional items were identified by TSG/S6 members and subsequently addressed. 2 Digital Audio Compression Standard, Table of Contents 22 November 2010 Table of Contents 1. SCOPE 19 2. INTRODUCTION 19 2.1 Motivation 20 2.2 Encoding 21 2.3 Decoding 22 3. REFERENCES 23 3.1 Normative References 23 3.2 Informative References 23 4. NOTATION, DEFINITIONS, AND TERMINOLOGY 24 4.1 Compliance Notation 24 4.2 Definitions 24 4.3 Terminology Abbreviations 25 5. BIT STREAM SYNTAX 29 5.1 Synchronization Frame 29 5.2 Semantics of Syntax Specification 29 5.3 Syntax Specification 30 5.3.1 syncinfo: Synchronization Information 30 5.3.2 bsi: Bit Stream Information 31 5.3.3 audioblk: Audio Block 32 5.3.4 auxdata: Auxiliary Data 37 5.3.5 errorcheck: Error Detection Code 38 5.4 Description of Bit Stream Elements 38 5.4.1 syncinfo: Synchronization Information 38 5.4.1.1 syncword: Synchronization Word, 16 bits 38 5.4.1.2 crc1: Cyclic Redundancy Check 1, 16 bits 38 5.4.1.3 fscod: Sample Rate Code, 2 bits 38 5.4.1.4 frmsizecod: Frame Size Code, 6 bits 38 5.4.2 bsi: Bit Stream Information 39 5.4.2.1 bsid: Bit Stream Identification, 5 bits 39 5.4.2.2 bsmod: Bit Stream Mode, 3 bits 39 5.4.2.3 acmod: Audio Coding Mode, 3 bits 39 5.4.2.4 cmixlev: Center Mix Level, 2 bits 40 5.4.2.5 surmixlev: Surround Mix Level, 2 bits 40 5.4.2.6 dsurmod: Dolby Surround Mode, 2 bits 40 5.4.2.7 lfeon: Low Frequency Effects Channel On, 1 bit 41 5.4.2.8 dialnorm: Dialogue Normalization, 5 bits 41 5.4.2.9 compre: Compression Gain Word Exists, 1 bit 41 5.4.2.10 compr: Compression Gain Word, 8 bits 41 5.4.2.11 langcode: Language Code Exists, 1 bit 41 5.4.2.12 langcod: Language Code, 8 bits 41 5.4.2.13 audprodie: Audio Production Information Exists, 1 bit 41 5.4.2.14 mixlevel: Mixing Level, 5 bits 42 3 Advanced Television Systems Committee Document A/52:2010 5.4.2.15 roomtyp: Room Type, 2 bits 42 5.4.2.16 dialnorm2: Dialogue Normalization, Ch2, 5 bits 42 5.4.2.17 compr2e: Compression Gain Word Exists, Ch2, 1 bit 42 5.4.2.18 compr2: Compression Gain Word, Ch2, 8 bits 42 5.4.2.19 langcod2e: Language Code Exists, Ch2, 1 bit 42 5.4.2.20 langcod2: Language Code, Ch2, 8 bits 42 5.4.2.21 audprodi2e: Audio Production Information Exists, Ch2, 1 bit 42 5.4.2.22 mixlevel2: Mixing Level, Ch2, 5 bits 43 5.4.2.23 roomtyp2: Room Type, Ch2, 2 bits 43 5.4.2.24 copyrightb: Copyright Bit, 1 bit 43 5.4.2.25 origbs: Original Bit Stream, 1 bit 43 5.4.2.26 timecod1e, timcode2e: Time Code (first and second) Halves Exist, 2 bits 43 5.4.2.27 timecod1: Time Code First Half, 14 bits 43 5.4.2.28 timecod2: Time Code Second Half, 14 bits 43 5.4.2.29 addbsie: Additional Bit Stream Information Exists, 1 bit 43 5.4.2.30 addbsil: Additional Bit Stream Information Length, 6 bits 44 5.4.2.31 addbsi: Additional Bit Stream Information, [(addbsil+1) ´ 8] bits 44 5.4.3 audblk: Audio Block 44 5.4.3.1 blksw[ch]: Block Switch Flag, 1 bit 44 5.4.3.2 dithflag[ch]: Dither Flag, 1 bit 44 5.4.3.3 dynrnge: Dynamic Range Gain Word Exists, 1 bit 44 5.4.3.4 dynrng: Dynamic Range Gain Word, 8 bits 44 5.4.3.5 dynrng2e: Dynamic Range Gain Word Exists, Ch2, 1 bit 44 5.4.3.6 dynrng2: Dynamic Range Gain Word Ch2, 8 bits 44 5.4.3.7 cplstre: Coupling Strategy Exists, 1 bit 44 5.4.3.8 cplinu: Coupling in Use, 1 bit 45 5.4.3.9 chincpl[ch]: Channel in Coupling, 1 bit 45 5.4.3.10 phsflginu: Phase Flags in Use, 1 bit 45 5.4.3.11 cplbegf: Coupling Begin Frequency Code, 4 bits 45 5.4.3.12 cplendf: Coupling End Frequency Code, 4 bits 45 5.4.3.13 cplbndstrc[sbnd]: Coupling Band Structure, 1 bit 45 5.4.3.14 cplcoe[ch]: Coupling Coordinates Exist, 1 bit 46 5.4.3.15 mstrcplco[ch]: Master Coupling Coordinate, 2 bits 46 5.4.3.16 cplcoexp[ch][bnd]: Coupling Coordinate Exponent, 4 bits 46 5.4.3.17 cplcomant[ch][bnd]: Coupling Coordinate Mantissa, 4 bits 46 5.4.3.18 phsflg[bnd]: Phase Flag, 1 bit 46 5.4.3.19 rematstr: Rematrixing Strategy, 1 bit 46 5.4.3.20 rematflg[rbnd]: Rematrix Flag, 1 bit 47 5.4.3.21 cplexpstr: Coupling Exponent Strategy, 2 bits 47 5.4.3.22 chexpstr[ch]: Channel Exponent Strategy, 2 bits 47 5.4.3.23 lfeexpstr: Low Frequency Effects Channel Exponent Strategy, 1 bit 47 5.4.3.24 chbwcod[ch]: Channel Bandwidth Code, 6 bits 47 5.4.3.25 cplabsexp: Coupling Absolute Exponent, 4 bits 47 5.4.3.26 cplexps[grp]: Coupling Exponents, 7 bits 47 5.4.3.27 exps[ch][grp]: Channel Exponents, 4 or 7 bits 48 4 Digital Audio Compression Standard, Table of Contents 22 November 2010 5.4.3.28 gainrng[ch]: Channel Gain Range Code, 2 bits 48 5.4.3.29 lfeexps[grp]: Low Frequency Effects Channel Exponents, 4 or 7 bits 48 5.4.3.30 baie: Bit Allocation Information Exists, 1 bit 48 5.4.3.31 sdcycod: Slow Decay Code, 2 bits 48 5.4.3.32 fdcycod: Fast Decay Code, 2 bits 48 5.4.3.33 sgaincod: Slow Gain Code, 2 bits 48 5.4.3.34 dbpbcod: dB per Bit Code, 2 bits 48 5.4.3.35 floorcod: Masking Floor Code, 3 bits 48 5.4.3.36 snroffste: SNR Offset Exists, 1 bit 48 5.4.3.37 csnroffst: Coarse SNR Offset, 6 bits 49 5.4.3.38 cplfsnroffst: Coupling Fine SNR Offset, 4 bits 49 5.4.3.39 cplfgaincod: Coupling Fast Gain Code, 3 bits 49 5.4.3.40 fsnroffst[ch]: Channel Fine SNR Offset, 4 bits 49 5.4.3.41 fgaincod[ch]: Channel Fast Gain Code, 3 bits 49 5.4.3.42 lfefsnroffst: Low Frequency Effects Channel Fine SNR Offset, 4 bits 49 5.4.3.43 lfefgaincod: Low Frequency Effects Channel Fast Gain Code, 3 bits 49 5.4.3.44 cplleake: Coupling Leak Initialization Exists, 1 bit 49 5.4.3.45 cplfleak: Coupling Fast Leak Initialization, 3 bits 49 5.4.3.46 cplsleak: Coupling Slow Leak Initialization, 3 bits 49 5.4.3.47 deltbaie: Delta Bit Allocation Information Exists, 1 bit 49 5.4.3.48 cpldeltbae: Coupling Delta Bit Allocation Exists, 2 bits 49 5.4.3.49 deltbae[ch]: Delta Bit Allocation Exists, 2 bits 50 5.4.3.50 cpldeltnseg: Coupling Delta Bit Allocation Number of Segments, 3 bits 50 5.4.3.51 cpldeltoffst[seg]: Coupling Delta Bit Allocation Offset, 5 bits 50 5.4.3.52 cpldeltlen[seg]: Coupling Delta Bit Allocation Length, 4 bits 50 5.4.3.53 cpldeltba[seg]: Coupling Delta Bit Allocation, 3 bits 50 5.4.3.54 deltnseg[ch]: Channel Delta Bit Allocation Number of Segments, 3 bits 51 5.4.3.55 deltoffst[ch][seg]: Channel Delta Bit Allocation Offset, 5 bits 51 5.4.3.56 deltlen[ch][seg]: Channel Delta Bit Allocation Length, 4 bits 51 5.4.3.57 deltba[ch][seg]: Channel Delta Bit Allocation, 3 bits 51 5.4.3.58 skiple: Skip Length Exists, 1 bit 51 5.4.3.59 skipl: Skip Length, 9 bits 51 5.4.3.60 skipfld: Skip Field, (skipl * 8) bits 51 5.4.3.61 chmant[ch][bin]: Channel Mantissas, 0 to 16 bits 51 5.4.3.62 cplmant[bin]: Coupling Mantissas, 0 to 16 bits 51 5.4.3.63 lfemant[bin]: Low Frequency Effects Channel Mantissas, 0 to 16 bits 52 5.4.4 auxdata: Auxiliary Data Field 52 5.4.4.1 auxbits: Auxiliary Data Bits, nauxbits bits 52 5.4.4.2 auxdatal: Auxiliary Data Length, 14 bits 53 5.4.4.3 auxdatae: Auxiliary Data Exists, 1 bit 54 5 Advanced Television Systems Committee Document A/52:2010 5.4.5 errorcheck:Frame Error Detection Field 54 5.4.5.1 crcrsv: CRC Reserved Bit, 1 bit 54 5.4.5.2 crc2: Cyclic Redundancy Check 2, 16 bits 54 5.5 Bit Stream Constraints 54 6. DECODING THE AC-3 BIT STREAM 54 6.1 Summary of the Decoding Process 55 6.1.1 Input Bit Stream 55 6.1.1.1 Continuous or Burst Input 55 6.1.1.2 Byte or Word Alignment 56 6.1.2 Synchronization and Error Detection 56 6.1.3 Unpack BSI, Side Information 56 6.1.4 Decode Exponents 57 6.1.5 Bit Allocation 57 6.1.6 Process Mantissas 57 6.1.7 Decoupling 57 6.1.8 Rematrixing 57 6.1.9 Dynamic Range Compression 58 6.1.10 Inverse Transform 58 6.1.11 Window, Overlap/Add 58 6.1.12 Downmixing 58 6.1.13 PCM Output Buffer 58 6.1.14 Output PCM 58 7. ALGORITHMIC DETAILS 58 7.1 Exponent coding 58 7.1.1 Overview 58 7.1.2 Exponent Strategy 59 7.1.3 Exponent Decoding 61 7.2 Bit Allocation 65 7.2.1 Overview 65 7.2.2 Parametric Bit Allocation 65 7.2.2.1 Initialization 66 7.2.2.1.1 Special Case Processing Step 66 7.2.2.2 Exponent Mapping into PSD 67 7.2.2.3 PSD Integration 67 7.2.2.4 Compute Excitation Function 68 7.2.2.5 Compute Masking Curve 70 7.2.2.6 Apply Delta Bit Allocation 71 7.2.2.7 Compute Bit Allocation 72 7.2.3 Bit Allocation Tables 73 7.3 Quantization and Decoding of Mantissas 80 7.3.1 Overview 80 7.3.2 Expansion of Mantissas for Asymmetric Quantization (6 £ bap £ 15) 81 7.3.3 Expansion of Mantissas for Symmetrical Quantization (1 £ bap £ 5) 81 7.3.4 Dither for Zero Bit Mantissas (bap = 0) 82 7.3.5 Ungrouping of Mantissas 84 7.4 Channel Coupling 85 6 Digital Audio Compression Standard, Table of Contents 22 November 2010 7.4.1 Overview 85 7.4.2 Sub-Band Structure for Coupling 85 7.4.3 Coupling Coordinate Format 86 7.5 Rematrixing 87 7.5.1 Overview 87 7.5.2 Frequency Band Definitions 88 7.5.2.1 Coupling Not in Use 88 7.5.2.2 Coupling in Use, cplbegf > 2 89 7.5.2.3 Coupling in use, 2 ³ cplbegf > 0 89 7.5.2.4 Coupling in Use, cplbegf=0 89 7.5.3 Encoding Technique 90 7.5.4 Decoding Technique 90 7.6 Dialogue Normalization 91 7.6.1 Overview 91 7.7 Dynamic Range Compression 92 7.7.1 Dynamic Range Control; dynrng, dynrng2 92 7.7.1.1 Overview 92 7.7.1.2 Detailed Implementation 94 7.7.2 Heavy Compression; compr, compr2 95 7.7.2.1 Overview 95 7.7.2.2 Detailed Implementation 96 7.8 Downmixing 97 7.8.1 General Downmix Procedure 97 7.8.2 Downmixing Into Two Channels 101 7.9 Transform Equations and Block Switching 103 7.9.1 Overview 103 7.9.2 Technique 103 7.9.3 Decoder Implementation 104 7.9.4 Transformation Equations 104 7.9.4.1 512-Sample IMDCT Transform 104 7.9.4.2 256-Sample IMDCT Transforms 106 7.9.5 Channel Gain Range Code 109 7.10 Error Detection 109 7.10.1 CRC Checking 110 7.10.2 Checking Bit Stream Consistency 113 8. ENCODING THE AC-3 BIT STREAM 115 8.1 Introduction 115 8.2 Summary of the Encoding Process 115 8.2.1 Input PCM 115 8.2.1.1 Input Word Length 115 8.2.1.2 Input Sample Rate 115 8.2.1.3 Input Filtering 115 8.2.2 Transient Detection 115 8.2.3 Forward Transform 117 8.2.3.1 Windowing 117 8.2.3.2 Time to Frequency Transformation 118 7 Advanced Television Systems Committee Document A/52:2010 8.2.4 Coupling Strategy 118 8.2.4.1 Basic Encoder 118 8.2.4.2 Advanced Encoder 118 8.2.5 Form Coupling Channel 118 8.2.5.1 Coupling Channel 118 8.2.5.2 Coupling Coordinates 119 8.2.6 Rematrixing 119 8.2.7 Extract Exponents 119 8.2.8 Exponent Strategy 119 8.2.9 Dither Strategy 119 8.2.10 Encode Exponents 120 8.2.11 Normalize Mantissas 120 8.2.12 Core Bit Allocation 120 8.2.13 Quantize Mantissas 121 8.2.14 Pack AC-3 Frame 121 Annex A: AC-3 Elementary Streams in the MPEG-2 Multiplex (Normative) 123 A1. SCOPE 123 A2. INTRODUCTION 123 A3. GENERIC IDENTIFICATION OF AN AC-3 STREAM 124 A4. DETAILED SPECIFICATION FOR SYSTEM A 124 A4.1 Stream Type 124 A4.2 Stream ID 124 A4.3 AC-3 Audio Descriptor 124 A4.4 STD Audio Buffer Size 129 A5. DETAILED SPECIFICATION FOR SYSTEM B 130 A5.1 Stream Type 130 A5.2 Stream ID 130 A5.3 Service Information 130 A5.3.1 AC-3 Descriptor 130 A5.3.2 AC-3 Descriptor Syntax 130 A5.3.3 AC-3 Component Types 132 A5.4 STD Audio Buffer Size 133 A6. PES CONSTRAINTS 133 A6.1 Encoding 133 A6.2 Decoding 134 A6.3 Byte-Alignment 134 Annex B: Bibliography (Informative) 135 Annex C: AC-3 Karaoke Mode (Informative) 137 C1. SCOPE 137 C2. INTRODUCTION 137 C3. DETAILED SPECIFICATION 138 C3.1 Karaoke Mode Indication 138 8 Digital Audio Compression Standard, Table of Contents 22 November 2010 C3.2 Karaoke Mode Channel Assignment 138 C3.3 Reproduction of Karaoke Mode Bit Streams 138 C3.3.1 Karaoke Aware Decoders 138 C3.3.2 Karaoke Capable Decoders 139 Annex D: Alternate Bit Stream Syntax (Normative) 141 D1. SCOPE 141 D2. SPECIFICATION 141 D2.1 Indication of Alternate Bit Stream Syntax 141 D2.2 Alternate Bit Stream Syntax Specification 141 D2.3 Description of Alternate Syntax Bit Stream Elements 143 D2.3.1 xbsi1e: Extra Bitstream Information #1 Exists, 1 bit 143 D2.3.2 dmixmod: Preferred Stereo Downmix Mode, 2 bits 143 D2.3.3 ltrtcmixlev: Lt/Rt Center Mix Level, 3 bits 143 D2.3.4 ltrtsurmixlev: Lt/Rt Surround Mix Level, 3 bits 144 D2.3.5 lorocmixlev: Lo/Ro Center Mix Level, 3 bits 144 D2.3.6 lorosurmixlev: Lo/Ro Surround Mix Level, 3 bits 145 D2.3.7 xbsi2e: Extra Bit Stream Information #2 Exists, 1 bit 145 D2.3.8 dsurexmod: Dolby Surround EX Mode, 2 bits 145 D2.3.9 dheadphonmod: Dolby Headphone Mode, 2 bits 146 D2.3.10 adconvtyp: A/D Converter Type, 1 bit 146 D2.3.11 xbsi2: Extra Bit Stream Information, 8 bits 146 D2.3.12 encinfo: Encoder Information, 1 bit 146 D3. DECODER PROCESSING 147 D3.1 Compliant Decoder Processing 147 D3.1.1 Two-Channel Downmix Selection 147 D3.1.2 Two-Channel Downmix Processing 147 D3.1.3 Informational Parameter Processing 147 D3.2 Legacy Decoder Processing 147 D4. ENCODER PROCESSING 147 D4.1 Encoder Processing Steps 148 D4.1.1 Dynamic Range Overload Protection Processing 148 D4.2 Encoder Requirements 148 D4.2.1 Legacy Decoder Support 148 D4.2.2 Original Bit Stream Syntax Support 148 Annex E: Enhanced AC-3 Bit Stream Syntax (Normative) 149 E1. SCOPE 149 E2. SPECIFICATION 149 E2.1 Indication of Enhanced AC-3 Bit Stream Syntax 149 E2.2 Syntax Specification 149 E2.2.1 syncinfo: Synchronization Information 150 E2.2.2 bsi: Bit Stream Information 150 E2.2.3 audfrm: Audio Frame 154 E2.2.4 audblk: Audio Block 157 9 Advanced Television Systems Committee Document A/52:2010 E2.2.5 auxdata: Auxiliary Data 170 E2.2.6 errorcheck: Error Detection Code 170 E2.3 Description of Enhanced AC-3 bit stream elements 170 E2.3.1 bsi: Bit Stream Information 170 E2.3.1.1 strmtyp: Stream Type, 2 bits 170 E2.3.1.2 substreamid: Substream Identification, 3 bits 171 E2.3.1.3 frmsiz: Frame Size, 11 bits 171 E2.3.1.4 fscod: Sample Rate Code, 2 bits 171 E2.3.1.5 numblkscod / fscod2: Number of Audio Blocks / Sample Rate Code 2, 2 bits 172 E2.3.1.6 bsid: Bit Stream Identification, 5 bits 172 E2.3.1.7 chanmape: Custom Channel Map Exists, 1 bit 172 E2.3.1.8 chanmap: Custom Channel Map, 16 bits 172 E2.3.1.9 mixmdate: Mixing Meta-Data Exists, 1 bit 173 E2.3.1.10 lfemixlevcode: LFE mix Level Code Exists, 1 bit 173 E2.3.1.11 femixlevcod: LFE Mix Level Code, 5 bits 173 E2.3.1.12 pgmscle: Program Scale Factor Exists, 1 bit 173 E2.3.1.13 pgmscl: Program Scale Factor, 6 bits 174 E2.3.1.14 pgmscl2e: Program Scale Factor #2 Exists, 1 bit 174 E2.3.1.15 pgmscl2: Program Scale Factor #2, 6 bits 174 E2.3.1.16 extpgmscle: External Program Scale Factor Exists, 1 bit 174 E2.3.1.17 extpgmscl: External Program Scale Factor, 6 bits 174 E2.3.1.18 mixdef: Mix Control Type, 2 bits 174 E2.3.1.19 mixdeflen: Length of Mixing Parameter Data Field, 5 bits 174 E2.3.1.20 mixdata: Mixing Parameter Data, (5 – 264) bits 174 E2.3.1.21 paninfoe: Pan Information Exists, 1 bit 174 E2.3.1.22 paninfo: Pan Information, 14 bits 175 E2.3.1.23 paninfo2e: Pan Information Exists, 1 bit 175 E2.3.1.24 paninfo2: Pan Information, 14 bits 175 E2.3.1.25 frmmixcnfginfoe: Frame Mixing Configuration Information Exists, 1 bit 175 E2.3.1.26 blkmixcfginfoe: Block Mixing Configuration Information Exists, 1 bit 175 E2.3.1.27 blkmixcfginfo[blk]: Block Mixing Configuration Information, 5 bits 175 E2.3.1.28 infomdate: Informational Meta-Data Exists, 1 bit 175 E2.3.1.29 sourcefscod: Source Sample Rate Code, 1 bit 175 E2.3.1.30 convsync: Converter Synchronization Flag, 1 bit 175 E2.3.1.31 blkid: Block Identification, 1 bit 175 E2.3.2 audfrm – Audio Frame 175 E2.3.2.1 expstre: Exponent Strategy Syntax Enabled, 1 bit 175 E2.3.2.2 ahte: Adaptive Hybrid Transform Enabled, 1 bit 176 E2.3.2.3 snroffststr: SNR Offset Strategy, 2 bits 176 E2.3.2.4 transproce: Transient Pre-Noise Processing Enabled, 1 bit 176 E2.3.2.5 blkswe: Block Switch Syntax Enabled, 1 bit 176 E2.3.2.6 dithflage: Dither Flag Syntax Enabled, 1 bit 176 10 Digital Audio Compression Standard, Table of Contents 22 November 2010 E2.3.2.7 bamode: Bit Allocation Model Syntax Enabled, 1 bit 176 E2.3.2.8 frmfgaincode: Fast Gain Codes Enabled, 1 bit 177 E2.3.2.9 dbaflde: Delta Bit Allocation Syntax Enabled, 1 bit 177 E2.3.2.10 skipflde: Skip Field Syntax Enabled, 1 bit 177 E2.3.2.11 spxattene: Spectral Extension Attenuation Enabled, 1 bit 177 E2.3.2.12 frmcplexpstr: Frame Based Coupling Exponent Strategy, 5 bits 177 E2.3.2.13 frmchexpstr[ch]: Frame Based Channel Exponent Strategy, 5 bits 177 E2.3.2.14 convexpstre: Converter Exponent Strategy Exists, 1 bit 177 E2.3.2.15 convexpstr[ch]: Converter Channel Exponent Strategy, 5 bits 177 E2.3.2.16 cplahtinu: Coupling Channel AHT in Use, 1bit 178 E2.3.2.17 chahtinu[ch]: Channel AHT in Use, 1 bit 178 E2.3.2.18 lfeahtinu: LFE Channel AHT in Use, 1 bit 179 E2.3.2.19 frmcsnroffst: Frame Coarse SNR Offset, 6 bits 179 E2.3.2.20 frmfsnroffst: Frame Fine SNR Offset, 4 bits 179 E2.3.2.21 chintransproc[ch]: Channel in Transient Pre-Noise Processing, 1 bit 179 E2.3.2.22 transprocloc[ch]: Transient Location Relative to Start of Frame, 10 bits 179 E2.3.2.23 transproclen[ch]: Transient Processing Length, 8 bits 179 E2.3.2.24 chinspxatten[ch]: Channel in Spectral Extension Attenuation Processing, 1 bit 179 E2.3.2.25 spxattencod[ch]: Spectral Extension Attenuation Code, 5 bits 179 E2.3.2.26 blkstrtinfoe: Block Start Information Exists, 1 bit 179 E2.3.2.27 blkstrtinfo: Block Start Information, nblkstrtbits 179 E2.3.2.28 firstspxcos[ch]: First Spectral Extension Coordinates States 180 E2.3.2.29 firstcplcos[ch]: First Coupling Coordinates States 180 E2.3.2.30 firstcplleak: First Coupling Leak State 180 E2.3.3 audblk: Audio Block 180 E2.3.3.1 spxstre: Spectral Extension Strategy Exists, 1 bit 180 E2.3.3.2 spxinu: Spectral Extension in Use, 1 bit 180 E2.3.3.3 chinspx[ch]: Channel Using Spectral Extension, 1 bit 180 E2.3.3.4 spxstrtf: Spectral Extension Start Copy Frequency Code, 2 bits 180 E2.3.3.5 spxbegf: Spectral Extension Begin Frequency Code, 3 bits 180 E2.3.3.6 spxendf: Spectral Extension End Frequency Code, 3 bits 181 E2.3.3.7 spxbndstrce: Spectral Extension Band Structure Exist, 1 bit 181 E2.3.3.8 spxbndstrc[bnd]: Spectral Extension Band Structure, 1 – 14 bits 181 E2.3.3.9 spxcoe[ch]: Spectral Extension Coordinates Exist, 1 bit 181 E2.3.3.10 spxblnd[ch]: Spectral Extension Blend, 5 bits 182 E2.3.3.11 mstrspxco[ch]: Master Spectral Extension Coordinate, 2 bits 182 E2.3.3.12 spxcoexp[ch][bnd]: Spectral Extension Coordinate Exponent, 4 bits 182 E2.3.3.13 spxcomant[ch][bnd]: Spectral Extension Coordinate Mantissa, 2 bits 182 E2.3.3.14 ecplinu: Enhanced Coupling in Use, 1 bit 182 E2.3.3.15 cplbndstrce: Coupling Band Structure Exist, 1 bit 182 E2.3.3.16 ecplbegf: Enhanced Coupling Begin Frequency Code, 4 bits 183 11 Advanced Television Systems Committee Document A/52:2010 E2.3.3.17 ecplendf: Enhanced Coupling End Frequency Code, 4 bits 183 E2.3.3.18 ecplbndstrce: Enhanced Coupling Band Structure Exists, 1 bit 183 E2.3.3.19 ecplbndstrc[sbnd]: Enhanced Coupling Band Structure, 1 bit 184 E2.3.3.20 ecplangleintrp: Enhanced Coupling Angle Interpolation Flag, 1 bit 185 E2.3.3.21 ecplparam1e[ch]: Enhanced Coupling Parameters 1 Exist, 1 bit 185 E2.3.3.22 ecplparam2e[ch]: Enhanced Coupling Parameters 2 Exist, 1 bit 185 E2.3.3.23 ecplamp[ch][bnd]: Enhanced Coupling Amplitude Scaling, 5 bits 185 E2.3.3.24 ecplangle[ch][bnd]: Enhanced Coupling Angle, 6 bits 185 E2.3.3.25 ecplchaos[ch][bnd]: Enhanced Coupling Chaos, 3 bits 186 E2.3.3.26 ecpltrans[ch]: Enhanced Coupling Transient Present, 1 bit 186 E2.3.3.27 blkfsnroffst: Block Fine SNR Offset, 4 bits 186 E2.3.3.28 fgaincode: Fast Gain Codes Exist, 1 bit 186 E2.3.3.29 convsnroffste: Converter SNR Offset Exists, 1 bit 186 E2.3.3.30 convsnroffst: Converter SNR Offset, 10 bits 186 E2.3.3.31 chgaqmod[ch]: Channel Gain Adaptive Quantization Mode, 2 bits 186 E2.3.3.32 chgaqgain[ch][n]: Channel Gain Adaptive Quantization gain, 1 or 5 bits 186 E2.3.3.33 pre_chmant[n][ch][bin]: Pre Channel Mantissas, 0 to 16 bits 186 E2.3.3.34 cplgaqmod: Coupling Channel Gain Adaptive Quantization Mode, 2 bits 186 E2.3.3.35 cplgaqgain[n]: Coupling Gain Adaptive Quantization Gain, 1 or 5 bits 187 E2.3.3.36 pre_cplmant[n][bin]: Pre Coupling Channel Mantissas, 0 to 16 bits 187 E2.3.3.37 lfegaqmod: LFE Channel Gain Adaptive Quantization Mode, 2 bits 187 E2.3.3.38 lfegaqgain[n]: LFE Gain Adaptive Quantization Gain, 1 or 5 bits 187 E2.3.3.39 pre_lfemant[n][bin]: Pre LFE Channel Mantissas, 0 to 16 bits 187 E3. ALGORITHMIC DETAILS 187 E3.1 Glitch-Free Switching Between Different Stream Types 187 E3.2 Error Detection and Concealment 187 E3.3 Adaptive Hybrid Transform Processing 188 E3.3.1 Overview 188 E3.3.2 Bit Stream Helper Variables 188 E3.3.3 Bit Allocation 195 E3.3.3.1 Parametric Bit Allocation 196 E3.3.3.2 Bit Allocation Tables 198 E3.3.4 Quantization 199 E3.3.4.1 Vector Quantization 199 E3.3.4.2 Gain Adaptive Quantization 200 E3.3.5 Transform Equations 203 E3.4 Enhanced Channel Coupling 204 E3.4.1 Overview 204 E3.4.2 Sub-Band Structure for Enhanced Coupling 204 E3.4.3 Enhanced coupling tables 206 E3.4.4 Enhanced Coupling Coordinate Format 210 12 Digital Audio Compression Standard, Table of Contents 22 November 2010 E3.4.5 Enhanced Coupling Processing 210 E3.4.5.1 Process Enhanced Coupling Channel 210 E3.4.5.2 Process Amplitude Parameters 212 E3.4.5.3 Process Angle Parameters 213 E3.4.5.4 Generate Channel Transform Coefficients 216 E3.5 Spectral Extension Processing 216 E3.5.1 Overview 216 E3.5.2 Sub-Band Structure for Spectral Extension 217 E3.5.3 Spectral Extension Coordinate Format 218 E3.5.4 High Frequency Transform Coefficient Synthesis 219 E3.5.4.1 Transform Coefficient Translation 219 E3.5.4.2 Transform Coefficient Noise Blending 221 E3.5.4.2.1 Blending Factor Calculation 221 E3.5.4.2.2 Banded RMS Energy Calculation 221 E3.5.4.2.3 Transform Coefficient Band Border Filtering 222 E3.5.4.2.4 Noise Scaling and Transform Coefficient Blending Calculation 224 E3.5.4.3 Blended Transform Coefficient Scaling 225 E3.6 Transient Pre-Noise Processing 226 E3.6.1 Overview 226 E3.6.2 Application of Transient Pre-Noise Processing Data 226 E3.7 Channel and Program Extensions 228 E3.7.1 Overview 228 E3.7.2 Decoding a Single Program with Greater than 5.1 Channels 229 E3.7.3 Decoding Multiple Programs with up to 5.1 Channels 230 E3.7.4 Decoding a Mixture of Programs with up to 5.1 Channels and Programs with Greater than 5.1 Channels 230 E3.7.5 Dynamic Range Compression for Programs Containing Greater than 5.1 Channels 230 E3.8 LFE Downmixing Decoder Description 231 E4. AHT VECTOR QUANTIZATION TABLES 232 13 Advanced Television Systems Committee Document A/52:2010 Index of Tables Table 4.1 ATSC Digital Audio Compression Standard Terms 26 Table 5.2 bsi Syntax and Word Size 31 Table 5.1 syncinfo Syntax and Word Size 31 Table 5.3 audioblk Syntax and Word Size 32 Table 5.4 auxdata Syntax and Word Size 37 Table 5.5 errorcheck Syntax and Word Size 38 Table 5.6 Sample Rate Codes 38 Table 5.7 Bit Stream Mode 39 Table 5.8 Audio Coding Mode 40 Table 5.9 Center Mix Level 40 Table 5.10 Surround Mix Level 40 Table 5.11 Dolby Surround Mode 41 Table 5.12 Room Type 42 Table 5.13 Time Code Exists 43 Table 5.14 Master Coupling Coordinate 46 Table 5.15 Number of Rematrixing Bands 47 Table 5.16 Delta Bit Allocation Exists States 50 Table 5.17 Bit Allocation Deltas 50 Table 5.18 Frame Size Code Table (1 word = 16 bits) 53 Table 7.1 Mapping of Differential Exponent Values, D15 Mode 60 Table 7.2 Mapping of Differential Exponent Values, D25 Mode 60 Table 7.3 Mapping of Differential Exponent Values, D45 Mode 60 Table 7.4 Exponent Strategy Coding 61 Table 7.5 LFE Channel Exponent Strategy Coding 61 Table 7.6 Slow Decay Table, slowdec[] 73 Table 7.7 Fast Decay Table, fastdec[] 73 Table 7.8 Slow Gain Table, slowgain[] 74 Table 7.9 dB/Bit Table, dbpbtab[] 74 Table 7.10 Floor Table, floortab[] 74 Table 7.11 Fast Gain Table, fastgain[] 74 Table 7.12 Banding Structure Tables, bndtab[], bndsz[] 75 Table 7.13 Bin Number to Band Number Table, masktab[bin], bin = (10 * A) + B 76 Table 7.14 Log-Addition Table, latab[val], val = (10 * A) + B 77 Table 7.15 Hearing Threshold Table, hth[fscod][band] 78 Table 7.16 Bit Allocation Pointer Table, baptab[] 79 Table 7.17 Quantizer Levels and Mantissa Bits vs. bap 80 Table 7.18 Mapping of bap to Quantizer 81 Table 7.19 bap = 1 (3-Level) Quantization 82 Table 7.20 bap = 2 (5-Level) Quantization 82 Table 7.21 bap = 3 (7-Level) Quantization 83 Table 7.22 bap = 4 (11-Level) Quantization 83 Table 7.23 bap = 5 (15-Level) Quantization 84 Table 7.24 Coupling Sub-Bands 86 14 Digital Audio Compression Standard, Table of Contents 22 November 2010 Table 7.25 Rematrix Banding Table A 89 Table 7.26 Rematrixing Banding Table B 89 Table 7.27 Rematrixing Banding Table C 89 Table 7.28 Rematrixing Banding Table D 90 Table 7.29 Meaning of 3 msb of dynrng 94 Table 7.30 Meaning of 4 msb of compr 96 Table 7.31 LoRo Scaled Downmix Coefficients 102 Table 7.32 LtRt Scaled Downmix Coefficients 102 Table 7.33 Transform Window Sequence (w[addr]), where addr = (10 * A) + B 108 Table 7.34 gainrng Maximum Absolute Value 109 Table 7.35 5/8_framesize Table; Number of Words in the First 5/8 of the Frame 112 Table 7.36 Known Bit Stream Error Conditions 113 Annex A: Table A4.1 AC-3 Audio Descriptor Syntax 125 Table A4.2 Sample Rate Code Table 126 Table A4.3 Bit Rate Code Table 126 Table A4.4 surround_mode Table 127 Table A4.5 num_channels Table 127 Table A4.6 Priority Field Coding 128 Table A5.1 AC-3 Descriptor Syntax 131 Table A5.2 AC-3 component_type Byte Value Assignments 133 Annex C: Table C3.1 Channel Array Ordering 138 Table C3.2 Coefficient Values for Karaoke Aware Decoders 139 Table C3.3 Default Coefficient Values for Karaoke Capable Decoders 139 Annex D: Table D2.1 Bit Stream Information; Alternate Bit Stream Syntax 141 Table D2.2 Preferred Stereo Downmix Mode 143 Table D2.3 Lt/Rt Center Mix Level 144 Table D2.4 Lt/Rt Surround Mix Level 144 Table D2.5 Lo/Ro Center Mix Level 145 Table D2.6 Lo/Ro Surround Mix Level 145 Table D2.7 Dolby Surround EX Mode 146 Table D2.8 Dolby Headphone Mode 146 Table D2.9 A/D Converter Type 146 Annex E: Table E2.1 syncinfo Syntax and Word Size 150 Table E2.2 bsi Syntax and Word Size 150 Table E2.3 audfrm Syntax and Word Size 154 Table E2.4 audblk Syntax and Word Size 157 Table E2.5 auxdata Syntax and Word Size 170 Table E2.6 errorcheck Syntax and Word Size 170 Table E2.7 Stream Type 170 Table E2.8 Sample Rate Codes 171 15 Advanced Television Systems Committee Document A/52:2010 Table E2.9 Number of Audio Blocks Per Syncframe 172 Table E2.10 Reduced Sampling Rates 172 Table E2.11 Custom Channel Map Locations 173 Table E2.12 Mix Control 174 Table E2.13 SNR Offset Strategy 176 Table E2.14 Frame Exponent Strategy Combinations 178 Table E2.15 Default Spectral Extension Banding Structure 181 Table E2.16 Default Coupling Banding Structure 183 Table E2.17 Default Enhanced Coupling Banding Structure 184 Table E3.1 High Efficiency Bit Allocation Pointers, hebaptab[] 198 Table E3.2 Quantizer Type, Quantizer Level, and Mantissa Bits vs. hebap 199 Table E3.3 Gain Adaptive Quantization Modes 201 Table E3.4 Mapping of Gain Elements, gaqmod = 0x3 202 Table E3.5 Gain Adaptive Quantizer Characteristics 202 Table E3.6 Large Mantissa Inverse Quantization (Remapping) Constants 203 Table E3.7 Enhanced Coupling Sub-bands 205 Table E3.8 Enhanced Coupling Start and End Indexes 206 Table E3.9 Sub-band Transform Start Coefficients: ecplsubbndtab[] 207 Table E3.10 Amplitudes: ecplampexptab[], ecplampmanttab[] 208 Table E3.11 Angles: ecplangletab[] 209 Table E3.12 Chaos Scaling: ecplchaostab[] 210 Table E3.13 Spectral Extension Band Table 218 Table E3.14 Spectral Extension Attenuation Table: spxattentab[][] 224 Table E4.1 VQ Table for hebap 1; 16-bit two’s complement 232 Table E4.2 VQ Table for hebap 2; 16-bit two’s complement 232 Table E4.3 VQ Table for hebap 3; 16-bit two’s complement 232 Table E4.4 VQ Table for hebap 4; 16-bit two’s complement 233 Table E4.5 VQ Table for hebap 5; 16-bit two’s complement 233 Table E4.6 VQ Table for hebap 6; 16-bit two’s complement 237 Table E4.7 VQ Table for hebap 7; 16-bit two’s complement 243 16 Digital Audio Compression Standard, Table of Contents 22 November 2010 Index of Figures Figure 2.1 Example application of AC-3 to satellite audio transmission. 21 Figure 2.2 The AC-3 encoder. 22 Figure 2.3 The AC-3 decoder. 23 Figure 5.1 AC-3 synchronization frame. 29 Figure 6.1 Flow diagram of the decoding process. 55 Figure 7.1 Example LFSR circuit. 110 Figure 8.1. Flow diagram of the encoding process. 116 Annex E: Figure E3.1 Flow diagram for GAQ mantissa dequantization. 200 Figure E3.2 Transient pre-noise time scaling synthesis summary. 227 Figure E3.3 Bitstream with a single program of greater than 5.1 channels. 229 Figure E3.4 Bitstream with multiple programs of up to 5.1 channels. 230 Figure E3.5 Bitstream with mixture of programs of up to 5.1 channels and programs of greater than 5.1 channels. 231 17 ATSC Standard: Digital Audio Compression Standard 1. SCOPE This standard defines how to create a coded representation of audio information, how to describe this representation, how to arrange the coded representation for storage or transmission and how to decode the data to create audio. The coded representation defined herein is intended for use in digital audio transmission and storage applications. A short form designation of the audio coding algorithm specified in the body of this Standard is “AC-3”. The short form designation of the audio coding algorithm specified in Annex E is “E- AC-3”. 2. INTRODUCTION The United States Advanced Television Systems Committee (ATSC), Inc., was formed by the member organizations of the Joint Committee on InterSociety Coordination (JCIC)1, recognizing that the prompt, efficient and effective development of a coordinated set of national standards is essential to the future development of domestic television services. One of the activities of the ATSC is exploring the need for and, where appropriate, coordinating the development of voluntary national technical standards for Advanced Television Systems (ATV). The ATSC Executive Committee assigned the work of documenting the U.S. ATV standard to a number of specialist groups working under the Technology Group on Distribution (T3). The Audio Specialist Group (T3/S7) was charged with documenting the ATV audio standard. This document was prepared initially by the Audio Specialist Group as part of its efforts to document the United States Advanced Television Broadcast Standard. It was approved by the Technology Group on Distribution on 26 September 1994, and by the full ATSC membership as an ATSC Standard on 10 November 1994. Annex A, “AC-3 Elementary Streams in an MPEG-2 Multiplex,” was approved by the Technology Group on Distribution on 23 February 1995, and by the full ATSC membership on 12 April 1995. Annex B, “AC-3 Data Stream in IEC958 Interface,” and Annex C, “AC-3 Karaoke Mode,” were approved by the Technology Group on Distribution on 24 October 1995 and by the full ATSC Membership on 20 December 1995. Revision A of this standard was approved by the full ATSC membership on 20 August 2001. Revision A corrected some errata in the detailed specifications, revised Annex A to include additional information about the DVB standard, removed Annex B that described an interface specification (superseeded by IEC and SMPTE standards), and added a new annex, “Alternate Bit Stream Syntax,” which contributes (in a compatible fashion) some new features to the AC-3 bit stream. Revision B of this standard was approved by the full ATSC membership on 14 June 2005. Revision B corrected some errata in the detailed specifications, and added a new annex, 1. The JCIC is presently composed of: the Electronic Industries Association (EIA), the Institute of Electrical and Electronic Engineers (IEEE), the National Association of Broadcasters (NAB), the National Cable Television Association (NCTA), and the Society of Motion Picture and Television Engineers (SMPTE). Page 19 Advanced Television Systems Committee, Inc. Document A/52:2010 “Enhanced AC-3 Bit Stream Syntax” which specifies a non-backwards compatible syntax that offers additional coding tools and features. Informative references were removed from the body of the document and placed in a new Annex B. Note: Revision A of this standard removed the informative annex “AC-3 Data Stream in IEC958 Interface” (Annex B). With this action, the former Annex C “AC-3 Karaoke Mode” became Annex B, and a new annex, “Alternate Bit Stream Syntax” became Annex C. Revision B of this standard restored the Annex “AC-3 Karaoke Mode” to its original designation of Annex C, moved the informative references to a bibliograpy in a new Annex B, changed the designation of the Annex “Alternate Bit Stream Syntax” to Annex D, and added a new Annex E, “Enhanced AC-3 Bit Stream Syntax,” documenting an enhanced syntax for audio coding (E-AC-3). ATSC Standard A/53, “Digital Television Standard”, references this document and describes how the audio coding algorithm described herein is applied in the ATSC DTV standard. The DVB/ETSI TS 101 154 document describes how AC-3 is applied in the DVB DTV standard. 2.1 Motivation In order to more efficiently broadcast or record audio signals, the amount of information required to represent the audio signals may be reduced. In the case of digital audio signals, the amount of digital information needed to accurately reproduce the original pulse code modulation (PCM) samples may be reduced by applying a digital compression algorithm, resulting in a digitally compressed representation of the original signal. (The term compression used in this context means the compression of the amount of digital information which must be stored or recorded, and not the compression of dynamic range of the audio signal.) The goal of the digital compression algorithm is to produce a digital representation of an audio signal which, when decoded and reproduced, sounds the same as the original signal, while using a minimum of digital information (bit-rate) for the compressed (or encoded) representation. The AC-3 digital compression algorithm specified in this document can encode from one to five full bandwidth audio channels, along with a low frequency enhancement channel. The six channels of source audio can be encoded from a PCM representation into a serial bit stream at data rates ranging from 32 kbps to 640 kbps. When all six channels are present this is referred to as 5.1 channels. The 0.1 channel refers to a fractional bandwidth channel intended to convey only low frequency (subwoofer) signals. While a wide range of encoded bit-rates is supported by this standard, a typical application of the algorithm is shown in Figure 2.1. In this example, a 5.1 channel audio program is converted from a PCM representation requiring more than 5 Mbps (6 channels × 48 kHz × 18 bits = 5.184 Mbps) into a 384 kbps serial bit stream by the AC-3 encoder. Satellite transmission equipment converts this bit stream to an RF transmission which is directed to a satellite transponder. The amount of bandwidth and power required by the transmission has been reduced by more than a factor of 13 by the AC-3 digital compression. The signal received from the satellite is demodulated back into the 384 kbps serial bit stream, and decoded by the AC-3 decoder. The result is the original 5.1 channel audio program. Digital compression of audio is useful wherever there is an economic benefit to be obtained by reducing the amount of digital information required to represent the audio. Typical 20 Digital Audio Compression Standard 22 November 2010 Transmission Input Audio Signals Left Encoded Center Bit-Stream Modulated Right 384 kb/s Transmission Signal Left Surround AC-3 Encoder Equipment Right Surround Low Frequency Satellite Dish Effects Reception Output Audio Signals Left Encoded Modulated Bit-Stream Center Signal 384 kb/s Right Reception AC-3 Decoder Left Surround Equipment Right Surround Satellite Dish Low Frequency Effects Figure 2.1 Example application of AC-3 to satellite audio transmission. applications are in satellite or terrestrial audio broadcasting, delivery of audio over metallic or optical cables, or storage of audio on magnetic, optical, semiconductor, or other storage media. 2.2 Encoding The AC-3 encoder accepts PCM audio and produces an encoded bit stream consistent with this standard. The specifics of the audio encoding process are not normative requirements of this standard. Nevertheless, the encoder must produce a bit stream matching the syntax described in Section 5, which, when decoded according to Sections 6 and 7, produces audio of sufficient quality for the intended application. Section 8 contains informative information on the encoding process. The encoding process is briefly described below. The AC-3 algorithm achieves high coding gain (the ratio of the input bit-rate to the output bit- rate) by coarsely quantizing a frequency domain representation of the audio signal. A block diagram of this process is shown in Figure 2.2. The first step in the encoding process is to transform the representation of audio from a sequence of PCM time samples into a sequence of blocks of frequency coefficients. This is done in the analysis filter bank. Overlapping blocks of 512 time samples are multiplied by a time window and transformed into the frequency domain. Due to the overlapping blocks, each PCM input sample is represented in two sequential transformed blocks. The frequency domain representation may then be decimated by a factor of two so that each block contains 256 frequency coefficients. The individual frequency coefficients are represented in binary exponential notation as a binary exponent and a mantissa. The set of exponents is encoded into a coarse representation of the signal spectrum which is referred to as the spectral envelope. This spectral envelope is used by the core bit allocation routine, which determines how many bits to use to encode each individual mantissa. The spectral envelope and the coarsely quantized mantissas for six audio blocks (1536 audio samples per channel) are formatted into an AC-3 frame. The AC-3 bit stream is a sequence of AC-3 frames. 21 Advanced Television Systems Committee, Inc. Document A/52:2010 Figure 2.2 The AC-3 encoder. The actual AC-3 encoder is more complex than indicated in Figure 2.2. The following functions not shown above are also included: 1. A frame header is attached which contains information (bit-rate, sample rate, number of encoded channels, etc.) required to synchronize to and decode the encoded bit stream. 2. Error detection codes are inserted in order to allow the decoder to verify that a received frame of data is error free. 3. The analysis filterbank spectral resolution may be dynamically altered so as to better match the time/frequency characteristic of each audio block. 4. The spectral envelope may be encoded with variable time/frequency resolution. 5. A more complex bit allocation may be performed, and parameters of the core bit allocation routine modified so as to produce a more optimum bit allocation. 6. The channels may be coupled together at high frequencies in order to achieve higher coding gain for operation at lower bit-rates. 7. In the two-channel mode, a rematrixing process may be selectively performed in order to provide additional coding gain, and to allow improved results to be obtained in the event that the two-channel signal is decoded with a matrix surround decoder. 2.3 Decoding The decoding process is basically the inverse of the encoding process. The decoder, shown in Figure 2.3, must synchronize to the encoded bit stream, check for errors, and de-format the various types of data such as the encoded spectral envelope and the quantized mantissas. The bit allocation routine is run and the results used to unpack and de-quantize the mantissas. The spectral envelope is decoded to produce the exponents. The exponents and mantissas are transformed back into the time domain to produce the decoded PCM time samples. The actual AC-3 decoder is more complex than indicated in Figure 2.3. The following functions not shown above are included: 22 Digital Audio Compression Standard 22 November 2010 Figure 2.3 The AC-3 decoder. 1. Error concealment or muting may be applied in case a data error is detected. 2. Channels which have had their high-frequency content coupled together must be de-coupled. 3. Dematrixing must be applied (in the 2-channel mode) whenever the channels have been rematrixed. 4. The synthesis filterbank resolution must be dynamically altered in the same manner as the encoder analysis filter bank had been during the encoding process. 3. REFERENCES At the time of publication, the editions indicated were valid. All referenced documents are subject to revision, and users of this Standard are encouraged to investigate the possibility of applying the most recent edition of the referenced document. 3.1 Normative References The following documents, in whole or in part, as referenced in this document, contain specific provisions that are to be followed strictly in order to implement a provision of this Standard. [1] ISO/IEC: ISO/IEC IS 13818-1, “Information technology – Generic coding of moving pictures and associated audio information: Systems”, 2007. [2] ISO: ISO 639-2, “Code for the representation of Names of Languages - Part 2: Alpha-3 code,” as maintained by the ISO 639/Joint Advisory Committee (ISO 639/JAC), http:// www.loc.gov/standards/iso639-2/iso639jac.html; ISO 639-2 standard online: http:// www.loc.gov/standards/iso639-2/langhome.html. [3] ISO/IEC: ISO/IEC 8859-1:1998, “Information technology -- 8-bit single-byte coded graphic character sets -- Part 1: Latin alphabet No. 1.” 3.2 Informative References The following documents contain information that may be helpful in applying this Standard. [4] ATSC: “Digital Television Standard: Part 1 - Digital Television System,” Doc. A/53 Part 1:2009, Advanced Television Systems Committee, Washington, D.C., 7 August 2009. 23 Advanced Television Systems Committee, Inc. Document A/52:2010 [5] ATSC: “Digital Television Standard: Part 3 - Service Multiplex and Transport SubsystemCharacteristics,” Doc. A/53 Part 3:2009, Advanced Television Systems Committee, Washington, D.C., 7 August 2009. [6] ATSC: “Digital Television Standard: Part 5 - AC-3 Audio System Characteristics,” Doc. A/ 53 Part 5:2010, Advanced Television Systems Committee, Washington, D.C., 6 July 2010. [7] ATSC: “Digital Television Standard: Part 6 - Enhanced AC-3 Audio System Characteristics,” Doc. A/53 Part 6:2010, Advanced Television Systems Committee, Washington, D.C., 6 July 2010. [8] DVB/ETSI: TS 101 154 V1.9.1, “Specification for the use of Video and Audio Coding in Broadcasting Applications based on the MPEG-2 Transport Stream,” 2009-09. [9] ETSI: TS 102 366 V1.2.1, “Digital Audio Compression (AC-3, Enhanced AC-3) Standard,” 2008-08. [10] ITU: ITU-R BT.1300-3, “Service multiplex, transport, and identification methods for digital terrestrial television broadcasting,” 2005. [11] DVB/ETSI: EN 300 468 V1.9.1, “Specification for Service Information (SI) in DVB systems,” 2009-03. [12] ATSC: “Program and System Information Protocol for Terrestrial Broadcast and Cable (PSIP),” Doc. A/65:2009, Advanced Television Systems Committee, Washington, D.C., 14 April 2009. 4. NOTATION, DEFINITIONS, AND TERMINOLOGY 4.1 Compliance Notation This section defines compliance terms for use by this document: shall – This word indicates specific provisions that are to be followed strictly (no deviation is permitted). shall not – This phrase indicates specific provisions that are absolutely prohibited. should – This word indicates that a certain course of action is preferred but not necessarily required. should not – This phrase means a certain possibility or course of action is undesirable but not prohibited. 4.2 Definitions A number of terms are used in this document. Below are definitions that explain the meaning of some of the terms used. audio block – A set of 512 audio samples consisting of 256 samples of the preceding audio block, and 256 new time samples. A new audio block occurs every 256 audio samples. Each audio sample is represented in two audio blocks. bin – The number of the frequency coefficient, as in frequency bin number n. The 512 point TDAC transform produces 256 frequency coefficients or frequency bins. 24 Digital Audio Compression Standard 22 November 2010 coefficient – The time domain samples are converted into frequency domain coefficients by the transform. coupled channel – A full bandwidth channel whose high frequency information is combined into the coupling channel. coupling band – A band of coupling channel transform coefficients covering one or more coupling channel sub-bands. coupling channel – The channel formed by combining the high frequency information from the coupled channels. coupling sub-band – A sub-band consisting of a group of 12 coupling channel transform coefficients. downmixing – Combining (or mixing down) the content of n original channels to produce m channels, where m < n. exponent set – The set of exponents for an independent channel, for the coupling channel, or for the low frequency portion of a coupled channel. full bandwidth (fbw) channel – An audio channel capable of full audio bandwidth. All channels (left, center, right, left surround, right surround) except the lfe channel are fbw channels. independent channel – A channel whose high frequency information is not combined into the coupling channel. (The lfe channel is always independent.) low frequency effects (lfe) channel – An optional single channel of limited (<120 Hz) bandwidth, which is intended to be reproduced at a level +10 dB with respect to the fbw channels. The optional lfe channel allows high sound pressure levels to be provided for low frequency sounds. spectral envelope – A spectral estimate consisting of the set of exponents obtained by decoding the encoded exponents. Similar (but not identical) to the original set of exponents. synchronization frame – A unit of the serial bit stream capable of being fully decoded. The synchronization frame begins with a sync code and contains 1536 coded audio samples. window – A time vector which is multiplied by an audio block to provide a windowed audio block. The window shape establishes the frequency selectivity of the filterbank, and provides for the proper overlap/add characteristic to avoid blocking artifacts. 4.3 Terminology Abbreviations A number of abbreviations are used to refer to elements employed in the AC-3 format. The following list is a cross reference from each abbreviation to the terminology which it represents. For most items, a reference to further information is provided. This document makes extensive use of these abbreviations. The abbreviations are lower case with a maximum length of 12 characters, and are suitable for use in either high level or assembly language computer software coding. Those who implement this standard are encouraged to use these same abbreviations in 25 Advanced Television Systems Committee, Inc. Document A/52:2010 any computer source code, or other hardware or software implementation documentation. Table 4.1 lists the abbreviations used in this document, their terminology and section reference. Table 4.1 ATSC Digital Audio Compression Standard Terms Abbreviation Terminology Reference acmod audio coding mode Section 5.4.2.3 addbsi additional bit stream information Section 5.4.2.31 addbsie additional bit stream information exists Section 5.4.2.29 addbsil additional bit stream information length Section 5.4.2.30 audblk audio block Section 5.4.3 audprodie audio production information exists Section 5.4.2.13 audprodi2e audio production information exists, ch2 Section 5.4.2.21 auxbits auxiliary data bits Section 5.4.4.1 auxdata auxiliary data field Section 5.4.4.1 auxdatae auxiliary data exists Section 5.4.4.3 auxdatal auxiliary data length Section 5.4.4.2 baie bit allocation information exists Section 5.4.3.30 bap bit allocation pointer bin frequency coefficient bin in index [bin] Section 5.4.3.13 blk block in array index [blk] blksw block switch flag Section 5.4.3.1 bnd band in array index [bnd] bsi bit stream information Section 5.4.2 bsid bit stream identification Section 5.4.2.1 bsmod bit stream mode Section 5.4.2.2 ch channel in array index [ch] chbwcod channel bandwidth code Section 5.4.3.24 chexpstr channel exponent strategy Section 5.4.3.22 chincpl channel in coupling Section 5.4.3.9 chmant channel mantissas Section 5.4.3.61 clev center mixing level coefficient Section 5.4.2.4 cmixlev center mix level Section 5.4.2.4 compr compression gain word Section 5.4.2.10 compr2 compression gain word, ch2 Section 5.4.2.18 compre compression gain word exists Section 5.4.2.9 compr2e compression gain word exists, ch2 Section 5.4.2.17 copyrightb copyright bit Section 5.4.2.24 cplabsexp coupling absolute exponent Section 5.4.3.25 cplbegf coupling begin frequency code Section 5.4.3.1 cplbndstrc coupling band structure Section 5.4.3.13 cplco coupling coordinate Section 7.4.3 cplcoe coupling coordinates exist Section 5.4.3.14 cplcoexp coupling coordinate exponent Section 5.4.3.16 cplcomant coupling coordinate mantissa Section 5.4.3.17 26 Digital Audio Compression Standard 22 November 2010 Table 4.1 ATSC Digital Audio Compression Standard Terms (Continued) Abbreviation Terminology Reference cpldeltba coupling dba Section 5.4.3.53 cpldeltbae coupling dba exists Section 5.4.3.48 cpldeltlen coupling dba length Section 5.4.3.52 cpldeltnseg coupling dba number of segments Section 5.4.3.50 cpldeltoffst coupling dba offset Section 5.4.3.51 cplendf coupling end frequency code Section 5.4.3.12 cplexps coupling exponents Section 5.4.3.26 cplexpstr coupling exponent strategy Section 5.4.3.21 cplfgaincod coupling fast gain code Section 5.4.3.39 cplfleak coupling fast leak initialization Section 5.4.3.45 cplfsnroffst coupling fine SNR offset Section 5.4.3.38 cplinu coupling in use Section 5.4.3.8 cplleake coupling leak initialization exists Section 5.4.3.44 cplmant coupling mantissas Section 5.4.3.61 cplsleak coupling slow leak initialization Section 5.4.3.46 cplstre coupling strategy exists Section 5.4.3.7 crc1 crc - cyclic redundancy check word 1 Section 5.4.1.2 crc2 crc - cyclic redundancy check word 2 Section 5.4.5.2 crcrsv crc reserved bit Section 5.4.5.1 csnroffst coarse SNR offset Section 5.4.3.37 d15 d15 exponent coding mode Section 5.4.3.21 d25 d25 exponent coding mode Section 5.4.3.21 d45 d45 exponent coding mode Section 5.4.3.21 dba delta bit allocation Section 5.4.3.47 dbpbcod dB per bit code Section 5.4.3.34 deltba channel dba Section 5.4.3.57 deltbae channel dba exists Section 5.4.3.49 deltbaie dba information exists Section 5.4.3.47 deltlen channel dba length Section 5.4.3.56 deltnseg channel dba number of segments Section 5.4.3.54 deltoffst channel dba offset Section 5.4.3.55 dialnorm dialogue normalization word Section 5.4.2.8 dialnorm2 dialogue normalization word, ch2 Section 5.4.2.16 dithflag dither flag Section 5.4.3.2 dsurmod Dolby surround mode Section 5.4.2.6 dynrng dynamic range gain word Section 5.4.3.4 dynrng2 dynamic range gain word, ch2 Section 5.4.3.6 dynrnge dynamic range gain word exists Section 5.4.3.3 dynrng2e dynamic range gain word exists, ch2 Section 5.4.3.5 exps channel exponents Section 5.4.3.27 fbw full bandwidth fdcycod fast decay code Section 5.4.3.32 27 Advanced Television Systems Committee, Inc. Document A/52:2010 Table 4.1 ATSC Digital Audio Compression Standard Terms (Continued) Abbreviation Terminology Reference fgaincod channel fast gain code Section 5.4.3.41 floorcod masking floor code Section 5.4.3.35 floortab masking floor table Section 7.2.2.7 frmsizecod frame size code Section 5.4.1.4 fscod sampling frequency code Section 5.4.1.3 fsnroffst channel fine SNR offset Section 5.4.3.40 gainrng channel gain range code Section 5.4.3.28 grp group in index [grp] langcod language code Section 5.4.2.12 langcod2 language code, ch2 Section 5.4.2.20 langcode language code exists Section 5.4.2.11 langcod2e language code exists, ch2 Section 5.4.2.19 lfe low frequency effects lfeexps lfe exponents Section 5.4.3.29 lfeexpstr lfe exponent strategy Section 5.4.3.23 lfefgaincod lfe fast gain code Section 5.4.3.43 lfefsnroffst lfe fine SNR offset Section 5.4.3.42 lfemant lfe mantissas Section 5.4.3.63 lfeon lfe on Section 5.4.2.7 mixlevel mixing level Section 5.4.2.14 mixlevel2 mixing level, ch2 Section 5.4.2.22 mstrcplco master coupling coordinate Section 5.4.3.15 nauxbits number of auxiliary bits Section 5.4.4.1 nchans number of channels Section 5.4.2.3 nchgrps number of fbw channel exponent groups Section 5.4.3.27 nchmant number of fbw channel mantissas Section 5.4.3.61 ncplbnd number of structured coupled bands Section 5.4.3.13 ncplgrps number of coupled exponent groups Section 5.4.3.26 ncplmant number of coupled mantissas Section 5.4.3.62 ncplsubnd number of coupling sub-bands Section 5.4.3.12 nfchans number of fbw channels Section 5.4.2.3 nlfegrps number of lfe channel exponent groups Section 5.4.3.29 nlfemant number of lfe channel mantissas Section 5.4.3.63 origbs original bit stream Section 5.4.2.25 phsflg phase flag Section 5.4.3.18 phsflginu phase flags in use Section 5.4.3.10 rbnd rematrix band in index [rbnd] rematflg rematrix flag Section 5.4.3.20 rematstr rematrixing strategy Section 5.4.3.19 roomtyp room type Section 5.4.2.15 roomtyp2 room type, ch2 Section 5.4.2.23 sbnd sub-band in index [sbnd] 28 Digital Audio Compression Standard 22 November 2010 Table 4.1 ATSC Digital Audio Compression Standard Terms (Continued) Abbreviation Terminology Reference sdcycod slow decay code Section 5.4.3.31 seg segment in index [seg] sgaincod slow gain code Section 5.4.3.33 skipfld skip field Section 5.4.3.60 skipl skip length Section 5.4.3.59 skiple skip length exists Section 5.4.3.58 slev surround mixing level coefficient Section 5.4.2.5 snroffste SNR offset exists Section 5.4.3.36 surmixlev surround mix level Section 5.4.2.5 syncframe synchronization frame Section 5.1 syncinfo synchronization information Section 5.3.1 syncword synchronization word Section 5.4.1.1 tdac time division aliasing cancellation timecod1 time code first half Section 5.4.2.27 timecod2 time code second half Section 5.4.2.28 timecod1e time code first half exists Section 5.4.2.26 timecod2e time code second half exists Section 5.4.2.26 5. BIT STREAM SYNTAX 5.1 Synchronization Frame An AC-3 serial coded audio bit stream is made up of a sequence of synchronization frames (see Figure 5.1). Each synchronization frame contains 6 coded audio blocks (AB), each of which represent 256 new audio samples per channel. A synchronization information (SI) header at the beginning of each frame contains information needed to acquire and maintain synchronization. A bit stream information (BSI) header follows SI, and contains parameters describing the coded audio service. The coded audio blocks may be followed by an auxiliary data (Aux) field. At the end of each frame is an error check field that includes a CRC word for error detection. An additional CRC word is located in the SI header, the use of which, by a decoder, is optional. 5.2 Semantics of Syntax Specification The following tables describe the order of arrival of information within the bit stream. The information contained in the tables is roughly based on C language syntax, but simplified for ease of reading. For bit stream elements that are larger than 1-bit, the order of the bits in the serial bit stream is either most-significant-bit-first (for numerical values), or left-bit-first (for bit-field Figure 5.1 AC-3 synchronization frame. 29 Advanced Television Systems Committee, Inc. Document A/52:2010 values). Fields or elements contained in the bit stream are indicated with bold type. Syntactic elements are typographically distinguished by the use of a different font (e.g., dynrng). Some AC-3 bit stream elements naturally form arrays. This syntax specification treats all bit stream elements individually, whether or not they would naturally be included in arrays. Arrays are thus described as multiple elements (as in blksw[ch] as opposed to simply blksw or blksw[]), and control structures such as for loops are employed to increment the index ([ch] for channel in this example). 5.3 Syntax Specification A continuous audio bit stream would consist of a sequence of synchronization frames: Syntax AC-3_bitstream() { while(true) { syncframe() ; } } /* end of AC-3 bit stream */ The syncframe consists of the syncinfo and bsi fields, the 6 coded audblk fields, the auxdata field, and the errorcheck field. Syntax syncframe() { syncinfo() ; bsi() ; for (blk = 0; blk < 6; blk++) { audblk() ; } auxdata() ; errorcheck() ; } /* end of syncframe */ Each of the bit stream elements, and their length, are itemized in the following tables. Note that all bit stream elements arrive most significant bit first, or left bit first, in time. 5.3.1 syncinfo: Synchronization Information 30 Digital Audio Compression Standard 22 November 2010 Table 5.1 syncinfo Syntax and Word Size Syntax Word Size syncinfo() { syncword 16 crc1 16 fscod 2 frmsizecod 6 } /* end of syncinfo */ 5.3.2 bsi: Bit Stream Information Table 5.2 bsi Syntax and Word Size Syntax Word Size bsi() { bsid 5 bsmod 3 acmod 3 if ((acmod & 0x1) && (acmod != 0x1)) /* if 3 front channels */ {cmixlev} 2 if (acmod & 0x4) /* if a surround channel exists */ {surmixlev} 2 if (acmod == 0x2) /* if in 2/0 mode */ {dsurmod} 2 lfeon 1 dialnorm 5 compre 1 if(compre) {compr} 8 langcode 1 if (langcode) {langcod} 8 audprodie 1 if (audprodie) { mixlevel 5 roomtyp 2 } if (acmod == 0) /* if 1+1 mode (dual mono, so some items need a second value) */ { dialnorm2 5 compr2e 1 if (compr2e) {compr2} 8 langcod2e 1 if (langcod2e) {langcod2} 8 audprodi2e 1 if (audprodi2e) { 31 Advanced Television Systems Committee, Inc. Document A/52:2010 Table 5.2 bsi Syntax and Word Size mixlevel2 5 roomtyp2 2 } } copyrightb 1 origbs 1 timecod1e 1 if (timecod1e) {timecod1} 14 timecod2e 1 if (timecod2e) {timecod2} 14 addbsie 1 if (addbsie) { addbsil 6 addbsi (addbsil+1)×8 } } /* end of bsi */ 5.3.3 audioblk: Audio Block Table 5.3 audioblk Syntax and Word Size Syntax Word Size audblk() { /* These fields for block switch and dither flags */ for (ch = 0; ch < nfchans; ch++) {blksw[ch]} 1 for (ch = 0; ch < nfchans; ch++) {dithflag[ch]} 1 /* These fields for dynamic range control */ dynrnge 1 if (dynrnge) {dynrng} 8 if (acmod == 0) /* if 1+1 mode */ { dynrng2e 1 if(dynrng2e) {dynrng2} 8 } /* These fields for coupling strategy information */ cplstre 1 if (cplstre) { cplinu 1 if (cplinu) { for (ch = 0; ch < nfchans; ch++) {chincpl[ch]} 1 32 Digital Audio Compression Standard 22 November 2010 Table 5.3 audioblk Syntax and Word Size (Continued) Syntax Word Size if (acmod == 0x2) {phsflginu} /* if in 2/0 mode */ 1 cplbegf 4 cplendf 4 /* ncplsubnd = 3 + cplendf - cplbegf */ for (bnd = 1; bnd < ncplsubnd; bnd++) {cplbndstrc[bnd]} 1 } } /* These fields for coupling coordinates, phase flags */ if (cplinu) { for (ch = 0; ch < nfchans; ch++) { if (chincpl[ch]) { cplcoe[ch] 1 if (cplcoe[ch]) { mstrcplco[ch] 2 /* ncplbnd derived from ncplsubnd, and cplbndstrc */ for (bnd = 0; bnd < ncplbnd; bnd++) { cplcoexp[ch][bnd] 4 cplcomant[ch][bnd] 4 } } } } if ((acmod == 0x2) && phsflginu && (cplcoe[0] || cplcoe[1])) { for (bnd = 0; bnd < ncplbnd; bnd++) {phsflg[bnd]} 1 } } /* These fields for rematrixing operation in the 2/0 mode */ if (acmod == 0x2) /* if in 2/0 mode */ { rematstr 1 if (rematstr) { if ((cplbegf > 2) || (cplinu == 0)) { for (rbnd = 0; rbnd < 4; rbnd++) {rematflg[rbnd]} 1 } 33 Advanced Television Systems Committee, Inc. Document A/52:2010 Table 5.3 audioblk Syntax and Word Size (Continued) Syntax Word Size if ((2 >= cplbegf > 0) && cplinu) { for (rbnd = 0; rbnd < 3; rbnd++) {rematflg[rbnd]} 1 } if ((cplbegf == 0) && cplinu) { for (rbnd = 0; rbnd < 2; rbnd++) {rematflg[rbnd]} 1 } } } /* These fields for exponent strategy */ if (cplinu) {cplexpstr} 2 for (ch = 0; ch < nfchans; ch++) {chexpstr[ch]} 2 if (lfeon) {lfeexpstr} 1 for (ch = 0; ch < nfchans; ch++) { if (chexpstr[ch] != reuse) { if (!chincpl[ch]) {chbwcod[ch]} 6 } } /* These fields for exponents */ if (cplinu) /* exponents for the coupling channel */ { if (cplexpstr != reuse) { cplabsexp 4 /* ncplgrps derived from ncplsubnd, cplexpstr */ for (grp = 0; grp< ncplgrps; grp++) {cplexps[grp]} 7 } } for (ch = 0; ch < nfchans; ch++) /* exponents for full bandwidth channels */ { if (chexpstr[ch] != reuse) { exps[ch][0] 4 /* nchgrps derived from chexpstr[ch], and cplbegf or chbwcod[ch] */ for (grp = 1; grp <= nchgrps[ch]; grp++) {exps[ch][grp]} 7 gainrng[ch] 2 } } if (lfeon) /* exponents for the low frequency effects channel */ 34 Digital Audio Compression Standard 22 November 2010 Table 5.3 audioblk Syntax and Word Size (Continued) Syntax Word Size { if (lfeexpstr != reuse) { lfeexps[0] 4 /* nlfegrps = 2 */ for (grp = 1; grp <= nlfegrps; grp++) {lfeexps[grp]} 7 } } /* These fields for bit-allocation parametric information */ baie 1 if (baie) { sdcycod 2 fdcycod 2 sgaincod 2 dbpbcod 2 floorcod 3 } snroffste 1 if (snroffste) { csnroffst 6 if (cplinu) { cplfsnroffst 4 cplfgaincod 3 } for (ch = 0; ch < nfchans; ch++) { fsnroffst[ch] 4 fgaincod[ch] 3 } if (lfeon) { lfefsnroffst 4 lfefgaincod 3 } } if (cplinu) { cplleake 1 if (cplleake) 35 Advanced Television Systems Committee, Inc. Document A/52:2010 Table 5.3 audioblk Syntax and Word Size (Continued) Syntax Word Size { cplfleak 3 cplsleak 3 } } /* These fields for delta bit allocation information */ deltbaie 1 if (deltbaie) { if (cplinu) {cpldeltbae} 2 for (ch = 0; ch < nfchans; ch++) {deltbae[ch]} 2 if (cplinu) { if (cpldeltbae==new info follows) { cpldeltnseg 3 for (seg = 0; seg <= cpldeltnseg; seg++) { cpldeltoffst[seg] 5 cpldeltlen[seg] 4 cpldeltba[seg] 3 } } } for (ch = 0; ch < nfchans; ch++) { if (deltbae[ch]==new info follows) { deltnseg[ch] 3 for (seg = 0; seg <= deltnseg[ch]; seg++) { deltoffst[ch][seg] 5 deltlen[ch][seg] 4 deltba[ch][seg] 3 } } } } /* These fields for inclusion of unused dummy data */ skiple 1 if (skiple) { 36 Digital Audio Compression Standard 22 November 2010 Table 5.3 audioblk Syntax and Word Size (Continued) Syntax Word Size skipl 9 skipfld skipl × 8 } /* These fields for quantized mantissa values */ got_cplchan = 0 for (ch = 0; ch < nfchans; ch++) { for (bin = 0; bin < nchmant[ch]; bin++) {chmant[ch][bin]} (0–16) if (cplinu && chincpl[ch] && !got_cplchan) { for (bin = 0; bin < ncplmant; bin++) {cplmant[bin]} (0–16) got_cplchan = 1 } } if (lfeon) /* mantissas of low frequency effects channel */ { for (bin = 0; bin < nlfemant; bin++) {lfemant[bin]} (0-16) } } /* end of audblk */ 5.3.4 auxdata: Auxiliary Data Table 5.4 auxdata Syntax and Word Size Syntax Word Size auxdata() { auxbits nauxbits if (auxdatae) { Auxdatal 14 } auxdatae 1 } /* end of auxdata */ 37 Advanced Television Systems Committee, Inc. Document A/52:2010 5.3.5 errorcheck: Error Detection Code Table 5.5 errorcheck Syntax and Word Size Syntax Word Size errorcheck() { crcrsv 1 crc2 16 } /* end of errorcheck */ 5.4 Description of Bit Stream Elements A number of bit stream elements have values which may be transmitted, but whose meaning has been reserved. If a decoder receives a bit stream which contains reserved values, the decoder may or may not be able to decode and produce audio. In the description of bit stream elements which have reserved codes, there is an indication of what the decoder can do if the reserved code is received. In some cases, the decoder can not decode audio. In other cases, the decoder can still decode audio by using a default value for a parameter which was indicated by a reserved code. 5.4.1 syncinfo: Synchronization Information 5.4.1.1 syncword: Synchronization Word, 16 bits The synchronization word, syncword, is always 0x0B77, or ‘0000 1011 0111 0111’. Transmission of syncword, like other bit field elements, is left bit first. 5.4.1.2 crc1: Cyclic Redundancy Check 1, 16 bits This 16 bit-CRC applies to the first 5/8 of the frame. Transmission of the CRC, like other numerical values, is most significant bit first. 5.4.1.3 fscod: Sample Rate Code, 2 bits This is a 2-bit code indicating sample rate according to Table 5.6. If the reserved code is indicated, the decoder should not attempt to decode audio and should mute. Table 5.6 Sample Rate Codes fscod Sampling Rate, kHz ‘00’ 48 ‘01’ 44.1 ‘10’ 32 ‘11’ reserved 5.4.1.4 frmsizecod: Frame Size Code, 6 bits The frame size code is used along with the sample rate code to determine the number of (2-byte) words before the next syncword. See Table 5.18. 38 Digital Audio Compression Standard 22 November 2010 5.4.2 bsi: Bit Stream Information 5.4.2.1 bsid: Bit Stream Identification, 5 bits This bit field shall have a value of ‘01000’ (= 8) when the stream_type is 0x81 unless the stream is constructed per one of the Annexs to this standard. The annexes to this standard define what other values signify and the degree of compatibility with decoders built to decode streams with bsid = 8. Thus, decoders built to this standard shall mute if the value of bsid is greater than 8 (unless the decoder is built in conformance with the optional provisions of Annex E), and should decode and reproduce audio if the value of bsid is less than or equal to 8. 5.4.2.2 bsmod: Bit Stream Mode, 3 bits This 3-bit code indicates the type of service that the bit stream conveys as defined in Table 5.7. Table 5.7 Bit Stream Mode bsmod acmod Type of Service ‘000’ any main audio service: complete main (CM) ‘001’ any main audio service: music and effects (ME) ‘010’ any associated service: visually impaired (VI) ‘011’ any associated service: hearing impaired (HI) ‘100’ any associated service: dialogue (D) ‘101’ any associated service: commentary (C) ‘110’ any associated service: emergency (E) ‘111’ ‘001’ associated service: voice over (VO) ‘111’ ‘010’ - ‘111’ main audio service: karaoke 5.4.2.3 acmod: Audio Coding Mode, 3 bits This 3-bit code, shown in Table 5.8, indicates which of the main service channels are in use, ranging from 3/2 to 1/0. If the msb of acmod is a ‘1’, surround channels are in use and surmixlev follows in the bit stream. If the msb of acmod is a ‘0’, the surround channels are not in use and surmixlev does not follow in the bit stream. If the lsb of acmod is a ‘0’, the center channel is not in use. If the lsb of acmod is a ‘1’, the center channel is in use. Note that the state of acmod sets the number of full-bandwidth channels parameter, nfchans, (e.g., for 3/2 mode, nfchans = 5; for 2/1 mode, nfchans = 3; etc.). The total number of channels, nchans, is equal to nfchans if the lfe channel is off, and is equal to 1 + nfchans if the lfe channel is on. If acmod is 0, then two completely independent program channels (dual mono) are encoded into the bit stream, and are referenced as Ch1, Ch2. In this case, a number of additional items are present in bsi or audblk to fully describe Ch2. Table 5.8 also indicates the channel ordering (the order in which the channels are processed) for each of the modes. 39 Advanced Television Systems Committee, Inc. Document A/52:2010 Table 5.8 Audio Coding Mode acmod Audio Coding Mode nfchans Channel Array Ordering ‘000’ 1+1 2 Ch1, Ch2 ‘001’ 1/0 1 C ‘010’ 2/0 2 L, R ‘011’ 3/0 3 L, C, R ‘100’ 2/1 3 L, R, S ‘101’ 3/1 4 L, C, R, S ‘110’ 2/2 4 L, R, SL, SR ‘111’ 3/2 5 L, C, R, SL, SR 5.4.2.4 cmixlev: Center Mix Level, 2 bits When three front channels are in use, this 2-bit code, shown in Table 5.9, indicates the nominal down mix level of the center channel with respect to the left and right channels. If cmixlev is set to the reserved code, decoders should still reproduce audio. The intermediate value of cmixlev (–4.5 dB) may be used in this case. Table 5.9 Center Mix Level cmixlev clev ‘00’ 0.707 (–3.0 dB) ‘01’ 0.595 (–4.5 dB) ‘10’ 0.500 (–6.0 dB) ‘11’ reserved 5.4.2.5 surmixlev: Surround Mix Level, 2 bits If surround channels are in use, this 2-bit code, shown in Table 5.10, indicates the nominal down mix level of the surround channels. If surmixlev is set to the reserved code, the decoder should still reproduce audio. The intermediate value of surmixlev (–6 dB) may be used in this case. Table 5.10 Surround Mix Level surmixlev slev ‘00’ 0.707 (–3 dB) ‘01’ 0.500 (–6 dB) ‘10’ 0 ‘11’ reserved 5.4.2.6 dsurmod: Dolby Surround Mode, 2 bits When operating in the two channel mode, this 2-bit code, as shown in Table 5.11, indicates whether or not the program has been encoded in Dolby Surround. This information is not used by the AC-3 decoder, but may be used by other portions of the audio reproduction equipment. If 40 Digital Audio Compression Standard 22 November 2010 dsurmodis set to the reserved code, the decoder should still reproduce audio. The reserved code may be interpreted as “not indicated”. Table 5.11 Dolby Surround Mode dsurmod Indication ‘00’ not indicated ‘01’ Not Dolby Surround encoded ‘10’ Dolby Surround encoded ‘11’ reserved 5.4.2.7 lfeon: Low Frequency Effects Channel On, 1 bit This bit has a value of 1 if the lfe (sub woofer) channel is on, and a value of 0 if the lfe channel is off. 5.4.2.8 dialnorm: Dialogue Normalization, 5 bits This 5-bit code indicates how far the average dialogue level is below digital 100 percent. Valid values are 1–31. The value of 0 is reserved. The values of 1 to 31 are interpreted as –1 dB to –31 dB with respect to digital 100 percent. If the reserved value of 0 is received, the decoder shall use –31 dB. The value of dialnorm shall affect the sound reproduction level. If the value is not used by the AC-3 decoder itself, the value shall be used by other parts of the audio reproduction equipment. Dialogue normalization is further explained in Section 7.6. 5.4.2.9 compre: Compression Gain Word Exists, 1 bit If this bit is a ‘1’, the following 8 bits represent a compression control word. 5.4.2.10 compr: Compression Gain Word, 8 bits This encoder-generated gain word may be present in the bit stream. If so, it may used to scale the reproduced audio level in order to reproduce a very narrow dynamic range, with an assured upper limit of instantaneous peak reproduced signal level in the monophonic downmix. The meaning and use of compr is described further in Section 7.7.2. 5.4.2.11 langcode: Language Code Exists, 1 bit If this bit is a ‘1’, the following 8 bits (i.e. the element langcod) shall be present in the bit stream. If this bit is a 0, the element langcod does not exist in the bit stream. 5.4.2.12 langcod: Language Code, 8 bits This is an 8 bit reserved value that shall be set to 0xFF if present. (This element was originally intended to carry an 8-bit value that would, via a table lookup, indicate the language of the audio program. Because modern delivery systems provide the ISO 639-2 language code in the signaling layer, indication of language within the AC-3 elementary stream was unnecessary, and so was removed from the AC-3 syntax to avoid confusion.) 5.4.2.13 audprodie: Audio Production Information Exists, 1 bit If this bit is a ‘1’, the mixlevel and roomtyp fields exist, indicating information about the audio production environment (mixing room). 41 Advanced Television Systems Committee, Inc. Document A/52:2010 5.4.2.14 mixlevel: Mixing Level, 5 bits This 5-bit code indicates the absolute acoustic sound pressure level of an individual channel during the final audio mixing session. The 5-bit code represents a value in the range 0 to 31. The peak mixing level is 80 plus the value of mixlevel dB SPL, or 80 to 111 dB SPL. The peak mixing level is the acoustic level of a sine wave in a single channel whose peaks reach 100 percent in the PCM representation. The absolute SPL value is typically measured by means of pink noise with an RMS value of –20 or –30 dB with respect to the peak RMS sine wave level. The value of mixlevel is not typically used within the AC-3 decoder, but may be used by other parts of the audio reproduction equipment. 5.4.2.15 roomtyp: Room Type, 2 bits This 2-bit code, shown in Table 5.12, indicates the type and calibration of the mixing room used for the final audio mixing session. The value of roomtyp is not typically used by the AC-3 decoder, but may be used by other parts of the audio reproduction equipment. If roomtyp is set to the reserved code, the decoder should still reproduce audio. The reserved code may be interpreted as “not indicated”. Table 5.12 Room Type roomtyp Type of Mixing Room ‘00’ not indicated ‘01’ large room, X curve monitor ‘10’ small room, flat monitor ‘11’ reserved 5.4.2.16 dialnorm2: Dialogue Normalization, Ch2, 5 bits This 5-bit code has the same meaning as dialnorm, except that it applies to the second audio channel when acmod indicates two independent channels (dual mono 1+1 mode). 5.4.2.17 compr2e: Compression Gain Word Exists, Ch2, 1 bit If this bit is a 1, the following 8 bits represent a compression gain word for Ch2. 5.4.2.18 compr2: Compression Gain Word, Ch2, 8 bits This 8-bit word has the same meaning as compr, except that it applies to the second audio channel when acmod indicates two independent channels (dual mono 1+1 mode). 5.4.2.19 langcod2e: Language Code Exists, Ch2, 1 bit If this bit is a ‘1’, the following 8 bits (i.e. the element langcod2) shall be present in the bit stream. If this bit is a 0, the element langcod2 does not exist in the bit stream. 5.4.2.20 langcod2: Language Code, Ch2, 8 bits This is an 8 bit reserved value that shall be set to 0xFF if present. See lancod, Section 5.4.2.12 above. 5.4.2.21 audprodi2e: Audio Production Information Exists, Ch2, 1 bit If this bit is a ‘1’, the following two data fields exist indicating information about the audio production for Ch2. 42 Digital Audio Compression Standard 22 November 2010 5.4.2.22 mixlevel2: Mixing Level, Ch2, 5 bits This 5-bit code has the same meaning as mixlevel, except that it applies to the second audio channel when acmod indicates two independent channels (dual mono 1+1 mode). 5.4.2.23 roomtyp2: Room Type, Ch2, 2 bits This 2-bit code has the same meaning as roomtyp, except that it applies to the second audio channel when acmod indicates two independent channels (dual mono 1+1 mode). 5.4.2.24 copyrightb: Copyright Bit, 1 bit If this bit has a value of ‘1’, the information in the bit stream is indicated as protected by copyright. It has a value of ‘0’ if the information is not indicated as protected. 5.4.2.25 origbs: Original Bit Stream, 1 bit This bit has a value of ‘1’ if this is an original bit stream. This bit has a value of ‘0’ if this is a copy of another bit stream. 5.4.2.26 timecod1e, timcode2e: Time Code (first and second) Halves Exist, 2 bits These values indicate, as shown in Table 5.13, whether time codes follow in the bit stream. The time code can have a resolution of 1/64th of a frame (one frame = 1/30th of a second). Since only the high resolution portion of the time code is needed for fine synchronization, the 28 bit time code is broken into two 14 bit halves. The low resolution first half represents the code in 8 second increments up to 24 hours. The high resolution second half represents the code in 1/64th frame increments up to 8 seconds. Table 5.13 Time Code Exists timecod2e,timecod1e Time Code Present ‘0’’0’ , not present ‘0’’1’ , first half (14 bits) present ‘1’’0’ , second half (14 bits) present ‘1’’1’ , both halves (28 bits) present 5.4.2.27 timecod1: Time Code First Half, 14 bits The first 5 bits of this 14-bit field represent the time in hours, with valid values of 0–23. The next 6 bits represent the time in minutes, with valid values of 0–59. The final 3 bits represents the time in 8 second increments, with valid values of 0–7 (representing 0, 8, 16, ... 56 seconds). 5.4.2.28 timecod2: Time Code Second Half, 14 bits The first 3 bits of this 14-bit field represent the time in seconds, with valid values from 0–7 (representing 0–7 seconds). The next 5 bits represents the time in frames, with valid values from 0–29. The final 6 bits represents fractions of 1/64 of a frame, with valid values from 0–63. 5.4.2.29 addbsie: Additional Bit Stream Information Exists, 1 bit If this bit has a value of 1 there is additional bit stream information, the length of which is indicated by the next field. If this bit has a value of 0, there is no additional bit stream information. 43 Advanced Television Systems Committee, Inc. Document A/52:2010 5.4.2.30 addbsil: Additional Bit Stream Information Length, 6 bits This 6-bit code, which exists only if addbside is a 1, indicates the length in bytes of additional bit stream information. The valid range of addbsil is 0–63, indicating 1–64 additional bytes, respectively. The decoder is not required to interpret this information, and thus shall skip over this number of bytes following in the data stream. 5.4.2.31 addbsi: Additional Bit Stream Information, [(addbsil+1) × 8] bits This field contains 1 to 64 bytes of any additional information included with the bit stream information structure. 5.4.3 audblk: Audio Block 5.4.3.1 blksw[ch]: Block Switch Flag, 1 bit This flag, for channel [ch], indicates whether the current audio block was split into 2 sub-blocks during the transformation from the time domain into the frequency domain. A value of 0 indicates that the block was not split, and that a single 512 point TDAC transform was performed. A value of 1 indicates that the block was split into 2 sub-blocks of length 256, that the TDAC transform length was switched from a length of 512 points to a length of 256 points, and that 2 transforms were performed on the audio block (one on each sub-block). Transform length switching is described in more detail in Section 7.9. 5.4.3.2 dithflag[ch]: Dither Flag, 1 bit This flag, for channel [ch], indicates that the decoder should activate dither during the current block. Dither is described in detail in Section 7.3.4. 5.4.3.3 dynrnge: Dynamic Range Gain Word Exists, 1 bit If this bit is a ‘1’, the dynamic range gain word follows in the bit stream. If it is ‘0’, the gain word is not present, and the previous value is reused, except for block 0 of a frame where if the control word is not present the current value of dynrng is set to 0. 5.4.3.4 dynrng: Dynamic Range Gain Word, 8 bits This encoder-generated gain word is applied to scale the reproduced audio as described in Section 7.7.1. 5.4.3.5 dynrng2e: Dynamic Range Gain Word Exists, Ch2, 1 bit If this bit is a ‘1’, the dynamic range gain word for channel 2 follows in the bit stream. If it is ‘0’, the gain word is not present, and the previous value is reused, except for block 0 of a frame where if the control word is not present the current value of dynrng2 is set to 0. 5.4.3.6 dynrng2: Dynamic Range Gain Word Ch2, 8 bits This encoder-generated gain word is applied to scale the reproduced audio of Ch2, in the same manner as dynrng is applied to Ch1, as described in Section 7.7.1. 5.4.3.7 cplstre: Coupling Strategy Exists, 1 bit If this bit is a ‘1’, coupling information follows in the bit stream. If it is ‘0’, new coupling information is not present, and coupling parameters previously sent are reused. This parameter shall not be set to 0 in block 0. 44 Digital Audio Compression Standard 22 November 2010 5.4.3.8 cplinu: Coupling in Use, 1 bit If this bit is a ‘1’, coupling is currently being utilized, and coupling parameters follow. If it is ‘0’, coupling is not being utilized (all channels are independent) and no coupling parameters follow in the bit stream. 5.4.3.9 chincpl[ch]: Channel in Coupling, 1 bit If this bit is a ‘1’, then the channel indicated by the index [ch] is a coupled channel. If the bit is a ‘0’, then this channel is not coupled. Since coupling is not used in the 1/0 mode, if any chincpl[] values exist there will be 2 to 5 values. Of the values present, at least two values will be 1, since coupling requires more than one coupled channel to be coupled. 5.4.3.10 phsflginu: Phase Flags in Use, 1 bit If this bit (defined for 2/0 mode only) is a 1, phase flags are included with coupling coordinate information. Phase flags are described in Section 7.4. 5.4.3.11 cplbegf: Coupling Begin Frequency Code, 4 bits This 4-bit code is interpreted as the sub-band number (0 to 15) which indicates the lower frequency band edge of the coupling channel (or the first active sub-band) as shown in Table 7.24. 5.4.3.12 cplendf: Coupling End Frequency Code, 4 bits This 4-bit code indicates the upper band edge of the coupling channel. The upper band edge (or last active sub-band) is cplendf+2, or a value between 2 and 17. See Table 7.24. The number of active coupling sub-bands is equal to ncplsubnd, which is calculated as ncplsubnd = 3 + cplendf – cplbegf 5.4.3.13 cplbndstrc[sbnd]: Coupling Band Structure, 1 bit There are 18 coupling sub-bands defined in Table 7.24, each containing 12 frequency coefficients. The fixed 12-bin wide coupling sub-bands are converted into coupling bands, each of which may be wider than (a multiple of) 12 frequency bins. Each coupling band may contain one or more coupling sub-bands. Coupling coordinates are transmitted for each coupling band. Each band’s coupling coordinate must be applied to all the coefficients in the coupling band. The coupling band structure indicates which coupling sub-bands are combined into wider coupling bands. When cplbndstrc[sbnd] is a ‘0’, the sub-band number [sbnd] is not combined into the previous band to form a wider band, but starts a new 12 wide coupling band. When cplbndstrc[sbnd] is a ‘1’, then the sub-band [sbnd] is combined with the previous band, making the previous band 12 bins wider. Each successive value of cplbndstrc which is a 1 will continue to combine sub-bands into the current band. When another cplbndstrc value of 0 is received, then a new band will be formed, beginning with the 12 bins of the current sub-band. The set of cplbndstrc[sbnd] values is typically considered an array. Each bit in the array corresponds to a specific coupling sub-band in ascending frequency order. The first element of the array corresponds to the sub-band cplbegf, is always 0, and is not transmitted. (There is no reason to send a cplbndstrc bit for the first sub-band at cplbegf, since this bit would always be ‘0’.) Thus, there are ncplsubnd–1 values of cplbndstrc transmitted. If there is only one coupling sub-band, then no cplbndstrc bits are sent. The number of coupling bands, ncplbnd, may be computed from ncplsubnd and cplbndstrc 45 Advanced Television Systems Committee, Inc. Document A/52:2010 ncplbnd = (ncplsubnd – (cplbndstrc[1] + ... + cplbndstrc[ncplsubnd – 1])) 5.4.3.14 cplcoe[ch]: Coupling Coordinates Exist, 1 bit Coupling coordinates indicate, for a given channel and within a given coupling band, the fraction of the coupling channel frequency coefficients to use to re-create the individual channel frequency coefficients. Coupling coordinates are conditionally transmitted in the bit stream. If new values are not delivered, the previously sent values remain in effect. See Section 7.4 for further information on coupling. If cplcoe[ch] is ‘1’, the coupling coordinates for the corresponding channel [ch] exist and follow in the bit stream. If the bit is ‘0’, the previously transmitted coupling coordinates for this channel are reused. This parameter shall not be set to 0 in block 0, or in any block for which the corresponding channel is participating in coupling but was not participating in coupling in the previous block. 5.4.3.15 mstrcplco[ch]: Master Coupling Coordinate, 2 bits This per channel parameter establishes a per channel gain factor (increasing the dynamic range) for the coupling coordinates as shown in Table 5.14. Table 5.14 Master Coupling Coordinate mstrcplco[ch] cplco[ch][bnd] gain multiplier ‘00’ 1 ‘01’ 2-3 ‘10’ 2-6 ‘11’ 2-9 5.4.3.16 cplcoexp[ch][bnd]: Coupling Coordinate Exponent, 4 bits Each coupling coordinate is composed of a 4-bit exponent and a 4-bit mantissa. This element is the value of the coupling coordinate exponent for channel [ch] and band [bnd]. The index [ch] only will exist for those channels which are coupled. The index [bnd] will range from 0 to ncplbnds. See Section 7.4.3 for further information on how to interpret coupling coordinates. 5.4.3.17 cplcomant[ch][bnd]: Coupling Coordinate Mantissa, 4 bits This element is the 4-bit coupling coordinate mantissa for channel [ch] and band [bnd]. 5.4.3.18 phsflg[bnd]: Phase Flag, 1 bit This element (only used in the 2/0 mode) indicates whether the decoder should phase invert the coupling channel mantissas when reconstructing the right output channel. The index [bnd] can range from 0 to ncplbnd. Phase flags are described in Section 7.4. 5.4.3.19 rematstr: Rematrixing Strategy, 1 bit If this bit is a ‘1’, then new rematrix flags are present in the bit stream. If it is ‘0’, rematrix flags are not present, and the previous values should be reused. The rematstr parameter is present only in the 2/0 audio coding mode. This parameter shall not be set to ‘0’ in block 0. 46 Digital Audio Compression Standard 22 November 2010 5.4.3.20 rematflg[rbnd]: Rematrix Flag, 1 bit This bit indicates whether the transform coefficients in rematrixing band [rbnd] have been rematrixed. If this bit is a ‘1’, then the transform coefficients in [rbnd] were rematrixed into sum and difference channels. If this bit is a ‘0’, then rematrixing has not been performed in band [rbnd]. The number of rematrixing bands (and the number of values of [rbnd]) depend on coupling parameters as shown in Table 5.15. Rematrixing is described in Section 7.5. Table 5.15 Number of Rematrixing Bands Condition No. of Rematrixing Bands cplinu == 0 4 (cplinu == 1) && (cplbegf > 2) 4 (cplinu == 1) && (2 ≥ cplbegf > 0) 3 (cplinu == 1) && (cplbegf == 0) 2 5.4.3.21 cplexpstr: Coupling Exponent Strategy, 2 bits This element indicates the method of exponent coding that is used for the coupling channel as shown in Table 7.4. See Section 7.1 for explanation of each exponent strategy. This parameter shall not be set to 0 in block 0, or in any block for which coupling is enabled but was disabled in the previous block. 5.4.3.22 chexpstr[ch]: Channel Exponent Strategy, 2 bits This element indicates the method of exponent coding that is used for channel [ch], as shown in Table 7.4. This element exists for each full bandwidth channel. This parameter shall not be set to 0 in block 0. 5.4.3.23 lfeexpstr: Low Frequency Effects Channel Exponent Strategy, 1 bit This element indicates the method of exponent coding that is used for the lfe channel, as shown in Table 7.5. This parameter shall not be set to 0 in block 0. 5.4.3.24 chbwcod[ch]: Channel Bandwidth Code, 6 bits The chbwcod[ch] element is an unsigned integer which defines the upper band edge for full- bandwidth channel [ch]. This parameter is only included for fbw channels which are not coupled. (See Section 7.1.3 on exponents for the definition of this parameter.) Valid values are in the range of 0–60. If a value greater than 60 is received, the bit stream is invalid and the decoder shall cease decoding audio and mute. 5.4.3.25 cplabsexp: Coupling Absolute Exponent, 4 bits This is an absolute exponent, which is used as a reference when decoding the differential exponents for the coupling channel. 5.4.3.26 cplexps[grp]: Coupling Exponents, 7 bits Each value of cplexps indicates the value of 3, 6, or 12 differentially-coded coupling channel exponents for the coupling exponent group [grp] for the case of D15, D25, or D45 coding, respectively. The number of cplexps values transmitted equals ncplgrps, which may be determined from cplbegf, cplendf, and cplexpstr. Refer to Section 7.1.3 for further information. 47 Advanced Television Systems Committee, Inc. Document A/52:2010 5.4.3.27 exps[ch][grp]: Channel Exponents, 4 or 7 bits These elements represent the encoded exponents for channel [ch]. The first element ([grp] = 0) is a 4-bit absolute exponent for the first (DC term) transform coefficient. The subsequent elements ([grp]>0) are 7-bit representations of a group of 3, 6, or 12 differentially coded exponents (corresponding to D15, D25, D45 exponent strategies respectively). The number of groups for each channel, nchgrps[ch], is determined from cplbegf if the channel is coupled, or chbwcod[ch] of the channel is not coupled. Refer to Section 7.1.3 for further information. 5.4.3.28 gainrng[ch]: Channel Gain Range Code, 2 bits This per channel 2-bit element may be used to determine a block floating-point shift value for the inverse TDAC transform filterbank. Use of this code allows increased dynamic range to be obtained from a limited word length transform computation. For further information see Section 7.9.5. 5.4.3.29 lfeexps[grp]: Low Frequency Effects Channel Exponents, 4 or 7 bits These elements represent the encoded exponents for the LFE channel. The first element ([grp] = 0) is a 4-bit absolute exponent for the first (dc term) transform coefficient. There are two additional elements (nlfegrps = 2) which are 7-bit representations of a group of 3 differentially coded exponents. The total number of lfe channel exponents (nlfemant) is 7. 5.4.3.30 baie: Bit Allocation Information Exists, 1 bit If this bit is a ‘1’, then five separate fields (totaling 11 bits) follow in the bit stream. Each field indicates parameter values for the bit allocation process. If this bit is a ‘0’, these fields do not exist. Further details on these fields may be found in Section 7.2. This parameter shall not be set to ‘0’ in block 0. 5.4.3.31 sdcycod: Slow Decay Code, 2 bits This 2-bit code specifies the slow decay parameter in the bit allocation process. 5.4.3.32 fdcycod: Fast Decay Code, 2 bits This 2-bit code specifies the fast decay parameter in the decode bit allocation process. 5.4.3.33 sgaincod: Slow Gain Code, 2 bits This 2-bit code specifies the slow gain parameter in the decode bit allocation process. 5.4.3.34 dbpbcod: dB per Bit Code, 2 bits This 2-bit code specifies the dB per bit parameter in the bit allocation process. 5.4.3.35 floorcod: Masking Floor Code, 3 bits This 3-bit code specifies the floor code parameter in the bit allocation process. 5.4.3.36 snroffste: SNR Offset Exists, 1 bit If this bit has a value of 1, a number of bit allocation parameters follow in the bit stream. If this bit has a value of 0, SNR offset information does not follow, and the previously transmitted values should be used for this block. The bit allocation process and these parameters are described in Section 7.2.2. This parameter shall not be set to 0 in block 0. 48 Digital Audio Compression Standard 22 November 2010 5.4.3.37 csnroffst: Coarse SNR Offset, 6 bits This 6-bit code specifies the coarse SNR offset parameter in the bit allocation process. 5.4.3.38 cplfsnroffst: Coupling Fine SNR Offset, 4 bits This 4-bit code specifies the coupling channel fine SNR offset in the bit allocation process. 5.4.3.39 cplfgaincod: Coupling Fast Gain Code, 3 bits This 3-bit code specifies the coupling channel fast gain code used in the bit allocation process. 5.4.3.40 fsnroffst[ch]: Channel Fine SNR Offset, 4 bits This 4-bit code specifies the fine SNR offset used in the bit allocation process for channel [ch]. 5.4.3.41 fgaincod[ch]: Channel Fast Gain Code, 3 bits This 3-bit code specifies the fast gain parameter used in the bit allocation process for channel [ch]. 5.4.3.42 lfefsnroffst: Low Frequency Effects Channel Fine SNR Offset, 4 bits This 4-bit code specifies the fine SNR offset parameter used in the bit allocation process for the lfe channel. 5.4.3.43 lfefgaincod: Low Frequency Effects Channel Fast Gain Code, 3 bits This 3-bit code specifies the fast gain parameter used in the bit allocation process for the lfe channel. 5.4.3.44 cplleake: Coupling Leak Initialization Exists, 1 bit If this bit is a ‘1’, leak initialization parameters follow in the bit stream. If this bit is a ‘0’, the previously transmitted values still apply. This parameter shall not be set to ‘0’ in block 0, or in any block for which coupling is enabled but was disabled in the previous block. 5.4.3.45 cplfleak: Coupling Fast Leak Initialization, 3 bits This 3-bit code specifies the fast leak initialization value for the coupling channel's excitation function calculation in the bit allocation process. 5.4.3.46 cplsleak: Coupling Slow Leak Initialization, 3 bits This 3-bit code specifies the slow leak initialization value for the coupling channel's excitation function calculation in the bit allocation process. 5.4.3.47 deltbaie: Delta Bit Allocation Information Exists, 1 bit If this bit is a ‘1’, some delta bit allocation information follows in the bit stream. If this bit is a ‘0’, the previously transmitted delta bit allocation information still applies, except for block 0. If deltbaie is ‘0’ in block 0, then cpldeltbae and deltbae[ch] are set to the binary value ‘10’, and no delta bit allocation is applied. Delta bit allocation is described in Section 7.2.2.6. 5.4.3.48 cpldeltbae: Coupling Delta Bit Allocation Exists, 2 bits This 2-bit code indicates the delta bit allocation strategy for the coupling channel, as shown in Table 5.16. If the reserved state is received, the decoder should not decode audio, and should 49 Advanced Television Systems Committee, Inc. Document A/52:2010 mute. This parameter shall not be set to ‘00’ in block 0, or in any block for which coupling is enabled but was disabled in the previous block. Table 5.16 Delta Bit Allocation Exists States cpldeltbae, deltbae Code ‘00’ reuse previous state ‘01’ new info follows ‘10’ perform no delta alloc ‘11’ reserved 5.4.3.49 deltbae[ch]: Delta Bit Allocation Exists, 2 bits This per full bandwidth channel 2-bit code indicates the delta bit allocation strategy for the corresponding channel, as shown in Table 5.16. This parameter shall not be set to ‘00’ in block 0. 5.4.3.50 cpldeltnseg: Coupling Delta Bit Allocation Number of Segments, 3 bits This 3-bit code indicates the number of delta bit allocation segments that exist for the coupling channel. The value of this parameter ranges from 1 to 8, and is calculated by adding 1 to the 3-bit binary number represented by the code. 5.4.3.51 cpldeltoffst[seg]: Coupling Delta Bit Allocation Offset, 5 bits The first 5-bit code ([seg] = 0) indicates the number of the first bit allocation band (as specified in 7.4.2) of the coupling channel for which delta bit allocation values are provided. Subsequent codes indicate the offset from the previous delta segment end point to the next bit allocation band for which delta bit allocation values are provided. 5.4.3.52 cpldeltlen[seg]: Coupling Delta Bit Allocation Length, 4 bits Each 4-bit code indicates the number of bit allocation bands that the corresponding segment spans. 5.4.3.53 cpldeltba[seg]: Coupling Delta Bit Allocation, 3 bits This 3-bit value is used in the bit allocation process for the coupling channel. Each 3-bit code indicates an adjustment to the default masking curve computed in the decoder. The deltas are coded as shown in Table 5.17. Table 5.17 Bit Allocation Deltas cpldeltba, deltba Adjustment ‘000’ –24 dB ‘001’ –18 dB ‘010’ –12 dB ‘011’ –6 dB ‘100’ +6 dB ‘101’ +12 dB ‘110’ +18 dB ‘111’ +24 dB 50 Digital Audio Compression Standard 22 November 2010 5.4.3.54 deltnseg[ch]: Channel Delta Bit Allocation Number of Segments, 3 bits These per full bandwidth channel elements are 3-bit codes indicating the number of delta bit allocation segments that exist for the corresponding channel. The value of this parameter ranges from 1 to 8, and is calculated by adding 1 to the 3-bit binary code. 5.4.3.55 deltoffst[ch][seg]: Channel Delta Bit Allocation Offset, 5 bits The first 5-bit code ([seg] = 0) indicates the number of the first bit allocation band (see Section 7.2.2.6) of the corresponding channel for which delta bit allocation values are provided. Subsequent codes indicate the offset from the previous delta segment end point to the next bit allocation band for which delta bit allocation values are provided. 5.4.3.56 deltlen[ch][seg]: Channel Delta Bit Allocation Length, 4 bits Each 4-bit code indicates the number of bit allocation bands that the corresponding segment spans. 5.4.3.57 deltba[ch][seg]: Channel Delta Bit Allocation, 3 bits This 3-bit value is used in the bit allocation process for the indicated channel. Each 3-bit code indicates an adjustment to the default masking curve computed in the decoder. The deltas are coded as shown in Table 5.17. 5.4.3.58 skiple: Skip Length Exists, 1 bit If this bit is a ‘1’, then the skipl parameter follows in the bit stream. If this bit is a ‘0’, skipl does not exist. 5.4.3.59 skipl: Skip Length, 9 bits This 9-bit code indicates the number of dummy bytes to skip (ignore) before unpacking the mantissas of the current audio block. 5.4.3.60 skipfld: Skip Field, (skipl * 8) bits This field contains the null bytes of data to be skipped, as indicated by the skipl parameter. 5.4.3.61 chmant[ch][bin]: Channel Mantissas, 0 to 16 bits The actual quantized mantissa values for the indicated channel. Each value may contain from 0 to as many as 16 bits. The number of mantissas for the indicated channel is equal to nchmant[ch], which may be determined from chbwcod[ch] (see Section 7.1.3) if the channel is not coupled, or from cplbegf (see Section 7.4.2) if the channel is coupled. Detailed information on packed mantissa data is in Section 7.3. 5.4.3.62 cplmant[bin]: Coupling Mantissas, 0 to 16 bits The actual quantized mantissa values for the coupling channel. Each value may contain from 0 to as many as 16 bits. The number of mantissas for the coupling channel is equal to ncplmant, which may be determined from ncplmant = 12 * ncplsubnd 51 Advanced Television Systems Committee, Inc. Document A/52:2010 5.4.3.63 lfemant[bin]: Low Frequency Effects Channel Mantissas, 0 to 16 bits The actual quantized mantissa values for the lfe channel. Each value may contain from 0 to as many as 16 bits. The value of nlfemant is 7, so there are 7 mantissa values for the lfe channel. 5.4.4 auxdata: Auxiliary Data Field Unused data at the end of a frame will exist whenever the encoder does not utilize all available data for encoding the audio signal. This may occur if the final bit allocation falls short of using all available bits, or if the input audio signal simply does not require all available bits to be coded transparently. Or, the encoder may be instructed to intentionally leave some bits unused by audio so that they are available for use by auxiliary data. Since the number of bits required for auxiliary data may be smaller than the number of bits available (which will be time varying) in any particular frame, a method is provided to signal the number of actual auxiliary data bits in each frame. 5.4.4.1 auxbits: Auxiliary Data Bits, nauxbits bits This field contains auxiliary data. The total number of bits in this field is nauxbits = (bits in frame) – (bits used by all bit stream elements except for auxbits) The number of bits in the frame can be determined from the frame size code (frmsizcod) and Table 5.18. The number of bits used includes all bits used by bit stream elements with the exception of auxbits. Any dummy data which has been included with skip fields (skipfld) is included in the used bit count. The length of the auxbits field is adjusted by the encoder such that the crc2 element falls on the last 16-bit word of the frame. If the number of user bits indicated by auxdatal is smaller than the number of available aux bits nauxbits, the user data is located at the end of the auxbits field. This allows a decoder to find and unpack the auxdatal user bits without knowing the value of nauxbits (which can only be determined by decoding the audio in the entire frame). The order of the user data in the auxbits field is forward. Thus the aux data decoder (which may not decode any audio) may simply look to the end of the AC-3 syncframe to find auxdatal, backup auxdatal bits (from the beginning of auxdatal) in the data stream, and then unpack auxdatal bits moving forward in the data stream. 52 Digital Audio Compression Standard 22 November 2010 Table 5.18 Frame Size Code Table (1 word = 16 bits) frmsizecod Nominal Bit Rate fs = 32 kHz fs = 44.1 kHz fs = 48 kHz words/syncframe words/syncframe words/syncframe ‘000000’ (0) 32 kbps 96 69 64 ‘000001’ (0) 32 kbps 96 70 64 ‘000010’ (1) 40 kbps 120 87 80 ‘000011’ (1) 40 kbps 120 88 80 ‘000100’ (2) 48 kbps 144 104 96 ‘000101’ (2) 48 kbps 144 105 96 ‘000110’ (3) 56 kbps 168 121 112 ‘000111’ (3) 56 kbps 168 122 112 ‘001000’ (4) 64 kbps 192 139 128 ‘001001’ (4) 64 kbps 192 140 128 ‘001010’ (5) 80 kbps 240 174 160 ‘001011’ (5) 80 kbps 240 175 160 ‘001100’ (6) 96 kbps 288 208 192 ‘001101’ (6) 96 kbps 288 209 192 ‘001110’ (7) 112 kbps 336 243 224 ‘001111’ (7) 112 kbps 336 244 224 ‘010000’ (8) 128 kbps 384 278 256 ‘010001’ (8) 128 kbps 384 279 256 ‘010010’ (9) 160 kbps 480 348 320 ‘010011’ (9) 160 kbps 480 349 320 ‘010100’ (10) 192 kbps 576 417 384 ‘010101’ (10) 192 kbps 576 418 384 ‘010110’ (11) 224 kbps 672 487 448 ‘010111’ (11) 224 kbps 672 488 448 ‘011000’ (12) 256 kbps 768 557 512 ‘011001’ (12) 256 kbps 768 558 512 ‘011010’ (13) 320 kbps 960 696 640 ‘011011’ (13) 320 kbps 960 697 640 ‘011100’ (14) 384 kbps 1152 835 768 ‘011101’ (14) 384 kbps 1152 836 768 ‘011110’ (15) 448 kbps 1344 975 896 ‘011111’ (15) 448 kbps 1344 976 896 ‘100000’ (16) 512 kbps 1536 1114 1024 ‘100001’ (16) 512 kbps 1536 1115 1024 ‘100010’ (17) 576 kbps 1728 1253 1152 ‘100011’ (17) 576 kbps 1728 1254 1152 ‘100100’ (18) 640 kbps 1920 1393 1280 ‘100101’ (18) 640 kbps 1920 1394 1280 5.4.4.2 auxdatal: Auxiliary Data Length, 14 bits This 14-bit integer value indicates the length, in bits, of the user data in the auxbits auxiliary field. 53 Advanced Television Systems Committee, Inc. Document A/52:2010 5.4.4.3 auxdatae: Auxiliary Data Exists, 1 bit If this bit is a ‘1’, then the auxdatal parameter precedes in the bit stream. If this bit is a ‘0’, auxdatal does not exist, and there is no user data. 5.4.5 errorcheck:Frame Error Detection Field 5.4.5.1 crcrsv: CRC Reserved Bit, 1 bit Reserved for use in specific applications to ensure crc2 will not be equal to syncword. Use of this bit is optional by encoders. If the crc2 calculation results in a value equal to syncword, the crcrsv bit may be inverted. This will result in a crc2 value which is not equal to syncword. 5.4.5.2 crc2: Cyclic Redundancy Check 2, 16 bits The 16 bit CRC applies to the entire frame. The details of the CRC checking are described in Section 7.10.1. 5.5 Bit Stream Constraints The following constraints shall be imposed upon the encoded bit stream by the AC-3 encoder. These constraints allow AC-3 decoders to be manufactured with smaller input memory buffers. 1) The combined size of the syncinfo fields, the bsi fields, block 0 and block 1 combined, shall not exceed 5/8 of the frame. 2) The combined size of the block 5 mantissa data, the auxiliary data fields, and the errorcheck fields shall not exceed the final 3/8 of the frame. 3) Block 0 shall contain all necessary information to begin correctly decoding the bit stream. 4) Whenever the state of cplinu changes from off to on, all coupling information shall be included in the block in which coupling is turned on. No coupling related information shall be reused from any previous blocks where coupling may have been on. 5) Coupling shall not be used in dual mono (1+1) or mono (1/0) modes. For blocks in which coupling is used, there shall be at least two channels in coupling. 6) Bit stream elements shall not be reused from a previous block if other bit stream parameters change the dimensions of the elements to be reused. For example, exponents shall not be reused if the start or end mantissa bin changes from the previous block. 6. DECODING THE AC-3 BIT STREAM Section 5 of this standard specifies the details of the AC-3 bit stream syntax. This section gives an overview of the AC-3 decoding process as diagrammed in Figure 6.1, where the decoding process flow is shown as a sequence of blocks down the center of the page, and some of the information flow is indicated by arrowed lines at the sides of the page. More detailed information on some of the processing blocks will be found in Section 7. The decoder described in this section should be considered one example of a decoder. Other methods may exist to implement decoders, and these other methods may have advantages in certain areas (such as instruction count, memory requirement, number of transforms required, etc.). 54 Digital Audio Compression Standard 22 November 2010 Figure 6.1 Flow diagram of the decoding process. 6.1 Summary of the Decoding Process 6.1.1 Input Bit Stream The input bit stream will typically come from a transmission or storage system. The interface between the source of AC-3 data and the AC-3 decoder is not specified in this standard. The details of the interface effect a number of decoder implementation details. 6.1.1.1 Continuous or Burst Input The encoded AC-3 data may be input to the decoder as a continuous data stream at the nominal bit-rate, or chunks of data may be burst into the decoder at a high rate with a low duty cycle. For 55 Advanced Television Systems Committee, Inc. Document A/52:2010 burst mode operation, either the data source or the decoder may be the master controlling the burst timing. The AC-3 decoder input buffer may be smaller in size if the decoder can request bursts of data on an as-needed basis. However, the external buffer memory may be larger in this case. 6.1.1.2 Byte or Word Alignment Most applications of this standard will convey the elementary AC-3 bit stream with byte or (16- bit) word alignment. The syncronization frame is always an integral number of words in length. The decoder may receive data as a continuous serial stream of bits without any alignment. Or, the data may be input to the decoder with either byte or word (16-bit) alignment. Byte or word alignment of the input data may allow some simplification of the decoder. Alignment does reduce the probability of false detection of the sync word. 6.1.2 Synchronization and Error Detection The AC-3 bit-steam format allows rapid synchronization. The 16-bit sync word has a low probability of false detection. With no input stream alignment the probability of false detection of the sync word is 0.0015 percent per input stream bit position. For a bit-rate of 384 kbps, the probability of false sync word detection is 19 percent per frame. Byte-alignment of the input stream drops this probability to 2.5 percent, and word alignment drops it to 1.2 percent. When a sync pattern is detected the decoder may be estimated to be in sync and one of the CRC words (crc1 or crc2) may be checked. Since crc1 comes first and covers the first 5/8 of the frame, the result of a crc1 check may be available after only 5/8 of the frame has been received. Or, the entire frame size can be received and crc2 checked. If either CRC checks, the decoder may safely be presumed to be in sync and decoding and reproduction of audio may proceed. The chance of false sync in this case would be the concatenation of the probabilities of a false sync word detection and a CRC misdetection of error. The CRC check is reliable to 0.0015 percent. This probability, concatenated with the probability of a false sync detection in a byte-aligned input bit stream, yield a probability of false synchronization of 0.000035 percent (or about once in 3 million synchronization attempts). If this small probability of false sync is too large for an application, there are several methods which may reduce it. The decoder may only presume correct sync in the case that both CRC words check properly. The decoder may require multiple sync words to be received with the proper alignment. If the data transmission or storage system is aware that data is in error, this information may be made known to the decoder. Additional details on methods of bit stream synchronization are not provided in this standard. Details on the CRC calculation are provided in Section 7.10. 6.1.3 Unpack BSI, Side Information Inherent to the decoding process is the unpacking (de-multiplexing) of the various types of information included in the bit stream. Some of these items may be copied from the input buffer to dedicated registers, some may be copied to specific working memory location, and some of the items may simply be located in the input buffer with pointers to them saved to another location for use when the information is required. The information which must be unpacked is specified in detail in Section 5.3. Further details on the unpacking of bsi and side information are not provided in this standard. 56 Digital Audio Compression Standard 22 November 2010 6.1.4 Decode Exponents The exponents are delivered in the bit stream in an encoded form. In order to unpack and decode the exponents two types of side information are required. First, the number of exponents must be known. For fbw channels this may be determined from either chbwcod[ch] (for uncoupled channels) or from cplbegf (for coupled channels). For the coupling channel, the number of exponents may be determined from cplbegf and cplendf. For the lfe channel (when on), there are always 7 exponents. Second, the exponent strategy in use (D15, etc.) by each channel must be known. The details on how to unpack and decode exponents are provided in Section 7.1. 6.1.5 Bit Allocation The bit allocation computation reveals how many bits are used for each mantissa. The inputs to the bit allocation computation are the decoded exponents, and the bit allocation side information. The outputs of the bit allocation computation are a set of bit allocation pointers (baps), one bap for each coded mantissa. The bap indicates the quantizer used for the mantissa, and how many bits in the bit stream were used for each mantissa. The bit allocation computation is described in detail in Section 7.2. 6.1.6 Process Mantissas The coarsely quantized mantissas make up the bulk of the AC-3 data stream. Each mantissa is quantized to a level of precision indicated by the corresponding bap. In order to pack the mantissa data more efficiently, some mantissas are grouped together into a single transmitted value. For instance, two 11-level quantized values are conveyed in a single 7-bit code (3.5 bits/value) in the bit stream. The mantissa data is unpacked by peeling off groups of bits as indicated by the baps. Grouped mantissas must be ungrouped. The individual coded mantissa values are converted into a de- quantized value. Mantissas which are indicated as having zero bits may be reproduced as either zero, or by a random dither value (under control of the dither flag). The mantissa processing is described in full detail in Section 7.3. 6.1.7 Decoupling When coupling is in use, the channels which are coupled must be decoupled. Decoupling involves reconstructing the high frequency section (exponents and mantissas) of each coupled channel, from the common coupling channel and the coupling coordinates for the individual channel. Within each coupling band, the coupling channel coefficients (exponent and mantissa) are multiplied by the individual channel coupling coordinates. The coupling process is described in detail in Section 7.4. 6.1.8 Rematrixing In the 2/0 audio coding mode rematrixing may be employed, as indicated by the rematrix flags (rematflg[rbnd]). Where the flag indicates a band is rematrixed, the coefficients encoded in the bit stream are sum and difference values instead of left and right values. Rematrixing is described in detail in Section 7.5. 57 Advanced Television Systems Committee, Inc. Document A/52:2010 6.1.9 Dynamic Range Compression For each block of audio a dynamic range control value (dynrng) may be included in the bit stream. The decoder, by default, shall use this value to alter the magnitude of the coefficient (exponent and mantissa) as specified in Section 7.7.1. 6.1.10 Inverse Transform The decoding steps described above will result in a set of frequency coefficients for each encoded channel. The inverse transform converts the blocks of frequency coefficients into blocks of time samples. The inverse transform is detailed in Section 7.9. 6.1.11 Window, Overlap/Add The individual blocks of time samples must be windowed, and adjacent blocks must be overlapped and added together in order to reconstruct the final continuous time output PCM audio signal. The window and overlap/add steps are described along with the inverse transform in Section 7.9. 6.1.12 Downmixing If the number of channels required at the decoder output is smaller than the number of channels which are encoded in the bit stream, then downmixing is required. Downmixing in the time domain is shown in this example decoder. Since the inverse transform is a linear operation, it is also possible to downmix in the frequency domain prior to transformation. Section 7.8 describes downmixing and specifies the downmix coefficients which decoders shall employ. 6.1.13 PCM Output Buffer Typical decoders will provide PCM output samples at the PCM sampling rate. Since blocks of samples result from the decoding process, an output buffer is typically required. This standard does not specify or describe output buffering in any further detail. 6.1.14 Output PCM The output PCM samples may be delivered in form suitable for interconnection to a digital to analog converter (DAC), or in any other form. This Standard does not specify the output PCM format. 7. ALGORITHMIC DETAILS The following sections describe various aspects of AC-3 coding in detail. 7.1 Exponent coding 7.1.1 Overview The actual audio information conveyed by the AC-3 bit stream consists of the quantized frequency coefficients. The coefficients are delivered in floating point form, with each coefficient consisting of an exponent and a mantissa. This section describes how the exponents are encoded and packed into the bit stream. Exponents are 5-bit values which indicate the number of leading zeros in the binary representation of a frequency coefficient. The exponent acts as a scale factor for each mantissa, equal to 2-exp. Exponent values are allowed to range from 0 (for the largest value coefficients with 58 Digital Audio Compression Standard 22 November 2010 no leading zeroes) to 24. Exponents for coefficients which have more than 24 leading zeroes are fixed at 24, and the corresponding mantissas are allowed to have leading zeros. Exponents require 5 bits in order to represent all allowed values. AC-3 bit streams contain coded exponents for all independent channels, all coupled channels, and for the coupling and low frequency effects channels (when they are enabled). Since audio information is not shared across frames, block 0 of every frame will include new exponents for every channel. Exponent information may be shared across blocks within a frame, so blocks 1 through 5 may reuse exponents from previous blocks. AC-3 exponent transmission employs differential coding, in which the exponents for a channel are differentially coded across frequency. The first exponent of a fbw or lfe channel is always sent as a 4-bit absolute value, ranging from 0–15. The value indicates the number of leading zeros of the first (dc term) transform coefficient. Successive (going higher in frequency) exponents are sent as differential values which must be added to the prior exponent value in order to form the next absolute value. The differential exponents are combined into groups in the audio block. The grouping is done by one of three methods, D15, D25, or D45, which are referred to as exponent strategies. The number of grouped differential exponents placed in the audio block for a particular channel depends on the exponent strategy and on the frequency bandwidth information for that channel. The number of exponents in each group depends only on the exponent strategy. An AC-3 audio block contains two types of fields with exponent information. The first type defines the exponent coding strategy for each channel, and the second type contains the actual coded exponents for channels requiring new exponents. For independent channels, frequency bandwidth information is included along with the exponent strategy fields. For coupled channels, and the coupling channel, the frequency information is found in the coupling strategy fields. 7.1.2 Exponent Strategy Exponent strategy information for every channel is included in every AC-3 audio block. Information is never shared across frames, so block 0 will always contain a strategy indication (D15, D25, or D45) for each channel. Blocks 1 through 5 may indicate reuse of the prior (within the same frame) exponents. The three exponent coding strategies provide a tradeoff between data rate required for exponents, and their frequency resolution. The D15 mode provides the finest frequency resolution, and the D45 mode requires the least amount of data. In all three modes, a number differential exponents are combined into 7-bit words when coded into an audio block. The main difference between the modes is how many differential exponents are combined together. The absolute exponents found in the bit stream at the beginning of the differentially coded exponent sets are sent as 4-bit values which have been limited in either range or resolution in order to save one bit. For fbw and lfe channels, the initial 4-bit absolute exponent represents a value from 0 to 15. Exponent values larger than 15 are limited to a value of 15. For the coupled channel, the 5-bit absolute exponent is limited to even values, and the lsb is not transmitted. The resolution has been limited to valid values of 0, 2, 4...24. Each differential exponent can take on one of five values: –2, –1, 0, +1, +2. This allows deltas of up to ±2 (±12 dB) between exponents. 59 Advanced Television Systems Committee, Inc. Document A/52:2010 These five values are mapped into the values 0, 1, 2, 3, 4 before being grouped, as shown in Table 7.1. Table 7.1 Mapping of Differential Exponent Values, D15 Mode diff exp Mapped Value +2 4 +1 3 0 2 –1 1 –2 0 mapped value = diff exp + 2 ; diff exp = mapped value – 2 ; In the D15 mode, the above mapping is applied to each individual differential exponent for coding into the bit stream. In the D25 mode, each pair of differential exponents is represented by a single mapped value in the bit stream. In this mode the second differential exponent of each pair is implied as a delta of 0 from the first element of the pair as indicated in Table 7.2. Table 7.2 Mapping of Differential Exponent Values, D25 Mode diff exp n diff exp n+1 Mapped Value +2 0 4 +1 0 3 0 0 2 –1 0 1 –2 0 0 The D45 mode is similar to the D25 mode except that quads of differential exponents are represented by a single mapped value, as indicated by Table 7.3. Table 7.3 Mapping of Differential Exponent Values, D45 Mode diff exp n diff exp n+1 diff exp n+2 diff exp n+3 Mapped Value +2 0 0 0 4 +1 0 0 0 3 0 0 0 0 2 –1 0 0 0 1 –2 0 0 0 0 Since a single exponent is effectively shared by 2 or 4 different mantissas, encoders must ensure that the exponent chosen for the pair or quad is the minimum absolute value (corresponding to the largest exponent) needed to represent all the mantissas. For all modes, sets of three adjacent (in frequency) mapped values (M1, M2, and M3) are grouped together and coded as a 7 bit value according to the following formula coded 7 bit grouped value = (25 * M1) + (5 * M2) + M3 60 Digital Audio Compression Standard 22 November 2010 The exponent field for a given channel in an AC-3 audio block consists of a single absolute exponent followed by a number of these grouped values. 7.1.3 Exponent Decoding The exponent strategy for each coupled and independent channel is included in a set of 2-bit fields designated chexpstr[ch]. When the coupling channel is present, a cplexpstr strategy code is also included. Table 7.4 shows the mapping from exponent strategy code into exponent strategy. Table 7.4 Exponent Strategy Coding chexpstr[ch], cplexpstr Exponent Strategy Exponents per Group ‘00’ reuse prior exponents 0 ‘01’ D15 3 ‘10’ D25 6 ‘11’ D45 12 When the low frequency effects channel is enabled the lfeexpstr field is present. It is decoded as shown in Table 7.5. Table 7.5 LFE Channel Exponent Strategy Coding lfeexpstr Exponent Strategy Exponents per Group ‘0’ reuse prior exponents 0 ‘1’ D15 3 Following the exponent strategy fields in the bit stream is a set of channel bandwidth codes, chbwcod[ch]. These are only present for independent channels (channels not in coupling) that have new exponents in the current block. The channel bandwidth code defines the end mantissa bin number for that channel according to the following endmant[ch] = ((chbwcod[ch] + 12) * 3) + 37; /* (ch is not coupled) */ For coupled channels the end mantissa bin number is defined by the starting bin number of the coupling channel endmant[ch] = cplstrtmant; /* (ch is coupled) */ where cplstrtmant is as derived below. By definition the starting mantissa bin number for independent and coupled channels is 0 strtmant[ch] = 0 For the coupling channel, the frequency bandwidth information is derived from the fields cplbegf and cplendf found in the coupling strategy information. The coupling channel starting and ending mantissa bins are defined as cplstrtmant = (cplbegf * 12) + 37 cplendmant = ((cplendf + 3) * 12) + 37 61 Advanced Television Systems Committee, Inc. Document A/52:2010 The low frequency effects channel, when present, always starts in bin 0 and always has the same number of mantissas lfestrtmant = 0 lfeendmant = 7 The second set of fields contains coded exponents for all channels indicated to have new exponents in the current block. These fields are designated as exps[ch][grp] for independent and coupled channels, cplexps[grp] for the coupling channel, and lfeexps[grp] for the low frequency effects channel. The first element of the exps fields (exps[ch][0]) and the lfeexps field (lfeexps[0]) is always a 4- bit absolute number. For these channels the absolute exponent always contains the exponent value of the first transform coefficient (bin #0). These 4 bit values correspond to a 5-bit exponent which has been limited in range (0 to 15, instead of 0 to 24); i.e., the most significant bit is zero. The absolute exponent for the coupled channel, cplabsexp, is only used as a reference to begin decoding the differential exponents for the coupling channel (i.e., it does not represent an actual exponent). The cplabsexp is contained in the audio block as a 4-bit value, however it corresponds to a 5-bit value. The LSB of the coupled channel initial exponent is always 0, so the decoder must take the 4-bit value which was sent, and double it (left shift by 1) in order to obtain the 5-bit starting value. For each coded exponent set the number of grouped exponents (not including the first absolute exponent) to decode from the bit stream is derived as follows: For independent and coupled channels: nchgrps[ch] = truncate {(endmant[ch] – 1) / 3} ; /* for D15 mode */ = truncate {(endmant[ch] – 1 + 3) / 6} ; /* for D25 mode */ = truncate {(endmant[ch] - 1 + 9) / 12} ; /* for D45 mode */ For the coupling channel: ncplgrps = (cplendmant – cplstrtmant) / 3 ; /* for D15 mode */ = (cplendmant – cplstrtmant) / 6 ; /* for D25 mode */ = (cplendmant – cplstrtmant) / 12 ; /* for D45 mode */ For the low frequency effects channel: nlfegrps =2 Decoding a set of coded grouped exponents will create a set of 5-bit absolute exponents. The exponents are decoded as follows: 1. Each 7 bit grouping of mapped values (gexp) is decoded using the inverse of the encoding procedure: M1 = truncate (gexp / 25) M2 = truncate {(gexp % 25} / 5) M3 = (gexp % 25) % 5 62 Digital Audio Compression Standard 22 November 2010 2. Each mapped value is converted to a differential exponent (dexp) by subtracting the mapping offset: dexp = M 2 3. The set of differential exponents if converted to absolute exponents by adding each differential exponent to the absolute exponent of the previous frequency bin: exp[n] = exp[n-1] + dexp[n] 4. For the D25 and D45 modes, each absolute exponent is copied to the remaining members of the pair or quad. The above procedure can be summarized as follows: 63 Advanced Television Systems Committee, Inc. Document A/52:2010 Pseudo Code /* unpack the mapped values */ for (grp = 0; grp < ngrps; grp++) { expacc = gexp[grp] ; dexp[grp * 3] = truncate (expacc / 25) ; expacc = expacc - ( 25 * dexp[grp * 3]) ; dexp[(grp * 3) + 1] = truncate ( expacc / 5) ; expacc = expacc - (5 * dexp[(grp * 3) + 1]) ; dexp[(grp * 3) + 2] = expacc ; } /* unbiased mapped values */ for (grp = 0; grp < (ngrps * 3); grp++) { dexp[grp] = dexp[grp] - 2 ; } /* convert from differentials to absolutes */ prevexp = absexp ; for (i = 0; i < (ngrps * 3); i++) { aexp[i] = prevexp + dexp[i] ; prevexp = aexp[i] ; } /* expand to full absolute exponent array, using grpsize */ exp[0] = absexp ; for (i = 0; i < (ngrps * 3); i++) { for (j = 0; j < grpsize; j++) { exp[(i * grpsize) + j +1] = aexp[i] ; } } Where,: ngrps = number of grouped exponents (nchgrps[ch], ncplgrps, or nlfegrps) grpsize = 1 for D15 = 2 for D25 = 4 for D45 absexp = absolute exponent (exps[ch][0], (cplabsexp<<1), or lfeexps[0]) For the coupling channel the above output array, exp[n], should be offset to correspond to the coupling start mantissa bin: cplexp[n + cplstrtmant] = exp[n + 1] ; For the remaining channels exp[n] will correspond directly to the absolute exponent array for that channel. 64 Digital Audio Compression Standard 22 November 2010 7.2 Bit Allocation 7.2.1 Overview The bit allocation routine analyzes the spectral envelope of the audio signal being coded with respect to masking effects to determine the number of bits to assign to each transform coefficient mantissa. In the encoder, the bit allocation is performed globally on the ensemble of channels as an entity, from a common bit pool. There are no preassigned exponent or mantissa bits, allowing the routine to flexibly allocate bits across channels, frequencies, and audio blocks in accordance with signal demand. The bit allocation contains a parametric model of human hearing for estimating a noise level threshold, expressed as a function of frequency, which separates audible from inaudible spectral components. Various parameters of the hearing model can be adjusted by the encoder depending upon signal characteristics. For example, a prototype masking curve is defined in terms of two piecewise continuous line segments, each with its own slope and y-axis intercept. One of several possible slopes and intercepts is selected by the encoder for each line segment. The encoder may iterate on one or more such parameters until an optimal result is obtained. When all parameters used to estimate the noise level threshold have been selected by the encoder, the final bit allocation is computed. The model parameters are conveyed to the decoder with other side information. The decoder executes the routine in a single pass. The estimated noise level threshold is computed over 50 bands of nonuniform bandwidth (an approximate 1/6 octave scale). The banding structure, defined by tables in the next section, is independent of sampling frequency. The required bit allocation for each mantissa is established by performing a table lookup based upon the difference between the input signal power spectral density (PSD) evaluated on a fine-grain uniform frequency scale, and the estimated noise level threshold evaluated on the coarse-grain (banded) frequency scale. Therefore, the bit allocation result for a particular channel has spectral granularity corresponding to the exponent strategy employed. More specifically, a separate bit allocation will be computed for each mantissa within a D15 exponent set, each pair of mantissas within a D25 exponent set, and each quadruple of mantissas within a D45 exponent set. The bit allocation must be computed in the decoder whenever the exponent strategy (chexpstr, cplexpstr, lfeexpstr) for one or more channels does not indicate reuse, or whenever baie, snroffste, or deltbaie = 1. Accordingly, the bit allocation can be updated at a rate ranging from once per audio block to once per 6 audio blocks, including the integral steps in between. A complete set of new bit allocation information is always transmitted in audio block 0. Since the parametric bit allocation routine must generate identical results in all encoder and decoder implementations, each step is defined exactly in terms of fixed-point integer operations and table lookups. Throughout the discussion below, signed two's complement arithmetic is employed. All additions are performed with an accumulator of 14 or more bits. All intermediate results and stored values are 8-bit values. 7.2.2 Parametric Bit Allocation This section describes the seven-step procedure for computing the output of the parametric bit allocation routine in the decoder. The approach outlined here starts with a single uncoupled or coupled exponent set and processes all the input data for each step prior to continuing to the next one. This technique, called vertical execution, is conceptually straightforward to describe and 65 Advanced Television Systems Committee, Inc. Document A/52:2010 implement. Alternatively, the seven steps can be executed horizontally, in which case multiple passes through all seven steps are made for separate subsets of the input exponent set. The choice of vertical vs. horizontal execution depends upon the relative importance of execution time vs. memory usage in the final implementation. Vertical execution of the algorithm is usually faster due to reduced looping and context save overhead. However, horizontal execution requires less RAM to store the temporary arrays generated in each step. Hybrid horizontal/vertical implementation approaches are also possible which combine the benefits of both techniques. 7.2.2.1 Initialization Compute start/end frequencies for the channel being decoded. These are computed from parameters in the bit stream as follows: Pseudo Code /* for fbw channels */ for (ch=0; ch lastbin) ; logadd(a, b) { c=a−b; address = min((abs(c) >> 1), 255) ; if (c >= 0) { return(a + latab(address)) ; } else { return(b + latab(address)) ; } } 7.2.2.4 Compute Excitation Function The excitation function is computed by applying the prototype masking curve selected by the encoder (and transmitted to the decoder) to the integrated PSD spectrum (bndpsd[]). The result of this computation is then offset downward in amplitude by the fgain and sgain parameters, which are also obtained from the bit stream. Pseudo Code bndstrt = masktab[start] ; bndend = masktab[end - 1] + 1 ; 68 Digital Audio Compression Standard 22 November 2010 if (bndstrt == 0) /* For fbw and lfe channels */ { /* Note: Do not call calc_lowcomp() for the last band of the lfe channel, (bin = 6) */ lowcomp = calc_lowcomp(lowcomp, bndpsd[0], bndpsd[1], 0) ; excite[0] = bndpsd[0] - fgain – lowcomp ; lowcomp = calc_lowcomp(lowcomp, bndpsd[1], bndpsd[2], 1) ; excite[1] = bndpsd[1] - fgain – lowcomp ; begin = 7 ; for (bin = 2; bin < 7; bin++) { if ((bndend != 7) || (bin != 6)) /* skip for last bin of lfe channels */ { lowcomp = calc_lowcomp(lowcomp, bndpsd[bin], bndpsd[bin+1], bin) ; } fastleak = bndpsd[bin] – fgain ; slowleak = bndpsd[bin] – sgain ; excite[bin] = fastleak – lowcomp ; if ((bndend != 7) || (bin != 6)) /* skip for last bin of lfe channel */ { if (bndpsd[bin] <= bndpsd[bin+1]) { begin = bin + 1 ; break ; } } } for (bin = begin; bin < min(bndend, 22); bin++) { if ((bndend != 7) || (bin != 6)) /* skip for last bin of lfe channel */ { lowcomp = calc_lowcomp(lowcomp, bndpsd[bin], bndpsd[bin+1], bin) ; } fastleak -= fdecay ; fastleak = max(fastleak, bndpsd[bin] - fgain) ; slowleak -= sdecay ; slowleak = max(slowleak, bndpsd[bin] - sgain) ; excite[bin] = max(fastleak – lowcomp, slowleak) ; } begin = 22 ; } else /* For coupling channel */ { begin = bndstrt ; } for (bin = begin; bin < bndend; bin++) 69 Advanced Television Systems Committee, Inc. Document A/52:2010 { fastleak -= fdecay ; fastleak = max(fastleak, bndpsd[bin] - fgain) ; slowleak -= sdecay ; slowleak = max(slowleak, bndpsd[bin] - sgain) ; excite[bin] = max(fastleak, slowleak) ; } calc_lowcomp(a, b0, b1, bin) { if (bin < 7) { if ((b0 + 256) == b1) ; { a = 384 ; } else if (b0 > b1) { a = max(0, a - 64) ; } } else if (bin < 20) { if ((b0 + 256) == b1) { a = 320 ; } else if (b0 > b1) { a = max(0, a - 64) ; } } else { a = max(0, a - 128) ; } return(a) ; } 7.2.2.5 Compute Masking Curve This step computes the masking (noise level threshold) curve from the excitation function, as shown below. The hearing threshold hth[][] is shown in Table 7.15. The fscod and dbpbcod variables are received by the decoder in the bit stream. 70 Digital Audio Compression Standard 22 November 2010 Pseudo Code for (bin = bndstrt; bin < bndend; bin++) { if (bndpsd[bin] < dbknee) { excite[bin] += ((dbknee - bndpsd[bin]) >> 2) ; } mask[bin] = max(excite[bin], hth[fscod][bin]) ; } 7.2.2.6 Apply Delta Bit Allocation The optional delta bit allocation information in the bit stream provides a means for the encoder to transmit side information to the decoder which directly increases or decreases the masking curve obtained by the parametric routine. Delta bit allocation can be enabled by the encoder for audio blocks which derive an improvement in audio quality when the default bit allocation is appropriately modified. The delta bit allocation option is available for each fbw channel and the coupling channel. In the event that delta bit allocation is not being used, and no dba information is included in the bit stream, the decoder must not modify the default allocation. One way to insure this is to initialize the cpldeltnseg and deltnseg[ch] delta bit allocation variables to 0 at the beginning of each frame. This makes the dba processing (shown below) to immediately terminate, unless dba information (including cpldeltnseg and deltnseg[ch]) is included in the bit stream. The dba information which modifies the decoder bit allocation are transmitted as side information. The allocation modifications occur in the form of adjustments to the default masking curve computed in the decoder. Adjustments can be made in multiples of ±6 dB. On the average, a masking curve adjustment of –6 dB corresponds to an increase of 1 bit of resolution for all the mantissas in the affected 1/6th octave band. The following code indicates, for a single channel, how the modification is performed. The modification calculation is performed on the coupling channel (where deltnseg below equals cpldeltnseg) and on each fbw channel (where deltnseg equals deltnseg[ch]). 71 Advanced Television Systems Committee, Inc. Document A/52:2010 Pseudo Code if ((deltbae == 0) || (deltbae == 1)) { band = 0 ; for (seg = 0; seg < deltnseg+1; seg++) { band += deltoffst[seg] ; if (deltba[seg] >= 4) { delta = (deltba[seg] - 3) << 7 ; } else { delta = (deltba[seg] - 4) << 7 ; } for (k = 0; k < deltlen[seg]; k++) { mask[band] += delta ; band++ ; } } } 7.2.2.7 Compute Bit Allocation The bit allocation pointer array (bap[]) is computed in this step. The masking curve, adjusted by snroffset in an earlier step and then truncated, is subtracted from the fine-grain psd[] array. The difference is right-shifted by 5 bits, thresholded, and then used as an address into baptab[] to obtain the final allocation. The baptab[] array is shown in Table 7.16. The sum of all channel mantissa allocations in one frame is constrained by the encoder to be less than or equal to the total number of mantissa bits available for that frame. The encoder accomplishes this by iterating on the values of csnroffst and fsnroffst (or cplfsnroffst or lfefsnroffst for the coupling and low frequency effects channels) to obtain an appropriate result. The decoder is guaranteed to receive a mantissa allocation which meets the constraints of a fixed transmission bit-rate. At the end of this step, the bap[] array contains a series of 4-bit pointers. The pointers indicate how many bits are assigned to each mantissa. The correspondence between bap pointer value and quantization accuracy is shown in Table 7.17. 72 Digital Audio Compression Standard 22 November 2010 Pseudo Code i = start ; j = masktab[start] ; do { lastbin = min(bndtab[j] + bndsz[j], end) ; mask[j] -= snroffset ; mask[j] -= floor ; if (mask[j] < 0) { mask[j] = 0 ; } mask[j] &= 0x1fe0 ; mask[j] += floor ; for (k = i; k < lastbin; k++) { address = (psd[i] - mask[j]) >> 5 ; address = min(63, max(0, address)) ; bap[i] = baptab[address] ; i++ ; } j++; } while (end > lastbin) ; 7.2.3 Bit Allocation Tables Table 7.6 Slow Decay Table, slowdec[] Address slowdec[address] 0 0x0f 1 0x11 2 0x13 3 0x15 Table 7.7 Fast Decay Table, fastdec[] Address fastdec[address] 0 0x3f 1 0x53 2 0x67 3 0x7b 73 Advanced Television Systems Committee, Inc. Document A/52:2010 Table 7.8 Slow Gain Table, slowgain[] Address slowgain[address] 0 0x540 1 0x4d8 2 0x478 3 0x410 Table 7.9 dB/Bit Table, dbpbtab[] Address dbpbtab[address] 0 0x000 1 0x700 2 0x900 3 0xb00 Table 7.10 Floor Table, floortab[] Address floortab[address] 0 0x2f0 1 0x2b0 2 0x270 3 0x230 4 0x1f0 5 0x170 6 0x0f0 7 0xf800 Table 7.11 Fast Gain Table, fastgain[] Address fastgain[address] 0 0x080 1 0x100 2 0x180 3 0x200 4 0x280 5 0x300 6 0x380 7 0x400 74 Digital Audio Compression Standard 22 November 2010 Table 7.12 Banding Structure Tables, bndtab[], bndsz[] Band # bndtab[band] bndsz[band] Band # bndtab[band] bndsz[band] 0 0 1 25 25 1 1 1 1 26 26 1 2 2 1 27 27 1 3 3 1 28 28 3 4 4 1 29 31 3 5 5 1 30 34 3 6 6 1 31 37 3 7 7 1 32 40 3 8 8 1 33 43 3 9 9 1 34 46 3 10 10 1 35 49 6 11 11 1 36 55 6 12 12 1 37 61 6 13 13 1 38 67 6 14 14 1 39 73 6 15 15 1 40 79 6 16 16 1 41 85 12 17 17 1 42 97 12 18 18 1 43 109 12 19 19 1 44 121 12 20 20 1 45 133 24 21 21 1 46 157 24 22 22 1 47 181 24 23 23 1 48 205 24 24 24 1 49 229 24 75 Advanced Television Systems Committee, Inc. Document A/52:2010 Table 7.13 Bin Number to Band Number Table, masktab[bin], bin = (10 * A) + B B=0 B=1 B=2 B=3 B=4 B=5 B=6 B=7 B=8 B=9 A=0 0 1 2 3 4 5 6 7 8 9 A=1 10 11 12 13 14 15 16 17 18 19 A=2 20 21 22 23 24 25 26 27 28 28 A=3 28 29 29 29 30 30 30 31 31 31 A=4 32 32 32 33 33 33 34 34 34 35 A=5 35 35 35 35 35 36 36 36 36 36 A=6 36 37 37 37 37 37 37 38 38 38 A=7 38 38 38 39 39 39 39 39 39 40 A=8 40 40 40 40 40 41 41 41 41 41 A=9 41 41 41 41 41 41 41 42 42 42 A=10 42 42 42 42 42 42 42 42 42 43 A=11 43 43 43 43 43 43 43 43 43 43 A=12 43 44 44 44 44 44 44 44 44 44 A=13 44 44 44 45 45 45 45 45 45 45 A=14 45 45 45 45 45 45 45 45 45 45 A=15 45 45 45 45 45 45 45 46 46 46 A=16 46 46 46 46 46 46 46 46 46 46 A=17 46 46 46 46 46 46 46 46 46 46 A=18 46 47 47 47 47 47 47 47 47 47 A=19 47 47 47 47 47 47 47 47 47 47 A=20 47 47 47 47 47 48 48 48 48 48 A=21 48 48 48 48 48 48 48 48 48 48 A=22 48 48 48 48 48 48 48 48 48 49 A=23 49 49 49 49 49 49 49 49 49 49 A=24 49 49 49 49 49 49 49 49 49 49 A=25 49 49 49 0 0 0 76 Digital Audio Compression Standard 22 November 2010 Table 7.14 Log-Addition Table, latab[val], val = (10 * A) + B B=0 B=1 B=2 B=3 B=4 B=5 B=6 B=7 B=8 B=9 A=0 0x0040 0x003f 0x003e 0x003d 0x003c 0x003b 0x003a 0x0039 0x0038 0x0037 A=1 0x0036 0x0035 0x0034 0x0034 0x0033 0x0032 0x0031 0x0030 0x002f 0x002f A=2 0x002e 0x002d 0x002c 0x002c 0x002b 0x002a 0x0029 0x0029 0x0028 0x0027 A=3 0x0026 0x0026 0x0025 0x0024 0x0024 0x0023 0x0023 0x0022 0x0021 0x0021 A=4 0x0020 0x0020 0x001f 0x001e 0x001e 0x001d 0x001d 0x001c 0x001c 0x001b A=5 0x001b 0x001a 0x001a 0x0019 0x0019 0x0018 0x0018 0x0017 0x0017 0x0016 A=6 0x0016 0x0015 0x0015 0x0015 0x0014 0x0014 0x0013 0x0013 0x0013 0x0012 A=7 0x0012 0x0012 0x0011 0x0011 0x0011 0x0010 0x0010 0x0010 0x000f 0x000f A=8 0x000f 0x000e 0x000e 0x000e 0x000d 0x000d 0x000d 0x000d 0x000c 0x000c A=9 0x000c 0x000c 0x000b 0x000b 0x000b 0x000b 0x000a 0x000a 0x000a 0x000a A=10 0x000a 0x0009 0x0009 0x0009 0x0009 0x0009 0x0008 0x0008 0x0008 0x0008 A=11 0x0008 0x0008 0x0007 0x0007 0x0007 0x0007 0x0007 0x0007 0x0006 0x0006 A=12 0x0006 0x0006 0x0006 0x0006 0x0006 0x0006 0x0005 0x0005 0x0005 0x0005 A=13 0x0005 0x0005 0x0005 0x0005 0x0004 0x0004 0x0004 0x0004 0x0004 0x0004 A=14 0x0004 0x0004 0x0004 0x0004 0x0004 0x0003 0x0003 0x0003 0x0003 0x0003 A=15 0x0003 0x0003 0x0003 0x0003 0x0003 0x0003 0x0003 0x0003 0x0003 0x0002 A=16 0x0002 0x0002 0x0002 0x0002 0x0002 0x0002 0x0002 0x0002 0x0002 0x0002 A=17 0x0002 0x0002 0x0002 0x0002 0x0002 0x0002 0x0002 0x0002 0x0001 0x0001 A=18 0x0001 0x0001 0x0001 0x0001 0x0001 0x0001 0x0001 0x0001 0x0001 0x0001 A=19 0x0001 0x0001 0x0001 0x0001 0x0001 0x0001 0x0001 0x0001 0x0001 0x0001 A=20 0x0001 0x0001 0x0001 0x0001 0x0001 0x0001 0x0001 0x0001 0x0001 0x0001 A=21 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 A=22 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 A=23 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 A=24 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 A=25 0x0000 0x0000 0x0000 0x0000 0x0000 0x0000 77 Advanced Television Systems Committee, Inc. Document A/52:2010 Table 7.15 Hearing Threshold Table, hth[fscod][band] Band hth[0][band] hth[1][band] hth[2][band] Band hth[0][band] hth[1][band] hth[2][band] No. (fs=48 kHz) (fs=44.1 kHz) (fs=32 kHz) No. (fs=48 kHz) (fs=44.1 kHz) (fs=32 kHz) 0 0x04d0 0x04f0 0x0580 25 0x0340 0x0350 0x0380 1 0x04d0 0x04f0 0x0580 26 0x0330 0x0340 0x0380 2 0x0440 0x0460 0x04b0 27 0x0320 0x0340 0x0370 3 0x0400 0x0410 0x0450 28 0x0310 0x0320 0x0360 4 0x03e0 0x03e0 0x0420 29 0x0300 0x0310 0x0350 5 0x03c0 0x03d0 0x03f0 30 0x02f0 0x0300 0x0340 6 0x03b0 0x03c0 0x03e0 31 0x02f0 0x02f0 0x0330 7 0x03b0 0x03b0 0x03d0 32 0x02f0 0x02f0 0x0320 8 0x03a0 0x03b0 0x03c0 33 0x02f0 0x02f0 0x0310 9 0x03a0 0x03a0 0x03b0 34 0x0300 0x02f0 0x0300 10 0x03a0 0x03a0 0x03b0 35 0x0310 0x0300 0x02f0 11 0x03a0 0x03a0 0x03b0 36 0x0340 0x0320 0x02f0 12 0x03a0 0x03a0 0x03a0 37 0x0390 0x0350 0x02f0 13 0x0390 0x03a0 0x03a0 38 0x03e0 0x0390 0x0300 14 0x0390 0x0390 0x03a0 39 0x0420 0x03e0 0x0310 15 0x0390 0x0390 0x03a0 40 0x0460 0x0420 0x0330 16 0x0380 0x0390 0x03a0 41 0x0490 0x0450 0x0350 17 0x0380 0x0380 0x03a0 42 0x04a0 0x04a0 0x03c0 18 0x0370 0x0380 0x03a0 43 0x0460 0x0490 0x0410 19 0x0370 0x0380 0x03a0 44 0x0440 0x0460 0x0470 20 0x0360 0x0370 0x0390 45 0x0440 0x0440 0x04a0 21 0x0360 0x0370 0x0390 46 0x0520 0x0480 0x0460 22 0x0350 0x0360 0x0390 47 0x0800 0x0630 0x0440 23 0x0350 0x0360 0x0390 48 0x0840 0x0840 0x0450 24 0x0340 0x0350 0x0380 49 0x0840 0x0840 0x04e0 78 Digital Audio Compression Standard 22 November 2010 Table 7.16 Bit Allocation Pointer Table, baptab[] Address baptab[address] Address baptab[address] 0 0 32 10 1 1 33 10 2 1 34 10 3 1 35 11 4 1 36 11 5 1 37 11 6 2 38 11 7 2 39 12 8 3 40 12 9 3 41 12 10 3 42 12 11 4 43 13 12 4 44 13 13 5 45 13 14 5 46 13 15 6 47 14 16 6 48 14 17 6 49 14 18 6 50 14 19 7 51 14 20 7 52 14 21 7 53 14 22 7 54 14 23 8 55 15 24 8 56 15 25 8 57 15 26 8 58 15 27 9 59 15 28 9 60 15 29 9 61 15 30 9 62 15 31 10 63 15 79 Advanced Television Systems Committee, Inc. Document A/52:2010 Table 7.17 Quantizer Levels and Mantissa Bits vs. bap bap Quantizer Mantissa Bits Levels (group bits / num in group) 0 0 0 1 3 1.67 (5/3) 2 5 2.33 (7/3) 3 7 3 4 11 3.5 (7/2) 5 15 4 6 32 5 7 64 6 8 128 7 9 256 8 10 512 9 11 1024 10 12 2048 11 13 4096 12 14 16,384 14 15 65,536 16 7.3 Quantization and Decoding of Mantissas 7.3.1 Overview All mantissas are quantized to a fixed level of precision indicated by the corresponding bap. Mantissas quantized to 15 or fewer levels use symmetric quantization. Mantissas quantized to more than 15 levels use asymmetric quantization which is a conventional two’s complement representation. Some quantized mantissa values are grouped together and encoded into a common codeword. In the case of the 3-level quantizer, 3 quantized values are grouped together and represented by a 5-bit codeword in the data stream. In the case of the 5-level quantizer, 3 quantized values are grouped and represented by a 7-bit codeword. For the 11-level quantizer, 2 quantized values are grouped and represented by a 7-bit codeword. In the encoder, each transform coefficient (which is always < 1.0) is left-justified by shifting its binary representation left the number of times indicated by its exponent (0 to 24 left shifts). The amplified coefficient is then quantized to a number of levels indicated by the corresponding bap. The following table indicates which quantizer to use for each bap. If a bap equals 0, no bits are sent for the mantissa. Grouping is used for baps of 1, 2, and 4 (3, 5, and 11 level quantizers.) 80 Digital Audio Compression Standard 22 November 2010 Table 7.18 Mapping of bap to Quantizer bap Quantizer Levels Quantization Type Mantissa Bits (qntztab[bap]) (group bits / num in group) 0 0 none 0 1 3 symmetric 1.67 (5/3) 2 5 symmetric 2.33 (7/3) 3 7 symmetric 3 4 11 symmetric 3.5 (7/2) 5 15 symmetric 4 6 32 asymmetric 5 7 64 asymmetric 6 8 128 asymmetric 7 9 256 asymmetric 8 10 512 asymmetric 9 11 1024 asymmetric 10 12 2048 asymmetric 11 13 4096 asymmetric 12 14 16,384 asymmetric 14 15 65,536 asymmetric 16 During the decode process, the mantissa data stream is parsed up into single mantissas of varying length, interspersed with groups representing combined coding of either triplets or pairs of mantissas. In the bit stream, the mantissas in each exponent set are arranged in frequency ascending order. However, groups occur at the position of the first mantissa contained in the group. Nothing is unpacked from the bit stream for the subsequent mantissas in the group. 7.3.2 Expansion of Mantissas for Asymmetric Quantization (6 ≤ bap ≤ 15) For bit allocation pointer array values, 6 ≤ bap ≤ 15, asymmetric fractional two’s complement quantization is used. Each mantissa, along with its exponent, are the floating point representation of a transform coefficient. The decimal point is considered to be to the left of the MSB; therefore the mantissa word represents the range of (1.0 – 2–(qntztab[bap] – 1)) to –1.0 The mantissa number k, of length qntztab[bap[k]], is extracted from the bit stream. Conversion back to a fixed point representation is achieved by right shifting the mantissa by its exponent. This process is represented by the formula transform_coefficient[k] = mantissa[k] >> exponent[k] ; No grouping is done for asymmetrically quantized mantissas. 7.3.3 Expansion of Mantissas for Symmetrical Quantization (1 ≤ bap ≤ 5) For bap values of 1 through 5 (1 ≤ bap ≤ 5), the mantissas are represented by coded values. The coded values are converted to standard 2’s complement fractional binary words by a table lookup. The number of bits indicated by a mantissa’s bap are extracted from the bit stream and right justified. This coded value is treated as a table index and is used to look up the mantissa value. 81 Advanced Television Systems Committee, Inc. Document A/52:2010 The resulting mantissa value is right shifted by the corresponding exponent to generate the transform coefficient value transform_coefficient[k] = quantization_table[mantissa_code[k]] >> exponent[k] ; The mapping of coded mantissa value into the actual mantissa value is shown in tables Table 7.19 through Table 7.23. 7.3.4 Dither for Zero Bit Mantissas (bap = 0) The AC-3 decoder uses random noise (dither) values instead of quantized values when the number of bits allocated to a mantissa is zero (bap = 0). The use of the random value is conditional on the value of dithflag. When the value of dithflag is 1, the random noise value is used. When the value of dithflag is 0, a true zero value is used. There is a dithflag variable for each channel. Dither is applied after the individual channels are extracted from the coupling channel. In this way, the dither applied to each channel's upper frequencies is uncorrelated. Any reasonably random sequence may be used to generate the dither values. The word length of the dither values is not critical. Eight bits is sufficient. The optimum scaling for the dither words is to take a uniform distribution of values between –1 and +1, and scale this by 0.707, resulting in a uniform distribution between +0.707 and –0.707. A scalar of 0.75 is close enough to also be considered optimum. A scalar of 0.5 (uniform distribution between +0.5 and –0.5) is also acceptable. Once a dither value is assigned to a mantissa, the mantissa is right shifted according to its exponent to generate the corresponding transform coefficient transform_coefficient[k] = scaled_dither_value >> exponent[k] ; Table 7.19 bap = 1 (3-Level) Quantization Mantissa Mantissa Value Code 0 –2./3 1 0 2 2./3 Table 7.20 bap = 2 (5-Level) Quantization Mantissa Mantissa Value Code 0 –4./5 1 –2./5 2 0 3 2./5 4 4./5 82 Digital Audio Compression Standard 22 November 2010 Table 7.21 bap = 3 (7-Level) Quantization Mantissa Mantissa Value Code 0 –6./7 1 –4./7 2 –2./7 3 0 4 2./7 5 4./7 6 6./7 Table 7.22 bap = 4 (11-Level) Quantization Mantissa Mantissa Value Code 0 –10./11 1 –8./11 2 –6./11 3 –4./11 4 –2./11 5 0 6 2./11 7 4./11 8 6./11 9 8./11 10 10./11 83 Advanced Television Systems Committee, Inc. Document A/52:2010 Table 7.23 bap = 5 (15-Level) Quantization Mantissa Mantissa Value Code 0 –14./15 1 –12./15 2 –10./15 3 –8./15 4 –6./15 5 –4./15 6 –2./15 7 0 8 2./15 9 4./15 10 6./15 11 8./15 12 10./15 13 12./15 14 14./15 7.3.5 Ungrouping of Mantissas In the case when bap = 1, 2, or 4, the coded mantissa values are compressed further by combining 3 level words and 5 level words into separate groups representing triplets of mantissas, and 11 level words into groups representing pairs of mantissas. Groups are filled in the order that the mantissas are processed. If the number of mantissas in an exponent set does not fill an integral number of groups, the groups are shared across exponent sets. The next exponent set in the block continues filling the partial groups. If the total number of 3 or 5 level quantized transform coefficient derived words are not each divisible by 3, or if the 11 level words are not divisible by 2, the final groups of a block are padded with dummy mantissas to complete the composite group. Dummies are ignored by the decoder. Groups are extracted from the bit stream using the length derived from bap. Three level quantized mantissas (bap = 1) are grouped into triples each of 5 bits. Five level quantized mantissas (bap = 2) are grouped into triples each of 7 bits. Eleven level quantized mantissas (bap = 4) are grouped into pairs each of 7 bits. 84 Digital Audio Compression Standard 22 November 2010 Encoder equations bap = 1: group_code = 9 * mantissa_code[a] + 3 * mantissa_code[b] + mantissa_code[c] ; bap = 2: group_code = 25 * mantissa_code[a] + 5 * mantissa_code[b] + mantissa_code[c] ; bap = 4: group_code = 11 * mantissa_code[a] + mantissa_code[b] ; Decoder equations bap = 1: mantissa_code[a] = truncate (group_code / 9) ; mantissa_code[b] = truncate ((group_code % 9) / 3 ) ; mantissa_code[c] = (group_code % 9) % 3 ; bap = 2: mantissa_code[a] = truncate (group_code / 25) ; mantissa_code[b] = truncate ((group_code % 25) / 5 ) ; mantissa_code[c] = (group_code % 25) % 5 ; bap = 4: mantissa_code[a] = truncate (group_code / 11) ; mantissa_code[b] = group_code % 11 ; where mantissa a comes before mantissa b, which comes before mantissa c 7.4 Channel Coupling 7.4.1 Overview If enabled, channel coupling is performed on encode by averaging the transform coefficients across channels that are included in the coupling channel. Each coupled channel has a unique set of coupling coordinates which are used to preserve the high frequency envelopes of the original channels. The coupling process is performed above a coupling frequency that is defined by the cplbegf value. The decoder converts the coupling channel back into individual channels by multiplying the coupled channel transform coefficient values by the coupling coordinate for that channel and frequency sub-band. An additional processing step occurs for the 2/0 mode. If the phsflginu bit = 1 or the equivalent state is continued from a previous block, then phase restoration bits are sent in the bit stream via phase flag bits. The phase flag bits represent the coupling sub-bands in a frequency ascending order. If a phase flag bit = 1 for a particular sub-band, all the right channel transform coefficients within that coupled sub-band are negated after modification by the coupling coordinate, but before inverse transformation. 7.4.2 Sub-Band Structure for Coupling Transform coefficients # 37 through # 252 are grouped into 18 sub-bands of 12 coefficients each, as shown in Table 7.24. The parameter cplbegf indicates the number of the coupling sub-band which is the first to be included in the coupling process. Below the frequency (or transform coefficient number) indicated by cplbegf, all channels are independently coded. Above the frequency indicated by cplbegf, channels included in the coupling process (chincpl[ch] = 1) share the 85 Advanced Television Systems Committee, Inc. Document A/52:2010 common coupling channel up to the frequency (or tc #) indicated by cplendf. The coupling channel is coded up to the frequency (or tc #) indicated by cplendf, which indicates the last coupling sub- band which is coded. The parameter cplendf is interpreted by adding 2 to its value, so the last coupling sub-band which is coded can range from 2–17. Table 7.24 Coupling Sub-Bands Coupling Low tc High tc # lf Cutoff (kHz) hf Cutoff (kHz) lf Cutoff (kHz) hf Cutoff (kHz) Subband # # @ fs=48 kHz @ fs=48 kHz @ fs=44.1 kHz @ fs=44.1 kHz 0 37 48 3.42 4.55 3.14 4.18 1 49 60 4.55 5.67 4.18 5.21 2 61 72 5.67 6.80 5.21 6.24 3 73 84 6.80 7.92 6.24 7.28 4 85 96 7.92 9.05 7.28 8.31 5 97 108 9.05 10.17 8.31 9.35 6 109 120 10.17 11.30 9.35 10.38 7 121 132 11.30 12.42 10.38 11.41 8 133 144 12.42 13.55 11.41 12.45 9 145 156 13.55 14.67 12.45 13.48 10 157 168 14.67 15.80 13.48 14.51 11 169 180 15.80 16.92 14.51 15.55 12 181 192 16.92 18.05 15.55 16.58 13 193 204 18.05 19.17 16.58 17.61 14 205 216 19.17 20.30 17.61 18.65 15 217 228 20.30 21.42 18.65 19.68 16 229 240 21.42 22.55 19.68 20.71 17 241 252 22.55 23.67 20.71 21.75 Note: At 32 kHz sampling rate the sub-band frequency ranges are 2/3 the values of those for 48 kHz. The coupling sub-bands are combined into coupling bands for which coupling coordinates are generated (and included in the bit stream). The coupling band structure is indicated by cplbndstrc[sbnd]. Each bit of the cplbndstrc[] array indicates whether the sub-band indicated by the index is combined into the previous (lower in frequency) coupling band. Coupling bands are thus made from integral numbers of coupling sub-bands. (See Section 5.4.3.13.) 7.4.3 Coupling Coordinate Format Coupling coordinates exist for each coupling band [bnd] in each channel [ch] which is coupled (chincp[ch] == 1). Coupling coordinates are sent in a floating point format. The exponent is sent as a 4-bit value (cplcoexp[ch][bnd]) indicating the number of right shifts which should be applied to the fractional mantissa value. The mantissas are transmitted as 4-bit values (cplcomant[ch][bnd]) which must be properly scaled before use. Mantissas are unsigned values so a sign bit is not used. Except for the limiting case where the exponent value = 15, the mantissa value is known to be between 0.5 and 1.0. Therefore, when the exponent value < 15, the msb of the mantissa is always equal to ‘1’ and is not transmitted; the next 4 bits of the mantissa are transmitted. This provides one additional bit of resolution. When the exponent value = 15 the mantissa value is generated by dividing the 4-bit value of cplcomant by 16. When the exponent value is < 15 the mantissa value is generated by adding 16 to the 4-bit value of cplcomant and then dividing the sum by 32. 86 Digital Audio Compression Standard 22 November 2010 Coupling coordinate dynamic range is increased beyond what the 4-bit exponent can provide by the use of a per channel 2-bit master coupling coordinate (mstrcplco[ch]) which is used to range all of the coupling coordinates within that channel. The exponent values for each channel are increased by 3 times the value of mstrcplco which applies to that channel. This increases the dynamic range of the coupling coordinates by an additional 54 dB. The following pseudo code indicates how to generate the coupling coordinate (cplco) for each coupling band [bnd] in each channel [ch]. Pseudo Code if (cplcoexp[ch, bnd] == 15) { cplco_temp[ch,bnd] = cplcomant[ch,bnd] / 16 ; } else { cplco_temp[ch,bnd] = (cplcomant[ch,bnd] + 16) / 32 ; } cplco[ch,bnd] = cplco_temp[ch,bnd] >> (cplcoexp[ch,bnd] + 3 * mstrcplco[ch]) ; Using the cplbndstrc[] array, the values of coupling coordinates which apply to coupling bands are converted (by duplicating values as indicated by values of ‘1’ in cplbandstrc[]) to values which apply to coupling sub-bands. Individual channel mantissas are then reconstructed from the coupled channel as follows: Pseudo Code for (sbnd = cplbegf; sbnd < 3 + cplendf; sbnd++) { for (bin = 0; bin < 12; bin++) { chmant[ch, sbnd*12+bin+37] = cplmant[sbnd*12+bin+37] * cplco[ch, sbnd] * 8 ; } } 7.5 Rematrixing 7.5.1 Overview Rematrixing in AC-3 is a channel combining technique in which sums and differences of highly correlated channels are coded rather than the original channels themselves. That is, rather than code and pack left and right in a two channel coder, we construct left' = 0.5 * (left + right) ; right' = 0.5 * (left – right) ; The usual quantization and data packing operations are then performed on left' and right'. Clearly, if the original stereo signal were identical in both channels (i.e., two-channel mono), this technique will result in a left' signal that is identical to the original left and right channels, and a 87 Advanced Television Systems Committee, Inc. Document A/52:2010 right' signal that is identically zero. As a result, we can code the right' channel with very few bits, and increase accuracy in the more important left' channel. This technique is especially important for preserving Dolby Surround compatibility. To see this, consider a two channel mono source signal such as that described above. A Dolby Pro Logic decoder will try to steer all in-phase information to the center channel, and all out-of-phase information to the surround channel. If rematrixing is not active, the Pro Logic decoder will receive the following signals received left = left + QN1 ; received right = right + QN2 ; where QN1 and QN2 are independent (i.e., uncorrelated) quantization noise sequences, which correspond to the AC-3 coding algorithm quantization, and are program-dependent. The Pro Logic decoder will then construct center and surround channels as center = 0.5 * (left + QN1) + 0.5 * (right + QN2) ; surround = 0.5 * (left + QN1) – 0.5 * (right + QN2) ; /* ignoring the 90 degree phase shift */ In the case of the center channel, QN1 and QN2 add, but remain masked by the dominant signal left + right. In the surround channel, however, left – right cancels to zero, and the surround speakers are left to reproduce the difference in the quantization noise sequences (QN1 – QN2). If channel rematrixing is active, the center and surround channels will be more easily reproduced as center = left' + QN1 ; surround = right' + QN2 ; In this case, the quantization noise in the surround channel QN2 is much lower in level, and it is masked by the difference signal, right'. 7.5.2 Frequency Band Definitions In AC-3, rematrixing is performed independently in separate frequency bands. There are four bands with boundary locations dependent on coupling information. The boundary locations are by coefficient bin number, and the corresponding rematrixing band frequency boundaries change with sampling frequency. The following tables indicate the rematrixing band frequencies for sampling rates of 48 kHz and 44.1 kHz. At 32 kHz sampling rate the rematrixing band frequencies are 2/3 the values of those shown for 48 kHz. 7.5.2.1 Coupling Not in Use If coupling is not in use (cplinu = 0), then there are 4 rematrixing bands, (nrematbd = 4). 88 Digital Audio Compression Standard 22 November 2010 Table 7.25 Rematrix Banding Table A Band # Low Coeff High Coeff # Low Freq High Freq (kHz) Low Freq (kHz) High Freq (kHz) # (kHz) fs = 48 kHz fs = 44.1 kHz fs = 44.1 kHz fs = 48 kHz 0 13 24 1.17 2.30 1.08 2.11 1 25 36 2.30 3.42 2.11 3.14 2 37 60 3.42 5.67 3.14 5.21 3 61 252 5.67 23.67 5.21 21.75 7.5.2.2 Coupling in Use, cplbegf > 2 If coupling is in use (cplinu = 1), and cplbegf > 2, there are 4 rematrixing bands (nrematbd = 4). The last (fourth) rematrixing band ends at the point where coupling begins. Table 7.26 Rematrixing Banding Table B Band # Low Coeff # High Coeff # Low Freq High Freq (kHz) Low Freq High Freq (kHz) (kHz) fs = 48 kHz (kHz) fs = 44.1 kHz fs = 48 kHz fs = 44.1 kHz 0 13 24 1.17 2.30 1.08 2.11 1 25 36 2.30 3.42 2.11 3.14 2 37 60 3.42 5.67 3.14 5.21 3 61 A 5.67 B 5.21 C A = 36 + cplbegf * 12 B = (A+1/2) * 0.09375 kHz C = (A+1/2) * 0.08613 kHz 7.5.2.3 Coupling in use, 2 ≥ cplbegf > 0 If coupling is in use (cplinu = 1), and 2 ≥ cplbegf > 0, there are 3 rematrixing bands (nrematbd = 3). The last (third) rematrixing band ends at the point where coupling begins. Table 7.27 Rematrixing Banding Table C Band # Low Coeff # High Coeff # Low Freq High Freq (kHz) Low Freq High Freq (kHz) (kHz) fs = 48 kHz (kHz) fs = 44.1 kHz fs = 48 kHz fs = 44.1 kHz 0 13 24 1.17 2.30 1.08 2.11 1 25 36 2.30 3.42 2.11 3.14 2 37 A 3.42 B 3.14 C A = 36 + cplbegf * 12 B = (A+1/2) * 0.09375 kHz C = (A+1/2) * 0.08613 kHz 7.5.2.4 Coupling in Use, cplbegf=0 If coupling is in use (cplinu = 1), and cplbegf = 0, there are 2 rematrixing bands (nrematbd = 2). 89 Advanced Television Systems Committee, Inc. Document A/52:2010 Table 7.28 Rematrixing Banding Table D Band # Low Coeff # High Coeff # Low Freq High Freq (kHz) Low Freq High Freq (kHz) (kHz) fs = 48 kHz (kHz) fs = 44.1 kHz fs = 48 kHz fs = 44.1 kHz 0 13 24 1.17 2.30 1.08 2.11 1 25 36 2.30 3.42 2.11 3.14 7.5.3 Encoding Technique If the 2/0 mode is selected, then rematrixing is employed by the encoder. The squares of the transform coefficients are summed up over the previously defined rematrixing frequency bands for the following combinations: L, R, L+R, L–R. Pseudo code if (minimum sum for a rematrixing sub-band n is L or R) { the variable rematflg[n] = 0 ; transmitted left = input L ; transmitted right = input R ; } if (minimum sum for a rematrixing sub-band n is L+R or L-R) { the variable rematflg[n] = 1 ; transmitted left = 0.5 * input (L+R) ; transmitted right = 0.5 * input (L-R) ; } This selection of matrix combination is done on a block by block basis. The remaining encoder processing of the transmitted left and right channels is identical whether or not the rematrixing flags are 0 or 1. 7.5.4 Decoding Technique For each rematrixing band, a single bit (the rematrix flag) is sent in the data stream, indicating whether or not the two channels have been rematrixed for that band. If the bit is clear, no further operation is required. If the bit is set, the AC-3 decoder performs the following operation to restore the individual channels: left(band n) = received left(band n) + received right(band n) ; right(band n) = received left(band n) – received right(band n) ; Note that if coupling is not in use, the two channels may have different bandwidths. As such, rematrixing is only applied up to the lower bandwidth of the two channels. Regardless of the actual bandwidth, all four rematrixing flags are sent in the data stream (assuming the rematrixing strategy bit is set). 90 Digital Audio Compression Standard 22 November 2010 7.6 Dialogue Normalization The AC-3 syntax provides elements which allow the encoded bit stream to satisfy listeners in many different situations. The dialnorm element allows for uniform reproduction of spoken dialogue when decoding any AC-3 bit stream. 7.6.1 Overview When audio from different sources is reproduced, the apparent loudness often varies from source to source. The different sources of audio might be different program segments during a broadcast (i.e., the movie vs. a commercial message); different broadcast channels; or different media (disc vs. tape). The AC-3 coding technology solves this problem by explicitly coding an indication of loudness into the AC-3 bit stream. The subjective level of normal spoken dialogue is used as a reference. The 5-bit dialogue normalization word which is contained in bsi, dialnorm, is an indication of the subjective loudness of normal spoken dialogue compared to digital 100 percent. The 5-bit value is interpreted as an unsigned integer (most significant bit transmitted first) with a range of possible values from 1 to 31. The unsigned integer indicates the headroom in dB above the subjective dialogue level. This value can also be interpreted as an indication of how many dB the subjective dialogue level is below digital 100 percent. The dialnorm value is not directly used by the AC-3 decoder. Rather, the value is used by the section of the sound reproduction system responsible for setting the reproduction volume; e.g., the system volume control. The system volume control is generally set based on listener input as to the desired loudness, or sound pressure level (SPL). The listener adjusts a volume control which generally directly adjusts the reproduction system gain. With AC-3 and the dialnorm value, the reproduction system gain becomes a function of both the listeners desired reproduction sound pressure level for dialogue, and the dialnorm value which indicates the level of dialogue in the audio signal. The listener is thus able to reliably set the volume level of dialogue, and the subjective level of dialogue will remain uniform no matter which AC-3 program is decoded. Example: The listener adjusts the volume control to 67 dB. (With AC-3 dialogue normalization, it is possible to calibrate a system volume control directly in sound pressure level, and the indication will be accurate for any AC-3 encoded audio source). A high quality entertainment program is being received, and the AC-3 bit stream indicates that dialogue level is 25 dB below 100 percent digital level. The reproduction system automatically sets the reproduction system gain so that full scale digital signals reproduce at a sound pressure level of 92 dB. The spoken dialogue (down 25 dB) will thus reproduce at 67 dB SPL. The broadcast program cuts to a commercial message, which has dialogue level at –15 dB with respect to 100 percent digital level. The system level gain automatically drops, so that digital 100 percent is now reproduced at 82 dB SPL. The dialogue of the commercial (down 15 dB) reproduces at a 67 dB SPL, as desired. In order for the dialogue normalization system to work, the dialnorm value must be communicated from the AC-3 decoder to the system gain controller so that dialnorm can interact 91 Advanced Television Systems Committee, Inc. Document A/52:2010 with the listener adjusted volume control. If the volume control function for a system is performed as a digital multiply inside the AC-3 decoder, then the listener selected volume setting must be communicated into the AC-3 decoder. The listener selected volume setting and the dialnorm value must be brought together and combined in order to adjust the final reproduction system gain. Adjustment of the system volume control is not an AC-3 function. The AC-3 bit stream simply conveys useful information which allows the system volume control to be implemented in a way which automatically removes undesirable level variations between program sources. It is mandatory that the dialnorm value and the user selected volume setting both be used to set the reproduction system gain. 7.7 Dynamic Range Compression 7.7.1 Dynamic Range Control; dynrng, dynrng2 The dynrng element allows the program provider to implement subjectively pleasing dynamic range reduction for most of the intended audience, while allowing individual members of the audience the option to experience more (or all) of the original dynamic range. 7.7.1.1 Overview A consistent problem in the delivery of audio programming is that different members of the audience wish to enjoy different amounts of dynamic range. Original high quality programming (such as feature films) are typically mixed with quite a wide dynamic range. Using dialogue as a reference, loud sounds like explosions are often 20 dB or more louder, and faint sounds like leaves rustling may be 50 dB quieter. In many listening situations it is objectionable to allow the sound to become very loud, and thus the loudest sounds must be compressed downwards in level. Similarly, in many listening situations the very quiet sounds would be inaudible, and must be brought upwards in level to be heard. Since most of the audience will benefit from a limited program dynamic range, soundtracks which have been mixed with a wide dynamic range are generally compressed: the dynamic range is reduced by bringing down the level of the loud sounds and bringing up the level of the quiet sounds. While this satisfies the needs of much of the audience, it removes the ability of some in the audience to experience the original sound program in its intended form. The AC-3 audio coding technology solves this conflict by allowing dynamic range control values to be placed into the AC-3 bit stream. The dynamic range control values, dynrng, indicate a gain change to be applied in the decoder in order to implement dynamic range compression. Each dynrng value can indicate a gain change of ± 24 dB. The sequence of dynrng values are a compression control signal. An AC-3 encoder (or a bit stream processor) will generate the sequence of dynrng values. Each value is used by the AC- 3 decoder to alter the gain of one or more audio blocks. The dynrng values typically indicate gain reduction during the loudest signal passages, and gain increases during the quiet passages. For the listener, it is desirable to bring the loudest sounds down in level towards dialogue level, and the quiet sounds up in level, again towards dialogue level. Sounds which are at the same loudness as the normal spoken dialogue will typically not have their gain changed. The compression is actually applied to the audio in the AC-3 decoder. The encoded audio has full dynamic range. It is permissible for the AC-3 decoder to (optionally, under listener control) ignore the dynrng values in the bit stream. This will result in the full dynamic range of the audio being reproduced. It is also permissible (again under listener control) for the decoder to use some fraction of the dynrng control value, and to use a different fraction of positive or negative values. 92 Digital Audio Compression Standard 22 November 2010 The AC-3 decoder can thus reproduce either fully compressed audio (as intended by the compression control circuit in the AC-3 encoder); full dynamic range audio; or audio with partially compressed dynamic range, with different amounts of compression for high level signals and low level signals. Example: A feature film soundtrack is encoded into AC-3. The original program mix has dialogue level at –25 dB. Explosions reach full scale peak level of 0 dB. Some quiet sounds which are intended to be heard by all listeners are 50 dB below dialogue level (or –75 dB). A compression control signal (sequence of dynrng values) is generated by the AC-3 encoder. During those portions of the audio program where the audio level is higher than dialogue level the dynrng values indicate negative gain, or gain reduction. For full scale 0 dB signals (the loudest explosions), gain reduction of –15 dB is encoded into dynrng. For very quiet signals, a gain increase of 20 dB is encoded into dynrng. A listener wishes to reproduce this soundtrack quietly so as not to disturb anyone, but wishes to hear all of the intended program content. The AC-3 decoder is allowed to reproduce the default, which is full compression. The listener adjusts dialogue level to 60 dB SPL. The explosions will only go as loud as 70 dB (they are 25 dB louder than dialogue but get –15 dB of gain applied), and the quiet sounds will reproduce at 30 dB SPL (20 dB of gain is applied to their original level of 50 dB below dialogue level). The reproduced dynamic range will be 70 dB – 30 dB = 40 dB. The listening situation changes, and the listener now wishes to raise the reproduction level of dialogue to 70 dB SPL, but still wishes to limit how loud the program plays. Quiet sounds may be allowed to play as quietly as before. The listener instructs the AC-3 decoder to continue using the dynrng values which indicate gain reduction, but to attenuate the values which indicate gain increases by a factor of 1/2. The explosions will still reproduce 10 dB above dialogue level, which is now 80 dB SPL. The quiet sounds are now increased in level by 20 dB / 2 = 10 dB. They will now be reproduced 40 dB below dialogue level, at 30 dB SPL. The reproduced dynamic range is now 80 dB – 30 dB = 50 dB. Another listener wishes the full original dynamic range of the audio. This listener adjusts the reproduced dialogue level to 75 dB SPL, and instructs the AC-3 decoder to ignore the dynamic range control signal. For this listener the quiet sounds reproduce at 25 dB SPL, and the explosions hit 100 dB SPL. The reproduced dynamic range is 100 dB – 25 dB = 75 dB. This reproduction is exactly as intended by the original program producer. In order for this dynamic range control method to be effective, it should be used by all program providers. Since all broadcasters wish to supply programming in the form that is most usable by their audience, nearly all broadcasters will apply dynamic range compression to any audio program which has a wide dynamic range. This compression is not reversible unless it is implemented by the technique embedded in AC-3. If broadcasters make use of the embedded AC- 3 dynamic range control system, then listeners can have some control over their reproduced 93 Advanced Television Systems Committee, Inc. Document A/52:2010 dynamic range. Broadcasters must be confident that the compression characteristic that they introduce into AC-3 will, by default, be heard by the listeners. Therefore, the AC-3 decoder shall, by default, implement the compression characteristic indicated by the dynrng values in the data stream. AC-3 decoders may optionally allow listener control over the use of the dynrng values, so that the listener may select full or partial dynamic range reproduction. 7.7.1.2 Detailed Implementation The dynrng field in the AC-3 data stream is 8-bits in length. In the case that acmod = 0 (1+1 mode, or 2 completely independent channels) dynrng applies to the first channel (Ch1), and dynrng2 applies to the second channel (Ch2). While dynrng is described below, dynrng2 is handled identically. The dynrng value may be present in any audio block. When the value is not present, the value from the previous block is used, except for block 0. In the case of block 0, if a new value of dynrng is not present, then a value of ‘0000 0000’ should be used. The most significant bit of dynrng (and of dynrng2) is transmitted first. The first three bits indicate gain changes in 6.02 dB increments which can be implemented with an arithmetic shift operation. The following five bits indicate linear gain changes, and require a 6-bit multiply. We will represent the 3 and 5 bit fields of dynrng as following: X0 X1 X2 . Y3 Y4 Y5 Y6 Y7 The meaning of the X values is most simply described by considering X to represent a 3-bit signed integer with values from –4 to 3. The gain indicated by X is then (X + 1) * 6.02 dB. Table 7.29 shows this in detail. Table 7.29 Meaning of 3 msb of dynrng X0 X1 X2 Integer Value Gain Indicated Arithmetic Shifts 0 1 1 3 +24.08 dB 4 left 0 1 0 2 +18.06 dB 3 left 0 0 1 1 +12.04 dB 2 left 0 0 0 0 +6.02 dB 1 left 1 1 1 –1 0 dB None 1 1 0 –2 –6.02 dB 1 right 1 0 1 –3 –12.04 dB 2 right 1 0 0 –4 –18.06 dB 3 right The value of Y is a linear representation of a gain change of up to 6 dB. Y is considered to be an unsigned fractional integer, with a leading value of 1, or: 0.1Y3 Y4 Y5 Y6 Y7 (base 2). Y can represent values between 0.1111112 (or 63/64) and 0.1000002 (or 1/2). Thus, Y can represent gain changes from –0.14 dB to –6.02 dB. The combination of X and Y values allows dynrng to indicate gain changes from 24.08 – 0.14 = +23.95 dB, to –18.06 – 6.02 = –24.08 dB. The bit code of ‘0000 0000’ indicates 0 dB (unity) gain. Partial Compression The dynrng value may be operated on in order to make it represent a gain change which is a fraction of the original value. In order to alter the amount of 94 Digital Audio Compression Standard 22 November 2010 compression which will be applied, consider the dynrng to represent a signed fractional number, or X0 . X1 X 2 Y3 Y 4 Y5 Y 6 Y7 where X0 is the sign bit and X1 X2 Y3 Y4 Y5 Y6 Y7 are a 7-bit fraction. This 8 bit signed fractional number may be multiplied by a fraction indicating the fraction of the original compression to apply. If this value is multiplied by 1/2, then the compression range of ±24 dB will be reduced to ±12 dB. After the multiplicative scaling, the 8-bit result is once again considered to be of the original form X0 X1 X2 . Y3 Y4 Y5 Y6 Y7 and used normally. 7.7.2 Heavy Compression; compr, compr2 The compr element allows the program provider (or broadcaster) to implement a large dynamic range reduction (heavy compression) in a way which assures that a monophonic downmix will not exceed a certain peak level. The heavily compressed audio program may be desirable for certain listening situations such as movie delivery to a hotel room, or to an airline seat. The peak level limitation is useful when, for instance, a monophonic downmix will feed an RF modulator and overmodulation must be avoided. 7.7.2.1 Overview Some products which decode the AC-3 bit stream will need to deliver the resulting audio via a link with very restricted dynamic range. One example is the case of a television signal decoder which must modulate the received picture and sound onto an RF channel in order to deliver a signal usable by a low cost television receiver. In this situation, it is necessary to restrict the maximum peak output level to a known value with respect to dialogue level, in order to prevent overmodulation. Most of the time, the dynamic range control signal, dynrng, will produce adequate gain reduction so that the absolute peak level will be constrained. However, since the dynamic range control system is intended to implement a subjectively pleasing reduction in the range of perceived loudness, there is no assurance that it will control instantaneous signal peaks adequately to prevent overmodulation. In order to allow the decoded AC-3 signal to be constrained in peak level, a second control signal, compr, (compr2 for Ch2 in 1+1 mode) may be present in the AC-3 data stream. This control signal should be present in all bit streams which are intended to be receivable by, for instance, a television set top decoder. The compr control signal is similar to the dynrng control signal in that it is used by the decoder to alter the reproduced audio level. The compr control signal has twice the control range as dynrng (±48 dB compared to ±24 dB) with 1/2 the resolution (0.5 dB vs. 0.25 dB). Also, since the compr control signal lives in BSI, it only has a time resolution of an AC-3 frame (32 ms) instead of a block (5.3 ms). Products which require peak audio level to be constrained should use compr instead of dynrng when compr is present in BSI. Since most of the time the use of dynrng will prevent large peak levels, the AC-3 encoder may only need to insert compr occasionally; i.e., during those instants when the use of dynrng would lead to excessive peak level. If the decoder has been instructed to use compr, and compr is not present for a particular frame, then the dynrng control signal shall be used for that frame. 95 Advanced Television Systems Committee, Inc. Document A/52:2010 In some applications of AC-3, some receivers may wish to reproduce a very restricted dynamic range. In this case, the compr control signal may be present at all times. Then, the use of compr instead of dynrng will allow the reproduction of audio with very limited dynamic range. This might be useful, for instance, in the case of audio delivery to a hotel room or an airplane seat. 7.7.2.2 Detailed Implementation The compr field in the AC-3 data stream is 8-bits in length. In the case that acmod = 0 (1+1 mode, or 2 completely independent channels) compr applies to the first channel (Ch1), and compr2 applies to the second channel (Ch2). While compr is described below (for Ch1), compr2 is handled identically (but for Ch2). The most significant bit is transmitted first. The first four bits indicate gain changes in 6.02 dB increments which can be implemented with an arithmetic shift operation. The following four bits indicate linear gain changes, and require a 5-bit multiply. We will represent the two 4-bit fields of compr as follows: X0 X1 X2 X3 . Y4 Y5 Y6 Y7 The meaning of the X values is most simply described by considering X to represent a 4-bit signed integer with values from –8 to +7. The gain indicated by X is then (X + 1) * 6.02 dB. Table 7.30 shows this in detail. Table 7.30 Meaning of 4 msb of compr X0 X1 X2 X3 Integer Value Gain Indicated Arithmetic Shifts 0 1 1 1 7 +48.16 dB 8 left 0 1 1 0 6 +42.14 dB 7 left 0 1 0 1 5 +36.12 dB 6 left 0 1 0 0 4 +30.10 dB 5 left 0 0 1 1 3 +24.08 dB 4 left 0 0 1 0 2 +18.06 dB 3 left 0 0 0 1 1 +12.04 dB 2 left 0 0 0 0 0 +6.02 dB 1 left 1 1 1 1 –1 0 dB None 1 1 1 0 –2 –6.02 dB 1 right 1 1 0 1 –3 –12.04 dB 2 right 1 1 0 0 –4 –18.06 dB 3 right 1 0 1 1 –5 –24.08 dB 4 right 1 0 1 0 –6 –30.10 dB 5 right 1 0 0 1 –7 –36.12 dB 6 right 1 0 0 0 –8 –42.14 dB 7 right The value of Y is a linear representation of a gain change of up to –6 dB. Y is considered to be an unsigned fractional integer, with a leading value of 1, or: 0.1 Y4 Y5 Y6 Y7 (base 2). Y can represent values between 0.111112 (or 31/32) and 0.100002 (or 1/2). Thus, Y can represent gain changes from –0.28 dB to –6.02 dB. The combination of X and Y values allows compr to indicate gain changes from 48.16 – 0.28 = +47.89 dB, to –42.14 – 6.02 = –48.16 dB. 96 Digital Audio Compression Standard 22 November 2010 7.8 Downmixing In many reproduction systems, the number of loudspeakers will not match the number of encoded audio channels. In order to reproduce the complete audio program, downmixing is required. It is important that downmixing be standardized so that program providers can be confident of how their program will be reproduced over systems with various numbers of loudspeakers. With standardized downmixing equations, program producers can monitor how the downmixed version will sound and make any alterations necessary so that acceptable results are achieved for all listeners. The program provider can make use of the cmixlev and smixlev syntactical elements in order to affect the relative balance of center and surround channels with respect to the left and right channels. Downmixing of the lfe channel is optional. An ideal downmix would have the lfe channel reproduce at an acoustic level of +10 dB with respect to the left and right channels. Since the inclusion of this channel is optional, any downmix coefficient may be used in practice. Care should be taken to assure that loudspeakers are not overdriven by the full scale low frequency content of the lfe channel. 7.8.1 General Downmix Procedure The following pseudo code describes how to arrive at un-normalized downmix coefficients. In a practical implementation it may be necessary to then normalize the downmix coefficients in order to prevent any possibility of overload. Normalization is achieved by attenuating all downmix coefficients equally, such that the sum of coefficients used to create any single output channel never exceeds 1. Pseudo code downmix() { if (acmod == 0) /* 1+1 mode, dual independent mono channels present */ { if (output_nfront == 1) /* 1 front loudspeaker (center) */ { if (dualmode == Chan 1) /* Ch1 output requested */ { route left into center ; } else if (dualmode == Chan 2) /* Ch2 output requested */ { route right into center ; } else { mix left into center with –6 dB gain ; mix right into center with –6 dB gain ; } } else if (output_nfront == 2) /* 2 front loudspeakers (left, right) */ 97 Advanced Television Systems Committee, Inc. Document A/52:2010 { if (dualmode == Stereo) /* output of both mono channels requested */ { route left into left ; route right into right ; } else if (dualmode == Chan 1) { mix left into left with –3 dB gain ; mix left into right with –3 dB gain ; } else if (dualmode == Chan 2) { mix right into left with –3 dB gain ; mix right into right with –3 dB gain ; } else /* mono sum of both mono channels requested */ { mix left into left with –6 dB gain ; mix right into left with –6 dB gain ; mix left into right with –6 dB gain ; mix right into right with –6 dB gain ; } } else /* output_nfront == 3 */ { if (dualmode == Stereo) { route left into left ; route right into right ; } else if (dualmode == Chan 1) { route left into center ; } else if (dualmode == Chan 2) { route right into center ; } else { mix left into center with –6 dB gain ; mix right into center with –6 dB gain ; } 98 Digital Audio Compression Standard 22 November 2010 } } else /* acmod > 0 */ { for i = { left, center, right, leftsur/monosur, rightsur } { if (exists(input_chan[i])) and (exists(output_chan[i])) { route input_chan[i] into output_chan[i] ; } } if (output_mode == 2/0 Dolby Surround compatible) /* 2 ch matrix encoded output requested */ { if (input_nfront != 2) { mix center into left with –3 dB gain ; mix center into right with –3 dB gain ; } if (input_nrear == 1) { mix -mono surround into left with –3 dB gain ; mix mono surround into right with –3 dB gain ; } else if (input_nrear == 2) { mix -left surround into left with –3 dB gain ; mix -right surround into left with –3 dB gain ; mix left surround into right with –3 dB gain ; mix right surround into right with –3 dB gain ; } } else if (output_mode == 1/0) /* center only */ { if (input_nfront != 1) { mix left into center with –3 dB gain ; mix right into center with –3 dB gain ; } if (input_nfront == 3) { mix center into center using clev and +3 dB gain ; } if (input_nrear == 1) { 99 Advanced Television Systems Committee, Inc. Document A/52:2010 mix mono surround into center using slev and –3 dB gain ; } else if (input_nrear == 2) { mix left surround into center using slev and –3 dB gain ; mix right surround into center using slev and –3 dB gain ; } } else /* more than center output requested */ { if (output_nfront == 2) { if (input_nfront == 1) { mix center into left with –3 dB gain ; mix center into right with –3 dB gain ; } else if (input_nfront == 3) { mix center into left using clev ; mix center into right using clev ; } } if (input_nrear == 1) /* single surround channel coded */ { if (output_nrear == 0) /* no surround loudspeakers */ { mix mono surround into left with slev and –3 dB gain ; mix mono surround into right with slev and –3 dB gain ; } else if (output_nrear == 2) /* two surround loudspeaker channels */ { mix mono srnd into left surround with –3 dB gain ; mix mono srnd into right surround with –3 dB gain ; } } else if (input_nrear == 2) /* two surround channels encoded */ { if (output_nrear == 0) { mix left surround into left using slev ; mix right surround into right using slev ; } else if (output_nrear == 1) . 100 Digital Audio Compression Standard 22 November 2010 { mix left srnd into mono surround with –3 dB gain ; mix right srnd into mono surround with –3 dB gain ; } } } } } The actual coefficients used for downmixing will affect the absolute level of the center channel. If dialogue level is to be established with absolute SPL calibration, this should be taken into account. 7.8.2 Downmixing Into Two Channels Let L, C, R, Ls, Rs refer to the 5 discrete channels which are to be mixed down to 2 channels. In the case of a single surround channel (n/1 modes), S refers to the single surround channel. Two types of downmix should be provided: downmix to an LtRt matrix surround encoded stereo pair; and downmix to a conventional stereo signal, LoRo. The downmixed stereo signal (LoRo, or LtRt) may be further mixed to mono, M, by a simple summation of the 2 channels. If the LtRt downmix is combined to mono, the surround information will be lost. The LoRo downmix is preferred when a mono signal is desired. Downmix coefficients shall have relative accuracy of at least ±0.25 dB. Prior to the scaling needed to prevent overflow, the general 3/2 downmix equations for an LoRo stereo signal are Lo = 1.0 * L + clev * C + slev * Ls ; Ro = 1.0 * R + clev * C + slev * Rs ; If LoRo are subsequently combined for monophonic reproduction, the effective mono downmix equation becomes M = 1.0 * L + 2.0 * clev * C + 1.0 * R + slev * Ls + slev * Rs ; If only a single surround channel, S, is present (3/1 mode) the downmix equations are Lo = 1.0 * L + clev * C + 0.7 * slev * S ; Ro = 1.0 * R + clev * C + 0.7 * slev * S ; M = 1.0 * L + 2.0 * clev * C + 1.0 * R + 1.4 * slev * S ; The values of clev and slev are indicated by the cmixlev and surmixlev bit fields in the bsi data, as shown in Table 5.9 and Table 5.10, respectively. If the cmixlev or surmixlev bit fields indicate the reserved state (value of ‘11’), the decoder should use the intermediate coefficient values indicated by the bit field value of 0 1. If the Center channel is missing (2/1 or 2/2 mode), the same equations may be used without the C term. If the surround channels are missing, the same equations may be used without the Ls, Rs, or S terms. 101 Advanced Television Systems Committee, Inc. Document A/52:2010 Prior to the scaling needed to prevent overflow, the 3/2 downmix equations for an LtRt stereo signal are Lt = 1.0 * L + 0.707 * C – 0.707 * Ls – 0.707 * Rs ; Rt = 1.0 * R + 0.707 * C + 0.707 * Ls + 0.707 * Rs ; If only a single surround channel, S, is present (3/1 mode) these equations become Lt = 1.0 L + 0.707 C – 0.707 S ; Rt = 1.0 R + 0.707 C + 0.707 S ; If the center channel is missing (2/2 or 2/1 mode) the C term is dropped. The actual coefficients used must be scaled downwards so that arithmetic overflow does not occur if all channels contributing to a downmix signal happen to be at full scale. For each audio coding mode, a different number of channels contribute to the downmix, and a different scaling could be used to prevent overflow. For simplicity, the scaling for the worst case may be used in all cases. This minimizes the number of coefficients required. The worst case scaling occurs when clev and slev are both 0.707. In the case of the LoRo downmix, the sum of the unscaled coefficients is 1 + 0.707 + 0.707 = 2.414, so all coefficients must be multiplied by 1/2.414 = 0.4143 (downwards scaling by 7.65 dB). In the case of the LtRt downmix, the sum of the unscaled coefficients is 1 + 0.707 + 0.707 + 0.707 = 3.121, so all coefficients must be multiplied by 1/ 3.121, or 0.3204 (downwards scaling by 9.89 dB). The scaled coefficients will typically be converted to binary values with limited wordlength. The 6-bit coefficients shown below have sufficient accuracy. In order to implement the LoRo 2-channel downmix, scaled (by 0.453) coefficient values are needed which correspond to the values of 1.0, 0.707, 0.596, 0.500, 0.354. Table 7.31 LoRo Scaled Downmix Coefficients Unscaled Scaled 6-bit Quantized Gain Relative Coefficient Coefficient Coefficient Coefficient Gain Error 1.0 0.414 26/64 –7.8 dB 0.0 dB --- 0.707 0.293 18/64 –11.0 dB –3.2 dB -0.2 dB 0.596 0.247 15/64 –12.6 –4.8 dB +0.3 dB dB 0.500 0.207 13/64 –13.8 –6.0 dB 0.0 dB dB 0.354 0.147 9/64 –17.0 dB –9.2 dB –0.2 dB In order to implement the LtRt 2-ch downmix, scaled (by 0.3204) coefficient values are needed which correspond to the values of 1.0 and 0.707. Table 7.32 LtRt Scaled Downmix Coefficients Unscaled Scaled 6-bit Quantized Gain Relative Coefficient Coefficient Coefficient Coefficient Gain Error 1.0 0.3204 20/64 –10.1 dB 0.0 dB --- 0.707 0.2265 14/64 –13.20 –3.1 dB –0.10 dB dB 102 Digital Audio Compression Standard 22 November 2010 If it is necessary to implement a mixdown to mono, a further scaling of 1/2 will have to be applied to the LoRo downmix coefficients to prevent overload of the mono sum of Lo+Ro. 7.9 Transform Equations and Block Switching 7.9.1 Overview The choice of analysis block length is fundamental to any transform-based audio coding system. A long transform length is most suitable for input signals whose spectrum remains stationary, or varies only slowly, with time. A long transform length provides greater frequency resolution, and hence improved coding performance for such signals. On the other hand, a shorter transform length, possessing greater time resolution, is more desirable for signals which change rapidly in time. Therefore, the time vs. frequency resolution tradeoff should be considered when selecting a transform block length. The traditional approach to solving this dilemma is to select a single transform length which provides the best tradeoff of coding quality for both stationary and dynamic signals. AC-3 employs a more optimal approach, which is to adapt the frequency/time resolution of the transform depending upon spectral and temporal characteristics of the signal being processed. This approach is very similar to behavior known to occur in human hearing. In transform coding, the adaptation occurs by switching the block length in a signal dependent manner. 7.9.2 Technique In the AC-3 transform block switching procedure, a block length of either 512 or 256 samples (time resolution of 10.7 or 5.3 ms for sampling frequency of 48 kHz) can be employed. Normal blocks are of length 512 samples. When a normal windowed block is transformed, the result is 256 unique frequency domain transform coefficients. Shorter blocks are constructed by taking the usual 512 sample windowed audio segment and splitting it into two segments containing 256 samples each. The first half of an MDCT block is transformed separately but identically to the second half of that block. Each half of the block produces 128 unique non-zero transform coefficients representing frequencies from 0 to fs/2, for a total of 256. This is identical to the number of coefficients produced by a single 512 sample block, but with two times improved temporal resolution. Transform coefficients from the two half-blocks are interleaved together on a coefficient-by-coefficient basis to form a single block of 256 values. This block is quantized and transmitted identically to a single long block. A similar, mirror image procedure is applied in the decoder during signal reconstruction. Transform coefficients for the two 256 length transforms arrive in the decoder interleaved together bin-by-bin. This interleaved sequence contains the same number of transform coefficients as generated by a single 512-sample transform. The decoder processes interleaved sequences identically to noninterleaved sequences, except during the inverse transformation described below. Prior to transforming the audio signal from time to frequency domain, the encoder performs an analysis of the spectral and/or temporal nature of the input signal and selects the appropriate block length. This analysis occurs in the encoder only, and therefore can be upgraded and improved without altering the existing base of decoders. A one bit code per channel per transform block (blksw[ch]) is embedded in the bit stream which conveys length information: (blksw[ch] = 0 or 1 for 512 or 256 samples, respectively). The decoder uses this information to deformat the bit stream, reconstruct the mantissa data, and apply the appropriate inverse transform equations. 103 Advanced Television Systems Committee, Inc. Document A/52:2010 7.9.3 Decoder Implementation TDAC transform block switching is accomplished in AC-3 by making an adjustment to the conventional forward and inverse transformation equations for the 256 length transform. The same window and FFT sine/cosine tables used for 512 sample blocks can be reused for inverse transforming the 256 sample blocks; however, the pre- and post-FFT complex multiplication twiddle requires an additional 128 table values for the block-switched transform. Since the input and output arrays for blksw[ch] = 1 are exactly one half of the length of those for blksw = 0, the size of the inverse transform RAM and associated buffers is the same with block switching as without. The adjustments required for inverse transforming the 256 sample blocks are: • The input array contains 128 instead of 256 coefficients. • The IFFT pre and post-twiddle use a different cosine table, requiring an additional 128 table values (64 cosine, 64 sine). • The complex IFFT employs 64 points instead of 128. The same FFT cosine table can be used with sub-sampling to retrieve only the even numbered entries. • The input pointers to the IFFT post-windowing operation are initialized to different start addresses, and operate modulo 128 instead of modulo 256. 7.9.4 Transformation Equations 7.9.4.1 512-Sample IMDCT Transform The following procedure describes the technique used for computing the IMDCT for a single N = 512 length real data block using a single N/4 point complex IFFT with simple pre- and post- twiddle operations. These are the inverse transform equations used when the blksw flag is set to zero (indicating absence of a transient, and 512 sample transforms). 1. Define the MDCT transform coefficients = X[k], k = 0, 1,...N/2–1. 2. Pre-IFFT complex multiply step. Compute N/4-point complex multiplication product Z[k], k = 0, 1,...N/4–1: Pseudo Code for (k=0; k max ≥ 0.25 1 0.25 > max ≥ 0.125 2 0.125 > max 3 If the encoder does not perform the step of finding the maximum absolute value within each block then the value of gainrng should be set to 0. The decoder may use the value of gainrng to pre-scale the transform coefficients prior to the transform and to post-scale the values after the transform. With careful design, the post-scaling process can be performed right at the PCM output stage allowing a 16-bit output buffer RAM to provide 18-bit dynamic range audio. 7.10 Error Detection There are several ways in which the AC-3 data may determine that errors are contained within a frame of data. The decoder may be informed of that fact by the transport system which has 109 Advanced Television Systems Committee, Inc. Document A/52:2010 b0 b1 + b2 b3 b13 b14 + b15 + u(x) Figure 7.1 Example LFSR circuit. delivered the data. The data integrity may be checked using the embedded CRCs. Also, some simple consistency checks on the received data can indicate that errors are present. The decoder strategy when errors are detected is user-definable. Possible responses include muting, block repeats, or frame repeats. The amount of error checking performed, and the behavior in the presence of errors are not specified in this standard, but are left to the application and implementation. 7.10.1 CRC Checking Each AC-3 frame contains two 16-bit CRC words. crc1 is the second 16-bit word of the frame, immediately following the sync word. crc2 is the last 16-bit word of the frame, immediately preceding the sync word of the following frame. crc1 applies to the first 5/8 of the frame, not including the sync word. crc2 provides coverage for the last 3/8 of the frame as well as for the entire frame (not including the sync word). Decoding of CRC word(s) allows errors to be detected. The following generator polynomial is used to generate each of the 16-bit CRC words x16 + x15 + x2 + 1 The 5/8 of a frame is defined in Table 7.35, and may be calculated by 5/8_framesize = truncate(framesize ÷ 2) + truncate(framesize ÷ 8) ; or 5/8_framesize = (int) (framesize>>1) + (int) (framesize>>3) ; where framesize is in units of 16-bit words. Table 7.35 shows the value of 5/8 of the frame size as a function of AC-3 bit-rate and audio sample rate. The CRC calculation may be implemented by one of several standard techniques. A convenient hardware implementation is a linear feedback shift register (LFSR). An example of an LFSR circuit for the above generator polynomial is given in Figure 7.1. Checking for valid CRC with the above circuit consists of resetting all registers to zero, and then shifting the AC-3 data bits serially into the circuit in the order in which they appear in the data stream. The sync word is not covered by either CRC (but is included in the indicated 5/ 8_framesize) so it should not be included in the CRC calculation. crc1 is considered valid if the above register contains all zeros after the first 5/8 of the frame has been shifted in. If the calculation is continued until all data in the frame has been shifted through, and the value is again equal to zero, then crc2 is considered valid. Some decoders may choose to only check crc2, and not check for a valid crc1 at the 5/8 point in the frame. If crc1 is invalid, it is possible to reset the 110 Digital Audio Compression Standard 22 November 2010 registers to zero and then check crc2. If crc2 then checks, then the last 3/8 of the frame is probably error free. This is of little utility however, since if errors are present in the initial 5/8 of a frame it is not possible to decode any audio from the frame even if the final 3/8 is error free. Note that crc1 is generated by encoders such that the CRC calculation will produce zero at the 5/8 point in the frame. It is not the value generated by calculating the CRC of the first 5/8 of the frame using the above generator polynomial. Therefore, decoders should not attempt to save crc1, calculate the CRC for the first 5/8 of the frame, and then compare the two. 111 Advanced Television Systems Committee, Inc. Document A/52:2010 Table 7.35 5/8_framesize Table; Number of Words in the First 5/8 of the Frame frmsizecod Nominal Bit- fs = 32 kHz fs = 44.1 kHz fs = 48 kHz Rate 5/8_framesize 5/8_framesize 5/8_framesize ‘000000’ (0) 32 kbps 60 42 40 ‘000001’ (0) 32 kbps 60 43 40 ‘000010’ (1) 40 kbps 75 53 50 ‘000011’ (1) 40 kbps 75 55 50 ‘000100’ (2) 48 kbps 90 65 60 ‘000101’ (2) 48 kbps 90 65 60 ‘000110’ (3) 56 kbps 105 75 70 ‘000111’ (3) 56 kbps 105 76 70 ‘001000’ (4) 64 kbps 120 86 80 ‘001001’ (4) 64 kbps 120 87 80 ‘001010’ (5) 80 kbps 150 108 100 ‘001011’ (5) 80 kbps 150 108 100 ‘001100’ (6) 96 kbps 180 130 120 ‘001101’ (6) 96 kbps 180 130 120 ‘001110’ (7) 112 kbps 210 151 140 ‘001111’ (7) 112 kbps 210 152 140 ‘010000’ (8) 128 kbps 240 173 160 ‘010001’ (8) 128 kbps 240 173 160 ‘010010’ (9) 160 kbps 300 217 200 ‘010011’ (9) 160 kbps 300 217 200 ‘010100’ (10) 192 kbps 360 260 240 ‘010101’ (10) 192 kbps 360 261 240 ‘010110’ (11) 224 kbps 420 303 280 ‘010111’ (11) 224 kbps 420 305 280 ‘011000’ (12) 256 kbps 480 347 320 ‘011001’ (12) 256 kbps 480 348 320 ‘011010’ (13) 320 kbps 600 435 400 ‘011011’ (13) 320 kbps 600 435 400 ‘011100’ (14) 384 kbps 720 521 480 ‘011101’ (14) 384 kbps 720 522 480 ‘011110’ (15) 448 kbps 840 608 560 ‘011111’ (15) 448 kbps 840 610 560 ‘100000’ (16) 512 kbps 960 696 640 ‘100001’ (16) 512 kbps 960 696 640 ‘100010’ (17) 576 kbps 1080 782 720 ‘100011’ (17) 576 kbps 1080 783 720 ‘100100’ (18) 640 kbps 1200 870 800 ‘100101’ (18) 640 kbps 1200 871 800 Syntactical block size restrictions within each frame (enforced by encoders), guarantee that blocks 0 and 1 are completely covered by crc1. Therefore, decoders may immediately begin processing block 0 when the 5/8 point in the data frame is reached. This may allow smaller input 112 Digital Audio Compression Standard 22 November 2010 buffers in some applications. Decoders that are able to store an entire frame may choose to process only crc2. These decoders would not begin processing block 0 of a frame until the entire frame is received. 7.10.2 Checking Bit Stream Consistency It is always possible that an AC-3 frame could have valid sync information and valid CRCs, but otherwise be undecodable. This condition may arise if a frame is corrupted such that the CRC word is nonetheless valid, or in the case of an encoder error (bug). One safeguard against this is to perform some error checking tests within the AC-3 decoder and bit stream parser. Despite its coding efficiency, there are some redundancies inherent in the AC-3 bit stream. If the AC-3 bit stream contains errors, a number of illegal syntactical constructions are likely to arise. Performing checks for these illegal constructs will detect a great many significant error conditions. Table 7.36 is a list of known bit stream error conditions. In some implementations it may be important that the decoder be able to benignly deal with these errors. Specifically, decoders may wish to ensure that these errors do not cause reserved memory to be overwritten with invalid data, and do not cause processing delays by looping with illegal loop counts. Invalid audio reproduction may be allowable, so long as system stability is preserved. Table 7.36 Known Bit Stream Error Conditions 1) (blknum == 0) && (cplstre == 0) ; 2) (cplinu == 1) && (fewer than two channels in coupling) ; 3) (cplinu == 1) && (cplbegf > (cplendf+2)) ; 4) (cplinu == 1) && ((blknum == 0) || (previous cplinu == 0)) && (chincpl[n] == 1) && (cplcoe[n] == 0) ; 5) (blknum == 0) && (acmod == 2) && (rematstr == 0) ; 6) (cplinu == 1) && ((blknum == 0) || (previous cplinu == 0)) && (cplexpstr == 0) ; 7) (cplinu == 1) && ((cplbegf != previous cplbegf) || (cplendf != previous cplendf)) && (cplexpstr == 0) ; 8) (blknum == 0) && (chexpstr[n] == 0) ; 9) (nchmant[n] != previous nchmant[n]) && (chexpstr[n] == 0) ; 10) (blknum == 0) && (lfeon == 1) && (lfeexpstr == 0) ; 113 Advanced Television Systems Committee, Inc. Document A/52:2010 Table 7.36 Known Bit Stream Error Conditions (Continued) 11) (chincpl[n] == 0) && (chbwcod[n] > 60) ; 12) (blknum == 0) && (baie == 0) ; 13) (blknum == 0) && (snroffste == 0) ; 14) (blknum == 0) && (cplinu == 1) && (cplleake == 0) ; 15) (cplinu == 1) && (expanded length of cpl delta bit allocation > 50) ; 16) expanded length of delta bit allocation[n] > 50 ; 17) compositely coded 5-level exponent value > 124 ; 18) compositely coded 3-level mantissa value > 26 ; 19) compositely coded 5-level mantissa value > 124 ; 20) compositely coded 11-level mantissa value > 120 ; 21) bit stream unpacking continues past the end of the frame ; 22) (cplinu == 1) && (acmod < 2) ; 23) (cplinu == 1) && ((cplbegf != previous cplbegf) || (cplendf != previous cplendf)) && (cplcoe[n] == 0) ; 24) (cplinu == 1) && (cplbndstrc != previous cplbndstrc) && (cplcoe[n] == 0) ; 25) (acmod == 2) && (number of rematrixing bands != previous number of rematrixing bands) && (rematstr == 0) ; 26) (cplinu == 1) && (previous cplinu == 0) && ((deltbaie == 0) || (cpldeltbae == 0)) ; 27) (cplinu == 1) && ((cplbegf != previous cplbegf) || (cplendf != previous cplendf)) && (previous cpl delta bit allocation active) && ((deltbaie == 0) || (cpldeltbae ==0)) ; 28) (nchmant[n] != previous nchmant[n]) && (previous delta bit allocation for channel n active) && ((deltbaie == 0) || (deltbae[n] == 0)) ; Note that some of these conditions (such as #17 through #20) can only be tested for at low- levels within the decoder software, resulting in a potentially significant MIPS impact. So long as these conditions do not affect system stability, they do not need to be specifically prevented. 114 Digital Audio Compression Standard 22 November 2010 8. ENCODING THE AC-3 BIT STREAM 8.1 Introduction This section provides some guidance on AC-3 encoding. Since AC-3 is specified by the syntax and decoder processing, the encoder is not precisely specified. The only normative requirement on the encoder is that the output elementary bit stream follow AC-3 syntax. Encoders of varying levels of sophistication may be produced. More sophisticated encoders may offer superior audio performance, and may make operation at lower bit-rates acceptable. Encoders are expected to improve over time. All decoders will benefit from encoder improvements. The encoder described in this section, while basic in operation, provides good performance. The description which follows indicates several avenues of potential improvement. A flow diagram of the encoding process is shown in Figure 8.1. 8.2 Summary of the Encoding Process 8.2.1 Input PCM 8.2.1.1 Input Word Length The AC-3 encoder accepts audio in the form of PCM words. The internal dynamic range of AC-3 allows input wordlengths of up to 24 bits to be useful. 8.2.1.2 Input Sample Rate The input sample rate must be locked to the output bit rate so that each AC-3 sync frame contains 1536 samples of audio per channel. If the input audio is available in a PCM format at a different sample rate than that required, sample rate conversion must be performed to conform the sample rate. 8.2.1.3 Input Filtering Individual input channels may be high-pass filtered. Removal of DC components of signals can allow more efficient coding since data rate is not used up encoding DC. However, there is the risk that signals which do not reach 100% PCM level before high-pass filtering will exceed 100% level after filtering, and thus be clipped. A typical encoder would high-pass filter the input signals with a single pole filter at 3 Hz. The lfe channel should be low-pass filtered at 120 Hz. A typical encoder would filter the lfe channel with an 8th order elliptic filter with a cutoff frequency of 120 Hz. 8.2.2 Transient Detection Transients are detected in the full-bandwidth channels in order to decide when to switch to short length audio blocks to improve pre-echo performance. High-pass filtered versions of the signals are examined for an increase in energy from one sub-block time-segment to the next. Sub-blocks are examined at different time scales. If a transient is detected in the second half of an audio block in a channel, that channel switches to a short block. A channel that is block-switched uses the D45 exponent strategy. The transient detector is used to determine when to switch from a long transform block (length 512), to the short block (length 256). It operates on 512 samples for every audio block. This is done in two passes, with each pass processing 256 samples. Transient detection is broken 115 Advanced Television Systems Committee, Inc. Document A/52:2010 Figure 8.1. Flow diagram of the encoding process. down into four steps: 1) high-pass filtering, 2) segmentation of the block into submultiples, 3) peak amplitude detection within each sub-block segment, and 4) threshold comparison. The transient detector outputs a flag blksw[n] for each full-bandwidth channel, which when set to “one” indicates the presence of a transient in the second half of the 512 length input block for the corresponding channel. 1. High-pass filtering: The high-pass filter is implemented as a cascaded biquad direct form I IIR filter with a cutoff of 8 kHz. 2. Block Segmentation: The block of 256 high-pass filtered samples are segmented into a hierarchical tree of levels in which level 1 represents the 256 length block, level 2 is two segments of length 128, and level 3 is four segments of length 64. 116 Digital Audio Compression Standard 22 November 2010 3. Peak Detection: The sample with the largest magnitude is identified for each segment on every level of the hierarchical tree. The peaks for a single level are found as follows: P[j][k] = max(x(n)) for n = (512 × (k-1) / 2^j), (512 × (k-1) / 2^j) + 1, ...(512 × k / 2^j) - 1 and k = 1, ..., 2^(j-1) ; Where: x(n) = the nth sample in the 256 length block j = 1, 2, 3 is the hierarchical level number k = the segment number within level j Note that P[j][0], (i.e., k = 0) is defined to be the peak of the last segment on level j of the tree calculated immediately prior to the current tree. For example, P[3][4] in the preceding tree is P[3][0] in the current tree. 4. Threshold Comparison: The first stage of the threshold comparator checks to see if there is significant signal level in the current block. This is done by comparing the overall peak value P[1][1] of the current block to a “silence threshold”. If P[1][1] is below this threshold then a long block is forced. The silence threshold value is 100/32768. The next stage of the comparator checks the relative peak levels of adjacent segments on each level of the hierarchical tree. If the peak ratio of any two adjacent segments on a particular level exceeds a pre-defined threshold for that level, then a flag is set to indicate the presence of a transient in the current 256 length block. The ratios are compared as follows: mag(P[j][k]) × T[j] > mag(P[j][(k-1)]) Where: T[j] is the pre-defined threshold for level j, defined as T[1] = .1 T[2] = .075 T[3] = .05 If this inequality is true for any two segment peaks on any level, then a transient is indicated for the first half of the 512 length input block. The second pass through this process determines the presence of transients in the second half of the 512 length input block. 8.2.3 Forward Transform 8.2.3.1 Windowing The audio block is multiplied by a window function to reduce transform boundary effects and to improve frequency selectivity in the filter bank. The values of the window function are included in Table 7.33. Note that the 256 coefficients given are used back-to-back to form a 512-point symmetrical window. 117 Advanced Television Systems Committee, Inc. Document A/52:2010 8.2.3.2 Time to Frequency Transformation Based on the block switch flags, each audio block is transformed into the frequency domain by performing one long N = 512 point transform, or two short N = 256 point transforms. Let x[n] represent the windowed input time sequence. The output frequency sequence, XD[k] is defined by N–1 –2 2π π X D [ k ] = ----- -  x [ n ] cos  ------ ( 2n + 1 ) ( 2k + 1 ) + -- ( 2k + 1 ) ( 1 + α ) for 0 ≤ k < N/2 - - N  4N 4  N=0 Where: α = –1 for the first short transform 0 for the long transform +1 for the second short transform 8.2.4 Coupling Strategy 8.2.4.1 Basic Encoder For a basic encoder, a static coupling strategy may be employed. Suitable coupling parameters are: cplbegf = 6 ; /* coupling starts at 10.2 kHz */ cplendf = 12 ; /* coupling channel ends at 20.3 kHz */ cplbndstrc = 0, 0, 1, 1, 0, 1, 1, 1; cplinu = 1; /* coupling always on */ /* all non-block switched channels are coupled */ for (ch=0; ch2 channels) 1 0 1 Reserved 1 1 0 Reserved 1 1 1 Reserved 1 0 0 0 X X X Complete Main (CM) 0 0 0 1 Music and Effects (ME) X 0 1 0 Visually Impaired (VI) X 0 1 1 Hearing Impaired (HI) 0 1 0 0 Dialogue (D) X 1 0 1 0 0 0 Commentary (C) 1 1 1 0 Emergency (E) 0 1 1 1 Voiceover (VO) 1 1 1 1 X X X Karaoke (mono and '1+1" prohibited) A5.4 STD Audio Buffer Size The main audio buffer size (BSn ) shall have a fixed value of 5696 bytes. Refer to ISO/IEC 13818-1 [1] for the derivation of (BSn ) for audio elementary streams. A6. PES CONSTRAINTS This section shall apply to both System A and System B. A6.1 Encoding In some applications, the audio decoder may be capable of simultaneously decoding two elementary streams containing different program elements, and then combining the program elements into a complete program. Most of the program elements are found in the main audio service. Another program element (such as a narration of the picture content intended for the visually impaired listener) may be found in the associated audio service. 133 Advanced Television Systems Committee, Inc. Document A/52:2010 In order to have the audio from the two elementary streams reproduced in exact sample synchronism, it is necessary for the original audio elementary stream encoders to have encoded the two audio program elements frame synchronously; i.e., if audio stream 1 has sample 0 of frame n taken at time t0, then audio stream 2 should also have frame n beginning with its sample 0 taken the identical time t0. If the encoding of multiple audio services is done frame and sample synchronous, and decoding is intended to be frame and sample synchronous, then the PES packets of these audio services shall contain identical values of PTS which refer to the audio access units intended for synchronous decoding. Audio services intended to be combined together for reproduction shall be encoded at an identical sample rate. A6.2 Decoding If audio access units from two audio services which are to be simultaneously decoded have identical values of PTS indicated in their corresponding PES headers, then the corresponding audio access units shall be presented to the audio decoder for simultaneous synchronous decoding. Synchronous decoding means that for corresponding audio frames (access units), corresponding audio samples are presented at the identical time. If the PTS values do not match (indicating that the audio encoding was not frame synchronous) then the audio frames (access units) of the main audio service may be presented to the audio decoder for decoding and presentation at the time indicated by the PTS. An associated service which is being simultaneously decoded may have its audio frames (access units), which are in closest time alignment (as indicated by the PTS) to those of the main service being decoded, presented to the audio decoder for simultaneous decoding. In this case the associated service may be reproduced out of sync by as much as 1/2 of a frame time. (This is typically satisfactory; a visually impaired narration does not require highly precise timing.) A6.3 Byte-Alignment This section applies to both System A and System B. The AC-3 elementary stream shall be byte- aligned within the MPEG-2 data stream. This means that the initial 8 bits of an AC-3 frame shall reside in a single byte which is carried by the MPEG-2 data stream. 134 A/52:2010, Annex B: Bibliography (Informative) The following documents contain information on the algorithm described in this standard, and may be useful to those who are using or attempting to understand this standard. In the case of conflicting information, the information contained in this standard should be considered correct. [1] Todd, C., et. al., “AC-3: Flexible Perceptual Coding for Audio Transmission and Storage”, AES 96th Convention, Preprint 3796, Audio Engineering Society, New York, NY, February 1994. [2] Fielder, L. D., M. A. Bosi, G. A. Davidson, M. F. Davis, C. Todd, and S. Vernon; “AC-2 and AC-3: Low-Complexity Transform-Based Audio Coding,” Collected Papers on Digital Audio Bit-Rate Reduction, Neil Gilchrist and Christer Grewin eds., pp. 54–72, Audio Engineering Society, New York, NY, 1996. [3] Davidson, G. A.; The Digital Signal Processing Handbook, V. K. Madisetti and D. B. Williams eds., pp. 41-1 – 41-21, CRC Press LLC, Boca Raton, FL, 1997. [4] Princen, J., and A. Bradley; “Analysis/synthesis filter bank design based on time domain aliasing cancellation,” IEEE Trans. Acoust. Speech and Signal Processing, vol. ASSP-34, pp. 1153–1161, IEEE, New York, NY, October 1986. [5] Davidson, G. A, L. D. Fielder, and B. D. Link; “Parametric Bit Allocation in Perceptual Audio Coder,” AES 97th Convention, Preprint 3921, Audio Engineering Society, New York, NY, November 1994. [6] Vernon, Steve; “Dolby Digital: Audio Coding for Digital Television and Storage Applications,” AES 17th International Conference: High-Quality Audio Coding, August 1999. [7] Vernon, Steve, Vlad Fruchter, and Sergio Kusevitzky; “A Single-Chip DSP Implementation of a High-Quality Low Bit-Rate Multichannel Audio Coder,” AES 95th Convention, Preprint 3775, Audio Engineering Society, New York, NY, September 1993. [8] Rao, R., and P. Yip; Discrete Cosine Transform, Academic Press, Boston, MA, pg. 11, 1990. [9] Cover, T. M., and J. A. Thomas; Elements of Information Theory, Wiley Series in Telecommunications, Wiley, New York, NY, pg. 13, 1991. [10] Gersho, A., and R. M. Gray; Vector Quantization and Signal Compression, Kluwer Academic Publisher, Boston, MA, pg. 309, 1992. [11] Truman, M. M., G. A. Davidson, A. Ubale, and L. D. Fielder; “Efficient Bit Allocation, Quantization, and Coding in an Audio Distribution System,” AES 107th Convention, Preprint 5068, Audio Engineering Society, New York, NY, August 1999. [12] Fielder, Louis D. and Grant A. Davidson; “Audio Coding Tools for Digital Television Distribution,” AES 108th Convention, Preprint 5104, Audio Engineering Society, New York, NY, January 2000. Page 135 Advanced Television Systems Committee, Inc. Document A/52:2010 [13] Crockett, B.; “High Quality Multi-Channel Time-Scaling and Pitch-Shifting using Auditory Scene Analysis,” AES 115th Convention, Preprint 5948, Audio Engineering Society, New York, NY, October 2003. [14] Crockett, B.; “Improved Transient Pre-Noise Performance of Low Bit Rate Audio Coders Using Time Scaling Synthesis,” AES 117th Convention, Audio Engineering Society, New York, NY, October 2004. [15] Fielder, L. D., R. L. Andersen, B. G. Crockett, G. A. Davidson, M. F. Davis, S. C. Turner, M. S. Vinton, and P. A. Williams; “Introduction to Dolby Digital Plus, an Enhancement to the Dolby Digital Coding System,” AES 117th Convention, Audio Engineering Society, New York, NY, October 2004. 136 A/52:2010, Annex C: AC-3 Karaoke Mode (Informative) C1. SCOPE This Annex contains specifications for how karaoke aware and karaoke capable AC-3 decoders should reproduce karaoke AC-3 bit streams. A minimum level of functionality is defined which allows a karaoke aware decoder to produce an appropriate 2/0 or 3/0 default output when presented with a karaoke mode AC-3 bit stream. An additional level of functionality is defined for the karaoke capable decoder so that the listener may optionally control the reproduction of the karaoke bit stream. C2. INTRODUCTION The AC-3 karaoke mode has been defined in order to allow the multi-channel AC-3 bit stream to convey audio channels designated as L, R (e.g., 2-channel stereo music), M (e.g., guide melody), and V1, V2 (e.g., one or two vocal tracks). This Annex does not specify the contents of L, R, M, V1, and V2, but does specify the behavior of AC-3 decoding equipment when receiving a karaoke bit stream containing these channels. An AC-3 decoder which is karaoke capable will allow the listener to optionally reproduce the V1 and V2 channels, and may allow the listener to adjust the relative levels (mixing balance) of the M, V1, and V2 channels. An AC-3 decoder which is karaoke aware will reproduce the L, R, and M channels, and will reproduce the V1 and V2 channels at a level indicated by the encoded bit stream. The 2-channel karaoke aware decoder will decode the karaoke bit stream using the Lo, Ro downmix. The L and R channels will be reproduced out of the left and right outputs, and the M channel will appear as a phantom center. The precise level of the M channel is determined by cmixlev which is under control of the program provider. The level of the V1 and V2 channels which will appear in the downmix is determined by surmixlev, which is under control of the program provider. A single V channel (V1 only) will appear as a phantom center. A pair of V channels (V1 and V2) will be reproduced with V1 in left output and V2 in right output. The 5-channel karaoke aware decoder will reproduce the L, R channels out of the left and right outputs, and the M channel out of the center output. A single V channel (V1 only) will be reproduced in the center channel output. A pair of V channels (V1 and V2) will be reproduced with V1 in left output and V2 in right output. The level of the V1 and V2 channels which will appear in the output is determined by surmixlev. The karaoke capable decoder gives some control of the reproduction to the listener. The V1, V2 channels may be selected for reproduction independent of the value of surmixlev in the bit stream. The decoder may optionally allow the reproduction level and location of the M, V1, and V2 channels to be adjusted by the listener. The detailed implementation of the flexible karaoke capable decoder is not specified; it is left up to the implementation as to the degree of adjustability to be offered to the listener. Page 137 Advanced Television Systems Committee, Inc. Document A/52:2010 C3. DETAILED SPECIFICATION C3.1 Karaoke Mode Indication AC-3 bit streams are indicated as karaoke type when bsmod = ‘111’ and acmod >= 0x2. C3.2 Karaoke Mode Channel Assignment The channel assignments for both the normal mode and the karaoke mode are shown in Table C3.1. Table C3.1 Channel Array Ordering acmod Audio Coding Normal Channel Assignment Karaoke Channel Assignment Mode (bsmod != ‘111’) (bsmod=‘111’) ‘010’ 2/0 L,R L,R ‘011’ 3/0 L,C,R L,M,R ‘100’ 2/1 L,R,S L,R,V1 ‘101’ 3/1 L,C,R,S L,M,R,V1 ‘110’ 2/2 L,R,Ls,Rs L,R,V1,V2 ‘111’ 3/2 L,C,R,Ls,Rs L,M,R,V1,V2 C3.3 Reproduction of Karaoke Mode Bit Streams This section contains the specifications which shall be met by decoders which are designated as karaoke aware or karaoke capable. The following general equations indicate how the AC-3 decoder’s output channels, Lk, Ck, Rk, are formed from the encoded channels L, M, R, V1, V2. Typically, the surround loudspeakers are not used when reproducing karaoke bit streams. Lk = L + a * V1 + b * V2 + c * M Ck = d * V1 + e * V2 + f * M Rk = R + g * V1 + h * V2 + i * M C3.3.1 Karaoke Aware Decoders The values of the coefficients a–i, which are used by karaoke aware decoders, are given in Table C3.2. Values are shown for both 2-channel (2/0) and multi-channel (3/0) reproduction. For each of these situations, a coefficient set is shown for the case of a single encoded V channel (V1 only) or two encoded V channels (V1, V2). The actual coefficients used must be scaled downwards so that arithmetic overflow does not occur if all channels contributing to an output channel happen to be at full scale. Monophonic reproduction would be obtained by summing the left and right output channels of the 2/0 reproduction. Any AC-3 decoder will produce the appropriate output if it is set to perform an Lo, Ro 2-channel downmix. 138 Digital Audio Compression Standard, Annex C 22 November 2010 Table C3.2 Coefficient Values for Karaoke Aware Decoders Coefficient 2/0 Reproduction 3/0 Reproduction 1 Vocal 2 Vocals 1 Vocal 2 Vocals a 0.7 * slev slev 0.0 slev b --- 0.0 --- 0.0 c clev clev 0.0 0.0 d --- --- slev 0.0 e --- --- --- 0.0 f --- --- 1.0 1.0 g 0.7 * slev 0.0 0.0 0.0 h --- slev --- slev i clev clev 0.0 0.0 C3.3.2 Karaoke Capable Decoders Karaoke capable decoders allow the user to choose to have the decoder reproduce none, one, or both of the V channels. The default coefficient values for the karaoke capable decoder are given in Table C3.2. When the listener selects to have none, one, or both of the V channels reproduced, the default coefficients are given in Table C3.3. Values are shown for both 2-channel (2/0) and multi-channel (3/0) reproduction, and for the cases of user selected reproduction of no V channel (None), one V channel (either V1 or V2), or both V channels (V1+V2). The M channel and a single V channel are reproduced out of the center output (phantom center in 2/0 reproduction), and a pair of V channels are reproduced out of the left (V1) and right (V2) outputs. The actual coefficients used must be scaled downwards so that arithmetic overflow does not occur if all channels contributing to an output happen to be at full scale. Table C3.3 Default Coefficient Values for Karaoke Capable Decoders Coefficient 2/0 Reproduction 3/0 Reproduction None V1 V2 V1+V2 None V1 V2 V1+V2 a 0.0 0.7 0.0 1.0 0.0 0.0 0.0 1.0 b 0.0 0.0 0.7 0.0 0.0 0.0 0.0 0.0 c clev clev clev clev 0.0 0.0 0.0 0.0 d --- --- --- --- 0.0 1.0 0.0 0.0 e --- --- --- --- 0.0 0.0 1.0 0.0 f --- --- --- --- 1.0 1.0 1.0 1.0 g 0.0 0.7 0.0 0.0 0.0 0.0 0.0 0.0 h 0.0 0.0 0.7 1.0 0.0 0.0 0.0 1.0 i clev clev clev clev 0.0 0.0 0.0 0.0 Additional flexibility may be offered optionally to the user of the karaoke decoder. For instance, the coefficients a, d, and g might be adjusted to allow the V1 channel to be reproduced in a different location and with a different level. Similarly the level and location of the V2 and M channels could be adjusted. The details of these additional optional user controls are not specified and are left up to the implementation. Also left up to the implementation is what use might be made of the Ls, Rs outputs of the 5-channel decoder, which would naturally reproduce the V1, V2 channels. 139 A/52:2010, Annex D: Alternate Bit Stream Syntax (Normative) D1. SCOPE This Annex contains specifications for an alternate bit stream syntax that may be implemented by some AC-3 encoders and interpreted by some AC-3 decoders. The new syntax redefines certain bit stream information (bsi) fields to carry new meanings. It is not necessary for decoders to be aware of this alternate syntax in order to properly reconstruct an audio soundfield; however those decoders that are aware of this syntax will be able to take advantage of the new system features described in this Annex. This alternate bit stream syntax is identified by setting the bsid to a value of 6. This Annex is Normative to the extent that when bsid is set to the value of 6, the alternate syntax elements shall have the meaning described in this Annex. Thus, this Annex may be considered Normative on encoders that set bsid to 6. This Annex is Informative for decoders. Interpretation and use of the new syntactical elements is optional for decoders. The new syntactical elements defined in this Annex are placed in the two 14-bit fields that are defined as timecod1 and timecod2 in the body of this document (these fields have never been applied for their originally anticipated purpose). D2. SPECIFICATION D2.1 Indication of Alternate Bit Stream Syntax An AC-3 bit stream shall have the alternate bit stream syntax described in this annex when the bit stream identification (bsid) field is set to 6. D2.2 Alternate Bit Stream Syntax Specification Table D2.1 shows the alternate bit stream syntax specification. Table D2.1 Bit Stream Information; Alternate Bit Stream Syntax Syntax Word Size bsi() { bsid 5 bsmod 3 acmod 3 if ((acmod & 0x1) && (acmod != 0x1)) /* if 3 front channels */ {cmixlev} 2 if (acmod & 0x4) /* if a surround channel exists */ {surmixlev} 2 if (acmod == 0x2) /* if in 2/0 mode */ {dsurmod} 2 lfeon 1 dialnorm 5 Page 141 Advanced Television Systems Committee, Inc. Document A/52:2010 Table D2.1 Bit Stream Information; Alternate Bit Stream Syntax (Continued) Syntax Word Size compre 1 if (compre) {compr} 8 langcode 1 if (langcode) {langcod} 8 audprodie 1 if (audprodie) { mixlevel 5 roomtyp 2 } if (acmod == 0) /* if 1+1 mode (dual mono, so some items need a second value) */ { dialnorm2 5 compr2e 1 if (compr2e) {compr2} 8 langcod2e 1 if (langcod2e) {langcod2} 8 audprodi2e 1 if (audprodi2e) { mixlevel2 5 roomtyp2 2 } } copyrightb 1 origbs 1 xbsi1e 1 if (xbsi1e) { dmixmod 2 ltrtcmixlev 3 ltrtsurmixlev 3 lorocmixlev 3 lorosurmixlev 3 } xbsi2e 1 if (xbsi2e) { dsurexmod 2 dheadphonmod 2 adconvtyp 1 xbsi2 8 142 Digital Audio Compression Standard, Annex D 22 November 2010 Table D2.1 Bit Stream Information; Alternate Bit Stream Syntax (Continued) Syntax Word Size encinfo 1 } addbsie 1 if (addbsie) { addbsil 6 addbsi(addbsil+1)×8 } } /* end of bsi */ D2.3 Description of Alternate Syntax Bit Stream Elements The following sections describe the meaning of the alternate syntax bit stream elements. Elements not specifically described retain the same meaning as specified in Section 5 of this document, except as noted in the alternate bit stream constraints section above. D2.3.1 xbsi1e: Extra Bitstream Information #1 Exists, 1 bit If this bit is a ‘1’, the following 14 bits contain extra bit stream information. D2.3.2 dmixmod: Preferred Stereo Downmix Mode, 2 bits This 2-bit code, as shown in Table D2.2, indicates the type of stereo downmix preferred by the mastering engineer. This information may be used by the AC-3 decoder to automatically configure the type of stereo downmix, but may also be overridden or ignored. If dmixmod is set to the reserved code, the decoder should still reproduce audio. The reserved code may be interpreted as “not indicated”. Table D2.2 Preferred Stereo Downmix Mode dmixmod Indication ‘00’ Not indicated ‘01’ Lt/Rt downmix preferred ‘10’ Lo/Ro downmix preferred ‘11’ Reserved Note: The meaning of this field is only defined as described if the audio coding mode is 3/0, 2/1, 3/1, 2/2 or 3/2. If the audio coding mode is 1+1, 1/0 or 2/0 then the meaning of this field is reserved. D2.3.3 ltrtcmixlev: Lt/Rt Center Mix Level, 3 bits This 3-bit code, shown in Table D2.3, indicates the nominal down mix level of the center channel with respect to the left and right channels in an Lt/Rt downmix. 143 Advanced Television Systems Committee, Inc. Document A/52:2010 Table D2.3 Lt/Rt Center Mix Level ltrtcmixlev clev ‘000’ 1.414 (+3.0 dB) ‘001’ 1.189 (+1.5 dB) ‘010’ 1.000 (0.0 dB) ‘011’ 0.841 (–1.5 dB) ‘100’ 0.707 (–3.0 dB) ‘101’ 0.595 (–4.5 dB) ‘110’ 0.500 (–6.0 dB) ‘111’ 0.000 (–inf dB) Note: The meaning of this field is only defined as described if the audio coding mode is 3/0, 3/1 or 3/2. If the audio coding mode is 1+1, 1/0, 2/0, 2/1 or 2/2 then the meaning of this field is reserved. D2.3.4 ltrtsurmixlev: Lt/Rt Surround Mix Level, 3 bits This 3-bit code, shown in Table D2.4, indicates the nominal down mix level of the surround channels with respect to the left and right channels in an Lt/Rt downmix. If one of the reserved values is received, the decoder should us a value of 0.841 for clev. Table D2.4 Lt/Rt Surround Mix Level ltrtsurmixlev slev ‘000’ reserved ‘001’ reserved ‘010’ reserved ‘011’ 0.841 (–1.5 dB) ‘100’ 0.707 (–3.0 dB) ‘101’ 0.595 (–4.5 dB) ‘110’ 0.500 (–6.0 dB) ‘111’ 0.000 (–inf dB) Note: The meaning of this field is only defined as described if the audio coding mode is 2/1, 3/1, 2/2 or 3/2. If the audio coding mode is 1+1, 1/0, 2/0 or 3/0 then the meaning of this field is reserved. D2.3.5 lorocmixlev: Lo/Ro Center Mix Level, 3 bits This 3-bit code, shown in Table D2.5, indicates the nominal down mix level of the center channel with respect to the left and right channels in an Lo/Ro downmix. 144 Digital Audio Compression Standard, Annex D 22 November 2010 Table D2.5 Lo/Ro Center Mix Level lorocmixlev clev ‘000’ 1.414 (+3.0 dB) ‘001’ 1.189 (+1.5 dB) ‘010’ 1.000 (0.0 dB) ‘011’ 0.841 (–1.5 dB) ‘100’ 0.707 (–3.0 dB) ‘101’ 0.595 (–4.5 dB) ‘110’ 0.500 (–6.0 dB) ‘111’ 0.000 (–inf dB) Note: The meaning of this field is only defined as described if the audio coding mode is 3/0, 3/1 or 3/2. If the audio coding mode is 1+1, 1/0, 2/0, 2/1 or 2/2 then the meaning of this field is reserved. D2.3.6 lorosurmixlev: Lo/Ro Surround Mix Level, 3 bits This 3-bit code, shown in Table D2.6, indicates the nominal down mix level of the surround channels with respect to the left and right channels in an Lo/Ro downmix. If one of the reserved values is received, the decoder should use a value of 0.841 for slev. Table D2.6 Lo/Ro Surround Mix Level lorosurmixlev slev ‘000’ reserved ‘001’ reserved ‘010’ reserved ‘011’ 0.841 (–1.5 dB) ‘100’ 0.707 (–3.0 dB) ‘101’ 0.595 (–4.5 dB) ‘110’ 0.500 (–6.0 dB) ‘111’ 0.000 (–inf dB) Note: The meaning of this field is only defined as described if the audio coding mode is 2/1, 3/1, 2/2 or 3/2. If the audio coding mode is 1+1, 1/0, 2/0 or 3/0 then the meaning of this field is reserved. D2.3.7 xbsi2e: Extra Bit Stream Information #2 Exists, 1 bit If this bit is a ‘1’, the following 14 bits contain extra bit stream information. D2.3.8 dsurexmod: Dolby Surround EX Mode, 2 bits This 2-bit code, as shown in Table D2.7, indicates whether or not the program has been encoded in Dolby Surround EX. This information is not used by the AC-3 decoder, but may be used by other portions of the audio reproduction equipment. If dsurexmod is set to the reserved code, the decoder should still reproduce audio. The reserved code may be interpreted as “not indicated”. 145 Advanced Television Systems Committee, Inc. Document A/52:2010 Table D2.7 Dolby Surround EX Mode dsurexmod Indication ‘00’ Not indicated ‘01’ Not Dolby Surround EX encoded ‘10’ Dolby Surround EX encoded ‘11’ Reserved Note: The meaning of this field is only defined as described if the audio coding mode is 2/2 or 3/2. If the audio coding mode is 1+1, 1/0, 2/0, 3/0, 2/1 or 3/1 then the meaning of this field is reserved. D2.3.9 dheadphonmod: Dolby Headphone Mode, 2 bits This 2-bit code, as shown in Table D2.8, indicates whether or not the program has been Dolby Headphone-encoded. This information is not used by the AC-3 decoder, but may be used by other portions of the audio reproduction equipment. If dheadphonmod is set to the reserved code, the decoder should still reproduce audio. The reserved code may be interpreted as “not indicated”. Table D2.8 Dolby Headphone Mode dheadphonmod Indication ‘00’ Not indicated ‘01’ Not Dolby Headphone encoded ‘10’ Dolby Headphone encoded ‘11’ Reserved Note: The meaning of this field is only defined as described if the audio coding mode is 2/0. If the audio coding mode is 1+1, 1/0, 3/0, 2/1, 3/1, 2/2 or 3/2 then the meaning of this field is reserved. D2.3.10 adconvtyp: A/D Converter Type, 1 bit This 1-bit code, as shown in Table D2.9, indicates the type of A/D converter technology used to capture the PCM audio. This information is not used by the AC-3 decoder, but may be used by other portions of the audio reproduction equipment. If the type of A/D converter used is not known, the “standard” setting should be chosen. Table D2.9 A/D Converter Type Adconvtyp Indication ‘0’ Standard ‘1’ HDCD D2.3.11 xbsi2: Extra Bit Stream Information, 8 bits This field is reserved for future assignment. Encoders shall set these bits to all 0’s. D2.3.12 encinfo: Encoder Information, 1 bit This field is reserved for use by the encoder, and is not used by the decoder. 146 Digital Audio Compression Standard, Annex D 22 November 2010 D3. DECODER PROCESSING There are two types of decoders: those that recognize the alternate syntax (compliant decoders), and those that do not (legacy decoders). This section specifies how each type of decoder will process bit streams that use the alternate bit stream syntax. Implementation of compliant decoding is optional. D3.1 Compliant Decoder Processing D3.1.1 Two-Channel Downmix Selection In the case of a two-channel downmix, compliant decoders should allow the end user to specify which two-channel downmix is chosen. Three separate options should be allowed: Lt/Rt downmix, Lo/Ro downmix, or automatic selection of either Lt/Rt or Lo/Ro based on the preferred downmix mode parameter dmixmod. D3.1.2 Two-Channel Downmix Processing Once a particular two-channel downmix has been selected, compliant decoders should use the new center mix level and surround mix level parameters associated with the selected downmix type (assuming they are included in the bit stream). If Lt/Rt downmix is selected, compliant decoders should use the ltrtcmixlev and ltrtsurmixlev parameters (if included). If Lo/Ro downmix is selected, compliant decoders should use the lorocmixlev and lorosurmixlev parameters (if included). If these parameters are not included in the bit stream, then downmixing should be performed as defined in the original specification. D3.1.3 Informational Parameter Processing Compliant decoders should provide a means for informational parameters (e.g., dsurexmod, dheadphonmod, etc.) to be accessed by external system components. Note that these parameters do not otherwise affect decoder processing. D3.2 Legacy Decoder Processing Legacy decoders do not recognize the alternate bit stream syntax, but rather interpret these bit fields according to their original definitions in the initial version of this document. The extra bit stream information words (xbsi1e, xbsi2e, dmixmod, etc.) are interpreted as time code words (timecod1e, timecod1, timecod2e, and timecod2). As described in the initial version of this document, the time code words do not affect the decoding process in legacy decoders. As a result, the alternate bit stream syntax can be safely decoded without causing incorrect decoder processing. However, legacy decoders will not be able to take advantage of new functionality provided by the alternate syntax. D4. ENCODER PROCESSING This section describes processing steps and requirements associated with encoders that create bits streams according to the alternate bit stream syntax. 147 Advanced Television Systems Committee, Inc. Document A/52:2010 D4.1 Encoder Processing Steps D4.1.1 Dynamic Range Overload Protection Processing If the alternate bit stream syntax is used, the dynamic range overload protection function within the encoder must account for potential overload in either legacy or compliant decoders, using any downmix mode. No assumption should be made that compliant decoders will necessarily use the preferred downmix mode. D4.2 Encoder Requirements D4.2.1 Legacy Decoder Support In order to support legacy decoder operations, it is necessary to continue to specify valid values for bit stream information parameters that are made obsolete by the alternate bit stream syntax. For example, the new ltrtcmixlev, ltrtsurmixlev, lorocmixlev, and lorosurmixlev fields (if included in the alternate bit stream) override the functionality of the previously defined cmixlev and surmixlev fields. Nonetheless, alternate bit stream syntax encoders must continue to specify valid values for the cmixlev and surmixlev fields. D4.2.2 Original Bit Stream Syntax Support Encoding equipment that is capable of creating bit streams according to the alternate bit stream syntax must also provide an option that allows for creation of bit streams according to this document, not including this Annex or Annex E. 148 A/52:2010, Annex E: Enhanced AC-3 Bit Stream Syntax (Normative) E1. SCOPE This Annex defines the bit stream syntax that shall be used by Enhanced AC-3 bit streams, and a reference decoding process. Enhanced AC-3 bit streams are similar in nature to standard AC-3 bit streams, but are not backwards compatible (i.e., they are not decodable by standard AC-3 decoders). This Annex outlines the differences between the stream types, and specifies the reference decoding process for Enhanced AC-3 bit streams. This Annex is normative in applications that specify the use of Enhanced AC-3. Encoders shall construct bit streams for decoding using the decoding process specified in this Annex. E2. SPECIFICATION E2.1 Indication of Enhanced AC-3 Bit Stream Syntax An AC-3 bit stream is indicated as using the Enhanced AC-3 bit stream syntax described in this Annex when the bit stream identification (bsid) field is set to 16. E2.2 Syntax Specification A continuous audio bit stream consists of a sequence of synchronization frames: Syntax bit stream() { while(true) { syncframe() ; } } /* end of bit stream */ The syncframe consists of the syncinfo, bsi and audfrm fields, up to 6 coded audblk fields, the auxdata field, and the errorcheck field. Page 149 Advanced Television Systems Committee, Inc. Document A/52:2010 Syntax syncframe() { syncinfo() ; bsi() ; audfrm() ; for (blk = 0; blk < number_of_blocks_per_syncframe; blk++) { audblk() ; } auxdata() ; errorcheck() ; } /* end of syncframe */ Each of the bit stream elements, and their length, are itemized in the following tables. Note that all bit stream elements arrive most significant bit first, or left bit first, in time. E2.2.1 syncinfo: Synchronization Information Table E2.1 syncinfo Syntax and Word Size Syntax Word Size syncinfo() { syncword 16 } /* end of syncinfo */ E2.2.2 bsi: Bit Stream Information Table E2.2 bsi Syntax and Word Size Syntax Word Size bsi() { strmtyp 2 substreamid 3 frmsiz 11 fscod 2 if (fscod == 0x3) { fscod2 2 numblkscod = 0x3 /* six blocks per frame */ } else { numblkscod 2 } 150 Digital Audio Compression Standard, Annex E 22 November 2010 Table E2.2 bsi Syntax and Word Size (Continued) Syntax Word Size acmod 3 lfeon 1 bsid 5 dialnorm 5 compre 1 if (compre) {compr} 8 if (acmod == 0x0) /* if 1+1 mode (dual mono, so some items need a second value) */ { dialnorm2 5 compr2e 1 if(compr2e) {compr2} 8 } if (strmtyp == 0x1) /* if dependent stream */ { chanmape 1 if(chanmape) {chanmap} 16 } mixmdate 1 if (mixmdate) /* Mixing metadata */ { if (acmod > 0x2) /* if more than 2 channels */ {dmixmod} 2 if ((acmod & 0x1) && (acmod > 0x2)) /* if three front channels exist */ { ltrtcmixlev 3 lorocmixlev 3 } if (acmod & 0x4) /* if a surround channel exists */ { ltrtsurmixlev 3 lorosurmixlev 3 } if (lfeon) /* if the LFE channel exists */ { lfemixlevcode 1 if (lfemixlevcode) {lfemixlevcod} 5 } if (strmtyp == 0x0) /* if independent stream */ { pgmscle 1 if (pgmscle) {pgmscl} 6 if (acmod == 0x0) /* if 1+1 mode (dual mono, so some items need a second value) */ { 151 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E2.2 bsi Syntax and Word Size (Continued) Syntax Word Size pgmscl2e 1 if (pgmscl2e) {pgmscl2} 6 } extpgmscle 1 if (extpgmscle) {extpgmscl} 6 mixdef 2 if (mixdef == 0x1) /* mixing option 2 */ {mixdata} 5 else if (mixdef == 0x2) /* mixing option 3 */ {mixdata} 12 else if (mixdef == 0x3) /* mixing option 4 */ { mixdeflen 5 mixdata 8*(mixdeflen+2) } if (acmod < 0x2) /* if mono or dual mono source */ { paninfoe 1 if (paninfoe) {paninfo} 14 if (acmod == 0x0) /* if 1+1 mode (dual mono, so some items need a second value) */ { paninfo2e 1 if (paninfo2e) {paninfo2} 14 } } frmmixcfginfoe 1 if (frmmixcfginfoe) /* mixing configuration information */ { if (numblkscod == 0x0) {blkmixcfginfo[0]} 5 else { for (blk = 0; blk < number_of_blocks_per_syncframe; blk++) { blkmixcfginfoe 1 if (blkmixcfginfoe){blkmixcfginfo[blk]} 5 } } } } } infomdate 1 if (infomdate) /* Informational metadata */ { bsmod 3 152 Digital Audio Compression Standard, Annex E 22 November 2010 Table E2.2 bsi Syntax and Word Size (Continued) Syntax Word Size copyrightb 1 origbs 1 if (acmod == 0x2) /* if in 2/0 mode */ { dsurmod 2 dheadphonmod 2 } if (acmod >= 0x6) /* if both surround channels exist */ {dsurexmod} 2 audprodie 1 if (audprodie) { mixlevel 5 roomtyp 2 adconvtyp 1 } if (acmod == 0x0) /* if 1+1 mode (dual mono, so some items need a second value) */ { audprodi2e 1 if (audprodi2e) { mixlevel2 5 roomtyp2 2 adconvtyp2 1 } } if (fscod < 0x3) /* if not half sample rate */ {sourcefscod} 1 } if ( (strmtyp == 0x0) && (numblkscod != 0x3) ) {convsync} 1 if (strmtyp == 0x2) /* if bit stream converted from AC-3 */ { if (numblkscod == 0x3) /* 6 blocks per frame */ {blkid = 1} else {blkid} 1 if (blkid) {frmsizecod} 6 } addbsie 1 if (addbsie) { addbsil 6 addbsi (addbsil+1)×8 } } /* end of bsi */ 153 Advanced Television Systems Committee, Inc. Document A/52:2010 E2.2.3 audfrm: Audio Frame Table E2.3 audfrm Syntax and Word Size Syntax Word Size audfrm() { /* These fields for audio frame exist flags and strategy data */ if (numblkscod == 0x3) /* six blocks per frame */ { expstre 1 ahte 1 } else { expstre = 1 ahte = 0 } snroffststr 2 transproce 1 blkswe 1 dithflage 1 bamode 1 frmfgaincode 1 dbaflde 1 skipflde 1 spxattene 1 /* These fields for coupling data */ if (acmod > 0x1) { cplstre[0] = 1 cplinu[0] 1 for (blk = 1; blk < number_of_blocks_per_sync_frame; blk++) { cplstre[blk] 1 if (cplstre[blk] == 1) {cplinu[blk]} 1 else {cplinu[blk] = cplinu[blk-1]} } } else { for (blk = 0; blk < number_of_blocks_per_sync_frame; blk++) {cplinu[blk] = 0} } /* These fields for exponent strategy data */ if (expstre) 154 Digital Audio Compression Standard, Annex E 22 November 2010 Table E2.3 audfrm Syntax and Word Size Syntax Word Size { for (blk = 0; blk < number_of_blocks_per_sync_frame; blk++) { if (cplinu[blk] == 1) {cplexpstr[blk]} 2 for (ch = 0; ch < nfchans; ch++) {chexpstr[blk][ch]} 2 } } else { ncplblks = 0 for (blk = 0; blk < number_of_blocks_per_sync_frame; blk++) {ncplblks += cplinu[blk]} if ( (acmod > 0x1) && (ncplblks > 0) ) {frmcplexpstr} 5 for (ch = 0; ch < nfchans; ch++) {frmchexpstr[ch]} 5 /* cplexpstr[blk] and chexpstr[blk][ch] derived from table lookups – see Table E2.14 */ } if (lfeon) { for (blk = 0; blk < number_of_blocks_per_sync_frame; blk++) {lfeexpstr[blk]} 1 } /* These fields for converter exponent strategy data */ if (strmtyp == 0x0) { if (numblkscod != 0x3) {convexpstre} 1 else {convexpstre = 1} if (convexpstre == 1) { for (ch = 0; ch < nfchans; ch++) {convexpstr[ch]} 5 } } /* These fields for AHT data */ if (ahte) { /* coupling can use AHT only when coupling in use for all blocks */ /* ncplregs derived from cplstre and cplexpstr – see Section E3.3.2 */ if ( (ncplblks == 6) && (ncplregs ==1) ) {cplahtinu} 1 else {cplahtinu = 0} for (ch = 0; ch < nfchans; ch++) { /* nchregs derived from chexpstr – see Section E3.3.2 */ if (nchregs[ch] == 1) {chahtinu[ch]} 1 else {chahtinu[ch] = 0} } 155 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E2.3 audfrm Syntax and Word Size Syntax Word Size if(lfeon) { /* nlferegs derived from lfeexpstr – see Section E3.3.2 */ if (nlferegs == 1) {lfeahtinu} 1 else {lfeahtinu = 0} } } /* These fields for audio frame SNR offset data */ if (snroffststr == 0x0) { frmcsnroffst 6 frmfsnroffst 4 } /* These fields for audio frame transient pre-noise processing data */ if (transproce) { for (ch = 0; ch < nfchans; ch++) { chintransproc[ch] 1 if (chintransproc[ch]) { transprocloc[ch] 10 transproclen[ch] 8 } } } /* These fields for spectral extension attenuation data */ if (spxattene) { for (ch = 0; ch < nfchans; ch++) { chinspxatten[ch] 1 if (chinspxatten[ch]) { spxattencod[ch] 5 } } } /* These fields for block start information */ if (numblkscod != 0x0) {blkstrtinfoe} 1 else {blkstrtinfoe = 0} if (blkstrtinfoe) 156 Digital Audio Compression Standard, Annex E 22 November 2010 Table E2.3 audfrm Syntax and Word Size Syntax Word Size { /* nblkstrtbits determined from frmsiz (see Section E2.3.2.27) */ blkstrtinfo nblkstrtbits } /* These fields for syntax state initialization */ for (ch = 0; ch < nfchans; ch++) { firstspxcos[ch] = 1 firstcplcos[ch] = 1 } firstcplleak = 1 } /* end of audfrm */ E2.2.4 audblk: Audio Block Table E2.4 audblk Syntax and Word Size Syntax Word Size audblk() { /* These fields for block switch and dither flags */ if (blkswe) { for (ch = 0; ch < nfchans; ch++) {blksw[ch]} 1 } else { for (ch = 0; ch < nfchans; ch++) {blksw[ch] = 0} } if (dithflage) { for (ch = 0; ch < nfchans; ch++) {dithflag[ch]} 1 } else { for (ch = 0; ch < nfchans; ch++) {dithflag[ch] = 1} /* dither on */ } /* These fields for dynamic range control */ dynrnge 1 if (dynrnge) {dynrng} 8 if (acmod == 0x0) /* if 1+1 mode */ { dynrng2e 1 157 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E2.4 audblk Syntax and Word Size (Continued) Syntax Word Size if(dynrng2e) {dynrng2} 8 } /* These fields for spectral extension strategy information */ if (blk == 0) {spxstre = 1} else {spxstre} 1 if (spxstre) { spxinu 1 if (spxinu) { if (acmod == 0x1) { chinspx[0] = 1 } else { for (ch = 0; ch < nfchans; ch++) {chinspx[ch]} 1 } spxstrtf 2 spxbegf 3 spxendf 3 if (spxbegf < 6) {spxbegf += 2} else {spxbegf = spxbegf * 2 – 3} if (spxendf < 3) {spxendf += 5} else {spxendf = spxendf * 2 + 3} spxbndstrce 1 if (spxbndstrce) { for (bnd = spxbegf+1; bnd < spxendf; bnd++) {spxbndstrc[bnd]} 1 } } else /* !spxinu */ { for (ch = 0; ch < nfchans; ch++) { chinspx[ch] = 0 firstspxcos[ch] = 1 } } } /* These fields for spectral extension coordinates */ if (spxinu) 158 Digital Audio Compression Standard, Annex E 22 November 2010 Table E2.4 audblk Syntax and Word Size (Continued) Syntax Word Size { for (ch = 0; ch < nfchans; ch++) { if (chinspx[ch]) { if (firstspxcos[ch]) { spxcoe[ch] = 1 firstspxcos[ch] = 0 } else /* !firstspxcos[ch] */ {spxcoe[ch]} 1 if (spxcoe[ch]) { spxblnd[ch] 5 mstrspxco[ch] 2 /* nspxbnds determined from spxbegf, spxendf, and spxbndstrc[ ] */ for (bnd = 0; bnd < nspxbnds; bnd++) { spxcoexp[ch][bnd] 4 spxcomant[ch][bnd] 2 } } } else /* !chinspx[ch] */ { firstspxcos[ch] = 1 } } } /* These fields for coupling strategy and enhanced coupling strategy information */ if (cplstre[blk]) { if (cplinu[blk]) { ecplinu 1 if (acmod == 0x2) { chincpl[0] = 1 chincpl[1] = 1 } else { 159 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E2.4 audblk Syntax and Word Size (Continued) Syntax Word Size for (ch = 0; ch < nfchans; ch++) {chincpl[ch]} 1 } if (ecplinu == 0) /* standard coupling in use */ { if (acmod == 0x2) {phsflginu} /* if in 2/0 mode */ 1 cplbegf 4 if (spxinu == 0) /* if SPX not in use */ { cplendf 4 cplendf = cplendf + 3 } else /* SPX in use */ { cplendf = spxbegf - 1 } cplbndstrce 1 if (cplbndstrce) { for (bnd = cplbegf+1; bnd < cplendf; bnd++) {cplbndstrc[bnd]} 1 } } else /* enhanced coupling in use */ { ecplbegf 4 if (ecplbegf < 3) {ecpl_start_subbnd = ecplbegf * 2} else if(ecplbegf < 13) {ecpl_start_subbnd = ecplbegf + 2} else {ecpl_start_subbnd = ecplbegf * 2 - 10} if (spxinu == 0) /* if SPX not in use */ { ecplendf 4 ecpl_end_subbnd = ecplendf + 7 } else /* SPX in use */ { if (spxbegf < 6) {ecpl_end_subbnd = spxbegf + 5} else {ecpl_end_subbnd = spxbegf * 2} } ecplbndstrce 1 if (ecplbndstrce) { for (sbnd = max(9, ecpl_start_subbnd+1); sbnd < ecpl_end_subbnd; sbnd++) { 160 Digital Audio Compression Standard, Annex E 22 November 2010 Table E2.4 audblk Syntax and Word Size (Continued) Syntax Word Size ecplbndstrc[sbnd] 1 } } } /* ecplinu[blk] */ } else /* !cplinu[blk] */ { for (ch = 0; ch < nfchans; ch++) { chincpl[ch] = 0 firstcplcos[ch] = 1 } firstcplleak = 1 phsflginu = 0 ecplinu = 0; } } /* cplstre[blk] */ /* These fields for coupling coordinates */ if (cplinu[blk]) { if (ecplinu == 0) /* standard coupling in use */ { for (ch = 0; ch < nfchans; ch++) { if (chincpl[ch]) { if (firstcplcos[ch]) { cplcoe[ch] = 1 firstcplcos[ch] = 0 } else /* !firstcplcos[ch] */ {cplcoe[ch]} 1 if (cplcoe[ch]) { mstrcplco[ch] 2 /* ncplbnd derived from cplbegf, cplendf, and cplbndstrc */ for (bnd = 0; bnd < ncplbnd; bnd++) { cplcoexp[ch][bnd] 4 cplcomant[ch][bnd] 4 } } /* cplcoe[ch] */ 161 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E2.4 audblk Syntax and Word Size (Continued) Syntax Word Size } else /* ! chincpl[ch] */ { firstcplcos[ch] = 1 } } /* ch */ if ((acmod == 0x2) && phsflginu && (cplcoe[0] || cplcoe[1])) { for (bnd = 0; bnd < ncplbnd; bnd++) {phsflg[bnd]} 1 } } else /* enhanced coupling in use */ { firstchincpl = -1 ecplangleintrp 1 for (ch = 0; ch < nfchans; ch++) { if (chincpl[ch]) { if (firstchincpl == -1) {firstchincpl = ch} if (firstcplcos[ch]) { ecplparam1e[ch] = 1 if (ch > firstchincpl) {ecplparam2e[ch] = 1} else {ecplparam2e[ch] = 0} firstcplcos[ch] = 0 } else /* !firstcplcos[ch] */ { ecplparam1e[ch] 1 if (ch > firstchincpl) {ecplparam2e[ch]} 1 else {ecplparam2e[ch] = 0} } if (ecplparam1e[ch]) { /* necplbnd derived from ecpl_start_subbnd, ecpl_end_subbnd, and ecplbndstrc */ for (bnd = 0; bnd < necplbnd; bnd++) {ecplamp[ch][bnd]} 5 } if (ecplparam2e[ch]) { /* necplbnd derived from ecpl_start_subbnd, ecpl_end_subbnd, and ecplbndstrc */ for (bnd = 0; bnd < necplbnd; bnd++) 162 Digital Audio Compression Standard, Annex E 22 November 2010 Table E2.4 audblk Syntax and Word Size (Continued) Syntax Word Size { ecplangle[ch][bnd] 6 ecplchaos[ch][bnd] 3 } } if (ch > firstchincpl) {ecpltrans[ch]} 1 } else /* !chincpl[ch] */ { firstcplcos[ch] = 1 } } /* ch */ } /* ecplinu[blk] */ } /* cplinu[blk] */ /* These fields for rematrixing operation in the 2/0 mode */ if (acmod == 0x2) /* if in 2/0 mode */ { if (blk == 0) {rematstr = 1} else {rematstr} 1 if (rematstr) { /* nrematbnds determined from cplinu, ecplinu, spxinu, cplbegf, ecplbegf and spxbegf */ for (bnd = 0; bnd < nrematbnds; bnd++) {rematflg[bnd]} 1 } } /* This field for channel bandwidth code */ for (ch = 0; ch < nfchans; ch++) { if (chexpstr[blk][ch] != reuse) { if ((!chincpl[ch]) && (!chinspx[ch])) {chbwcod[ch]} 6 } } /* These fields for exponents */ if (cplinu[blk]) /* exponents for the coupling channel */ { if (cplexpstr[blk] != reuse) { cplabsexp 4 /* ncplgrps derived from cplbegf, ecplbegf, cplendf, ecplendf, and cplexpstr */ for (grp = 0; grp< ncplgrps; grp++) {cplexps[grp]} 7 } 163 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E2.4 audblk Syntax and Word Size (Continued) Syntax Word Size } for (ch = 0; ch < nfchans; ch++) /* exponents for full bandwidth channels */ { if (chexpstr[blk][ch] != reuse) { exps[ch][0] 4 /* nchgrps derived from chexpstr[ch], and cplbegf or chbwcod[ch] */ for (grp = 1; grp <= nchgrps[ch]; grp++) {exps[ch][grp]} 7 gainrng[ch] 2 } } if (lfeon) /* exponents for the low frequency effects channel */ { if (lfeexpstr[blk] != reuse) { lfeexps[0] 4 nlfegrps = 2 for(grp = 1; grp <= nlfegrps; grp++) {lfeexps[grp]} 7 } } /* These fields for bit-allocation parametric information */ if (bamode) { baie 1 if (baie) { sdcycod 2 fdcycod 2 sgaincod 2 dbpbcod 2 floorcod 3 } } else { sdcycod = 0x2 fdcycod = 0x1 sgaincod = 0x1 dbpbcod = 0x2 floorcod = 0x7 } if (snroffststr == 0x0) 164 Digital Audio Compression Standard, Annex E 22 November 2010 Table E2.4 audblk Syntax and Word Size (Continued) Syntax Word Size { if (cplinu[blk]) {cplfsnroffst = frmfsnroffst} for (ch = 0; ch < nfchans; ch++) {fsnroffst[ch] = frmfsnroffst} if (lfeon) {lfefsnroffst = frmfsnroffst} } else { if (blk == 0) {snroffste = 1} else {snroffste} 1 if (snroffste) { csnroffst 6 if (snroffststr == 0x1) { blkfsnroffst 4 cplfsnroffst = blkfsnroffst for (ch=0; ch < nfchans; ch++) {fsnroffst[ch] = blkfsnroffst} lfefsnroffst = blkfsnroffst } else if(snroffststr == 0x2) { if (cplinu[blk]) {cplfsnroffst} 4 for (ch = 0; ch < nfchans; ch++) {fsnroffst[ch]} 4 if (lfeon) {lfefsnroffst} 4 } } } if (frmfgaincode) {fgaincode} 1 else {fgaincode = 0} if (fgaincode) { if (cplinu[blk]) {cplfgaincod} 3 for (ch = 0; ch < nfchans; ch++) {fgaincod[ch]} 3 if (lfeon) {lfefgaincod} 3 } else { if (cplinu[blk]) {cplfgaincod = 0x4} for (ch= 0; ch < nfchans; ch++) {fgaincod[ch] = 0x4} if (lfeon) {lfefgaincod = 0x4} } if (strmtyp == 0x0) 165 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E2.4 audblk Syntax and Word Size (Continued) Syntax Word Size { convsnroffste 1 if (convsnroffste) {convsnroffst} 10 } if (cplinu[blk]) { if (firstcplleak) { cplleake = 1 firstcplleak = 0 } else /* !firstcplleak */ { cplleake 1 } if (cplleake) { cplfleak 3 cplsleak 3 } } /* These fields for delta bit allocation information */ if (dbaflde) { deltbaie 1 if (deltbaie) { if (cplinu[blk]) {cpldeltbae} 2 for (ch = 0; ch < nfchans; ch++) {deltbae[ch]} 2 if (cplinu[blk]) { if (cpldeltbae==new info follows) { cpldeltnseg 3 for (seg = 0; seg <= cpldeltnseg; seg++) { cpldeltoffst[seg] 5 cpldeltlen[seg] 4 cpldeltba[seg] 3 } } } 166 Digital Audio Compression Standard, Annex E 22 November 2010 Table E2.4 audblk Syntax and Word Size (Continued) Syntax Word Size for (ch = 0; ch < nfchans; ch++) { if (deltbae[ch]==new info follows) { deltnseg[ch] 3 for (seg = 0; seg <= deltnseg[ch]; seg++) { deltoffst[ch][seg] 5 deltlen[ch][seg] 4 deltba[ch][seg] 3 } } } } /* if (deltbaie) */ }/* if (dbaflde) */ /* These fields for inclusion of unused dummy data */ if (skipflde) { skiple 1 if (skiple) { skipl 9 skipfld skipl × 8 } } /* These fields for quantized mantissa values */ got_cplchan = 0 for (ch = 0; ch < nfchans; ch++) { if (chahtinu[ch] == 0) { for (bin = 0; bin < nchmant[ch]; bin++) {chmant[ch][bin]} (0–16) } else if(chahtinu[ch] == 1) { chgaqmod[ch] 2 if ( (chgaqmod[ch] > 0x0) && (chgaqmod[ch] < 0x3) ) { for (n = 0; n < chgaqsections[ch]; n++) {chgaqgain[ch][n]} 1 } else if(chgaqmod[ch] == 0x3) { 167 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E2.4 audblk Syntax and Word Size (Continued) Syntax Word Size for (n = 0; n < chgaqsections[ch]; n++) {chgaqgain[ch][n]} 5 } for (bin = 0; bin < nchmant[ch]; bin++) { if (chgaqbin[ch][bin]) { for (n = 0; n < 6; n++) {pre_chmant[n][ch][bin]} (0–16) } else {pre_chmant[0][ch][bin]} (0–9) } chahtinu[ch] = -1 /* AHT info for this frame has been read – do not read again */ } if (cplinu[blk] && chincpl[ch] && !got_cplchan) { if (cplahtinu == 0) { for (bin = 0; bin < ncplmant; bin++) {cplmant[bin]} (0–16) got_cplchan = 1 } else if(cplahtinu == 1) { cplgaqmod 2 if ( (cplgaqmod > 0x0) && (cplgaqmod < 0x3) ) { for (n = 0; n < cplgaqsections; n++) {cplgaqgain[n]} 1 } else if (cplgaqmod == 0x3) { for (n = 0; n < cplgaqsections; n++) {cplgaqgain[n]} 5 } for (bin = 0; bin < ncplmant; bin++) { if (cplgaqbin[bin]) { for (n = 0; n < 6; n++) {pre_cplmant[n][bin]} (0–16) } else {pre_cplmant[0][bin]} (0–9) } got_cplchan = 1 cplahtinu = -1 /* AHT info for this frame has been read – do not read again */ } else {got_cplchan = 1} 168 Digital Audio Compression Standard, Annex E 22 November 2010 Table E2.4 audblk Syntax and Word Size (Continued) Syntax Word Size } } if (lfeon) /* mantissas of low frequency effects channel */ { if (lfeahtinu == 0) { for (bin = 0; bin < nlfemant; bin++) {lfemant[bin]} (0–16) } else if (lfeahtinu == 1) { lfegaqmod 2 if ( (lfegaqmod > 0x0) && (lfegaqmod < 0x3) ) { for (n = 0; n < lfegaqsections; n++) {lfegaqgain[n]} 1 } else if (lfegaqmod == 0x3) { for (n = 0; n < lfegaqsections; n++) {lfegaqgain[n]} 5 } for (bin = 0; bin < nlfemant; bin++) { if (lfegaqbin[bin]) { for (n = 0; n < 6; n++) {pre_lfemant[n][bin]} (0–16) } else {pre_lfemant[0][bin]} (0–9) } lf eahtinu = -1 /* AHT info for this frame has been read – do not read again */ } } } /* end of audblk */ 169 Advanced Television Systems Committee, Inc. Document A/52:2010 E2.2.5 auxdata: Auxiliary Data Table E2.5 auxdata Syntax and Word Size Syntax Word Size auxdata() { auxbits nauxbits if (auxdatae) { auxdatal 14 } auxdatae 1 } /* end of auxdata */ E2.2.6 errorcheck: Error Detection Code Table E2.6 errorcheck Syntax and Word Size Syntax Word Size errorcheck() { encinfo 1 crc2 16 } /* end of errorcheck */ E2.3 Description of Enhanced AC-3 bit stream elements Unless otherwise indicated, all bit stream elements retain the same meaning and purpose as described in this document, including Annex D, “Alternate Bit Stream Syntax.” E2.3.1 bsi: Bit Stream Information E2.3.1.1 strmtyp: Stream Type, 2 bits This 2-bit code, as shown in Table E2.7, indicates the stream type. Table E2.7 Stream Type strmtyp Indication ‘00’ Type 0 ‘01’ Type 1 ‘10’ Type 2 ‘11’ Type 3 The stream types are defined as follows: Type 0: These frames comprise an independent stream or substream. The program may be decoded independently of any other substreams that might exist in the bit stream. Type 1: These frames comprise a dependent substream. The program must be decoded in conjunction with the independent substream with which it is associated. 170 Digital Audio Compression Standard, Annex E 22 November 2010 Type 2: These frames comprise an independent stream or substream that was previously coded in AC-3. Type 2 streams must be independently decodable, and may not have any dependent streams associated with them. Type 3: Reserved. E2.3.1.2 substreamid: Substream Identification, 3 bits This field indicates the substream identification parameter. The substream identification parameter can be used, in conjunction with additional bit stream metadata, to enable carriage of a single program of more than 5.1 channels, multiple programs of up to 5.1 channels, or a mixture of programs with up to 5.1 channels and programs with greater than 5.1 channels. All Enhanced AC-3 bit streams must contain an independent substream assigned substream ID 0. The independent substream assigned substream ID 0 must be the first substream present in the bit stream. If an AC-3 bitstream is present in the Enhanced AC-3 bitstream, then the AC-3 bitstream shall be treated as an independent substream assigned substream ID 0. Enhanced AC-3 bit streams may also contain up to 7 additional independent substreams assigned substream ID’s 1 – 7. Independent substream ID’s must be assigned sequentially in the order the independent substreams are present in the bit stream. Independent substreams 1 – 7 must contain the same number of blocks per sync frame as independent substream 0. Each independent substream may have up to 8 dependent substreams associated with it. Dependent substreams must immediately follow the independent substream with which they are associated. Dependent substreams are assigned substream ID’s 0 – 7, which must be assigned sequentially according to the order the dependent substreams are present in the bit stream. Dependent substreams 0 – 7 must contain the same number of blocks per sync frame as the independent substream with which they are associated. For more information about usage of the substreamid parameter, please refer to Section E3.7. E2.3.1.3 frmsiz: Frame Size, 11 bits This field indicates a value one less than the overall size of the coded frame in 16-bit words. That is, this field may assume a value ranging from 0 to 2047, and these values correspond to frame sizes ranging from 1 to 2048. Note that some values at the lower end of this range may not be valid, as they may not represent enough words to convey a complete frame. It is the responsibility of the encoder to ensure that this does not occur in practice. E2.3.1.4 fscod: Sample Rate Code, 2 bits This is a 2-bit code indicating sample rate according to Table E2.8. If the fscod2 is indicated, the decoder should interpret the following 2-bits as fscod2. Table E2.8 Sample Rate Codes fscod Sampling Rate, kHz ‘00’ 48 ‘01’ 44.1 ‘10’ 32 ‘11’ fscod2 171 Advanced Television Systems Committee, Inc. Document A/52:2010 E2.3.1.5 numblkscod / fscod2: Number of Audio Blocks / Sample Rate Code 2, 2 bits numblkscod – This 2-bit code, as shown in Table E2.9, indicates the number of audio blocks per syncframe if the fscod indicates 32, 44.1, or 48 kHz sampling rate: Table E2.9 Number of Audio Blocks Per Syncframe numblkscod Indication ‘00’ 1 block per syncframe ‘01’ 2 blocks per syncframe ‘10’ 3 blocks per syncframe ‘11’ 6 blocks per syncframe fscod2 – If the fscod field indicates fscod2 then this 2-bit code indicates the reduced sample rate, as shown in Table E2.10. When using reduced sample rates, numblkscod shall be ‘11’ (6 blocks per syncframe). Table E2.10 Reduced Sampling Rates fscod2 Sampling Rate, kHz ‘00’ 24 ‘01’ 22.05 ‘10’ 16 ‘11’ reserved E2.3.1.6 bsid: Bit Stream Identification, 5 bits This bit field has a value of ‘10000’ (=16) for bitstreams compliant with this Annex. Values of bsid smaller than 16 and greater than 10 are used for versions of E-AC-3 which are backwards compatible with version 16 decoders. Decoders which can decode version 16 will thus be able to decode version numbers less than 16 and greater than 10. Additionally, E-AC-3 decoders must also be able to decode AC-3 bitstreams with bsid values 0 through 8. Decoders compliant with this Annex are not able to decode bit streams with bsid=9 or 10. Thus, decoders compliant with this Annex shall mute if the value of bsid is 9, 10, or greater than 16, and should decode and reproduce audio if the value of bsid is 0 – 8, or 11 – 16. E2.3.1.7 chanmape: Custom Channel Map Exists, 1 bit If this bit is a ‘0’, the channel map for a dependent substream is defined by the audio coding mode (acmod) and LFE on (lfeon) parameters. If this bit is a 1, the following 16 bits define the custom channel map for this dependent substream. Only dependent substreams can have a custom channel map. E2.3.1.8 chanmap: Custom Channel Map, 16 bits This 16-bit field specifies the custom channel map for a dependent substream. The channel locations supported by the custom channel map are defined in Table E2.11. Shaded entries in Table E2.11 represent channel locations present in the independent substream with which the 172 Digital Audio Compression Standard, Annex E 22 November 2010 dependent substream is associated. Non-shaded entries in Table E2.11 represent channel locations not present in the independent substream with which the dependent substream is associated. Table E2.11 Custom Channel Map Locations Bit Location Bit Location 0 Left 8 TBD 1 Center 9 TBD 2 Right 10 TBD 3 Left Surround 11 TBD 4 Right Surround 12 TBD 5 TBD 13 TBD 6 TBD 14 TBD 7 TBD 15 LFE The custom channel map indicates both which coded channels are present in the dependent substream and the order of the coded channels in the dependent substream. For each channel present in the dependent substream, the corresponding location bit in the chanmap is set to ‘1’. The order of the coded channels in the dependent substream is the same as the order of the enabled location bits in the chanmap. For example, if bits 0, 3, and 4 of the chanmap field are set to ‘1’, and the dependent stream is coded with acmod = 3 and lfeon = 0, the first coded channel in the dependent stream is the Left channel, the second coded channel is the Left Surround channel, and the third coded channel is the Right Surround channel. Note that the number of channel locations indicated by the chanmap field must equal the total number of coded channels present in the dependent substream, as indicated by the acmod and lfeon bit stream parameters. For more information about usage of the chanmap parameter, please refer to Section E3.7. E2.3.1.9 mixmdate: Mixing Meta-Data Exists, 1 bit If this bit is a ‘1’, mixing and mapping information follows in the bit stream. E2.3.1.10 lfemixlevcode: LFE mix Level Code Exists, 1 bit If this bit is a ‘1’, the LFE mix level code follows in the bit stream. If this bit is a ‘0’, the LFE mix level code is not present in the bit stream, and LFE mixing is disabled. E2.3.1.11 femixlevcod: LFE Mix Level Code, 5 bits This 5 bit code specifies the level at which the LFE data is mixed into the Left and Right channels during downmixing. The LFE mix level (in dB) can be derived from the LFE mix level code according to the following formula: LFE mix level (dB) = LFE mix level code + 10 Valid values for the LFE mix level code are 0 to 31, and valid values for the LFE mix level are therefore +10 to –21 dB. For more information on LFE mixing, please refer to Section E3.8. E2.3.1.12 pgmscle: Program Scale Factor Exists, 1 bit If this bit is a ‘1’, the program scale factor word follows in the bit stream. If it is ‘0’, the program scale factor word is defaulted to 0 dB (no scaling). 173 Advanced Television Systems Committee, Inc. Document A/52:2010 E2.3.1.13 pgmscl: Program Scale Factor, 6 bits This field specifies a scale factor that should be applied to the program during decoding. Valid values are 0 – 63, with 0 interpreted as mute, and 1 – 63 interpreted as –50 dB to +12 dB of scaling in 1 dB steps. E2.3.1.14 pgmscl2e: Program Scale Factor #2 Exists, 1 bit If this bit is a ‘1’, the program scale factor #2 word follows in the bit stream. If it is ‘0’, the program scale factor #2 word is defaulted to 0 dB (no scaling). E2.3.1.15 pgmscl2: Program Scale Factor #2, 6 bits This field has the same meaning as pgmscl, except that it applies to the second audio channel when acmod indicates two independent channels (dual mono 1+1 mode). E2.3.1.16 extpgmscle: External Program Scale Factor Exists, 1 bit If this bit is a ‘1’, the external program scale factor word follows in the bit stream. If it is ‘0’, the external program scale factor word is defaulted to 0 dB (no scaling). E2.3.1.17 extpgmscl: External Program Scale Factor, 6 bits In some applications, two bit streams may be decoded and mixed together. This field specifies a scale factor that should be applied to the external program (i.e., the program that is not carried in this bit stream) during mixing. This field uses the same scale as pgmscl. E2.3.1.18 mixdef: Mix Control Type, 2 bits This 2-bit code, as shown in Table E2.12, indicates the mode and parameter field lengths for program mixing. Table E2.12 Mix Control mixdef Indication ‘00’ mixing option 1, no additional bits ‘01’ mixing option 2, 5 bits reserved ‘10’ mixing option 3, 12 bits reserved ‘11’ mixing option 4, 16-264 bits reserved by mixdeflen E2.3.1.19 mixdeflen: Length of Mixing Parameter Data Field, 5 bits This defines the mixing data field size for the most flexible mode. The length is given in bytes: mixdeflen = {0, 1, 2, 3 … 31) represents mixdata lengths = {2, 3, 4, 5 … 33} bytes. E2.3.1.20 mixdata: Mixing Parameter Data, (5 – 264) bits This data field contains control parameters for mixing the program and external program streams during decoding. E2.3.1.21 paninfoe: Pan Information Exists, 1 bit If this bit is a ‘1’, panning information follows in the bit stream. If it is ‘0’, the pan position word is defaulted to “center”. 174 Digital Audio Compression Standard, Annex E 22 November 2010 E2.3.1.22 paninfo: Pan Information, 14 bits This 14-bit word defines how a single channel is reproduced in a two dimensional sound field. E2.3.1.23 paninfo2e: Pan Information Exists, 1 bit If this bit is a ‘1’, panning information #2 follows in the bit stream. If it is ‘0’, the pan position word is defaulted to “center”. E2.3.1.24 paninfo2: Pan Information, 14 bits This field has the same meaning as paninfo, except that it applies to the second audio channel when acmod indicates two independent channels (dual mono 1+1 mode). E2.3.1.25 frmmixcnfginfoe: Frame Mixing Configuration Information Exists, 1 bit This flag indicates whether frame mixing configuration information follows in the bit stream. If the flag is set to ‘0’, no frame mixing configuration information follows in the bit stream. If the flag is set to ‘1’, frame mixing configuration information follows in the bit stream. E2.3.1.26 blkmixcfginfoe: Block Mixing Configuration Information Exists, 1 bit This flag indicates whether block mixing configuration information follows in the bit stream. If the flag is set to ‘0’, no block mixing configuration information follows in the bit stream. If the flag is set to ‘1’, block mixing configuration information follows in the bit stream. Note that in the case where the number of blocks per frame is ‘1’, this flag is assumed to be ‘1’ and is not transmitted. E2.3.1.27 blkmixcfginfo[blk]: Block Mixing Configuration Information, 5 bits This field contains block mixing configuration information. E2.3.1.28 infomdate: Informational Meta-Data Exists, 1 bit If this bit is a ‘1’, informational meta-data follows in the bit stream. E2.3.1.29 sourcefscod: Source Sample Rate Code, 1 bit If the sourcefscod bit is a ‘1’, the source material was sampled at twice the rate indicated by fscod. E2.3.1.30 convsync: Converter Synchronization Flag, 1 bit This bit is used for synchronization by a device that converts an Enhanced AC-3 bit stream to an AC-3 bit stream. E2.3.1.31 blkid: Block Identification, 1 bit If strmtyp indicates a Type 2 bit stream, this bit is set to 1 to indicate that the first block in this Enhanced AC-3 frame was the first block in the original standard AC-3 frame. E2.3.2 audfrm – Audio Frame E2.3.2.1 expstre: Exponent Strategy Syntax Enabled, 1 bit If this bit is a ‘1’, full exponent strategy syntax exists in each audio block. If this bit is a ‘0’, then the exponent strategy is specified by the frame-based exponent strategy defined in Sections E2.3.2.12 and E2.3.2.13. 175 Advanced Television Systems Committee, Inc. Document A/52:2010 E2.3.2.2 ahte: Adaptive Hybrid Transform Enabled, 1 bit If this bit is a ‘1’, an Adaptive Hybrid Transform is used to code at least one of the independent channels, the coupling channel, or the LFE channel in the current frame. If this bit is a ‘0’, the entire frame is coded using the standard bit allocation and quantization model as described in Sections 7.2 and 7.3 in the main body of this document. E2.3.2.3 snroffststr : SNR Offset Strategy, 2 bits This field indicates how SNR offsets are transmitted Table E2.13 SNR Offset Strategy snroffststr Indication ‘00’ SNR offset strategy 1 ‘01’ SNR offset strategy 2 ‘10’ SNR offset strategy 3 ‘11’ Reserved SNR Offset Strategy 1: When SNR Offset Strategy 1 is used, one coarse SNR offset value and one fine SNR offset value are transmitted in the bit stream. These SNR offset values are used for every channel of every block in the frame, including the coupling and LFE channels. SNR Offset Strategy 2: When SNR Offset Strategy 2 is used, one coarse SNR offset value and one fine SNR offset value are transmitted in the bit stream as often as once per block. The fine SNR offset value is used for every channel in the block, including the coupling and LFE channels. For blocks in which coarse and fine SNR offset values are not transmitted in the bit stream, the decoder must reuse the coarse and fine SNR offset values from the previous block. One coarse and one fine SNR offset value must be transmitted in block 0. SNR Offset Strategy 3: When SNR Offset Strategy 3 is used, coarse and fine SNR offset values are transmitted in the bit stream as often as once per block. Separate fine SNR offset values are transmitted for each channel, including the coupling and LFE channels. For blocks in which coarse and fine SNR offset values are not transmitted in the bit stream, the decoder must reuse the coarse and fine SNR offset values from the previous block. Coarse and fine SNR offset values must be transmitted in block 0. E2.3.2.4 transproce: Transient Pre-Noise Processing Enabled, 1 bit If this bit is a ‘1’, at least one channel in the current frame contains transient pre-noise processing data. If it is ‘0’, transient pre-noise processing is not being utilized in this frame. E2.3.2.5 blkswe: Block Switch Syntax Enabled, 1 bit If this bit is a ‘1’, full block switch syntax exists in each audio block. E2.3.2.6 dithflage: Dither Flag Syntax Enabled, 1 bit If this bit is a ‘1’, full dither flag syntax exists in each audio block. E2.3.2.7 bamode: Bit Allocation Model Syntax Enabled, 1 bit If this bit is a ‘1’, full bit allocation syntax exists in each audio block. 176 Digital Audio Compression Standard, Annex E 22 November 2010 E2.3.2.8 frmfgaincode: Fast Gain Codes Enabled, 1 bit If this bit is a ‘1’, fast gain codes are transmitted in the bit stream as often as once per audio block. If this bit is a ‘0’, no fast gain codes are transmitted in the bit stream, and default fast gain code values are used for every channel of every block in the frame. E2.3.2.9 dbaflde: Delta Bit Allocation Syntax Enabled, 1 bit If this bit is a ‘1’, full delta bit allocation syntax exists in each audio block. E2.3.2.10 skipflde: Skip Field Syntax Enabled, 1 bit If this bit is a ‘1’, full skip field syntax exists in each audio block. E2.3.2.11 spxattene: Spectral Extension Attenuation Enabled, 1 bit If this bit is a ‘1’, at least one channel in the current frame contains spectral extension attenuation data. If it is a ‘0’, spectral extension attenuation processing is not being utilized in the frame. E2.3.2.12 frmcplexpstr : Frame Based Coupling Exponent Strategy, 5 bits This 5-bit code specifies the coupling channel exponent strategy for all audio blocks, as defined in Table E2.14. The number of blocks per frame is required to be six. Note that exponent strategies D15, D25, and D45 are as defined in Section 7.1 in the main body of this document, while ‘R’ indicates that exponents from the previous block are reused. E2.3.2.13 frmchexpstr[ch]: Frame Based Channel Exponent Strategy, 5 bits This 5-bit code specifies the channel exponent strategy for all audio blocks, as defined in Table E2.14. The number of blocks per frame is required to be six. Note that exponent strategies D15, D25, and D45 are as defined in Section 7.1 in the main body of this document, while ‘R’ indicates that exponents from the previous block are reused. E2.3.2.14 convexpstre: Converter Exponent Strategy Exists, 1 bit If this parameter is one, exponent strategy data used required by the E-AC-3 to AC-3 converter follows. Exponent strategy shall be provided once every 6 blocks. E2.3.2.15 convexpstr[ch]: Converter Channel Exponent Strategy, 5 bits This 5-bit code specifies the exponent strategy, as defined in Table E2.14, for each full bandwidth channel of each block of an AC-3 frame converted from a set of 1 or more E-AC-3 frames. 177 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E2.14 Frame Exponent Strategy Combinations frmcplexpstr Audio Block Number 0 1 2 3 4 5 0 D15 R R R R R 1 D15 R R R R D45 2 D15 R R R D25 R 3 D15 R R R D45 D45 4 D25 R R D25 R R 5 D25 R R D25 R D45 6 D25 R R D45 D25 R 7 D25 R R D45 D45 D45 8 D25 R D15 R R R 9 D25 R D25 R R D45 10 D25 R D25 R D25 R 11 D25 R D25 R D45 D45 12 D25 R D45 D25 R R 13 D25 R D45 D25 R D45 14 D25 R D45 D45 D25 R 15 D25 R D45 D45 D45 D45 16 D45 D15 R R R R 17 D45 D15 R R R D45 18 D45 D25 R R D25 R 19 D45 D25 R R D45 D45 20 D45 D25 R D25 R R 21 D45 D25 R D25 R D45 22 D45 D25 R D45 D25 R 23 D45 D25 R D45 D45 D45 24 D45 D45 D15 R R R 25 D45 D45 D25 R R D45 26 D45 D45 D25 R D25 R 27 D45 D45 D25 R D45 D45 28 D45 D45 D45 D25 R R 29 D45 D45 D45 D25 R D45 30 D45 D45 D45 D45 D25 R 31 D45 D45 D45 D45 D45 D45 E2.3.2.16 cplahtinu: Coupling Channel AHT in Use, 1bit If this bit is a ‘1’, the coupling channel is coded using an Adaptive Hybrid Transform. If this bit is a ‘0’, conventional coupling channel coding is used for that region. E2.3.2.17 chahtinu[ch]: Channel AHT in Use, 1 bit If this bit is a ‘1’, channel ch is coded using an Adaptive Hybrid Transform. If this bit is a ‘0’, conventional channel coding is used for that region. 178 Digital Audio Compression Standard, Annex E 22 November 2010 E2.3.2.18 lfeahtinu: LFE Channel AHT in Use, 1 bit If this bit is a ‘1’, the LFE channel is coded using an Adaptive Hybrid Transform. If this bit is a ‘0’, conventional LEF channel coding is used for that region. E2.3.2.19 frmcsnroffst: Frame Coarse SNR Offset, 6 bits This field contains the frame coarse SNR offset value. This coarse SNR offset value is used for every block in the frame. E2.3.2.20 frmfsnroffst: Frame Fine SNR Offset, 4 bits This field contains the frame fine SNR offset value. This fine SNR offset value is used for every channel of every block in the frame, including the coupling and LFE channels. E2.3.2.21 chintransproc[ch]: Channel in Transient Pre-Noise Processing, 1 bit Transient pre-noise processing exist bit for each full bandwidth channel. If set to ‘1’, then the corresponding channel has associated transient pre-noise processing data. E2.3.2.22 transprocloc[ch]: Transient Location Relative to Start of Frame, 10 bits This field provides the location of the transient relative to the start of the current frame. The transient location (in samples) is calculated by multiplying this value by 4. It is possible for the transient to be located in a later audio frame and therefore this number can exceed the number of PCM samples contained within the current frame. E2.3.2.23 transproclen[ch]: Transient Processing Length, 8 bits This field provides the transient pre-noise processing length in samples, relative to the location of the transient provided by the value of transprocloc[ch]. E2.3.2.24 chinspxatten[ch]: Channel in Spectral Extension Attenuation Processing, 1 bit If this bit is a ‘1’, channel [ch] is using spectral extension attenuation processing. If it is a ‘0’, channel [ch] is not using spectral extension attenuation processing. E2.3.2.25 spxattencod[ch]: Spectral Extension Attenuation Code, 5 bits This 5-bit code specifies the index for channel [ch] into Table E3.14 from which spectral extension attenuation values are derived. E2.3.2.26 blkstrtinfoe: Block Start Information Exists, 1 bit If this bit is a ‘1’, block start information follows in the bit stream. If this bit is a ‘0’, no block start information follows in the bit stream. E2.3.2.27 blkstrtinfo: Block Start Information, nblkstrtbits This field contains the block start information. The number of bits of block start information is given by the formula nblkstrtbits = (numblks – 1) * (4 + ceiling (log2 (words_per_frame))) where numblks is derived from the numblkscod in Table E2.15 and ceiling(n) is a function which rounds the fractional number n up to the next higher integer. 179 Advanced Television Systems Committee, Inc. Document A/52:2010 For example, ceiling(2.1) = 3 log2(n) is the base 2 logarithm of n words_per_frame = frmsiz + 1 E2.3.2.28 firstspxcos[ch]: First Spectral Extension Coordinates States The firstspxcos[ch] state determines when new spectral extension coordinates can be assumed to exist in the bit stream. If firstspxcos[ch] is set to ‘1’, the spxcoe[ch] bit is assumed to be ‘1’ for the current block and is not transmitted in the bit stream. E2.3.2.29 firstcplcos[ch]: First Coupling Coordinates States The firstcplcos[ch] state determines when new coupling coordinates can be assumed to exist in the bit stream. If firstcplcos[ch] is set to 1, the cplcoe[ch] bit is assumed to be 1 for the current block and is not transmitted in the bit stream. E2.3.2.30 firstcplleak: First Coupling Leak State The firstcplleak state determines when new coupling leak values can be assumed to exist in the bit stream. If firstcplleak is set to ‘1’, the cplleake bit is assumed to be ‘1’ for the current block and is not transmitted in the bit stream. E2.3.3 audblk: Audio Block E2.3.3.1 spxstre: Spectral Extension Strategy Exists, 1 bit If this bit is a ‘1’, spectral extension information follows in the bit stream. If it is ‘0’, new spectral extension information is not present, and spectral extension parameters previously sent are reused. E2.3.3.2 spxinu: Spectral Extension in Use, 1 bit If this bit is a ‘1’, then the spectral extension technique is used in this block. If this bit is a ‘0’, then the spectral extension technique is not used in this block. E2.3.3.3 chinspx[ch]: Channel Using Spectral Extension, 1 bit If this bit is a ‘1’, then the channel indicated by the index [ch] is utilizing spectral extension. If the bit is a ‘0’, then this channel is not utilizing spectral extension. E2.3.3.4 spxstrtf: Spectral Extension Start Copy Frequency Code, 2 bits This 2-bit code is used to derive the number of the lowest frequency sub-band of the spectral extension copy region. See Table E3.13 for the definition of the spectral extension sub-bands. E2.3.3.5 spxbegf: Spectral Extension Begin Frequency Code, 3 bits This 3-bit code is used to derive the number of the lowest frequency sub-band of the spectral extension region. 180 Digital Audio Compression Standard, Annex E 22 November 2010 E2.3.3.6 spxendf: Spectral Extension End Frequency Code, 3 bits This 3-bit code is used to derive a number one greater than the highest frequency sub-band of the spectral extension region. E2.3.3.7 spxbndstrce: Spectral Extension Band Structure Exist, 1 bit If this parameter is one, the spectral extension band structure follows. If it is zero in the first block using spectral extension, a default spectral extension band structure is used. If it is zero in any other block, the band structure from the previous block is reused. The default banding structure defspxbndstrc[] is shown in Table E2.15. Table E2.15 Default Spectral Extension Banding Structure spx sub-band # defspxbndstrc[] 0 0 1 0 2 0 3 0 4 0 5 0 6 0 7 0 8 1 9 0 10 1 11 0 12 1 13 0 14 1 15 0 16 1 E2.3.3.8 spxbndstrc[bnd]: Spectral Extension Band Structure, 1 – 14 bits This data structure determines the grouping of subbands in spectral extension, and operates in the same fashion as the coupling band structure. For each subband: • A zero represents the beginning of a new band • A one indicates that the subband should be combined into the previous band Note that it is assumed that the first band begins at the first subband. Therefore, the first band is assumed to be zero and not sent. The first band in the structure corresponds to the second subband. E2.3.3.9 spxcoe[ch]: Spectral Extension Coordinates Exist, 1 bit If this parameter is ‘1’, spectral extension coordinate information follows. If it is ‘0’, the spectral extension coordinates from the previous block are used. 181 Advanced Television Systems Committee, Inc. Document A/52:2010 E2.3.3.10 spxblnd[ch]: Spectral Extension Blend, 5 bits This per channel parameter determines the per channel noise blending factor (translated signal mixed with random noise) for the spectral extension process. E2.3.3.11 mstrspxco[ch]: Master Spectral Extension Coordinate, 2 bits This per channel parameter establishes a per channel gain factor (increasing the dynamic range) for the spectral extension coordinates as shown in Table 5.14 in the main body of this document which describes the mstrcplco[ch] element. E2.3.3.12 spxcoexp[ch][bnd]: Spectral Extension Coordinate Exponent, 4 bits Each spectral extension coordinate is composed of a 4-bit exponent and a 2-bit mantissa. This element is the value of the spectral extension coordinate exponent for channel [ch] and band [bnd]. The index [ch] only will exist for those channels that are in spectral extension. The index [bnd] will range from zero to nspxbnds. E2.3.3.13 spxcomant[ch][bnd]: Spectral Extension Coordinate Mantissa, 2 bits This element is the 2-bit spectral extension coordinate mantissa for the channel [ch] and band [bnd]. E2.3.3.14 ecplinu: Enhanced Coupling in Use, 1 bit If this bit is a ‘1’, enhanced coupling is used for the current block. If this bit is a ‘0’, standard coupling is used for the current block. E2.3.3.15 cplbndstrce: Coupling Band Structure Exist, 1 bit If this parameter is ‘1’, the coupling band structure follows. If it is ‘0’ in the first block using coupling, a default coupling band structure is used. If it is ‘0’ in any other block, the band structure from the previous block is reused. The default coupling banding structure defcplbndstrc[] is shown in Table E2.16. 182 Digital Audio Compression Standard, Annex E 22 November 2010 Table E2.16 Default Coupling Banding Structure couple sub-band # defcplbndstrc[] 0 1 0 2 0 3 0 4 0 5 0 6 0 7 0 8 1 9 0 10 1 11 1 12 0 13 1 14 1 15 1 16 1 17 1 E2.3.3.16 ecplbegf: Enhanced Coupling Begin Frequency Code, 4 bits This 4-bit code is used to derive the number of the lowest frequency edge of the enhanced coupling channel (or the first active enhanced coupling sub-band) as shown in Table E3.8. The index of the first active enhanced coupling sub-band is equal to ecpl_start_subbnd and is calculated as: if (ecplbegf < 3) {ecpl_start_subbnd = ecplbegf * 2} else if (ecplbegf < 13) {ecpl_start_subbnd = ecplbegf + 2} else {ecpl_start_subbnd = ecplbegf * 2 - 10} E2.3.3.17 ecplendf: Enhanced Coupling End Frequency Code, 4 bits This 4-bit code is used to derive a number one greater than the highest frequency sub-band of the enhanced coupling region. See Table E3.8. The index of one greater than the highest active enhanced coupling sub-band is equal to ecpl_end_subbnd and is calculated as: if (spxinu == 0) {ecpl_end_subbnd = ecplendf + 7} else if (spxbegf < 6) {ecpl_end_subbnd = spxbegf + 5} else {ecpl_end_subbnd = spxbegf * 2} E2.3.3.18 ecplbndstrce: Enhanced Coupling Band Structure Exists, 1 bit If this parameter is one, the enhanced coupling band structure follows. If it is zero in the first block using enhanced coupling, a default enhanced coupling band structure is used. If it is zero in 183 Advanced Television Systems Committee, Inc. Document A/52:2010 any other block, the band structure from the previous block is reused. The default enhanced coupling banding structure defecplbndstrc[] is shown in Table E2.17. Table E2.17 Default Enhanced Coupling Banding Structure Enhanced Coupling defecplbndstrc[] Sub-Band # 0 to 8 0 9 1 10 0 11 1 12 0 13 1 14 0 15 1 16 1 17 1 18 0 19 1 20 1 21 1 E2.3.3.19 ecplbndstrc[sbnd]: Enhanced Coupling Band Structure, 1 bit There are 22 enhanced coupling sub-bands defined in Table E3.7, each containing either 6 or 12 frequency coefficients. The fixed 12-bin wide enhanced coupling sub-bands 8 and above are converted into enhanced coupling bands, each of which may be wider than (a multiple of) 12 frequency bins. Sub-bands 0 through 7 are never grouped together to form larger enhanced coupling bands, and are thus each considered enhanced coupling bands. Each enhanced coupling band may contain one or more enhanced coupling sub-bands. Enhanced coupling coordinates are transmitted for each enhanced coupling band. Each band’s enhanced coupling coordinate must be applied to all the coefficients in the enhanced coupling band. The enhanced coupling band structure indicates which enhanced coupling sub-bands are combined into wider enhanced coupling bands. When ecplbndstrc[sbnd] is a ‘0’, the sub-band number [sbnd] is not combined into the previous band to form a wider band, but starts a new 12-bin wide enhanced coupling band. When cplbndstrc[sbnd] is a ‘1’, then the sub-band [sbnd] is combined with the previous band, making the previous band 12 bins wider. Each successive value of ecplbndstrc which is a ‘1’ will continue to combine sub-bands into the current band. When another ecplbndstrc value of ‘0’ is received, then a new band will be formed, beginning with the 12 bins of the current sub-band. The set of ecplbndstrc[sbnd] values is typically considered an array. Each bit in the array corresponds to a specific enhanced coupling sub-band in ascending frequency order. The elements of the array corresponding to the sub-bands up to and including ecpl_start_subbnd or 8 (whichever is greater), are always 0, and are not transmitted. (There is no reason to send an ecplbndstrc bit for these sub-bands, since these bits are always 0.) If there is only one enhanced coupling sub-band above sub-band 7, then no ecplbndstrc bits are sent. 184 Digital Audio Compression Standard, Annex E 22 November 2010 The total number of enhanced coupling bands, necplbnd, may be computed as follows: necplbnd = ecpl_end_subbnd - ecpl_start_subbnd; necplbnd -= ecplbndstrc[ecpl_start_subbnd] + … + ecplbndstrc[ecpl_end_subbnd -1] A default setting of ecplbndstrc[], when all bands are used in enhanced coupling, is given in Table E2.17. E2.3.3.20 ecplangleintrp: Enhanced Coupling Angle Interpolation Flag, 1 bit If this element is set to ‘1’, then interpolation is used to derive enhanced coupling bin angle values between band angle values according to the pseudo-code specified in Section E3.4.5.3. If this element is set to ‘0’, then interpolation is not used and each enhanced coupling band value should be applied to all the bin angle values within the band. E2.3.3.21 ecplparam1e[ch]: Enhanced Coupling Parameters 1 Exist, 1 bit Enhanced coupling parameters are used to derive the enhanced coupling coordinates which indicate, for a given channel and within a given enhanced coupling band, the fraction of the enhanced coupling channel frequency coefficients to use to re-create the individual channel frequency coefficients. Enhanced coupling parameters are conditionally transmitted in the bit stream. If new values are not delivered, the previously sent values remain in effect. See Section E3.4 for further information on enhanced coupling. Each enhanced coupling coordinate is derived from a 5-bit amplitude, a 6-bit angle, a 3-bit chaos measure and a 1-bit transient present flag. With the exception of the transient present flag, enhanced coupling parameters are signaled by two exist bits. If ecplparam1e[ch] is ‘1’, the amplitudes for the corresponding channel [ch] exist and follow in the bit stream. If the bit is ‘0’, the previously transmitted amplitudes for this channel are reused. All amplitudes are always transmitted in the first block in which enhanced coupling is enabled. E2.3.3.22 ecplparam2e[ch]: Enhanced Coupling Parameters 2 Exist, 1 bit If ecplparam2e[ch] is ‘1’, the angle and chaos values for the corresponding channel [ch] exist and follow in the bit stream. If the bit is ‘0’, the previously transmitted angle and chaos values for this channel are reused. The angle and chaos parameters are always transmitted in the first block in which enhanced coupling is enabled. E2.3.3.23 ecplamp[ch][bnd]: Enhanced Coupling Amplitude Scaling, 5 bits This element is the value of the enhanced coupling amplitude for channel [ch] and band [bnd]. The index [ch] will only exist for those channels in enhanced coupling. The index [bnd] will range from 0 to necplbnds–1. See Section E3.4.5 for more information on how to interpret enhanced coupling parameters. E2.3.3.24 ecplangle[ch][bnd]: Enhanced Coupling Angle, 6 bits This element is the 6-bit enhanced coupling angle for channel [ch] and band [bnd]. The enhanced coupling angle is assumed to be 0 for the first channel [ch] in enhanced coupling, and is not transmitted in the bit stream. 185 Advanced Television Systems Committee, Inc. Document A/52:2010 E2.3.3.25 ecplchaos[ch][bnd]: Enhanced Coupling Chaos, 3 bits This element is the 3-bit enhanced coupling chaos for channel [ch] and band [bnd]. The enhanced coupling chaos is assumed to be 0 for the first channel [ch] in enhanced coupling, and is not transmitted in the bit stream. E2.3.3.26 ecpltrans[ch]: Enhanced Coupling Transient Present, 1 bit This element is the 1-bit enhanced coupling transient present indication for channel [ch]. The enhanced coupling transient present bit is not transmitted in the bit stream for the first channel [ch] in enhanced coupling. E2.3.3.27 blkfsnroffst: Block Fine SNR Offset, 4 bits This 4-bit code specifies the fine SNR offset value used by all channels, including the coupling and LFE channels. E2.3.3.28 fgaincode: Fast Gain Codes Exist, 1 bit If this parameter is set to ‘1’, fast gain codes for each channel are transmitted in the bit stream. If this parameter is set to ‘0’ in block 0, no fast gain codes are transmitted in the bit stream, and default fast gain codes are used. If parameter is set to 0 in any other block, no fast gain codes are transmitted in the bit stream, and fast gain codes from the previous block are re-used. E2.3.3.29 convsnroffste: Converter SNR Offset Exists, 1 bit If this parameter is one, a SNR offset for the converter follows. E2.3.3.30 convsnroffst: Converter SNR Offset, 10 bits This 10 bit word is the SNR offset required to convert the current frame to an AC-3 frame. E2.3.3.31 chgaqmod[ch]: Channel Gain Adaptive Quantization Mode, 2 bits This 2-bit code specifies which one of four possible quantization modes is used for mantissas in the given channel. If chgaqmod[ch] is 0, conventional scalar quantization is used for channel ch. Otherwise, gain adaptive quantization is used and chgaqgain[ch][n] words follow in the bit stream. E2.3.3.32 chgaqgain[ch][n]: Channel Gain Adaptive Quantization gain, 1 or 5 bits This code signals the adaptive quantizer gain value or values associated with one or more exponents. If chgaqmod[ch] is either 1 or 2, chgaqgain[ch][n] is 1 bit in length, signaling two possible gain states. If chgaqmod[ch] is 3, chgaqgain[ch][n] is 5 bits in length, representing a triplet of gains coded compositely. In this case, each gain signals three possible gain states. E2.3.3.33 pre_chmant[n][ch][bin]: Pre Channel Mantissas, 0 to 16 bits These values represent the channel mantissas coded either with scalar, vector or gain adaptive quantization. E2.3.3.34 cplgaqmod: Coupling Channel Gain Adaptive Quantization Mode, 2 bits This 2-bit code specifies which one of four possible quantization modes is used for mantissas in the coupling channel. If cplgaqmod is 0, conventional scalar quantization is used. Otherwise, gain adaptive quantization is used and cplgaqgain[n] words follow in the bit stream. 186 Digital Audio Compression Standard, Annex E 22 November 2010 E2.3.3.35 cplgaqgain[n]: Coupling Gain Adaptive Quantization Gain, 1 or 5 bits This code signals the adaptive quantizer gain value or values associated with one or more exponents. If cplgaqmod is either 1 or 2, cplgaqgain[n] is 1 bit in length, signaling two possible gain states. If cplgaqmod is 3, cplgaqgain[n] is 5 bits in length, representing a triplet of gains coded compositely. In this case, each gain signals three possible gain states. E2.3.3.36 pre_cplmant[n][bin]: Pre Coupling Channel Mantissas, 0 to 16 bits These values represent the coupling channel mantissas coded either with scalar, vector or gain adaptive quantization. E2.3.3.37 lfegaqmod: LFE Channel Gain Adaptive Quantization Mode, 2 bits This 2-bit code specifies which one of four possible quantization modes is used for mantissas in the LFE channel. If lfegaqmod is 0, conventional scalar quantization is used. Otherwise, gain adaptive quantization is used and lfegaqgain[n] words follow in the bit stream. E2.3.3.38 lfegaqgain[n]: LFE Gain Adaptive Quantization Gain, 1 or 5 bits This code signals the adaptive quantizer gain value or values associated with one or more exponents. If lfegaqmod is either 1 or 2, lfegaqgain[n] is 1 bit in length, signaling two possible gain states. If lfegaqmod is 3, lfegaqgain[n] is 5 bits in length, representing a triplet of gains coded compositely. In this case, each gain signals three possible gain states. E2.3.3.39 pre_lfemant[n][bin]: Pre LFE Channel Mantissas, 0 to 16 bits These values represent the LFE channel mantissas coded either with scalar, vector or gain adaptive quantization. E3. ALGORITHMIC DETAILS This section specifies how the reference Enhanced AC-3 decoder shall process bit streams that use the Enhanced AC-3 bit stream syntax. Some of the decoding process is shown in the form of pseudo code; all pseudo code is normative. E3.1 Glitch-Free Switching Between Different Stream Types Enhanced AC-3 decoders should be designed to switch between all supported bit stream types without introducing audible clicks or pops. E3.2 Error Detection and Concealment Enhanced AC-3 decoders are required to implement error detection based on the bit stream CRC word. Enhanced AC-3 bit streams contain only one CRC word, which covers the entire frame. When decoding bit streams that use the Enhanced AC-3 bit stream syntax, Enhanced AC-3 decoders must verify the CRC word prior to decoding any of the blocks in the frame. If the CRC word for an Enhanced AC-3 bit stream is found to be invalid, all blocks in the frame must be substituted with an appropriate error concealment signal. For most applications, this can be easily accomplished by simply repeating the last known-good block (before the overlap-add window process). 187 Advanced Television Systems Committee, Inc. Document A/52:2010 E3.3 Adaptive Hybrid Transform Processing E3.3.1 Overview The Adaptive Hybrid Transform (AHT) is composed of two linear transforms connected in cascade. The first transform is identical to that employed in AC-3 – a windowed Modified Discrete Cosine Transform (MDCT) of length 128 or 256 frequency samples. This feature provides compatibility with AC-3 without the need to return to the time domain in the decoder. For frames containing audio signals which are not time-varying in nature (stationary), a second transform can optionally be applied by the encoder, and inverted by the decoder. The second transform is composed of a non-windowed, non-overlapped Discrete Cosine Transform (DCT Type II). When the DCT is employed, the effective audio transform length increases from 256 to 1536 audio samples. This results in significantly improved coding gain and perceptual coding performance for stationary signals. The AHT transform is enabled by setting the ahte bit stream parameter to 1. If ahte is 1, at least one of the independent channels, the coupling channel, or the LFE channel has been coded with AHT. The chahtinu[ch], cplahtinu, and lfeahtinu bit stream parameters are used to indicate which channels are channels coded with AHT. In order to realize gain made available by the AHT, the AC-3 scalar quantizers have been augmented with two new coding tools. When AHT is in use, both 6-dimensional vector quantization (VQ) and gain-adaptive quantization (GAQ) are employed. VQ is employed for the largest step sizes (coarsest quantization), and GAQ is employed for the smallest stepsizes (finest quantization). The selection of quantizer step size is performed using the same parametric bit allocation method as AC-3, except the conventional bit allocation pointer (bap) table is replaced with a high-efficiency bap table (hebap[]). The hebap[] table employs finer-granularity than the conventional bap table, enabling more efficient allocation of bits. E3.3.2 Bit Stream Helper Variables Several helper variables must be computed during the decode process in order to decode a frame containing at least one channel using AHT (ahte = 1). These variables are not transmitted in the bit stream itself, but are computed from other bit stream parameters. The first helper variables of this type are denoted in the bit stream syntax as ncplregs, nchregs[ch], and nlferegs. The method for computing these variables is presented in the following three sections of pseudo code. Generally speaking, the nregs variables are set equal to the number of times exponents are transmitted in the frame. 188 Digital Audio Compression Standard, Annex E 22 November 2010 Pseudo Code /* Only compute ncplregs if coupling in use for all 6 blocks */ ncplregs = 0; /* AHT is only available in 6 block mode (numblkscod ==0x3) */ for (blk = 0; blk < 6; blk++) { if ( (cplstre[blk] == 1) || (cplexpstr[blk] != reuse) ) { ncplregs++; } } Pseudo Code for (ch = 0; ch < nfchans; ch++) { nchregs[ch] = 0; /* AHT is only available in 6 block mode (numblkscod ==0x3) */ for (blk = 0; blk < 6; blk++) { if (chexpstr[blk][ch] != reuse) { nchregs[ch]++; } } } Pseudo Code nlferegs = 0; /* AHT is only available in 6 block mode (numblkscod ==0x3) */ for (blk = 0; blk < 6; blk++) { if ( lfeexpstr[blk] != reuse) { nlferegs++; } } A second set of helper variables are required for identifying which and how many mantissas employ GAQ. The arrays identifying which bins are GAQ coded are called chgaqbin[ch][bin], cplgaqbin[bin], and lfegaqbin[bin]. Since the number and position of GAQ-coded mantissas varies from frame to frame, these variables need to be computed after the corresponding hebap[] array is available, but prior to mantissa unpacking. This procedure is shown in the following pseudo-code. 189 Advanced Television Systems Committee, Inc. Document A/52:2010 Pseudo Code if (cplahtinu == 0) { for (bin = cplstrtmant; bin < cplendmant; bin++) { cplgaqbin[bin] = 0; } } else { if (cplgaqmod < 2) { endbap = 12; } else { endbap = 17; } cplactivegaqbins = 0; for (bin = cplstrtmant; bin < cplendmant; bin++) { if (cplhebap[bin] > 7 && cplhebap[bin] < endbap) { cplgaqbin[bin] = 1; /* Gain word is present */ cplactivegaqbins++; } else if (cplhebap[bin] >= endbap) { cplgaqbin[bin] = -1; /* Gain word is not present */ } else { cplgaqbin[bin] = 0; } } } 190 Digital Audio Compression Standard, Annex E 22 November 2010 Pseudo Code for (ch = 0; ch < nfchans; ch++) { if (chahtinu[ch] == 0) { for (bin = 0; bin < endmant[ch]; bin++) { chgaqbin[ch][bin] = 0; } } else { if (chgaqmod < 2) { endbap = 12; } else { endbap = 17; } chactivegaqbins[ch] = 0; for (bin = 0; bin < endmant[ch]; bin++) { if (chhebap[ch][bin] > 7 && chhebap[ch][bin] < endbap) { chgaqbin[ch][bin] = 1; /* Gain word is present */ chactivegaqbins[ch]++; } else if (chhebap[ch][bin] >= endbap) { chgaqbin[ch][bin] = -1;/* Gain word not present */ } else { chgaqbin[ch][bin] = 0; } } } } 191 Advanced Television Systems Committee, Inc. Document A/52:2010 Pseudo Code if (lfeahtinu == 0) { for (bin = 0; bin < lfeendmant; bin++) { lfegaqbin[bin] = 0; } } else { if (lfegaqmod < 2) { endbap = 12; } else { endbap = 17; } lfeactivegaqbins = 0; for (bin = 0; bin < lfeendmant; bin++) { if (lfehebap[bin] > 7 && lfehebap[bin] < endbap) { lfegaqbin[bin] = 1; /* Gain word is present */ lfeactivegaqbins++; } else if (lfehebap[bin] >= endbap) { lfegaqbin[bin] = -1; /* Gain word is not present */ } else { lfegaqbin[bin] = 0; } } } In a final set of helper variables, the number of gain words to be read from the bitstream is computed. These variables are called chgaqsections[ch], cplgaqsections, and lfegaqsections for the independent channels, coupling channel, and LFE channel, respectively. They denote the number of GAQ gain words transmitted in the bit stream, and are computed as shown in the following pseudo code. 192 Digital Audio Compression Standard, Annex E 22 November 2010 Pseudo Code if (cplahtinu == 0) { cplgaqsections = 0; } else { switch(cplgaqmod) { case 0: /* No GAQ gains present */ { cplgaqsections = 0; break; } case 1: /* GAQ gains 1 and 2 */ case 2: /* GAQ gains 1 and 4 */ { cplgaqsections = cplactivegaqbins;/* cplactivegaqbins was computed earlier */ break; } case 3: /* GAQ gains 1, 2, and 4 */ { cplgaqsections = cplactivegaqbins / 3; if (cplactivegaqbins % 3) cplgaqsections++; break; } } } 193 Advanced Television Systems Committee, Inc. Document A/52:2010 Pseudo Code for (ch = 0; ch > 5 ; address = min(63, max(0, address)) ; hebap[i] = hebaptab[address] ; i++ ; } 196 Digital Audio Compression Standard, Annex E 22 November 2010 j++; } while (end > lastbin) ; } else { i = start ; j = masktab[start] ; do { lastbin = min(bndtab[j] + bndsz[j], end); mask[j] -= snroffset ; mask[j] -= floor ; if (mask[j] < 0) { mask[j] = 0 ; } mask[j] &= 0x1fe0 ; mask[j] += floor ; for (k = i; k < lastbin; k++) { address = (psd[i] - mask[j]) >> 5 ; address = min(63, max(0, address)) ; bap[i] = baptab[address] ; i++ ; } j++; } while (end > lastbin) ; } 197 Advanced Television Systems Committee, Inc. Document A/52:2010 E3.3.3.2 Bit Allocation Tables Table E3.1 High Efficiency Bit Allocation Pointers, hebaptab[] Address hebaptab[address] Address hebaptab[address] 0 0 32 14 1 1 33 14 2 2 34 14 3 3 35 15 4 4 36 15 5 5 37 15 6 6 38 15 7 7 39 16 8 8 40 16 9 8 41 16 10 8 42 16 11 8 43 17 12 9 44 17 13 9 45 17 14 9 46 17 15 10 47 18 16 10 48 18 17 10 49 18 18 10 50 18 19 11 51 18 20 11 52 18 21 11 53 18 22 11 54 18 23 12 55 19 24 12 56 19 25 12 57 19 26 12 58 19 27 13 59 19 28 13 60 19 29 13 61 19 30 13 62 19 31 14 63 19 198 Digital Audio Compression Standard, Annex E 22 November 2010 Table E3.2 Quantizer Type, Quantizer Level, and Mantissa Bits vs. hebap hebap Quantizer Type Levels Mantissa Bits 0 NA NA 0 1 VQ NA (2/6) 2 VQ NA (3/6) 3 VQ NA (4/6) 4 VQ NA (5/6) 5 VQ NA (7/6) 6 VQ NA (8/6) 7 VQ NA (9/6) 8 symmetric + GAQ 7 3 9 symmetric + GAQ 15 4 10 symmetric + GAQ 31 5 11 symmetric + GAQ 63 6 12 symmetric + GAQ 127 7 13 symmetric + GAQ 255 8 14 symmetric + GAQ 511 9 15 symmetric + GAQ 1023 10 16 symmetric + GAQ 2047 11 17 symmetric + GAQ 4095 12 18 symmetric + GAQ 16,383 14 19 symmetric + GAQ 65,535 16 E3.3.4 Quantization Depending on the bit allocation pointer (hebap) calculated in Section E3.3.3.1, the mantissa values are either coded using vector quantization or gain adaptive quantization. The following section describes both of these coding techniques. E3.3.4.1 Vector Quantization Vector quantization is a quantization technique that takes advantage of similarities and patterns in an ordered series of values, or vector, to reduce redundancy and hence improve coding efficiency. For AHT processing, 6 mantissa values across blocks within a single spectral bin are grouped together to create a 6-dimensional Euclidean space. If AHT is in use and the bit allocation pointer is between 1 and 7 inclusive, then vector quantization (VQ) is used to encode the mantissas. The range of hebap values that use VQ are shown in Table E3.2. If VQ is applied to a set of 6 mantissa values then the data in the bit stream represents an N bit index into a 6-dimensional look up table, where N is dependent on the hebap value as defined in Table E3.2. When vector quantization is used, the values shall be compared to the values in the vector tables for each bit allocation pointer between 1 and 7 inclusive as shown in Section E3.8. The values in the vector tables are represented as 16-bit, signed (two's complement) values. If a hebap value is within the VQ range, the encoder selects the best vector to transmit to the decoder by locating the vector which minimizes the Euclidean distance between the actual 199 Advanced Television Systems Committee, Inc. Document A/52:2010 Figure E3.1 Flow diagram for GAQ mantissa dequantization. mantissa vector and the table vector. The index of the closest matching vector is then transmitted to the decoder. In the decoder, the index is read from the bit stream and the mant values are replaced with the values from the appropriate vector table. E3.3.4.2 Gain Adaptive Quantization Gain-adaptive quantization (GAQ) is a method for quantizing mantissas using variable-length codewords. In the encoder, the technique is based upon conditionally amplifying one or more of the smaller and typically more frequently occurring transform coefficient mantissas in one DCT block, and representing these with a shorter length code. Larger transform coefficients are not gain amplified, but are transmitted using longer codes since these occur relatively infrequently for typical audio signals. The gain words selected by the encoder, one per GAQ-coded DCT block of length six, are packed together with the mantissa codewords and transmitted as side information. With this system, the encoder can adapt to changing local signal statistics from frame to frame, and/or from channel to channel. Since a coding mode using constant-length output symbols is included as a subset, gain-adaptive quantization cannot cause a noticeable coding loss compared to the fixed-length codes used in AC-3. In the decoder, the individual gain words are unpacked first, followed by a bit stream parsing operation (using the gains) to reconstruct the individual transform coefficient mantissas. To compensate for amplification applied in the encoder, the decoder applies an attenuation factor to the small mantissas. The level of large mantissas is unaffected by these gain factors in both the encoder and decoder. The decoder structure for gain-adaptive quantization is presented in Figure E3.1. Decoder processing consists of a bit stream deformatter connected in cascade with the switched gain attenuation element, labeled as 1/Gk in the figure. The three inputs to the deformatter are the packed mantissa bit stream, the hebap[] output from the parametric bit allocation, and the gaqgain[] array received from the encoder. The hebap[] array is used by the deformatter to determine if the current (kth) DCT block of six mantissas to be unpacked is coded with GAQ, and if so, what the small and large mantissa bit lengths are. The gaqgain[] array is processed by the deformatter to 200 Digital Audio Compression Standard, Annex E 22 November 2010 produce the gain attenuation element corresponding to each DCT mantissa block identified in the bit stream. The switch position is also derived by the deformatter for each GAQ-coded mantissa. The switch position is determined from the presence or absence of a unique bit stream tag, as discussed in the next paragraph. When the deformatting operation is complete, the dequantized and level-adjusted mantissas are available for the next stage of processing. As a means for signaling the two mantissa lengths to the decoder, quantizer output symbols for large mantissas are flagged in the bit stream using a unique identifier tag. In Enhanced AC-3, the identifier tag is the quantizer symbol representing a full-scale negative output (e.g., the ‘100’ symbol for a 3-bit two’s complement quantizer). In a conventional mid-tread quantizer, this symbol is often deliberately unused since it results in an asymmetric quantizer characteristic. In gain-adaptive quantization, this symbol is employed to indicate the presence of a large mantissa. The tag length is equal to the length of the small mantissa codeword (computed from hebap[] and gaqgain[]), allowing unique bit stream decoding. If an identifier tag is found, additional bits immediately following the tag (also of known length) convey the quantizer output level for the corresponding large mantissas. Four different gain transmission modes are available for use in the encoder. The different modes employ switched 0, 1 or 1.67-bit gains. For each independent, coupling, and LFE channel in which AHT is in use, a 2-bit parameter called gaqmod is transmitted once per frame to the decoder. The bitstream parameters, values, and active hebap range are shown for each mode in Table E3.3. If gaqmod = 0x0, GAQ is not in use and no gains are present in the bitstream. If gaqmod = 0x1, a 1-bit gain value is present for each block of DCT coefficients having an hebap value between 8 and 11, inclusive. Coefficients with hebap higher than 11 are decoded using the same quantizer as gaqmod 0x0. If gaqmod = 0x2 or 0x3, gain values are present for each block of DCT coefficients having an hebap value between 8 and 16, inclusive. Coefficients with hebap higher than 16 are decoded using the same quantizer as gaqmod 0x0. The difference between the two last modes lies in the gain word length, as shown in Table E3.3. Table E3.3 Gain Adaptive Quantization Modes chgaqmod[ch], GAQ Mode for Frame Active hebap Range cplgaqmod, and (for which gains are transmitted) lfegaqmod 0x0 GAQ not in use None 0x1 1-bit gains (Gk = 1 or 2) 8 ≤ hebap ≤ 11 0x2 1-bit gains (Gk = 1 or 4) 8 ≤ hebap ≤ 16 0x3 1.67 bit gains (Gk = 1, 2, or 4) 8 ≤ hebap ≤ 16 For the case of gaqmod = 0x1 and 0x2, the gains are coded using binary 0 to signal Gk = 1, and binary 1 to signal Gk = 2 or 4. For the case of gaqmod = 0x3, the gains are composite-coded in triplets (three 3-state gains packed into 5-bit words). The gains are unpacked in a manner similar to exponent unpacking as described in the main body of this document. For example, for a 5-bit composite gain triplet grpgain: M1 = truncate (grpgain / 9) M2 = truncate ((grpgain % 9) / 3) M3 = (grpgain % 9) % 3 201 Advanced Television Systems Committee, Inc. Document A/52:2010 In this example, M1, M2, and M3 correspond to mapped values derived from consecutive gains in three ascending frequency blocks, respectively, each ranging in value from 0 to 2 inclusive as shown in Table E3.4. Table E3.4 Mapping of Gain Elements, gaqmod = 0x3 Gain, Gk Mapped Value 1 0 2 1 4 2 Details of the GAQ quantizer characteristics are shown in Table E3.5. If the received gain is 1, or no gain was received at all, a single quantizer with no tag is used. If the received gain is either 2 or 4, both the small and large mantissas (and associated tags) must be decoded using the quantizer characteristics shown. Both small and large mantissas are decoded by interpreting them as signed two’s complement fractional values. The variable m in the table represents the number of mantissa bits associated with a given hebap value as shown in Table E3.2. Table E3.5 Gain Adaptive Quantizer Characteristics Gk = 1 Gk = 2 Gk = 4 Quantizer Small Large Small Large Quantizer Quantizer Quantizer Quantizer Length of quantizer codeword m m–1 m–1 m–2 m Number of reconstruction 2m –1 2m–1 –1 2m–1 2m–2 –1 2m (output) points Step size 2/(2m – 1) 1/(2m–1) 1/(2m–1 – 1) 1/(2m–1) 3/(2m+1 – 2) Since the large mantissas are coded using a dead-zone quantizer, a post-processing step is required to transform (remap) large mantissa codewords received by the decoder into a reconstructed mantissa. This remapping is applied when Gk = 2 or 4. An identical post-processing step is required to implement a symmetric quantizer characteristic when Gk = 1, and for all gaqmod = 0x0 quantizers. The post-process is a computation of the form y = x + ax + b. In this equation, x represents a mantissa codeword (interpreted as a signed two’s complement fractional value), and the constants a and b are provided in Table E3.6. The constants are also interpreted as 16-bit signed two’s complement fractional values. The expression for y was arranged for implementation convenience so that all constants will have magnitude less than one. For decoders where this is not a concern, the remapping can be implemented as y = a’x + b, where the new coefficient a’ = 1 + a. The sign of x must be tested prior to retrieving b from the table. Remapping is not applicable to the table entries marked N/A. 202 Digital Audio Compression Standard, Annex E 22 November 2010 Table E3.6 Large Mantissa Inverse Quantization (Remapping) Constants hebap Gk = 1 Gk = 2 Gk = 4 a b a b a b 8 x≥0 0x1249 0x0000 0xd555 0x4000 0xedb7 0x2000 x<0 0x1249 0x0000 0xd555 0xeaab 0xedb7 0xfb6e 9 x≥0 0x0889 0x0000 0xc925 0x4000 0xe666 0x2000 x<0 0x0889 0x0000 0xc925 0xd249 0xe666 0xeccd 10 x≥0 0x0421 0x0000 0xc444 0x4000 0xe319 0x2000 x<0 0x0421 0x0000 0xc444 0xc889 0xe319 0xe632 11 x≥0 0x0208 0x0000 0xc211 0x4000 0xe186 0x2000 x<0 0x0208 0x0000 0xc211 0xc421 0xe186 0xe30c 12 x≥0 0x0102 0x0000 0xc104 0x4000 0xe0c2 0x2000 x<0 0x0102 0x0000 0xc104 0xc208 0xe0c2 0xe183 13 x≥0 0x0081 0x0000 0xc081 0x4000 0xe060 0x2000 x<0 0x0081 0x0000 0xc081 0xc102 0xe060 0xe0c1 14 x≥0 0x0040 0x0000 0xc040 0x4000 0xe030 0x2000 x<0 0x0040 0x0000 0xc040 0xc081 0xe030 0xe060 15 x≥0 0x0020 0x0000 0xc020 0x4000 0xe018 0x2000 x<0 0x0020 0x0000 0xc020 0xc040 0xe018 0xe030 16 x≥0 0x0010 0x0000 0xc010 0x4000 0xe00c 0x2000 x<0 0x0010 0x0000 0xc010 0xc020 0xe00c 0xe018 17 x≥0 0x0008 0x0000 N/A N/A N/A N/A x<0 0x0008 0x0000 N/A N/A N/A N/A 18 x≥0 0x0002 0x0000 N/A N/A N/A N/A x<0 0x0002 0x0000 N/A N/A N/A N/A 19 x≥0 0x0000 0x0000 N/A N/A N/A N/A x<0 0x0000 0x0000 N/A N/A N/A N/A E3.3.5 Transform Equations The AHT processing uses a DCT to achieve higher coding efficiency. Hence, if AHT is in use, the DCT must be inverted prior to applying the exponents. The inverse DCT (IDCT) for AHT is given in the following equation. Any fast technique may be used to invert the DCT in Enhanced AC-3 decoders. In the following equation, C(k,m) is the MDCT spectrum for the kth bin and mth block, and X(k,j) is the AHT spectrum for the kth bin and jth block. 5  j (2m + 1)π  C (k , m ) = 2  R j X (k , j ) cos  j =0  12  m = 0,1,...,5 Where  1 j≠0 Rj =  1 / 2 j=0 and k is the bin index, m is the block index, and j is the AHT transform index. 203 Advanced Television Systems Committee, Inc. Document A/52:2010 E3.4 Enhanced Channel Coupling E3.4.1 Overview Enhanced channel coupling is a spatial coding technique that elaborates on conventional channel coupling, principally by adding phase compensation, a de-correlation mechanism, variable time constants, and more compact amplitude representation. The intent is to reduce coupling cancellation artifacts in the encode process by adjusting inter-channel phase before downmixing, and to improve dimensionality of the reproduced signal by restoring the phase angles and degrees of correlation in the decoder. This also allows the process to be used at lower frequencies than conventional channel coupling. The decoder converts the enhanced coupling channel back into individual channels principally by applying an amplitude scaling and phase adjustment for each channel and frequency sub-band. Additional processing occurs when transients are indicated in one or more channels. E3.4.2 Sub-Band Structure for Enhanced Coupling Enhanced coupling transform coefficients are transmitted in exactly the same manner as conventional coupling. That is, coefficients are reconstructed from exponents and quantized mantissas. Transform coefficients # 13 through # 252 are grouped into 22 sub-bands of either 6 or 12 coefficients each, as shown in Table E3.7. The parameter ecplbegf is used to derive the value ecpl_start_subbnd which indicates the number of the enhanced coupling sub-band which is the first to be included in the enhanced coupling process. Below the frequency (or transform coefficient number) indicated by ecplbegf, all channels are independently coded. Above the frequency indicated by ecplbegf, channels included in the enhanced coupling process (chincpl[ch] = 1) share the common enhanced coupling channel up to the frequency (or tc #) indicated by ecplendf. The enhanced coupling channel is coded up to the frequency (or tc #) indicated by ecplendf, which is used to derive ecpl_end_subbnd. The value ecpl_end_subbnd is one greater than the last coupling sub- band which is coded. 204 Digital Audio Compression Standard, Annex E 22 November 2010 Table E3.7 Enhanced Coupling Sub-bands enhanced low tc # high tc # lf cutoff (kHz) hf cutoff (kHz) lf cutoff (kHz) hf cutoff (kHz) coupling @ fs=48 kHz @ fs=48 kHz @ fs=44.1 kHz @ fs=44.1 kHz sub-band # 0 13 18 1.17 1.73 1.08 1.59 1 19 24 1.73 2.30 1.59 2.11 2 25 30 2.30 2.86 2.11 2.63 3 31 36 2.86 3.42 2.63 3.14 4 37 48 3.42 4.55 3.14 4.18 5 49 60 4.55 5.67 4.18 5.21 6 61 72 5.67 6.80 5.21 6.24 7 73 84 6.80 7.92 6.24 7.28 8 85 96 7.92 9.05 7.28 8.31 9 97 108 9.05 10.17 8.31 9.35 10 109 120 10.17 11.30 9.35 10.38 11 121 132 11.30 12.42 10.38 11.41 12 133 144 12.42 13.55 11.41 12.45 13 145 156 13.55 14.67 12.45 13.48 14 157 168 14.67 15.80 13.48 14.51 15 169 180 15.80 16.92 14.51 15.55 16 181 192 16.92 18.05 15.55 16.58 17 193 204 18.05 19.17 16.58 17.61 18 205 216 19.17 20.30 17.61 18.65 19 217 228 20.30 21.42 18.65 19.68 20 229 240 21.42 22.55 19.68 20.71 21 241 252 22.55 23.67 20.71 21.75 Note: At 32 kHz sampling rate the sub-band frequency ranges are 2/3 the values of those for 48 kHz. 205 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E3.8 Enhanced Coupling Start and End Indexes ecpl sub-band # low tc # high tc # ecplbegf ecplendf 0 13 18 0 1 19 24 2 25 30 1 3 31 36 4 37 48 2 5 49 60 3 6 61 72 4 7 73 84 5 0 8 85 96 6 1 9 97 108 7 2 10 109 120 8 3 11 121 132 9 4 12 133 144 10 5 13 145 156 11 6 14 157 168 12 7 15 169 180 8 16 181 192 13 9 17 193 204 10 18 205 216 14 11 19 217 228 12 20 229 240 15 13 21 241 252 14 22 253 15 The enhanced coupling sub-bands are combined into enhanced coupling bands for which coupling coordinates are generated (and included in the bit stream). The coupling band structure is indicated by ecplbndstrc[sbnd]. Each bit of the ecplbndstrc[] array indicates whether the sub-band indicated by the index is combined into the previous (lower in frequency) enhanced coupling band. Enhanced coupling bands are thus made from integral numbers of enhanced coupling sub- bands. (See Section E2.3.3.19.) E3.4.3 Enhanced coupling tables The following tables are used to lookup various parameter values used by the enhanced coupling process. 206 Digital Audio Compression Standard, Annex E 22 November 2010 Table E3.9 Sub-band Transform Start Coefficients: ecplsubbndtab[] sbnd ecplsubbndtab[sbnd] 0 13 1 19 2 25 3 31 4 37 5 49 6 61 7 73 8 85 9 97 10 109 11 121 12 133 13 145 14 157 15 169 16 181 17 193 18 205 19 217 20 229 21 241 22 253 207 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E3.10 Amplitudes: ecplampexptab[], ecplampmanttab[] ecplamp ecplampexptab[ecplamp] ecplampmanttab[ecplamp] 0 0 0x20 1 0 0x1b 2 0 0x17 3 0 0x13 4 0 0x10 5 1 0x1b 6 1 0x17 7 1 0x13 8 1 0x10 9 2 0x1b 10 2 0x17 11 2 0x13 12 2 0x10 13 3 0x1b 14 3 0x17 15 3 0x13 16 3 0x10 17 4 0x1b 18 4 0x17 19 4 0x13 20 4 0x10 21 5 0x1b 22 5 0x17 23 5 0x13 24 5 0x10 25 6 0x1b 26 6 0x17 27 6 0x13 28 6 0x10 29 7 0x1b 30 7 0x17 31 - 0x00 208 Digital Audio Compression Standard, Annex E 22 November 2010 Table E3.11 Angles: ecplangletab[] ecplangl ecplangletab[ecplangle] ecplangle ecplangletab[ecplangle] e 0 0.00000 32 –1.00000 1 0.03125 33 –0.96875 2 0.06250 34 –0.93750 3 0.09375 35 –0.90625 4 0.12500 36 –0.87500 5 0.15625 37 –0.84375 6 0.18750 38 –0.81250 7 0.21875 39 –0.78125 8 0.25000 40 –0.75000 9 0.28125 41 –0.71875 10 0.31250 42 –0.68750 11 0.34375 43 –0.65625 12 0.37500 44 –0.62500 13 0.40625 45 –0.59375 14 0.43750 46 –.56250 15 0.46875 47 –0.53125 16 0.50000 48 –0.50000 17 0.53125 49 –0.46875 18 0.56250 50 –0.43750 19 0.59375 51 –0.40625 20 0.62500 52 –0.37500 21 0.65625 53 –0.34375 22 0.68750 54 –0.31250 23 0.71875 55 –0.28125 24 0.75000 56 –0.25000 25 0.78125 57 –0.21875 26 0.81250 58 –0.18750 27 0.84375 59 –0.15625 28 0.87500 60 –0.12500 29 0.90625 61 –0.09375 30 0.93750 62 –0.06250 31 0.96875 63 –0.03125 209 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E3.12 Chaos Scaling: ecplchaostab[] ecplchaos ecplchaostab[ecplchaos] 0 0.000000 1 –0.142857 2 –0.285714 3 –0.428571 4 –0.571429 5 –0.714286 6 –0.857143 7 –1.000000 E3.4.4 Enhanced Coupling Coordinate Format Enhanced coupling coordinates exist for each enhanced coupling band [bnd] in each channel [ch] which is coupled (chincp[ch]==1). Enhanced coupling coordinates are derived from three parameters; a 5-bit amplitude scaling value (ecplamp[ch][bnd]), a 6-bit phase angle value (ecplangle[ch][bnd]) and a 3-bit chaos measure (ecplchaos[ch][bnd]). These values will always be transmitted in the first block containing a coupled channel and are optionally transmitted in subsequent blocks, as indicated by the enhanced coupling parameter exists flags (ecplparam1e[ch] and ecplparam2e[ch]). If ecplparam1e[ch] or ecplparam2e[ch] are set to 0, corresponding coordinate values from the previous block are reused. The ecplamp values 0 to 30 represent gains between 0 dB and –45.01 dB quantized to increments of approximately 1.5 dB, and the value 31 represents minus infinity dB. The ecplangle values represent angles between 0 and 2π radians, quantized to increments of 2π/64 radians. The ecplchaos values each represent a scaling value between 0.0 and –1.0. E3.4.5 Enhanced Coupling Processing This section specifies the processing steps the reference decoder shall employ to recover transform coefficients for each coupled channel from the enhanced coupling data. The following steps are performed for each block. • Process the enhanced coupling channel • Prepare amplitudes for each channel and band • Prepare angles for each channel and band • Generate transform coefficients for each channel from the processed enhanced coupling channel, amplitudes and angles E3.4.5.1 Process Enhanced Coupling Channel This section assumes that the enhanced coupling channel mantissas and exponents have been extracted from the bitstream and have been denormalized into fixed point transform coefficients. Angle adjustment of the enhanced coupling channel requires that time domain aliasing not be present. Therefore the non-aliased enhanced coupling channel must be reconstructed using the enhanced coupling transform coefficients from the previous, current and next blocks. If enhanced coupling is not in use in the previous block, enhanced coupling transform coefficients for the previous block shall be set to zero. Likewise if enhanced coupling is not in use in the next block, enhanced coupling transform coefficients for the next block shall be set to zero. The following procedure describes how the non-aliased coupling channel is obtained. 210 Digital Audio Compression Standard, Annex E 22 November 2010 1. Define the MDCT transform coefficient buffers for the previous, current and next blocks (of length k = 0, 1,…,N/2–1 where N = 512) as: XPREV[k] = ecplmantPREV[k] where k = ecplstartmantPREV to ecplendmantPREV - 1 =0 elsewhere XCURR[k] = ecplmantCURR[k] where k = ecplstartmantCURR to ecplendmantCURR - 1 =0 elsewhere XNEXT[k] = ecplmantNEXT[k] where k = ecplstartmantNEXT to ecplendmantNEXT - 1 =0 elsewhere where ecplstartmant = ecplsubbndtab[ecplbegf] ecplendmant = ecplsubbndtab[ecplendf] 2. Compute the windowed time domain samples xPREV[n], xCURR[n] and xNEXT[n] using the 512- sample IMDCT (as described in steps 1 to 5 of Section 7.9.4.1 in the main body of this document). 3. Overlap and add the second half of the previous sample block and the first half of the next sample block with the current sample block as follows: Pseudo Code for (n=0; n> ecplampexptab[ecplamp[ch][bnd]]; } Modifications are made to the amplitude values using the transmitted chaos measure and transient parameter. Firstly, chaos values for each enhanced coupling band [bnd] in each channel [ch] are obtained from the ecplchaos parameters as follows. Pseudo Code if (ch == firstchincpl) { chaos[ch][bnd] = 0; } else { chaos[ch][bnd] = ecplchaostab[ecplchaos[ch][bnd]]; } The chaos modification is then performed as: Pseudo Code if ( (ecpltrans[ch] == 0) && (ch != firstchincpl) ) { amp[ch][bnd] *= 1 + 0.38 * chaos[ch][bnd]; } Using the ecplbndstrc[] array, the amplitude values amp[ch][bnd] which apply to enhanced coupling bands are converted to values which apply to enhanced coupling sub-bands amp[ch][sbnd] 212 Digital Audio Compression Standard, Annex E 22 November 2010 by duplicating values as indicated by values of ‘1’ in ecplbndstrc[]. Amplitude values for individual transform coefficients [bin] are then reconstructed as follows. Pseudo Code bnd = -1; for (sbnd=ecpl_start_sbnd; sbnd 1.0) angle_curr -= 2.0; while ((angle_prev – angle_curr) > 1.0) angle_curr += 2.0; slope = (angle_curr – angle_prev)/((nbins_curr + nbins_prev)/2.0); /* floating point calculation*/ / * do lower half of first band */ if ((bnd == 1) && (nbins_prev > 1)) { if (iseven(nbins_prev)) /* iseven() returns 1 if value is even, 0 if value is odd */ { y = angle_prev - slope/2; bin = nbins_prev/2 - 1; } else { y = angle_prev - slope; bin = (nbins_prev - 3)/2; } count = bin + 1; for (j = 0; j < count; j++) { ytmp = y; while (y > 1.0) y -= 2.0; while (y < (-1.0)) y += 2.0; angle[ch][bin--] = y; y = ytmp; y -= slope; } bin = count; } if (iseven(nbins_prev)) { y = angle_prev + slope/2; count = nbins_curr/2 + nbins_prev/2; /* integer calculation */ } else { y = angle_prev; count = nbins_curr/2 + (nbins_prev + 1)/2; /* integer calculation */ } 214 Digital Audio Compression Standard, Annex E 22 November 2010 for (j = 0; j < count; j++) { ytmp = y; while (y > 1.0) y -= 2.0; while (y < (-1.0)) y += 2.0; angle[ch][bin++] = y; y = ytmp; y += slope; } } /* Finish last band */ if (iseven(nbins_curr)) count = nbins_curr/2; /* integer calculation */ else count = nbins_curr/2 + 1; /* integer calculation */ for (j = 0; j < count; j++) { ytmp = y; while (y > 1.0) y -= 2.0; while (y < (-1.0)) y += 2.0; angle[ch][bin++] = y; y = ytmp; y += slope; } } To assist in de-correlating complex continuous signals, a scaled array of random values is added to each bin angle. The random values depend on whether or not a transient is present in the channel being processed as indicated by ecpltrans[ch]. For channels without a transient, the random values rand_notrans[ch][bin] have the following properties: • They are uniformly distributed between –1.0 and 1.0. • They must be unique for each bin [bin] and channel [ch]. • They must only be generated once (for example during decoder initialization) and must stay the same for every block of every frame. For channels with a transient, the random values rand_trans[ch][bnd] have the following properties: • They are uniformly distributed between –1.0 and 1.0. • They must be unique for each band [bnd] and channel [ch]. • New values must be generated for each block. Using the ecplbndstrc[] array, the banded values for chaos[ch][bnd] and for rand_trans[ch][bnd] are converted to individual bin values by duplicating the band values across each subband and then 215 Advanced Television Systems Committee, Inc. Document A/52:2010 across each bin within a subband. The chaos and random values are then used to modify each angle value as follows. Pseudo Code if (ecpltrans[ch] == 0) { rand[ch][bin] = rand_notrans[ch][bin] } else { rand[ch][bin] = rand_trans[ch][bin] } angle[ch][bin] += chaos[ch][bin] * rand[ch][bin]; if (angle[ch][bin] < -1.0) { angle[ch][bin] += 2.0; } else if(angle[ch][bin] >= 1.0) { angle[ch][bin] -= 2.0; } E3.4.5.4 Generate Channel Transform Coefficients Individual channel transform coefficients are then reconstructed from the coupling channel by computing the following complex products. Pseudo Code Zr[ch][bin] = Zr[bin] * amp[ch][bin] * cos(π * angle[ch][bin]) - Zi[bin] * amp[ch][bin] * sin(π * angle[ch][bin]); Zi[ch][bin] = Zi[bin] * amp[ch][bin] * cos(π * angle[ch][bin]) + Zr[bin] * amp[ch][bin] * sin(π * angle[ch][bin]); chmant[ch][bin] = -2 * ( y[bin] * Zr[ch][bin] + y[N/2-1-bin] * Zi[ch][bin] ); Where: Zr[bin] = real(Z[k]); Zi[bin] = imag(Z[k]); y[bin] = cos(2π * (N/4 + 0.5) / N * (k + 0.5)); for bin=k=0,1,…,N/2–1 E3.5 Spectral Extension Processing Enhanced AC-3 supports a new coding technique, based on high frequency regeneration, called spectral extension. This section contains a detailed description of the spectral extension process that the reference decoder shall implement. E3.5.1 Overview When spectral extension is in use, high frequency transform coefficients of the channels that are participating in spectral extension are synthesized. Transform coefficient synthesis involves 216 Digital Audio Compression Standard, Annex E 22 November 2010 copying low frequency transform coefficients, inserting them as high frequency transform coefficients, blending the inserted transform coefficients with pseudo-random noise, and scaling the blended transform coefficients to match the coarse (banded) spectral envelope of the original signal. To enable the decoder to scale the blended transform coefficients to match the spectral envelope of the original signal, scale factors are computed by the encoder and transmitted to the decoder on a banded basis for all channels participating in the spectral extension process. For a given channel and spectral extension band, the blended transform coefficients for that channel and band are multiplied by the scale factor associated with that channel and band. The spectral extension process is performed beginning at the spectral extension begin frequency, and ending at the spectral extension end frequency. The spectral extension begin frequency is derived from the spxbegf bit stream parameter. The spectral extension end frequency is derived from the spxendf bit stream parameter. In some cases, it may be desirable to use channel coupling for a mid-range portion of the frequency spectrum and spectral extension for the higher-range portion of the frequency spectrum. In this configuration, the highest coupled transform coefficient number must be 1 less than the lowest transform coefficient number generated by spectral extension. E3.5.2 Sub-Band Structure for Spectral Extension Transform coefficients #25 through #228 are grouped into 17 sub-bands of 12 coefficients each, as shown in Table E3.13. The final table entry does not represent an actual sub-band, but is included for the case when the spxendf parameter is 17. The spectral extension sub-bands containing transform coefficients #37 through #228 coincide with coupling sub-bands. The parameter spxbegf, derived from the bit stream parameter of the same name, indicates the number of the first spectral extension sub-band. The parameter spxendf, derived from the bit stream parameter of the same name, indicates a number one greater than the last spectral extension sub- band. From the sub-band indicated by spxbegf to the sub-band indicated by spxendf, transform coefficients are synthesized for all channels participating in the spectral extension process (chinspx[ch] == 1). Below the sub-band indicated by spxbegf, channels may be independently coded. Alternatively, channels may be coded independently below the coupling begin frequency, and coupled from the coupling begin frequency to the spectral extension begin frequency. Spectral extension sub-bands are combined into spectral extension bands for which spectral extension coordinates are generated (and included in the bit stream). Like channel coupling, each spectral extension band is made up of one or more consecutive spectral extension sub-bands. The number of spectral extension bands and the size of each band are determined from the spectral extension band structure array (spxbndstrc[]). Upon frame initialization, the default spectral extension banding structure is copied into the spxbndstrc[] array. If (spxbndstrce == 1), the spxbndstrc[sbnd] bit stream parameters are present in the bit stream and are used to fill the spxbndstrc[] array. If (spxbndstrce == 0), the existing values in the spxbndstrc[] array are used to compute the number of spectral extension bands and the size of each band. The following pseudo code indicates how to determine the number of spectral extension bands and the size of each band. 217 Advanced Television Systems Committee, Inc. Document A/52:2010 Pseudo Code nspxbnds = 1; spxbndsztab[0] = 12; for (bnd = spxbegf+1; bnd < spxendf; bnd ++) { if (spxbndstrc[bnd] == 0) { spxbndsztab[nspxbnds] = 12; nspxbnds++; } else { spxbndsztab[nspxbnds – 1] += 12; } } Table E3.13 Spectral Extension Band Table spx sub-band # low tc # high tc # spxbegf spxendf 0 25 36 1 37 48 2 49 60 0 3 61 72 1 4 73 84 2 5 85 96 3 0 6 97 108 4 1 7 109 120 5 2 8 121 132 9 133 144 6 3 10 145 156 11 157 168 7 4 12 169 180 13 181 192 5 14 193 204 15 205 216 6 16 217 228 17 229 7 E3.5.3 Spectral Extension Coordinate Format Spectral extension coordinates exist for each spectral extension band [bnd] of each channel [ch] that is using spectral extension (chinspx[ch] ==1). Spectral extension coordinates must be sent at least once per frame, and may be sent as often as once per block. The spxcoe[ch] bit stream parameter informs the decoder when spectral extension coordinates are present in the bit stream. If (spxcoe[ch] == 0), no spectral extension coordinates for channel [ch] are present in the bit stream, and the 218 Digital Audio Compression Standard, Annex E 22 November 2010 previous spectral extension coordinates should be reused. If (spxcoe[ch] == 1), spectral extension coordinates are present in the bit stream for channel [ch]. When present in the bit stream, spectral extension coordinates are transmitted in a floating point format. The exponent is sent as a 4-bit value (spxcoexp[ch][bnd]) indicating the number of right shifts which should be applied to the fractional mantissa value. The mantissas are sent as 2-bit values (spxcomant[ch][bnd]) which must be properly scaled before use. Mantissas are unsigned values so a sign bit is not used. Except for the limiting case where the exponent value = 15, the mantissa value is known to be between 0.5 and 1.0. Therefore, when the exponent value < 15, the msb of the mantissa is always equal to ‘1’ and is not transmitted; the next 2 bits of the mantissa are transmitted. This provides one additional bit of resolution. When the exponent value = 15 the mantissa value is generated by dividing the 2-bit value of spxcomant by 4. When the exponent value is < 15 the mantissa value is generated by adding 4 to the 2-bit value of spxcomant and then dividing the sum by 8. Spectral extension coordinate dynamic range is increased beyond what the 4-bit exponent can provide by the use of a per channel 2-bit master spectral extension coordinate (mstrspxco[ch]) which is used to scale all of the spectral extension coordinates within that channel. The exponent values for each channel are increased by 3 times the value of mstrspxco which applies to that channel. This increases the dynamic range of the spectral extension coordinates by an additional 54 dB. The following pseudo code indicates how to generate the spectral extension coordinate (spxco) for each spectral extension band [bnd] in each channel [ch]. Pseudo Code if (spxcoexp[ch][bnd] == 15) { spxco_temp[ch][bnd] = spxcomant[ch][bnd] / 4; } else { spxco_temp[ch][bnd] = (spxcomant[ch][bnd] + 4) / 8; } spxco[ch][bnd] = spxco_temp[ch][bnd] >> (spxcoexp[ch][bnd] + 3*mstrspxco[ch]); E3.5.4 High Frequency Transform Coefficient Synthesis This process synthesizes transform coefficients above the spectral extension begin frequency. The synthesis process consists of a number of different steps, described in the following sections. E3.5.4.1 Transform Coefficient Translation The first step of the high frequency transform coefficient synthesis process is transform coefficient translation. Transform coefficient translation consists of making copies of a channel’s low frequency transform coefficients and inserting them as the channel’s high frequency transform coefficients. The parameter spxstrtf, derived from the bit stream parameter of the same name, is used as the index into a table to determine the first transform coefficient to be copied. The parameter spxbegf, derived from the bit stream parameter of the same name, is used as the index into a table to determine the first transform coefficient to be inserted. The parameter spxendf, 219 Advanced Television Systems Committee, Inc. Document A/52:2010 derived from the bit stream parameter of the same name, is used as the index into a table to determine the last transform coefficient to be inserted. Transform coefficient translation is performed on a banded basis. For each spectral extension band, coefficients are copied sequentially starting with the transform coefficient at copyindex and ending with the transform coefficient at (copyindex + bandsize – 1). Transform coefficients are inserted sequentially starting with the transform coefficient at insertindex and ending with the transform coefficient at (insertindex + bandsize – 1). Prior to beginning the translation process for each band, the value of (copyindex + bandsize – 1) is compared to the copyendmant parameter. If (copyindex + bandsize – 1) is greater than or equal to the copyendmant parameter, the copyindex parameter is reset to the copystartmant parameter and wrapflag[bnd] is set to 1. Otherwise, wrapflag[bnd] is set to 0. The following pseudo code indicates how the spectral component translation process is carried out for channel [ch]. Pseudo Code copystartmant = spxbandtable[spxstrtf]; copyendmant = spxbandtable[spxbegf]; copyindex = copystartmant; insertindex = copyendmant; for (bnd = 0; bnd < nspxbnds; band++) { bandsize = spxbndsztab[bnd]; if ((copyindex + bandsize) > copyendmant) { copyindex = copystartmant; wrapflag[bnd] = 1; } else { wrapflag[bnd] = 0; } for (bin = 0; bin < bandsize; bin++) { if (copyindex == copyendmant) { copyindex = copystartmant; } tc[ch][insertindex] = tc[ch][copyindex]; insertindex++; copyindex++; } } 220 Digital Audio Compression Standard, Annex E 22 November 2010 E3.5.4.2 Transform Coefficient Noise Blending The next step of the high frequency transform coefficient synthesis process is transform coefficient noise blending. In this step, the translated transform coefficients are blended with pseudo-random noise in order to create a more natural sounding signal. E3.5.4.2.1 Blending Factor Calculation The first step of the transform coefficient noise blending process is to determine blending factors for the pseudo-random noise and the translated transform coefficients. The blending factor calculation for each band is based on both the spxblend bit stream parameter and the frequency mid-point of the band. This enables unique blending factors to be computed for each band from a single bit stream parameter. Because the spxblnd parameter exists in the bit stream only when new spectral extension coordinates exist in the bit stream, the blending factors can be reused for all blocks in which spectral extension coordinates are reused. The following pseudo code indicates how the blending factors for a channel [ch] are determined. Pseudo Code noffset[ch] = spxblend[ch] / 32.0; spxmant = spxbandtable[spxbegf]; if (spxcoe[ch]) { for (bnd = 0; bnd < nspxbnds; bnd++) { bandsize = spxbndsztab[bnd]; nratio = ((spxmant + 0.5*bandsize) / spxbandtable[spxendf]) – noffset[ch]; if (nratio < 0.0) { nratio = 0.0; } else if (nratio > 1.0) { nratio = 1.0; } nblendfact[ch][bnd] = squareroot(nratio); sblendfact[ch][bnd] = squareroot(1 – nratio); spxmant += bandsize; } } E3.5.4.2.2 Banded RMS Energy Calculation The next step is to compute the banded RMS energy of the translated transform coefficients. The banded RMS energy measures are needed to properly scale the pseudo-random noise samples prior to blending. 221 Advanced Television Systems Committee, Inc. Document A/52:2010 The following pseudo code indicates how to compute the banded RMS energy of the translated transform coefficients for channel [ch]. Pseudo Code spxmant = spxbandtab[spxbegf]; for (bnd = 0; bnd < nspxbnds; bnd++) { bandsize = spxbndsztab[bnd]; accum = 0; for (bin = 0; bin < bandsize; bin++) { accum = accum + (tc[ch][spxmant] * tc[ch][spxmant]); spxmant++; } rmsenergy[ch][band] = squareroot(accum / bandsize); } E3.5.4.2.3 Transform Coefficient Band Border Filtering When spectral extension attenuation is enabled for channel [ch], a notch filter is applied to the transform coefficients surrounding the border between the baseband and extension region. The filter is symmetric about the first bin of the extension region, and covers a total of 5 bins. The first 3 attenuation values of the filter are determined by lookup into Table E3.14 with index spxattencod[ch]. The last two attenuation values of the filter are determined by symmetry and are not explicitly stored in the table. The filter is also applied to the transform coefficients surrounding each border between bands where wrapping occurs during the transform coefficient translation operation, as indicated by wrapflag[bnd]. It is important that filtering occurs after the transform coefficient translation and banded RMS energy calculation but prior to the noise scaling and transform coefficient blending calculation. The following pseudo code demonstrates the application of the notch filter at the border between the baseband and extension region and all wrap points for each channel [ch]. 222 Digital Audio Compression Standard, Annex E 22 November 2010 Pseudo Code if (chinspxatten[ch]) { /* apply notch filter at baseband / extension region border */ filtbin = spxbandtable[spxbegf] - 2; for (bin = 0; bin < 3; bin++) { tc[ch][filtbin] *= spxattentab[spxattencod[ch]][binindex]; filtbin++; } for (bin = 1; bin >= 0; bin--) { tc[ch][filtbin] *= spxattentab[spxattencod[ch]][binindex]; filtbin++; } filtbin += spxbndsztab[0]; /* apply notch at all other wrap points */ for (bnd = 1; bnd < nspxbnds; bnd++) { if (wrapflag[bnd])/* wrapflag[bnd] set during transform coefficient translation */ { filtbin = filtbin – 5; for (binindex = 0; binindex < 3; bin++) { tc[ch][filtbin] *= spxattentab[spxattencod[ch]][binindex]; filtbin++; } for (bin = 1; bin >= 0; bin--) { tc[ch][filtbin] *= spxattentab[spxattencod[ch]][binindex]; filtbin++; } } filtbin += spxbndsztab[bnd]; } } 223 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E3.14 Spectral Extension Attenuation Table: spxattentab[][] spxattencod binindex 0 1 2 0 0.954841604 0.911722489 0.870550563 1 0.911722489 0.831237896 0.757858283 2 0.870550563 0.757858283 0.659753955 3 0.831237896 0.690956440 0.574349177 4 0.793700526 0.629960525 0.500000000 5 0.757858283 0.574349177 0.435275282 6 0.723634619 0.523647061 0.378929142 7 0.690956440 0.477420802 0.329876978 8 0.659753955 0.435275282 0.287174589 9 0.629960525 0.396850263 0.250000000 10 0.601512518 0.361817309 0.217637641 11 0.574349177 0.329876978 0.189464571 12 0.548412490 0.300756259 0.164938489 13 0.523647061 0.274206245 0.143587294 14 0.500000000 0.250000000 0.125000000 15 0.477420802 0.227930622 0.108818820 16 0.455861244 0.207809474 0.094732285 17 0.435275282 0.189464571 0.082469244 18 0.415618948 0.172739110 0.071793647 19 0.396850263 0.157490131 0.062500000 20 0.378929142 0.143587294 0.054409410 21 0.361817309 0.130911765 0.047366143 22 0.345478220 0.119355200 0.041234622 23 0.329876978 0.108818820 0.035896824 24 0.314980262 0.099212566 0.031250000 25 0.300756259 0.090454327 0.027204705 26 0.287174589 0.082469244 0.023683071 27 0.274206245 0.075189065 0.020617311 28 0.261823531 0.068551561 0.017948412 29 0.250000000 0.062500000 0.015625000 30 0.238710401 0.056982656 0.013602353 31 0.227930622 0.051952369 0.011841536 E3.5.4.2.4 Noise Scaling and Transform Coefficient Blending Calculation In order to properly blend the translated transform coefficients with pseudo-random noise, the noise components for each band must be scaled to match the energy of the translated transform coefficients in the band. The energy matching can be achieved by scaling all the noise components in a given band by the RMS energy of the translated transform coefficients in that band, provided the noise components are generated by a zero-mean, unity-variance noise generator. Once the zero-mean, unity-variance noise components for each band have been scaled by the RMS energy for that band, the scaled noise components can be blended with the translated transform coefficients. 224 Digital Audio Compression Standard, Annex E 22 November 2010 The following pseudo code indicates how the translated transform coefficients and pseudo- random noise for a channel [ch] are blended. The function noise() returns a pseudo-random number generated from a zero-mean, unity-variance noise generator. Pseudo Code spxmant = spxbandtable[spxbegf]; for (bnd = 0; bnd < nspxbnds; bnd++) { bandsize = spxbndsztab[bnd]; nscale = rmsenergy[ch][bnd] * nblendfact[ch][bnd]; sscale = sblendfact[ch][bnd]; for (bin = 0; bin < bandsize; bin++) { tctemp = tc[ch][spxmant]; ntemp = noise(); tc[ch][spxmant] = tctemp * sscale + ntemp * nscale; spxmant++; } } E3.5.4.3 Blended Transform Coefficient Scaling The final step of the high frequency transform coefficient synthesis process is blended transform coefficient scaling. In this step, blended transform coefficients are scaled by the spectral extension coordinates to form the final synthesized high frequency transform coefficients. After this step, the banded energy of the synthesized high frequency transform coefficients should match the banded energy of the high frequency transform coefficients of the original signal. The blended transform coefficient scaling process for channel [ch] is shown in the following pseudo code. Pseudo Code spxmant = spxbandtable[spxbegf]; for (bnd = 0; bnd < nspxbnds; bnd++) { bandsize = spxbndsztab[bnd]; spxcotemp = spxco[ch][bnd]; for (bin = 0; bin < bandsize; bin++) { tctemp = tc[ch][spxmant]; tc[ch][spxmant] = tctemp * spxcotemp * 32; spxmant++; } } 225 Advanced Television Systems Committee, Inc. Document A/52:2010 E3.6 Transient Pre-Noise Processing Transient pre-noise processing is a new audio coding improvement technique, which reduces the duration of pre-noise introduced by low-bit rate audio coding of transient material. This section contains a detailed description of transient pre-noise processing that the reference decoder shall implement. E3.6.1 Overview When transient pre-noise processing is used, decoded PCM audio located prior to transient material is used to overwrite the transient pre-noise, thereby improving the perceived quality of low-bit rate audio coded transient material. To enable the decoder to efficiently perform transient pre-noise processing with minimal decoding complexity, transient location detection and time scaling synthesis analysis is performed by the encoder and the information transmitted to the decoder. The encoder performs transient pre-noise processing for each full bandwidth audio channel and transmits the information once per frame. The transmitted transient location and time scaling synthesis information are relative to the first decoded PCM sample contained in the audio frame containing the bit stream information. It should be noted that it is possible for the time scaling synthesis parameters contained in audio frame N, to reference PCM samples and transients located in audio frame N + 1, but this does not create a requirement for multi-frame decoding. E3.6.2 Application of Transient Pre-Noise Processing Data The bit stream syntax and high level description of the transient pre-noise parameters contained in the audio frame field are outlined in Sections E2.2.3 and E2.3.2, respectively. The parameter transproce indicates whether any of the full bandwidth channels in the current audio frame have associated transient pre-noise time scaling synthesis processing information. If transproce is set to a value of ‘1’, then the parameter chintransproc[ch] can be set for each full bandwidth channel. For each full bandwidth channel where chintransproc[ch] is set to a value of ‘1’, the transient location parameter transprocloc[ch] and time scaling length parameter transproclen[ch] are each set to values that have been calculated by the encoder. Figure E3.2 provides an overview of how the transient pre-noise parameters that are computed and transmitted by the encoder are applied in the decoder. As shown in Figure E3.2a, the parameter transprocloc[ch] identifies the location of the transient relative to the first sample of decoded PCM channel data in the audio frame that contains the transient pre-noise processing parameters. As defined, transprocloc[ch] has four sample resolution to reduce the data rate required to transmit the transient location and must be multiplied by 4 to get the location of the transient in samples. As also shown in Figure E3.2a, the parameter transproclen[ch] provides the time scaling length, in samples, relative to the leading edge of the audio coding block prior to the block in which the transient is located. As shown in Figure E3.2b, the location of the leading edge of the audio coding block prior to the block containing the transient indicates the start of the transient pre-noise. The start of the previous audio coding block and location of the transient provide the total length of the transient pre-noise in samples, PN. As part of the normal decoding operation, the decoder inherently knows the starting location of the audio coding block that contains the transient and this does not need to be transmitted. Also shown in Figure E3.2b is how the time scaling synthesis audio buffer, which is used to modify the transient pre-noise, is defined relative to the decoded audio frame. The time scaling synthesis buffer is (2*TC1 + PN) PCM samples in length, where TC1 is a time scaling synthesis 226 Digital Audio Compression Standard, Annex E 22 November 2010 Decoded PCM Audio 4* transprocloc[ch] samples transproclen[ch] a) samples First PCM sample Audio coding block Location of from Decoded frame leading edge Transient b) Synthesis buffer = (2*TC1 + PN samples) First PCM sample Audio coding block Location of from Decoded frame leading edge Transient Pre-noise = PN samples trtransproclen[ch] + PN + TC1 c) Synthesis buffer First PCM sample TC1 TC2 from Decoded frame samples samples Figure E3.2 Transient pre-noise time scaling synthesis summary. system parameter equal to 256 samples. The first sample of the time scaling synthesis buffer is located (2*TC1 + 2*PN) samples before the location of the transient. Figure E3.2c outlines how the time scaling synthesis buffer is used along with the transproclen[ch] parameter to remove the transient pre-noise. As shown in Figure E3.2c the original decoded audio data is cross-faded with the time scaling synthesis buffer starting at the sample located (PN + TC1 + transproclen[ch]) samples before the location of the transient. The length of the cross-fade is TC1 or 256 samples. Nearly any pair of constant amplitude cross-fade windows may be used to perform the overlap-add between the original data and the synthesis buffer, although standard Hanning windows have been shown to provide good results. The time scaling synthesis buffer is then used to overwrite the decoded PCM audio data that is located before the transient, including the transient pre-noise. This overwriting continues until TC2 samples before the transient where TC2 is another time scaling synthesis system parameter equal to 128 samples. At TC2 samples before the transient, the time scaling synthesis audio buffer is cross-faded with the original decoded PCM data using a set of constant amplitude cross-fade windows. The following pseudo code outlines how to implement the transient pre-noise time scaling synthesis functionality in the decoder for a single full bandwidth channel, [ch]. Where: win_fade_out1 = TC1 sample length cross-fade out window (unity to zero in value) win_fade_in1 = TC1 sample length cross-fade in window (zero to unity in value) 227 Advanced Television Systems Committee, Inc. Document A/52:2010 win_fade_out2 = TC2 sample length cross-fade out window (unity to zero in value) win_fade_in2 = TC2 sample length cross-fade in window (zero to unity in value) Pseudo Code /* unpack the transient location relative to first decoded pcm sample. */ transloc = transprocloc[ch]; /* unpack time scaling length relative to first decoded pcm sample. */ translen = transproclen[ch]; /* compute the transient pre-noise length using audio coding block first sample, aud_blk_samp_loc. */ pnlen = (transloc – aud_blk_samp_loc); /* compute the total number of samples corrected in the output buffer. */ tot_corr_len = (pnlen + translen + TC1); /* create time scaling synthesis buffer from decoded output pcm buffer, pcm_out[ ]. */ for (samp = 0; samp < (2*TC1 + pnlen); samp++) synth_buf[samp] = pcm_out[(transloc – (2*TC + 2*pnlen) + samp)]; end /* use time scaling synthesis buffer to overwrite and correct pre-noise in output pcm buffer. */ start_samp = (transloc – tot_corr_len); for (samp = 0; samp < TC1; samp++) { pcm_out[start_samp + samp] =(pcm_out[start_samp + samp] * win_fade_out1[samp]) + (synth_buf[samp] * win_fade_in1[samp]); } for (samp = TC1; samp < (tot_corr_len – TC2); samp++) { pcm_out[start_samp + samp] = synth_buf[samp]; } for (samp = (tot_corr_len – TC2); samp < tot_corr_len; samp++) { pcm_out[start_samp + samp] =(pcm_out[start_samp + samp] * win_fade_in2[samp]) + (synth_buf[samp] * win_fade_out2[samp]); } E3.7 Channel and Program Extensions The Enhanced AC-3 bit stream syntax allows for time-multiplexed substreams to be present in a single bit stream. By allowing time-multiplexed substreams, the Enhanced AC-3 bit stream syntax enables a single program with greater than 5.1 channels, multiple programs of up to 5.1 channels, or a mixture of programs with up to 5.1 channels and programs with greater than 5.1 channels, to be carried in a single bit stream. E3.7.1 Overview An Enhanced AC-3 bit stream must consist of at least one independently decodable stream (type 0 or 2). Optionally, Enhanced AC-3 bit streams may consist of multiple independent substreams (type 0 or 2) or a combination of multiple independent (type 0 and 2) and multiple dependent (type 1) substreams. 228 Digital Audio Compression Standard, Annex E 22 November 2010 Figure E3.3 Bitstream with a single program of greater than 5.1 channels. The reference enhanced AC-3 decoder must be able to decode independent substream 0, and skip over any additional independent and dependent substreams present in the bit stream. Optionally, Enhanced AC-3 decoders may use the information present in the acmod, lfeon, strmtyp, substreamid, chanmape, and chanmap bit stream parameters to decode bit streams with a single program with greater than 5.1 channels, multiple programs of up to 5.1 channels, or a mixture of programs with up to 5.1 channels and programs with greater than 5.1 channels. E3.7.2 Decoding a Single Program with Greater than 5.1 Channels When a bit stream contains a single program with greater than 5.1 channels, independent substream 0 contains a 5.1 channel downmix of the program for compatibility with playback systems containing 5.1 speakers. The audio in independent substream 0 can also be downmixed for compatibility with playback systems containing less than 5.1 speakers. Decoders reproducing 5.1 or fewer channels from a program containing greater than 5.1 channels shall decode only independent substream 0 and skip all associated dependent substreams. In order to accommodate playback by systems with greater than 5.1 speakers, the Enhanced AC-3 bit stream will carry one or more dependent substreams that contain channels that either replace or supplement the 5.1 channel data carried in independent substream 0. (See Figure E3.3.) If the chanmape parameter of a dependent substream is set to 0, then the acmod and lfeon parameters of the dependent substream are used to identify the channels present in the dependent substream, and the corresponding audio channels in the independent substream are overwritten with the dependent audio channel data. For example, if the dependent substream uses acmod 1/0 (center channel only) and has lfeon set to 1, then the center channel audio data carried in the dependent stream will replace the center channel audio data carried in the independent stream, and the LFE audio data carried in the dependent stream will replace the LFE data carried in the independent stream. If the chanmape parameter of a dependent substream is set to 1, then the chanmap parameter is used to determine the channel mapping for all channels contained in the dependent stream. Each bit of the chanmap parameter corresponds to a particular channel location. Audio data is contained in the dependent substream for each chanmap bit that is set to 1. The order of the coded channels in the dependent substream is the same as the order of the bits set to 1 in the chanmap parameter. For example, if the Left channel bit is set to 1 in the channel map field, then Left channel audio data will be contained in the first coded channel of data in the dependent substream. If channels are present in the dependent substream that correspond to channels in the associated independent substream, then the dependent substream data for those channels replaces the independent substream data for the corresponding channels. All channels present in the dependent substream that do not correspond to channels in the independent substream are used to enable output for speaker configurations with greater than 5.1 channels. The maximum number of channels rendered for a single program is 14. 229 Advanced Television Systems Committee, Inc. Document A/52:2010 Figure E3.4 Bitstream with multiple programs of up to 5.1 channels. E3.7.3 Decoding Multiple Programs with up to 5.1 Channels When an Enhanced AC-3 bit stream contains multiple independent substreams, each independent substream corresponds to an independent audio program. The application interface may inform the decoder which independent audio program should be decoded by selecting a specific independent substream ID. The decoder should then only decode substreams with the desired independent substream ID, and skip over any other programs present in the bit stream with different substream ID’s. The default program selection should always be Program 1. In some cases, it may be desirable to decode multiple independent audio programs. In these cases, the application interface should inform the decoder which independent audio programs to decode by selecting specific independent substream ID’s. The decoder should then decode all substreams with the desired independent substream ID’s, and skip over any other programs present in the bit stream with different substream ID’s. (See Figure E3.4.) E3.7.4 Decoding a Mixture of Programs with up to 5.1 Channels and Programs with Greater than 5.1 Channels When an Enhanced AC-3 bit stream contains multiple independent and dependent substreams, each independent substream and its associated dependent substreams correspond to an independent audio program. The application interface may inform the decoder which independent audio program should be decoded by selecting a specific independent substream ID. The decoder should then only decode the desired independent substream and all its associated dependent substreams, and skip over all other independent substreams and their associated dependent substreams. If the selected independent audio program contains greater than 5.1 channels, the decoder should decode the selected independent audio program as explained in Section E3.7.2. The default program selection should always be Program 1. In some cases, it may be desirable to decode multiple independent audio programs. In these cases, the application interface should inform the decoder which independent audio programs to decode by selecting specific independent substream ID’s. The decoder should then decode the desired independent substreams and their associated dependent substreams, and skip over all other independent substreams and associated dependent substreams present in the bit stream. (See Figure E3.5.) E3.7.5 Dynamic Range Compression for Programs Containing Greater than 5.1 Channels A program using channel extensions to convey greater than 5.1 channels may require two different sets of compr and dynrng metadata words: one set for the 5.1 channel downmix carried by independent substream 0 and a separate set for the complete (greater than 5.1 channel) mix. If a decoder is reproducing the complete mix, the compr and dynrng metadata words carried in independent substream 0 shall be ignored. The decoder shall instead use the compr and dynrng metadata words carried by the associated dependent substream. If multiple associated dependent substreams are present, only the last dependent substream may carry compr and dynrng metadata 230 Digital Audio Compression Standard, Annex E 22 November 2010 Figure E3.5 Bitstream with mixture of programs of up to 5.1 channels and programs of greater than 5.1 channels. words, and these metadata words shall apply to all substreams in the program, including the independent substream. The compre bit is used by the decoder to determine which dependent substream in a program is the last dependent substream of the program. Therefore, the compre bit in the last dependent substream of a program must be set to 1, and the compre bit in all other dependent substreams of the program must be set to 0. Additionally, the compr2e, dynrnge, and dynrng2e bits for all but the last dependent substream of a program must be set to 0. The compr2e, dynrnge, and dynrng2e bits for the last dependent substream shall be set as required to transmit the proper compr2, dynrng, and dynrng2 words for the program. Note that the compr2e, compr2, dynrng2e, and dynrng2 metadata words are only present in the bit stream when acmod = 0. E3.8 LFE Downmixing Decoder Description For decoders with only 2-channel or mono outputs, where a dedicated LFE/subwoofer output is not available, Enhanced AC-3 enables the LFE channel audio to be mixed into the Left and Right channels at a level indicated by the LFE mix level code bit stream parameter. LFE downmixing occurs only if the LFE mix level code parameter is present in the bit stream and the decoder is operating in 1/0 (C only) or 2/0 (L/R) output modes with the LFE channel output disabled. For all other output modes, the LFE mixing information, if present, is ignored. Note that lfemixlevcode should be assumed to be 0 when it is not transmitted in the bit stream. For the 1/0 case, the decoder should perform a standard 2/0 downmix with the LFE mixed into the Left and Right channels, followed by a subsequent mix of the L/R channels to a mono C channel. The following pseudo code indicates how the decoder should perform the LFE downmix. Pseudo Code if (output mode == 1/0 or 2/0) && (lfeoutput == disabled) && (lfemixlevcode == 1)) { mix LFE into left with (LFE mix level - 4.5) dB gain mix LFE into right with (LFE mix level - 4.5) dB gain } if (output mode == 1/0) { mix left into center with -6 dB gain mix right into center with -6 dB gain } 231 Advanced Television Systems Committee, Inc. Document A/52:2010 E4. AHT VECTOR QUANTIZATION TABLES Table E4.1 VQ Table for hebap 1; 16-bit two’s complement Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16- (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s bit two’s complement) complement) complement) complement) complement) complement) 0 0x1bff 0x1283 0x0452 0x10ad 0x28ac 0x12d4 1 0xe9ba 0xf38d 0xc76d 0xfa90 0xf815 0x0351 2 0x0279 0x1837 0x1b61 0xce15 0xf6fe 0xf5b4 3 0xfa44 0xe489 0x1da8 0x2979 0xe8c6 0xf40a Table E4.2 VQ Table for hebap 2; 16-bit two’s complement Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16- (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s bit two’s complement) complement) complement) complement) complement) complement) 0 0xd0d7 0x0260 0xe495 0x024e 0x0fa0 0x0365 1 0x1a24 0x3d49 0xe7de 0xdbe9 0xffb6 0x0085 2 0x073f 0xfc23 0x5074 0xf498 0xee85 0x00e1 3 0xfb56 0xf0c3 0xfccb 0xe65a 0xfc95 0xb0b6 4 0xf536 0xf393 0xf002 0xea09 0xbdcf 0x2625 5 0x060b 0x1ab7 0x07bc 0x4f09 0xfbd1 0xec86 6 0x184d 0xba05 0xea74 0x187a 0x0166 0x048a 7 0x0ea9 0xfbd6 0x10bb 0xf365 0x3e38 0x27ca Table E4.3 VQ Table for hebap 3; 16-bit two’s complement Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16- (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s bit two’s complement) complement) complement) complement) complement) complement) 0 0xd8d4 0x512b 0x2ae6 0xee30 0x031e 0xffbc 1 0x2b2a 0x500a 0xe627 0xeb22 0xf8fb 0xf9a1 2 0x0f89 0xfde2 0x1bce 0xfb72 0x499c 0x3956 3 0xef20 0xffa0 0xe381 0xfe14 0xa9de 0xef4b 4 0x0a84 0x16e0 0x159a 0x5566 0xe3d4 0xeb33 5 0xff79 0xa4a1 0x03c2 0x1fb3 0xfd7c 0x017e 6 0xf9e5 0x0d48 0xf31d 0x1255 0xe514 0x577e 7 0x0dcf 0x0bd6 0x1c80 0x1846 0x4ffc 0xd0bd 8 0x0039 0xe559 0x0738 0xa8b3 0xe8e1 0x1aa7 9 0xfccb 0xf1b9 0xfe7d 0xe793 0xf939 0xa89b 10 0xe862 0x0632 0xb636 0xc7c8 0x23fe 0x02c1 11 0xe9ac 0x0108 0xb9d4 0x391a 0x1ef1 0xfeaf 12 0xff92 0x006c 0x0008 0x004a 0xffa7 0xffce 13 0x19d4 0xfa13 0x54b7 0xf986 0xe0f3 0xff0a 14 0x54a3 0xe741 0xdf9e 0xff9b 0xfabb 0xffea 15 0xaa0d 0xe6b4 0x1f26 0x0288 0x0806 0xfeb5 232 Digital Audio Compression Standard, Annex E 22 November 2010 Table E4.4 VQ Table for hebap 4; 16-bit two’s complement index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 0 0x5903 0x15c0 0xe9e6 0xff64 0xfe06 0xffdf 1 0x19ec 0xee0f 0x375d 0xbc6f 0xbf75 0x0360 2 0x0e4a 0x580c 0x0068 0xf91d 0xffac 0x0006 3 0x544c 0xba69 0xe38e 0xf9d9 0xf7e2 0xfec0 4 0xf747 0x2721 0xf558 0x3a5a 0xcab8 0xbb05 5 0xf9e4 0xbab6 0xb527 0x35a7 0x0ac5 0x0b87 6 0x11a8 0x1586 0x1ce1 0x2a2f 0x4b1f 0xca36 7 0x018f 0x0ba0 0xfbb5 0x1395 0xfb79 0x564f 8 0x0e28 0xf6c9 0x1248 0xf742 0x58ae 0x0eb5 9 0xef97 0xdfa3 0xe566 0xcf9a 0xb812 0x3c16 10 0xebb2 0xe52b 0xd8c1 0xdf54 0xc16a 0xafae 11 0xff72 0xa771 0xfe90 0x1127 0xfe30 0xfff3 12 0x032e 0xfba2 0xfbbf 0xa9fd 0x004a 0x0611 13 0xf9ae 0x4b16 0xbb16 0xcb4e 0x034a 0xf6fb 14 0x1251 0x406a 0x514d 0xc3e5 0xefbc 0xf080 15 0xf314 0x2bce 0xcb1a 0x351f 0xb3ef 0x35ca 16 0x0719 0x0356 0x52e9 0xfc3a 0xf995 0xfef4 17 0xf5e5 0xff95 0xb146 0x0178 0x0496 0xfed0 18 0xf499 0x01c5 0xeaf2 0x02ee 0xa9ee 0xfc2e 19 0xb5bc 0x41c7 0x2710 0xf204 0x08a3 0x05b3 20 0x0553 0xf59e 0xffdf 0xf01d 0x048d 0xaa1f 21 0xde70 0xf538 0xbb90 0xc18f 0x3a31 0x052b 22 0x028c 0xdb8d 0x0cb5 0xc6e2 0x2f95 0x4cec 23 0xe727 0x168d 0xc3dd 0x438b 0x40ce 0xf496 24 0xfd6b 0xfda7 0x0649 0x5852 0x03e0 0xfbeb 25 0x1361 0x2393 0x2bd9 0x1e95 0x3fc0 0x48c3 26 0xaa90 0xfa67 0x008a 0x05be 0xf89d 0xff3c 27 0xb3d5 0xb8e5 0x2b30 0xfdfc 0x09ef 0xf737 28 0xfb54 0xbb5a 0x4eb6 0x2cc6 0xfe6f 0x0a3b 29 0x121e 0xe026 0x2e73 0xc271 0x44cf 0xc595 30 0xffad 0x0116 0x0143 0x0037 0xff66 0x00e8 31 0x1e6c 0x05b6 0x47db 0x3bc0 0xc26d 0xfb95 Table E4.5 VQ Table for hebap 5; 16-bit two’s complement Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 0 0xf2be 0xb2ee 0x0b93 0x2576 0x1234 0x4cd9 1 0xc2cf 0xe6fb 0x4507 0x0f14 0xdfd8 0xb41b 2 0x1173 0x019c 0xba2f 0xe09b 0x02b3 0xbc65 3 0x0dfc 0x093b 0x1ae6 0x0eb3 0x18eb 0xafd6 233 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E4.5 VQ Table for hebap 5; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 4 0xbcb2 0xc8cb 0xfa8c 0xa27d 0x20b5 0xcf07 5 0xe077 0xac23 0xc2ea 0x0c8e 0x1fa9 0xe8b3 6 0x10f7 0x1431 0x0a7b 0xbe4a 0xebe6 0xbf52 7 0x18cc 0xd654 0x32c3 0x9c64 0xa9b6 0x0ffb 8 0xf4c0 0xdf52 0xe8b0 0xbcfa 0xf5b2 0x5a5c 9 0xec19 0xc837 0xa89d 0x54ed 0x0e69 0x0b91 10 0xf675 0xbab5 0x6243 0x0a93 0x063a 0x0007 11 0xb835 0x2332 0x10ae 0x02db 0xfe56 0xfd80 12 0xa371 0x609c 0x160a 0x0264 0xfecc 0xfc3c 13 0xfd01 0x04f4 0x00e1 0x0663 0x00ad 0x0394 14 0x154f 0x195d 0x1326 0x2940 0x5a01 0xbd0c 15 0x4343 0xadc2 0xb6e4 0x1348 0xf2a4 0x0d1d 16 0xfa92 0x3c80 0xaa46 0xc6ed 0x053b 0x021e 17 0xe52e 0xf732 0xd0da 0xf3fd 0xb1f3 0xaf72 18 0xf8f5 0x2dff 0x053f 0x22d5 0x02b5 0x5fb1 19 0xab96 0x24f6 0x1249 0x2426 0xe179 0x3e20 20 0xea49 0xf436 0xdc2f 0xfabd 0xa7ed 0x3244 21 0xfe92 0x13d4 0xf941 0x4fcb 0xfee5 0xf495 22 0xf8a2 0xe757 0xfc55 0xf7df 0xfa89 0x0db9 23 0xf3a7 0xfde7 0xec2d 0x2c04 0x4bc4 0x03dd 24 0x0929 0x1039 0x1689 0xef4f 0x00e9 0xfe71 25 0xaa7a 0xfb8e 0xbfa6 0x170e 0x1570 0xf375 26 0x2717 0xcf0e 0x498d 0x51c4 0xfb7a 0x06fe 27 0xfb73 0x1396 0xfb51 0x190f 0xdf1e 0xadd2 28 0x0764 0xf232 0x0ee7 0xe92a 0x402b 0x4f40 29 0xf598 0xd295 0xefcd 0xb879 0xa74a 0x3a00 30 0x4368 0x28b3 0x1e54 0x2f08 0x4a0c 0x09dd 31 0xac55 0xb703 0xd56f 0x1110 0xe475 0x11bb 32 0xf9da 0x0802 0x1680 0x60b4 0x3e6f 0x450e 33 0xfde6 0xa6ad 0x2b3b 0x283d 0x0181 0x0210 34 0xdeef 0xf42f 0xc01b 0xa53b 0x406b 0x0e46 35 0x16d0 0x023f 0x2e72 0x079b 0x6245 0x19fd 36 0x19e1 0xf244 0xf854 0x0f0a 0xfe7a 0xff8c 37 0x4655 0x51a4 0x37f3 0xe23b 0xd556 0x2e1a 38 0xed07 0xf48c 0xcbea 0xe179 0x5476 0x08db 39 0xfdbd 0xdb29 0xfd14 0xacb7 0x304f 0x2049 40 0xdf83 0x055f 0xba49 0x0b69 0x2366 0x561e 41 0x47de 0x21bb 0xfa21 0xf68e 0xb889 0xc672 42 0xf455 0x3b19 0xf2fd 0x571c 0x3636 0xcab9 43 0x16f2 0xb5ae 0x3ce3 0x2c56 0xaefe 0x07b3 44 0x062d 0xe4d5 0xac40 0x0997 0x0041 0x019e 234 Digital Audio Compression Standard, Annex E 22 November 2010 Table E4.5 VQ Table for hebap 5; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 45 0x0203 0xee8c 0xfd67 0xedc0 0x007d 0xb4ea 46 0x53f7 0xb0b3 0xf8b0 0xf87a 0xff2d 0xfc02 47 0x1445 0xd026 0xf911 0xa402 0xee3e 0x16b5 48 0x0141 0xe745 0x3936 0x1b3f 0xf913 0x0363 49 0xca0a 0x0c6c 0x1ef7 0x01bc 0x4c60 0x0c4a 50 0xe5fc 0x2fdc 0xf84c 0x4400 0xa128 0xcd64 51 0xfd17 0x3814 0xfbad 0x5cbe 0xda61 0xb858 52 0x476c 0xe11b 0xe295 0x4aae 0x1e29 0xce8d 53 0x0786 0x3afd 0xcdd0 0x0869 0x547f 0x0748 54 0xf7ae 0x5b78 0x42a0 0xc313 0xf9f8 0x0057 55 0x207a 0xd1d0 0x38f5 0xaf91 0x1ed3 0xf7cd 56 0x4c90 0x591e 0xbc68 0xf808 0x011d 0xf0e9 57 0xdfea 0xb86e 0x29e4 0xca50 0xcb63 0xf97e 58 0x380f 0x1310 0xb1be 0x03c4 0xef83 0xff4c 59 0x9fea 0xbf05 0x4d0c 0x1725 0x12a9 0x113e 60 0xf641 0x5bc5 0xc0f3 0x0b66 0xfbf2 0xf826 61 0x4a1e 0xf614 0x341f 0x057d 0xe7ce 0xfb90 62 0x09b9 0x3566 0x586e 0xe371 0xff7f 0xf518 63 0xc976 0x4187 0x59e4 0x02d8 0x0d45 0x00a2 64 0x0bde 0x03e1 0x2246 0xaa2f 0xe62f 0x038e 65 0xcf64 0xa88e 0xf5be 0xeb51 0x4c40 0x2690 66 0xf889 0xb89e 0xb7b6 0xc58e 0x1298 0x1bcf 67 0x206a 0xf45e 0x651e 0x1dec 0xe127 0x03fc 68 0x17f4 0x3b17 0x4945 0xa0ce 0xe67f 0xe61d 69 0x1ef4 0x2f5d 0xdb0d 0xa266 0x157e 0x03a9 70 0xbd60 0xeb03 0x09da 0x0147 0x0469 0xfe7a 71 0x3d9e 0x4df3 0xd774 0x2ba4 0xf3dd 0x3a05 72 0xd180 0xe065 0xb9f8 0xa8f1 0xbcab 0xe56d 73 0xcdc2 0xf784 0xe693 0x1727 0x30aa 0xa8ad 74 0xfe0f 0x0142 0x040e 0xe60d 0xeae4 0x4f57 75 0x043b 0xa638 0xded2 0x2f62 0xfd06 0x0a3f 76 0x13cb 0x4d00 0xf893 0xffe2 0xfebb 0x0055 77 0x03db 0xe93a 0x1074 0xdcba 0x23a1 0x9e32 78 0xe144 0x1c74 0xcffc 0x3272 0xaba8 0x51cd 79 0xf9a2 0xe1f2 0xf775 0xdeb3 0xea1c 0x9d94 80 0xe5f4 0x0184 0xa7f9 0x05f6 0x237a 0x00c1 81 0xe145 0xa8dc 0x142b 0x016a 0x03b0 0xfefd 82 0x0ef0 0xd1b6 0x1da7 0xa578 0x62fe 0x5cdb 83 0xd6f8 0x101b 0xad89 0x52b5 0x57a7 0xfcba 84 0xed8d 0x5523 0x1828 0xff86 0x066a 0xfd33 85 0x5fb8 0x4daf 0xf805 0x03da 0x0007 0xffc9 235 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E4.5 VQ Table for hebap 5; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 86 0x954f 0xff79 0x0985 0x0103 0x0059 0x0133 87 0x5f7e 0xf0df 0xeaf1 0xfccc 0xf6ad 0x0169 88 0x1599 0x1698 0x48fa 0x00f2 0xaa78 0xf05d 89 0x5720 0x1183 0x02d2 0xd02e 0x1d92 0x3c58 90 0x21e1 0x0bc1 0x4fd5 0x5274 0xad94 0xf3f8 91 0xfb94 0x0a91 0xf8df 0x152c 0xfcef 0x4864 92 0x4224 0xcb33 0xbf83 0xc5f6 0xb099 0xc873 93 0x08ab 0x0564 0x53e2 0xfb98 0x0147 0x0053 94 0xf77f 0x540d 0xf0f0 0xc89c 0xff34 0xf771 95 0x03b9 0xdb2e 0x3e02 0xd62a 0xf361 0x5226 96 0xfe5b 0xfa9f 0x0280 0xdfd1 0xae10 0x087e 97 0x10d5 0x4852 0xdc74 0xb871 0xc362 0x0e78 98 0xe8c9 0x01c1 0xdf3d 0x0433 0xa93e 0xec80 99 0x0b89 0x31f4 0x476d 0x0596 0x3a59 0x54e3 100 0xf49f 0x0191 0xed7d 0xb177 0x06a3 0xfb85 101 0x0d79 0x1479 0x2295 0x5676 0xe285 0x05ab 102 0xf796 0x2188 0x46c8 0xc302 0x4b77 0xe899 103 0x0dad 0x0b19 0x1709 0x18fd 0x21b6 0x59ea 104 0x09a3 0x0b8c 0x017b 0x1647 0xa9e1 0xf773 105 0xbdbd 0xfdae 0x4986 0xeb51 0x0668 0x0306 106 0x0b50 0xfa70 0x0e02 0xf70c 0x4dc6 0xf8e2 107 0xb771 0x52e3 0xc94f 0xcee3 0x4052 0x027b 108 0xf832 0xb48e 0xbf71 0x2fb0 0xbf40 0xe152 109 0xda36 0x03f4 0xab73 0x0b43 0xce58 0x0911 110 0xfc13 0x01d7 0xf1d3 0x1f6d 0xd4b1 0x63bd 111 0x102d 0xac20 0xf58f 0x02f4 0xfd69 0xfdf5 112 0x195a 0x2153 0x4b59 0x4a05 0x17cc 0xdb7d 113 0x4245 0x6017 0x36c8 0x2758 0xfde8 0xd73a 114 0xe02d 0x0861 0xa60c 0xbd4f 0x154b 0xeecf 115 0xc5e7 0x5028 0xb881 0xda0b 0xd193 0xba59 116 0xf70e 0xc8d8 0x0816 0x57c3 0x0687 0x02d5 117 0xdea6 0x3925 0x0dc1 0xaf9f 0x1a11 0x2008 118 0x4f18 0x113a 0xfaaa 0xfdb7 0x04cd 0xf66f 119 0x1d2b 0xe414 0x3563 0xdfca 0x5778 0xbc58 120 0xf874 0x0f23 0xdc98 0xf11c 0x03be 0x0109 121 0xeed1 0x0b8f 0xc1d9 0x4c8e 0x135a 0xfbaf 122 0x4659 0xd93d 0xb927 0xf0ea 0x2baa 0x16bd 123 0xc6fc 0xfb35 0x25bc 0x5473 0x2bdc 0xd237 124 0xfd2f 0xf95c 0x006d 0xf7a2 0x003d 0xe58c 125 0x9fd5 0xa808 0x15e8 0xf85b 0xf91f 0xfc0c 126 0xa350 0xee9d 0xf580 0xc6a9 0xef56 0x26bf 236 Digital Audio Compression Standard, Annex E 22 November 2010 Table E4.5 VQ Table for hebap 5; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 127 0x212f 0xfc82 0x4fd6 0xca04 0xbc8d 0x008b Table E4.6 VQ Table for hebap 6; 16-bit two’s complement Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 0 0x27aa 0x1cc5 0x41dd 0x48f9 0xa693 0xf1cc 1 0xf5c5 0xf134 0xea67 0xebb8 0xdccf 0xb0b6 2 0xea31 0xa6f0 0x5331 0x1b64 0x02e9 0x02d0 3 0x01ac 0xfa4d 0x006d 0xf3f6 0x0169 0xdf2d 4 0x1fe1 0x5781 0x00f1 0x06db 0xfc96 0xf4f8 5 0x0474 0x3163 0x0902 0x56f7 0x9dc6 0xbb6b 6 0xf5cf 0x0d33 0x2861 0xb2ee 0xc394 0xa278 7 0xf038 0xce04 0x9b54 0x3328 0x0f41 0x0523 8 0x1210 0xa309 0x3528 0x62e5 0xfb69 0x087d 9 0xff9f 0x35b3 0xebfe 0x5ad7 0x1076 0xa97f 10 0x1ade 0xfebe 0x4758 0xfcaa 0xd174 0xfd23 11 0x4380 0xce83 0xda23 0x5c0b 0xc090 0xfae3 12 0x16aa 0xec98 0x4c46 0xad36 0x9fd2 0xff49 13 0x16db 0xc0f7 0x3b7d 0xdae8 0xf9fe 0x0179 14 0x3710 0x61e1 0x346b 0x2062 0x5b18 0x41b2 15 0xe3a3 0x0076 0xc205 0x4a99 0x2635 0xfeeb 16 0xef40 0x5455 0xcc18 0xc07d 0x40f9 0xed02 17 0x132d 0xb4ef 0x5b73 0x3971 0xfd2e 0x007d 18 0x4c06 0xed84 0xf878 0xd2f9 0x5122 0x1531 19 0x9456 0xf4bf 0xef15 0x0180 0xf7c9 0x0557 20 0xfef6 0xdc29 0x1541 0x66dd 0xf87c 0x107d 21 0xf466 0xb136 0xaac8 0x154a 0xe2fe 0x14e0 22 0xff23 0xe5d8 0x025b 0xdc4c 0x051c 0x948e 23 0x2595 0xdf44 0xf851 0x24bb 0xf98d 0x5921 24 0x1d8e 0xeb7e 0xefbb 0x0569 0xfc22 0x0230 25 0xfb12 0x60a2 0xb58f 0x29f5 0x1da1 0xe446 26 0x01c3 0x4ea2 0xd923 0xe881 0xf774 0xfa4e 27 0x56e9 0x24a4 0x2388 0x2acf 0xf6c3 0xf174 28 0x48ec 0xfd76 0xfb2e 0x2b54 0x1dfe 0x1751 29 0x4b07 0xfa33 0xfbcc 0xfd25 0xfd54 0x002b 30 0xec93 0x3476 0x4eab 0x003c 0x01dc 0xfc59 31 0xb1c3 0x2206 0x09c3 0x03f8 0xfb7a 0x014f 32 0x98d3 0x48a6 0xf767 0xfd63 0x0d51 0x0319 33 0xed8a 0x22ab 0x9fe1 0xda52 0x0e3b 0xfee5 34 0x33f7 0xac64 0xf195 0xfb60 0xf84e 0x064c 237 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E4.6 VQ Table for hebap 6; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 35 0x00ad 0x003c 0x0397 0x04cd 0x1b1e 0xfd67 36 0x3ff9 0x425f 0x14dd 0xc941 0xf700 0xb05a 37 0x62f6 0xd68f 0x2eab 0xe21b 0xe725 0x36ea 38 0x5d79 0xcc35 0xe3c6 0xfc57 0x00ea 0xff45 39 0x18a7 0xf8ab 0x30da 0xf8a9 0x493f 0xa4d3 40 0x026d 0x192d 0x0d1a 0xa12e 0x20d6 0x14c3 41 0xf31f 0xec56 0xeda0 0xec28 0x9b7e 0x14e3 42 0xfb05 0xcc11 0xfc3b 0xa4ea 0x04be 0x6693 43 0xe794 0x2733 0xb177 0x3bc5 0xc137 0x1410 44 0x255a 0xf0b9 0xb3ca 0x1289 0x56fe 0xefb5 45 0x1f2a 0xb370 0x36c8 0xe98f 0xae89 0x22eb 46 0x0007 0xf039 0x03df 0xe84f 0x0034 0xb421 47 0x0d9d 0x0b99 0x1e34 0x1e6a 0x62e0 0x183e 48 0xfc41 0xcdf4 0xf8d0 0xa729 0x1c9c 0x2a4e 49 0xedb2 0x068e 0xd844 0xebab 0x10c6 0xfb09 50 0x0f31 0x0516 0x1d1a 0x027e 0x4f96 0xf3c3 51 0xcf30 0xdc5d 0x481f 0xcfc9 0xe3ba 0x4878 52 0xe7d7 0x21c9 0xe509 0xfc81 0x42d5 0x40dc 53 0xd958 0x6fa3 0x0b1d 0x0668 0x0b6d 0xfed6 54 0x3a78 0x9a7c 0x3a1e 0xa234 0x0717 0xe6b6 55 0x65fb 0x142e 0x52e9 0x3e01 0x5471 0x39e9 56 0xab4c 0x4036 0x5018 0xc7f6 0xe436 0xefbe 57 0x6fe7 0x005a 0xf9dc 0x0315 0xfc7a 0xffb5 58 0xfa39 0x09a7 0xf023 0x0e1c 0xf740 0x2aa2 59 0x21a8 0x4453 0x4367 0xbbd0 0x427e 0xc01b 60 0xaf0e 0xb75b 0x62ba 0x4538 0xf20b 0x069f 61 0xfc1b 0x17f1 0xe761 0x2bf2 0xd3a1 0xb2e5 62 0xffb6 0xf05f 0xf9d0 0x3448 0x00a2 0xff70 63 0xfdef 0x524c 0x1ef3 0xd37c 0x01a6 0xffe6 64 0x1bbe 0xcb25 0xb1a9 0x0a45 0xff4e 0xfe53 65 0x23f1 0x0558 0xa922 0x0a3f 0xafed 0x6139 66 0xfe50 0x1a13 0xfef6 0x2213 0x0050 0x6d78 67 0x4c25 0xf3dc 0xdbd3 0x0776 0xaaef 0x14e1 68 0x36ff 0xd31f 0x313c 0x17bf 0x4da5 0x0523 69 0x2ac3 0x266d 0xb74c 0x3d7e 0x12b8 0x025d 70 0xf90f 0x0eae 0xf009 0x54c0 0x1788 0xfdc0 71 0x0def 0xf206 0x3ffb 0x0a78 0xf928 0x02cc 72 0xec47 0xfa89 0xee3a 0xfd74 0xbac7 0xf2da 73 0xf1cd 0xeeec 0xe686 0xa978 0x1cd6 0x05b2 74 0x2fd2 0x4af6 0x160e 0xe179 0xb0bf 0x5360 75 0xe2ac 0x4df0 0x5bf6 0xd9e7 0x1625 0xf83a 238 Digital Audio Compression Standard, Annex E 22 November 2010 Table E4.6 VQ Table for hebap 6; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 76 0xf71d 0x3c4e 0x2a9b 0xba29 0x1961 0x350e 77 0xc1ea 0xc2e2 0xed94 0x1783 0x5eba 0xe7dd 78 0xf7ff 0xe538 0xfb48 0x0396 0x4547 0xffbb 79 0xf177 0x238b 0xc13f 0xa3bb 0x175d 0xf6d8 80 0x1eb6 0xdd2a 0x5de1 0x63a4 0xd4e7 0xfd1b 81 0xced4 0x4c0c 0x3939 0x3c5b 0xad16 0x0493 82 0x0836 0x047b 0x0ae5 0x1000 0x0883 0x222e 83 0xb8da 0xbaa2 0xd782 0xebad 0xfbd6 0xf22b 84 0xf4fd 0xb20a 0xd16f 0x1790 0x207b 0x2886 85 0xdc8a 0xf7cc 0x4be7 0xffef 0x02dc 0xfd4f 86 0xc750 0xb4e8 0xe449 0x4927 0x074e 0x597a 87 0x0f48 0x0293 0x63fd 0xf05a 0x2593 0x036d 88 0x0a38 0x58a7 0xe976 0x4600 0x0ee4 0x4efc 89 0x0a01 0x68df 0xeb83 0xd564 0x08d0 0xfdfb 90 0xec92 0x00c6 0xaa21 0xf1e8 0x569e 0xb614 91 0x533c 0xfb45 0x4ac8 0x4133 0xf9cc 0x2c7e 92 0xf902 0x0f77 0xf260 0x1b5b 0xe43d 0x518d 93 0xe824 0xb9dd 0xb6de 0x60bb 0x407c 0x0c8b 94 0x4fee 0x9e65 0xb077 0x1184 0xec09 0xfe22 95 0xe716 0xf832 0xd80b 0xfdcf 0xa9e9 0xa8bd 96 0x0be7 0xb65e 0x1da2 0x3997 0xb26a 0x18cf 97 0xec49 0x057d 0xda38 0x041f 0xaa87 0x2ba2 98 0x0d99 0xda1d 0x197e 0xbef1 0x591d 0x5593 99 0xb776 0x445d 0x3948 0x050b 0x13a2 0x4cdc 100 0x3f06 0xb29e 0xbdc4 0xb9ed 0xbddb 0x16a8 101 0xdfe0 0x132c 0x22e7 0x08e0 0xfb8c 0xa54f 102 0x0624 0x0ac1 0xf9c2 0x085f 0xf2ee 0xaa5a 103 0xd998 0xfbdc 0x9356 0x04be 0x1c79 0x0096 104 0x0062 0x0602 0x0217 0x4415 0xa562 0xfc7b 105 0x535c 0xb14e 0x0ce1 0xf930 0xdff1 0xac2a 106 0xefba 0xede7 0xba12 0x1566 0x0505 0x0088 107 0x4919 0x520b 0x60f2 0x2c9d 0x0502 0xedf6 108 0xf231 0x1dd4 0xfef7 0x085d 0xfcc3 0xf80d 109 0xf390 0x4d01 0x0ad7 0xfffe 0x0442 0x0068 110 0xe58d 0xb127 0x0b7a 0xf7b3 0xffdc 0x04f4 111 0x2558 0x24d6 0x2572 0x5654 0x3603 0x1898 112 0xfde9 0xb1ce 0x10b4 0xf8b4 0xfe40 0xbce1 113 0xa0e0 0x37a4 0xcab1 0xadd0 0x08df 0x2d23 114 0xf5aa 0x3c4d 0xee13 0x48ce 0xef35 0xfd92 115 0xb1a0 0x1049 0x46c3 0xfa84 0x359a 0xf8df 116 0xc019 0x2378 0x02e8 0x5605 0x007d 0x2a2a 239 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E4.6 VQ Table for hebap 6; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 117 0x25ac 0xc6f1 0xb7d1 0xc686 0x2ba6 0xaeee 118 0xfeba 0xa32e 0x1800 0x1ee5 0x025a 0x0604 119 0xe606 0x19ea 0xce75 0x5394 0x5131 0xe549 120 0x109c 0xadcd 0x15fc 0x48ff 0x5d34 0x2088 121 0x4642 0x1648 0xeb83 0xb953 0xfdd5 0x0c93 122 0x17cb 0x3798 0xec03 0xbbd0 0xb404 0xd27f 123 0xabae 0x2c26 0x3c4a 0x63f6 0x1a79 0x97c5 124 0x536b 0xdfcc 0x16f5 0xf22c 0x17bf 0xf5f9 125 0x0a2b 0xf669 0x152d 0xd002 0xb564 0x15c6 126 0xf947 0x98e7 0xa390 0x5978 0xfea3 0x0ecb 127 0x088d 0xfb4d 0x14dc 0x0cb1 0xa7a7 0x0068 128 0xf980 0xd4f4 0xf4d7 0xaf0d 0xa20f 0x4dbc 129 0x5959 0xe34f 0xb7cf 0xc6e8 0xdf30 0xcd5b 130 0x0ec1 0x0f76 0x202f 0x500e 0xe4b1 0xfb4f 131 0xff60 0xf9b3 0xfce7 0xde17 0x023d 0x0308 132 0x10c9 0xf136 0x4f95 0x17c2 0xeb37 0xb820 133 0x4939 0x099f 0x3102 0xe1bb 0xe1ca 0xf779 134 0x2b42 0xed90 0x5667 0x0721 0xa0e1 0x0ff0 135 0x05df 0xb516 0xf9df 0x000d 0xfec7 0x0177 136 0x013e 0xfdc1 0x09f0 0x00b2 0x0066 0x0028 137 0xc184 0x96ef 0x1390 0x0cf8 0x02ae 0x0487 138 0x649b 0x6906 0x023e 0xe8d6 0xf0b4 0x057f 139 0xdc44 0xe20f 0xf4c5 0xdf40 0xb719 0x6720 140 0xe2eb 0xb988 0xb824 0x2262 0xf734 0xaa82 141 0x1eab 0x2dfd 0x6b5d 0xcdd1 0xfa7e 0x4c86 142 0x08c0 0x173b 0x2bef 0x3e6c 0xe69d 0x5ed8 143 0x54a9 0xb7ad 0x262b 0x1996 0xf556 0x014e 144 0xefcb 0x0628 0xd4fe 0x0059 0xa093 0xe9b2 145 0x1e28 0x05c6 0x53a4 0x9e3f 0xdf3f 0x0009 146 0xf670 0x27ea 0xce2c 0xc131 0x0489 0xacdc 147 0xddcb 0xc7a3 0xa67a 0xc624 0x0a45 0x3614 148 0xe3ac 0x0b1b 0xda59 0x0b42 0xc6df 0x5fb1 149 0xfd5e 0xe67e 0x019e 0xa4db 0xaca1 0x01c6 150 0x0838 0xe758 0x2a87 0x46a7 0xfb51 0x00af 151 0xfe13 0xfdce 0xf54d 0x0076 0xfbce 0x005d 152 0xd8e5 0xf015 0x9259 0x56a4 0x3ae5 0xfd84 153 0xede3 0xbfe8 0xdcd5 0xb03e 0xd2a8 0xae3c 154 0x12cf 0x3e14 0x5eae 0xcabe 0xf3fe 0xfbdd 155 0xe5bc 0x1202 0xb6ac 0xc44d 0xbed3 0x5db4 156 0x3bf5 0xfd5e 0xf19e 0x54af 0x117b 0xd0c8 157 0x1294 0x0a21 0x14ea 0x1771 0x3ad7 0x677a 240 Digital Audio Compression Standard, Annex E 22 November 2010 Table E4.6 VQ Table for hebap 6; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 158 0xa2f9 0xbc9d 0x1b20 0x017a 0x02b6 0x029e 159 0x5b60 0xdd79 0xc687 0x1d78 0xfc94 0x2b50 160 0x0e38 0x0d08 0x5841 0xf259 0xf6e8 0xff8f 161 0x011c 0x1b02 0x0c19 0x27bb 0x19ee 0xb743 162 0x09a8 0x1758 0x2b2e 0xd160 0xfda5 0xfd69 163 0x3f2f 0x4039 0x336c 0xf035 0x123b 0x1d07 164 0x4b8a 0x3cae 0xe67b 0x0691 0xed07 0x4298 165 0x4283 0x0214 0xb588 0xfa5f 0xebf6 0x043d 166 0xceb7 0xbb37 0x080e 0x9d0c 0x4a41 0xc107 167 0x2748 0xadf8 0xcabe 0xf47b 0x3c07 0x4dde 168 0xfd78 0xf9bb 0x273e 0xf9c8 0x33f0 0x4d60 169 0xfbe2 0x29f8 0x021a 0x616a 0x259e 0xdca4 170 0xd88d 0x0be2 0x9e0c 0xa20c 0x3693 0x0064 171 0x1993 0x1afb 0x1b77 0x286c 0x5cdf 0xba22 172 0xa6f7 0xf840 0xfa8f 0xf2fe 0x2433 0x37ed 173 0xc7f6 0xf081 0x0be2 0x3f7e 0xbc69 0x25ae 174 0xac6f 0x5c4c 0x4185 0x02cc 0x0a67 0x0072 175 0xb5b8 0xf422 0x0626 0xff0b 0x05b7 0xfce7 176 0x578a 0x5b91 0xc6d3 0xfdee 0x439e 0x3531 177 0xd2c2 0x1eff 0xc97e 0x5ba9 0x9fcc 0x67b6 178 0xfbeb 0x0e5f 0xf756 0x294c 0x5207 0xf18a 179 0xc367 0x00c5 0x414e 0x9fe5 0x1351 0x0005 180 0x2a1d 0x10ef 0x68a6 0xdc9d 0xc0e8 0xf4e8 181 0x3ecb 0xa1dc 0xf0a3 0xe54f 0x3165 0xe48b 182 0x0830 0x9c1c 0xdf4e 0x1a9e 0x000b 0x049a 183 0xd1b8 0xfdb9 0xdd47 0xafc1 0xd719 0xfe84 184 0xf649 0x60c9 0xab79 0xb473 0x067c 0xfd24 185 0x0909 0x356f 0x0ff5 0x5fe5 0x6073 0xad45 186 0xf6c2 0xfe08 0xefde 0xd6b6 0x5c74 0x07a9 187 0x4f9b 0x4591 0xdade 0x0e95 0xb5f6 0xe76c 188 0xf0f0 0x41a2 0xfc5f 0xb0aa 0xbab5 0x1a8d 189 0x308f 0x17be 0xd3f8 0xc78e 0x1b01 0x5bb4 190 0x1dd4 0xf989 0x59e9 0x29df 0xdf9c 0x0346 191 0xde91 0xfb2d 0xb950 0x0f39 0x3edd 0x05d2 192 0xf1fe 0x2054 0x3b3d 0xf131 0xad63 0x06cd 193 0xee6f 0x54eb 0x093e 0xfeea 0xed48 0x3cbd 194 0xa5ae 0xca74 0x1df4 0x3f68 0x5e38 0x3ab1 195 0xb1b5 0x3215 0xb140 0x4133 0xd279 0xc12f 196 0xcec7 0x4f0f 0x0da8 0xf60b 0xe5a7 0xd1b6 197 0x1159 0x1e84 0x512f 0x42b8 0x2d03 0xda55 198 0x60be 0x212e 0xa4fe 0xf342 0x2b5d 0xe430 241 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E4.6 VQ Table for hebap 6; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 199 0xd885 0xe239 0xa978 0xb881 0x6815 0x254e 200 0x9c33 0x01dd 0x1ec2 0xf9fe 0x0463 0xff58 201 0x01d6 0x266a 0xfea5 0x5d89 0xd773 0xdb05 202 0xf000 0xda1a 0xe538 0xabd8 0x516d 0x1c06 203 0x14fa 0x2614 0xa32b 0xfb5a 0x0200 0xf9fe 204 0xfc12 0xd8c2 0xce97 0x4b22 0xf902 0xfc86 205 0x3b04 0x5c44 0xc2e2 0xf626 0xfb4d 0xfad3 206 0xe312 0xf5d3 0x0447 0xff09 0xfe27 0x00b1 207 0x1f99 0x0004 0x3088 0xa8f4 0x28a5 0xe1d0 208 0x56b4 0x2a17 0xec4d 0x02b2 0x0216 0xff2c 209 0xf3af 0xfa76 0xbe3d 0x47fa 0x3dcd 0x59ac 210 0x1631 0xf74b 0x0c7c 0xf2aa 0xaac7 0xc629 211 0x0013 0x0313 0x0408 0x00aa 0xdf99 0xfd7b 212 0xfc8e 0xf6f1 0x961f 0x01b0 0xeed8 0x05db 213 0xfab6 0xd1d5 0xffb4 0xb064 0xd7cb 0x2c40 214 0x00d3 0xed6f 0xedbd 0xe4eb 0xcb1e 0x388f 215 0x179b 0x148c 0xfe35 0xfe32 0x008f 0xffbf 216 0xf5f4 0x1c58 0xf30b 0x23fc 0xa570 0xd8fa 217 0x9ece 0xdac4 0x49ba 0x17d5 0x097d 0xc76e 218 0x207a 0x08e5 0x3770 0x0db8 0x6519 0x55f0 219 0x00d0 0x4efa 0xfee7 0x9f36 0xffc1 0xfb61 220 0x0447 0xe86e 0x0a92 0xaa51 0xf5a1 0x0233 221 0x0017 0xe8d6 0x00f3 0xdce3 0x14e1 0x504e 222 0xc396 0x319b 0x1040 0x2b4f 0x508d 0xd750 223 0x5203 0xffab 0xdeec 0x00c2 0x03eb 0xdad5 224 0xb34b 0xf2f9 0xc8ff 0x0df6 0xa4ab 0xfd65 225 0xf7e4 0x0da1 0xf388 0xb459 0x021b 0xfa06 226 0x1cb8 0xc493 0x5844 0x4ba9 0x0413 0x40f3 227 0xf8b0 0xfe63 0x04d3 0xeb64 0xf222 0x558f 228 0x1efb 0xf828 0x4248 0xe571 0x72d1 0xf655 229 0xcaeb 0x20c5 0xa3ac 0xa9b5 0xc89e 0xc827 230 0xd2c9 0xb186 0x3eb8 0xf8c8 0x3d69 0x1194 231 0x0f09 0xbf3b 0x4ec1 0xad5d 0x1e62 0x2e58 232 0xe66d 0xfb07 0xb66b 0xd42e 0x2d74 0x0414 233 0x09e0 0xe5dd 0xba03 0xd39e 0xece2 0xfc10 234 0x04d9 0x10a4 0x090f 0x17df 0x0d9d 0x4ef1 235 0x0bc6 0xf418 0x14c4 0xee45 0x515f 0x21fe 236 0xf902 0xc6a5 0x0116 0x3684 0xd8af 0xd6cd 237 0xa734 0xe0eb 0xfb7e 0x35fd 0xfa34 0xfb21 238 0xe36b 0xfd99 0x3326 0x49ef 0x26a9 0x05ac 239 0x09f8 0xf6de 0x0d60 0xedea 0x2b74 0xb380 242 Digital Audio Compression Standard, Annex E 22 November 2010 Table E4.6 VQ Table for hebap 6; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 240 0xd48b 0xafb7 0xd599 0xd5e1 0xaee1 0x1ab1 241 0x03d8 0xc509 0x168f 0x6225 0x1501 0xb2a9 242 0x0205 0x33d8 0xe2de 0xf951 0x5084 0xe883 243 0xac57 0x33c3 0xaec5 0x3489 0x4381 0x3330 244 0xc23d 0xc088 0x5a35 0xf03b 0xdffd 0x0367 245 0x0246 0x311b 0xad77 0xc652 0xdc1d 0x1635 246 0x10de 0xf910 0x2ca1 0xba9d 0xd93f 0x0241 247 0x177d 0x41be 0x44f7 0x9b5a 0xeed0 0xf222 248 0xca50 0xbf63 0x0e34 0xf2fe 0xad9d 0xc1f2 249 0x19a5 0xd475 0x21c9 0xccc6 0x5b31 0xcb03 250 0xf612 0xdcaa 0xe27a 0x7238 0x0e75 0xfe81 251 0xd68c 0x61a3 0x0765 0xdfee 0x51b8 0xc0ae 252 0x149c 0x4156 0x29a3 0x4de4 0xed41 0xb484 253 0xfdec 0xdbac 0x6cd0 0x1365 0xff0f 0x0218 254 0xfd03 0xaf1e 0xf2ac 0x49b6 0x0acd 0x058c 255 0xf40d 0x0a94 0xb5b2 0xfeb5 0x0dd1 0x0074 Table E4.7 VQ Table for hebap 7; 16-bit two’s complement Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 0 0xad4b 0x5585 0x2896 0x354e 0x29de 0xdc27 1 0xa809 0xdfff 0x4798 0xe61b 0x63ae 0xd5a0 2 0x1a90 0xca42 0xcc22 0x5792 0x394b 0xae36 3 0x092b 0x2914 0x0465 0xf281 0x15c1 0x6a00 4 0xe627 0x2e4b 0xa034 0x5999 0x4f8a 0xe87d 5 0xaaf8 0x29b9 0xb361 0xc553 0xdee2 0xf7df 6 0xf547 0x21ec 0xece1 0x6c5b 0x1d82 0xd147 7 0xfc04 0x099c 0xfc46 0x1292 0xfd8d 0xc010 8 0xb30a 0x5a39 0x004b 0xca8c 0xf5ac 0x083c 9 0x0fd1 0xf4c8 0x16db 0xee95 0x5686 0x3110 10 0xacc8 0xbd17 0xfd26 0x1cfb 0xd276 0xd6e5 11 0x2c44 0x0700 0x682a 0x5bde 0xb397 0xfe15 12 0xba5d 0xbe77 0xcada 0xc7cb 0xa9f3 0xf660 13 0x0443 0xe8b1 0xe0d9 0xbdaf 0xb030 0xaa55 14 0x47d4 0xfbb1 0x078d 0x341e 0xbbc9 0x46c2 15 0x5876 0x43c1 0xd912 0x45ff 0x4762 0x02ba 16 0x05cc 0x4f49 0xe986 0x986d 0x134d 0xa909 17 0xf5d5 0x11eb 0xe92e 0x4820 0x223f 0xf5f8 18 0xf513 0xf9be 0x54d1 0x0c1b 0x9bad 0x0c98 19 0xb5ad 0x1255 0xec71 0x17ac 0x07b4 0xc509 243 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E4.7 VQ Table for hebap 7; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 20 0xf773 0x252c 0xfdee 0x50bd 0xedca 0xdf93 21 0xa8cb 0xdd49 0x09e1 0xd3a8 0x1564 0x03e6 22 0x5654 0xec44 0x0673 0xf59f 0x1207 0x090f 23 0x5177 0xf3fa 0xf2fe 0x1009 0x349e 0x0bfd 24 0x0055 0x4389 0x2818 0xc660 0x00d6 0x005a 25 0x9903 0xb65f 0xb468 0x4b2c 0xd816 0x26b5 26 0xd9f5 0x5011 0xe64d 0xe4b9 0x0b4b 0xfd1e 27 0x505f 0xc20c 0xa67f 0x1ad6 0x004c 0x0147 28 0x2228 0xcdb3 0xa65f 0xf6bc 0xb420 0xd9d5 29 0xcdaa 0x3f37 0x525c 0x0eed 0x02ed 0xca20 30 0xc185 0x47df 0x0957 0xbb03 0x4c1c 0xe87e 31 0x058f 0x2dd6 0x0fd3 0x4b5a 0x1ac9 0xb31f 32 0xebb0 0x2626 0x4746 0x099f 0x494c 0xed0c 33 0xfdab 0x4c2a 0x052b 0xdc78 0xfecc 0xfbb0 34 0xf3e5 0x9b7d 0xc2cf 0x62f4 0x121a 0x0a4b 35 0x4ca7 0xf6b0 0xe117 0x2e14 0xdb8b 0xc8fc 36 0x0a52 0x6755 0xad9d 0xd78e 0xf963 0xf951 37 0x560f 0x5479 0x2d3c 0xa67d 0xefd3 0x0081 38 0xe816 0x0dd6 0x0393 0xfefb 0xffef 0xfe81 39 0x06a0 0x1a30 0xfa6f 0x5166 0x0359 0xeec0 40 0x058f 0xc450 0xde9a 0xda3d 0x145a 0x1637 41 0xee58 0xfd9b 0xd25d 0x15f2 0x1086 0x026b 42 0x03a9 0xec9d 0xc8ea 0xbd30 0xe506 0xe8c0 43 0xc524 0xfe1f 0xe3bb 0xc5d2 0x49bc 0x54a9 44 0x9bc6 0x0b5e 0x0477 0xfeb9 0xfe36 0xfc1d 45 0xda48 0xfccd 0x9ebc 0x0af4 0x4f01 0x043b 46 0xfba9 0xf19e 0xf904 0xb3dc 0x03c6 0x0335 47 0x1c7d 0xab01 0x2a26 0xe46d 0xa503 0xf945 48 0xfee6 0xd4ab 0x00aa 0xae2a 0x8f02 0x3147 49 0x4612 0x0e81 0xf9e5 0x0375 0x0005 0x0234 50 0x17e4 0x58a8 0x08c2 0xe4d9 0x26f7 0xe80c 51 0x10f2 0x68b8 0xf187 0x07b8 0xfbc9 0xf5f6 52 0xfd6b 0xe123 0xf594 0xc4a6 0x453a 0x1117 53 0xefb2 0xd4d3 0x02cd 0xa816 0x061a 0x2fdc 54 0xe6fb 0x479e 0x17d7 0x1b47 0x1744 0x4713 55 0x267b 0x14fa 0x5c34 0xe533 0xe657 0xffc2 56 0x55cc 0x342f 0xfd55 0x0ec9 0x0878 0x00d1 57 0xf20f 0xfb99 0xb2f4 0xf9f8 0x051c 0xfcdd 58 0xf3f5 0x3eb1 0xca21 0xf3fb 0x10c6 0x5ca1 59 0xd8f1 0x26d7 0xc200 0x3286 0xa3b1 0x54c3 60 0x25fa 0x5935 0x2fa0 0x3af3 0x159d 0x12e5 244 Digital Audio Compression Standard, Annex E 22 November 2010 Table E4.7 VQ Table for hebap 7; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 61 0x08c3 0x0833 0x04db 0x0ff9 0x128c 0x329c 62 0x0fa7 0xf65c 0x0d19 0xf3ec 0x228b 0x4280 63 0x10ea 0x17ef 0x15ad 0x2421 0x2bda 0x6fb0 64 0xda8e 0xdd87 0x00ec 0x03f1 0x01c7 0xfc3c 65 0x1aad 0x4b5a 0xfc06 0x00c8 0x071d 0x0242 66 0x144c 0x03bd 0x2884 0x0d02 0xce00 0xff81 67 0xf432 0xdfff 0xc723 0x562d 0x1720 0x041d 68 0x2ae6 0x6556 0xa01e 0xa512 0xd17f 0xe57b 69 0x588b 0xd4fe 0x1668 0x0a07 0x5c99 0xd7f3 70 0xf2f1 0xef77 0xeaae 0x50bb 0xd5a5 0xf1eb 71 0xefdd 0xf1e4 0x11df 0xfcc3 0xfea2 0xfcb1 72 0xf319 0x0d7b 0xe31a 0xd2ac 0x0bcf 0x01c7 73 0x0c80 0xdab5 0x0c82 0xa693 0x2bb0 0x989e 74 0xc8f3 0xefeb 0x3c16 0x37d7 0xd55d 0xb067 75 0x0edf 0xd4f8 0x5624 0x3822 0xc420 0xe1cb 76 0xe76d 0xbac9 0xf9e8 0x2f10 0xb2a3 0xfe45 77 0xe7fd 0xef76 0xff60 0x20ab 0x586e 0x2e87 78 0x4afd 0x0497 0x1cfe 0xd96d 0xefd8 0x1260 79 0xffb8 0xe21c 0xff90 0xd14d 0xf362 0x6a27 80 0x0cca 0x174b 0x1d4d 0xbd85 0x0362 0x9c94 81 0x02e1 0x0745 0x0729 0x07e6 0x0950 0x1293 82 0xeb9f 0x1d58 0x0cfb 0x0a9b 0x0bf9 0xf9ba 83 0x1097 0x0235 0x15fd 0x09c1 0x4663 0xecc8 84 0xf4ef 0xba00 0xe082 0x3d7a 0xfc06 0x0858 85 0x0bea 0xb3e5 0x4222 0x748b 0xd812 0x3b31 86 0xd3ae 0x0076 0x9b3c 0xca3f 0x3bd8 0xfe2c 87 0xed28 0x1360 0xef59 0x0627 0xd69f 0x4c69 88 0xdff5 0xfa7f 0xfd05 0xfb8d 0xfda1 0x0580 89 0xf765 0xd369 0x07e5 0xe70c 0xf5d8 0x02c7 90 0xfe63 0xf631 0xff28 0xf241 0x9195 0x06b7 91 0xc792 0x429a 0x365d 0x34bb 0x9b5e 0xc107 92 0x4b1c 0x1cad 0xcff3 0x02aa 0xf131 0xff39 93 0xef9f 0x510a 0xc2dd 0x2c55 0x16e4 0xfcc8 94 0xac0e 0xf226 0xfffd 0xf957 0xf089 0x23fd 95 0x3c55 0xf8ac 0x07dc 0xb355 0x3f64 0xed13 96 0xf4cd 0xf16b 0xe346 0xff51 0xb169 0x2ba6 97 0xf20d 0x9ff5 0x4cf4 0x19fe 0x03d3 0xfd72 98 0x553c 0xe2fa 0xe611 0xd5f1 0xdf56 0x3cb7 99 0x39eb 0x4639 0xe3dc 0xf2af 0x0772 0xbc78 100 0x0dc5 0xf095 0xfa79 0xf512 0x44f0 0x0822 101 0xe64c 0xc469 0xba07 0x0539 0x3bea 0x52a6 245 Table E4.7 VQ Table for hebap 7; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 102 0x1842 0x25e2 0x3b33 0x9fa6 0xa815 0xf061 103 0xf934 0xfdaf 0x0447 0xe19d 0x61e2 0x15e1 104 0x53a7 0xfe50 0xf986 0xe50e 0xfa62 0xc78a 105 0xe4e1 0x02bc 0xd095 0xfd17 0xa185 0x57c2 106 0x188f 0x0cd3 0x2afe 0x0f04 0x4af0 0x39bd 107 0xa81a 0x3baa 0x1543 0xf508 0xfc36 0xf2f1 108 0x0cb9 0xf184 0x1288 0xdf93 0x591e 0xd820 109 0x5f1a 0xae16 0x4d86 0x03db 0xd14a 0xe77b 110 0x0f42 0xb30b 0x3304 0xf9b7 0x48d1 0x1d2a 111 0x98d7 0xa7eb 0x3fb1 0x07de 0x2adf 0x4670 112 0xe481 0x122f 0xc61e 0x4933 0x3dad 0x0510 113 0x245e 0xf96f 0x394b 0xf302 0x67a7 0xd1b3 114 0x1660 0x171d 0x3458 0x2724 0xf744 0x9f80 115 0x06cd 0xe5b9 0x3197 0xaa07 0x0ff0 0x154a 116 0xf5c3 0x24b1 0x5297 0x9aae 0xf3a6 0xf61f 117 0x50c0 0x49ce 0xc98d 0x1b4e 0xdfbc 0x3dc3 118 0xa2f6 0x2baf 0xcab9 0x2e5c 0x3ead 0x0a46 119 0x47b9 0xd814 0x033d 0x0358 0xfc0e 0x009d 120 0x3840 0xedba 0x1421 0xcc16 0x94d6 0xd4ec 121 0x546d 0x2bf8 0x442d 0x1db4 0x334a 0xfe1c 122 0x0007 0x04d4 0x023d 0x1076 0x15c8 0xf3f7 123 0x0394 0xdc7c 0x0505 0xdd02 0x04a1 0x8fe5 124 0x5453 0x5c8f 0x4aac 0xf4bb 0xc836 0xdf0a 125 0x5b76 0xe7ef 0x32b2 0x0bf5 0xdb79 0x08bc 126 0xf402 0xe350 0xb154 0x169c 0x0246 0xfdd9 127 0xf067 0x013b 0xe1a3 0x2020 0x924e 0xcf4f 128 0x35c6 0xc403 0x4b05 0xaf70 0x32f3 0xb4d1 129 0x0ec1 0xff4f 0x1f5d 0xfc17 0x4594 0x142a 130 0xe374 0xef19 0xb950 0xfd94 0xfaba 0x3a54 131 0x39a4 0xfb3b 0xcded 0xc5b6 0xfddd 0x69f5 132 0x08ba 0x06ac 0x0acc 0x1528 0x1f32 0x9db5 133 0x0b39 0x0e34 0x0f98 0x14e0 0x279e 0x530b 134 0x0486 0x1503 0x01fc 0xd6ee 0x0122 0xf9b1 135 0x045a 0x60d5 0x40bf 0x9db0 0xfed6 0xf4f0 136 0xfbad 0xe800 0xf882 0xe191 0xf465 0xa514 137 0x0fb0 0x2a29 0x43a5 0xef0a 0xae0a 0xf2c9 138 0xee72 0xff31 0xd921 0xf209 0x1f0b 0x0482 139 0xe268 0x1fb5 0xc921 0x4256 0x98a7 0x946c 140 0xc4c4 0x3ee0 0xbe34 0xdd4a 0xa358 0x3e22 141 0x615a 0x1630 0xf8ae 0x01a4 0x0084 0x0075 142 0xfe06 0xb492 0xff3a 0x019c 0xfec9 0x02f0 143 0xf88e 0x0f8d 0xe1f8 0x40b6 0xb4a5 0xc67e Digital Audio Compression Standard, Annex E 22 November 2010 Table E4.7 VQ Table for hebap 7; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 144 0xfe71 0xfd27 0xf121 0xef9c 0xcf95 0x1dd7 145 0x0d28 0x091a 0x2384 0x5c86 0xd7ce 0xf957 146 0xf3b4 0x0a24 0xe0ce 0x390a 0xed39 0x40f3 147 0x1f79 0x05c9 0x0031 0x4335 0x6125 0x1d32 148 0xb498 0xfdff 0x2e81 0x092a 0x15d4 0x0d25 149 0xec39 0xaacc 0x2c6a 0x2a90 0x1311 0x0105 150 0x12ba 0x5061 0x13f5 0xe877 0xe08f 0xfa0f 151 0x1fbd 0xc65c 0x509f 0xc5ba 0x5d85 0xf1be 152 0x30a3 0x0565 0x0e1d 0x21ef 0xa23e 0x12f0 153 0x1a46 0x2993 0x2766 0x6281 0x9db9 0xfbd7 154 0x19a1 0x3699 0x0b5f 0x54e9 0x4051 0x9a3e 155 0xf910 0x0a0f 0xb36a 0xbe60 0x0bd8 0x1a17 156 0x3aa4 0xba0a 0xdf0a 0xabce 0x9619 0x2e20 157 0x0d78 0xfc64 0xc1d7 0xfb91 0x1406 0xaf7b 158 0x1e28 0x08b2 0x4437 0x153a 0x710e 0x4490 159 0x04de 0x3cfe 0xd221 0x602a 0xbb7d 0x0cc8 160 0x0c8f 0x461e 0x0adf 0xfd2e 0xa770 0x175b 161 0xe9d2 0xf390 0x9a19 0x65b2 0x19b7 0x0ce6 162 0x4f56 0xf21d 0xf565 0xfe44 0xfa31 0x05f6 163 0xaf60 0xaa2e 0xd051 0x9b3f 0x229f 0xfbf4 164 0x45e0 0x023a 0xc11a 0x2089 0xf607 0x3bab 165 0xf58b 0x26de 0xf8a9 0x405d 0xce26 0x8eb1 166 0xff88 0xf753 0x00db 0x0061 0x016d 0x0023 167 0x04f6 0xfd32 0x05c8 0xf57f 0x078a 0xe299 168 0x0768 0x222e 0x0772 0x473b 0xce6c 0xe7e2 169 0xf16b 0x3591 0xd966 0xc1a8 0xfaa0 0xe416 170 0xd698 0x2130 0x3e5f 0xdda8 0x1d6c 0x4fd7 171 0x0be1 0xcb6f 0x0408 0x96b8 0x169b 0x6198 172 0xee12 0xdfe4 0xdb96 0xe820 0xbca5 0x6491 173 0xba70 0x1b3a 0x0ea8 0x0272 0xff8e 0x0882 174 0x1161 0xed02 0x1b8e 0xeae4 0x1282 0xf4f5 175 0x133a 0xfd75 0x49fb 0xd976 0x0350 0x075e 176 0xfeb0 0xeade 0x1c42 0x4fdc 0xda91 0xfda8 177 0x030d 0xb3ee 0xce98 0x19ea 0x0586 0x01c2 178 0xf2b9 0xbe7e 0x2b63 0x3390 0xea86 0x549f 179 0xf33f 0x12fb 0xe8b7 0x1d6a 0xd5ab 0x6db6 180 0x286e 0xcd9b 0x6463 0x6428 0xfd81 0x015f 181 0x048b 0x494b 0xeaa6 0xc511 0xff6f 0xfa9f 182 0xc773 0x6a5d 0x8569 0x8073 0x53bf 0xf4b2 183 0x3c3c 0x4987 0x5670 0x4bc6 0x5837 0x3513 184 0xd64e 0x29d6 0x13e1 0xed6c 0x038d 0xaee8 247 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E4.7 VQ Table for hebap 7; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 185 0xcd6c 0xaf4c 0x1cf2 0x0aa2 0x0d63 0x2d41 186 0xfbaf 0x47c6 0x4d13 0xda4e 0x57aa 0x4cb2 187 0xfed8 0xe572 0xc6ab 0x5463 0x4d54 0x5397 188 0xb46a 0xe2b3 0x6366 0x3358 0x218c 0x9d2e 189 0x0c14 0xd686 0x51a0 0x2421 0xf302 0x0704 190 0xfcd5 0x05a9 0x0c22 0x128c 0x2f29 0xc84a 191 0xaf10 0x37c3 0xef14 0x9b12 0xe96b 0xad63 192 0xebf4 0x293a 0xc93c 0xa97a 0x0b18 0xfdd6 193 0x63bd 0x44f0 0x3a26 0xadae 0x099b 0x6236 194 0xdb66 0xf904 0xcdc2 0xe912 0x9b2d 0xd4f1 195 0x1a2a 0x0333 0x2849 0x00a6 0x6bbd 0x020b 196 0x0065 0xb444 0x0d55 0x25a6 0x0040 0x0326 197 0xf54a 0xb9f5 0xf5f0 0x5922 0x2169 0x0466 198 0x0b9c 0x3b63 0x0700 0x635a 0xe9a0 0xbc8f 199 0xfa75 0x0644 0x112e 0x2cbc 0x06c3 0x5ceb 200 0xebf0 0x1211 0xd663 0x6d4d 0x26a9 0xf632 201 0xd6e0 0x927f 0x0bb7 0xfa06 0xfcc0 0xfcc2 202 0xd483 0xcf21 0x56be 0xe3b5 0xa3e6 0xab3e 203 0x4227 0xaa7c 0x0745 0xda7a 0x24d8 0x4a52 204 0x2825 0x252c 0x68bf 0x07da 0xecb1 0xdc88 205 0x15ab 0xf75e 0x37be 0xc43c 0xb48c 0x071e 206 0xed0e 0xfcf1 0xdd01 0xf3fc 0xb1a8 0xf383 207 0x2028 0xf516 0xbaa8 0x33fc 0x0c9d 0xfc21 208 0xd033 0xe64b 0x284b 0xdab0 0x08d4 0xaf58 209 0xe4a8 0x1599 0xe27f 0xe2be 0xd79a 0xd7e6 210 0x0e39 0x4c17 0xe8ac 0xb567 0xb776 0x3205 211 0x0503 0xefbc 0x1066 0x90c7 0xf63e 0x074a 212 0x3eaf 0x68ca 0xcd03 0xe754 0x03d9 0xf9c3 213 0xfe6d 0x3570 0x1939 0x61ee 0x69f4 0xaf1a 214 0xb96a 0xf902 0x9e66 0x1741 0xfc46 0x67e8 215 0xa160 0xc3e9 0x60d4 0x07a1 0xfb90 0x00bb 216 0xf70f 0x30d9 0xaefe 0xfc78 0x4794 0x530a 217 0x0a62 0xe804 0x3f33 0x5704 0xfdd4 0x086a 218 0xe839 0x367e 0x9bae 0x93bf 0x0fd1 0xed45 219 0xac34 0x6769 0x4beb 0xdc5c 0x037f 0x012f 220 0xa948 0x99bf 0xe876 0x6099 0xa672 0xdcba 221 0xc83c 0xc192 0x5cb4 0xa6bd 0x2434 0xf0ff 222 0x732a 0x55a3 0xe7bb 0x068f 0xf7f5 0xfba0 223 0xfe4d 0x264a 0xf0cd 0x3047 0xef40 0xb5e5 224 0x4d38 0xffaa 0x09a3 0x07c6 0xfc03 0xeb16 225 0x51fa 0xddb1 0xeb2f 0xa3f6 0xed86 0x0a71 248 Digital Audio Compression Standard, Annex E 22 November 2010 Table E4.7 VQ Table for hebap 7; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 226 0xec19 0x15e5 0xedeb 0x4ace 0x65b5 0xd01d 227 0x03cc 0x1aca 0x11c7 0x6d2d 0xf047 0xf720 228 0x17bb 0xf344 0xec83 0xfe8b 0xf9dd 0xf16e 229 0xe3a8 0xcd40 0xdd8c 0xec0b 0x5a0e 0x13be 230 0x0398 0x0a37 0x1ee8 0xe347 0xecd7 0x4eda 231 0xff06 0x154e 0x0c44 0x1b10 0xb6dd 0xf7fd 232 0xd7c5 0xeeec 0x4c98 0x130f 0xfd6b 0xf8a3 233 0x39e0 0xde65 0xb299 0x17f7 0xad26 0x371c 234 0xd157 0xf751 0x139a 0x2e74 0x58d5 0x0196 235 0xcc80 0xf5cb 0xcc38 0xa85f 0xcf3e 0xdf44 236 0x42aa 0x62b3 0xf71f 0x13c0 0xfeaa 0x0091 237 0x20d1 0xbaed 0x4aa8 0x2977 0xb403 0x42bb 238 0x5155 0xe9bc 0x300a 0x9c02 0x2897 0x1e0c 239 0x11c6 0x3da3 0x43ba 0xb44d 0xed60 0x04b6 240 0xe1d5 0x2a54 0x95e4 0xd351 0x1ab3 0xf910 241 0x09ee 0x0c7f 0x115a 0x4469 0xf181 0xfc6e 242 0x51e0 0xbe7a 0xe94a 0x2b4f 0xffba 0x59b1 243 0x0ce9 0x0b67 0x1870 0xed40 0xae1a 0xf362 244 0x1724 0xbf5d 0x0887 0x0aad 0x0d76 0xa4f6 245 0xe853 0xff3e 0xc9e4 0xd525 0x4c20 0x0405 246 0x1173 0xe8b4 0xb5c4 0x05ef 0xfe99 0x0357 247 0xf9d3 0xe249 0x5636 0xd2c4 0xd8d0 0x42ce 248 0xcf84 0x09f9 0x10e4 0x57e4 0x1677 0x2f8a 249 0x9dd9 0x464c 0xe710 0x049c 0x049e 0x2596 250 0x5ba6 0xdee9 0xeed8 0xf593 0x1dd6 0xbe3d 251 0xea79 0xf4b9 0xd5fb 0xae6d 0x1c4e 0x041d 252 0x0a8f 0xaf86 0xe27e 0x1d5c 0xe1c4 0x16ec 253 0x50be 0x558d 0x01c9 0x3a79 0xbb07 0xd16f 254 0x0e13 0xf9c5 0xf77f 0xff63 0xffd5 0x025d 255 0x09d1 0x22fa 0x291f 0x581f 0xc11c 0xc157 256 0x1772 0x1357 0x1a8b 0xed02 0xa880 0x49a1 257 0x1da6 0xf963 0x9f90 0xf2b4 0x3759 0x04be 258 0xeed2 0xe5f9 0xe52a 0xd89d 0x9fec 0x2425 259 0x28e4 0x4557 0xe1bc 0x0093 0xe756 0x1143 260 0x3f3b 0xbf53 0xefe9 0x10ce 0x1dc9 0x1521 261 0x0ce7 0x0aaf 0x1d22 0xb242 0xf732 0xf18a 262 0xf7e3 0x5469 0x3a16 0x3101 0xe83f 0xf91c 263 0x1246 0x2ddc 0x0b2b 0x1b29 0x077f 0xf0e1 264 0x0dc2 0xaaa3 0xf65b 0xd72b 0x49cd 0xd60a 265 0x0eaf 0xd831 0xecfe 0xf59d 0xba59 0xfb26 266 0x3a8f 0x2487 0x2e5e 0xf9db 0xed10 0x5815 249 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E4.7 VQ Table for hebap 7; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 267 0x2525 0x95f0 0x29ee 0x5173 0x99b7 0xba2a 268 0xe3fe 0xfa90 0xb3d9 0x31ca 0x2006 0xf83c 269 0x075b 0x6dfe 0xfcb2 0xe3bd 0x00f9 0x00e9 270 0xe3e0 0x029d 0xfe8d 0xf47c 0x5ac2 0xe9fd 271 0x0c45 0x0120 0x0c97 0xfb16 0xff9e 0x9429 272 0x43dd 0xa53d 0x13f6 0xd441 0xf5f2 0xd321 273 0xecc0 0x05ee 0xeab0 0x029e 0xb89a 0x079f 274 0x285e 0xb267 0xedd7 0x0169 0xff60 0xfc65 275 0x492c 0x37b8 0xf3ad 0xe2c3 0xf300 0x1747 276 0xf1e2 0x5255 0x1c6c 0x0dd0 0x1fb9 0xfa08 277 0xdf1a 0x01f4 0xb512 0x49f1 0x6718 0xfbf1 278 0x3d17 0x6444 0x20b7 0x076f 0x0799 0xd135 279 0xf564 0x0d3d 0x68e2 0xee15 0x070b 0x0016 280 0x0499 0xfd71 0x04d1 0xf7b0 0x1ea4 0x06e7 281 0xfd07 0x2011 0xb4a6 0xee0f 0x0783 0xfea9 282 0xfd4f 0xf236 0xf33d 0xf124 0xf53f 0x4886 283 0xf7c2 0x07aa 0xfab7 0x4103 0x0acd 0xa5c2 284 0xfe4f 0x1329 0x012e 0x32d8 0x3e3d 0xe8ef 285 0x0c83 0x101e 0x2bad 0xea88 0xf61f 0xfb78 286 0xfbbd 0xe6bb 0xfa79 0x1632 0xfef4 0x0247 287 0xdb43 0xb38c 0x1848 0x067a 0x03e1 0xffb5 288 0xf961 0xee68 0xf70f 0xf008 0xe664 0xbf3f 289 0x1298 0xfc84 0xd56a 0x1974 0x5e87 0xe885 290 0xff03 0x03e8 0x003f 0xffaf 0xff8d 0xfe82 291 0xfacb 0x5ea0 0xfd46 0xedc5 0xf50f 0xb538 292 0xfc94 0x8f3e 0xaa8f 0x3185 0xe738 0x0ca3 293 0x41cf 0x5299 0x99c4 0xf391 0xfe74 0x00e6 294 0x4778 0xe192 0xcdc7 0xfd59 0xfa3f 0x0005 295 0xd708 0x2ca5 0x64cd 0xfb9e 0x0579 0xfe4a 296 0x0ec6 0xe2fb 0x6860 0x449f 0x4b39 0x30fe 297 0x18bc 0xfd16 0x31f5 0x2464 0xa7f2 0xeb16 298 0x0d5a 0xa738 0x6962 0x477f 0x0434 0x03bc 299 0x954d 0xf454 0x0398 0x00eb 0x08b9 0x0051 300 0x1837 0x14b0 0x3edd 0x39b0 0xdf13 0xfba8 301 0xe6e0 0x4b2c 0x26c1 0xf34b 0x04fe 0xfc46 302 0x5e95 0x0801 0xa66d 0x0a19 0xf696 0xef88 303 0x2446 0x37ca 0xb2e9 0xf06f 0xf6d8 0x0404 304 0xb160 0x4649 0xdb0e 0x59e4 0xbda9 0x21b1 305 0xe510 0xaf06 0x0eb2 0x4407 0x5745 0x4a50 306 0x034a 0x5e75 0x61e6 0xe931 0xffb2 0x03a9 307 0xfd93 0x4d0a 0xa174 0xf856 0xc5fa 0xffc8 250 Digital Audio Compression Standard, Annex E 22 November 2010 Table E4.7 VQ Table for hebap 7; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 308 0x58ee 0xec01 0x43d5 0x5d3c 0xb3e8 0xe662 309 0xf792 0x4452 0xac45 0x0d0c 0xcded 0xb0b9 310 0xda6b 0x43ad 0x02cb 0x08d9 0xefe5 0xfe14 311 0x23c4 0x3293 0x6aa7 0xad49 0xe848 0xdb0f 312 0xcc94 0xa51b 0xc94a 0xefa8 0x1b42 0x0002 313 0x03aa 0xcbbb 0x0dc0 0xa117 0x5976 0x4c85 314 0xecd1 0xb2c2 0x4d34 0xdba2 0xce96 0x0eeb 315 0xeaaa 0xef67 0xe4b5 0xe78c 0xc989 0x9dc2 316 0x247d 0x2881 0xc9da 0xe5d0 0x581c 0xfdf9 317 0x19fb 0x4950 0xed09 0x311a 0x398a 0xd81f 318 0xfcc9 0x46c7 0x018e 0xf9d2 0xff8c 0xfe95 319 0xe4e9 0xce6a 0x9118 0x2168 0x1b31 0xff11 320 0xf5d6 0xeda0 0xfc03 0x07df 0x1409 0x5c76 321 0xcef1 0xe002 0x9e3c 0x4870 0x3763 0x067f 322 0x0ee5 0x522c 0xda6c 0xec45 0xf8f8 0xfbc1 323 0xa9d7 0x4123 0x3a70 0x24f3 0x0ae2 0x425f 324 0x9a48 0xb48a 0xe6f2 0x0450 0x16a6 0xb989 325 0xf937 0x60f9 0x28b1 0xd4b1 0x0380 0xeb67 326 0xf8c1 0x2d8d 0xf50d 0x60e9 0xac45 0xb2b0 327 0xa44f 0xbea7 0xe96a 0x160b 0x0a4c 0x134c 328 0xf944 0x1124 0x97cf 0xca81 0x294a 0x9ad9 329 0x3bfe 0xb3d8 0x6682 0xb7c3 0x06c8 0x1f76 330 0x1634 0x519a 0x0ffb 0xb564 0xc704 0xd71c 331 0x436c 0xc05d 0x3a0b 0xbad1 0xb51a 0x3093 332 0x95cf 0xcee3 0x1a57 0xfdce 0x03d0 0xfeff 333 0x306b 0xde56 0xa918 0xb27d 0x2b05 0x1e52 334 0x0ed7 0x2e4d 0x941a 0xdee7 0x0441 0xfa29 335 0x102d 0xf77a 0x97a0 0xfd21 0xfcfa 0x05bd 336 0x0c35 0x35c2 0x11fe 0x7249 0x4953 0xd91a 337 0xbbc7 0xdb1b 0xbb66 0xf61e 0xe6dd 0xf172 338 0xf932 0x10ff 0xe547 0xb2c3 0x259b 0xd662 339 0x1c53 0x0dc5 0x2a53 0x15e1 0x626e 0xa4cc 340 0xd7c4 0xba5a 0x0277 0x2d78 0x07fc 0xae72 341 0xfc97 0xdeca 0xfbd9 0xc2c6 0xd63b 0x3a56 342 0xc1ab 0x6de9 0x1494 0x01dd 0xfbe3 0x0486 343 0xfa29 0xdd92 0xe97c 0x9e7b 0x6584 0x1ee3 344 0xfbf2 0xff8e 0xf6fc 0xfad9 0xe6b0 0x05c0 345 0x131f 0xba17 0x9b06 0x14b5 0xff44 0x062d 346 0x0c80 0x4349 0x10fa 0x5655 0xb791 0x560c 347 0xd7f6 0x0221 0xd54c 0x08e4 0x925a 0x1fb6 348 0x3bef 0x0919 0x2464 0x5039 0x3a3c 0x521d 251 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E4.7 VQ Table for hebap 7; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 349 0x18b9 0x17f2 0xa044 0x0345 0xde43 0xe92c 350 0x1cda 0xfe0b 0x2907 0x4ea3 0x2cab 0xed6d 351 0xf547 0x5e6e 0xdbc6 0x3ba9 0xdf3b 0xe935 352 0x0bb0 0xf4d0 0x17a0 0xe2cf 0x2da7 0xb1e4 353 0xfc8d 0xd14e 0xd908 0xaabb 0xeeac 0x95d6 354 0x0d82 0x4caa 0x0500 0x0a25 0x4d89 0x1487 355 0xeb3d 0x4abd 0xc74a 0xdd0e 0x35b5 0xfab8 356 0x48d2 0x44f7 0x2af9 0x1aa1 0xb80e 0x18c0 357 0xf95f 0x08c4 0xede0 0x0f6c 0xcda6 0xeb67 358 0x4fcc 0x292e 0x104a 0xfc0c 0x4bef 0x54bb 359 0xf481 0xb2e9 0xef90 0x0528 0x038d 0xdd3f 360 0x2487 0xe07e 0xf5c6 0xcd7b 0x67d6 0x0db3 361 0x25e9 0xa79c 0x2077 0x1fe7 0xcc13 0x15e8 362 0x0c96 0x0ea5 0xfa1c 0x00a5 0xffcc 0xff3c 363 0x0066 0xa728 0xdd80 0x0387 0xd363 0xc6ba 364 0xff88 0x176e 0x4d35 0x3459 0x0e2c 0x144d 365 0x2150 0x16c3 0xfbd6 0x0306 0xffd9 0xff5a 366 0x24c3 0xdafc 0x256d 0xcd34 0x5f88 0x6144 367 0x45d6 0x08bb 0xab79 0x4ffe 0x126c 0xe3ea 368 0xf64e 0x2527 0x064b 0xaa49 0x3796 0xfaf7 369 0x2448 0xf70d 0x5aaf 0xf284 0xd5a6 0x000b 370 0x2518 0x0be1 0x140a 0xf0ce 0xad1d 0xa7c3 371 0x37b6 0xd992 0x4ee3 0x36c3 0x005b 0xbcd0 372 0xb761 0x03d4 0x0011 0x0335 0x0078 0xfdc2 373 0x2ffd 0xb4bb 0x35ae 0x3ff5 0xff5f 0x1789 374 0xf2dc 0x05fa 0xf05b 0x0996 0xd588 0xa2e1 375 0x0069 0x13dd 0xfefc 0x169e 0xfdb4 0x4ae2 376 0x1019 0x1049 0x347f 0x3934 0x51a3 0x1d0a 377 0xff51 0x332d 0xf188 0x5ac1 0x0f43 0x277a 378 0xe82b 0x5bab 0x1454 0xfac3 0x063f 0x3376 379 0xf36f 0xf25a 0x3b0d 0xdf3d 0xd20e 0xed72 380 0x047a 0x1243 0xb44e 0x3a45 0xec1d 0x00f9 381 0xabfe 0x2798 0xbfa7 0xcc07 0x47ce 0xde67 382 0x0274 0x098f 0x0d10 0x0c3a 0xec05 0x0077 383 0x45ec 0xa86a 0xbb1f 0x55cf 0xc05b 0xe204 384 0x41df 0x5e96 0x15ec 0xf0ee 0xfcd7 0x0eee 385 0xf70d 0x276b 0xf6c8 0x9deb 0xfb36 0x0138 386 0x0b8d 0x2bf8 0x6879 0xcc2e 0xf281 0xfb98 387 0xb2ce 0xf56c 0x11fc 0x18d3 0x0666 0x639d 388 0xb377 0xe1b7 0x0c57 0xffab 0xfe17 0xf8c1 389 0x032e 0x30de 0x4a85 0xedb7 0xf5ce 0xfa3e 252 Digital Audio Compression Standard, Annex E 22 November 2010 Table E4.7 VQ Table for hebap 7; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 390 0xa490 0xb5ad 0x1fc9 0x4da6 0x1ee8 0xfee6 391 0x0347 0xb33c 0x2e97 0x6a8e 0xf375 0x08da 392 0x0fb4 0xfbaa 0x2022 0xfb06 0x51ba 0x61e4 393 0x67d0 0x0145 0xde0b 0xff18 0xf756 0xfd45 394 0xd3e3 0xef98 0x070d 0xe5ef 0xa664 0xfac5 395 0xf82b 0xc1f2 0xfbe9 0x93d9 0xcc4d 0x3822 396 0xa9c7 0x079d 0x3377 0xc2d8 0xf8ca 0x1f77 397 0x0bdf 0x2ef9 0x1bdc 0x9fc8 0x019d 0xf6d5 398 0xa210 0xff32 0x30ab 0xe602 0xfe5f 0xd895 399 0x4703 0xa378 0xafdd 0xbff4 0x1c3e 0x02fb 400 0x161b 0xec23 0x3636 0xa34f 0xd4bb 0xb37d 401 0x2c4c 0x01f5 0x61d0 0x1dc0 0xb336 0x0645 402 0x97e6 0x22ae 0x2930 0x01a1 0x0513 0x0105 403 0x387c 0x2c69 0xf341 0x2706 0x2002 0x46bf 404 0x054b 0xae9a 0xdc14 0xc144 0xde91 0xfd26 405 0xf882 0xae37 0xb88b 0xf663 0xf5a5 0x10dc 406 0xf506 0x5fc9 0xd50c 0x9b87 0x0134 0xfb2e 407 0xdc8d 0xbc80 0xf8d7 0x8d62 0xa16b 0xbf09 408 0xf4e5 0x27b5 0xeb25 0xf4b8 0x5562 0xaca4 409 0xc228 0x3a01 0xa31c 0x1440 0x2781 0xaf61 410 0xb3b1 0xd39f 0x20dd 0x05ce 0xa396 0xe981 411 0xe2a8 0x0403 0xaec6 0x35a4 0x4db4 0xaa52 412 0xd09c 0xe492 0xb519 0xdd78 0x566d 0xbf96 413 0x0791 0x145a 0xe752 0xa314 0x3355 0x2b4a 414 0xff33 0x1794 0xfe84 0x21d2 0xff17 0x6d74 415 0xea6d 0x1d35 0x1dd3 0x5c2b 0x2623 0xf5e2 416 0x549a 0x9167 0xf3f2 0xfed4 0xfbf8 0x06d0 417 0xa8b0 0x4106 0x00d0 0x1a09 0xbc08 0xf42c 418 0x4832 0x2478 0xf54f 0xb454 0x0197 0xeedb 419 0xeccf 0xbc26 0x4983 0xbb0a 0x3468 0x3b80 420 0x1e45 0x18e0 0x5a5f 0xb902 0x1da0 0xef68 421 0xfa2f 0xe685 0x024a 0xd853 0x3a74 0x63e0 422 0x0f04 0xe7f4 0x1321 0xcd0b 0xa802 0x160f 423 0xded5 0xf7c7 0x9f3a 0x0389 0xdb92 0x05b0 424 0xf420 0xfa3c 0x048e 0xeeb4 0x2be4 0x23f4 425 0x0d45 0xfa55 0x351e 0xc21f 0x5fdc 0x16bb 426 0x2123 0xf44f 0x542b 0xbdec 0x1e3d 0x5dd2 427 0xc5ac 0xa332 0xeb2c 0xe5f8 0xee6f 0x33d3 428 0x4bb3 0x3274 0xf7a2 0xfd1f 0x526c 0xa9ab 429 0x0d41 0xedeb 0x1667 0xb61f 0xe4c7 0x0a7f 430 0x047c 0xc0ed 0xac47 0x9241 0xfd8a 0xc78f 253 Advanced Television Systems Committee, Inc. Document A/52:2010 Table E4.7 VQ Table for hebap 7; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 431 0x1c84 0x02a0 0x4862 0xbbd4 0xd85b 0x015f 432 0x2c5c 0xd522 0x433c 0x1210 0x0091 0x457f 433 0xfd39 0xf269 0xf742 0x3e0f 0x07eb 0x0000 434 0x9270 0x0702 0xfdaf 0xf53a 0xaaa4 0x2d0f 435 0xb31d 0x1349 0x55f4 0x5413 0xf3b4 0x06fe 436 0x032d 0x2027 0x0a49 0x2ecd 0xf41d 0x56b9 437 0x22f8 0x9f48 0xfd4e 0x3a19 0xf6c2 0xeb04 438 0x20d6 0xeac1 0xfeee 0xfd7e 0xff6f 0x030a 439 0xe633 0x1c5a 0x512c 0xa42d 0xb73f 0x58fe 440 0xa690 0x9c70 0x2724 0xf9b2 0x05e4 0xfa8f 441 0x1db7 0x0197 0x9f9a 0xbfff 0xf8f4 0xeda5 442 0xd6a0 0xb53d 0x28de 0xf15d 0x2211 0xe4f9 443 0x32d2 0x14ac 0xe7aa 0xecec 0xae58 0xf8fb 444 0x41fb 0xca36 0xfe2f 0x4b8f 0xd60b 0xcd61 445 0x6269 0xc631 0xe9cf 0xfdf7 0xfebf 0xfb45 446 0x1b05 0xf3eb 0x4ed7 0x96e9 0xd106 0x050f 447 0x0131 0x07c8 0x4c01 0xfc27 0x0019 0xfdf7 448 0x1a33 0xf18e 0x20ad 0xde11 0x55a1 0x95e2 449 0x123c 0x176d 0x1bcd 0x2db0 0x5f51 0xd5d6 450 0x02e8 0xdb38 0x4db5 0x07ab 0x1ef2 0xd9a0 451 0x0d66 0x5322 0xf938 0x2a5c 0x2275 0x6987 452 0xdd93 0x05f1 0xa21a 0x0673 0x1e9e 0xfb48 453 0x0f47 0xd42b 0x0cc9 0xcf03 0x1c00 0x47e2 454 0x548a 0x239d 0xd2f0 0xeb78 0x1ba5 0x094e 455 0x0064 0x0ee9 0xe5c7 0x04dc 0x05ee 0xfebf 456 0x1f0a 0xb712 0x29ab 0xecfe 0x02d7 0x0308 457 0xc1f5 0xe02a 0xf7d9 0x58d3 0x061f 0xf266 458 0x111c 0xf551 0x2115 0xe48f 0xd360 0x0525 459 0x695a 0x1129 0x1df1 0x4499 0xfd36 0x028a 460 0xc0c1 0xfcbd 0x20ad 0x0703 0xc816 0x3fa9 461 0x1198 0xd8c0 0x1dee 0x97be 0xbbec 0x0a14 462 0x0030 0xf070 0x0234 0xe911 0x0a62 0xb71e 463 0x3123 0x9a60 0xc2e6 0x0a70 0xfabd 0xfc89 464 0xecaa 0x1070 0xe565 0x0a09 0xaf73 0xde2e 465 0xf8d4 0xc61e 0xea3d 0xa4e6 0xc630 0x650b 466 0x153a 0x9215 0xf6cb 0xf4bd 0xfdc6 0x097f 467 0x3328 0xf52d 0x61a2 0xcf30 0x9f6d 0xfbff 468 0xe9d4 0xef0d 0x0774 0x48c4 0xacb5 0x43d6 469 0x6c0c 0x9307 0xc3cf 0x059c 0xe438 0xf73f 470 0x1f53 0x0f07 0x5ff8 0xfe2b 0x25ca 0x29bb 471 0xfc79 0xd85b 0x0709 0xacf4 0x12bb 0xddd1 254 Digital Audio Compression Standard, Annex E 22 November 2010 Table E4.7 VQ Table for hebap 7; 16-bit two’s complement (Continued) Index val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s (16-bit two’s complement) complement) complement) complement) complement) complement) 472 0x0462 0xda92 0x0a41 0x5907 0x03bc 0x0372 473 0x1ec4 0x4a83 0xd954 0xa136 0x1d48 0x243d 474 0x03d4 0x9774 0xeaf6 0x1514 0x043e 0x0670 475 0x70a6 0xfb0a 0xfe41 0x0005 0xfe53 0xffec 476 0xc44d 0x17f4 0x591c 0x04e4 0xd915 0x01ff 477 0x0353 0x1ef5 0xfe37 0xd04e 0x10a5 0x1d9b 478 0xee4e 0x2104 0xfb22 0x38a5 0x9e89 0xe980 479 0xba6a 0xd619 0x269f 0xa287 0xcb88 0x0756 480 0xc537 0x27b5 0x406b 0xc6f5 0xd240 0xad5c 481 0xf30b 0x0348 0xe9cd 0x578d 0x07ca 0x024a 482 0x5a76 0xe964 0xc53d 0xd77c 0xdbc9 0xcb2d 483 0xfcfb 0xdadb 0xf067 0xa138 0x210f 0x16ac 484 0xde9f 0xfd41 0xcf68 0xf06f 0x9dde 0x920d 485 0xbeed 0x3e81 0x0aba 0x064b 0x13e9 0xfbed 486 0x0029 0xe3f3 0x4dbf 0x7b43 0x8213 0x3667 487 0xe9e6 0x034d 0xce1a 0x1649 0x4137 0xffaa 488 0x14a2 0x3a1b 0x6992 0x5284 0x3da0 0xd713 489 0x3978 0x4cc0 0xd321 0xcbcf 0xb11c 0xc483 490 0x2195 0xdc4e 0xfd8e 0x2a8b 0xe881 0x18ca 491 0xfa30 0xfb08 0xfa39 0xfae9 0xf188 0xea93 492 0xf2d6 0x45cf 0xe634 0x6162 0x651e 0xf3c9 493 0x20e0 0x6c87 0xfa97 0x14e6 0xef5c 0x4e19 494 0x1638 0x016a 0x435e 0x0ee1 0xf352 0x0440 495 0xff97 0x8c59 0x0abb 0x3b77 0xff59 0x0e8a 496 0x0dae 0xf385 0x219a 0x1e5c 0xf9e2 0xfc6d 497 0xfe15 0x0cb9 0xf689 0x1592 0x507e 0xff9c 498 0xc984 0xd398 0xc3f1 0xaa96 0xdeb8 0x2fbd 499 0xfd98 0x0978 0xf819 0x112e 0xf123 0x1fac 500 0xe3dc 0x5233 0x52db 0xdb4d 0xb441 0x0380 501 0xe997 0xc4c8 0xacce 0x42aa 0xfc12 0xfe92 502 0x1875 0x0ca8 0xd15f 0xc0ab 0xc234 0x19b5 503 0xf3ad 0x60dc 0x0aad 0xfb17 0xfc95 0xf9c3 504 0xb00b 0x2b56 0x5e07 0xdce5 0x3738 0x08ac 505 0xc8e6 0x2eb7 0xa821 0x1027 0xfbe1 0xead4 506 0x0321 0xf5a1 0x003c 0xeb34 0xfcea 0x1731 507 0xe334 0xf91c 0xa85f 0x9a34 0x54cb 0x1052 508 0xe9ad 0xe608 0xc5c4 0x052d 0xa214 0x05d5 509 0xe878 0xcf38 0x5d7a 0x0b86 0x0641 0x0495 510 0x4a7b 0x44de 0x4609 0xd662 0x2ab0 0xeca2 511 0x0c9f 0xf32c 0x6ac8 0x104e 0xf96d 0x01f1 255 Advanced Television Systems Committee, Inc. 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