Digital Audio Compression Standard (AC-3, E-AC-3)

Revision B

Document A/52B, 14 June 2005

Advanced Television Systems Committee, Inc.

1750 K Street, N.W., Suite 1200
Washington, D.C. 20006
Advanced Television Systems Committee, Inc. Document A/52B

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 Television Association (NCTA), and the Society of Motion Picture and Television
Engineers (SMPTE). Currently, there are approximately 140 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.
Mailing address Advanced Television Systems Commmittee, Inc.
1750 K Street, N.W., Suite 1200
Washington, D.C. 20006
Telephone 202-872-9160 (voice)
202-872-9161 (fax)
Web site http://www.atsc.org
E-mail standards@atsc.org

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.

2
Digital Audio Compression Standard, Table of Contents 14 June 2005

Table of Contents

1. INTRODUCTION 19
1.1 Motivation 20
1.2 Encoding 20
1.3 Decoding 22
2. SCOPE 23
3. REFERENCES 23
3.1 Normative References 23
4. NOTATION, DEFINITIONS, AND TERMINOLOGY 23
4.1 Compliance Notation 23
4.2 Definitions 24
4.3 Terminology Abbreviations 25
5. BIT STREAM SYNTAX 28
5.1 Synchronization Frame 28
5.2 Semantics of Syntax Specification 28
5.3 Syntax Specification 28
5.3.1 syncinfo: Synchronization Information 29
5.3.2 bsi: Bit Stream Information 30
5.3.3 audioblk: Audio Block 31
5.3.4 auxdata: Auxiliary Data 35
5.3.5 errorcheck: Error Detection Code 35
5.4 Description of Bit Stream Elements 35
5.4.1 syncinfo: Synchronization Information 35
5.4.1.1 syncword: Synchronization Word 35
5.4.1.2 crc1: Cyclic Redundancy Check 1 36
5.4.1.3 fscod: Sample Rate Code 36
5.4.1.4 frmsizecod: Frame Size Code 36
5.4.2 bsi: Bit Stream Information 36
5.4.2.1 bsid: Bit Stream Identification 36
5.4.2.2 bsmod: Bit Stream Mode 36
5.4.2.3 acmod: Audio Coding Mode 36
5.4.2.4 cmixlev: Center Mix Level 37
5.4.2.5 surmixlev: Surround Mix Level 37
5.4.2.6 dsurmod: Dolby Surround Mode 38
5.4.2.7 lfeon: Low Frequency Effects Channel On 38
5.4.2.8 dialnorm: Dialogue Normalization 38
5.4.2.9 compre: Compression Gain Word Exists 38
5.4.2.10 compr: Compression Gain Word 38
5.4.2.11 langcode: Language Code Exists 38
5.4.2.12 langcod: Language Code 38
5.4.2.13 audprodie: Audio Production Information Exists 39
5.4.2.14 mixlevel: Mixing Level 39
5.4.2.15 roomtyp: Room Type 39

3
Advanced Television Systems Committee, Inc. Document A/52B

5.4.2.16 dialnorm2: Dialogue Normalization, Ch2 39

5.4.2.17 compr2e: Compression Gain Word Exists, Ch2 39
5.4.2.18 compr2: Compression Gain Word, Ch2 39
5.4.2.19 langcod2e: Language Code Exists, Ch2 39
5.4.2.20 langcod2: Language Code, Ch2 39
5.4.2.21 audprodi2e: Audio Production Information Exists, Ch2 40
5.4.2.22 mixlevel2: Mixing Level, Ch2 40
5.4.2.23 roomtyp2: Room Type, Ch2 40
5.4.2.24 copyrightb: Copyright Bit 40
5.4.2.25 origbs: Original Bit Stream 40
5.4.2.26 timecod1e, timcode2e: Time Code (first and second) Halves Exist 40
5.4.2.27 timecod1: Time Code First Half 40
5.4.2.28 timecod2: Time Code Second Half 40
5.4.2.29 addbsie: Additional Bit Stream Information Exists 41
5.4.2.30 addbsil: Additional Bit Stream Information Length 41
5.4.2.31 addbsi: Additional Bit Stream Information 41
5.4.3 audblk: Audio Block 41
5.4.3.1 blksw[ch]: Block Switch Flag 41
5.4.3.2 dithflag[ch]: Dither Flag 41
5.4.3.3 dynrnge: Dynamic Range Gain Word Exists 41
5.4.3.4 dynrng: Dynamic Range Gain Word 41
5.4.3.5 dynrng2e: Dynamic Range Gain Word Exists, Ch2 41
5.4.3.6 dynrng2: Dynamic Range Gain Word Ch2 41
5.4.3.7 cplstre: Coupling Strategy Exists 42
5.4.3.8 cplinu: Coupling in Use 42
5.4.3.9 chincpl[ch]: Channel in Coupling 42
5.4.3.10 phsflginu: Phase Flags in Use 42
5.4.3.11 cplbegf: Coupling Begin Frequency Code 42
5.4.3.12 cplendf: Coupling End Frequency Code 42
5.4.3.13 cplbndstrc[sbnd]: Coupling Band Structure 42
5.4.3.14 cplcoe[ch]: Coupling Coordinates Exist 43
5.4.3.15 mstrcplco[ch]: Master Coupling Coordinate 43
5.4.3.16 cplcoexp[ch][bnd]: Coupling Coordinate Exponent 43
5.4.3.17 cplcomant[ch][bnd]: Coupling Coordinate Mantissa 43
5.4.3.18 phsflg[bnd]: Phase Flag 43
5.4.3.19 rematstr: Rematrixing Strategy 43
5.4.3.20 rematflg[rbnd]: Rematrix Flag 44
5.4.3.21 cplexpstr: Coupling Exponent Strategy 44
5.4.3.22 chexpstr[ch]: Channel Exponent Strategy 44
5.4.3.23 lfeexpstr: Low Frequency Effects Channel Exponent Strategy 44
5.4.3.24 chbwcod[ch]: Channel Bandwidth Code 44
5.4.3.25 cplabsexp: Coupling Absolute Exponent 44
5.4.3.26 cplexps[grp]: Coupling Exponents 44
5.4.3.27 exps[ch][grp]: Channel Exponents 45
5.4.3.28 gainrng[ch]: Channel Gain Range Code 45
5.4.3.29 lfeexps[grp]: Low Frequency Effects Channel Exponents 45

4
Digital Audio Compression Standard, Table of Contents 14 June 2005

5.4.3.30 baie: Bit Allocation Information Exists 45

5.4.3.31 sdcycod: Slow Decay Code 45
5.4.3.32 fdcycod: Fast Decay Code 45
5.4.3.33 sgaincod: Slow Gain Code 45
5.4.3.34 dbpbcod: dB per Bit Code 45
5.4.3.35 floorcod: Masking Floor Code 45
5.4.3.36 snroffste: SNR Offset Exists 45
5.4.3.37 csnroffst: Coarse SNR Offset 46
5.4.3.38 cplfsnroffst: Coupling Fine SNR Offset 46
5.4.3.39 cplfgaincod: Coupling Fast Gain Code 46
5.4.3.40 fsnroffst[ch]: Channel Fine SNR Offset 46
5.4.3.41 fgaincod[ch]: Channel Fast Gain Code 46
5.4.3.42 lfefsnroffst: Low Frequency Effects Channel Fine SNR Offset 46
5.4.3.43 lfefgaincod: Low Frequency Effects Channel Fast Gain Code 46
5.4.3.44 cplleake: Coupling Leak Initialization Exists 46
5.4.3.45 cplfleak: Coupling Fast Leak Initialization 46
5.4.3.46 cplsleak: Coupling Slow Leak Initialization 46
5.4.3.47 deltbaie: Delta Bit Allocation Information Exists 46
5.4.3.48 cpldeltbae: Coupling Delta Bit Allocation Exists 46
5.4.3.49 deltbae[ch]: Delta Bit Allocation Exists 47
5.4.3.50 cpldeltnseg: Coupling Delta Bit Allocation Number of Segments 47
5.4.3.51 cpldeltoffst[seg]: Coupling Delta Bit Allocation Offset 47
5.4.3.52 cpldeltlen[seg]: Coupling Delta Bit Allocation Length 47
5.4.3.53 cpldeltba[seg]: Coupling Delta Bit Allocation 47
5.4.3.54 deltnseg[ch]: Channel Delta Bit Allocation Number of Segments 48
5.4.3.55 deltoffst[ch][seg]: Channel Delta Bit Allocation Offset 48
5.4.3.56 deltlen[ch][seg]: Channel Delta Bit Allocation Length 48
5.4.3.57 deltba[ch][seg]: Channel Delta Bit Allocation 48
5.4.3.58 skiple: Skip Length Exists 48
5.4.3.59 skipl: Skip Length 48
5.4.3.60 skipfld: Skip Field 48
5.4.3.61 chmant[ch][bin]: Channel Mantissas 48
5.4.3.62 cplmant[bin]: Coupling Mantissas 48
5.4.3.63 lfemant[bin]: Low Frequency Effects Channel Mantissas 49
5.4.4 auxdata: Auxiliary Data Field 49
5.4.4.1 auxbits: Auxiliary Data Bits 49
5.4.4.2 auxdatal: Auxiliary Data Length 50
5.4.4.3 auxdatae: Auxiliary Data Exists 50
5.4.5 errorcheck:Frame Error Detection Field 51
5.4.5.1 crcrsv: CRC Reserved Bit 51
5.4.5.2 crc2: Cyclic Redundancy Check 2 51
5.5 Bit Stream Constraints 51
6. DECODING THE AC-3 BIT STREAM 51
6.1 Summary of the Decoding Process 51
6.1.1 Input Bit Stream 51
6.1.1.1 Continuous or Burst Input 52

5
Advanced Television Systems Committee, Inc. Document A/52B

6.1.1.2 Byte or Word Alignment 52

6.1.2 Synchronization and Error Detection 53
6.1.3 Unpack BSI, Side Information 53
6.1.4 Decode Exponents 53
6.1.5 Bit Allocation 54
6.1.6 Process Mantissas 54
6.1.7 Decoupling 54
6.1.8 Rematrixing 54
6.1.9 Dynamic Range Compression 54
6.1.10 Inverse Transform 54
6.1.11 Window, Overlap/Add 55
6.1.12 Downmixing 55
6.1.13 PCM Output Buffer 55
6.1.14 Output PCM 55
7. ALGORITHMIC DETAILS 55
7.1 Exponent coding 55
7.1.1 Overview 55
7.1.2 Exponent Strategy 56
7.1.3 Exponent Decoding 57
7.2 Bit Allocation 60
7.2.1 Overview 60
7.2.2 Parametric Bit Allocation 61
7.2.2.1 Initialization 62
7.2.2.1.1 Special Case Processing Step 62
7.2.2.2 Exponent Mapping into PSD 63
7.2.2.3 PSD Integration 63
7.2.2.4 Compute Excitation Function 64
7.2.2.5 Compute Masking Curve 66
7.2.2.6 Apply Delta Bit Allocation 66
7.2.2.7 Compute Bit Allocation 67
7.2.3 Bit Allocation Tables 68
7.3 Quantization and Decoding of Mantissas 75
7.3.1 Overview 75
7.3.2 Expansion of Mantissas for Asymmetric Quantization (6 ≤ bap ≤ 15) 76
7.3.3 Expansion of Mantissas for Symmetrical Quantization (1 ≤ bap ≤ 5) 76
7.3.4 Dither for Zero Bit Mantissas (bap = 0) 77
7.3.5 Ungrouping of Mantissas 78
7.4 Channel Coupling 79
7.4.1 Overview 79
7.4.2 Sub-Band Structure for Coupling 79
7.4.3 Coupling Coordinate Format 80
7.5 Rematrixing 81
7.5.1 Overview 81
7.5.2 Frequency Band Definitions 82
7.5.2.1 Coupling Not in Use 82
7.5.2.2 Coupling in Use, cplbegf > 2 83

6
Digital Audio Compression Standard, Table of Contents 14 June 2005

7.5.2.3 Coupling in use, 2 ≥ cplbegf > 0 83

7.5.2.4 Coupling in Use, cplbegf = 0 83
7.5.3 Encoding Technique 83
7.5.4 Decoding Technique 84
7.6 Dialogue Normalization 84
7.6.1 Overview 84
7.7 Dynamic Range Compression 85
7.7.1 Dynamic Range Control; dynrng, dynrng2 85
7.7.1.1 Overview 86
7.7.1.2 Detailed Implementation 87
7.7.2 Heavy Compression; compr, compr2 88
7.7.2.1 Overview 89
7.7.2.2 Detailed Implementation 89
7.8 Downmixing 90
7.8.1 General Downmix Procedure 91
7.8.2 Downmixing Into Two Channels 94
7.9 Transform Equations and Block Switching 96
7.9.1 Overview 96
7.9.2 Technique 96
7.9.3 Decoder Implementation 97
7.9.4 Transformation Equations 97
7.9.4.1 512-Sample IMDCT Transform 97
7.9.4.2 256-Sample IMDCT Transforms 99
7.9.5 Channel Gain Range Code 102
7.10 Error Detection 102
7.10.1 CRC Checking 102
7.10.2 Checking Bit Stream Consistency 104
8. ENCODING THE AC-3 BIT STREAM 106
8.1 Introduction 106
8.2 Summary of the Encoding Process 107
8.2.1 Input PCM 107
8.2.1.1 Input Word Length 107
8.2.1.2 Input Sample Rate 107
8.2.1.3 Input Filtering 108
8.2.2 Transient Detection 108
8.2.3 Forward Transform 109
8.2.3.1 Windowing 109
8.2.3.2 Time to Frequency Transformation 109
8.2.4 Coupling Strategy 110
8.2.4.1 Basic Encoder 110
8.2.4.2 Advanced Encoder 110
8.2.5 Form Coupling Channel 110
8.2.5.1 Coupling Channel 110
8.2.5.2 Coupling Coordinates 110
8.2.6 Rematrixing 110
8.2.7 Extract Exponents 111

7
Advanced Television Systems Committee, Inc. Document A/52B

8.2.8 Exponent Strategy 111

8.2.9 Dither Strategy 111
8.2.10 Encode Exponents 111
8.2.11 Normalize Mantissas 111
8.2.12 Core Bit Allocation 112
8.2.13 Quantize Mantissas 112
8.2.14 Pack AC-3 Frame 112

Annex A: AC-3 Elementary Streams in the MPEG-2 Multiplex (Normative) 113

A1. SCOPE 113
A2. INTRODUCTION 113
A3. DETAILED SPECIFICATION FOR SYSTEM A (ATSC) 113
A3.1 Stream Type 113
A3.2 Stream ID 113
A3.3 Registration Descriptor 114
A3.4 AC-3 Audio Descriptor 114
A3.5 ISO-639 Language Code 119
A3.6 STD Audio Buffer Size 119
A4. DETAILED SPECIFICATION FOR SYSTEM B (DVB) 119
A4.1 Stream Type 119
A4.2 Stream ID 120
A4.3 Service Information 120
A4.3.1 AC-3 Descriptor 120
A4.3.2 AC-3 Descriptor Syntax 120
A4.3.3 AC-3 Component Types 122
4.4 STD Audio Buffer Size 123
A5. PES CONSTRAINTS 123
A5.1 Encoding 123
A5.2 Decoding 124
A5.3 Byte-Alignment 124

Annex B: Informative References 125

Annex C: AC-3 Karaoke Mode (Informative) 127

C1. SCOPE 127
C2. INTRODUCTION 127
C3. DETAILED SPECIFICATION 128
C3.1 Karaoke Mode Indication 128
C3.2 Karaoke Mode Channel Assignment 128
C3.3 Reproduction of Karaoke Mode Bit Streams 128
C3.3.1 Karaoke Aware Decoders 128
C3.3.2 Karaoke Capable Decoders 129

Annex D: Alternate Bit Stream Syntax (Normative) 131

D1. SCOPE 131

8
Digital Audio Compression Standard, Table of Contents 14 June 2005

D2. SPECIFICATION 131

D2.1 Indication of Alternate Bit Stream Syntax 131
D2.2 Alternate Bit Stream Syntax Specification 131
D2.3 Description of Alternate Syntax Bit Stream Elements 133
D2.3.1 xbsi1e: Extra Bitstream Information #1 Exists 133
D2.3.2 dmixmod: Preferred Stereo Downmix Mode 133
D2.3.3 ltrtcmixlev: Lt/Rt Center Mix Level 133
D2.3.4 ltrtsurmixlev: Lt/Rt Surround Mix Level 134
D2.3.5 lorocmixlev: Lo/Ro Center Mix Level 134
D2.3.6 lorosurmixlev: Lo/Ro Surround Mix Level 134
D2.3.7 xbsi2e: Extra Bit Stream Information #2 Exists 135
D2.3.8 dsurexmod: Dolby Surround EX Mode 135
D2.3.9 dheadphonmod: Dolby Headphone Mode 135
D2.3.10 adconvtyp: A/D Converter Type 136
D2.3.11 xbsi2: Extra Bit Stream Information 136
D2.3.12 encinfo: Encoder Information 136
D3. DECODER PROCESSING 136
D3.1 Compliant Decoder Processing 136
D3.1.1 Two-Channel Downmix Selection 136
D3.1.2 Two-Channel Downmix Processing 136
D3.1.3 Informational Parameter Processing 137
D3.2 Legacy Decoder Processing 137
D4. ENCODER PROCESSING 137
D4.1 Encoder Processing Steps 137
D4.1.1 Dynamic Range Overload Protection Processing 137
D4.2 Encoder Requirements 137
D4.2.1 Legacy Decoder Support 137
D4.2.2 Original Bit Stream Syntax Support 137

Annex E: Enhanced AC-3 Bit Stream Syntax (Normative) 139

E1. SCOPE 139
E2. SPECIFICATION 139
E2.1 Indication of Enhanced AC-3 Bit Stream Syntax 139
E2.2 Syntax Specification 139
E2.2.1 syncinfo: Synchronization Information 140
E2.2.2 bsi: Bit Stream Information 140
E2.2.3 audfrm: Audio Frame 143
E2.2.4 audblk: Audio Block 146
E2.2.5 auxdata: Auxiliary Data 156
E2.2.6 errorcheck: Error Detection Code 156
E2.3 Description of Enhanced AC-3 bit stream elements 156
E2.3.1 bsi: Bit Stream Information 157
E2.3.1.1 strmtyp: Stream Type 157
E2.3.1.2 substreamid: Substream Identification 157
E2.3.1.3 frmsiz: Frame Size 158
E2.3.1.4 fscod: Sample Rate Code 158

9
Advanced Television Systems Committee, Inc. Document A/52B

E2.3.1.5 numblkscod / fscod2: Number of Audio Blocks / Sample Rate Code 2 158
E2.3.1.6 bsid: Bit Stream Identification 158
E2.3.1.7 chanmape: Custom Channel Map Exists 159
E2.3.1.8 chanmap: Custom Channel Map 159
E2.3.1.9 mixmdate: Mixing Meta-Data Exists 159
E2.3.1.10 lfemixlevcode: LFE mix Level Code Exists 160
E2.3.1.11 femixlevcod: LFE Mix Level Code 160
E2.3.1.12 pgmscle: Program Scale Factor Exists 160
E2.3.1.13 pgmscl: Program Scale Factor 160
E2.3.1.14 pgmscl2e: Program Scale Factor #2 Exists 160
E2.3.1.15 pgmscl2: Program Scale Factor #2 160
E2.3.1.16 extpgmscle: External Program Scale Factor Exists 160
E2.3.1.17 extpgmscl: External Program Scale Factor 160
E2.3.1.18 mixdef: Mix Control Type 160
E2.3.1.19 mixdeflen: Length of Mixing Parameter Data Field 161
E2.3.1.20 mixdata: Mixing Parameter Data 161
E2.3.1.21 paninfoe: Pan Information Exists 161
E2.3.1.22 paninfo: Pan Information 161
E2.3.1.23 paninfo2e: Pan Information Exists 161
E2.3.1.24 paninfo2: Pan Information 161
E2.3.1.25 frmmixcnfginfoe: Frame Mixing Configuration Information Exists 161
E2.3.1.26 blkmixcfginfoe: Block Mixing Configuration Information Exists 161
E2.3.1.27 blkmixcfginfo[blk]: Block Mixing Configuration Information 161
E2.3.1.28 infomdate: Informational Meta-Data Exists 161
E2.3.1.29 sourcefscod: Source Sample Rate Code 162
E2.3.1.30 convsync: Converter Synchronization Flag 162
E2.3.1.31 blkid: Block Identification 162
E2.3.2 audfrm – Audio Frame 162
E2.3.2.1 expstre: Exponent Strategy Syntax Enabled 162
E2.3.2.2 ahte: Adaptive Hybrid Transform Enabled 162
E2.3.2.3 snroffststr: SNR Offset Strategy 162
E2.3.2.4 transproce: Transient Pre-Noise Processing Enabled 163
E2.3.2.5 blkswe: Block Switch Syntax Enabled 163
E2.3.2.6 dithflage: Dither Flag Syntax Enabled 163
E2.3.2.7 bamode: Bit Allocation Model Syntax Enabled 163
E2.3.2.8 frmfgaincode: Fast Gain Codes Enabled 163
E2.3.2.9 dbaflde: Delta Bit Allocation Syntax Enabled 163
E2.3.2.10 skipflde: Skip Field Syntax Enabled 163
E2.3.2.11 spxattene: Spectral Extension Attenuation Enabled 163
E2.3.2.12 frmcplexpstr: Frame Based Coupling Exponent Strategy 163
E2.3.2.13 frmchexpstr[ch]: Frame Based Channel Exponent Strategy 163
E2.3.2.14 convexpstre: Converter Exponent Strategy Exists 164
E2.3.2.15 convexpstr[ch]: Converter Channel Exponent Strategy 164
E2.3.2.16 cplahtinu: Coupling Channel AHT in Use 164
E2.3.2.17 chahtinu[ch]: Channel AHT in Use 165
E2.3.2.18 lfeahtinu: LFE Channel AHT in Use 165

10
Digital Audio Compression Standard, Table of Contents 14 June 2005

E2.3.2.19 frmcsnroffst: Frame Coarse SNR Offset 165

E2.3.2.20 frmfsnroffst: Frame Fine SNR Offset 165
E2.3.2.21 chintransproc[ch]: Channel in Transient Pre-Noise Processing 165
E2.3.2.22 transprocloc[ch]: Transient Location Relative to Start of Frame 165
E2.3.2.23 transproclen[ch]: Transient Processing Length 165
E2.3.2.24 chinspxatten[ch]: Channel in Spectral Extension Attenuation Proc. 165
E2.3.2.25 spxattencod[ch]: Spectral Extension Attenuation Code 165
E2.3.2.26 blkstrtinfoe: Block Start Information Exists 165
E2.3.2.27 blkstrtinfo: Block Start Information 165
E2.3.2.28 firstspxcos[ch]: First Spectral Extension Coordinates States 166
E2.3.2.29 firstcplcos[ch]: First Coupling Coordinates States 166
E2.3.2.30 firstcplleak: First Coupling Leak State 166
E2.3.3 audblk: Audio Block 166
E2.3.3.1 spxstre: Spectral Extension Strategy Exists 166
E2.3.3.2 spxinu: Spectral Extension in Use 166
E2.3.3.3 chinspx[ch]: Channel Using Spectral Extension 166
E2.3.3.4 spxstrtf: Spectral Extension Start Copy Frequency Code 166
E2.3.3.5 spxbegf: Spectral Extension Begin Frequency Code 166
E2.3.3.6 spxendf: Spectral Extension End Frequency Code 167
E2.3.3.7 spxbndstrce: Spectral Extension Band Structure Exist 167
E2.3.3.8 spxbndstrc[bnd]: Spectral Extension Band Structure 167
E2.3.3.9 spxcoe[ch]: Spectral Extension Coordinates Exist 167
E2.3.3.10 spxblnd[ch]: Spectral Extension Blend 167
E2.3.3.11 mstrspxco[ch]: Master Spectral Extension Coordinate 168
E2.3.3.12 spxcoexp[ch][bnd]: Spectral Extension Coordinate Exponent 168
E2.3.3.13 spxcomant[ch][bnd]: Spectral Extension Coordinate Mantissa 168
E2.3.3.14 ecplinu: Enhanced Coupling in Use 168
E2.3.3.15 cplbndstrce: Coupling Band Structure Exist 168
E2.3.3.16 ecplbegf: Enhanced Coupling Begin Frequency Code 168
E2.3.3.17 ecplendf: Enhanced Coupling End Frequency Code 169
E2.3.3.18 ecplbndstrce: Enhanced Coupling Band Structure Exists 169
E2.3.3.19 ecplbndstrc[sbnd]: Enhanced Coupling Band Structure 169
E2.3.3.20 ecplangleintrp: Enhanced Coupling Angle Interpolation Flag 170
E2.3.3.21 ecplparam1e[ch]: Enhanced Coupling Parameters 1 Exist 170
E2.3.3.22 ecplparam2e[ch]: Enhanced Coupling Parameters 2 Exist 170
E2.3.3.23 ecplamp[ch][bnd]: Enhanced Coupling Amplitude Scaling 171
E2.3.3.24 ecplangle[ch][bnd]: Enhanced Coupling Angle 171
E2.3.3.25 ecplchaos[ch][bnd]: Enhanced Coupling Chaos 171
E2.3.3.26 ecpltrans[ch]: Enhanced Coupling Transient Present 171
E2.3.3.27 blkfsnroffst: Block Fine SNR Offset 171
E2.3.3.28 fgaincode: Fast Gain Codes Exist 171
E2.3.3.29 convsnroffste: Converter SNR Offset Exists 171
E2.3.3.30 convsnroffst: Converter SNR Offset 171
E2.3.3.31 chgaqmod[ch]: Channel Gain Adaptive Quantization Mode 171
E2.3.3.32 chgaqgain[ch][n]: Channel Gain Adaptive Quantization gain 171
E2.3.3.33 pre_chmant[n][ch][bin]: Pre Channel Mantissas 172

11
Advanced Television Systems Committee, Inc. Document A/52B

E2.3.3.34 cplgaqmod: Coupling Channel Gain Adaptive Quantization Mode 172

E2.3.3.35 cplgaqgain[n]: Coupling Gain Adaptive Quantization Gain 172
E2.3.3.36 pre_cplmant[n][bin]: Pre Coupling Channel Mantissas 172
E2.3.3.37 lfegaqmod: LFE Channel Gain Adaptive Quantization Mode 172
E2.3.3.38 lfegaqgain[n]: LFE Gain Adaptive Quantization Gain 172
E2.3.3.39 pre_lfemant[n][bin]: Pre LFE Channel Mantissas 172
E3. ALGORITHMIC DETAILS 172
E3.1 Glitch-Free Switching Between Different Stream Types 172
E3.2 Error Detection and Concealment 173
E3.3 Adaptive Hybrid Transform Processing 173
E3.3.1 Overview 173
E3.3.2 Bit Stream Helper Variables 173
E3.3.3 Bit Allocation 180
E3.3.3.1 Parametric Bit Allocation 180
E3.3.3.2 Bit Allocation Tables 183
E3.3.4 Quantization 184
E3.3.4.1 Vector Quantization 184
E3.3.4.2 Gain Adaptive Quantization 185
3.3.5 Transform Equations 188
E3.4 Enhanced Channel Coupling 189
E3.4.1 Overview 189
E3.4.2 Sub-Band Structure for Enhanced Coupling 189
E3.4.3 Enhanced coupling tables 191
E3.4.4 Enhanced Coupling Coordinate Format 194
E3.4.5 Enhanced Coupling Processing 195
E3.4.5.1 Process Enhanced Coupling Channel 195
E3.4.5.2 Process Amplitude Parameters 196
E3.4.5.3 Process Angle Parameters 197
E3.4.5.4 Generate Channel Transform Coefficients 200
E3.5 Spectral Extension Processing 200
E3.5.1 Overview 200
E3.5.2 Sub-Band Structure for Spectral Extension 201
E3.5.3 Spectral Extension Coordinate Format 202
E3.5.4 High Frequency Transform Coefficient Synthesis 203
E3.5.4.1 Transform Coefficient Translation 203
E3.5.4.2 Transform Coefficient Noise Blending 204
E3.5.4.2.1 Blending Factor Calculation 204
E3.5.4.2.2 Banded RMS Energy Calculation 205
E3.5.4.2.3 Transform Coefficient Band Border Filtering 205
E3.5.4.2.4 Noise Scaling and Transform Coefficient Blending Calculation 207
E3.5.4.2.5 Blended Transform Coefficient Scaling 208
E3.6 Transient Pre-Noise Processing 208
E3.6.1 Overview 208
E3.6.2 Application of Transient Pre-Noise Processing Data 209
E3.7 Channel and Program Extensions 211
E3.7.1 Overview 211

12
Digital Audio Compression Standard, Table of Contents 14 June 2005

E3.7.2 Decoding a Single Program with Greater than 5.1 Channels 212
E3.7.3 Decoding Multiple Programs with up to 5.1 Channels 212
E3.7.4 Decoding a Mixture of Programs with up to 5.1 Ch and < 5.1 Ch 213
E3.7.5 Dynamic Range Compression for Programs Containing < 5.1 Ch 213
E3.8 LFE Downmixing Decoder Description 214
E4. AHT VECTOR QUANTIZATION TABLES 215

13
Advanced Television Systems Committee, Inc. Document A/52B

Index of Tables

Table 4.1 ATSC Digital Audio Compression Standard Terms 25

Table 5.1 syncinfo Syntax and Word Size 29
Table 5.2 bsi Syntax and Word Size 30
Table 5.3 audioblk Syntax and Word Size 31
Table 5.4 auxdata Syntax and Word Size 35
Table 5.5 errorcheck Syntax and Word Size 35
Table 5.6 Sample Rate Codes 36
Table 5.7 Bit Stream Mode 36
Table 5.8 Audio Coding Mode 37
Table 5.9 Center Mix Level 37
Table 5.10 Surround Mix Level 37
Table 5.11 Dolby Surround Mode 38
Table 5.12 Room Type 39
Table 5.13 Time Code Exists 40
Table 5.14 Master Coupling Coordinate 43
Table 5.15 Number of Rematrixing Bands 44
Table 5.16 Delta Bit Allocation Exists States 47
Table 5.17 Bit Allocation Deltas 47
Table 5.18 Frame Size Code Table (1 word = 16 bits) 50
Table 7.1 Mapping of Differential Exponent Values, D15 Mode 56
Table 7.2 Mapping of Differential Exponent Values, D25 Mode 57
Table 7.3 Mapping of Differential Exponent Values, D45 Mode 57
Table 7.4 Exponent Strategy Coding 57
Table 7.5 LFE Channel Exponent Strategy Coding 58
Table 7.6 Slow Decay Table, slowdec[] 68
Table 7.7 Fast Decay Table, fastdec[] 68
Table 7.8 Slow Gain Table, slowgain[] 68
Table 7.9 dB/Bit Table, dbpbtab[] 69
Table 7.10 Floor Table, floortab[] 69
Table 7.11 Fast Gain Table, fastgain[] 69
Table 7.12 Banding Structure Tables, bndtab[], bndsz[] 70
Table 7.13 Bin Number to Band Number Table, masktab[bin], bin = (10 * A) + B 71
Table 7.14 Log-Addition Table, latab[val], val = (10 * A) + B 72
Table 7.15 Hearing Threshold Table, hth[fscod][band] 73
Table 7.16 Bit Allocation Pointer Table, baptab[] 74
Table 7.17 Quantizer Levels and Mantissa Bits vs. bap 75
Table 7.18 Mapping of bap to Quantizer 76
Table 7.19 bap = 1 (3-Level) Quantization 77
Table 7.20 bap = 2 (5-Level) Quantization 77
Table 7.21 bap = 3 (7-Level) Quantization 77
Table 7.22 bap = 4 (11-Level) Quantization 78
Table 7.23 bap = 5 (15-Level) Quantization 78
Table 7.24 Coupling Sub-Bands 80

14
Digital Audio Compression Standard, Table of Contents 14 June 2005

Table 7.25 Rematrix Banding Table A 82

Table 7.26 Rematrixing Banding Table B 83
Table 7.27 Rematrixing Banding Table C 83
Table 7.28 Rematrixing Banding Table D 83
Table 7.29 Meaning of 3 msb of dynrng 88
Table 7.30 Meaning of 4 msb of compr 90
Table 7.31 LoRo Scaled Downmix Coefficients 95
Table 7.32 LtRt Scaled Downmix Coefficients 95
Table 7.33 Transform Window Sequence (w[addr]), where addr = (10 * A) + B 101
Table 7.34 gainrng Maximum Absolute Value 102
Table 7.35 5/8_framesize Table; Number of Words in the First 5/8 of the Frame 104
Table 7.36 Known Bit Stream Error Conditions 105
Annex A:
Table A3.1 AC-3 Registration Descriptor 114
Table A3.2 AC-3 Audio Descriptor Syntax 115
Table A3.3 Sample Rate Code Table 116
Table A3.4 Bit Rate Code Table 116
Table A3.5 dsurmod Table 116
Table A3.6 num_channels Table 117
Table A3.7 Priority Field Coding 118
Table A4.1 AC-3 Descriptor Syntax 121
Table A4.2 AC-3 component_type Byte Value Assignments 123
Annex C:
Table C3.1 Channel Array Ordering 128
Table C3.3 Default Coefficient Values for Karaoke Capable Decoders 129
Table C3.2 Coefficient Values for Karaoke Aware Decoders 129
Annex D:
Table D2.1 Bit Stream Information; Alternate Bit Stream Syntax 131
Table D2.2 Preferred Stereo Downmix Mode 133
Table D2.3 Lt/Rt Center Mix Level 133
Table D2.4 Lt/Rt Surround Mix Level 134
Table D2.5 Lo/Ro Center Mix Level 134
Table D2.6 Lo/Ro Surround Mix Level 135
Table D2.7 Dolby Surround EX Mode 135
Table D2.8 Dolby Headphone Mode 135
Table D2.9 A/D Converter Type 136
Annex E:
Table E2.1 syncinfo Syntax and Word Size 140
Table E2.2 bsi Syntax and Word Size 140
Table E2.3 audfrm Syntax and Word Size 143
Table E2.4 audblk Syntax and Word Size 146
Table E2.5 auxdata Syntax and Word Size 156
Table E2.6 errorcheck Syntax and Word Size 156
Table E2.7 Stream Type 157

15
Advanced Television Systems Committee, Inc. Document A/52B

Table E2.8 Sample Rate Codes 158

Table E2.9 Number of Audio Blocks Per Syncframe 158
Table E2.10 Reduced Sampling Rates 158
Table E2.11 Custom Channel Map Locations 159
Table E2.12 Mix Control 161
Table E2.13 SNR Offset Strategy 162
Table E2.14 Frame Exponent Strategy Combinations 164
Table E2.15 Default Spectral Extension Banding Structure 167
Table E2.16 Default Coupling Banding Structure 168
Table E2.17 Default Enhanced Coupling Banding Structure 169
Table E3.1 High Efficiency Bit Allocation Pointers, hebaptab[] 183
Table E3.2 Quantizer Type, Quantizer Level, and Mantissa Bits vs. hebap 184
Table E3.3 Gain Adaptive Quantization Modes 186
Table E3.4 Mapping of Gain Elements, gaqmod = 0x3 187
Table E3.5 Gain Adaptive Quantizer Characteristics 187
Table E3.6 Large Mantissa Inverse Quantization (Remapping) Constants 188
Table E3.7 Enhanced Coupling Sub-bands 190
Table E3.8 Enhanced Coupling Start and End Indexes 191
Table E3.9 Sub-band Transform Start Coefficients: ecplsubbndtab[] 192
Table E3.10 Amplitudes: ecplampexptab[], ecplampmanttab[] 193
Table E3.11 Angles: ecplangletab[] 194
Table E3.12 Chaos Scaling: ecplchaostab[] 194
Table E3.13 Spectral Extension Band Table 202
Table E3.14 Spectral Extension Attenuation Table: spxattentab[][] 207
Table E4.1 VQ Table for Hebap 1; 16-bit two’s complement 215
Table E4.2 VQ Table for Hebap 2; 16-bit two’s complement 215
Table E4.3 VQ Table for Hebap 3; 16-bit two’s complement 215
Table E4.5 VQ Table for Hebap 5; 16-bit two’s complement 216
Table E4.4 VQ Table for Hebap 4; 16-bit two’s complement 216
Table E4.6 VQ Table for Hebap 6; 16-bit two’s complement 219
Table E4.7 VQ Table for Hebap 7; 16-bit two’s complement 224

16
Digital Audio Compression Standard, Table of Contents 14 June 2005

Index of Figures

Figure 1.1 Example application of AC-3 to satellite audio transmission. 21

Figure 1.2 The AC-3 encoder. 21
Figure 1.3 The AC-3 decoder. 22
Figure 5.1 AC-3 synchronization frame. 28
Figure 6.1 Flow diagram of the decoding process. 52
Figure 7.1 Example LFSR circuit. 103
Figure 8.1. Flow diagram of the encoding process. 107
Annex E:
Figure E3.1 Flow diagram for GAQ mantissa dequantization. 186
Figure E3.2 Transient pre-noise time scaling synthesis summary. 209
Figure E3.3 Bitstream with a single program of greater than 5.1 channels. 212
Figure E3.4 Bitstream with multiple programs of up to 5.1 channels. 213
Figure E3.5 Bitstream with mixture of programs of up to 5.1 ch and greater than 5.1 ch. 213

17
 ATSC Standard:
Digital Audio Compression Standard, Revision B

1. 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,
“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.

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).
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. The patent holder has, however, filed a statement of
willingness to grant a license under these rights on reasonable and nondiscriminatory terms and conditions to
applicants desiring to obtain such a license. Details may be obtained from the publisher.

Page 19
Advanced Television Systems Committee, Inc. Document A/52B

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/53C, “Digital Television Standard”, references this document and describes
how the audio coding algorithm described herein is applied in the ATSC DTV standard. The ETSI
TR 101 154 document describes how AC-3 is applied in the DVB DTV standard.

1.1 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 1 to 5.1 channels of source
audio from a PCM representation into a serial bit stream at data rates ranging from 32 kbps to 640
kbps. The 0.1 channel refers to a fractional bandwidth channel intended to convey only low
frequency (subwoofer) signals.
A typical application of the algorithm is shown in Figure 1.1. 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 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.

1.2 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

20
Digital Audio Compression Standard 14 June 2005

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 1.1 Example application of AC-3 to satellite audio transmission.

Figure 1.2 The AC-3 encoder.

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 1.2. 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

21
Advanced Television Systems Committee, Inc. Document A/52B

Figure 1.3 The AC-3 decoder.

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.
The actual AC-3 encoder is more complex than indicated in Figure 1.2. 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.

1.3 Decoding
The decoding process is basically the inverse of the encoding process. The decoder, shown in
Figure 1.3, 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

22
Digital Audio Compression Standard 14 June 2005

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 1.3. The following
functions not shown above are included:
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.

2. SCOPE
The normative portions of this standard specify a coded representation of audio information, and
specify the decoding process. Informative information on the encoding process is included. The
coded representation specified herein is suitable for use in digital audio transmission and storage
applications. The coded representation may convey from 1 to 5 full bandwidth audio channels,
along with a low frequency enhancement channel. A wide range of encoded bit-rates is supported
by this specification.
A short form designation of the audio coding algorithm specified in the body of this standard
(whether or not Annex D is included) is “AC-3”. The short form designation of the audio coding
algorithm specified in Annex E is "E-AC-3".

3. REFERENCES

3.1 Normative References

The following documents contain provisions which, through reference in this text, constitute
provisions of this standard. At the time of publication, the editions indicated were valid. All
standards are subject to revision, and parties to agreement based on this standard are encouraged
to investigate the possibility of applying the most recent editions of the documents listed below.
[1] ISO/IEC IS 13818-1, “Information technology – Generic coding of moving pictures and
associated audio information: Systems”, 2000.
[2] ISO 639-2, “Codes for the representation of names of languages – Part 2: Alpha-3 code,”
1998.
[3] ISO/IEC 8859-1:1998, “Information technology -- 8-bit single-byte coded graphic character
sets -- Part 1: Latin alphabet No. 1.”

4. NOTATION, DEFINITIONS, AND TERMINOLOGY

4.1 Compliance Notation

As used in this document, “must” or “shall” denotes a mandatory provision of this standard.
“Should” denotes a provision that is recommended but not mandatory. “May” denotes a feature

23
Advanced Television Systems Committee, Inc. Document A/52B

whose presence does not preclude compliance, and that may or may not be present at the option of
the implementor.

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.
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.

24
Digital Audio Compression Standard 14 June 2005

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 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

25
Advanced Television Systems Committee, Inc. Document A/52B

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
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

26
Digital Audio Compression Standard 14 June 2005

Table 4.1 ATSC Digital Audio Compression Standard Terms (Continued)

Abbreviation Terminology Reference
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]
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

27
Advanced Television Systems Committee, Inc. Document A/52B

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.

Figure 5.1 AC-3 synchronization frame.

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
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 */

28
Digital Audio Compression Standard 14 June 2005

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

Table 5.1 syncinfo Syntax and Word Size

Syntax Word Size
syncinfo()
{
syncword 16
crc1 16
fscod 2
frmsizecod 6
} /* end of syncinfo */

29
Advanced Television Systems Committee, Inc. Document A/52B

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
lngcod2e 1
if(langcod2e) {langcod2} 8
audprodi2e 1
if(audprodi2e)
{
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 */

30
Digital Audio Compression Standard 14 June 2005

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
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]))

31
Advanced Television Systems Committee, Inc. Document A/52B

Table 5.3 audioblk Syntax and Word Size (Continued)

Syntax Word Size
{
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
}
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] */

32
Digital Audio Compression Standard 14 June 2005

Table 5.3 audioblk Syntax and Word Size (Continued)

Syntax Word Size
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 != 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)
{
cplfleak 3
cplsleak 3
}

33
Advanced Television Systems Committee, Inc. Document A/52B

Table 5.3 audioblk Syntax and Word Size (Continued)

Syntax Word Size
}
/* 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)
{
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
}

34
Digital Audio Compression Standard 14 June 2005

Table 5.3 audioblk Syntax and Word Size (Continued)

Syntax Word Size
}
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 */

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.

35
Advanced Television Systems Committee, Inc. Document A/52B

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.

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

36
Digital Audio Compression Standard 14 June 2005

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.

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

37
Advanced Television Systems Committee, Inc. Document A/52B

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
dsurmod is 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 reserved. 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. (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 multiplexing layer, indication of language
within the AC-3 elementary stream was unnecessary, and so was removed from the AC-3 syntax.)

38
Digital Audio Compression Standard 14 June 2005

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).

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 reserved. 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. See lancod, Section 5.4.2.12 above.

39
Advanced Television Systems Committee, Inc. Document A/52B

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.

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.

40
Digital Audio Compression Standard 14 June 2005

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.

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.

41
Advanced Television Systems Committee, Inc. Document A/52B

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.

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

42
Digital Audio Compression Standard 14 June 2005

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
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.

43
Advanced Television Systems Committee, Inc. Document A/52B

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.

44
Digital Audio Compression Standard 14 June 2005

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.

45
Advanced Television Systems Committee, Inc. Document A/52B

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

46
Digital Audio Compression Standard 14 June 2005

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

47
Advanced Television Systems Committee, Inc. Document A/52B

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

48
Digital Audio Compression Standard 14 June 2005

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.

49
Advanced Television Systems Committee, Inc. Document A/52B

Table 5.18 Frame Size Code Table (1 word = 16 bits)

fs = 32 kHz fs = 44.1 kHz fs = 48 kHz
frmsizecod Nominal Bit Rate
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.

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.

50
Digital Audio Compression Standard 14 June 2005

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.).

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.

51
Advanced Television Systems Committee, Inc. Document A/52B

Figure 6.1 Flow diagram of the decoding process.

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
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.

52
Digital Audio Compression Standard 14 June 2005

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.

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.

53
Advanced Television Systems Committee, Inc. Document A/52B

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.

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.

54
Digital Audio Compression Standard 14 June 2005

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
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

55
Advanced Television Systems Committee, Inc. Document A/52B

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.
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

56
Digital Audio Compression Standard 14 June 2005

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

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

57
Advanced Television Systems Committee, Inc. Document A/52B

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

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

58
Digital Audio Compression Standard 14 June 2005

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

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:

59
Advanced Television Systems Committee, Inc. Document A/52B

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.

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

60
Digital Audio Compression Standard 14 June 2005

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
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

61
Advanced Television Systems Committee, Inc. Document A/52B

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<nfchans; ch++)
{
strtmant[ch] = 0;
if(chincpl[ch]) endmant[ch] = 37 + (12 × cplbegf) ; /* channel is coupled */
else endmant[ch] = 37 + (3 × (chbwcod + 12)) ; /* channel is not coupled */
}
/* for coupling channel */
cplstrtmant = 37 + (12 × cplbegf) ;
cplendmant = 37 + [12 × (cplendf + 3)] ;
/* for lfe channel */
lfestartmant = 0 ;
lfeendmant = 7 ;

7.2.2.1.1 Special Case Processing Step

Before continuing with the initialization procedure, all SNR offset parameters from the bit stream
should be evaluated. These include csnroffst, fsnroffst[ch], cplfsnroffst, and lfefsnroffst. If they are all
found to be equal to zero, then all elements of the bit allocation pointer array bap[] should be set to
zero, and no other bit allocation processing is required for the current audio block.
Perform table lookups to determine the values of sdecay, fdecay, sgain, dbknee, and floor from
parameters in the bit stream as follows:
Pseudo Code
sdecay = slowdec[sdcycod] ; /* Table 7.6 */
fdecay = fastdec[fdcycod] /* Table 7.7 */
sgain = slowgain[sgaincod] /* Table 7.8 */
dbknee = dbpbtab[dbpbcod] /* Table 7.9 */
floor = floortab[floorcod] /* Table 7.10 */

Initialize as follows for the uncoupled portion of fbw channel:

Pseudo Code
start = strtmant[ch] ;
end = endmant[ch] ;
lowcomp = 0 ;
fgain = fastgain[fgaincod[ch]]; /* Table 7.11 */
snroffset[ch] = (((csnroffst −15) << 4) + fsnroffst[ch]) << 2 ;

Initialize as follows for coupling channel:

62
Digital Audio Compression Standard 14 June 2005

Pseudo Code
start = cplstrtmant ;
end = cplendmant ;
fgain = fastgain[cplfgaincod] ; /* Table 7.11 */
snroffset = (((csnroffst −15) << 4) + cplfsnroffst) << 2 ;
fastleak = (cplfleak << 8) + 768 ;
slowleak = (cplsleak << 8) + 768 ;

Initialize as follows for lfe channel:

Pseudo Code
start = lfestrtmant ;
end = lfeendmant ;
lowcomp = 0 ;
fgain = fastgain[lfefgaincod] ;
snroffset = (((csnroffst - 15) << 4) + lfefsnroffst) << 2 ;

7.2.2.2 Exponent Mapping into PSD

This step maps decoded exponents into a 13-bit signed log power-spectral density function.
Pseudo Code
for (bin=start; bin<end; bin++)
{
psd[bin] = (3072 - (exp[bin] << 7)) ;
}

Since exp[k] assumes integral values ranging from 0 to 24, the dynamic range of the psd[] values
is from 0 (for the lowest-level signal) to 3072 for the highest-level signal. The resulting function is
represented on a fine-grain, linear frequency scale.

7.2.2.3 PSD Integration

This step of the algorithm integrates fine-grain PSD values within each of a multiplicity of 1/6th
octave bands. Table 7.12 contains the 50 array values for bndtab[] and bndsz. The bndtab[] array gives
the first mantissa number in each band. The bndsz[] array provides the width of each band in
number of included mantissas. Table 7.13 contains the 256 array values for masktab[], showing the
mapping from mantissa number into the associated 1/6 octave band number. These two tables
contain duplicate information, all of which need not be available in an actual implementation.
They are shown here for simplicity of presentation only.
The integration of PSD values in each band is performed with log-addition. The log-addition
is implemented by computing the difference between the two operands and using the absolute
difference divided by 2 as an address into a length 256 lookup table, latab[], shown in Table 7.14.

63
Advanced Television Systems Committee, Inc. Document A/52B

Pseudo Code
j = start ;
k = masktab[start] ;
do
{
lastbin = min(bndtab[k] + bndsz[k], end);
bndpsd[k] = psd[j] ;
j++ ;
for (i = j; i < lastbin; i++)
{
bndpsd[k] = logadd(bndpsd[k], psd[j]) ;
j++ ;
}
k++ ;
}
while (end > 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 ;
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 ;

64
Digital Audio Compression Standard 14 June 2005

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++)
{
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)
{

65
Advanced Television Systems Committee, Inc. Document A/52B

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.
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

66
Digital Audio Compression Standard 14 June 2005

channel (where deltnseg below equals cpldeltnseg) and on each fbw channel (where deltnseg equals
deltnseg[ch]).

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.

67
Advanced Television Systems Committee, Inc. Document A/52B

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

Table 7.8 Slow Gain Table, slowgain[]

Address slowgain[address]
0 0x540
1 0x4d8
2 0x478
3 0x410

68
Digital Audio Compression Standard 14 June 2005

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

69
Advanced Television Systems Committee, Inc. Document A/52B

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

70
Digital Audio Compression Standard 14 June 2005

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

71
Advanced Television Systems Committee, Inc. Document A/52B

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

72
Digital Audio Compression Standard 14 June 2005

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

73
Advanced Television Systems Committee, Inc. Document A/52B

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

74
Digital Audio Compression Standard 14 June 2005

Table 7.17 Quantizer Levels and Mantissa Bits vs. bap

Mantissa Bits
bap Quantizer 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.)

75
Advanced Television Systems Committee, Inc. Document A/52B

Table 7.18 Mapping of bap to Quantizer

Mantissa Bits (qntztab[bap])
bap Quantizer Levels Quantization Type
(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.
The resulting mantissa value is right shifted by the corresponding exponent to generate the
transform coefficient value

76
Digital Audio Compression Standard 14 June 2005

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 Code Mantissa Value
0 –2./3
1 0
2 2./3

Table 7.20 bap = 2 (5-Level) Quantization

Mantissa Code Mantissa Value
0 –4./5
1 –2./5
2 0
3 2./5
4 4./5

Table 7.21 bap = 3 (7-Level) Quantization

Mantissa Code Mantissa Value
0 –6./7
1 –4./7
2 –2./7
3 0
4 2./7
5 4./7
6 6./7

77
Advanced Television Systems Committee, Inc. Document A/52B

Table 7.22 bap = 4 (11-Level) Quantization

Mantissa Code Mantissa Value
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

Table 7.23 bap = 5 (15-Level) Quantization

Mantissa Code Mantissa Value
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.

78
Digital Audio Compression Standard 14 June 2005

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
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-

79
Advanced Television Systems Committee, Inc. Document A/52B

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 lf Cutoff (kHz) hf Cutoff (kHz) lf Cutoff (kHz) hf Cutoff (kHz)
Low tc # High tc #
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.
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.

80
Digital Audio Compression Standard 14 June 2005

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
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

81
Advanced Television Systems Committee, Inc. Document A/52B

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).

Table 7.25 Rematrix Banding Table A

Low Freq (kHz) High Freq (kHz) Low Freq (kHz) High Freq (kHz)
Band # Low Coeff # High Coeff #
fs = 48 kHz fs = 48 kHz fs = 44.1 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 61 252 5.67 23.67 5.21 21.75

82
Digital Audio Compression Standard 14 June 2005

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

Low Freq (kHz) High Freq (kHz) Low Freq (kHz) High Freq (kHz)
Band # Low Coeff # High Coeff #
fs = 48 kHz fs = 48 kHz fs = 44.1 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
Low Freq (kHz) High Freq (kHz) Low Freq (kHz) High Freq (kHz)
Band # Low Coeff # High Coeff #
fs = 48 kHz fs = 48 kHz fs = 44.1 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).

Table 7.28 Rematrixing Banding Table D

Low Freq (kHz) High Freq (kHz) Low Freq (kHz) High Freq (kHz)
Band # Low Coeff # High Coeff #
fs = 48 kHz fs = 48 kHz fs = 44.1 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.

83
Advanced Television Systems Committee, Inc. Document A/52B

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).

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

84
Digital Audio Compression Standard 14 June 2005

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
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.

85
Advanced Television Systems Committee, Inc. Document A/52B

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.
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.

86
Digital Audio Compression Standard 14 June 2005

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
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

87
Advanced Television Systems Committee, Inc. Document A/52B

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
compression which will be applied, consider the dynrng to represent a signed
fractional number, or
X0 . X1 X2 Y3 Y4 Y5 Y6 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

88
Digital Audio Compression Standard 14 June 2005

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.
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

89
Advanced Television Systems Committee, Inc. Document A/52B

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.

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.

90
Digital Audio Compression Standard 14 June 2005

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) */
{
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 ;
}

91
Advanced Television Systems Committee, Inc. Document A/52B

}
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 ;
}
}
}
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 */
{

92
Digital Audio Compression Standard 14 June 2005

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)
{
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 ;
}

93
Advanced Television Systems Committee, Inc. Document A/52B

else if (output_nrear == 1) .
{
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.

94
Digital Audio Compression Standard 14 June 2005

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 Relative Coefficient
Gain
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 dB –4.8 dB +0.3 dB
0.500 0.207 13/64 –13.8 dB –6.0 dB 0.0 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 Relative Coefficient
Gain
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 dB –3.1 dB –0.10 dB

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.

95
Advanced Television Systems Committee, Inc. Document A/52B

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.

96
Digital Audio Compression Standard 14 June 2005

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<N/4; k++)
{
/* Z[k] = (X[N/2-2*k-1] + j * X[2*k]) * (xcos1[k] + j * xsin1[k]) ; */
Z[k]=(X[N/2-2*k-1]*xcos1[k]-X[2*k]*xsin1[k])+j*(X[2*k]*xcos1[k]+X[N/2-2*k-1]*xsin1[k]);
}

Where:
xcos1[k] = –cos (2 p * (8 * k + 1)/(8 * N))
xsin1[k] = –sin (2 p * (8 * k + 1)/(8 * N))

3) Complex IFFT step.

Compute N/4-point complex IFFT of Z(k) to generate complex-valued sequence z(n).

97
Advanced Television Systems Committee, Inc. Document A/52B

Pseudo Code
for(n=0; n<N/4; n++)
{
z[n] = 0 ;
for(k=0; k<N/4; k++)
{
z[n] + = Z[k] * (cos(8*π*k*n/N) + j * sin(8*π*k*n/N)) ;
}
}

4) Post-IFFT complex multiply step.

Compute N/4-point complex multiplication product y(n), n = 0, 1,...N/4–1 as:

Pseudo Code
for(n=0; n<N/4; n++)
{
/* y[n] = z[n] * (xcos1[n] + j * xsin1[n]) ; */
y[n] = (zr[n] * xcos1[n] - zi[n] * xsin1[n]) + j * (zi[n] * xcos1[n] + zr[n] * xsin1[n]) ;
}

Where:
zr[n] = real(z[n])
zi[n] = imag(z[n])
xcos1[n] and xsin1[n] are as defined in step 2 above
5) Windowing and de-interleaving step.
Compute windowed time-domain samples x[n]:

Pseudo Code
for(n=0; n<N/8; n++)
{
x[2*n] = -yi[N/8+n] * w[2*n] ;
x[2*n+1] = yr[N/8-n-1] * w[2*n+1] ;
x[N/4+2*n] = -yr[n] * w[N/4+2*n] ;
x[N/4+2*n+1] = yi[N/4-n-1] * w[N/4+2*n+1] ;
x[N/2+2*n] = -yr[N/8+n] * w[N/2-2*n-1] ;
x[N/2+2*n+1] = yi[N/8-n-1] * w[N/2-2*n-2] ;
x[3*N/4+2*n] = yi[n] * w[N/4-2*n-1] ;
x[3*N/4+2*n+1] = -yr[N/4-n-1] * w[N/4-2*n-2] ;
}

Where:
yr[n] = real(y[n])
yi[n] = imag(y[n])
w[n] is the transform window sequence (see Table 7.33)
6) Overlap and add step.

98
Digital Audio Compression Standard 14 June 2005

The first half of the windowed block is overlapped with the second half of the previous block
to produce PCM samples (the factor of 2 scaling undoes headroom scaling performed in the
encoder):

Pseudo Code
for(n=0; n<N/2; n++)
{
pcm[n] = 2 * (x[n] + delay[n]) ;
delay[n] = x[N/2+n) ;
}

Note that the arithmetic processing in the overlap/add processing must use saturation
arithmetic to prevent overflow (wraparound). Since the output signal consists of the original
signal plus coding error, it is possible for the output signal to exceed 100 percent level even
though the original input signal was less than or equal to 100 percent level.

7.9.4.2 256-Sample IMDCT Transforms

The following equations should be used for computing the inverse transforms in the case of blksw
= 1, indicating the presence of a transient and two 256 sample transforms (N below still equals
512).
1) Define the MDCT transform coefficients = X[k], k = 0, 1,...N/2.
Pseudo Code
for(k=0; k<N/4; k++)
{
X1[k] = X[2*k] ;
X2[k] = X[2*k+1] ;
}

2) Pre-IFFT complex multiply step.

Compute N/8-point complex multiplication products Z1(k) and Z2(k), k = 0, 1,...N/8–1.

Pseudo Code
for(k=0; k<N/8; k++)
{
/* Z1[k] = (X1[N/4-2*k-1] + j * X1[2*k]) * (xcos2[k] + j * xsin2[k]); */
Z1[k]=(X1[N/4-2*k-1]*xcos2[k]-X1[2k]*xsin2[k])+j*(X1[2*k]*xcos2[k]+X1[N/4-2*k-1]*xsin2[k]) ;
/* Z2[k] = (X2[N/4-2*k-1] + j * X2[2*k]) * (xcos2[k] + j * xsin2[k]) ; */
Z2[k]=(X2[N/4-2*k-1]*xcos2[k]-X2[2*k]*xsin2[k])+j*(X2[2*k]*xcos2[k]+X2[N/4-2*k-1]*xsin2[k]) ;
}

Where:
xcos2[k] = -cos(2p*(8*k+1)/(4*N)), xsin2(k) = -sin(2p*(8*k+1)/(4*N))

3) Complex IFFT step.

Compute N/8-point complex IFFTs of Z1[k] and Z2[k] to generate complex-valued sequences
z1[n] and z2[n].

99
Advanced Television Systems Committee, Inc. Document A/52B

Pseudo Code
for(n=0; n<N/8; n++)
{
z1[n] = 0. ;
z2[n] = 0. ;
for(k=0; k<N/8; k++)
{
z1[n] + = Z1[k] * (cos(16*π*k*n/N) + j * sin(16*π*k*n/N)) ;
z2[n] + = Z2[k] * (cos(16*π*k*n/N) + j * sin(16*π*k*n/N)) ;
}
}

4) Post-IFFT complex multiply step:

Compute N/8-point complex multiplication products y1[n] and y2[n], n = 0, 1,...N/8–1.
Pseudo Code
for(n=0; n<N/8; n++)
{
/* y1[n] = z1[n] * (xcos2[n] + j * xsin2[n]) ; */
y1[n] = (zr1[n] * xcos2[n] - zi1[n] * xsin2[n]) + j * (zi1[n] * xcos2[n] + zr1[n] * xsin2[n]) ;
/* y2[n] = z2[n] * (xcos2[n] + j * xsin2[n]) ; */
y2[n] = (zr2[n] * xcos2[n] - zi2[n] * xsin2[n]) + j * (zi2[n] * xcos2[n] + zr2[n] * xsin2[n]) ;
}

Where:
zr1[n] = real(z1[n])
zi1[n] = imag(z1[n])
zr2[n] = real(z2[n])
zi2[n] = imag(z2[n])
xcos2[n] and xsin2[n] are as defined in step 2 above
5) Windowing and de-interleaving step.
Compute windowed time-domain samples x[n].

Pseudo Code
for(n=0; n<N/8; n++)
{
x[2*n] = -yi1[n] * w[2*n] ;
x[2*n+1] = yr1[N/8-n-1] * w[2*n+1] ;
x[N/4+2*n] = -yr1[n] * w[N/4+2*n] ;
x[N/4+2*n+1] = yi1[N/8-n-1] * w[N/4+2*n+1] ;
x[N/2+2*n] = -yr2[n] * w[N/2-2*n-1] ;
x[N/2+2*n+1] = yi2[N/8-n-1] * w[N/2-2*n-2] ;
x[3N/4+2*n] = yi2[n] * w[N/4-2*n-1] ;
x[3N/4+2*n+1] = -yr2[N/8-n-1] * w[N/4-2*n-2] ;
}

Where:
yr1[n] = real(y1[n])
yi1[n] = imag(y1[n])
yr2[n] = real(y2[n])

100
Digital Audio Compression Standard 14 June 2005

yi2[n] = imag(y2[n])
w[n] is the transform window sequence (see Table 7.33)
Table 7.33 Transform Window Sequence (w[addr]), where addr = (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 0.00014 0.00024 0.00037 0.00051 0.00067 0.00086 0.00107 0.00130 0.00157 0.00187
A=1 0.00220 0.00256 0.00297 0.00341 0.00390 0.00443 0.00501 0.00564 0.00632 0.00706
A=2 0.00785 0.00871 0.00962 0.01061 0.01166 0.01279 0.01399 0.01526 0.01662 0.01806
A=3 0.01959 0.02121 0.02292 0.02472 0.02662 0.02863 0.03073 0.03294 0.03527 0.03770
A=4 0.04025 0.04292 0.04571 0.04862 0.05165 0.05481 0.05810 0.06153 0.06508 0.06878
A=5 0.07261 0.07658 0.08069 0.08495 0.08935 0.09389 0.09859 0.10343 0.10842 0.11356
A=6 0.11885 0.12429 0.12988 0.13563 0.14152 0.14757 0.15376 0.16011 0.16661 0.17325
A=7 0.18005 0.18699 0.19407 0.20130 0.20867 0.21618 0.22382 0.23161 0.23952 0.24757
A=8 0.25574 0.26404 0.27246 0.28100 0.28965 0.29841 0.30729 0.31626 0.32533 0.33450
A=9 0.34376 0.35311 0.36253 0.37204 0.38161 0.39126 0.40096 0.41072 0.42054 0.43040
A=10 0.44030 0.45023 0.46020 0.47019 0.48020 0.49022 0.50025 0.51028 0.52031 0.53033
A=11 0.54033 0.55031 0.56026 0.57019 0.58007 0.58991 0.59970 0.60944 0.61912 0.62873
A=12 0.63827 0.64774 0.65713 0.66643 0.67564 0.68476 0.69377 0.70269 0.71150 0.72019
A=13 0.72877 0.73723 0.74557 0.75378 0.76186 0.76981 0.77762 0.78530 0.79283 0.80022
A=14 0.80747 0.81457 0.82151 0.82831 0.83496 0.84145 0.84779 0.85398 0.86001 0.86588
A=15 0.87160 0.87716 0.88257 0.88782 0.89291 0.89785 0.90264 0.90728 0.91176 0.91610
A=16 0.92028 0.92432 0.92822 0.93197 0.93558 0.93906 0.94240 0.94560 0.94867 0.95162
A=17 0.95444 0.95713 0.95971 0.96217 0.96451 0.96674 0.96887 0.97089 0.97281 0.97463
A=18 0.97635 0.97799 0.97953 0.98099 0.98236 0.98366 0.98488 0.98602 0.98710 0.98811
A=19 0.98905 0.98994 0.99076 0.99153 0.99225 0.99291 0.99353 0.99411 0.99464 0.99513
A=20 0.99558 0.99600 0.99639 0.99674 0.99706 0.99736 0.99763 0.99788 0.99811 0.99831
A=21 0.99850 0.99867 0.99882 0.99895 0.99908 0.99919 0.99929 0.99938 0.99946 0.99953
A=22 0.99959 0.99965 0.99969 0.99974 0.99978 0.99981 0.99984 0.99986 0.99988 0.99990
A=23 0.99992 0.99993 0.99994 0.99995 0.99996 0.99997 0.99998 0.99998 0.99998 0.99999
A=24 0.99999 0.99999 0.99999 1.00000 1.00000 1.00000 1.00000 1.00000 1.00000 1.00000
A=25 1.00000 1.00000 1.00000 1.00000 1.00000 1.00000

6) Overlap and add step.

The first half of the windowed block is overlapped with the second half of the previous block
to produce PCM samples (the factor of 2 scaling undoes headroom scaling performed in the
encoder):

Pseudo Code
for(n=0; n<N/2; n++)
{
pcm[n] = 2 * (x[n] + delay[n]) ;
delay[n] = x[N/2+n] ;
}

Note that the arithmetic processing in the overlap/add processing must use saturation
arithmetic to prevent overflow (wraparound). Since the output signal consists of the original
signal plus coding error, it is possible for the output signal to exceed 100 percent level even
though the original input signal was less than or equal to 100 percent level.

101
Advanced Television Systems Committee, Inc. Document A/52B

7.9.5 Channel Gain Range Code

When the signal level is low, the dynamic range of the decoded audio is typically limited by the
wordlength used in the transform computation. The use of longer wordlength improves dynamic
range but increases cost, as the wordlength of both the arithmetic units and the working RAM
must be increased. In order to allow the wordlength of the transform computation to be reduced,
the AC-3 bit stream includes a syntactic element gainrng[ch]. This 2-bit element exists for each
encoded block for each channel.
The gainrng element is a value in the range of 0–3. The value is an indication of the maximum
sample level within the coded block. Each block represents 256 new audio samples and 256
previous audio samples. Prior to the application of the 512 point window, the maximum absolute
value of the 512 PCM values is determined. Based on the maximum value within the block, the
value of gainrng is set as indicated in Table 7.34:

Table 7.34 gainrng Maximum Absolute Value

Maximum Absolute Value (max) gainrng
max ≥ 0.5 0
0.5 > 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
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

102
Digital Audio Compression Standard 14 June 2005

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.

b0 b1 + b2 b3 b13 b14 + b15 +

u(x)

Figure 7.1 Example LFSR circuit.

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
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.

103
Advanced Television Systems Committee, Inc. Document A/52B

Table 7.35 5/8_framesize Table; Number of Words in the First 5/8 of the Frame
fs = 32 kHz fs = 44.1 kHz fs = 48 kHz
frmsizecod Nominal Bit-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
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

104
Digital Audio Compression Standard 14 June 2005

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) ;
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 ;

105
Advanced Television Systems Committee, Inc. Document A/52B

Table 7.36 Known Bit Stream Error Conditions (Continued)

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.

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.

106
Digital Audio Compression Standard 14 June 2005

Figure 8.1. Flow diagram of the encoding process.

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

107
Advanced Television Systems Committee, Inc. Document A/52B

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 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.
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

108
Digital Audio Compression Standard 14 June 2005

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.

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
XD [ k ] –----2- ⎛ -2---π--- ( 2 n + 1 ) ( 2 k + 1 ) + π
-- ( 2 k + 1 ) ( 1 + α)⎞
=
N ∑ x[n ] cos
⎝ 4N 4 ⎠
for 0 ≤ k < N/2
N=0
Where:
α = –1 for the first short transform
0 for the long transform
+1 for the second short transform

109
Advanced Television Systems Committee, Inc. Document A/52B

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; ch<nfchans; ch++) if(blksw[ch]) chincpl[ch] = 0; else chincpl[ch] = 1;

Coupling coordinates for all channels may be transmitted for every other block; i.e. blocks 0,
2, and 4. During blocks 1, 3, and 5, coupling coordinates are reused.

8.2.4.2 Advanced Encoder

More advanced encoders may make use of dynamically variable coupling parameters. The
coupling frequencies may be made variable based on bit demand and on a psychoacoustic model
which compares the audibility of artifacts caused by bit starvation vs. those caused by the
coupling process. Channels with a rapidly time varying power level may be removed from
coupling. Channels with slowly varying power levels may have their coupling coordinates sent
less often. The coupling band structure may be made dynamic.

8.2.5 Form Coupling Channel

8.2.5.1 Coupling Channel

The most basic encoder can form the coupling channel by simply adding all of the individual
channel coefficients together, and dividing by 8. The division by 8 prevents the coupling channel
from exceeding a value of 1. Slightly more sophisticated encoders can alter the sign of individual
channels before adding them into the sum so as to avoid phase cancellations.

8.2.5.2 Coupling Coordinates

Coupling coordinates are formed by taking magnitude ratios within of each coupling band. The
power in the original channel within a coupling band is divided by the power in the coupling
channel within the coupling band, and the square root of this result is then computed. This
magnitude ratio becomes the coupling coordinate. The coupling coordinates are converted to
floating point format and quantized. The exponents for each channel are examined to see if they
can be further scaled by 3, 6, or 9. This generates the 2-bit master coupling coordinate for that
channel. (The master coupling coordinates allow the dynamic range represented by the coupling
coordinate to be increased.)

8.2.6 Rematrixing
Rematrixing is active only in the 2/0 mode. Within each rematrixing band, power measurements
are made on the L, R, L+R, and L–R signals. If the maximum power is found in the L or R
channels, the rematrix flag is not set for that band. If the maximum power is found in the L+R or
L–R signal, then the rematrix flag is set. When the rematrix flag for a band is set, the encoder
codes L+R and L–R instead of L and R. Rematrixing is described in Section 7.5.

110
Digital Audio Compression Standard 14 June 2005

8.2.7 Extract Exponents

The binary representation of each frequency coefficient is examined to determine the number of
leading zeros. The number of leading zeroes (up to a maximum of 24) becomes the initial
exponent value. These exponents are extracted and the exponent sets (one for each block for each
channel, including the coupling channel) are used to determine the appropriate exponent
strategies.

8.2.8 Exponent Strategy

For each channel, the variation in exponents over frequency and time is examined. There is a
tradeoff between fine frequency resolution, fine time resolution, and the number of bits required
to send exponents. In general, when operating at very low bit rates, it is necessary to trade off time
vs. frequency resolution.
In a basic encoder a simple algorithm may be employed. First, look at the variation of
exponents over time. When the variation exceeds a threshold new exponents will be sent. The
exponent strategy used is made dependent on how many blocks the new exponent set is used for. If
the exponents will be used for only a single block, then use strategy D45. If the new exponents
will be used for 2 or 3 blocks, then use strategy D25. If the new exponents will be used for 4, 5, or
6 blocks, use strategy D15.

8.2.9 Dither Strategy

The encoder controls, on a per channel basis, whether coefficients which will be quantized to zero
bits will be reproduced with dither. The intent is to maintain approximately the same energy in the
reproduced spectrum even if no bits are allocated to portions of the spectrum. Depending on the
exponent strategy, and the accuracy of the encoded exponents, it may be beneficial to defeat dither
for some blocks.
A basic encoder can implement a simple dither strategy on a per channel basis. When blksw[ch]
is 1, defeat dither for that block and for the following block.

8.2.10 Encode Exponents

Based on the selected exponent strategy, the exponents of each exponent set are preprocessed.
D25 and D45 exponent strategies require that a single exponent be shared over more than one
mantissa. The exponents will be differentially encoded for transmission in the bit stream. The
difference between successive raw exponents does not necessarily produce legal differential codes
(maximum value of ±2) if the slew rate of the raw exponents is greater than that allowed by the
exponent strategy. Preprocessing adjusts exponents so that transform coefficients that share an
exponent have the same exponent and so that differentials are legal values. The result of this
processing is that some exponents will have their values decreased, and the corresponding
mantissas will have some leading zeroes.
The exponents are differentially encoded to generate the encoded spectral envelope. As part of
the encoder processing, a set of exponents is generated which is equal to the set of exponents
which the decoder will have when it decodes the encoded spectral envelope.

8.2.11 Normalize Mantissas

Each channel's transform coefficients are normalized by left shifting each coefficient the number
of times given by its corresponding exponent to create normalized mantissas. The original binary

111
Advanced Television Systems Committee, Inc. Document A/52B

frequency coefficients are left shifted according to the exponents which the decoder will use.
Some of the normalized mantissas will have leading zeroes. The normalized mantissas are what
are quantized.

8.2.12 Core Bit Allocation

A basic encoder may use the core bit allocation routine with all parameters fixed at nominal
default values.
sdcycod = 2;
fdcycod = 1;
sgaincod = 1;
dbpbcod = 2;
floorcod = 4;
cplfgaincod = 4;
fgaincod[ch] = 4;
lfegaincod = 4;
cplsnroffst = fsnroffst[ch] = lfesnroffst = fineoffset;

Since the bit allocation parameters are static, they are only sent during block 0. Delta bit
allocation is not used, so deltbaie = 0. The core bit allocation routine (described in Section 7.2) is
run, and the coarse and fine SNR offsets are adjusted until all available bits in the frame are used
up. The coarse SNR offset adjusts in 3 dB increments, and the fine offset adjusts in 3/16 dB
increments. Bits are allocated globally from a common bit pool to all channels. The combination
of csnroffst and fineoffset are chosen which uses the largest number of bits without exceeding the
frame size. This involves an iterative process. When, for a given iteration, the number of bits
exceeds the pool, the SNR offset is decreased for the next iteration. On the other hand, if the
allocation is less than the pool, the SNR offset is increased for the next iteration. When the SNR
offset is at its maximum without causing the allocation to exceed the pool, the iterating is
complete. The result of the bit allocation routine are the final values of csnroffst and fineoffset, and
the set of bit allocation pointers (baps). The SNR offset values are included in the bit stream so
that the decoder does not need to iterate.

8.2.13 Quantize Mantissas

The baps are used by the mantissa quantization block. There is a bap for each individual transform
coefficient. Each normalized mantissas is quantized by the quantizer indicated by the
corresponding bap. Asymmetrically quantized mantissas are quantized by rounding to the number
of bits indicated by the corresponding bap. Symmetrically quantized mantissas are quantized
through the use of a table lookup. Mantissas with baps of 1, 2, and 4 are grouped into triples or
duples.

8.2.14 Pack AC-3 Frame

All of the data is packed into the encoded AC-3 frame. Some of the quantized mantissas are
grouped together and coded by a single codeword. The output format is dependent on the
application. The frame may be output in a burst, or delivered as a serial data stream at a constant
rate.

112
 A/52B, Annex A:
AC-3 Elementary Streams in the MPEG-2 Multiplex
(Normative)

A1. SCOPE
This Annex contains specifications on how to combine one or more AC-3 elementary streams into
the ATSC (Recommendation ITU-R BT.1300-1 System A) or DVB (Recommendation ITU-R
BT.1300-1, System B) MPEG-2 “Transport Stream” (ISO/IEC 13818-1).

A2. INTRODUCTION
The AC-3 elementary bit stream is included in an MPEG-2 multiplex bit stream in much the same
way an MPEG-1 audio stream would be included. The AC-3 bit stream is packetized into PES
packets. An MPEG-2 multiplex bit stream containing AC-3 elementary streams must meet all
audio constraints described in the STD model in §3.6 (System A) or §4.4 (System B). It is
necessary to unambiguously indicate that an AC-3 stream is, in fact, an AC-3 stream (and not an
MPEG audio stream). The MPEG-2 standard does not explicitly indicate codes to be used to
indicate an AC-3 stream. Also, the MPEG-2 standard does not have an audio descriptor adequate
to describe the contents of the AC-3 bit stream in the PSI tables.
The AC-3 audio access unit (AU) or presentation unit (PU) is an AC-3 sync frame. The AC-3
sync frame contains 1536 audio samples. The duration of an AC-3 access (or presentation) unit is
32 ms for audio sampled at 48 kHz, approximately 34.83 ms for audio sampled at 44.1 kHz, and
48 ms for audio sampled at 32 kHz.
The items which need to be specified in order to include AC-3 within the MPEG-2 bit stream
are: stream_type, stream_id, AC-3 audio descriptor, and, for system A only, registration descriptor.
The registration descriptor is not required in System B since the AC-3_descriptor is regarded as a
public descriptor in this system. The ISO 639 language descriptor may be employed to indicate
language. Some constraints are placed on the PES layer for the case of multiple audio streams
intended to be reproduced in exact sample synchronism. In System A (ATSC) the AC-3 audio
descriptor is titled “audio_stream_descriptor” while in System B (DVB) the AC-3 audio descriptor is
titled “AC-3 descriptor”. It should be noted that the syntax of these descriptors differs
significantly between the two systems.

A3. DETAILED SPECIFICATION FOR SYSTEM A (ATSC)

A3.1 Stream Type

The value of stream_type for AC-3 shall be 0x81.

A3.2 Stream ID
The value of stream_id in the PES header shall be 0xBD (indicating private_stream_1). Multiple AC-3
streams may share the same value of stream_id since each stream is carried with a unique PID
value. The mapping of values of PID to stream_type is indicated in the transport stream program
map table (PMT).

Page 113
Advanced Television Systems Committee, Inc. Document A/52B

A3.3 Registration Descriptor

The syntax of the AC-3 registration descriptor is shown in Table A3.1. The AC-3 registration
descriptor shall be included in the TS_program_map_section in the program element descriptor loop.
The AC-3 format_identifier shall be 0x41432D33 (“AC-3”).

Table A3.1 AC-3 Registration Descriptor

Syntax No. of Bits Mnemonic
registration_descriptor() {
descriptor_tag 8 uimsbf
descriptor_length 8 uimsbf
format_identifier 32 uimsbf
}
descriptor_tag — 0× 05.
descriptor_length — 0× 04.
format_identifier — 0× 41432D33 (“AC-3”).

A3.4 AC-3 Audio Descriptor

The AC-3_audio_stream_descriptor shall be constructed per Table A3.2 with field meanings and
usages as defined below. This descriptor allows information about individual AC-3 elementary
streams to be included in the program specific information (PSI) tables. This information is useful
to enable decision making as to the appropriate AC-3 stream(s) that are present in a current
broadcast to be directed to the audio decoder, and also to enable the announcement of
characteristics of audio streams that will be included in future broadcasts. Note that horizontal
lines in the table indicate allowable termination points for the descriptor.

114
Digital Audio Compression Standard, Annex A 14 June 2005

Table A3.2 AC-3 Audio Descriptor Syntax

Syntax No. of Bits Mnemonic
AC-3_audio_stream_descriptor() {
descriptor_tag 8 uimsbf
descriptor_length 8 uimsbf
sample_rate_code 3 bslbf
bsid 5 bslbf
bit_rate_code 6 bslbf
surround_mode 2 bslbf
bsmod 3 bslbf
num_channels 4 bslbf
full_svc 1 bslbf
langcod 8 bslbf
if(num_channels==0) /* 1+1 mode */
langcod2 8 bslbf
if(bsmod<2) {
mainid 3 uimsbf
priority 2 bslbf
reserved 3 ‘111’
}
else asvcflags 8 bslbf
textlen 7 uimsbf
text_code 1 bslbf
for(i=0; i<M; i++) {
text[i] 8 bslbf
}
language_flag 1 bslbf
language_flag_2 1 bslbf
reserved 6 ‘111111’
if(language_flag==1) {
language 3*8 uimsbf
}
if(language_flag_2==1) {
language_2 3*8 uimsbf
}
for(i=0; i<N; i++) {
additional_info[i] N× 8 bslbf
}
}

descriptor_tag – The value for the AC-3 descriptor tag is 0x81.

– This is an 8-bit field specifying the number of bytes of the descriptor
descriptor_length
immediately following descriptor_length field.
sample_rate_code – This is a 3-bit field which indicates the sample rate of the encoded audio. The
indication may be of one specific sample rate, or may be of a set of values which include the
sample rate of the encoded audio (see Table A3.3).

115
Advanced Television Systems Committee, Inc. Document A/52B

Table A3.3 Sample Rate Code Table

sample_rate_code Sample Rate (kHz)
‘000’ 48
‘001’ 44.1
‘010’ 32
‘011’ Reserved
‘100’ 48 or 44.1
‘101’ 48 or 32
‘110’ 44.1 or 32
‘111’ 48 or 44.1 or 32

bsid – This is a 5-bit field which is set to the same value as the bsid field in the AC-3 elementary
stream.
– This is a 6-bit field. The lower 5 bits indicate a nominal bit rate. The MSB indicates
bit_rate_code
whether the indicated bit rate is exact (MSB = 0) or an upper limit (MSB = 1) (see Table
A3.4).

Table A3.4 Bit Rate Code Table

Exact Bit Rate Bit Rate Upper Limit
bit_rate_code bit_rate_code
(kbit/s) (kbit/s)
‘000000’ (0.) 32 ‘100000’ (32.) 32
‘000001’ (1.) 40 ‘100001’ (33.) 40
‘000010’ (2.) 48 ‘100010’ (34.) 48
‘000011’ (3.) 56 ‘100011’ (35.) 56
‘000100’ (4.) 64 ‘100100’ (36.) 64
‘000101’ (5.) 80 ‘100101’ (37.) 80
‘000110’ (6.) 96 ‘100110’ (38.) 96
‘000111’ (7.) 112 ‘100111’ (39.) 112
‘001000’ (8.) 128 ‘101000’ (40.) 128
‘001001’ (9.) 160 ‘101001’ (41.) 160
‘001010’ (10.) 192 ‘101010’ (42.) 192
‘001011’ (11.) 224 ‘101011’ (43.) 224
‘001100’ (12.) 256 ‘101100’ (44.) 256
‘001101’ (13.) 320 ‘101101’ (45.) 320
‘001110’ (14.) 384 ‘101110’ (46.) 384
‘001111’ (15.) 448 ‘101111’ (47.) 448
‘010000’ (16.) 512 ‘110000’ (48.) 512
‘010001’ (17.) 576 ‘110001’ (49.) 576
‘010010’ (18.) 640 ‘110010’ (50.) 640

dsurmod – This is a 2-bit field which may be set to the same value as the dsurmod field in the AC-3
elementary stream, or which may be set to ‘00’ (not indicated) (see Table A3.5).

Table A3.5 dsurmod Table

surround_mode Meaning
‘00’ Not indicated
‘01’ NOT Dolby surround encoded
‘10’ Dolby surround encoded
‘11’ Reserved

116
Digital Audio Compression Standard, Annex A 14 June 2005

bsmod – This is a 3-bit field which is set to the same value as the bsmod field in the AC-3
elementary stream.
num_channels – This is a 4-bit field which indicates the number of channels in the AC-3
elementary stream. When the MSB is 0, the lower 3 bits are set to the same value as the acmod
field in the AC-3 elementary stream. When the MSB field is 1, the lower 3 bits indicate the
maximum number of encoded audio channels (counting the lfe channel as 1). If the value of
acmod in the AC-3 elementary stream is ‘000’ (1+1 mode), then the value of num_channels shall
be set to ‘0000’ (see Table A3.6).

Table A3.6 num_channels Table

Audio coding mode Number of
num_channels num_channels
(acmod) encoded channels
‘0000’ 1+1 ‘1000’ 1
‘0001’ 1/0 ‘1001’ ≤2
‘0010’ 2/0 ‘1010’ ≤3
‘0011’ 3/0 ‘1011’ ≤4
‘0100’ 2/1 ‘1100’ ≤5
‘0101’ 3/1 ‘1101’ ≤6
‘0110’ 2/2 ‘1110’ Reserved
‘0111’ 3/2 ‘1111’ Reserved

full_svc – This is a 1-bit field which indicates whether or not this audio service is a full service
suitable for presentation, or whether this audio service is only a partial service which should
be combined with another audio service before presentation. This bit should be set to a “1” if
this audio service is sufficiently complete to be presented to the listener without being
combined with another audio service (for example, a visually impaired service which contains
all elements of the programme; music, effects, dialogue, and the visual content descriptive
narrative). This bit should be set to a “0” if the service is not sufficiently complete to be
presented without being combined with another audio service (e.g., a visually impaired
service which only contains a narrative description of the visual program content and which
needs to be combined with another audio service which contains music, effects, and dialogue).
langcod – This is an 8-bit field which is set to the same value as the langcod field in the AC-3
elementary stream. If the AC-3 elementary stream langcod field is not present, then this 8-bit
field shall be set to 0xFF if present.
langcod2 – This is an 8-bit field which is set to the value of the langcod2 field in the AC-3
elementary stream. If the AC-3 elementary stream langcod2 field is not present, then this 8-bit
field shall be set to 0xFF if present.
Note: The langcod and langcod2 fields are not (that is, are no longer) used to indicate
language. Indication of language is done by means of the ISO 639 language codes
within the AC3_audio_stream_descriptor() or within an MPEG
ISO_639_language_descriptor().

mainid – This is a 3-bit field which contains a number in the range 0–7 which identifies a main
audio service. Each main service should be tagged with a unique number. This value is used as
an identifier to link associated services with particular main services.

117
Advanced Television Systems Committee, Inc. Document A/52B

priority – This is a 2-bit field that indicates the priority of the audio service. This field allows a
Main audio service (bsmod equal to 0 or 1) to be marked as the primary audio service. Other
audio services may be explicitly marked or not specified. Table A3.7 below shows how this
field is encoded.
Table A3.7 Priority Field Coding
Bit Field Meaning
00 reserved
01 Primary Audio
10 Other Audio
11 Not specified

asvcflags – This is an 8-bit field. Each bit (0–7) indicates with which main service(s) this
associated service is associated. The left most bit, bit 7, indicates whether this associated
service may be reproduced along with main service number 7. If the bit has a value of 1, the
service is associated with main service number 7. If the bit has a value of 0, the service is not
associated with main service number 7.
textlen– This is an unsigned integer which indicates the length, in bytes, of a descriptive text field
which follows.
text_code – This is a 1-bit field which indicates how the following text field is encoded. If this bit
is a “1”, the text is encoded as 1-byte characters using the ISO Latin-1 alphabet (ISO 8859-1).
If this bit is a “0”, the text is encoded with 2-byte unicode characters.
text[i] – The text field may contain a brief textual description of the audio service.
language_flag – This is a 1-bit flag that indicates whether or not the 3-byte language field is present
in the descriptor. If this bit is set to ‘1’, then the 3-byte language field is present. If this bit is
set to ‘0”, then the language field is not present.
language_flag_2 – This is a 1-bit flag that indicates whether or not the 3-byte language_2 field is
present in the descriptor. If this bit is set to ‘1’, then the 3-byte language_2 field is present. If
this bit is set to ‘0”, then the language_2 field is not present. This bit shall always be set to ‘0’,
unless the num_channels field is set to ‘0000’ indicating the audio coding mode is 1+1 (dual
mono). If the num_channels field is set to ‘0000’ then this bit may be set to ‘1’ and and the
language_2 field may be included in this descriptor.

language – This field is a 3-byte language code per ISO 639-2/B defining the language of this
audio service. If the AC-3 stream audio coding mode is 1+1 (dual mono), this field indicates
the language of the first channel (channel 1, or “left” channel). The language field shall
contain a three-character code as specified by ISO 639-2/B [2]. Each character is coded into 8
bits according to ISO 8859-1 (ISO Latin-1) and inserted in order into the 24-bit field1. The
coding is identical to that used in the MPEG-2 ISO_639_language_code value in the
ISO_639_language_descriptor specified in ISO/IEC 13818-1.

language_2 – This field is only present if the AC-3 stream audio coding mode is 1+1 (dual mono).
This field is a 3-byte language code per ISO 639-2/B defining the language of the second

1. Note: In the event that there is a single Main service that alternates between different languages, the ISO 639
Language descriptor may be used to communicate that additional information.

118
Digital Audio Compression Standard, Annex A 14 June 2005

channel (channel 2, or “right” channel) in the AC-3 bit stream. The language_2 field shall
contain a three-character code as specified by ISO 639-2/B [2]. Each character is coded into 8
bits according to ISO 8859-1 (ISO Latin-1) and inserted in order into the 24-bit field. The
coding is identical to that used in the MPEG-2 ISO_639_language_code value in the
ISO_639_language_descriptor specified in ISO/IEC 13818-1.

– This is a set of additional bytes filling out the remainder of the descriptor. The
additional_info[j]
purpose of these bytes is not currently defined. This field is provided to allow the ATSC to
extend this descriptor. No other use is permitted.

A3.5 ISO-639 Language Code

The ISO_639_language_code descriptor allows a stream to be tagged with the 24-bit ISO 639
language code.

A3.6 STD Audio Buffer Size

For an MPEG-2 transport stream, the T-STD model defines the main audio buffer size BSn as:
BSn = BSmux + BSdec + BSoh

Where:
BSmux = 736 bytes
BSoh = PES header overhead
BSdec = access unit buffer

MPEG-2 specifies a fixed value for BSn (3584 bytes) and indicates that any excess buffer may
be used for additional multiplexing.
When an AC-3 elementary stream is carried by an MPEG-2 transport stream, the transport
stream shall be compliant with a main audio buffer size of
BSn = BSmux + BSpad + BSdec

Where:
BSmux = 736 bytes
BSpad = 64 bytes

The value of BSdec employed shall be that of the highest bit rate supported by the system (i.e.,
the buffer size is not decreased when the audio bit rate is less than the maximum value allowed by
a specific system). The 64 bytes in BSpad are available for BSoh and additional multiplexing.
This constraint makes it possible to implement decoders with the minimum possible memory
buffer.

A4. DETAILED SPECIFICATION FOR SYSTEM B (DVB)

A4.1 Stream Type

The value of stream_type for an AC-3 elementary stream shall be 0x06 (indicating PES packets
containing private data).

119
Advanced Television Systems Committee, Inc. Document A/52B

A4.2 Stream ID
The value of stream_id in the PES header shall be 0xBD (indicating private_stream_1). Multiple AC-3
streams may share the same value of stream_id since each stream is carried with a unique PID
value. The mapping of values of PID to stream_type is indicated in the transport stream program
map table (PMT).

A4.3 Service Information

4.3.1 AC-3 Descriptor

The AC-3_descriptor identifies an AC-3 audio elementary stream that has been coded in accordance
with this Recommendation. The intended purpose is to provide configuration information for the
IRD. The descriptor is located in the PSI PMT, and used once in a program map section following
the relevant ES_info_length field for any stream containing AC-3.
The descriptor tag provides a unique identification of the presence of the AC-3 elementary
stream. Other optional fields in the descriptor may be used to provide identification of the
component type mode of the AC-3 audio coded in the stream (AC-3_type field) and indicate if the
stream is a main AC-3 audio service (mainid field) or an associated AC-3 service (asvc field).
The descriptor has a minimum length of one byte, but may be longer depending upon the state
of the flags and the additional info loop.

4.3.2 AC-3 Descriptor Syntax

The AC-3 descriptor (constructed per Table A4.1) shall be used in the PSI PMT to identify
streams which carry AC-3 audio. The descriptor is to be located once in a program map section
following the relevant ES_info_length field.

120
Digital Audio Compression Standard, Annex A 14 June 2005

Table A4.1 AC-3 Descriptor Syntax

Syntax No. of Bits Identifier
AC-3_ descriptor(){
descriptor_tag 8 uimsbf
descriptor_length 8 uimsbf
AC-3_type_flag 1 bslbf
bsid_flag 1 bslbf
mainid_flag 1 bslbf
asvc_flag 1 bslbf
reserved 1 bslbf
reserved 1 bslbf
reserved 1 bslbf
reserved 1 bslbf
if (AC-3_type_flag)==1{
AC-3_type 8 uimsbf
}
if (bsid_flag)==1{
bsid 8 uimsbf
{
if (mainid_flag)==1{
mainid 8 uimsbf
}
if (asvc_flag)==1{
asvc 8 bslbf
}
for(i=0;i<N;i++){
additional_info[i] Nx8 uimsbf
}
}

descriptor_tag − The descriptor tag is an 8-bit field which identifies each descriptor. The AC-3
descriptor_tag shall have a value of 0x6A.

descriptor_length − This 8-bit field specifies the total number of bytes of the data portion of the
descriptor following the byte defining the value of this field. The AC-3 descriptor has a
minimum length of one byte but may be longer depending on the use of the optional flags and
the additional_info loop.
AC-3_type_flag − This 1-bit field is mandatory. It should be set to “1” to include the optional AC-
3_type field in the descriptor.

bsid_flag−This 1-bit field is mandatory. It should be set to “1” to include the optional bsid field in
the descriptor.
mainid_flag −This 1-bit field is mandatory. It should be set to “1” to include the optional mainid field
in the descriptor.
asvc_flag−This 1-bit field is mandatory. It should be set to “1” to include the optional asvc field in
the descriptor.
reserved flags −These 1-bit fields are reserved for future use. They should always be set to “0”.

121
Advanced Television Systems Committee, Inc. Document A/52B

AC-3_type − This optional 8-bit field indicates the type of audio carried in the AC-3 elementary
stream. It is set to the same value as the component type field of the component descriptor
(refer to Table A7).
bsid −This optional 8-bit field indicates the AC-3 coding version. The three MSBs should always
be set to “0”. The five LSBs are set to the same value as the bsid field in the AC-3 elementary
stream, ‘01000’ (=8) in the current version of AC-3.
mainid − This optional 8-bit field identifies a main audio service and contains a number in the
range 0–7 which identifies a main audio service. Each main service should be tagged with a
unique number. This value is used as an identifier to link associated services with particular
main services.
asvc − This 8-bit field is optional. Each bit (0–7) identifies with which main service(s) this
associated service is associated. The left most bit, bit 7, indicates whether this associated
service may be reproduced along with main service number 7. If the bit has a value of 1, the
service is associated with main service number 7. If the bit has a value of 0, the service is not
associated with main service number 7.
additional_info −These optional bytes are reserved for future use.

4.3.3 AC-3 Component Types

Table A4.2 shows the assignment of component_type values in the component_descriptor in the case
that the stream_content value is set to 0x04, indicating the reference to an AC-3 stream.

122
Digital Audio Compression Standard, Annex A 14 June 2005

Table A4.2 AC-3 component_type Byte Value Assignments

component_type Byte Values (permitted settings)
Full Description
Reserved Service Type Number of
Service
Status Flag Flags Channels Flags
Flag
b7 b6 b5 b4 b3 b2 b1 b0
1 X X X X X X X Reserved
0 X X X X X X X Interpret b0-b6 as indicated below
1 X X X X X X Decoded audio stream is a full service (suitable
for decoding and presentation to the listener)
0 Decoded audio stream is intended to be
combined with another decoded audio stream
before presentation to the listener
X X X X 0 0 0 Mono
0 0 1 1+1 mode
0 1 0 2 Channel (stereo)
0 1 1 2 Channel Dolby surround encoded (stereo)
1 0 0 Multichannel audio (>2 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)

A4.4 STD Audio Buffer Size

The main audio buffer size (BSn ) shall have a fixed value of 5696 bytes. Refer to ISO/IEC
13818-1 (1996) [1] for the derivation of (BSn ) for audio elementary streams.

A5. PES CONSTRAINTS

This section applies to both System A and System B.

A5.1 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.
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 programme 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

123
Advanced Television Systems Committee, Inc. Document A/52B

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.

A5.2 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.)

A5.3 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.

124
 A/52B, Annex B:
Informative References

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] ITU-R Rec. BT.1300-1, “Service multiplex, transport, and identification methods for digital
terrestrial television broadcasting,” 2000.
[2] 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.
[3] 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.
[4] 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.
[5] 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.
[6] 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.
[7] Vernon, Steve; “Dolby Digital: Audio Coding for Digital Television and Storage
Applications,” AES 17th International Conference: High-Quality Audio Coding, August
1999.
[8] 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.
[9] Rao, R., and P. Yip; Discrete Cosine Transform, Academic Press, Boston, MA, pg. 11, 1990.
[10] Cover, T. M., and J. A. Thomas; Elements of Information Theory, Wiley Series in
Telecommunications, Wiley, New York, NY, pg. 13, 1991.
[11] Gersho, A., and R. M. Gray; Vector Quantization and Signal Compression, Kluwer
Academic Publisher, Boston, MA, pg. 309, 1992.
[12] 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.

Page 125
Advanced Television Systems Committee, Inc. Document A/52B

[13] 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.
[14] 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.
[15] 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.
[16] 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.

126
 A/52B, 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 127
Advanced Television Systems Committee, Inc. Document A/52B

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

Normal Channel Assignment Karaoke Channel Assignment
acmod Audio Coding 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.

128
Digital Audio Compression Standard, Annex C 14 June 2005

Table C3.2 Coefficient Values for Karaoke Aware Decoders

2/0 Reproduction 3/0 Reproduction
Coefficient
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

2/0 Reproduction 3/0 Reproduction
Coefficient
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.

129
 A/52B, 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 A/52B (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
compre 1
if(compre) {compr} 8

Page 131
Advanced Television Systems Committee, Inc. Document A/52B

Table D2.1 Bit Stream Information; Alternate Bit Stream Syntax (Continued)
Syntax Word Size
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
encinfo 1
}
addbsie 1
if(addbsie)
{
addbsil 6
addbsi(addbsil+1)× 8
}
} /* end of bsi */

132
Digital Audio Compression Standard, Annex D 14 June 2005

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 ATSC document
A/52B, 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.

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.

133
Advanced Television Systems Committee, Inc. Document A/52B

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.

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.

134
Digital Audio Compression Standard, Annex D 14 June 2005

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”.

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

135
Advanced Television Systems Committee, Inc. Document A/52B

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.

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.

136
Digital Audio Compression Standard, Annex D 14 June 2005

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 A/52B. The extra bit stream information words
(xbsi1e, xbsi2e, dmixmod, etc.) are interpreted as time code words (timecod1e, timecod1, timecod2e, and
timecod2).
As described in A/52B, 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.

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 A/52B not
including this Annex.

137
 A/52B, Annex E:
Enhanced AC-3 Bit Stream Syntax
(Normative)

E1. SCOPE
This Annex to ATSC document A/52B 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 139
Advanced Television Systems Committee, Inc. Document A/52B

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
}
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) */

140
Digital Audio Compression Standard, Annex E 14 June 2005

Table E2.2 bsi Syntax and Word Size (Continued)

Syntax Word Size
{
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) */
{
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

141
Advanced Television Systems Committee, Inc. Document A/52B

Table E2.2 bsi Syntax and Word Size (Continued)

Syntax Word Size
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
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
}
}

142
Digital Audio Compression Standard, Annex E 14 June 2005

Table E2.2 bsi Syntax and Word Size (Continued)

Syntax Word Size
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 */

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

143
Advanced Television Systems Committee, Inc. Document A/52B

Table E2.3 audfrm Syntax and Word Size (Continued)

Syntax Word Size
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)
{
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}

144
Digital Audio Compression Standard, Annex E 14 June 2005

Table E2.3 audfrm Syntax and Word Size (Continued)

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)
{
/* 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

145
Advanced Television Systems Committee, Inc. Document A/52B

Table E2.3 audfrm Syntax and Word Size (Continued)

Syntax Word Size
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
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

146
Digital Audio Compression Standard, Annex E 14 June 2005

Table E2.4 audblk Syntax and Word Size (Continued)

Syntax Word Size
}
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)
{
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

147
Advanced Television Systems Committee, Inc. Document A/52B

Table E2.4 audblk Syntax and Word Size (Continued)

Syntax Word Size
}
}
}
/* 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
{
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}

148
Digital Audio Compression Standard, Annex E 14 June 2005

Table E2.4 audblk Syntax and Word Size (Continued)

Syntax Word Size
else {ecpl_end_subbnd = spxbegf * 2}
}
ecplbndstrce 1
if (ecplbndstrce)
{
for(sbnd = max(9, ecpl_start_subbnd+1); sbnd < ecpl_end_subbnd; sbnd++)
{
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] */
}
else /* ! chincpl[ch] */

149
Advanced Television Systems Committee, Inc. Document A/52B

Table E2.4 audblk Syntax and Word Size (Continued)

Syntax Word Size
{
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++)
{
ecplangle[ch][bnd] 6
ecplchaos[ch][bnd] 3
}
}
if(ch > firstchincpl) {ecpltrans[ch]} 1
}
else /* !chincpl[ch] */
{
firstcplcos[ch] = 1
}

150
Digital Audio Compression Standard, Annex E 14 June 2005

Table E2.4 audblk Syntax and Word Size (Continued)

Syntax Word Size
} /* 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
}
}
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
}
}

151
Advanced Television Systems Committee, Inc. Document A/52B

Table E2.4 audblk Syntax and Word Size (Continued)

Syntax Word Size
/* 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)
{
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}

152
Digital Audio Compression Standard, Annex E 14 June 2005

Table E2.4 audblk Syntax and Word Size (Continued)

Syntax Word Size
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)
{
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

153
Advanced Television Systems Committee, Inc. Document A/52B

Table E2.4 audblk Syntax and Word Size (Continued)

Syntax Word Size
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
}
}
}
} /* 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)
{
for(n = 0; n < chgaqsections[ch]; n++) {chgaqgain[ch][n]} 5
}
for(bin = 0; bin < nchmant[ch]; bin++)
{
if(chgaqbin[ch][bin])

154
Digital Audio Compression Standard, Annex E 14 June 2005

Table E2.4 audblk Syntax and Word Size (Continued)

Syntax Word Size
{
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}
}
}
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

155
Advanced Television Systems Committee, Inc. Document A/52B

Table E2.4 audblk Syntax and Word Size (Continued)

Syntax Word Size
}
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)
}
lfeahtinu = -1 /* AHT info for this frame has been read – do not read again */
}
}
} /* end of audblk */

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 ATSC A/52B, including Annex D, “Alternate Bit Stream Syntax.”

156
Digital Audio Compression Standard, Annex E 14 June 2005

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.
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.

157
Advanced Television Systems Committee, Inc. Document A/52B

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

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) in this version of this standard. Future modifications of
this standard may define other values. Values of bsid smaller than 16 and greater than 10 will be
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. If this standard is extended by the addition of additional elements or features

158
Digital Audio Compression Standard, Annex E 14 June 2005

that are not compatible with decoders that can decode version 16, a value of bsid greater than 16
will be used. Decoders built to this version of the standard would likely not be able to decode
versions with bsid greater than 16. Additionally, decoders built to this version of the standard will
not be able to decode bit streams with bsid = 9 or 10. Thus, decoders built to this standard 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
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.

159
Advanced Television Systems Committee, Inc. Document A/52B

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).

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.

160
Digital Audio Compression Standard, Annex E 14 June 2005

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”.

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.

161
Advanced Television Systems Committee, Inc. Document A/52B

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.

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 A/52B.

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

162
Digital Audio Compression Standard, Annex E 14 June 2005

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.

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 A/52B, 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 A/52B, while ‘R’ indicates that
exponents from the previous block are reused.

163
Advanced Television Systems Committee, Inc. Document A/52B

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.

Table E2.14 Frame Exponent Strategy Combinations

Audio Block Number
frmcplexpstr
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.

164
Digital Audio Compression Standard, Annex E 14 June 2005

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.

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

165
Advanced Television Systems Committee, Inc. Document A/52B

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.
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.

166
Digital Audio Compression Standard, Annex E 14 June 2005

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 one, spectral extension coordinate information follows. If it is zero, the
spectral extension coordinates from the previous block are used.

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.

167
Advanced Television Systems Committee, Inc. Document A/52B

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 A/52B 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 one, the coupling band structure follows. If it is zero in the first block using
coupling, a default coupling band structure is used. If it is zero 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.

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

168
Digital Audio Compression Standard, Annex E 14 June 2005

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
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.

169
Advanced Television Systems Committee, Inc. Document A/52B

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.
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.

170
Digital Audio Compression Standard, Annex E 14 June 2005

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.

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

171
Advanced Television Systems Committee, Inc. Document A/52B

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.

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.

172
Digital Audio Compression Standard, Annex E 14 June 2005

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).

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.

173
Advanced Television Systems Committee, Inc. Document A/52B

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.

174
Digital Audio Compression Standard, Annex E 14 June 2005

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;
}
}
}

175
Advanced Television Systems Committee, Inc. Document A/52B

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;
}
}
}
}

176
Digital Audio Compression Standard, Annex E 14 June 2005

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.

177
Advanced Television Systems Committee, Inc. Document A/52B

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;
}
}
}

178
Digital Audio Compression Standard, Annex E 14 June 2005

Pseudo Code
for(ch = 0; ch <nfchans; ch ++)
{
if(chahtinu[ch] == 0)
{
chgaqsections[ch] = 0;
}
else
{
switch(chgaqmod[ch])
{
case 0: /* No GAQ gains present */
{
chgaqsections[ch] = 0;
break;
}
case 1: /* GAQ gains 1 and 2 */
case 2: /* GAQ gains 1 and 4 */
{
chgaqsections[ch] = chactivegaqbins[ch]; /* chactivegaqbins[ch] was computed earlier */
break;
}
case 3: /* GAQ gains 1, 2, and 4 */
{
chgaqsections[ch] = chactivegaqbins[ch] / 3;
if (chactivegaqbins[ch] % 3) chgaqsections[ch]++;
break;
}
}
}
}

179
Advanced Television Systems Committee, Inc. Document A/52B

Pseudo Code
if(lfeahtinu == 0)
{
lfegaqsections = 0;
}
else
{
sumgaqbins = 0;
for(bin = 0; bin < lfeendmant; bin++)
{
sumgaqbins += lfegaqbin[bin];
}
switch(lfegaqmod)
{
case 0: /* No GAQ gains present */
{
lfegaqsections = 0;
break;
}
case 1: /* GAQ gains 1 and 2 */
case 2: /* GAQ gains 1 and 4 */
{
lfegaqsections = lfeactivegaqbins; /* lfeactivegaqbins was computed earlier */
break;
}
case 3: /* GAQ gains 1, 2, and 4 */
{
lfegaqsections = lfeactivegaqbins / 3;
if(lfeactivegaqbins % 3) lfegaqsections++;
break;
}
}
}

If the gaqmod bit stream parameter bits are set to 0, conventional scalar quantization is used in
place of GAQ coding. If the gaqmod bits are set to 1 or 2, a 1-bit gain is present for each mantissa
coded with GAQ. If the gaqmod bits are set to 3, the GAQ gains for three individual mantissas are
compositely coded as a 5-bit word.

E3.3.3 Bit Allocation

When AHT is in use for any independent channel, the coupling channel, or the LFE channel,
higher coding efficiency is achieved by allowing quantization noise to be allocated with higher
precision. The high precision allocation is achieved using a combination of a new bit allocation
pointer look up table and vector quantization. The following section describes the changes to the
bit allocation routines defined in the main body of A/52B in order to achieve higher precision
allocation.

E3.3.3.1 Parametric Bit Allocation

If the ahtinu flag is set for any independent channel, the coupling channel, or the LFE channel then
the bit allocation routine for that channel is modified to incorporate the new high efficiency bit
allocation pointers. When AHT is in use, the exponents are first decoded and the PSD, excitation

180
Digital Audio Compression Standard, Annex E 14 June 2005

function, and masking curve are calculated. The delta bit allocation, if present in the bit stream, is
then applied. The final computation of the bit allocation, however, is modified as follows:
The high efficiency bit allocation array (hebap[]) is now computed. The masking curve,
adjusted by the snroffset and then truncated, is subtracted from the fine-grain psd[] array. The
difference is right shifted by 5 bits, limited, and then used as an address into the hebaptab[] to find
the final bit allocation and quantizer type applied to the mantissas. When the hebap[] array is
computed, the hebaptab[] array values shall be as shown in Table E3.1.
At the end of the bit allocation procedure, shown in the following pseudo-code, the hebap[]
array contains a series of 5-bit pointers. The pointers indicate how many bits have been allocated
to each mantissa and the type of quantizer applied to the mantissas. The correspondence between
the hebap pointer and quantizer type and quantizer levels is shown in Table E3.2.
Note that if AHT is not in use for a given independent channel, the coupling channel, or the
LFE channel, then the bit allocation procedure and resulting bap[] arrays for that channel are the
same as described in the main body of A/52B.

181
Advanced Television Systems Committee, Inc. Document A/52B

Pseudo Code
if(ahtinu == 1) /* cplAHTinu, chAHTinu[ch], or lfeAHTinu */
{
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)) ;
hebap[i] = hebaptab[address] ;
i++ ;
}
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) ;
}

182
Digital Audio Compression Standard, Annex E 14 June 2005

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

183
Advanced Television Systems Committee, Inc. Document A/52B

Table E3.2 Quantizer Type, Quantizer Level, and Mantissa Bits vs. hebap
Mantissa
hebap Quantizer Type Levels
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
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.

184
Digital Audio Compression Standard, Annex E 14 June 2005

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
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

185
Advanced Television Systems Committee, Inc. Document A/52B

Figure E3.1 Flow diagram for GAQ mantissa dequantization.

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],
Active hebap Range
cplgaqmod, and GAQ Mode for Frame
(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 A/52B. For example, for a 5-bit
composite gain triplet grpgain:
M1 = truncate (grpgain / 9)
M2 = truncate ((grpgain % 9) / 3)
M3 = (grpgain % 9) % 3

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.

186
Digital Audio Compression Standard, Annex E 14 June 2005

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
Small Large Small Large
Quantizer
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.

187
Advanced Television Systems Committee, Inc. Document A/52B

Table E3.6 Large Mantissa Inverse Quantization (Remapping) Constants

Gk = 1 Gk = 2 Gk = 4
hebap
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 X (k , j ) cos⎛⎜ ⎟
j =0
j

⎝ 12 ⎠ m = 0,1,...,5
Where

⎧ 1 j≠0
R =⎨
j

⎩1 / 2 j=0
and k is the bin index, m is the block index, and j is the AHT transform index.

188
Digital Audio Compression Standard, Annex E 14 June 2005

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.

189
Advanced Television Systems Committee, Inc. Document A/52B

Table E3.7 Enhanced Coupling Sub-bands

enhanced
lf cutoff (kHz) hf cutoff (kHz) lf cutoff (kHz) hf cutoff (kHz)
coupling low tc # high tc #
@ 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.

190
Digital Audio Compression Standard, Annex E 14 June 2005

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.

191
Advanced Television Systems Committee, Inc. Document A/52B

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

192
Digital Audio Compression Standard, Annex E 14 June 2005

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

193
Advanced Television Systems Committee, Inc. Document A/52B

Table E3.11 Angles: ecplangletab[]

ecplangle ecplangletab[ecplangle] ecplangle ecplangletab[ecplangle]
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

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

194
Digital Audio Compression Standard, Annex E 14 June 2005

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.
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 A/52B).

195
Advanced Television Systems Committee, Inc. Document A/52B

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<N/2; n++)
{
pcm[n] = xPREV[n+N/2] + xCURR[n];
pcm[n+N/2] = xCURR[n+N/2] + xNEXT [n];
}

4) Adjust the enhanced coupling channel samples such that the following DFT (FFT) output is an
oddly stacked filterbank (as per the MDCT). The window w[n] is defined in Table 7.33 in the
main body of A/52B.
Pseudo Code
for(n=0; n<N/2; n++)
{
pcm_real[n] = pcm[n] * w[n] * xcos3[n];
pcm_real[n+N/2] = pcm[n+N/2] * w[N/2-n-1] * xcos3[n+N/2];
pcm_imag[n] = pcm[n] * w[n] * xsin3[n];
pcm_imag[n+N/2] = pcm[n+N/2] * w[N/2-n-1] * xsin3[n+N/2];
}

Where:
xcos3[n] = cos(π * n / N) ;
xsin3[n] = -sin(π * n / N) ;

5. Perform a Discrete Fourier Transform (as an FFT) on the complex samples to create the
complex frequency coefficients Z[k], k = 0, 1,…,N–1

N–1
π kn⎞ π kn⎞ ⎞
Z [ k ] = ---1-
N ∑ ( pcm_real [ n ] + j × pcm_imag [ n ] ) × ⎛⎝ cos ⎛⎝ 2------------ ⎛ 2------------
N ⎠ – j × sin ⎝ N ⎠ ⎠
n=0

E3.4.5.2 Process Amplitude Parameters

Amplitude values for each enhanced coupling band [bnd] in each channel [ch] are obtained from the
ecplamp parameters as:

Pseudo Code
if (ecplamp[ch][bnd] == 31)
{
amp[ch][bnd] = 0;
}
else
{
amp[ch][bnd] = ( ecplampmanttab[ecplamp[ch][bnd]] / 32 ) >> ecplampexptab[ecplamp[ch][bnd]];
}

196
Digital Audio Compression Standard, Annex E 14 June 2005

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]
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<ecpl_end_sbnd; sbnd++)
{
if(ecplbndstrc[sbnd] == 0)
{
bnd++;
}
for(bin=ecplsubbndtab[sbnd]; bin<ecplsubbndtab[sbnd+1]; bin++)
{
amp[ch][bin] = amp[ch][bnd];
}
}

E3.4.5.3 Process Angle Parameters

Angle values for each enhanced coupling band [bnd] in each channel [ch] are obtained from the
ecplangle parameters as follows. Each angle has a value in the range –1.0 to 1.0 (representing –π to
π). Arithmetic operations performed on these angles “wrap around” such that the results are
within the range –1.0 to 1.0. The following pseudo code derives the band angle value associated
with a given channel and enhanced coupling angle, ecplangle[ch][bnd].

197
Advanced Television Systems Committee, Inc. Document A/52B

Pseudo Code
if (ch == firstchincpl)
{
angle[ch][bnd] = 0;
}
else
{
angle[ch][bnd] = ecplangletab[ecplangle[ch][bnd]];
}

The above band angle values are used to derive bin angle values associated with individual
transform coefficients in one of two ways depending on the ecplangleintrp flag.
If ecplangleintrp is set to 0, then no interpolation is used and the band angle values are applied to
bin angle values according to the ecplbndstrc[] array.
If ecplangleintrp is set to 1, then the band angle values are converted to bin angle values using
linear interpolation between the centers of each band. The following pseudo code interpolates the
band angles (angle[ch][bnd]) into bin angles (angle[ch][bin]) for channel [ch].
Pseudo Code
if (ecpangleintrp == 1)
{
bin = ecplsubbndtab[ecpl_start_subbnd];
for (bnd = 1; bnd < nbands; bnd++)
{
nbins_prev = nbins_per_bnd_array[bnd-1]; /* array of length nbands containing band sizes */
nbins_curr = nbins_per_bnd_array[bnd];
angle_prev = angle[ch][bnd-1];
angle_curr = angle[ch][bnd];
while ((angle_curr – angle_prev) > 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;

198
Digital Audio Compression Standard, Annex E 14 June 2005

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 */
}
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].

199
Advanced Television Systems Committee, Inc. Document A/52B

• 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
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
copying low frequency transform coefficients, inserting them as high frequency transform

200
Digital Audio Compression Standard, Annex E 14 June 2005

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.

201
Advanced Television Systems Committee, Inc. Document A/52B

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
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

202
Digital Audio Compression Standard, Annex E 14 June 2005

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,
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

203
Advanced Television Systems Committee, Inc. Document A/52B

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++;
}
}

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.

204
Digital Audio Compression Standard, Annex E 14 June 2005

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.
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

205
Advanced Television Systems Committee, Inc. Document A/52B

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].

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];
}
}

206
Digital Audio Compression Standard, Annex E 14 June 2005

Table E3.14 Spectral Extension Attenuation Table: spxattentab[][]

binindex
spxattencod
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.
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.

207
Advanced Television Systems Committee, Inc. Document A/52B

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++;
}
}

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

208
Digital Audio Compression Standard, Annex E 14 June 2005

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.

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

209
Advanced Television Systems Committee, Inc. Document A/52B

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
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)
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)

210
Digital Audio Compression Standard, Annex E 14 June 2005

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.
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.

211
Advanced Television Systems Committee, Inc. Document A/52B

Figure E3.3 Bitstream with a single program of 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.

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

212
Digital Audio Compression Standard, Annex E 14 June 2005

Figure E3.4 Bitstream with multiple programs of up to 5.1 channels.

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.)

Figure E3.5 Bitstream with mixture of programs of up to 5.1 channels and programs of greater
than 5.1 channels.

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

213
Advanced Television Systems Committee, Inc. Document A/52B

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
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
}

214
Digital Audio Compression Standard, Annex E 14 June 2005

E4. AHT VECTOR QUANTIZATION TABLES

Table E4.1 VQ Table for Hebap 1; 16-bit two’s complement

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-
Index (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

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-
Index (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

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5] (16-
Index (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

215
Advanced Television Systems Committee, Inc. Document A/52B

Table E4.4 VQ Table for Hebap 4; 16-bit two’s complement

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
index (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

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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
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

216
Digital Audio Compression Standard, Annex E 14 June 2005

Table E4.5 VQ Table for Hebap 5; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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
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

217
Advanced Television Systems Committee, Inc. Document A/52B

Table E4.5 VQ Table for Hebap 5; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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 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
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

218
Digital Audio Compression Standard, Annex E 14 June 2005

Table E4.5 VQ Table for Hebap 5; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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
127 0x212f 0xfc82 0x4fd6 0xca04 0xbc8d 0x008b

Table E4.6 VQ Table for Hebap 6; 16-bit two’s complement

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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

219
Advanced Television Systems Committee, Inc. Document A/52B

Table E4.6 VQ Table for Hebap 6; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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
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

220
Digital Audio Compression Standard, Annex E 14 June 2005

Table E4.6 VQ Table for Hebap 6; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
75 0xe2ac 0x4df0 0x5bf6 0xd9e7 0x1625 0xf83a
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
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

221
Advanced Television Systems Committee, Inc. Document A/52B

Table E4.6 VQ Table for Hebap 6; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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
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

222
Digital Audio Compression Standard, Annex E 14 June 2005

Table E4.6 VQ Table for Hebap 6; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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
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

223
Advanced Television Systems Committee, Inc. Document A/52B

Table E4.6 VQ Table for Hebap 6; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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
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

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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

224
Digital Audio Compression Standard, Annex E 14 June 2005

Table E4.7 VQ Table for Hebap 7; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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
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

225
Advanced Television Systems Committee, Inc. Document A/52B

Table E4.7 VQ Table for Hebap 7; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
59 0xd8f1 0x26d7 0xc200 0x3286 0xa3b1 0x54c3
60 0x25fa 0x5935 0x2fa0 0x3af3 0x159d 0x12e5
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
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

226
Digital Audio Compression Standard, Annex E 14 June 2005

Table E4.7 VQ Table for Hebap 7; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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
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

227
Advanced Television Systems Committee, Inc. Document A/52B

Table E4.7 VQ Table for Hebap 7; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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
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

228
Digital Audio Compression Standard, Annex E 14 June 2005

Table E4.7 VQ Table for Hebap 7; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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
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

229
Advanced Television Systems Committee, Inc. Document A/52B

Table E4.7 VQ Table for Hebap 7; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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
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

230
Digital Audio Compression Standard, Annex E 14 June 2005

Table E4.7 VQ Table for Hebap 7; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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
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
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

231
Advanced Television Systems Committee, Inc. Document A/52B

Table E4.7 VQ Table for Hebap 7; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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
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

232
Digital Audio Compression Standard, Annex E 14 June 2005

Table E4.7 VQ Table for Hebap 7; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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
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

233
Advanced Television Systems Committee, Inc. Document A/52B

Table E4.7 VQ Table for Hebap 7; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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
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

234
Digital Audio Compression Standard, Annex E 14 June 2005

Table E4.7 VQ Table for Hebap 7; 16-bit two’s complement (Continued)

val[index][0] val[index][1] val[index][2] val[index][3] val[index][4] val[index][5]
Index (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)
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

End of document.

235
Advanced Television Systems Committee, Inc.
1750 K Street, N.W., Suite 1200
Washington, D.C. 20006