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AN163 APPLICATION NOTE
AVR/OUTBOARD DECODER SYSTEMS: APPLICATION CODE USER'S GUIDE FOR THE CS4932X FAMILY
Contents
l l l l l l l l l l l How to Control Common Application Modules
Dolby DigitalTM (AC-3TM) Description DTS Digital SurroundTM Description MPEG-1, Audio Layer 3 (MP3) Description MPEG Multichannel Description AAC Description Crystal Original Surround Description Software Naming Convention Document Strategy Understanding Application Messaging Hardware Configurations Supported by the CS4932X and AC-3, DTS, MP3, MPEG Multichannel, AAC and Crystal Original Surround
(R)
such as:
-- -- -- -- Audio Manager Generalized Bass Manager Pro Logic Module Tone Control
l How to Control Specific Application Modules
such as:
-- -- -- -- AC-3: AC-3 Manager, PCM Processor DTS: DTS Manager MP3: MP3 Manager MPEG Multichannel: MPEG Multichannel Manager -- AAC: AAC Manager -- Crystal Original Surround: C.O.S. Effects Manager C.O.S. PCM Processor
(R)
(R)
DIGITAL SOUND
TM
CRYSTAL
(R)
PROCESSING
Dualzone Specific to AC-3, DTS, MPEG-MC and C.O.S. Code 5 Channels (Ls=Rs) ProLogic Enable & Stereo OR Dolby Surround Encoded Input
5 Discrete Channels Specific to AC-3, DTS, MPEG-MC and AAC Code AC-3, DTS, MP3, MPEG-MC, AAC or PCM Enable Autodetect (AC-3, DTS, MP3, MPEGMC, AAC and PCM) CMPDAT/ DAI
2 Channels (Lt & Rt)
Tone Control Parameters Bass Manager
Pro Logic Decode Up to 5 Discrete Channels Downmix Tone Control (L, C, R)
Downmixed 2 Channel Output (Formatted as either Digital Audio or S/PDIF) Up to 5.1 Discrete Channels
Audio Manager
Downmixed 2 Channel Output (Formatted as either Digital Audio or S/PDIF) Up to 5.1 Discrete Channels
Autodetect Enable AC-3, DTS, MP3, MPEGMC, and AAC Decode / PCM Processing
5.1 Discrete Channels
5 Discrete Channels
Output Mode Control
0.1 (Sub)
LFE Enable
Bass Parameter Output Mode
Volume/Mute Channel Remap Delay
P.O. Box 17847, Austin, Texas 78760 (512) 445 7222 FAX: (512) 445 7581 http://www.cirrus.com
Copyright Cirrus Logic, Inc. 2001 (All Rights Reserved)
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TABLE OF CONTENTS
1. OVERVIEW................................................................................................................................ 7 1.1 Document Strategy ............................................................................................................. 7 1.1.1 Hardware Documentation ...................................................................................... 7 1.1.2 CS493XX Application Code User's Guides ........................................................... 8 1.2 Using the CS4932X .......................................................................................................... 14 1.3 Software Naming Convention........................................................................................... 15 1.4 Understanding the Differences between IBA Application Codes and Full Download Application Codes ............................................................................................................ 15 1.5 Understanding the Feature Set of a Device vs. the Application Code.............................. 17 1.6 Dolby Digital Description .................................................................................................. 18 1.7 DTS Digital Surround Description..................................................................................... 20 1.8 MP3 Description ............................................................................................................... 21 1.9 MPEG Multichannel Description ....................................................................................... 22 1.10 AAC Description ............................................................................................................. 23 1.11 Crystal Original Surround Description ............................................................................ 24 2. HARDWARE CONFIGURATION ............................................................................................ 25 2.1 Supported Input/Output Modes ........................................................................................ 25 2.1.1 DCLK = 86MHz .................................................................................................... 26 2.1.2 Configuring the XMT958/AUDAT3 Data Format.................................................. 26 2.1.3 S/PDIF (AES/EBU) Channel Status Block Control .............................................. 26 2.1.4 S/PDIF (AES/EBU) Validity Bit Control ................................................................ 27
Contacting Cirrus Logic Support
For a complete listing of Direct Sales, Distributor, and Sales Representative contacts, visit the Cirrus Logic web site at: http://www.cirrus.com/corporate/contacts
Dolby Digital, AC-3, Dolby Pro Logic, Dolby Pro Logic II, Dolby Surround, Surround EX, Virtual Dolby Digital and the "AAC" logo are trademarks and the "Dolby Digital" logo, "Dolby Digital with Pro Logic II" logo, "Dolby" and the double-"D" symbol are registered trademarks of Dolby Laboratories Licensing Corporation. DTS, DTS Digital Surround, DTS-ES Extended Surround, DTS Neo:6, and DTS Virtual 5.1 are trademarks and the "DTS", "DTS-ES", "DTS Virtual 5.1" logos are registered trademarks of the Digital Theater Systems Corporation. The "MPEG Logo" is a registered trademark of Philips Electronics N. V. Home THX Cinema and THX are registered trademarks of Lucasfilm Ltd. Surround EX is a jointly developed technology of THX and Dolby Labs, Inc. AAC (Advanced Audio Coding) is an "MPEG-2standard-based" digital audio compression algorithm (offering up 5.1 discrete decoded channels for this implementation) collaboratively developed by AT&T, the Fraunhofer Institute, Dolby Laboratories, and the Sony Corporation. In regards to the MP3 capable functionality of the CS49300 Family DSP (via downloading of mp3_493xxx_vv.ld and mp3e_493xxx_vv.ld application codes) the following statements are applicable: "Supply of this product conveys a license for personal, private and non-commercial use. MPEG Layer-3 audio decoding technology licensed from Fraunhofer IIS and THOMSON Multimedia." MLP and Meridian Lossless Packing are registered trademarks of Meridian Audio Ltd. Harman VMAx is a registered trademark of Harman International. The Logic 7 logo and Logic 7 are registered trademarks of Lexicon. SRS CircleSurround, and SRS TruSurround are trademarks of SRS Labs, Inc. The HDCD logo, HDCD, High Definition Compatible Digital and Pacific Microsonics are either registered trademarks or trademarks of Pacific Microsonics, Inc. in the United States and/or other countries. HDCD technology provided under license from Pacific Microsonics, Inc. This product's software is covered by one or more of the following in the United States: 5,479,168; 5,638,074; 5,640,161; 5,872,531; 5,808,574; 5,838,274; 5,854,600; 5,864,311; and in Australia: 669114; with other patents pending. Intel is a registered trademark of Intel Corporation. Motorola is a registered trademark of Motorola, Inc. I2C is a registered trademark of Philips Semiconductor. Purchase of I2C Components of Cirrus Logic, Inc., or one of its sublicensed Associated Companies conveys a license under the Philips I2C Patent Rights to use those components in a standard I2C system. The "Crystal Logo" and the "Crystal Digital Sound Processing Logo" are registered trademarks of Cirrus Logic, Inc. All other names are trademarks, registered trademarks, or service marks of their respective companies. Preliminary product information describes products which are in production, but for which full characterization data is not yet available. Advance product information describes products which are in development and subject to development changes. Cirrus Logic, Inc. has made best efforts to ensure that the information contained in this document is accurate and reliable. However, the information is subject to change without notice and is provided "AS IS" without warranty of any kind (express or implied). No responsibility is assumed by Cirrus Logic, Inc. for the use of this information, nor for infringements of patents or other rights of third parties. This document is the property of Cirrus Logic, Inc. and implies no license under patents, copyrights, trademarks, or trade secrets. No part of this publication may be copied, reproduced, stored in a retrieval system, or transmitted, in any form or by any means (electronic, mechanical, photographic, or otherwise) without the prior written consent of Cirrus Logic, Inc. Items from any Cirrus Logic website or disk may be printed for use by the user. However, no part of the printout or electronic files may be copied, reproduced, stored in a retrieval system, or transmitted, in any form or by any means (electronic, mechanical, photographic, or otherwise) without the prior written consent of Cirrus Logic, Inc.Furthermore, no part of this publication may be used as a basis for manufacture or sale of any items without the prior written consent of Cirrus Logic, Inc. The names of products of Cirrus Logic, Inc. or other vendors and suppliers appearing in this document may be trademarks or service marks of their respective owners which may be registered in some jurisdictions. A list of Cirrus Logic, Inc. trademarks and service marks can be found at http://www.cirrus.com.
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3. APPLICATION MESSAGING.................................................................................................. 29 3.1 Indexed Module Communication Protocol ........................................................................ 29 3.1.1 Write Session....................................................................................................... 29 3.1.2 Solicited Read Message Format.......................................................................... 30 3.1.3 Unsolicited Read Message Format...................................................................... 31 4. COMMON APPLICATION MODULES .................................................................................... 32 4.1 Audio Manager for Dolby Digital, DTS, MP3, MPEG Multichannel, AAC and Crystal Original Surround.......................................................................................... 33 4.1.1 Audio_Control: (Index 0x00) ................................................................................ 37 4.1.2 Noise_Control (Index 0x01): (Applicable to AC-3 Code Only) ............................. 38 4.1.3 Sampling Frequency Code (Index 0x14) ............................................................. 39 4.2 Generalized Bass Manager for Dolby Digital, DTS, MP3, MPEG Multichannel, AAC, and Crystal Original Surround ................................................................................ 40 4.2.1 Bass Manager Block Diagrams............................................................................ 44 4.2.2 Dolby Digital Bass Management Configurations.................................................. 46 4.3 Unsolicited Messages (Read-Only) .................................................................................. 54 4.3.1 Autodetect Operation ........................................................................................... 55 4.3.2 Special Considerations for Autodetection ............................................................ 58 4.4 Generalized Pro Logic Manager for Dolby Digital, DTS, MPEG Multichannel and AAC .. 59 4.4.1 Using Pro Logic in Normal, Wide and Phantom Modes....................................... 60 4.4.1.1 Normal Mode ....................................................................................... 60 4.4.1.2 Wide Mode........................................................................................... 61 4.4.1.3 Phantom Mode..................................................................................... 61 4.4.2 Pro Logic decode of 48 kHz input PCM (Using AC-3 Application Code) ............. 61 4.5 Generalized Tone Control Manager for Dolby Digital, DTS, MPEG Multichannel and C.O.S. ....................................................................................................................... 63 4.6 Controlling the Level for Treble and Bass Boost/Cut........................................................ 65 4.6.1 Level Settings for +12 dB to -12 dB Tone Control ............................................... 65 4.6.2 Tone Control Look-Up Tables (LUT0, LUT1, LUT2) ............................................ 66 4.7 Controlling the Corner Frequencies of the LPF/HPF for Bass and Treble Control ........... 69 5. AC-3 SPECIFIC MODULES AND OPERATIONS................................................................... 71 5.1 AC-3 Manager .................................................................................................................. 71 5.1.1 OUTPUT_MODE_CONTROL.............................................................................. 75 5.1.2 Downmix Equation for Lt, Rt ................................................................................ 75 5.1.3 AC-3 Stream Information ..................................................................................... 75 5.1.4 Special Considerations For AC-3......................................................................... 76 5.1.5 AC-3 decode with Autodetect .............................................................................. 76 5.1.6 AC-3 with Pro Logic Decode and Autodetect....................................................... 77 5.2 PCM Manager (for AC-3 Application Code) ..................................................................... 78 5.2.1 PCM_CONTROL ................................................................................................. 80 5.2.2 Karaoke Capable Function: ................................................................................. 80 5.2.3 Special Considerations For PCM......................................................................... 81 5.2.4 Special Considerations For Pink/White Noise ..................................................... 81 5.2.5 PCM Pass-through (Using AC-3 Application Code)............................................. 81 5.2.6 Pink/White Noise Generation............................................................................... 82 6. DTS MANAGER AND OPERATIONS..................................................................................... 83 6.1 DTS Manager ................................................................................................................... 83 6.1.1 DTS_CONTROL .................................................................................................. 86 6.1.2 Downmix Equation for Lt, Rt ................................................................................ 86 6.1.3 DTS Stream Information ...................................................................................... 86 6.1.4 DTS Decode with Autodetect............................................................................... 86 6.1.5 Special Considerations For DTS CDs and Autodetection.................................... 88
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7. MP3 MANAGER AND OPERATIONS ..................................................................................... 89 7.1 MP3 Manager ................................................................................................................... 89 7.1.1 MP3_CONTROL .................................................................................................. 90 7.2 MP3 Decode with Autodetect............................................................................................ 90 7.2.1 Special Considerations MP3 Autodetect.............................................................. 91 7.2.2 For System Designers: When to Download MP3 Application code... .................. 93 8. MPEG MULTICHANNEL MANAGER AND OPERATIONS .................................................... 95 8.1 MPEG Multichannel Manager ........................................................................................... 95 8.1.1 MPEG_CONTROL ............................................................................................... 98 8.1.2 Downmix Equation for Lt, Rt ................................................................................ 98 8.1.3 MPEG Stream Information ................................................................................... 98 8.2 MPEG Multichannel Decode with Autodetect ................................................................... 99 9. AAC MANAGER AND OPERATIONS .................................................................................. 100 9.1 AAC Manager ................................................................................................................. 100 9.1.1 AAC_CONTROL ................................................................................................ 102 9.1.2 Downmix Equation for Lt, Rt .............................................................................. 102 9.1.3 AAC Stream Information .................................................................................... 102 9.2 AAC Decode with Autodetect.......................................................................................... 103 9.2.1 Implicit Speaker Mapping ................................................................................... 104 10. CRYSTAL ORIGINAL SURROUND (C.O.S.) SPECIFIC MODULES AND OPERATIONS 105 10.1 C.O.S. Effects Manager ................................................................................................ 105 10.1.1 EFFECTS_MGR_CONTROL........................................................................... 107 10.1.2 Basic Crystal Surround Effects Explanation..................................................... 108 10.2 C.O.S. PCM Processor ................................................................................................. 112 10.2.1 PCM_CONTROL.............................................................................................. 113 10.2.2 Creating Custom Effects Using Advanced Settings ......................................... 114 10.2.2.1 FIR Section: Early Reflections.......................................................... 114 10.2.2.2 Comb Filter Section: After-reflections............................................... 114 10.2.2.3 All-Pass Filter Section: More After-reflections and Decorrelation .... 114 10.2.3 Crystal Original Surround Effects Modes ......................................................... 116 10.2.4 Non - Reverberative Effects ............................................................................. 116 10.2.4.1 Passthru ........................................................................................... 117 10.2.4.2 Mono ................................................................................................ 117 10.2.4.3 Classical ........................................................................................... 117 10.2.4.4 Panorama......................................................................................... 117 10.2.4.5 Movie 1-4.......................................................................................... 117 10.2.4.6 Music 1-4.......................................................................................... 117 10.2.5 Reverberative Effects....................................................................................... 117 10.2.5.1 t1r1a1i1.cfg - Room 1....................................................................... 119 10.2.5.2 t2r3a1i2.cfg - Room 2....................................................................... 119 10.2.5.3 t1r1a2i1.cfg - Theater 1 .................................................................... 119 10.2.5.4 t2r3a2i2.cfg -- Theater 2................................................................... 119 10.2.5.5 t1r1a3i1.cfg -- Hall 1 ......................................................................... 119 10.2.5.6 t2r3a3i2.cfg -- Hall 2 ......................................................................... 119 10.2.5.7 t2r3a4i2.cfg -- Stadium ..................................................................... 119 10.3 Effects Configuration Files ............................................................................................ 119 10.3.1 COEFMAKR.EXE and EFFECTS.EXE ............................................................ 120 10.3.2 Command Line Options for COEFMAKR.EXE................................................. 121
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LIST OF FIGURES
Figure 1. Dolby Digital Block Diagram, Configured for PCM Pass-Through/Pink Noise Generation19 Figure 2. Dolby Digital Block Diagram, Configured for AC-3 Decoding ........................................ 19 Figure 3. DTS Block Diagram ....................................................................................................... 20 Figure 4. MP3 Block Diagram ....................................................................................................... 21 Figure 5. MPEG Multichannel Block Diagram............................................................................... 22 Figure 6. AAC Block Diagram ....................................................................................................... 23 Figure 7. Crystal Original Surround Block Diagram ...................................................................... 24 Figure 8. Filter Topology ............................................................................................................... 44 Figure 9. Subwoofer Summing Module......................................................................................... 45 Figure 10. Bass Manager Processing Unit.................................................................................... 45 Figure 11. Generic Autodetect Flow Chart................................................................................... 57 Figure 12. Typical Download Sequence ....................................................................................... 58 Figure 13. Normal Mode ............................................................................................................... 60 Figure 14. Wide Mode................................................................................................................... 61 Figure 15. Phantom Mode............................................................................................................. 61 Figure 16. Composite Graph of a Frequency Sweep with Bass and Treble ................................. 69 Figure 17. Composite Graph of a Frequency Sweep with Bass ................................................... 70 Figure 18. Composite Graph of a Frequency Sweep with Treble ................................................. 70 Figure 19. AC-3 Decode with Autodetect...................................................................................... 76 Figure 20. DTS decode with Autodetect ....................................................................................... 87 Figure 21. DTS Specific Autodetection Flowchart......................................................................... 88 Figure 22. MP3 Decode with Autodetect....................................................................................... 92 Figure 23. When to Download MP3 Code..................................................................................... 94 Figure 24. MPEG Multichannel decode with Autodetect............................................................... 99 Figure 25. AAC decode with Autodetect ..................................................................................... 103 Figure 26. Illustrated Basic Reverb Theory................................................................................. 110 Figure 27. Block Diagram of Effects Code .................................................................................. 110 Figure 28. Block Diagram of Effects Channel Mixer ................................................................... 111 Figure 29. Block Diagram of Effects Reverberator Module......................................................... 115
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LIST OF TABLES
Table 1. Hardware Configurations Supported By: AC-3, DTS, MP3, MPEG Multichannel, and AAC........................................................................................................................ 25 Table 2. Hardware Configurations Supported By: PCM Pass-Through (AC-3 Code) and Crystal Original Surround....................................................................................... 25 Table 3. Professional Channel Status bytes 0, 1, and 3 ............................................................... 28 Table 5. Write Message Format .................................................................................................... 29 Table 6. Read Command Message Format .................................................................................. 30 Table 7. Read Response Message Format................................................................................... 30 Table 8. Unsolicited Read Message Format ................................................................................. 31 Table 9. Audio Manager ................................................................................................................ 33 Table 10. Bass Manager ............................................................................................................... 40 Table 11. Dolby Configuration 0.................................................................................................... 46 Table 12. Dolby Configuration 1.................................................................................................... 47 Table 13. Dolby Configuration 2.................................................................................................... 48 Table 14. Dolby Alternative Configuration 2.................................................................................. 49 Table 15. Dolby Configuration 3 (Normal - No Sub Out)............................................................... 50 Table 16. Dolby Configuration 3 (Sub Out) ................................................................................... 51 Table 17. Dolby DVD Configuration (Bass to L/R) ........................................................................ 52 Table 18. Dolby DVD Configuration (Bass to Subwoofer)............................................................. 53 Table 19. Unsolicited Messages ................................................................................................... 54 Table 20. Generalized Pro Logic Manager.................................................................................... 59 Table 21. Enabling Pro Logic decode of 48 kHz PCM .................................................................. 62 Table 22. Tone Control.................................................................................................................. 63 Table 23. Look-Up Table 0; "LUT0"............................................................................................... 67 Table 24. Look-Up Table 1; "LUT1"............................................................................................... 67 Table 25. Look-Up Table 2; "LUT2"............................................................................................... 68 Table 26. AC-3 Manager ............................................................................................................... 71 Table 27. Enabling AC-3 Decode with Autodetect ........................................................................ 76 Table 28. Enabling AC-3 with Pro Logic Decode with Autodetect................................................. 77 Table 29. PCM Processor ............................................................................................................. 78 Table 30. Enabling PCM Pass-through of 44.1 kHz PCM............................................................. 81 Table 31. Enabling Pink/White Noise Generation ......................................................................... 82 Table 32. Pink/White Noise Generation for All 5.1 Channel.......................................................... 82 Table 33. DTS Manager ................................................................................................................ 83 Table 34. Enabling DTS Decode with Autodetect ......................................................................... 87 Table 35. MP3 Manager................................................................................................................ 89 Table 36. Enabling MP3 Decode with Autodetect ......................................................................... 91 Table 37. MPEG Manager............................................................................................................. 95 Table 38. Enabling MPEG Decode with Autodetect...................................................................... 99 Table 39. AAC Manager.............................................................................................................. 100 Table 40. Enabling AAC Decode with Autodetect ....................................................................... 103 Table 41. Implicit Speaker Mapping ............................................................................................ 104 Table 42. C.O.S. Effects Manager .............................................................................................. 105 Table 43. C.O.S. PCM Processor ............................................................................................... 112
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1. OVERVIEW
The CS4932X is a sub-family of the CS49300 Family DSPs which are system-on-a-chip solutions for Multichannel audio decompression and digital signal processing. Since the part is RAM-based, a download of application software is required each time the CS4932X is powered up. The CS49300 Family is generally targeted at three different market segments. The broadcast market, where audio/video (A/V) synchronization is always required, the DVD Player market where audio/video (A/V) synchronization is sometimes required, and the outboard decoder markets where audio/video synchronization is not required. Another important differentiation between these applications is the format in which the data will be received by the CS49300 Family. In systems where A/V synchronization is required from the CS4931X, the incoming data is typically PES encoded. In an outboard decoder application, the data typically comes in the IEC61937 format (as specified by the DVD consortium). In DVD players, where A/V synchronization is not required, data is typically delivered in an elementary stream format. An important point to remember is that the CS49300 Family can support all three environments, but different device numbers and downloads are required depending on the input data type. Broadcast applications include (but are not limited to) digital cable and satellite set-top box applications and digital TVs. Data input can be ES or PES in format, while data delivery is either serial or parallel and is typically bursty in nature. DVD-Audio Players are unique in that they may have the additional requirement of MLP(R) (Meridian Lossless Packing) Technology and other compressed audio decoding algorithms that are not needed in the Broadcast market, yet may or may not require A/V synchronization. Data delivery can be ES, PES, or IEC61937 depending on the implementation. Often times, a system may be a hybrid between an outboard decoder and a broadcast system depending on its functionality.
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Outboard decoder applications include stand-alone decoders and audio/video receivers. Where data is delivered by way of a S/PDIF cable where the data is formatted IEC61937 (for compressed data) or IEC60958 (for PCM data). This User's Guide covers application code that provides decoding of compressed data streams such as:
l Dolby Digital (AC-3), delivered via an IEC61937
stream
l DTS and DTS-ES, delivered via an IEC61937
stream; DTS Format-16 and DTS Format-14 (via IEC60958) l MPEG-1, Layer 3 (MP3) delivered via an IEC61937 stream l MPEG-2, Audio Layer 2 Stereo and Multichannel (with and without extension), delivered via an IEC61937 stream l MPEG-2, AAC (Advanced Audio Coding) delivered via an IEC61937 stream.
This guide also covers application code that provides effects processing of Linear PCM delivered in an IEC60958 stream, called Crystal Original Surround.
1.1 Document Strategy
Multiple documents are needed to fully define, understand and implement the functionality of the CS4932X. They can be split up into two basic groups: hardware and application code documentation. Please note that hardware and application code are co-dependent and one can not successfully use the part without an understanding of both. The "ANXXX" notation denotes the application note number under which the respective user's guide was released.
1.1.1 Hardware Documentation CS49300 Family Data Sheet
This document describes the electrical characteristics of the device from timing to base functionality. This is the hardware designers tool to learn the part's electrical and systems requirements.
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1.1.2 CS493XX Application Code User's Guides
The following application notes describe the application codes used with the CS493XX. Whenever an application code user's guide is referred to, it should be assumed that one or more of the below documents are being referenced. This list will grow with each new application released. For the most current list of released user's guides and application codes please refer to the CS4931x/CS49330 Part Matrix vs. Code Matrix and the CS4932x/CS49330 Part Matrix vs. Code Matrix available from the CS49300 Page. All appendices covering proprietary algorithms such as: DTS-ESTM, DTS Neo:6, Dolby Digital with Dolby Pro Logic II, Virtual Dolby DigitalTM, VMAx, SRS CircleSurroundTM, HDCD(R), Logic 7TM, SRS TruSurroundTM, and THX(R)/THX(R) Surround EXTM, in addition to the corresponding device-specific codes are available from Cirrus Logic. All application codes are only available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after the proper IP rights have been confirmed (i.e. - the end-customer has a Dolby Digital License, DTS License, etc.). A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a production-level design. AN161 DVD Audio/Video Systems: Application Code User's Guide for the CS4930X This guide describes all application code (e.g. Dolby Digital; DTS; MPEG-1, Layer 3 (MP3); MPEG-2, Layer 3 (a.k.a. MP3); MPEG-2, Layer 2 Multichannel.) intended specifically for DVD Players. This guide also describes the additional application codes designed for DVD Audio Players such as: Meridian Lossless Packing (with both elementary and DVD Audio Pack support), all supported LPCM formats, in addition to a discussion of audio/video synchronization considerations. All application codes referred to in AN161 are only available for a 90 day evaluation period upon signing of a Crystal Ware Software
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Evaluation Agreement (CWSEA) after the proper IP rights have been confirmed (i.e. - the endcustomer has a MLP License, Dolby Digital License, DTS License, and/or AAC License). A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a production-level design. AN162 Broadcast Systems: Application Code User's Guide for the CS4931X This guide describes all application code (e.g. Dolby Digital; Stereo MPEG-1, Layers 1 and 2, Stereo MPEG-2, Layer 2; and MPEG-2, AAC) designed for broadcast systems such as HDTV and set-top box receivers. This document also provides a discussion of broadcast system considerations and dependencies, such as bursty data delivery, and audio/video synchronization. All application codes referred to in AN162 are only available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after the proper IP rights have been confirmed (i.e. - the end-customer has a Dolby Digital License, and/or AAC License). A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a productionlevel design. AN162A Appendix A to Application Note 162: PCMP (SMPTE 302M) User's Guide for the CS4931X Family and CS49330 The PCMP application code is a SMPTE 302M decoder designed for broadcast applications where PCM is delivered in an MPEG-2 transport stream PES packet. It provides support for AV synchronization, channel change, bitstream parameter reporting and PLL control, among other features offered in other broadcast-based application codes discussed in AN162. In addition, a mutually exclusive PCM pass-through application mode is available with the PCMP release. This feature is similar to the AVR-based AC-3 code that is capable of passing PCM via I2S data delivery. PCMP application code and AN162A are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA). A Crystal Ware
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Licensing Agreement (CWLA) must be signed before these application codes may be used in a production-level design. AN162B Appendix B to Application Note 162: AC-3 + PCM Mixer (AC3N) User's Guide for the CS4931X AC3N - Dolby Digital decoder supporting PES parsing and AV Sync for data that is delivered according to the "PES over I2S" Cirrus Logic proprietary packing technique. Additionally, an IEC60958 Lt, Rt output or IEC61937 compressed output is supported from the AUDAT3/XMT958 pin. AC3N application code and AN162B are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA). A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a productionlevel design. AN162C Appendix C to Application Note 162: Interfacing the TeraLogic TL8XX Digital TV ICs to the CS4931X Audio Decoder ICs using AACT or AAST AN162C includes a brief description of hardware configurations and software feature set supported by the Cougar DTV Development Platform in combination with the CS4931X when using two separate AAC application codes developed exclusively for TeraLogic. These two codes are AACT and AAST. The AACT code offers up to 5.1 Channel decoded outputs for an 5.1 Channel AAC Digital Audio Source (Low Complexity, ADTS, Elementary Stream Data packed according to the IEC61937 specification) delivered via an I 2 S (or LJ) output from the TL85X. The AACT code is also capable of simultaneously mixing PCM via the DAI port on the CS4931X. The AAST code offers up to 2.0 Channel (Lt, Rt downmix) decoded outputs for a 5.1 Channel AAC Digital Audio Source (Low Complexity, ADTS, Elementary Stream Data packed according to the IEC61937 specification) delivered via an I 2 S (or LJ) output from the TL85x. AACT and AAST application codes and AN162C are available for a 90 day
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evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA). A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a production-level design. AN163 AVR/Outboard Decoder Systems: Application Code User's Guide for the CS4932X Family This guide describes all application code (e.g. Dolby Digital; DTS; MPEG-1, Layer 3 (MP3); MPEG-2, Layer 3 (a.k.a. MP3); MPEG-2, Layer 2 Multichannel; MPEG-2, AAC; and Crystal Original Surround) designed for outboard decoder and A/V Receiver systems. All application codes referred to in AN163 are only available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after the proper IP rights have been confirmed (i.e. - the end-customer has a Dolby Digital License, DTS License, and/or AAC License). A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a production-level design. AN163A Appendix A to Application Note 163: Virtual Dolby DigitalTM User's Guide for the CS4932X Family This appendix offers how to control the softwarespecific features displayed in the Virtual Dolby Digital Manager. Release of this document and code requires confirmation of an approved Virtual Dolby Digital license. Please contact Dolby Laboratories Licensing Corporation for licensing information. Virtual Dolby Digital application code and AN163A are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has a Virtual Dolby Digital License and is in good standing with Dolby Laboratories. A Crystal Ware Licensing Agreement (CWLA) must be signed before this application code may be used in a production-level design.
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AN163B Appendix B to Application Note 163: VMAx VirtualTheaterTM User's Guide for the CS4932X Family This appendix offers how to control the softwarespecific features displayed in the VMAx VirtualTheater Manager. Please contact Harman International and Dolby Laboratories Licensing Corporation for licensing information. VMAx VirtualTheater application code and AN163B are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has a VMAx License as well as a Virtual Dolby Digital License and is in good standing with both Harman International and Dolby Laboratories Licensing Corporation. A Crystal Ware Licensing Agreement (CWLA) must be signed before this application code may be used in a production-level design. AN163C Appendix C to Application Note 163: SRS CircleSurroundTM User's Guide for the CS4932X Family and CS49330 This appendix offers how to control the softwarespecific features displayed in the SRS CircleSurround Manager. Release of this document and code is currently only authorized to licensees of SRS CircleSurround technology. Please contact SRS Labs for licensing information. SRS CircleSurround application code and AN163C are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has a SRS CircleSurround License, and is in good standing with SRS Labs. A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a productionlevel design. AN163D Appendix D to Application Note 163: Pacific MicrosonicsTM HDCD(R) User's Guide for the CS4932X Family and CS49330 This appendix offers how to control the softwarespecific features displayed in the HDCD Manager. Release of this document and code is currently only
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authorized to licensees of HDCD technology. Please contact Pacific Microsonics for licensing information. HDCD application code and AN163D are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has an HDCD License and is in good standing with Pacific Microsonics. A Crystal Ware Licensing Agreement (CWLA) must be signed before this application code may be used in a production-level design. AN163E Appendix E to Application Note 163: Logic 5TM 5.1 Channel PCM Enhancement User's Guide for the CS4932X Family This appendix offers how to control the softwarespecific features displayed in the Logic 5 Manager. Release of this document and code is currently only authorized to licensees of Logic 5 technology. Please contact Lexicon, Inc. for licensing information. Logic 5 application code and AN163E are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has a Logic 5 License and is in good standing with Lexicon, Inc. A Crystal Ware Licensing Agreement (CWLA) must be signed before this application code may be used in a production-level design. AN163F Appendix F to Application Note 163: SRS TruSurroundTM User's Guide for the CS4932X Family This appendix offers how to control the softwarespecific features displayed in the SRS TruSurround Manager. Release of this document and code requires confirmation of an approved SRS TruSurround license. Please contact SRS Labs for licensing information. Furthermore, if the version of code that is to be released is based on Virtual Dolby Digital technology, a Virtual Dolby Digital License will be required. Please contact Dolby Laboratories Licensing Corporation for licensing information. If the version of this code is based on DTS technology, a DTS license will be required. Please contact Digital Theater Systems for licensing information. If the version of this code is
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based on MPEG-2, Layer 2 Multichannel technology, no additional licensing is required other than the SRS TruSurround license. All SRS TruSurround application codes and AN163F are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has a SRS TruSurround License, Virtual Dolby Digital License and DTS Virtual 5.1 License and is in good standing with SRS Labs, Dolby Laboratories Licensing Corporation, and DTS Corporation. A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a production-level design. AN163G Appendix G to Application Note 163: Crystal 5.1 Channel Digital Post-Processor User's Guide for the CS49330 This appendix offers how to control the softwarespecific features for a non-license based multichannel PCM digital post-processing code. This code offers the following features: Multichannel PCM input, LR2LsRs copy, Test Noise and Impulse Generator, PCM Mixing, Downmixing, Tone Control and 3-band Parametric EQ control of up to 5 full-bandwidth input channels, Bass Manager and Delay. Crystal 5.1 Channel Digital Post-Processor application code and AN163G are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA). A Crystal Ware Licensing Agreement (CWLA) must be signed before this application code may be used in a production-level design. AN163H Appendix H to Application Note 163: THX(R)/THX(R) Surround EXTM 5.1 Channel Post-Processor User's Guide for the CS49330 This appendix offers how to control the softwarespecific features for THX and THX Surround EX application codes. Release of this document and code(s) requires confirmation of an approved THX license status. Please contact the Mr. Mark Paddack at the THX Division of Lucasfilm, Ltd. for licensing information. Both Home THX Cinema
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and/or THX Surround EX 5.1 Channel PostProcessing application codes and AN163H are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has a Home THX Cinema and/or THX Surround EX License and is in good standing with the THX Division of Lucasfilm, Ltd. A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a production-level design. AN163I Appendix I to Application Note 163: Crystal P.D.F. User's Guide for the CS4932X This appendix offers how to control the softwarespecific features for Crystal P.D.F. (Pro Logic Double-Frequency) application code. This code allows for stereo or Pro Logic encoded PCM Stereo streams with sampling rates up to 96kHz. This code also allows for 1:2 upsampling for all input sampling frequencies up to 48kHz. Digital deemphasis for all input sampling frequencies is also available with dual-precision bass management. A Dolby Pro Logic License is required. Crystal P.D.F. application code and AN163I are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has a Dolby Pro Logic License and is in good standing with Dolby Laboratories Licensing Corporation. A Crystal Ware Licensing Agreement (CWLA) must be signed before this application code may be used in a production-level design. AN163J Appendix J to Application Note 163: Digital/Multimedia Speaker User's Guide for the CS4932X Family This appendix describes the Digital/Multimedia Speaker (D.M.S.) licensed software that can process up to two PCM stereo streams, as well as perform an AC-3 decode or DTS (depending on the version of code requested). It also controls independent 5-channel tone control and 3-band parametric EQ on all full-bandwidth channels. With the appropriate Dolby licenses, codes for
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various virtualization modules are available that support Virtualization of Dolby Digital or Dolby Pro Logic decoded output down to two channels. A Dual-Precision Bass Manager is also available with this code. A Dolby Digital, DTS or additional virtualizer technology license may be required, depending on which application code is required. All Digital/Multimedia application codes and AN163J are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has the proper IP owner's licenses and is in good standing with all respective companies. A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a production-level design. AN163K Appendix K to Application Note 163: 5-Channel, 5-Band Graphic Equalizer User's Guide For the CS4932X Family This appendix describes how to control the licensed application code that runs on the CS4932X Family DSP and offers the designer a 5channel, 5- band graphic equalizer. This graphic equalizer features 24-bits of precision at the backend of an AC-3 decoder (which offers additional Pass-Through). The application code also offers a DTS decoder or an MPEG Multichannel decoder, instead of a Dolby Digital decoder. All application codes with a 5-Band, 5 Channel Graphic EQ and AN163K are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has a Dolby Digital License and DTS Digital Surround License and is in good standing with Dolby Laboratories Licensing Corporation, and the DTS Corporation. A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a production-level design. AN163L Appendix L to Application Note 163: DTS-ES Discrete 6.1 and Matrix 6.1 User's Guide for the CS4932X Family This appendix describes the licensed software designed to run on: the CS49326 and CS49329 which provides DTS Digital Surround 5.1 channel decoding in addition to DTS-ES Discrete 6.1 and DTS-ES Matrix 6.1 decoding. Descriptions for software configuration and in-depth descriptions of application control modules are also included in this document. DTS-ES Discrete 6.1 and Matrix 6.1 application code and AN163L are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has a DTS-ES License, and is in good standing with the DTS Corporation. A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a productionlevel design. AN163M Appendix M to Application Note 163: Dolby Pro Logic II TM PCM Enhancement User's Guide for the CS4932X Family This appendix describes how to control the licensed software designed to run on: the CS4932X Family, which provides Dolby Pro Logic II decoding of 2 channel PCM material and AC-3 DTS, MPEG. To find out about the licensing requirements for using these Dolby features, contact Dolby Laboratories Licensing Corporation for licensing information. All application codes that contain Dolby Pro Logic II decoding technology and AN163M are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has a Dolby Pro Logic II License in addition to a Dolby Digital License and DTS Digital Surround License and is in good standing with Dolby Laboratories Licensing Corporation, and the DTS Corporation. A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a production-level design. AN163N Appendix N to Application Note 163: DTS Neo:6 PCM Enhancement User's Guide for the CS4932X Family
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This appendix describes the licensed software designed to run on: the CS49326 and CS49329 which provides DTS NEO:6 decoding for 2channel PCM. This appendix documents how to enable the DTS NEO:6 functionality. DTS Neo:6 and AN163N are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has a DTS Neo:6 License and is in good standing with the DTS Corporation. A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a productionlevel design. AN163O Appendix H to Application Note 163: THX(R)/THX(R) Surround EXTM 7.1 Channel Post-Processor User's Guide for the CS49330 This appendix offers how to control the softwarespecific features for Home THX Cinema and THX Surround EX 7.1 Channel Post-Processor application codes. This application code is capable of post-processing DTS-ES Discrete 6.1 channel streams. Release of this document and code(s) requires confirmation of an approved THX license status. Please contact Mr. Mark Paddack at the THX Division of Lucasfilm, Ltd. for licensing information. Both Home THX Cinema and/or THX Surround EX 7.1 Channel Post-Processing application codes and AN163O are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has a Home THX Cinema and/or THX Surround EX License and is in good standing with the THX Division of Lucasfilm, Ltd. A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a production-level design. AN163P Appendix P to Application Note 163: Crystal PL2_2FS User's Guide for the CS4932X This appendix offers how to control the softwarespecific features for Crystal PL2_2FS (Pro Logic II Double-Frequency) application code. This code
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allows for stereo or Pro Logic encoded PCM Stereo streams with sampling rates up to 96kHz. This code also allows for 1:2 upsampling for all input sampling frequencies up to 48kHz. Digital deemphasis for all input sampling frequencies is also available with dual-precision bass management. A Dolby Pro Logic II License is required. Crystal PL2_2FS application code and AN163P are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has a Dolby Pro Logic II License and is in good standing with Dolby Laboratories Licensing Corporation. A Crystal Ware Licensing Agreement (CWLA) must be signed before this application code may be used in a production-level design. AN163Q Appendix Q to Application Note 163: Crystal 7.1 Channel Digital Post-Processor User's Guide for the CS49330 This appendix offers how to control the softwarespecific features for a non-license based multichannel PCM digital post-processing code. This code offers the following features: Multichannel PCM input, LR2LsRs copy, Test Noise and Impulse Generator, PCM Mixing, Downmixing, Tone Control and 3-band Parametric EQ control of up to 7 full-bandwidth input channels, Bass Manager and Delay. Crystal 7.1 Channel Digital Post-Processor application code and AN163Q are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA). A Crystal Ware Licensing Agreement (CWLA) must be signed before this application code may be used in a production-level design. AN163R Appendix R to Application Note 163: Crystal Extra Surround 6.1/7.1 Channel Decoder Enhancement User's Guide for the CS4932X This appendix offers how to control all application codes which offer the Crystal Extra Surround 6.1/7.1 Channel Decoder Enhancement. Crystal Extra Surround (CES) algorithm processes the
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surround channels (Ls and Rs) to produce the extra surround channel (BsC) or channels (BsL, BsR) using proprietary matrix techniques developed by Cirrus Logic. The Crystal Extra Surround module can operate in "Matrix Encoded Mode" (e.g. Surround EXTM encoded Dolby DigitalTM streams or DTS-ESTM Extended Surround streams) or "NonMatrix Encoded Mode" (e.g. - normal Dolby Digital, DTS Digital SurroundTM, or MPEG Multichannel streams) which is user controlled. These two different matrix processing modes are available to help provide optimum surround channel steering performance depending on whether the input surround channels (Ls and Rs) are matrix encoded or not. All Crystal Extra Surround enabled application codes and AN163R are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA). A Crystal Ware Licensing Agreement (CWLA) must be signed before these application codes may be used in a productionlevel design. AN163S Appendix S to Application Note 163: Logic 7TM 7.1 Channel PCM Enhancement User's Guide for the CS4932X Family This appendix offers how to control the softwarespecific features displayed in the Logic 7 Manager. Release of this document and code is currently only authorized to licensees of Logic 7 technology. Please contact Lexicon, Inc. for licensing information. Logic 7 application code and AN163S are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has a Logic 7 License and is in good standing with Lexicon, Inc. A Crystal Ware Licensing Agreement (CWLA) must be signed before this application code may be used in a production-level design. AN163T Appendix T to Application Note 163: SRS CircleSurround IITM User's Guide for the CS4932X Family and CS49330 This appendix offers how to control the softwarespecific features displayed in the SRS CircleSurround II Manager. Release of this document and code is currently only authorized to licensees of SRS CircleSurround II technology. Please contact SRS Labs for licensing information. SRS CircleSurround II application code and AN163T are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA) after Cirrus Logic has confirmed that the end-user has a SRS CircleSurround II License, and is in good standing with SRS Labs. A Crystal Ware Licensing Agreement (CWLA) must be signed before this application code may be used in a production-level design. AN163U Appendix U to Application Note 163: Crystal DVD Audio Bass Management User's Guide for the CS4932X Family and CS49330 This appendix offers how to control the softwarespecific features of the Crystal DVD Audio Bass Manager application code in addition to to showing a system block diagram of how to deliver 6 channels of 96kHz, 24-bit audio or 2 channels of 192kHz, 24-bit audio into the DSP from 3 stereo, high-performance A/Ds. Crystal DVD Audio Bass Management application code and AN163U are available for a 90 day evaluation period upon signing of a Crystal Ware Software Evaluation Agreement (CWSEA). A Crystal Ware Licensing Agreement (CWLA) must be signed before this application code may be used in a production-level design.
1.2 Using the CS4932X
No matter what application is being used on the chip, the following four steps are always followed to use the CS4932X in system. Please refer to the associated documents (seen below in parentheses). 1) Reset and/or Download Code. (CS49300 Family Datasheet) 2) Hardware Configuration. (CS49300 Family Datasheet) 3) Application Configuration. (AN163)
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4) Kickstart - This is the "Go" command to the CS4932X once the system is properly configured. (AN163) CS49326, and CS49328 (DTS and COS only). The IBA code must be downloaded just like any other full download code following the instructions in the CS49300 Family Datasheet to initiate the respective application. Please see Section 1.4 "Understanding the Differences between IBA Application Codes and Full Download Application Codes" on page 15.
1.3 Software Naming Convention
To keep up with the different applications, parts, hardware revisions and software revisions the following naming convention has been adopted for the CS4932X software: ANAME_CCCCCC_VV.LD where: ANAME = Application Name = 2...12 chars (can include "_"s) CCCCCC = Chip Number = 6 chars NOTE: This is exactly what is marked on the chip except "x" wherever wildcard applies VV = Version Number = 2 digits This document uses the terms Dolby Digital and AC-3 interchangeably. For the AC-3 Section, CS49325, 326, and 329 has been replaced with CS4932X. For the DTS Section, CS49326 and 329 have been replaced with CS4932X. For the MPEG Multichannel Section, CS49325, 326, 329 and 330 have been replaced with CS493XX. For the MP3 Section, CS49325, 326, 329, and 330 have been replaced with CS493XX. For the AAC Section, CS49329 is the only device applicable. For the C.O.S. Section, CS49325, 326, 329, and 330 have been replaced with CS493XX. Unless otherwise specified, CS4932X should be interpreted as applying to the CS49325, 326, 328 and 329. Please refer to Figure 1 and Figure 2 for more detailed information. It should be noted that some AC-3 codes (which also handle PCM Pass-Through), DTS codes and C.O.S. codes have an "I" after the first three characters, such as ac3i_493263_05.LD, dtsi_493263_05.LD, and effi_4932xx_04.ld. This "I" stands for IBA or Internal Boot Assist. This is due to the fact that the code for Dolby Digital, and/or DTS (along with DTS Tables), and Crystal Original Surround Effects code are already located in the ROM of the CS49325 (AC-3 and COS only),
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1.4 Understanding the Differences between IBA Application Codes and Full Download Application Codes
The Internal Boot Assist (IBA) code simply loads the internal code from ROM into active memory, whereas the Full Download downloads directly to active memory. The typical size of an IBA code is around 300 bytes, where a Full Download code can be as large as 32 Kbytes. IBA codes are only for applications that accept IEC60958 or IEC61937 streams. The system designer should be aware that while both codes may be downloaded from the host microcontroller or autobooted from external memory, there are some fundamental feature set differences between the two types of application codes for AC-3, DTS and C.O.S. Please see below for all major feature differences between AC3_4932XX_VV.LD vs. AC3I_4932XX_VV.LD: AC3_4932XX_VV.LD has: 1) Fs-dependent digital deemphasis for PCM input. This means that the deemphasis coefficients are embedded inside the code (for 48 kHz, 44.1 kHz and 32 kHz), so that the user need only select the appropriate Fs, and when digital deemphasis is enabled, the response will be correct. For PCM streams with pre-emphasis and an Fs higher than 48 kHz, additional steps must be taken to enable deemphasis. Once the host determines that the incoming PCM stream has changed to 96 kHz, 99.2 kHz, or 64 kHz from the S/PDIF receiver and pre-emphasis was applied earlier on in the mastering process, the host must either reset the DSP (due to the Fs
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change on the DIR) and re-download the ac3_4932xx_vv.ld code with appropriate Fs digital deemphasis coefficient set OR perform an Application Restart, and then download the appropriate Fs digital deemphasis coefficient set.
Note: This extra message patch download is only necessary if the host wants to enable digital deemphasis.
or 4 type DTS streams. 3) Audio_Change_Config_Notification for the DTS stream-specific variable of pcmr. This means that if a stream changes from just a DTS 5.1 stream to one that is DTS-ES Discrete 6.1 or DTS-ES Matrix 6.1 encoded, the DSP will perform an unsolicited message. DTSI32XX.LD does not incorporate the features of DTS_32XX.LD. Please see below for all major feature differences between EFF_32XX.LD vs. EFFI32XX.LD: EFF_32XX.LD has: 1) Tone Control. 2) Fs-dependent digital deemphasis for PCM input. This means that the deemphasis coefficients are embedded inside the code (for 48 kHz, 44.1 kHz, and 32 kHz), so that the user need only select the appropriate Fs, and when digital deemphasis is enabled, the response will be correct. For PCM streams with pre-emphasis and an Fs higher than 48 kHz, additional steps must be taken to enable deemphasis. Once the host determines that the incoming PCM stream has changed to 96 kHz, 99.2 kHz, or 64 kHz from the S/PDIF receiver and pre-emphasis was applied earlier on in the mastering process, the host must either: (a) Reset the DSP (due to the Fs change on the DIR), and re-download the EFF_32XX.LD code with appropriate Fs digital deemphasis coefficient set. (b) Or, perform an Application Restart, and then download the appropriate Fs digital deemphasis coefficient set.
Note: This extra message patch download is only necessary if the host wants to enable digital deemphasis.
2) Tone Control. 3) Audio_Change_Config_Notification for the Dolby Digital stream-specific variable of timecod2 (also know as dsurmodex). This means that if a stream changes from just a Dolby Digital stream to one that is Dolby Digital Surround EX encoded, the DSP will perform an unsolicited message. AC3I_4932XX_VV.LD has: 1) There exists a message patch for 96 kHz, 88.2 kHz, 64 kHz, 48 kHz, 44.1 kHz, and 32 kHz Fs digital deemphasis coefficients. This means that once the host determines that the incoming PCM stream has changed to 96 kHz, 99.2 kHz, 64 kHz, 48 kHz, 44.1 kHz, or 32 kHz from the S/PDIF receiver and preemphasis was applied earlier on in the mastering process, the host must either: (a) Reset the DSP (due to the Fs change on the DIR), and re-download the AC3I32XX.LD code with appropriate Fs digital deemphasis coefficient set. (b) Or, perform an Application Restart, and then download the appropriate Fs digital deemphasis coefficient set.
Note: This extra message patch download is only necessary if the host wants to enable digital deemphasis.
Please see below for all major feature differences between DTS_32XX.LD vs. DTSI32XX.LD: DTS_32XX.LD has: 1) Tone Control. 2) Pro Logic decoding support for AMODE = 2, 3,
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EFFI32XX.LD has: 1) There exists a message patch for 96 kHz, 88.2 kHz, 64 kHz, 48 kHz, 44.1 kHz, and 32 kHz Fs digital deemphasis coefficients. This means that once the host determines that the
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incoming PCM stream has changed to 96 kHz, 99.2 kHz, 64 kHz, 48 kHz, 44.1 kHz or 32 kHz from the S/PDIF receiver and pre-emphasis was applied earlier on in the mastering process, the host must either: (a) Reset the DSP (due to the Fs change on the DIR), and re-download the EFFI32XX.LD code with appropriate Fs digital deemphasis coefficient set. (b) Or, perform an Application Restart, and then download the appropriate Fs digital deemphasis coefficient set.
Note: This extra message patch download is only necessary if the host wants to enable digital deemphasis.
1.5 Understanding the Feature Set of a Device vs. the Application Code
Please refer to CS4932X/CS49330 Part Matrix vs. Code Matrix PDF available from the CS49300 Page.
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1.6 Dolby Digital Description
Dolby Digital, also known as AC-3, is a digital audio compression algorithm developed by Dolby Laboratories for up to 5.1 channels of audio. This User's Guide covers the software that decodes Dolby Digital, and is designed to run on the CS49325, CS49326, CS49328, and CS49329 devices. This User's Guide covers all code that is named with AAAA == AC3_, AC3D, AC3X, AC3R and CCC == 325, 326 and 329 in addition to AAAA == AC3I and CCC == 325 and 326, as described in the previous section. Along with Dolby Digital, this document describes how to use the code for Dolby Pro Logic decode, tone control, pink/white noise generation, and bass management, as well as other features. It is assumed that the reader is familiar with the requirements and features of Dolby Digital as specified in: * Dolby AC-3 Specification: Digital Audio Compression Standard (AC-3), Advanced Television Systems Committee, Dec. 20, 1995 * Licensee Information Manual: Dolby Digital Multichannel Digital Audio Decoding for Consumer Products, Licensee Information Manual, Version 2.0, April, 1997 Special versions of the Dolby Digital decoder exist with different features, such as Virtual Dolby DigitalTM, VMAx VirtualTheaterTM, and SRS TruSurroundTM. Dolby Digital and/or Pro Logic decoding must be enabled in order for proper operation. For each such version of Dolby Digital decode, a separate appendix covers the incremental changes with respect to this application note. For commands specific to Virtual Dolby Digital software, please refer to AN163A. For commands specific to VMAx VirtualTheater software, please refer to AN163B. For commands specific to SRS TruSurround software, please refer to AN163F. Figure 1 displays a functional block diagram of the Dolby Digital application code when configured for PCM pass-through or Pink/White Noise Generation. It gives an idea of the interaction between the various application modules when PCM is enabled. Figure 2 displays a functional block diagram when the part is configured for decoding AC-3. AC-3 and PCM should be considered as separate applications when using this code.
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Dualzone
2 Discrete Channels
2 Channels (Lt & Rt) 5 Channels (C, Ls, Rs, LFE = 0) during PCM PassThrough
5 Channels (Ls=Rs)
Dolby Surround Encoded Input
Tone Control Parameters Bass Manager
2 Channel Output (Formatted as either Digital Audio or S/PDIF) Up to 5.1 Discrete Channels
Audio Manager
2 Channel Output (Formatted as either Digital Audio or S/PDIF) Up to 5.1 Discrete Channels
Pro Logic Decode Up to 5 Discrete Channels Downmix Tone Control (L, C, R)
PCM Enable
Autodetect Enable
Autodetect (PCM)
PCM PassThrough
Pink/White Noise Generation
CMPDAT/ DAI Pink Noise Control (Autodetect should be disabled when using pink noise)
5 Discrete Channels (C, Ls, Rs = 0 during PCM PassThrough
Output Mode Control
0.1 (Sub) (LFE = 0) During PCM PassThrough
LFE Enable
Bass Parameter Output Mode
Volume/Mute Channel Remap Delay
Figure 1. Dolby Digital Block Diagram, Configured for PCM Pass-Through/Pink Noise Generation
Dualzone
5 Discrete Channels ProLogic Enable & Stereo OR Dolby Surround Encoded Input
2 Channels (Lt & Rt)
5 Channels (Ls=Rs)
Tone Control Parameters Bass Manager
Pro Logic Decode AC-3 Enable Autodetect Enable 5.1 Discrete Channels Up to 5 Discrete Channels Downmix Tone Control (L, C, R)
Downmixed 2 Channel Output (Formatted as either Digital Audio or S/PDIF) Up to 5.1 Discrete Channels
Audio Manager
Downmixed 2 Channel Output (Formatted as either Digital Audio or S/PDIF) Up to 5.1 Discrete Channels
Autodetect (AC-3)
AC-3 Decode
CMPDAT/ DAI
5 Discrete Channels
Output Mode Control
0.1 (Sub)
LFE Enable
Bass Parameter Output Mode
Volume/Mute Channel Remap Delay
Figure 2. Dolby Digital Block Diagram, Configured for AC-3 Decoding
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1.7 DTS Digital Surround Description
DTS Digital Surround is a digital audio compression algorithm developed by Digital Theater Systems for up to 5.1 channels of audio. This User's Guide covers the software that decodes DTS and is designed to run on the CS49326, CS49328, and CS49329 devices. This user's guide covers all code that is named with AAAA == DTSI and DTS_, and CCC==326, 328 and 329, as described in Section 1.3 "Software Naming Convention" on page 15. Along with DTS, this document describes how to use the code for Dolby Pro Logic decode, tone control and bass management, as well as other features. It is assumed that the reader is familiar with the requirements and features of DTS as specified in: * DTS Specification: DTS Coherent Acoustics Decoder, DTS Technology, Version 1.0, January 20, 1998 Licensee Manual: DTS Master Quality Multichannel Digital Audio Decoding System for Consumer Products Licensee Manual, Version 1.0, July 1998
*
A special version of the DTS decoder exists with a virtualization feature called SRS TruSurroundTM. DTS and/or Pro Logic decoding must be enabled in order for proper operation. For commands specific to SRS TruSurround software, please refer to AN163F. Figure 3 displays a functional block diagram of the DTS application code and gives an idea of the interaction between the various application modules.
Dualzone
5 Discrete Channels Pro Logic Enable & AMODE == 3, 4 or 5
2 Channels (Lt & Rt)
5 Channels (Ls=Rs)
Tone Control Parameters Bass Manager
Pro Logic Decode DTS Enable Autodetect Enable 5.1 Discrete Channels Up to 5 Discrete Channels Downmix Tone Control (L, C, R)
Downmixed 2 Channel Output (Formatted as either Digital Audio or S/PDIF) Up to 5.1 Discrete Channels
Audio Manager
Downmixed 2 Channel Output (Formatted as either Digital Audio or S/PDIF) Up to 5.1 Discrete Channels
Autodetect (DTS)
DTS Decode
CMPDAT/ DAI
5 Discrete Channels
Output Mode Control
0.1 (Sub)
LFE Enable
Bass Parameter Output Mode
Volume/Mute Channel Remap Delay
Figure 3. DTS Block Diagram
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1.8 MP3 Description
MPEG-1, Audio Layer 3 (also known as MP3) is a digital audio compression algorithm developed by the Moving Pictures Experts Group for up to 2 discrete decoded channels of audio. This User's Guide covers the software that decodes MPEG-1, Audio Layer 3 and is designed to run on the CS49325, CS49326, CS49328, and CS49329 devices. This User's Guide covers all code that is named with AAAA == MP3_ and CCC==325, 326, 328 and 329, as described in the previous Section 1.3 "Software Naming Convention" on page 15. Along with MP3, this document describes how to use the code for bass management, as well as other features. It is assumed that the reader is familiar with the features of MP3 as specified in: ISO/IEC 11172-3, Information technology Coding of moving pictures and associated audio for digital storage media at up to about 1.5 Mbit/s - Part 3: Audio Figure 4 displays a functional block diagram of the MP3 application code and gives you an idea of the interaction between the various application modules.
Dualzone
MP3 Enable
Autodetect Enable 2 Discrete Channels Autodetect (MP3) Bass Manager
2.1 Channels Audio Manager
2 Decoded Channels (PCM) 2.1 Channels
MP3 Decoding
CMPDAT/ DAI
Bass Parameter Output Mode
Volume/Mute Channel Remap Delay
Figure 4. MP3 Block Diagram
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1.9 MPEG Multichannel Description
MPEG Multichannel is a digital audio compression algorithm developed by the Moving Pictures Experts Group for multiple discrete encoded channels of audio. This User's Guide covers the software that decodes MPEG-1, Audio Layer 2 as well as MPEG-2, Audio Layer 2 (with and without extension) Stereo and Multichannel streams with a decoded output of up to 5.1 discrete channels and is designed to run on the CS49325, CS49326, CS49328, and CS49329. This User's Guide covers all code that is named with AAAA == MPG_ and CCC==325, 326, 328 and 329, as described in Section 1.3. Along with MPEG Multichannel this document describes how to use the code for Dolby Pro Logic decode, tone control and bass management, as well as other features. It is assumed that the reader is familiar with the features of MPEG Multichannel as specified in: ISO/IEC 13818-3, MPEG Audio Coding Standard A special version of the MPEG Multichannel decoder exists with a virtualization feature, called SRS TruSurroundTM. MPEG Multichannel and/or Pro Logic decoding must be enabled in order for proper operation. For commands specific to SRS TruSurround software, please refer to AN163F. Figure 5 displays a functional block diagram of the MPEG Multichannel application code and gives an idea of the interaction between the various application modules.
Dualzone
5 Discrete Channels
2 Channels (Lt & Rt)
5 Channels (Ls=Rs)
Pro Logic Enable & Audio_Config == 9
Tone Control Parameters Bass Manager
Pro Logic Decode MPEG-MC Enable Autodetect Enable 5.1 Discrete Channels Up to 5 Discrete Channels Downmix Tone Control (L, C, R)
Downmixed 2 Channel Output (Formatted as either Digital Audio or S/PDIF) Up to 5.1 Discrete Channels
Audio Manager
Downmixed 2 Channel Output (Formatted as either Digital Audio or S/PDIF) Up to 5.1 Discrete Channels
Autodetect (MPEG-MC)
MPEG-MC Decode
CMPDAT/ DAI
5 Discrete Channels
Output Mode Control
0.1 (Sub)
LFE Enable
Bass Parameter Output Mode
Volume/Mute Channel Remap Delay
Figure 5. MPEG Multichannel Block Diagram
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1.10 AAC Description
AAC (Advanced Audio Coding) is an "MPEG-2standard-based" digital audio compression algorithm (offering up 5.1 discrete decoded channels for this implementation) collaboratively developed by AT&T, the Fraunhofer Institute, Dolby Laboratories, and the Sony Corporation. This User's Guide covers the software that decodes and is designed to run on the CS49329. This User's Guide covers all code that is named with AAAA == AAC_ and CCC==329 as described in the previous section. Along with AAC this document describes how to use the code for Dolby Pro Logic decode and bass management, as well as other features. It is assumed that the reader is familiar with the features of AAC as specified in: IS 13818-7, MPEG-2 Advanced Audio Coding (AAC) Specification, April, 1997 Figure 6 displays a functional block diagram of the AAC application code and gives you an idea of the interaction between the various application modules.
Dualzone
5 Discrete Channels
2 Channels (Lt & Rt)
5 Channels (Ls=Rs)
Pro Logic Enable & Audio_Config == 2
Tone Control Parameters Bass Manager
Pro Logic Decode AAC Enable Autodetect Enable 5.1 Discrete Channels Up to 5 Discrete Channels Downmix Tone Control (L, C, R)
Downmixed 2 Channel Output (Formatted as either Digital Audio or S/PDIF) Up to 5.1 Discrete Channels
Audio Manager
Downmixed 2 Channel Output (Formatted as either Digital Audio or S/PDIF) Up to 5.1 Discrete Channels
Autodetect (AAC)
AAC Decode
CMPDAT/ DAI
5 Discrete Channels
Output Mode Control
0.1 (Sub)
LFE Enable
Bass Parameter Output Mode
Volume/Mute Channel Remap
Figure 6. AAC Block Diagram
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1.11 Crystal Original Surround Description
Crystal Original Surround is an application code designed to take process PCM data and convert a two-channel source into 5.1 channels with effects processing. This User's Guide covers the software that processes PCM using the C.O.S. Application Code designed to run on the CS49325, CS49326, CS49328, CS49329, and CS49330. This User's Guide covers all code that is named with AAAA == EFF_, and CCC==325, 326, 328, 329, and 330, as described in the previous section. It is assumed that the reader is familiar with effects processing, however a brief description of reverb, echo, and other filtering techniques are covered. Along with Crystal Original Surround, this document describes how to use the code for tone control and bass management, as well as other features. Figure 7 displays a functional block diagram of the Crystal Original Surround application code and gives you an idea of the interaction between the various application modules. Other Stereo PCM processing application codes are available and are discussed in separate appendices to AN163. They are: SRS Circle Surround (AN163C), Pacific Microsonics HDCD (AN163D) and Logic 7 (AN163E).
Dualzone 2 Channels (Ls, Rs) Autodetect Enable Test Impulse Enable EQ Parameters 5 Discrete Channels (Reverb Out)
FIR Enable 5.1 Channels Audio Manager
Crystal Original Surround Effects Enable
2 Channels (Pass-Through)
5.1 Channels Autodetect (PCM) Test Impulse (L, R) 3-Band Parametric Equalizer Stereo PCM to Multichannel PCM Processing Divider FIR Channel Mixer Tone Control (L, C, R) Bass Manager
CMPDAT/ DAI 2 Discrete Channels
1 Channel (FIR Out) 2 Discrete Channels 2 Discrete Channels
3.1 Channels (L, C, R, LFE)
Tone Control Parameters
Bass Parameter Output Mode
Volume/Mute Channel Remap Delay
Figure 7. Crystal Original Surround Block Diagram
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2. HARDWARE CONFIGURATION
After download or soft reset, and before kickstarting the application (please see Section 4.1 "Audio Manager for Dolby Digital, DTS, MP3, MPEG Multichannel, AAC and Crystal Original Surround" on page 33 for more information on kickstarting), the host has the option of changing the default hardware configuration. Address checking for serial communication and data type and format for digital data input and output can be changed through hardware configuration settings. The entire hardware configuration can only be changed immediately after download or soft reset. The input and output port configurations, however, can also be changed following a special run-time restart message (please see Section 4.1 "Audio Manager for Dolby Digital, DTS, MP3, MPEG Multichannel, AAC and Crystal Original Surround" on page 33 for more information about the Application Restart message).
2.1 Supported Input/Output Modes
The CS4932X has two input ports and one output port. The CS49300 Family Datasheet describes the digital audio formats supported by the ports and gives a description of the ports themselves. The capabilities of each port are presented in the CS49300 Family Datasheet. Table 1 and Table 2 specify the exact input/output modes supported by each application. Please refer to the CS49300 Family Datasheet for more information on these hardware configurations. The hardware modes are supported when the AC-3, DTS, MPEG Multichannel, or AAC code is active: The hardware modes are supported by the PCM pass-through portion of the AC-3 application or when Crystal Original Surround Audio Effects Processing Code is loaded.
Hardware Configuration INPUTA INPUTB INPUTC OUTPUTA OUTPUTB OUTPUTC OUTPUTD OUTPUTE
Description Input Data Type Input Data Format Input SCLK/Data Edge Output Master/Slave Clock Setup Output Data Format Output MCLK Rate Output SCLK Rate Output SCLK/Data Edge
Supported Modes 0,1 0,1 0,1 0,1 0,1,2,3 0,1,2,3 0,1,2 0,1
Hardware Configuration INPUTA INPUTB INPUTC OUTPUTA OUTPUTB OUTPUTC OUTPUTD OUTPUTE
Description Input Data Type Input Data Format Input SCLK/Data Edge Output Master/Slave Clock Setup Output Data Format Output MCLK Rate Output SCLK Rate Output SCLK/Data Edge
Supported Modes 0,1,2 0,1 0,1 0,1 0,1,2,3 0,1,2,3 0,1,2 0,1
Table 1. Hardware Configurations Supported By: AC-3, DTS, MP3, MPEG Multichannel, and AAC
Table 2. Hardware Configurations Supported By: PCM Pass-Through (AC-3 Code) and Crystal Original Surround
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2.1.1 DCLK = 86MHz
The switching characteristics of the Motorola and Intel parallel communication modes, discussed in the CS49300 Data Sheet, have notes that state "It should be noted that DCLK for the internal clock mode is application specific. The application code user's guide should be checked to confirm DCLK for the particular application." For all the application codes discussed in this application code user's guide, DCLK is equal to 86 MHz. 0x800271 0xF80FFF 0x800270 0xF01301 The following commands must be sent to configure the data present on the XMT958/AUDAT3 pin to Left Justified 24-bit format that is driven from the supplied MCLK (256Fs/512Fs/128Fs/384Fs) set by the OUTPUT C Parameter in the CS49300 Data Sheet: 0x800271 0xF80FFF 0x800270 0xF01700 The following commands must be sent to configure the data present on the XMT958/AUDAT3 pin to Left Justified 20-bit format that is driven from the supplied MCLK (256Fs/512Fs/128Fs/384Fs) set by the OUTPUT C Parameter in the CS49300 Data Sheet: 0x800271 0xF80FFF 0x800270 0xF01300 One of the configurations listed above must be sent before kickstart.
2.1.2 Configuring the XMT958/AUDAT3 Data Format
The following commands must be sent to configure the data present on the XMT958/AUDAT3 pin to S/PDIF (IEC60958) format that is driven from the supplied MCLK at 256 Fs: 0x800271 0xf7ffff 0x800171 0x021000 The following commands must be sent to configure the data present on the XMT958/AUDAT3 pin to S/PDIF (IEC60958) format that is driven from the supplied MCLK at 512 Fs: 0x800271 0xf7ffff 0x800171 0x061000 The following commands must be sent to configure the data present on the XMT958/AUDAT3 pin to I2S 24-bit format that is driven from the supplied MCLK (256Fs/512Fs/128Fs/384Fs) set by the OUTPUT C Parameter in the CS49300 Data Sheet: 0x800271 0xF80FFF 0x800270 0xF01701 The following commands must be sent to configure the data present on the XMT958/AUDAT3 pin to I2S 20-bit format that is driven from the supplied MCLK (256Fs/512Fs/128Fs/384Fs) set by the OUTPUT C Parameter in the CS49300 Data Sheet:
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2.1.3 S/PDIF (AES/EBU) Channel Status Block Control
The CS4932X, in combination with all of the codes mentioned in this document, offers the "minimum" professional implementation of control over the IEC60958 channel status bits. According to the IEC60958 Specification, the host has control over the setting of the "PRO" bit in the channel status word (Channel Status Byte 0, Bit 0). "Professional use of the channel status block" will be implemented if the host sets the PRO bit to a state of logic 1. These additional channel status bits may also be set by the host: Channel Status Byte 0, Bits 1-5; Channel Status Byte 1, Bits 0-7; and Channel Status Byte 3, Bits 0 and 1. Please refer to Table 3 for the IEC60958 definition for the setting of each bit or set of bits.
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"Consumer use of the channel status block" will be implemented if the host sets the PRO bit to logic state 0. These additional channel status bits may also be set by the host: Channel Status Byte 0, Bits 1-5; Channel Status Byte 1, Bits 0-7; and Channel Status Byte 3, Bits 0 and 1. Please refer to Table 4 for the IEC6096 definition for the setting of each bit or set of bits. The channel status bits map to the DSP Write Data Word in the following manner (please refer to Table 5): * Channel Status Byte 0, Bit 0 = maps to bit 8 in the DSP Write Data Word. * * * Channel Status Byte 0, Bits 1:5 = map to bits 9:13 in the DSP Write Data Word. Channel Status Byte 1, Bits 0:7 = map to bits 14:21 in the DSP Write Data Word. Channel Status Byte 3, Bits 0 and 1 = map to bits 22 and 23 in the DSP Write Data Word. For example the following command sets the Channel Status Block of the Right Subframe (Channel B) for: Consumer Mode (Byte 0, Bit 0 = 0), 32kHz Fs (Byte 3, Bits 0 and 1 = 11); Category Code = General (Byte 1, Bits 0:7 = 000000000); 2 Audio Channels without Pre-Emphasis and Copy Prohibited (Byte 0, Bits 1:5 = 00000): 0x800273 0x0000FF 0x800173 0xC00000 The first word performs a logical "AND" of the register with the second word, which acts as a mask. The values located in the bit locations "AND'ed" with a one are preserved, while the locations "AND'ed" with a zero are cleared. The third word performs a logical "OR" of the same register with the fourth word, which contains the necessary CSW bits that the user wants to set.
Please refer to the IEC60958 Specification for more a more detailed explanation of the channel status block definitions. For example, the following command sets the Channel Status Block of the Left Subframe (Channel A) for: Consumer Mode (Byte 0, Bit0 = 0); 48kHz Fs (Byte 3, Bit 0 and 1 = 01); Category Code = General (Byte 1, Bits 0:7 = 000000000); 2 Audio Channels without Pre-Emphasis and Copy Prohibited (Byte 0, Bits 1:5 = 00000): 0x800272 0x0000FF 0x800172 0x800000
2.1.4 S/PDIF (AES/EBU) Validity Bit Control
In addition to the Channel Status Block Control, the Validity bit in the IEC60958 stream may also be set by the host to indicate if the according audio sample is fit for conversion to analog. The following command must be sent to CLEAR the IEC60958 Validity bit: 0x800271 0xFFBFFF The following command must be sent to SET the IEC60958 Validity bit: 0x800171 0x004000
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B
Bit
0 0 1 Bit 1 0 1 Bits 2 3 4 000
100 110 111 XXX 5 0 1 bits 6 7 00 01 10 11
Bit
BYTE 0 PRO = 1 Consumer use of channel status block Professional use of channel status block Audio Normal Audio Non-Audio Encoded audio signal emphasis Emphasis not indicated. Receiver defaults to no emphasis with manual override enabled None. Receiver manual override disabled 50/15 S. Receiver manual override disabled CCITT J.17. Receiver manual override disabled All other states of bits 2-4 are reserved Lock: Source Sample Frequency Locked - default Unlocked Fs: Sample Frequency Not indicated. Receiver default to 48 kHz and manual override or auto set enabled 48 kHz. Manual override or auto disabled 44.1 kHz. Manual override or auto disabled 32 kHz. Manual override or auto disabled
BYTE 1 Bits 0 1 2 3 Channel Mode 0 0 0 0 Mode not indicated. Receiver default to 2-channel mode. Manual override enabled 0 0 0 1 Two-channels. Manual override disabled 0 0 1 0 Single channel. Manual override disabled 0 0 1 1 Primary/Secondary (Ch. A is primary) Manual override disabled 0 1 0 0 Stereophonic. (Ch. A is left) Manual override disabled. 0 1 0 1 Reserved for user-defined application 0 1 1 0 Reserved for user-defined application 1 1 1 1 Vector to byte 3. Reserved X X X X All other states of Bits 0-3 are reserved. Bits 4 5 6 7 User bits management 0 0 0 0 Default, no user information indicated 0 0 0 1 192-bit block structure Preamble "Z" starts block 0 0 1 0 Reserved 0 0 1 1 User-defined application X X X X All other states of Bits 4-7 are reserved. BYTE 3 Vectored target byte Reserved
Bits 0-7 XXXXXX
Table 3. Professional Channel Status bytes 0, 1, and 3
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3. APPLICATION MESSAGING
While using the CS4932X it may be necessary to control or monitor the application to take full advantage of the rich feature set employed by the CS4932X and its software. By controlling application messaging, the user controls the application. Whether it is configuring the part after download (e.g. enabling MPEG decode), or changing run-time parameters (e.g. adjusting channel delays), the host uses application messaging to communicate with the CS4932X. While communicating with the CS4932X using indexed modules, a strict software protocol must be used in conjunction with the hardware protocol discussed in the CS49300 Family Data Sheet. This section will cover both the format of the messages and the different configuration modules available for use with the CS4932X.
Caution: The host must strictly adhere to the hardware and software protocols described in the software and hardware documentation for the CS4932X part to ensure successful communication.
3.1 Indexed Module Communication Protocol
Each indexed module of the application can be thought of as a block of software registers or variables. The index identifies a unique variable within the module. When the opcode for a module and an index are combined, a unique variable can be read or written. This section covers how to communicate with the CS4932X using indexed modules. The software protocol is presented for the following types of messages: Write, Solicited Read, Read Response, and Unsolicited Read Response. When the protocol presented in this section is used with the application modules in Section 4 "Common Application Modules" on page 32, the host will be able to fully configure the application running on the CS4932X.
3.1.1 Write Session
A write session with the CS4932X consists of one 6-byte message from the host to the CS4932X. The write message consists of a Command word followed by an associated data word. Table 5 shows the format of a Write Message:
Write Command Word:
23 22 21 20 19 18 17 16 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 OPCODE[7:0] INDEX[15:0]
Write Data Word:
23 22 21 20 19 18 17 16 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 DATA[23:0]
OPCODE[7:0] - 8-bit (1-byte) field containing opcode for targeted application module. This field will choose which module is written. INDEX[15:0] - 16-bit (2-byte) field containing the index for the desired variable in the module chosen by OPCODE. This field will choose the actual variable to be altered. DATA[23:0] - 24-bit (3-byte) data word to be written into the variable specified by INDEX in the module specified by OPCODE. Table 5. Write Message Format
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3.1.2 Solicited Read Message Format
A solicited read session consists of one 3-byte Read Request message from the host to the CS4932X, followed by a 6-byte Read Response message from the CS4932X to the host. The Read Request message simply consists of a Read Command word whose format is shown in Table 6.
Read Command Word: 23 22 21 20 19 18 17 16 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 OPCODE[7:0] INDEX[15:0]
After the host sends the Read Command word it should wait for the INTREQ line to fall. See the CS49300 Family Data Sheet for more on hardware communication with the CS4932X. After INTREQ falls, the host should read out the 6byte Read Response message which consists of a 3byte Read Response Command word followed by the requested data word. The format of the Read Response message is shown in Table 7.
OPCODE[7:0] - 8-bit (1-byte) field containing opcode for targeted application module. This field will choose the module to be read. INDEX[15:0] - 16-bit (2-byte) field containing the index for the desired variable in the module chosen by OPCODE. This field will choose the actual variable to be read. Table 6. Read Command Message Format
Read Response Command Word: 23 22 21 20 19 18 17 16 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 OPCODE[7:0] Read Response Data Word 23 22 21 20 19 18 17 16 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 DATA[23:0] INDEX[15:0]
OPCODE[7:0] - 8-bit (1-byte) field containing the Read Response opcode for the targeted application module. This field will show which module was read. INDEX[15:0] - 16-bit (2-byte) field containing the index for the desired variable in the module chosen by OPCODE. This field will show the actual variable that was read. DATA[23:0] - 24-bit (3-byte) data word that was read from the variable specified by INDEX in the module specified by OPCODE. Table 7. Read Response Message Format
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3.1.3 Unsolicited Read Message Format
Unsolicited messages will typically be used in systems where INTREQ can generate interrupts. These messages will come from the CS4932X to indicate a change in the system that must be addressed. One example is when the part is in autodetect mode and detects a new stream. An Unsolicited Read message will be sent by the CS4932X to indicate the new stream type. The 6-byte unsolicited read messages from the CS4932X consist of a 3-byte Read Command word
Unsolicited Read Command Word: 23 22 21 20 19 18 17 16 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 OPCODE[7:0] Unsolicited Read Data Word: 23 22 21 20 19 18 17 16 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0 DATA[23:0] INDEX[15:0]
which defines the type of unsolicited message (as described in Section 4.3 "Unsolicited Messages (Read-Only)" on page 54and a 3-byte associated data word that contains more information describing a system condition. Every time the existence of a message is detected (by sensing that INTREQ has gone LOW), the host should read out the 6-byte Unsolicited Read message. Table 8 shows the format of an Unsolicited Read Message:
OPCODE[7:0] - 8-bit (1-byte) field containing opcode for the Unsolicited Read message INDEX[15:0] - 16-bit (2-byte) field containing the index for the variable in the module designated by OPCODE. DATA[23:0] - 24-bit (3-byte) data word that corresponds to the variable specified by INDEX in the module specified by OPCODE. Table 8. Unsolicited Read Message Format
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4. COMMON APPLICATION MODULES
Both the Generalized Audio Manager and Generalized Bass Manager are generic in their use and apply to Dolby Digital, DTS, MP3, MPEG Multichannel, AAC, and Crystal Original Surround code, in addition to the codes covered in AN163 appendices. The Generalized Pro Logic Manager is applicable to all codes with the exception of MP3 and C.O.S. codes in addition to all IBA-Based codes. The Generalized Tone Control Manager is applicable to all codes with the exception of MP3 and AAC. The block diagram on the cover page of this document portrays the interaction between the various application modules and the AC-3, DTS, MP3, MPEG Multichannel, AAC and Crystal Original Surround code. The following should be noted about all values in the application modules: * Variables marked by "Default*" will only be initialized after download. These variables will retain their values after a soft reset (or Application Restart, where applicable). * Variables marked by `Default' will be reinitialized to the values shown in this application note after download, soft reset (or Application Restart, where applicable). Variables marked by `' can be modified during runtime of any application code. Variables NOT marked by `' can NOT be modified during runtime of any application code. Only those values which are presented as valid or which fall within the specified range should be written to the application module variables. If a value which falls outside the stated range is written to a variable, functionality of the application is not guaranteed and erroneous output could result. All bits that are not defined should be considered reserved and written with 0's unless
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specified otherwise. * For variables such as volume and scaling factors, the real number range of 0.0-1.0 is written as 0x000000-0x7FFFFF. This range is linear, i.e. 1/4 volume, or -12 dB, is represented as 0x1FFFFF and is equivalent to a value of 0.25. The formula for converting variable settings from a hexadecimal number into dB is as follows:
20 * log (variable setting/max value for variable) e.g. 20 * log (0x1FFFFF/0x7FFFFF) == -12 dB
*
Numbers preceded by 0x should be interpreted as hex, and numbers followed by `b' should be interpreted as binary. All values sent to the CS4932X should be converted to hexadecimal. Likewise all values read from the part are in hexadecimal.
*
*
*
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4.1 Audio Manager for Dolby Digital, DTS, MP3, MPEG Multichannel, AAC and Crystal Original Surround
Write Opcode = 0x88; Read Opcode = 0x09; Read Response Opcode = 0x89 Write Message = 0x8800HH 0xhhhhhh Read Request Message = 0x0900HH; Read Response Message = 0x8900HH 0xhhhhhh 0xHH = index, 0xhhhhhh = data value
Index
0x00
Variable
AUDIO_MGR_CONTROL
Dataword Content
Bit 15 = Audio_Config_Change_Notification_Enable = 0/1= Disable/Enable unsolicited message notification of any change in audio configuration of any input bitstream. Note: This bit is applicable only for the following algorithms: AAC, AC-3, MPEG and DTS. The exact fields are specified for each application separately. Please refer to Section 4.1.1 "Audio_Control: (Index 0x00)" on page 37 for more detail. Bit 14 = Startup_Autodetect_Bypass_Enable= 0/1= Disable/Enable bypassing of autodetection at start-up, i.e. for the first time after download/reset. (Valid only with Autodetect_Enable==1). Note: If the above mode is enabled, during readback this bit is reserved and may read 0 or 1. Setting of this bit is not recommended with DTS code. Bit 13 = Null_Pause_Report_Enable = 0/1= Disable/Enable reporting of Null and Pause IEC61937 data types (Valid only with Autodetect_Enable==1). Note: Care should be taken when enabling this bit as it could cause multiple interrupts to be taken by the host. Using this bit is not suggested unless IEC61937 streams and null/pause reporting are understood in the context of how it will affect the host microcontroller and interrupts. Use of null/pause reporting is not required for a working system. Bit 12 = Autodetect_Enable = 0/1= Disable/Enable autodetect functionality. Bit 9 = Delay_Granularity 0/1 = ms/16 sample. Note: Not applicable to AAC code. Bit 8 = Application_Restart_Enable = 0/1= Disable/Enable restart of software application. Note: This setting is ONLY applicable for AC-3 and C.O.S. application codes. This bit should be set to 0 for all other application codes. Bit 4 = PLL_Enable = 0/1 = Disable/Enable Phase Locked Loop generation of DSP clock. It is mandatory that the PLL be enabled for the CS4932X. Bit 0 = Kickstart_Enable = 0/1 = Disable/Enable Kickstart of application. Default = 0x000000 Table 9. Audio Manager
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Index
0x01
Variable
NOISE_CONTROL
Dataword Content
Bit 6 = High_Pass_Filtered_White_Noise_Enable = 0/1 = Disable/Enable. Note: Bit 6 and Bit 4 are mutually exclusive. Set only one HIGH. Bit 5 = Level_Select = 0/1 = Dolby Level / High Level White Noise output. Dolby Level must be used for Dolby Certification. Refer to Section 4.1.2. Note: Bit 5 should only be set HIGH if Bit 4 or Bit 6 has been set HIGH. Bit 4 = White_Noise_Enable = 0/1 = Pink/White Noise output. Note: Bit 6 and Bit 4 are mutually exclusive. Set only one HIGH. Bits 3:0 = Output _Channel = 0...5 = L, C, R, Ls, Rs, LFE channel output. Disabled if > 5. Note: Only applicable when AC-3 application code is loaded and configured for PCM pass-through. Autodetection must be disabled.
0x02
C_DELAY
Default = 0x7FFFFF 0...15 ms Delay if Delay_Granularity = 0 0...45 * 16 sample Delay if Delay_Granularity = 1 Note: Not applicable to AAC Code. Default* = 0 1...24 = Precision of output PCM. Default* = 24 0...40 = mS delay (automatic +15ms for Pro Logic) if Delay_Granularity = 0 0...120 * 16 sample Delay (automatic +15ms for Pro Logic) if Delay_Granularity = 1 Note: AAC Code only supports 15ms automatic + user defined (up to 15ms) delay for left/right surround channels when Prologic is switched on. However, when Prologic is switched off, the user settings for left/right surround delays are ignored (i.e. AAC Code does not support delays for any channels during normal operation).
0x03 0x04
PCM_PRECISION LS_DELAY
0x05
RS_DELAY
Default* = 0 0...40 = mS delay (automatic +15ms for Pro Logic) if Delay_Granularity = 0 0...120 * 16 sample Delay (automatic +15ms for Pro Logic) if Delay_Granularity = 1 Note: AAC Code only supports 15ms automatic + user defined (up to 15ms) delay for left/right surround channels when Prologic is switched on. However, when Prologic is switched off, the user settings for left/right surround delays are ignored (i.e. AAC Code does not support delays for any channels during normal operation).
0x06 0x07 0x08 0x09 0x0a 0x0b 0x0c
MASTER_VOLUME L_VOLUME C_VOLUME R_VOLUME LS_VOLUME RS_VOLUME LFE_VOLUME
Default* = 0 0x800000-0x7FFFFF (-1.0 to 1.0). Default* = 0x7FFFFF 0x800000-0x7FFFFF (-1.0 to 1.0). Default* = 0x7FFFFF 0x800000-0x7FFFFF (-1.0 to 1.0). Default* = 0x7FFFFF 0x800000-0x7FFFFF (-1.0 to 1.0). Default* = 0x7FFFFF 0x800000-0x7FFFFF (-1.0 to 1.0). Default* = 0x7FFFFF 0x800000-0x7FFFFF (-1.0 to 1.0). Default* = 0x7FFFFF 0x800000-0x7FFFFF (-1.0 to 1.0). Default* = 0x7FFFFF Table 9. Audio Manager (Continued)
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Index
0x0d 0x0e 0x0f 0x10 0x11 0x12 0x13 0x14 MUTE DAO0_CHANNEL DAO1_CHANNEL DAO2_CHANNEL DAO3_CHANNEL DAO4_CHANNEL DAO5_CHANNEL
Variable
Dataword Content
0/1 = Unmute/mute audio. This is a soft mute. Default = 0 0...7 = Channel typea for Digital Audio Output 0. Any channel may be mapped to any valid number of outputs. Default* = 0(L) (note a) 0...7 = Channel typea for Digital Audio Output 1. Any channel may be mapped to any valid number of outputs. Default* = 2(R) (note a) 0...7 = Channel typea for Digital Audio Output 2. Any channel may be mapped to any valid number of outputs. Default* = 3(Ls) (note a) 0...7 = Channel typea for Digital Audio Output 3. Any channel may be mapped to any valid number of outputs. Default* = 4(Rs) (note a) 0...7 = Channel typea for Digital Audio Output 4. Any channel may be mapped to any valid number of outputs. Default* = 1(C) (note a)
0...7 = Channel typea for Digital Audio Output 5. Any channel may be mapped to any valid number of outputs. Default* = 5(LFE) (note a) SAMPLING_FREQUENCY_ 0..2 = Sampling Frequency code to be set by host in case of PCM-only input CODE applications. Irrelevant for non-PCM processing. Note: This setting affects both the Pro Logic and Bass Manager filter coefficients and Surround Channel Delays, as well as the deemphasis filter applied when digital deemphasis is enabled. Please refer to Section 5.2.1 "PCM_CONTROL" on page 80 for more on using digital deemphasis with IBA codes. 0 = 48 KHz (note b) 1 = 44.1 KHz (note c) 2 = 32 KHz (note d) Default* = 0 0..8388606 = Number of samples of silence after which CS4932X will declare Out-Of-PCM while playing PCM with autodetection enabled. 8388607 (0x7FFFFF) = Disable silence detection when playing PCM. CS4932X will never declare Out-Of-PCM once it starts playing PCM (equivalent to infinite threshold). Note: Valid only for AC-3 code when configured for PCM pass-through, in addition to all other PCM processing application codes. It is recommended that system designers set this value large enough to avoid inter-track silence from PCM Compact Discs.
0x15
PCM_AUTODETECT_ SILENCE_THRESHOLD
0x16
LAST_UNSOLICITED_ MESSAGE PLL_REGISTER_1
Default* = 48000 Last Unsolicited Message Value (see Section 4.3) READ ONLY Default=0x000000 Register 1 value for Phase Locked Loop (no Fs generation). The following values should be used for each corresponding CLKIN frequency: 0x05F1BF (CLKIN = 12.288MHz, DCLK = 86MHz, Output A1) 0x0BF1BF (CLKIN = 24.576MHz, DCLK = 86MHz, Output A1) 0x0d214f (CLKIN = 27MHz, DCLK = 86MHz, Output A1) Default* = 0x05F1BF Table 9. Audio Manager (Continued)
0x17
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Index
0x18
Variable
PLL_REGISTER_2
Dataword Content
Register 2 value for Phase Locked Loop (no Fs generation). The following values should be used for each corresponding CLKIN frequency: 0x002F7D (CLKIN = 12.288MHz, DCLK = 86MHz, Output A1) 0x002F7D (CLKIN = 24.576MHz, DCLK = 86MHz, Output A1) 0x00237D (CLKIN = 27MHz, DCLK = 86MHz, Output A1)
0x19
L_DELAY
Default* = 0x002F7D 0...15 ms Delay if Delay_Granularity = 0 0...45 * 16 sample Delay if Delay_Granularity = 1 Note: Not applicable to AAC Code. Default* = 0 0...15 ms Delay if Delay_Granularity = 0 0...45 * 16 sample Delay if Delay_Granularity = 1 Note: Not applicable to AAC Code. Default* = 0 0...10 ms Delay if Delay_Granularity = 0 0...30 * 16 sample Delay if Delay_Granularity = 1 Note: Not applicable to AAC Code. Default* = 0 0x800000-0x7FFFFF (-1.0 to 1.0). Default* = 0x7FFFFF 0x800000-0x7FFFFF (-1.0 to 1.0). Default* = 0x7FFFFF 0...7 = Channel typea for Digital Audio Output 6. Any channel may be mapped to any valid number of outputs. Default* = 6(Lz) (note a) 0...7 = Channel typea for Digital Audio Output 7. Any channel may be mapped to any valid number of outputs. Default* = 7(Rz) (note a) 0x800000-0x7FFFFF (-1.0 to 1.0). Default* = 0x7FFFFF Table 9. Audio Manager (Continued)
0x1a
R_DELAY
0x1b
LFE_DELAY
0x1c 0x1d 0x1e 0x1f 0x20
LZ_VOLUME RZ_VOLUME DAO6_CHANNEL DAO7_CHANNEL DUALZONE_ ATTENUATION
Notes: a. Channel type 0...7 corresponds to Channels L, C, R, Ls, Rs, LFE, Lz, and Rz respectively. Lz and Rz refer dual-zone outputs. If the source made available to the decoder is a 5.1 Channel AC-3 stream, these outputs will be Pro Logic encoded and should be considered to be Lt and Rt. However, for a 2.0 Channel AC-3 stream, these outputs will be simply L and R. b. Pre-loaded coefficients will provide a 80 Hz Crossover if input Fs is 48 kHz. A different set of coefficients must be downloaded for 96 kHz or 192 kHz Fs support. c. Pre-loaded coefficients will provide a 80 Hz Crossover if input Fs is 44.1 kHz. A different set of coefficients must be downloaded for 88.2 kHz or 176.4 kHz Fs support. d. Pre-loaded coefficients will provide a 80 Hz Crossover if input Fs is 32 kHz. A different set of coefficients must be downloaded for 64 kHz or 128 kHz Fs support. See definition on page 32. "Default" vs. "Default*" See definition page 32.
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4.1.1 Audio_Control: (Index 0x00)
Bit_15: Audio_Config_Change_Notification_Enable Setting this bit HIGH will cause the CS4932X to generate interrupts when a change to a specific stream definition variable occurs. Stream definition variables are embedded in the compressed bit stream.
Caution: The host must wait at least 20 ms before attempting to read any of the stream definition variables after receiving an Unsolicited Message indicating that there has been a change to one of the variables. Variable values read from the DSP before this time has passed may not be valid.
For the AC-3 code if any of acmod, bsid, lfeon, dsurmod, dialnorm or timecod2 variables change in the incoming stream, CS4932X will issue an unsolicited message (see Section 4.3 "Unsolicited Messages (Read-Only)" on page 54). For the DTS code if any of amode, lff, or pcmr variables change in the incoming stream, CS4932X will issue an unsolicited message (see Section 4.3 "Unsolicited Messages (Read-Only)" on page 54). For the MP3 code if the mode variable changes in the incoming stream, CS4932X will issue an unsolicited message (see Section 4.3 "Unsolicited Messages (Read-Only)" on page 54). For the MPEG code if any of mode, center, surround, or lfe variables change in the incoming stream, CS4932X will issue an unsolicited message (see Section 4.3 "Unsolicited Messages (ReadOnly)" on page 54). For the AAC code if the channel_configuration variable changes in the incoming stream, CS4932X will issue an unsolicited message (see Section 4.3 "Unsolicited Messages (Read-Only)" on page 54). 0 - Disables Audio_Config_Change_Notification 1 - Enables Audio_Config_Change_Notification
Bit 14: Startup_Autodetect_Bypass_Enable Maintains the CS4932X in autodetect mode (if Bit 12 if set), but bypasses autodetection (directly enters input data processing) at start-up, i.e., for the first time after download/reset. This bit should only be changed after download, soft reset, (or Application Restart, where applicable). The state of autodetect should not be modified during run-time. 0 - Disables Startup_Autodetect_Bypass 1 - Enables Startup_Autodetect_Bypass No unsolicited message is generated by CS4932X if the processing is successful. However, if the data format is subsequently found to be the wrong type, autodetect messaging will be generated by CS4932X as usual. The Startup_Autodetect_Bypass mode above is useful in saving the autodetect latency when the controller already knows that the current input data format is decodable by the downloaded application. The controller is usually aware of this due to the autodetection message from the previous (most recent) downloaded application.
Notes: 1. If this mode is enabled, Bit 14 readback value is reserved (either 0 or 1 may be received and should be ignored). 2. The autodetection module in the MPEG code and AAC code will assume a particular format (Pc = 5 in MPEG and Pc = 28 in AAC) and will start decoding. It is important to note that autodetection will be ALWAYS bypassed, however, an unsolicited message will be sent TWICE if the incoming data format does not have the assumed Pc. More specifically, if MPEG material has been detected while decoding a different format stream, the host will download the MPEG code with Startup_Autodetect_Bypass_Enable set HIGH, and will receive the Pc = 5 message. If the MPEG stream happens to have Pc = 6, then that will be reported to the host as well. Similarly, in the AAC case, the default Pc is chosen to be 28, but the Pc = 7 will be detected and reported as well.
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Bit 12: Autodetect_Enable Places the CS4932X into autodetect mode (see Section 4.3 "Unsolicited Messages (Read-Only)" on page 54). This bit should only be changed after download, soft reset, (or Application Restart, where applicable). The state of autodetect should not be modified during run-time. 0 - Disables Autodetect 1 - Enables Autodetect Bit 9: Delay Granularity Delay Granularity should be setup at kickstart (not after). This setting is used to choose the delay granularity as mS or in 16-sample units. 0 - The delay indices are interpreted as mS 1 - The delay indices are interpreted as the number of 16-sample-units to delay. In other words, the individual channel delay setting is multiplied by 16 to determine the amount of delay in samples. Bit 8: Application_Restart_Enable Used along with Kickstart_Enable in only the AC3 and C.O.S. application codes, restarts the downloaded software application. 0 - Disables Application_Restart 1 - Enables Application_Restart When enabled, this action allows input unit settings to be changed on the fly without resetting the output unit (which may cause clicks on the DAC if in master mode). This action also allows changes of processing parameters (such as filter coefficients) without resetting the output unit. Application Restart effectively places the CS4932X in a startup state as if it had been issued a hardware reset plus a soft reset, except that the preset output and control port configuration is maintained. Except for issuing output and host initialization messages, after an Application_Restart, the host should treat the CS4932X as it would at start-up, i.e., issue input unit initialization, setup application parameters, and then kickstart. After sending an Application Restart, the host should wait a certain amount of time after sending
38
Application Restart before sending any further commands to the CS4932X. For example, here are several delay sequences that should be followed: If the AC-3 Application Code was processing an AC-3 Stream, the host must wait 35 ms. If the C.O.S. application code or AC-3 application code was processing PCM, the host must wait 12 ms.
Note: Just as with Soft Reset, all Default* settings are untouched by Application Restart.
Bit 4: PLL_Enable Enables the use of the Phase Locked Loop (PLL) to generate internal DSPCLK from the desired external CLKIN. 0 - Disables PLL 1 - Enables PLL The register values in PLL_Register_1 (0x17) and PLL_Register_2 (0x18) define the desired DSP clock and reference CLKIN used. The default values at download, set up a 86 MHz DSPCLK from a 12.288 MHz external CLKIN. These register values should be downloaded prior to Kickstart with PLL_Enable. Use the corresponding PLL values located in Table 9 for 24.576 MHz external CLKIN or 27 MHz external CLKIN. For other system requirements, please contact the factory for appropriate register values. Bit 0: Kickstart Enable Puts CS4932X into run-time mode. Indicates that all hardware and software configuration activities have been completed and CS4932X can start the application. 0 - Application continues waiting for kickstart 1 - Application is kickstarted
4.1.2 Noise_Control (Index 0x01): (Applicable to AC-3 Code Only)
Switches ON Pink/White Noise in Channels L, C, R, Ls, Rs and LFE. Pink/White noise will not be enabled unless the AC-3 application code is configured for PCM pass-through and the Output_Channel is set to a value in the range
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0x0 - 0x5. Autodetect should not be enabled when using Pink/White Noise. Bit 5: Level_Select Level_Select sets up output level. 0 - Default level (Dolby-compatible) 1 - 10.5 dB boosted with respect to default level. Bit 4: White_Noise_Enable White_Noise_Enable enables White noise if set (otherwise output is Pink noise). 0 - Pink Noise 1 - White Noise Bits 3:0: Output Channel Output Channel selects channel for noise output 0x0 - Left 0x1 -Center 0x2 -Right 0x3 -Left Surround (This value controls both Ls and Rs in Pro Logic Mode) 0x4 -Right Surround (This value has no effect in Pro Logic mode) 0x5 -LFE 0x6 -Lz 0x7 -Rz 0x8-0xf - Disabled For Pro Logic processing of PCM streams or when digital deemphasis is enabled (refer to Section 5.2 "PCM Manager (for AC-3 Application Code)" on page 78), the correct sampling frequency must also be given to the Pro Logic Manager for proper operation of filters. For compressed inputs, such as AC-3, DTS, MPEG, or AAC, the decoder application will set up sampling frequency information based on the information parsed from the bitstream; consequently, this field does not need to be set. Bits 2:0: Sampling_Frequency_Code 0x0 - 48kHz 0x1 - 44.1kHz 0x2 - 32kHz For 192 kHz or 96 kHz Stereo PCM support, the host should set the Fs to 48 kHz and download the correct 192 kHz or 96 kHz coefficients in the Generalized Bass Manager in the place of the 48 kHz coefficients. For 176.4 kHz or 88.2 kHz Stereo PCM support, the host should set the Fs to 44.1 kHz and download the correct 176.4 kHz or 88.2 kHz coefficients in the Generalized Bass Manager in the place of the 44.1 kHz coefficients. For 128 kHz or 64 kHz Stereo PCM support, the host should set the Fs to 32 kHz and download the correct 128 kHz or 64 kHz coefficients in the Generalized Bass Manager in the place of the 32 kHz coefficients. Please contact your FAE for the appropriate 192 kHz, 176.4 kHz, 128 kHz, 96 kHz, 88.2 kHz or 64 kHz coefficient files, along with the desired crossover frequencies you need. These coefficients only effect the Generic Bass Manager module and do not effect the generic Pro Logic module. Currently, the Pro Logic module only supports sampling frequencies up to 48 kHz for the standard application code releases of AC-3, DTS, MPEG Multichannel, and AAC.
4.1.3 Sampling Frequency Code (Index 0x14)
This parameter controls the sampling frequency for PCM applications. This must be set by the host during while a PCM application is loaded, but not during runtime.
Note: For PCM input, the DSP does NOT know the sampling frequency setting. If the system expects to receive PCM from a S/PDIF receiver, it is up to the host to inquire about the sampling frequency information from the receiver, and then inform the DSP.
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4.2 Generalized Bass Manager for Dolby Digital, DTS, MP3, MPEG Multichannel, AAC, and Crystal Original Surround
Write Opcode = 0x94; Read Opcode = 0x15; Read Response Opcode = 0x95 Write = 0x9400HH 0xhhhhhh Read Request = 0x1500HH; Read Response = 0x9500HH 0xhhhhhh 0xHH = index, 0xhhhhhh = data value
Index
0x00
Variable
BASS_MGR_CONTROL
Dataword Content
Bit 16 = LFE_To_Sub_Enable = 0/1 = Disable/Enable LFE Only to subwoofer. Bit 12 = LPF_Enable = 0/1 = Disable/Enable 4th Order LPF at the output of summer. Bit 8 = HPF_L_Enable = 0/1/ = Disable/Enable L channel 2nd Order HPF. Bit 7 = HPF_C_Enable = 0/1/ = Disable/Enable C channel 2nd Order HPF. Bit 6 = HPF_R_Enable = 0/1/ = Disable/Enable R channel 2nd Order HPF. Bit 5 = HPF_Ls_Enable = 0/1/ = Disable/Enable Ls channel 2nd Order HPF. Bit 4 = HPF_Rs_Enable = 0/1/ = Disable/Enable Rs channel 2nd Order HPF. Bit 0 = Bass_Mgr_Enable = 0/1 = Disable/Enable Bass Manager post-processing. All other bits should be 0. Default = 0x000000 0.0-1.0 = Level adjustment for input L channel pass-through. Default* = 0x7FFFFF 0.0-1.0 = Level adjustment for input C channel input pass-through. Default* = 0x7FFFFF 0.0-1.0 = Level adjustment for input R channel input pass-through. Default* = 0x7FFFFF 0.0-1.0 = Level adjustment for input Ls channel pass-through. Note: Setting not applicable when Pro Logic is Enabled. Bass information from surround channels will never be input. Default* = 0x7FFFFF 0.0-1.0 = Level adjustment for input Rs channel pass-through. Note: Setting not applicable when Pro Logic is Enabled. Bass information from surround channels will never be input. Default* = 0x7FFFFF 0.0-1.0 = Level adjustment for input LFE channel pass-through. Default* = 0x7FFFFF 0.0-1.0 = Level adjustment for bass summer output contribution to L channel output. Default* = 0x000000 Table 10. Bass Manager
0x01 0x02 0x03 0x04
BASS_MGR_INPUT_ L_LEVEL BASS_MGR_INPUT_ C_LEVEL BASS_MGR_INPUT_ R_LEVEL BASS_MGR_INPUT_ LS_LEVEL
0x05
BASS_MGR_INPUT_ RS_LEVEL
0x06 0x07
BASS_MGR_INPUT_ LFE_LEVEL BASS_MGR_SUM_ OUTPUT_L_LEVEL
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Index
0x08
Variable
BASS_MGR_SUM_ OUTPUT_C_LEVEL BASS_MGR_SUM_ OUTPUT_R_LEVEL BASS_MGR_SUM_ OUTPUT_LS_LEVEL BASS_MGR_SUM_ OUTPUT_RS_LEVEL BASS_MGR_LFE_ L_LEVEL BASS_MGR_LFE _C_LEVEL BASS_MGR_LFE_ R_LEVEL BASS_MGR_LFE_ LS_LEVEL BASS_MGR_LFE_ RS_LEVEL BASS_MGR_SUM_ INPUT_L_LEVEL BASS_MGR_SUM_ INPUT_C_LEVEL BASS_MGR_SUM_ INPUT_R_LEVEL BASS_MGR_SUM_ INPUT_LS_LEVEL BASS_MGR_SUM_ INPUT_RS_LEVEL BASS_MGR_SUM_ INPUT_LFE_LEVEL
Dataword Content
0.0-1.0 = Level adjustment for bass summer output contribution to C channel output. Default* = 0x000000 0.0-1.0 = Level adjustment for bass summer output contribution to R channel output. Default* = 0x000000 0.0-1.0 = Level adjustment for bass summer output contribution to Ls channel output. Default* = 0x000000 0.0-1.0 = Level adjustment for bass summer output contribution to Rs channel output. Default* = 0x000000 0.0-1.0 = Level adjustment for input LFE contribution to L channel output. Default* = 0x000000 0.0-1.0 = Level adjustment for input LFE contribution to C channel output. Default* = 0x000000 0.0-1.0 = Level adjustment for input LFE contribution to R channel output. Default* = 0x000000 0.0-1.0 = Level adjustment for input LFE contribution to Ls channel output. Default* = 0x000000 0.0-1.0 = Level adjustment for input LFE contribution to Rs channel output. Default* = 0x000000 0.0-1.0 = Level adjustment for input L channel contribution to bass summer input. Default* = 0x16c311 0.0-1.0 = Level adjustment for input C channel contribution to bass summer input. Default* = 0x16c311 0.0-1.0 = Level adjustment for input R channel contribution to bass summer input. Default* = 0x16c311 0.0-1.0 = Level adjustment for input Ls channel contribution to bass summer input. Default* = 0x16c311 0.0-1.0 = Level adjustment for input Rs channel contribution to bass summer input. Default* = 0x16c311 0.0-1.0 = Level adjustment for input LFE channel contribution to bass summer.
0x09
0x0a
0x0b
0x0c 0x0d 0x0e 0x0f 0x10 0x11
0x12
0x13
0x14
0x15
0x16 0x17
Default* = 0x47facd BASS_MGR_LPF_48_B0 0.0-1.0 = b0 coefficient for Low Pass Filter, Fs = 48 kHz. (Note a) Default* = 0x000072 Table 10. Bass Manager (Continued)
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Index
0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x1e 0x1f 0x20 0x21 0x22 0x23 0x24 0x25 0x26 0x27 0x28 0x29 0x2a 0x2b
Variable
Dataword Content
BASS_MGR_LPF_48_B1 0.0-1.0 = b1 coefficient for Low Pass Filter, Fs = 48 kHz. (Note a) Default* = 0x0000e4 BASS_MGR_LPF_48_B2 0.0-1.0 = b2 coefficient for Low Pass Filter, Fs = 48 kHz. (Note a) Default* =0x000072 BASS_MGR_LPF_48_A1 0.0-1.0 = a1 coefficient for Low Pass Filter, Fs = 48 kHz. (Note a) Default* = 0x7f0d13 BASS_MGR_LPF_48_A2 0.0-1.0 = a2 coefficient for Low Pass Filter, Fs = 48 kHz. (Note a) Default* = 0xc0f124 BASS_MGR_HPF_48_B0 0.0-1.0 = b0 coefficient for High Pass Filter, Fs = 48 kHz. (Note a) Default* = 0x3f8745 BASS_MGR_HPF_48_B1 0.0-1.0 = b1 coefficient for High Pass Filter, Fs = 48 kHz. (Note a) Default* = 0x80f176 BASS_MGR_HPF_48_B2 0.0-1.0 = b2 coefficient for High Pass Filter, Fs = 48 kHz. (Note a) Default* = 0x3f8745 BASS_MGR_HPF_48_A1 0.0-1.0 = a1 coefficient for High Pass Filter, Fs = 48 kHz. (Note e) Default* = 0x7f0da6 BASS_MGR_HPF_48_A2 0.0-1.0 = a2 coefficient for High Pass Filter, Fs = 48 kHz. (Note a) Default* = 0xc0f092 BASS_MGR_LPF_44_B0 0.0-1.0 = b0 coefficient for Low Pass Filter, Fs = 44.1 kHz. (Note b) Default* = 0x000087 BASS_MGR_LPF_44_B1 0.0-1.0 = b1 coefficient for Low Pass Filter, Fs = 44.1 kHz. (Note b) Default* = 0x00010e BASS_MGR_LPF_44_B2 0.0-1.0 = b2 coefficient for Low Pass Filter, Fs = 44.1 kHz. (Note b) Default* = 0x000087 BASS_MGR_LPF_44_A1 0.0-1.0 = a1 coefficient for Low Pass Filter, Fs = 44.1 kHz. (Note b) Default* = 0x7ef797 BASS_MGR_LPF_44_A2 0.0-1.0 = a2 coefficient for Low Pass Filter, Fs = 44.1 kHz. (Note b) Default* = 0xc1064b BASS_MGR_HPF_44_B0 0.0-1.0 = b0 coefficient for High Pass Filter, Fs = 44.1 kHz. (Note b) Default* = 0x3f7ca3 BASS_MGR_HPF_44_B1 0.0-1.0 = b1 coefficient for High Pass Filter, Fs = 44.1 kHz. (Note b) Default* = 0x8106ba BASS_MGR_HPF_44_B2 0.0-1.0 = b2 coefficient for High Pass Filter, Fs = 44.1 kHz. (Note f) Default* = 0x3f7ca3 BASS_MGR_HPF_44_A1 0.0-1.0 = a1 coefficient for High Pass Filter, Fs = 44.1 kHz. (Note f) Default* = 0x7ef838 BASS_MGR_HPF_44_A2 0.0-1.0 = a2 coefficient for High Pass Filter, Fs = 44.1 kHz. (Note b) Default* = 0xc105ad BASS_MGR_LPF_32_B0 0.0-1.0 = b0 coefficient for Low Pass Filter, Fs = 32 kHz. (Note c) Default* = 0x000100 Table 10. Bass Manager (Continued)
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Index
0x2c 0x2d 0x2e 0x2f 0x30 0x31 0x32 0x33 0x34
Variable
Dataword Content
BASS_MGR_LPF_32_B1 0.0-1.0 = b1 coefficient for Low Pass Filter, Fs = 32 kHz. (Note c) Default* = 0x000200 BASS_MGR_LPF_32_B2 0.0-1.0 = b2 coefficient for Low Pass Filter, Fs = 32 kHz. (Note c) Default* =0x000100 BASS_MGR_LPF_32_A1 0.0-1.0 = a1 coefficient for Low Pass Filter, Fs = 32 kHz. (Note c) Default* = 0x7e939f BASS_MGR_LPF_32_A2 0.0-1.0 = a2 coefficient for Low Pass Filter, Fs = 32 kHz. (Note c) Default* = 0xc1685f BASS_MGR_HPF_32_B0 0.0-1.0 = b0 coefficient for High Pass Filter, Fs = 32 kHzc. (note c) Default* = 0x2a77ee BASS_MGR_HPF_32_B1 0.0-1.0 = b1 coefficient for High Pass Filter, Fs = 32 kHz. (note c) Default* = 0xf8c361 BASS_MGR_HPF_32_B2 0.0-1.0 = b2 coefficient for High Pass Filter, Fs = 32 kHz. (note g) Default* = 0x2a77ee BASS_MGR_HPF_32_A1 0.0-1.0 = a1 coefficient for High Pass Filter, Fs = 32 kHz. (note c) Default* = 0x7e947c BASS_MGR_HPF_32_A2 0.0-1.0 = a2 coefficient for High Pass Filter, Fs = 32 kHz. (note c) Default* = 0xc16787 Table 10. Bass Manager (Continued)
Notes: a. Pre-loaded coefficients will provide a 80 Hz Crossover if input Fs is 48 kHz. A different set of coefficients must be downloaded for 96 kHz or 192 kHz Fs support. b. Pre-loaded coefficients will provide a 80 Hz Crossover if input Fs is 44.1 kHz. A different set of coefficients must be downloaded for 88.2 kHz or 176.4 kHz Fs support. c. Pre-loaded coefficients will provide a 80 Hz Crossover if input Fs is 32 kHz. A different set of coefficients must be downloaded for 64 kHz or 128 kHz Fs support. See definition on page 32. "Default" vs. "Default*" See definition page 32.
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4.2.1 Bass Manager Block Diagrams
The Bass Manager is best described with the use of a block diagram. Figure 8 shows the topology of the filters used for both the low pass and the high pass filters. The coefficients of LPF and HPF represent the second order digital filter H(z) = (b0 + b1z[-1]+ b2z[-2]) / (1 - a1 z[-1] - a2[-2]). The LPF implementation cascades 2 such identical filters to produce a 4th order filter Linkwitz-Riley filter. Therefore, the corner frequency of the LPF is 6 dB. Hence, if it is important to ensure a particular overall 3 dB corner frequency (rather than other attenuation), it is required to specially design the individual LPF and HPF components to have appropriate attenuation at this desired corner frequency. The above default values have been designed to result in an 3 dB corner frequency of 80 Hz for the HPF and 6dB corner frequency of 80 Hz for LPF. Figure 9, "Subwoofer Summing Module," on page 45 shows the summing module used to redirect bass to the subwoofer channel. Figure 10, "Bass Manager Processing Unit," on page 45 shows the Bass Manager processing module. Figure 10 also shows the processing unit on each of the output channels (except the LFE).
b0
+
Z-1
Z-1
b1
a1
Z-1
Z-1
b2
a2
Bass Manager High Pass Filter
b0
+
b0
+
Z-1
Z-1
Z-1
Z-1
b1
a1
b1
a1
Z-1
Z-1
Z-1
Z-1
b2
a2
b2
a2
Bass Manager Low Pass Filter
Figure 8. Filter Topology
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Sum_Input_X_Level L C R Ls To Subwoofer Rs LFE LFE Input_LFE_Level
Figure 9. Subwoofer Summing Module
To Processing Unit
LPF_Enable
LFE_Only_Sub_Enable
Input_X_Level Input Channel X Output Channel X
HPF_X_Enable
Sum_Output_X_Level
LFE_X_Level
To Summing Module
Summing Module
LFE
Figure 10. Bass Manager Processing Unit
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4.2.2 Dolby Digital Bass Management Configurations
Tables 11 through 18 show the values that need to be written for the different Dolby Digital Bass Management configurations. All variables (except filter coefficients) can be changed during runtime. The following tables are in no way inclusive of all of the different modes possible, they are just presented to show the different configurations required for Dolby Digital Certification. Different configurations are required depending on the level of certification that is desired. Any questions regarding certification should be directed towards Dolby Laboratories.
Write Opcode
0x940000
Bass Manager Variables
BASS_MGR_CONTROL
Values
LFE_ONLY_SUB DISABLED LPF DISABLED HPF_L DISABLED HPF_C DISABLED HPF_R DISABLED HPF_LS DISABLED HPF_RS DISABLED BASS_MGR ENABLED 0 dB 0 dB 0 dB 0 dB 0 dB 0 dB - dB - dB - dB - dB - dB - dB - dB - dB - dB - dB -15 dB -15 dB -15 dB -15 dB -15 dB -5 dB Table 11. Dolby Configuration 0
HEX Value
0x000001
INPUT_L_LEVEL INPUT_C_LEVEL INPUT_R_LEVEL INPUT_LS_LEVEL INPUT_RS_LEVEL INPUT_LFE_LEVEL SUM_OUTPUT_L_LEVEL SUM_OUTPUT_C_LEVEL SUM_OUTPUT_R_LEVEL SUM_OUTPUT_LS_LEVEL SUM_OUTPUT_RS_LEVEL LFE_L_LEVEL LFE_C_LEVEL LFE_R_LEVEL LFE_LS_LEVEL LFE_RS_LEVEL SUM_INPUT_L_LEVEL SUM_INPUT_C_LEVEL SUM_INPUT_R_LEVEL SUM_INPUT_LS_LEVEL SUM_INPUT_RS_LEVEL SUM_INPUT_LFE_LEVEL
0x940001 0x940002 0x940003 0x940004 0x940005 0x940006 0x940007 0x940008 0x940009 0x94000A 0x94000B 0x94000C 0x94000D 0x94000E 0x94000F 0x940010 0x940011 0x940012 0x940013 0x940014 0x940015 0x940016
0x7FFFFF 0x7FFFFF 0x7FFFFF 0x7FFFFF 0x7FFFFF 0x7FFFFF 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x16C311 0x16C311 0x16C311 0x16C311 0x16C311 0x47FACD
46
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Write Opcode
0x940000
Bass Manager Variables
BASS_MGR_CONTROL
Values
LFE_Only_Sub Disabled LPF Enabled HPF_L Enabled HPF_C Enabled HPF_R Enabled HPF_Ls Enabled HPF_Rs Enabled Bass_Mgr Enabled 0 dB 0 dB 0 dB 0 dB 0 dB 0 dB - dB - dB - dB - dB - dB - dB - dB - dB - dB - dB -15 dB -15 dB -15 dB -15 dB -15 dB -5 dB Table 12. Dolby Configuration 1
HEX Value
0x0011F1
INPUT_L_LEVEL INPUT_C_LEVEL INPUT_R_LEVEL INPUT_LS_LEVEL INPUT_RS_LEVEL INPUT_LFE_LEVEL SUM_OUTPUT_L_LEVEL SUM_OUTPUT_C_LEVEL SUM_OUTPUT_R_LEVEL SUM_OUTPUT_LS_LEVEL SUM_OUTPUT_RS_LEVEL LFE_L_LEVEL LFE_C_LEVEL LFE_R_LEVEL LFE_LS_LEVEL LFE_RS_LEVEL SUM_INPUT_L_LEVEL SUM_INPUT_C_LEVEL SUM_INPUT_R_LEVEL SUM_INPUT_LS_LEVEL SUM_INPUT_RS_LEVEL SUM_INPUT_LFE_LEVEL
0x940001 0x940002 0x940003 0x940004 0x940005 0x940006 0x940007 0x940008 0x940009 0x94000A 0x94000B 0x94000C 0x94000D 0x94000E 0x94000F 0x940010 0x940011 0x940012 0x940013 0x940014 0x940015 0x940016
0x7FFFFF 0x7FFFFF 0x7FFFFF 0x7FFFFF 0x7FFFFF 0x7FFFFF 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x16C311 0x16C311 0x16C311 0x16C311 0x16C311 0x47FACD
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Write Opcode
0x940000
Bass Manager Variables
BASS_MGR_CONTROL
Values
LFE_Only_Sub Disabled LPF Enabled HPF_L Disabled HPF_C Enabled HPF_R Disabled HPF_Ls Enabled HPF_Rs Enabled Bass_Mgr Enabled -12 dB ADJ, default = 0 dB -12 dB ADJ, default = 0 dB ADJ, default = 0 dB 0 dB -1.5 dB - dB -1.5 dB - dB - dB - dB - dB - dB - dB - dB - dB -15 dB - dB -15 dB -15 dB -5 dB Table 13. Dolby Configuration 2
HEX Value
0x0010B1
INPUT_L_LEVEL INPUT_C_LEVEL INPUT_R_LEVEL INPUT_LS_LEVEL INPUT_RS_LEVEL INPUT_LFE_LEVEL SUM_OUTPUT_L_LEVEL SUM_OUTPUT_C_LEVEL SUM_OUTPUT_R_LEVEL SUM_OUTPUT_LS_LEVEL SUM_OUTPUT_RS_LEVEL LFE_L_LEVEL LFE_C_LEVEL LFE_R_LEVEL LFE_LS_LEVEL LFE_RS_LEVEL SUM_INPUT_L_LEVEL SUM_INPUT_C_LEVEL SUM_INPUT_R_LEVEL SUM_INPUT_LS_LEVEL SUM_INPUT_RS_LEVEL SUM_INPUT_LFE_LEVEL
0x940001 0x940002 0x940003 0x940004 0x940005 0x940006 0x940007 0x940008 0x940009 0x94000A 0x94000B 0x94000C 0x94000D 0x94000E 0x94000F 0x940010 0x940011 0x940012 0x940013 0x940014 0x940015 0x940016
0x2026F2 0x7FFFFF 0x2026F2 0x7FFFFF 0x7FFFFF 0x7FFFFF 0x6BB2D5 0x000000 0x6BB2D5 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x16C311 0x000000 0x16C311 0x16C311 0x47FACD
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Write Opcode
0x940000
Bass Manager Variables
BASS_MGR_CONTROL
Values
LFE_Only_Sub Disabled LPF Enabled HPF_L Disabled HPF_C Enabled HPF_R Disabled HPF_Ls Enabled HPF_Rs Enabled Bass_Mgr Enabled 0 dB 0 dB 0 dB 0 dB 0 dB 0 dB - dB - dB - dB - dB - dB - dB - dB - dB - dB - dB - dB -15 dB - dB -15 dB -15 dB -5 dB
HEX Value
0x0010B1
INPUT_L_LEVEL INPUT_C_LEVEL INPUT_R_LEVEL INPUT_LS_LEVEL INPUT_RS_LEVEL INPUT_LFE_LEVEL SUM_OUTPUT_L_LEVEL SUM_OUTPUT_C_LEVEL SUM_OUTPUT_R_LEVEL SUM_OUTPUT_LS_LEVEL SUM_OUTPUT_RS_LEVEL LFE_L_LEVEL LFE_C_LEVEL LFE_R_LEVEL LFE_LS_LEVEL LFE_RS_LEVEL SUM_INPUT_L_LEVEL SUM_INPUT_C_LEVEL SUM_INPUT_R_LEVEL SUM_INPUT_LS_LEVEL SUM_INPUT_RS_LEVEL SUM_INPUT_LFE_LEVEL
0x940001 0x940002 0x940003 0x940004 0x940005 0x940006 0x940007 0x940008 0x940009 0x94000A 0x94000B 0x94000C 0x94000D 0x94000E 0x94000F 0x940010 0x940011 0x940012 0x940013 0x940014 0x940015 0x940016
0x7FFFFF 0x7FFFFF 0x7FFFFF 0x7FFFFF 0x7FFFFF 0x7FFFFF 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x16C311 0x000000 0x16C311 0x16C311 0x47FACD
Table 14. Dolby Alternative Configuration 2
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Write Opcode
0x940000
Bass Manager Variables
BASS_MGR_CONTROL
Values
LFE_Only_Sub Disabled LPF Enabled HPF_L Disabled HPF_C Enabled HPF_R Disabled HPF_Ls Disabled HPF_Rs Disabled Bass_Mgr Enabled -8 dB -8 dB -8 dB -8 dB -8 dB -8 dB -4.5 dB - dB -4.5 dB - dB - dB -8 dB - dB -8 dB -8 dB -8 dB - dB -8 dB - dB - dB - dB - dB
HEX Value
0x001081
INPUT_L_LEVEL INPUT_C_LEVEL INPUT_R_LEVEL INPUT_LS_LEVEL INPUT_RS_LEVEL INPUT_LFE_LEVEL SUM_OUTPUT_L_LEVEL SUM_OUTPUT_C_LEVEL SUM_OUTPUT_R_LEVEL SUM_OUTPUT_LS_LEVEL SUM_OUTPUT_RS_LEVEL LFE_L_LEVEL LFE_C_LEVEL LFE_R_LEVEL LFE_LS_LEVEL LFE_RS_LEVEL SUM_INPUT_L_LEVEL SUM_INPUT_C_LEVEL SUM_INPUT_R_LEVEL SUM_INPUT_LS_LEVEL SUM_INPUT_RS_LEVEL SUM_INPUT_LFE_LEVEL
0x940001 0x940002 0x940003 0x940004 0x940005 0x940006 0x940007 0x940008 0x940009 0x94000A 0x94000B 0x94000C 0x94000D 0x94000E 0x94000F 0x940010 0x940011 0x940012 0x940013 0x940014 0x940015 0x940016
0x32F52C 0x32F52C 0x32F52C 0x32F52C 0x32F52C 0x32F52C 0x4C3EA7 0x000000 0x4C3EA7 0x000000 0x000000 0x32F52C 0x000000 0x32F52C 0x32F52C 0x32F52C 0x000000 0x32F52C 0x000000 0x000000 0x000000 0x000000
Table 15. Dolby Configuration 3 (Normal - No Sub Out)
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Write Opcode
0x940000
Bass Manager Variables
BASS_MGR_CONTROL
Values
LFE_Only_Sub Disabled LPF Enabled HPF_L Disabled HPF_C Enabled HPF_R Disabled HPF_Ls Disabled HPF_Rs Disabled Bass_Mgr Enabled -4 dB -4 dB -4 dB -4 dB -4 dB -4 dB -4.5 dB - dB -4.5 dB - dB - dB - dB - dB - dB - dB - dB - dB -4 dB - dB - dB - dB - dB Table 16. Dolby Configuration 3 (Sub Out)
HEX Value
0x011081
INPUT_L_LEVEL INPUT_C_LEVEL INPUT_R_LEVEL INPUT_LS_LEVEL INPUT_RS_LEVEL INPUT_LFE_LEVEL SUM_OUTPUT_L_LEVEL SUM_OUTPUT_C_LEVEL SUM_OUTPUT_R_LEVEL SUM_OUTPUT_LS_LEVEL SUM_OUTPUT_RS_LEVEL LFE_L_LEVEL LFE_C_LEVEL LFE_R_LEVEL LFE_LS_LEVEL LFE_RS_LEVEL SUM_INPUT_L_LEVEL SUM_INPUT_C_LEVEL SUM_INPUT_R_LEVEL SUM_INPUT_LS_LEVEL SUM_INPUT_RS_LEVEL SUM_INPUT_LFE_LEVEL
0x940001 0x940002 0x940003 0x940004 0x940005 0x940006 0x940007 0x940008 0x940009 0x94000A 0x94000B 0x94000C 0x94000D 0x94000E 0x94000F 0x940010 0x940011 0x940012 0x940013 0x940014 0x940015 0x940016
0x50C335 0x50C335 0x50C335 0x50C335 0x50C335 0x50C335 0x4C3EA7 0x000000 0x4C3EA7 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x50C335 0x000000 0x000000 0x000000 0x000000
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Write Opcode
0x940000
Bass Manager Variables
BASS_MGR_CONTROL
Values
LFE_Only_Sub Enabled LPF Enabled HPF_L Disabled HPF_C Enabled HPF_R Disabled HPF_Ls Enabled HPF_Rs Enabled Bass_Mgr Enabled 0 dB 0 dB 0 dB 0 dB 0 dB -5 dB 0 dB - dB 0 dB - dB - dB - dB - dB - dB - dB - dB - dB -4.5 dB - dB -4.5 dB -4.5 dB - dB
HEX Value
0x0110B1
INPUT_L_LEVEL INPUT_C_LEVEL INPUT_R_LEVEL INPUT_LS_LEVEL INPUT_RS_LEVEL INPUT_LFE_LEVEL SUM_OUTPUT_L_LEVEL SUM_OUTPUT_C_LEVEL SUM_OUTPUT_R_LEVEL SUM_OUTPUT_LS_LEVEL SUM_OUTPUT_RS_LEVEL LFE_L_LEVEL LFE_C_LEVEL LFE_R_LEVEL LFE_LS_LEVEL LFE_RS_LEVEL SUM_INPUT_L_LEVEL SUM_INPUT_C_LEVEL SUM_INPUT_R_LEVEL SUM_INPUT_LS_LEVEL SUM_INPUT_RS_LEVEL SUM_INPUT_LFE_LEVEL
0x940001 0x940002 0x940003 0x940004 0x940005 0x940006 0x940007 0x940008 0x940009 0x94000A 0x94000B 0x94000C 0x94000D 0x94000E 0x94000F 0x940010 0x940011 0x940012 0x940013 0x940014 0x940015 0x940016
0x7FFFFF 0x7FFFFF 0x7FFFFF 0x7FFFFF 0x7FFFFF 0x47FACC 0x7FFFFF 0x000000 0x7FFFFF 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x4C3EA7 0x000000 0x4C3EA7 0x4C3EA7 0x000000
Table 17. Dolby DVD Configuration (Bass to L/R)
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Write Opcode
0x940000
Bass Manager Variables
BASS_MGR_CONTROL
Values
LFE_Only_Sub Disabled LPF Enabled HPF_L Disabled HPF_C Enabled HPF_R Disabled HPF_Ls Enabled HPF_Rs Enabled Bass_Mgr Enabled 0 dB 0 dB 0 dB 0 dB 0 dB 0 dB - dB - dB - dB - dB - dB - dB - dB - dB - dB - dB - dB -15 dB - dB -15 dB -15 dB -5 dB
HEX Value
0x0010B1
INPUT_L_LEVEL INPUT_C_LEVEL INPUT_R_LEVEL INPUT_LS_LEVEL INPUT_RS_LEVEL INPUT_LFE_LEVEL SUM_OUTPUT_L_LEVEL SUM_OUTPUT_C_LEVEL SUM_OUTPUT_R_LEVEL SUM_OUTPUT_LS_LEVEL SUM_OUTPUT_RS_LEVEL LFE_L_LEVEL LFE_C_LEVEL LFE_R_LEVEL LFE_LS_LEVEL LFE_RS_LEVEL SUM_INPUT_L_LEVEL SUM_INPUT_C_LEVEL SUM_INPUT_R_LEVEL SUM_INPUT_LS_LEVEL SUM_INPUT_RS_LEVEL SUM_INPUT_LFE_LEVEL
0x940001 0x940002 0x940003 0x940004 0x940005 0x940006 0x940007 0x940008 0x940009 0x94000A 0x94000B 0x94000C 0x94000D 0x94000E 0x94000F 0x940010 0x940011 0x940012 0x940013 0x940014 0x940015 0x940016
0x7FFFFF 0x7FFFFF 0x7FFFFF 0x7FFFFF 0x7FFFFF 0x7FFFFF 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x000000 0x16C310 0x000000 0x16C310 0x16C310 0x47FACC
Table 18. Dolby DVD Configuration (Bass to Subwoofer)
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4.3 Unsolicited Messages (Read-Only)
No Write Message. No Read Request. Unsolicited Read Response = 0x8700HH 0xhhhhhh 0xHH = index, 0xhhhhhh = data value
Index
0x00
Variable
AUTODETECT_ RESPONSE
Dataword Content
Bit 23 = Decodable_Stream_Flag= 0/1 = This stream is not/is decodable by the application (no need for new download if 1). Bits 22:6 = Reserved. Bit 5 Non_IEC61937_Stream_Flag= 1/0 = This stream is not/is IEC61937 compressed data. If Non_IEC61937_Stream_Flag=1 Bits 4:0 = Non_IEC61937 Stream Descriptor. 0x00 = Silent Input Data (Out of Application Sync). 0x01 = DTS Format-16 elementary stream. 0x02 = DTS Format-14 elementary stream. 0x03 = Linear PCM stream. 0x04 = HDCD PCM stream (only available in HDCD application). 0x05-0x1F = Reserved. If Non_IEC61937_Stream_Flag=0 Bits 4:0 = IEC61937 Stream Descriptor = Identical to bits [4:0] of the Pc burst data type descriptor in IEC61937 specification. Description of the data-type field of Pc reproduced below from IEC61937 Specification (current as of 11/97): 0x00 = Never Reported. (Null data is ignored by CS4932X, unless Bit 13 in Audio_Mgr_Control is set, see Section 4.1). 0x01 = AC-3 data. 0x02 = Reserved. 0x03 = Never Reported. (Pause is ignored by CS4932X, unless Bit 13 in Audio_Mgr_Control is set, see Section 4.1). 0x04 = MPEG-1 Layer 1 data. 0x05 = MPEG-1 Layer 2 or 3 data or MPEG-2 without extension. 0x06 = MPEG-2 data with extension. 0x07 = Reserved. 0x08 = MPEG-2 Layer 1 Low sampling frequency. 0x09 = MPEG-2 Layer 2 or 3 Low sampling frequency. 0x0A = Reserved. 0x0B = DTS-1 data (512-sample bursts). 0x0C = DTS-2 data (1024-sample bursts). 0x0D = DTS-3 data (2048-sample bursts). 0x0E - 0x1B = Reserved. 0x1C = MPEG-2 AAC ADTS data. 0x1D - 0x1F = Reserved. Bits 23:0 = Reserved. Please see Section 4.1.1 "Audio_Control: (Index 0x00)" on page 37 bit 15 for more information. Bit 23 = 1. Bits 22:0 = Reserved. Table 19. Unsolicited Messages
0x03
0x10
AUDIO_ CONFIGURATION_ CHANGE PLL_OUT_OF_LOCK
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4.3.1 Autodetect Operation
The sequence of events involving autodetection are described below from the host's perspective:
Note: This example assumes CS49326, CS49328, or CS49329 is being used since only these devices can play DTS. In the case of DTS being detected on CS49325 or AC-3 detected on CS49326, the host should display an appropriate user message to the front panel stating that the detected stream (DTS or AC-3) cannot be played on this receiver.
outputs, and issues an Unsolicited Message to the host indicating the data type with Decodable_Bitstream_Flag=0. In our example, the message would be 0x870000 0x000021. On receiving this message, host repeats Steps 1 onwards but this time downloads the DTS code to the CS4932X (along with correct hardware configuration and application configuration for DTS with Autodetect enabled). Subsequently, DTS will be detected within 100 mS and successfully played by the new DTS code, after sending the corresponding unsolicited message (0x870000 0x800021). If the host desires, it can reduce the latency of this second autodetection, by enabling Startup_Autodetect_Bypass in the Kickstart. This will allow the DTS decoder to directly start decoding the input data (which is known to be DTS), while still retaining subsequent autodetect functionality. 6) After the steps above have taken place and while CS4932X is successfully playing the input bitstream (still AC-3 in our example), if the host receives external information that the input has been changed (e.g., the user selects a new source using the front panel buttons), then before switching the input data to the CS4932X, the host should send a Soft Reset (or an Application Restart, for AC-3 or C.O.S. code) message. (See Section 4.1 "Audio Manager for Dolby Digital, DTS, MP3, MPEG Multichannel, AAC and Crystal Original Surround" on page 33.) If Application Restart is used, this effectively puts CS4932X in Step 2, without changing the output hardware configuration (i.e., output clocks are not interrupted). The input configuration would need to be changed if it is different than the default. If Soft Reset is issued, then the entire hardware configuration needs to be resent.
55
1) Host downloads CS4932X with a tentative application code, for this example we will use AC3I32X7.LD. 2) Host then configures the CS4932X hardware appropriately and sets up application parameters as desired, including enabling the desired application. For this example we will say the code is configured for AC-3 decode. 3) Host then kickstarts CS4932X with Autodetect enabled (see Section 4.1 "Audio Manager for Dolby Digital, DTS, MP3, MPEG Multichannel, AAC and Crystal Original Surround" on page 33 for details). 4) The Autodetect module of the enabled application of the CS4932X analyzes the input for a maximum of 100 ms of non-silent/nonpause data and determines the content of the input bitstream. 5a) If the enabled application can play the detected input (i.e., if AC-3 was detected in this case), then the CS4932X issues an Unsolicited Message to the host indicating the data type with Decodable_Bitstream_Flag=1. In our example of the IEC61937 AC-3 stream, the message would be 0x870000 0x800001. The CS4932X then goes ahead and processes it according to the application parameters as setup in Step 2 above. 5b) If the enabled application cannot play the detected input (say Non-IEC61937 LD DTS was detected), then the CS4932X soft mutes the
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The host should then repeat Steps 2, 3, 4, 5a/b as described above after delivering the new input stream to the CS4932X. If the new input content is detected as unchanged (still AC-3 in our example), the CS4932X responds and continues processing it as in Step 5a. This situation will happen if the new stream selected by the user is also AC-3. If the input content is detected as different (non-AC-3 in our example), the CS4932X responds as in Step 5b and continues monitoring the input stream for change in content. 7) During runtime, while successfully playing the input bitstream, the CS4932X also simultaneously monitors the input.
Note: The CS4932X has only one active input. The "input" is defined as the pin receiving data for which the application is configured (e.g. If the application is configured for PCM, the pin receiving PCM data is the input. If the application is configured for compressed data, the pin receiving compressed data is the input). As soon as the CS4932X detects a change in the bitstream (no longer AC-3, in our original example), then the CS4932X automatically reverts to Step 4., i.e., analyzes the input to determine the content. This is an automatic version of Step 6 above, but is intended to cover only the cases where the host is not aware of any possible upstream content changes. Whenever possible, the host should convey information about a possible change in input as in Step 6.
If the input content is detected as unchanged (still AC-3 in our example), the CS4932X continues processing it like in Step 5a, without requiring any further action from the host. This situation could arise due to a pause or track change upstream in the source (such as a DVD player). In the case of compressed data being played currently (like AC-3 in our example), the host could see an out-of-sync Unsolicited Message followed by an Unsolicited Message indicating MPEG data. The presence of out-of-sync messages due to "special" functions like pause or track change is completely dependent upon the audio source (such as the DVD player). In the case of a PCM application that is currently active, if the silence is less than PCM_Autodetect_Silence_Threshold (default 48000 samples, i.e. 1 Second at 48 KHz) before transitioning to new PCM, the CS4932X continues to process the input data as if no change had occurred. In case this threshold is set to 8388607 = 0x7fffff, then silence is completely ignored (threshold is effectively infinite). However, during PCM processing, if the silence is more than PCM_Autodetect_Silence_Threshold, the CS4932X jumps to a Silent Input Data state, and the output is muted (transparent due to silent input anyway). Transition to this Silent Input Data state is reported via an Unsolicited Message (0x870000 0x800020). This message is informative only, and no action should be taken by the host. CS4932X is effectively in Step 4 above now, waiting to autodetect the input once non-silent data appears. Once input data is detected, a corresponding Unsolicited Message is issued to the host and the process continues as above. Please see Figure 11, for an overview of the previously described flow.
If the input content is detected as different (non-AC-3 in our example), the CS4932X reverts to Step 5b. For compressed data streams the code will also report the Unsolicited Message 0x870000 0x800020 to indicate an out-of-sync condition when the decoder loses sync with the incoming stream due to silent input data. This is an informative message and no action is needed by the host.
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Perform Kickstart with Autodetect
Run Time (Autodetect Enabled) INTREQ goes LOW User Reads Unsolicited Response
Last Input = Current Input OR Current Input = Silence
Host Does Nothing Yes
No
Last Input = Compressed Bitstream AND Current Input = Different Compressed Bitstream
Yes
Host Loads Corresponding AC-3, DTS, MP3, MPEG-MC or AAC Code, Configures Hardware and Performs Kickstart with Autodetect Bypass
No
Last Input = PCM AND Current Input = Compressed Bitstream
Host Loads Corresponding AC-3, DTS, MP3, MPEG-MC or AAC Yes Code, Configures Hardware and Performs Kickstart Note 1
No
Last Input = Compressed Bitstream AND Current Input =PCM
Yes
Host Loads Preffered PCM Code, (e.g. - COS) Configures Hardware and Performs Kickstart Note 2
Figure 11. Generic Autodetect Flow Chart
Notes: 1. If AC-3 is detected, the host does not need to load AC-3 code if AC-3 is already loaded and passingthrough PCM. The host only needs to: (a) Perform a Soft Reset, Configure the Hardware and perform a Kickstart. (b) Or, perform an Application Restart, Configure the Inputs, and perform a Kickstart. 2. If PCM is detected, host does need to reload AC-3 code if AC-3 is already loaded, as it will pass-through PCM. The host only needs to: (a) Perform a Soft Reset, Configure the Hardware and perform a Kickstart. (b) Or, perform an Application Restart, Configure the Inputs, and perform a Kickstart.
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4.3.2 Special Considerations for Autodetection
First, autodetect should only be enabled or disabled when sending the Kickstart command. Changing the state of Autodetect at any other time can produce unpredictable results as it is a function of the input and the application that is currently enabled. If the host needs to change the state of Autodetect at run time for some reason, a soft reset (or Application Restart, if possible) should be sent. The application must be enabled before sending the autodetect enable with Kickstart. Particularly, for the AC-3 code, either AC-3 or PCM must be enabled before the autodetect message is sent with the kickstart. If the Autodetect function senses the data for the enabled application, then the appropriate Unsolicited Message is sent and decode (or PCM processing) will start automatically without any direction from the host. If the host desires complete control of the output, then it should set master volume to 0 before enabling Autodetect. Upon receiving the autodetect message that the appropriate data is present at the input, the host would then set the master volume to the desired level. Alternatively, the host could use Mute instead of Master_Volume. In summary this is the recommended procedure when using autodetect: 1) Download code (or Soft Reset or Application Restart) 2) Hardware Configuration (for download or Soft Reset) / Input Configuration (for Application Restart) 3) Application Configuration (including enable of desired application) 4) Kickstart with autodetection enabled. Please see Figure 12, for an overview of the previously described summary.
Try one/all of the follwing: See Notes 1,2,3 No
Download Application Code (AC-3, DTS, MP3, MPEG-MC, AAC, COS)
Download Successful
Yes
Configure Hardware (Includes Configuring Inputs)
Configure Application (Audio Manager, etc.)
Perform Kickstart with Autodetect
Run Time (Autodetect Enabled)
Figure 12. Typical Download Sequence
Notes: 1. Check .LD file version. Contact your FAE for the most recent version. 2. Check to make sure .LD file is correct for specific chip. (E.G. code for CS49329 will NOT load on a CS49326). 3. Check communications protocol technique. Check microcontroller code. Check powerup pin logic. Remove other devices from the communications bus (to rule-out bus contention).
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4.4 Generalized Pro Logic Manager for Dolby Digital, DTS, MPEG Multichannel and AAC
Write Opcode = 0x8E; Read Opcode = 0x0F; Read Response Opcode = 0x8F Write = 0x8E00HH 0xhhhhhh Read Request = 0x0F00HH; Read Response = 0x8F00HH 0xhhhhhh 0xHH = index, 0xhhhhhh = data value
Index
0x00
Variable
PRO_LOGIC_CONTROL
Dataword Content
Bit 8 = Autobalance_Enable = 0/1= Disable/Enable Autobalance. Bit 4 = Btype_NR_Enable = 0/1= Disable/Enable B-Type Noise Reduction. Bit 0 = ProLogic_Enable = 0/1 = Disable/Enable Pro Logic decoding. Default = 0x000000 Table 20. Generalized Pro Logic Manager
For AC-3, the code will not automatically enable Pro Logic if the stream is 2/0 Dolby Surround encoded. This task is the responsibility of the host. The end-user may choose to override the host controller's default setting when a Dolby Surround encoded 2/0 AC-3 stream is detected. This ability to disable or enable Pro Logic decoding is controlled via commands from this generalized Pro Logic Manager. If the input is NOT Dolby Surround encoded or is more or less than 2 channels, it is not recommended that Pro Logic decoding occur. For PCM input, the AC-3 module is still used, but is configured for PCM PassThrough and AC-3 decoding is disabled. Please refer to Section 4.4.2 "Pro Logic decode of 48 kHz input PCM (Using AC-3 Application Code)" on page 61 for an example of how to configure the AC-3 code for PCM Pass-Through and Pro Logic decoding.
For DTS, when the input stream is 2/0 Stereo or Dolby Surround encoded material, this module gives the additional flexibility to perform a Pro Logic decode. If the input is NOT Dolby Surround encoded or is more or less than 2 channels, it is not recommended that Pro Logic decoding occur. For MPEG, when the input stream is 2/0 Stereo or Dolby Surround encoded material, this module gives the additional flexibility to perform a Pro Logic decode. If the input is NOT Dolby Surround encoded, two independent mono channels, or is more or less than 2 channels, it is not recommended that Pro Logic decoding occur. For AAC, when the input stream is 2/0 Stereo or Dolby Surround encoded material, this module gives the additional flexibility to perform a Pro Logic decode. If the input is NOT Dolby Surround encoded, two independent mono channels, or is more or less than 2 channels, it is not recommended that Pro Logic decoding occur.
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4.4.1 Using Pro Logic in Normal, Wide and Phantom Modes 4.4.1.1 Normal Mode
Figure 13 is a example of what commands must be sent to the CS4932X, in order to enable Pro Logic in Normal mode, when decoding AC-3. This mode is especially helpful for systems that contain a center channel that is not capable of reproducing low frequency information (included in the fullbandwidth range that is directed to the center channel).
#-------BASS_MGR_CONTROL for Dolby config 3 subwoofer on-----940000 011081 #-------BASS_MGR_INPUT_LEVELS----940001 50c335 940002 50c335 940003 50c335 940004 50c335 940005 7fffff 940006 50c335 #-------BASS_MGR_SUM_OUTPUT_LEVELS----940007 4c3ea8 940008 000000 940009 4c3ea8 94000a 000000 94000b 000000 #-------BASS_MGR_LFE_OUTPUT_LEVELS----94000c 000000 94000d 000000 94000e 000000 94000f 000000 940010 000000 #-------BASS_MGR_SUM_INPUT_LEVELS----940011 000000 940012 50c335 940013 000000 940014 000000 940015 000000 940016 000000 #-------Enables LFE Channel (if present) and AC-3 Decoding-----8a0000 000011 #-------Enable Dolby Pro Logic processing, Dolby B N.R., and Autobalance-----8e0000 000111
Figure 13. Normal Mode 60 AN163REV3
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4.4.1.2 Wide Mode
When Pro Logic is enabled, without the Bass Manager enabled, the CS4932X is (by default) placed in Wide mode, when decoding AC-3. A summary of the commands enable Wide mode is found in Table 12. This mode is especially helpful for systems that contain a center channel that is capable of reproducing low frequency information (included in the full-bandwidth that is directed to the center channel).
4.4.1.3 Phantom Mode
Figure 15 is a summary of what commands must be sent to the CS4932X, in order to enable Pro Logic in Phantom mode, when decoding AC-3. This mode is especially helpful for systems do not contain a center channel.
4.4.2 Pro Logic decode of 48 kHz input PCM (Using AC-3 Application Code)
In this mode the input should be 48 kHz Pro Logic encoded PCM data. The data will be taken from the PCM input port as designated by the Hardware Configuration message.
#-------Enables LFE Channel (if present) and AC-3 Decoding------8a0000 000011 #-------Enable Dolby Pro Logic processing, Dolby B N.R., and Autobalance------8e0000 000111
Figure 14. Wide Mode
#-------Enables LFE Channel (if present) and AC-3 Decoding------8a0000 000011 #-------Enable Dolby Pro Logic processing, Dolby B N.R., and Autobalance------8e0000 000111 # Enable Phantom Mode - Output Mode set to L, R, Ls, Rs (Ls=Rs=S-3dB)------8a0001 000006
Figure 15. Phantom Mode
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Module AC-3 Manager PCM Manager Pro Logic Manager Audio Manager Audio Manager
Index AC3_CONTROL PCM_Control Pro_Logic_Control Sampling_Frequency Audio_Manager _Control
Description AC-3 Decoding Disabled PCM Pass-Through Enabled Enabled Autobalance Enabled B-Type NR Enabled 48 kHz Kickstart Application
Opcode & Index 0x8A0000 0x9c0000 0x8e0000 0x880014 0x880000
Value 0x000000 0x000001 0x000111 0x000000 0x001011
Table 21. Enabling Pro Logic decode of 48 kHz PCM
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4.5 Generalized Tone Control Manager for Dolby Digital, DTS, MPEG Multichannel and C.O.S.
Write Opcode = 0x9A; Read Opcode = 0x1B; Read Response Opcode = 0x9B Write = 0x9A00HH 0xhhhhhh Read Request = 0x1B00HH; Read Response = 0x9B00HH 0xhhhhhh 0xHH = index, 0xhhhhhh = data value
Index
0x00
Variable
EQ_TONE_CONTROL Bit 4 = Reserved.
Dataword Content
Bit 0 = Tone_Control_Enable = 0/1 = Disable/Enable Tone Control (Bass/Treble) just before Bass Manager on L, C, and R.
0x01
Default = 0x000000 PRE_TONE_ATTENUATION -1.0 to 1.0 = Pre-Tone Control attenuation scale factor to setup desired headroom for Tone control. Default* = 0.03125 (0x040000) L_PASS_TROUGH_LEVEL -4.0 to 4.0 in 6.18 format = L Channel pass through level setting.1 Default* = 0.0 -4.0 to 4.0 in 6.18 format = L Channel bass LPF level setting.1 Default* = 0.0 -4.0 to 4.0 in 6.18 format = L Channel bass HPF level setting.1 Default* = 0.0 -4.0 to 4.0 in 6.18 format = L Channel treble HPF level setting.1 Default* = 0.0 -4.0 to 4.0 in 6.18 format = L Channel treble LPF level setting.1 Default* = 0.0 -4.0 to 4.0 in 6.18 format = C Channel pass through level setting.1 Default* = 0.0 -4.0 to 4.0 in 6.18 format = C Channel bass LPF level setting.1 Default* = 0.0 -4.0 to 4.0 in 6.18 format = C Channel bass HPF level setting.1 Default* = 0.0 -4.0 to 4.0 in 6.18 format = C Channel treble HPF level setting.1 Default* = 0.0 -4.0 to 4.0 in 6.18 format = C Channel treble LPF level setting.1 Default* = 0.0 -4.0 to 4.0 in 6.18 format = R Channel pass through level setting.1 Default* = 0.0 Table 22. Tone Control
0x02
0x03
L_BASS_LPF_LEVEL
0x04
L_BASS_HPF_LEVEL
0x05
L_TREBLE_HPF_LEVEL
0x06
L_TREBLE_LPF_LEVEL
0x07
C_PASS_TROUGH_LEVEL
0x08
C_BASS_LPF_LEVEL
0x09
C_BASS_HPF_LEVEL
0x0a
C_TREBLE_HPF_LEVEL
0x0b
C_TREBLE_LPF_LEVEL
0x0c
R_PASS_TROUGH_LEVEL
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Index
0x0d
Variable
R_BASS_LPF_LEVEL Default* = 0.0
Dataword Content
-4.0 to 4.0 in 6.18 format = R Channel bass LPF level setting.1 -4.0 to 4.0 in 6.18 format = R Channel bass HPF level setting.1 Default* = 0.0 -4.0 to 4.0 in 6.18 format = R Channel treble HPF level setting.1 Default* = 0.0 -4.0 to 4.0 in 6.18 format = R Channel treble LPF level setting.1 Default* = 0.0 -1.0 to 1.0 in 1.23 format = I order b0 filter coefficient for LPF (default corner=90 Hz at Fs=48 KHz). Default* = 0x00bfe4 Note: b1 = b0 assumed for LPF. -1.0 to 1.0 in 1.23 format = I order a1 filter coefficient for LPF (default corner=90 Hz at Fs=48 KHz). Default* = 0x7e8037 -1.0 to 1.0 in 1.23 format = I order b0 filter coefficient for HPF (default corner=350 Hz at Fs=48 KHz). Default* = 0x7d220d Note: b1 = - b0 assumed for HPF. -1.0 to 1.0 in 1.23 format = I order a1 filter coefficient for HPF (default corner=350 Hz at Fs=48 KHz). Default* = 0x7a441a -1.0 to 1.0 in 1.23 format = I order b0 filter coefficient for HPF (default corner=7500 Hz at Fs=48 KHz). Default* = 0x536a08 Note: b1 = - b0 assumed for HPF. -1.0 to 1.0 in 1.23 format = I order a1 filter coefficient for HPF (default corner=7500 Hz at Fs=48 KHz). Default* = 0x26d410 -1.0 to 1.0 in 1.23 format = I order b0 filter coefficient for LPF (default corner=1900 Hz at Fs=48 KHz). Default* = 0x0e38e2 Note: b1 = b0 assumed for LPF. -1.0 to 1.0 in 1.23 format = I order a1 filter coefficient for LPF (default corner=1900 Hz at Fs=48 KHz). Default* = 0x638e3c Table 22. Tone Control (Continued)
0x0e
R_BASS_HPF_LEVEL
0x0f
R_TREBLE_HPF_LEVEL
0x10
R_TREBLE_LPF_LEVEL
0x1b
BASS_LPF_B0
0x1c
BASS_LPF_A1
0x1d
BASS_HPF_B0
0x1e
BASS_HPF_A1
0x1f
TREBLE_HPF_B0
0x20
TREBLE_HPF_A1
0x21
TREBLE_LPF_B0
0x22
TREBLE_LPF_A1
1Please
see Section 4.6.1 "Level Settings for +12 dB to -12 dB Tone Control" on page 65 on details of how to set up this register
Notes: "Default" vs. "Default*" See definition page 32.
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The tone control application code enhancement allows the host to send commands to the DSP that allows a parametric boost of up to +12 dB to treble and/or bass frequencies in addition to a parametric cut of up to -12 dB to treble and/or bass frequencies on the front three channels (L, C, R). The only application codes covered by AN163 that have Tone Control ability are: ac3_32x8.ld, dts_32x8.ld, mpg_32x8.ld, and eff_32x7.ld (or later releases). Other application codes not covered specifically in AN163, such as SRS Circle Surround (cs__3xx3.ld), also support tone control. Please refer to the appropriate AN163 appendix for more information.
Note: For system certification purposes (such as Dolby Laboratories), tone control should be disabled during certification testing.
For each channel (L, C, R), the tone control module accepts 5 independent parameters Pass_Through_Level, Bass_LPF_Level, Bass_HPF_Level, Treble_LPF_Level, and Treble_HPF_Level to adjust the amount of cut/boost in the low and high frequency bands for that channel. The following section describes the procedure by which appropriate values for these parameters can be arrived at for a desired tone control setting.
4.6.1 Level Settings for +12 dB to -12 dB Tone Control
The desired bass and treble values are mapped to five values which are plugged into a particular channel's X_Pass_Through_Level, X_Bass_LPF_Level, X_Bass_HPF_Level, X_Treble_HPF_Leve,l and X_Treble_LPF_Level registers, where X denotes L, C, R, Ls, or Rs. The procedure for mapping the desired bass and treble values is given below. These variables can be changed on the fly without audible artifacts, if the step is smaller than 1 dB. Obviously, a one-step change from -12 dB to +12 dB would result in an audible artifact. Please note that the procedure assumes integer values for the tone levels. Also, the abbreviation LUT is used to denote a "Look-Up Table". Table 23, Table 24 and Table 25 are the three Look-Up Tables used in the mapping scheme and are labeled LUT0, LUT1, and LUT2. The mapping is sensitive to the "polarity" of the tone levels. The four possible cases are handled differently as follows: * Bass Level > = 0 and Treble > = 0:
X_Pass_Through_Level = 0x03FFFF; X_Bass_LPF_Level = LUT1(Bass_Level); X_Bass_HPF_Level = 0; X_Treble_HPF_Level = LUT1(Treble_Level); X_Treble_LPF_Level = 0; Bass Level > = 0 and Treble < 0: X_Pass_Through_Level = LUT0( abs (Treble_ Level) ); X_Bass_LPF_Level = LUT1( (Bass_Level) );
65
4.6 Controlling the Level for Treble and Bass Boost/Cut
When tone control is enabled, the pre-Bass Manager channel PCM is attenuated by the Pre_Tone_Attenuation scale factor and passed through tone control shelving filters for bass and treble.
Note: Pre_Tone_Attenuation affects only channels which have tone control (L, C, and R) and leaves the other channels (Ls, Rs, LFE) untouched at the full level. Thus, in order to equalize all the levels coming out of the CS4932X, the controller should set the volumes of the other channels (Ls, Rs, LFE) to be the same as Pre_Tone_Attenuation. Pre_Tone_Attenuation and the above matching volume setting will result in reduced level which has to be compensated by analog gain after the digital-to-analog conversion in the analog chain The controller has the knowledge of this attenuation setting at all times and can thus adjust the post-DAC analog gain accordingly. However, this increased digital headroom will result in smaller dynamic range numbers and thus it is recommended that a passthrough (no Tone Control) mode be implemented for testing purposes.
*
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X_Bass_HPF_Level = 0; X_Treble_HPF_Level = 0; X_Treble_LPF_Level = LUT2(abs (Treble_ Level) ); Bass Level < 0 and Treble > = 0: X_Pass_Through_Level = LUT0(abs (Bass_ Level) ); X_Bass_LPF_Level = 0; X_Bass_HPF_Level = LUT2(abs (Bass_ Level) ); X_Treble_HPF_Level = LUT1(Treble_Level); X_Treble_LPF_Level = 0; Bass Level < 0 and Treble < 0: X_Pass_Through_Level = LUT0(abs (Bass_ Level) ) - LUT2(abs (Treble_Level) ) + CF; X_Bass_LPF_Level = -CF; X_Bass_HPF_Level = LUT2(abs (Bass_ Level) ); X_Treble_HPF_Level = -CF; X_Treble_LPF_Level = LUT2(abs (Treble_ Level) );
*
4.6.2 Tone Control Look-Up Tables (LUT0, LUT1, LUT2)
The LUT0, LUT1, and LUT2 look-up tables have the following characteristics: The bass and treble setting values range from -12 to +12 dB. In order to accommodate this range in 24-bit signed, fixedpoint binary format, the Hex values are all represented in 6.18 format. The tables can be stored in the microcontroller and used with the algorithm described in the previous section to generate the various filter settings for a given user setting.
*
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dB
0 1 2 3 4 5 6 7 8 9 10 11 12
Real Value
1 0.891240713 0.794310008 0.707921418 0.630928389 0.562309067 0.501152734 0.44664772 0.398070632 0.354776754 0.316191487 0.281802726 0.251154063
Hex Value in 6.18
40000 390A1 32D60 2D4E9 28612 23FCE 2012E 1C95E 197A0 16B4B 143C8 12091 1012F
Table 23. Look-Up Table 0; "LUT0"
dB
0 1 2 3 4 5 6 7 8 9 10 11 12 0
Real Value
0
Hex Value in 6.18
7CF6 1092B 1A67D 25701 31D10 3FB4A 4F4A2 60C68 74652 8A68B A31BF BED2E
0.122031327 0.258954299 0.412586164 0.584965928 0.778381424 0.99539967 1.23890094 1.512116993 1.818673965 2.16264049 2.548581706 2.981619842
Table 24. Look-Up Table 1; "LUT1"
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dB
0 1 2 3 4 5 6 7 8 9 10 11 12 0
Real Value
0
Hex Value in 6.18
6F5F D2A0 12B17 179EE 1C032 1FED2 236A2 26860 294B5 2BC38 2DF6F 2FED1
0.108759287 0.205689992 0.292078582 0.369071611 0.437690933 0.498847266 0.55335228 0.601929368 0.645223246 0.683808513 0.718197274 0.748845937
Table 25. Look-Up Table 2; "LUT2"
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4.7 Controlling the Corner Frequencies of the
LPF/HPF for Bass and Treble Control
The tone control module implements shelving filters using two first order LPFs and two first order HPFs. The LPFs default corner frequencies are 90 Hz and 1900 Hz at Fs = 48 KHz. The HPFs default frequencies are 350 Hz and 7500 Hz at Fs = 48 KHz. Different filter coefficients can be downloaded (before Kickstart only) for any desired corner frequencies or to match any change in Fs.
30
Only b0 and a1 are required to be downloaded since it is assumed that b1 = b0 for the LPF, and b1 = -b0 for the HPF. These values can be obtained using any standard filter design procedure for first order LPF and HPF with unity gain in the passband. The coefficients are all expected to be downloaded in 1.23 format before Kickstart.
Note: LPF/HPF coefficients should be set up before Kickstart and should not be modified on the fly during runtime.
Frequency Response of Tone Control
20
10
Output level (dB)
0
-10
-20
-30 10
2
10 Frequency (Hz)
3
10
4
Figure 16. Composite Graph of a Frequency Sweep with Bass and Treble Note: Boosted by +4 dB, +8 db, and +12 dB along with Bass and Treble Cut by -4 dB, -8 dB, and -12 dB
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Frequency Response of Tone Control 30
20
10 Output level (dB)
0
-10
-20
-30 10
2
10 Frequency (Hz)
3
10
4
Figure 17. Composite Graph of a Frequency Sweep with Bass Note: Boosted by +4 dB, +8 db, and +12 dB along with Bass Cut by -4 dB, -8 dB, and -12 dB
Frequency Response of Tone Control 30
20
10 Output level (dB)
0
-10
-20
-30 10
2
10 Frequency (Hz)
3
10
4
Figure 18. Composite Graph of a Frequency Sweep with Treble Note: Boosted by +4 dB, +8 db, and +12 dB along with Treble Cut by -4 dB, -8 dB, and -12 dB
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5. AC-3 SPECIFIC MODULES AND OPERATIONS
The following two modules are specific to AC-3 code. They are the AC-3 Manager and PCM Manager. In addition to AC-3 specific modules, certain operations such as enabling Autodetection of an AC-3 stream and enabling Autodetection of a Pro Logic encoded AC-3 stream is also included in this section. Please refer to Section 4.2 "Generalized Bass Manager for Dolby Digital, DTS, MP3, MPEG Multichannel, AAC, and Crystal Original Surround" on page 40 section for the listing of the Dolby Bass Manager configurations. AC-3 application code (ac3_32xx.ld) supports the following types of data: AC-3 (Pc = 1). Pass-Through of PCM is also possible using this code.
5.1 AC-3 Manager
Write Opcode = 0x8A; Read Opcode = 0x0B; Read Response Opcode = 0x8B Write = 0x8A00HH 0xhhhhhh Read Request = 0x0B00HH; Read Response = 0x8B00HH 0xhhhhhh 0xHH = index, 0xhhhhhh = data value
Index
0x00
Variable
AC3_CONTROL
Dataword Content
Bit 4 = LFE_Enable = 0/1 = Disable/Enable LFE channel decoding (if it exists in the stream). Bit 0 = AC3_Enable = 0/1 = Disable/Enable AC-3 decoding. Default = 0x000000 Bit 8 = Karaoke_Capable_Enable = 0/1 = Enable/Disable Karaoke capable downmixing (see Section 5.2 "PCM Manager (for AC-3 Application Code)" on page 78 for Karaoke Capable user settings). Bits 5:4 = Dualmode = 0...3 = -u setting of Dolby C decoder. Valid only if acmod=0 (dual mono). 00b = Stereo. 01b = Left Mono. 10b = Right Mono. 11b = Mixed-Mono. Bits 3:0 = Output_Mode = 0...7 = -o setting of Dolby C decoder. 0x0 = 2/0 Lt, Rt Dolby Surround compatible. (Equation in Section 5.1.2 "Downmix Equation for Lt, Rt" on page 75). 0x1 = 1/0 C. 0x2 = 2/0 L, R. 0x3 = 3/0 L, C, R. 0x4 = 2/1 L, R, S (Ls = Rs = S - 3 dB). 0x5 = 3/1 L, C, R, S (Ls = Rs = S - 3 dB). 0x6 = 2/2 L, R, Ls, Rs. 0x7 = 3/2 L, C, R, Ls, Rs. Default* = 0x000007
0x01
OUTPUT_MODE_CONTROL
Note: Undesired Channel outputs must be explicitly muted by setting the corresponding Ch_Vol (see Section 4.1 "Audio Manager for Dolby Digital, DTS, MP3, MPEG Multichannel, AAC and Crystal Original Surround" on page 33) to 0.0 Table 26. AC-3 Manager AN163REV3 71
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Index
0x02
Variable
COMPRESSION_CONTROL
Dataword Content
Bits 1:0 = 0...3 = -k setting of Dolby C decoder. 00b = Custom Mode 0 (Analog Dialnorm). 01b = Custom Mode 1 (Digital Dialnorm). 10b = Line Out Mode (This mode is REQUIRED for all AVR systems). 11b = RF Remodulation Mode. Default* = 1 0x000000-0x7FFFFF = Compression cut scale factor = -x setting of Dolby C decoder. Cut_x is a scaling factor used on the absolute compression limits. The absolute compression limits are set during the encoding process. Default* = 0 0x000000-0x7FFFFF = Compression boost scale factor = -y setting of Dolby C decoder. Boost_y is a scaling factor used on the absolute compression limits. The absolute compression limits are set during the encoding process. Default* = 0 Bits 1:0 = Current Sample Rate Code (fscod) value. 00b = 48kHz. 01b = 44.1kHz. 10b = 32kHz. 11b = Reserved. READ-ONLY Bits 3:0 = Current Bit Stream (bsmod) value. 0x0 = Main audio service: complete main (CM). 0x1 = Main audio service: music and effects (ME). 0x2 = Associated service: visually impaired (VI). 0x3 = Associated service: hearing impaired (HI). 0x4 = Associated service: dialogue (D). 0x5 = Associated service: commentary (C). 0x6 = Associated service: emergency (E). 0x7: if ACMOD == 1: Associated service: voice over (VO). if ACMOD == 010b to 111b: main audio service: karaoke. READ-ONLY
0x03
CUT_X
0x04
BOOST_Y
0x05
FSCOD
0x06
BSMOD
Table 26. AC-3 Manager (Continued)
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Index
0x07 ACMOD
Variable
Dataword Content
Bits 3:0 = Current Audio Coding Mode (acmod) value 0x0 = 1+1 Ch1, Ch2. 0x1 = 1/0 C. 0x2 = 2/0 L, R. 0x3 = 3/0 L, C, R. 0x4 = 2/1 L, R, S. 0x5 = 3/1 L, C, R, S. 0x6 = 2/2 L, R, Ls, Rs. 0x7 = 3/2 L, C, R, Ls, Rs. Note: Changes in this stream variable will result in an Unsolicited Message from the DSP if Audio_Config_Change_Notification has been enabled in the Audio Manager. Please refer to Section 4.1.1 "Audio_Control: (Index 0x00)" on page 37 for more information.
0x08
DSURMOD
READ-ONLY Bits 1:0 = Current Dolby Surround Mode (dsurmod value) valid only when (acmod=2). 00b = Not indicated. 01b = NOT Dolby Surround encoded. 10b = Dolby Surround Encoded. 11b = Reserved. Note: Changes in this stream variable will result in an Unsolicited Message from the DSP if Audio_Config_Change_Notification has been enabled in the Audio Manager. Please refer to Section 4.1.1 "Audio_Control: (Index 0x00)" on page 37 for more information.
0x09
CLEV
READ-ONLY 0x000000-0x7FFFFF = Current center mixing level (interpreted stream value as 1.23 fractional integer). Valid only if 3 front channels exist. READ-ONLY 0x000000-0x7FFFFF = Current surround mixing level (interpreted stream value as 1.23 fractional integer). Valid only if a surround channel exists. READ-ONLY Bit 0 = Current lfeon value. 0 = LFE channel is off. 1 = LFE channel is on. Notes: Changes in this stream variable will result in an Unsolicited Message from the DSP if Audio_Config_Change_Notification has been enabled in the Audio Manager. Please refer to Section 4.1.1 "Audio_Control: (Index 0x00)" on page 37 for more information. READ-ONLY Table 26. AC-3 Manager (Continued)
0x0a
SLEV
0x0b
LFEON
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Index
0x0c
Variable
DIALNORM
Dataword Content
0...31 = Current dialnorm value. Note: Changes in this stream variable will result in an Unsolicited Message from the DSP if Audio_Config_Change_Notification has been enabled in the Audio Manager. Please refer to Section 4.1.1 "Audio_Control: (Index 0x00)" on page 37 for more information.
0x0d 0x0e
COMPR DIALNORM2
READ-ONLY 0xFFFFFF = Nonexistent in stream. 0x00-0xFF = Current compr value. READ-ONLY 0...31 = Current dialnorm2 value. Valid only in 1+1 dual mono mode (acmod=0). READ-ONLY 0xFFFFFF = Nonexistent in stream. 0x00-0xFF = Current compr2 value. Valid only in 1+1 dual mono mode (acmod=0). READ-ONLY 0...31 = Current bsid value. READ-ONLY 0xFFFFFF = Nonexistent in stream. 0...255 = Current langcod value. READ-ONLY 0xFFFFFF = Nonexistent in stream. 0x00...0x7F: Bits 6:2 = Current mixlevel. Bits 1:0 = Current roomtyp. READ-ONLY 0xFFFFFF = Nonexistent in stream. 0...255 = Current langcod value. Valid only in 1+1 dual mono mode (acmod=0). READ-ONLY 0xFFFFFF = Nonexistent in stream. 0x00...0x7F: Bits 6:2 = Current mixlevel2. Bits 1:0 = Current roomtyp2. Valid only in 1+1 dual mono mode (acmod=0). READ-ONLY 0/1 = Current copyrightb value. READ-ONLY 0/1 = Current origbs value. READ-ONLY Current frame size code. READ-ONLY Current time code. READ-ONLY Table 26. AC-3 Manager (Continued)
0x0f
COMPR2
0x10 0x11 0x12
BSID LANGCOD MIXLEVEL_ROOMTYP
0x13
LANGCOD2
0x14
MIXLEVEL2_ROOMTYP2
0x15 0x16 0x17 0x18
COPYRIGHTB ORIGBS FRMSIZCOD TIMECOD
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Index
0x19
Variable
TIMECOD2
Dataword Content
Current time code 2. This variable can also indicate that the AC-3 stream is Dolby Digital Surround-EX encoded. The host must also take into account the BSID variable setting in order to confirm how a change in TIMECOD2 should be interpreted. Please refer to the AC-3 Specification for more details. Note: Changes in this stream variable will result in an Unsolicited Message from the DSP if Audio_Config_Change_Notification has been enabled in the Audio Manager. Please refer to Section 4.1.1 "Audio_Control: (Index 0x00)" on page 37 for more information. READ-ONLY Table 26. AC-3 Manager (Continued)
Notes:
See definition on page 32.
"Default" vs. "Default*" See definition page 32.
5.1.1 OUTPUT_MODE_CONTROL
Sets up particular output speaker downmix configuration. When configuring the Output _Mode_Control, first the individual volumes of each undesired channel should be set to 0. Next, the desired output mode should be specified. This should typically be done before kickstart of the application. For example, if output mode 3/0 (L, C, R) is chosen, Ls and Rs volume must be explicitly set to 0 in the Audio Manager module to guarantee no output on those channels. Bit 8: KARAOKE_CAPABLE_ENABLE - If set, downmixing of Karaoke streams (bsmod=7, acmod = 2...7) will be performed according to Karaoke Capable decoder specification by employing user settings. See Section 5.2 "PCM Manager (for AC-3 Application Code)" on page 78 for more details of the user settings. If Karaoke_Capable_Enable is not set, then the CS4932X defaults to a Karaoke Aware downmix using levels indicated in the stream. Bits 5:4: DUALMODE = 0...3 = -u setting of Dolby C decoder. Valid only if ACMOD = 0 (dual mono). 00b = Stereo. 01b = Left Mono. 10b = Right Mono. 11b = Mixed-Mono.
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Bits 3:0: OUTPUT_MODE = 0...7 = -o setting of Dolby C decoder 0x0 = 2/0 L, R Dolby Surround compatible (a.k.a. Lt, Rt). 0x1 = 1/0 C. 0x2 = 2/0 L, R. 0x3 = 3/0 L, C, R. 0x4 = 2/1 L, R, S (Ls = Rs = S - 3 dB). 0x5 = 3/1 L, C, R, S (Ls = Rs = S - 3 dB). 0x6 = 2/2 L, R, Ls, Rs. 0x7 = 3/2 L, C, R, Ls, Rs.
5.1.2 Downmix Equation for Lt, Rt
Lt = (((1/3.1)L) + ((0.7/3.1)C) - ((0.7/3.1)Ls) ((0.7/3.1)Rs) Rt = (((1/3.1)L) + ((0.7/3.1)C) + ((0.7/3.1)Ls) + ((0.7/3.1)Rs)
5.1.3 AC-3 Stream Information
Table 26 lists a complete set of READ-ONLY variables that are reported to the host by the AC-3 Decoder module. These values can be read out using the appropriate Read Request, Read Response sessions. For example, to read the ACMOD value, host should issue the 3-byte Read Request 0x0b0007 and then read the 6-byte Read Response 0x9b0007, 0x00000h, where the dataword (latter 3 bytes)
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indicates the reported value of ACMOD. Please refer to the Dolby AC-3 Specification for more information on these stream variables. DTS, MP3, MPEG Multichannel, AAC and Crystal Original Surround" on page 33.
5.1.5 AC-3 decode with Autodetect
In this mode, the input should be compressed AC3. The data will be taken from the compressed input port as designated by the Hardware Configuration message. If the input on this port is not AC-3, the Autodetect function will notify the host as described in Section 4.3.1 "Autodetect Operation" on page 55. Figure 19 is an example of pseudocode to configure the part for AC-3 decoding with autodetection. This pseudocode can be used as a template for other configurations by swapping out the message array and the size portions of the pseudocode.
5.1.4 Special Considerations For AC-3
When using the AC-3 code, note that AC-3 and PCM are completely independent applications, and both are embedded inside the same code download. The host must treat them as such and only enable either AC-3 or PCM at any one time. Transitions from AC-3 to PCM, or vice-versa, should be done with a Soft Reset followed by complete Hardware Configuration and Application Configuration. Conversely, an Application Restart message could be used that would only require that a new input port configuration and application configuration be sent -- output clocks would not be affected. For more information on Application Restart please see Section 4.1 "Audio Manager for Dolby Digital,
Module AC-3 AC-3 Audio Manager Index AC3_CONTROL
Description AC-3 Enabled LFE Enabled Autodetect Enabled PLL Enabled Kickstart Application
Opcode & Index 0x8A0000 0x8A0002 0x880000
Value 0x000011 0x000002 0x001011
COMPRESSION_CONTROL Set to Line Out Mode AUDIO_MANAGER _CONTROL
Table 27. Enabling AC-3 Decode with Autodetect
void AC-3_AD_Config() { unsigned char AC-3_AD_config_message[] = {0x8A, 0x00, 0x00, 0x00, 0x00, 0x11, 0x8A, 0x00, 0x02 0x00, 0x00, 0x02 0x88, 0x00, 0x00, 0x00, 0x10, 0x11}; Write_*(AC-3_AD_config_message,12); /* Replace * with I2C or SPI depending on protocol */ }
Figure 19. AC-3 Decode with Autodetect
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5.1.6 AC-3 with Pro Logic Decode and Autodetect
In this mode, the input should be two-channel compressed AC-3 (acmod = 2) with Pro Logic encoded data (dsurmod = 2). The data will be taken
Module Pro Logic AC-3 AC-3 Audio Manager Index ProLogic Control AC-3 Control Compression_Control Audio Manager Control
from the compressed input port as designated by the Hardware Configuration message. If the input on this port is not AC-3 the Autodetect function will notify the host as described in Section 4.3.1 "Autodetect Operation" on page 55.
Opcode & Index 0x8e0000 0x8A0000 0x8A0002 0x880000
Description Enabled Autobalance Enabled B Type NR Enabled AC-3 Enabled LFE Enabled Set to Line Out Mode Autodetect Enabled PLL Enabled Kickstart Application
Value 0x000111 0x000011 0x000002 0x001011
Table 28. Enabling AC-3 with Pro Logic Decode with Autodetect
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5.2 PCM Manager (for AC-3 Application Code)
Write Opcode = 0x9C; Read Opcode = 0x1D; Read Response Opcode = 0x9D Write = 0x9C00HH 0xhhhhhh Read Request = 0x1D00HH; Read Response = 0x9D00HH 0xhhhhhh 0xHH = index, 0xhhhhhh = data value
Index
0x00
Variable
PCM_CONTROL
Dataword Content
Bit 4 = Deemphasis_Enable = 0/1 = Disable/Enable PCM Deemphasis. Note: Please refer to the deemphasis section, located after this table, for more information about using digital deemphasis for PCM sampling frequencies other than 44.1 kHz when using IBAbased codes. Bit 0 = PCM_Enable = 0/1 = Disable/Enable PCM. This bit should only be enabled if AC-3 is disabled.
0x01
K_CPL_MLEV_1_F
Default = 0x000000 M level for Karaoke capable 1 front output channel. 0 <= K_Cpl_mlev_1_F <= 0x7FFFFF. *Default = 0X1E2A53 V1 level for Karaoke capable 1 front output channel. 0 <= K_Cpl_v1lev_1_F <= 0x7FFFFF. *Default = 0X1E2A53 V2 level for Karaoke capable 1 front output channel. 0 <= K_Cpl_v2lev_1_F <= 0x7FFFFF. *Default = 0X1E2A53 m L level for Karaoke capable 2 front output channels. 0 <= K_Cpl_mllev_2_F <= 0x7FFFFF. *Default = 0X34B4B3 V1 L level for Karaoke capable 2 front output channels. 0 <= K_Cpl_v1llev_2_F <= 0x7FFFFF. *Default = 0X4B4B4A V2 L level for Karaoke capable 2 front output channels. 0 <= K_Cpl_v2llev_2_F <= 0x7FFFFF. *Default = 0x0 M R level for Karaoke capable 2 front output channels. 0 <= K_Cpl_mrlev_2_F <= 0x7FFFFF. *Default = 0X34B4B3 Table 29. PCM Processor
0x02
K_CPL_V1LEV_1_F
0x03
K_CPL_V2LEV_1_F
0x04
K_CPL_MLLEV_2_F
0x05
K_CPL_V1LLEV_2_F
0x06
K_CPL_V2LLEV_2_F
0x07
K_CPL_MRLEV_2_F
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Index
0x08
Variable
K_CPL_V1RLEV_2_F
Dataword Content
V1 R level for Karaoke capable 2 front output channels. 0 <= K_Cpl_v1rlev_2_F <= 0x7FFFFF. *Default = 0x0 V2 R level for Karaoke capable 2 front output channels. 0 <= K_Cpl_v2rlev_2_F <= 0x7FFFFF. *Default = 0X4B4B4A M C level for Karaoke capable 3 front output channels. 0 <= K_Cpl_mcpan_3_F <= 0x7FFFFF. *Default = 0x7FFFFF. V1 C level for Karaoke capable 3 front output channels. 0 <= K_Cpl_v1cpan_3_F <= 0x7FFFFF. *Default = 0x0. V2 C level for Karaoke capable 3 front output channels. 0 <= K_Cpl_v2cpan_3_F <= 0x7FFFFF. *Default = 0x0. M L level for Karaoke capable 3 front output channels. 0 <= K_Cpl_mlpan_3_F <= 0x7FFFFF. *Default = 0x0. V1 L level for Karaoke capable 3 front output channels. 0 <= K_Cpl_v1lpan_3_F <= 0x7FFFFF. *Default = 0x7FFFFF. V2 L level for Karaoke capable 3 front output channels. 0 <= K_Cpl_v2lpan_3_F <= 0x7FFFFF. *Default = 0x0. M R level for Karaoke capable 3 front output channels. 0 <= K_Cpl_mrpan_3_F <= 0x7FFFFF. *Default = 0x0. V1 R level for Karaoke capable 3 front output channels. 0 <= K_Cpl_v1rpan_3_F <= 0x7FFFFF. *Default = 0x0. V2 R level for Karaoke capable 3 front output channels. 0 <= K_Cpl_v2rpan_3_F <= 0x7FFFFF. *Default = 0x7FFFFF. Table 29. PCM Processor
0x09
K_CPL_V2RLEV_2_F
0x0a
K_CPL_MCPAN_3_F
0x0b
K_CPL_V1CPAN_3_F
0x0c
K_CPL_V2CPAN_3_F
0x0d
K_CPL_MLPAN_3_F
0x0e
K_CPL_V1LPAN_3_F
0x0f
K_CPL_V2LPAN_3_F
0x10
K_CPL_MRPAN_3_F
0x11
K_CPL_V1RPAN_3_F
0x12
K_CPL_V2RPAN_3_F
Notes:
See definition on page 32.
"Default" vs. "Default*" See definition page 32.
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5.2.1 PCM_CONTROL
Bit 4: DEEMPHASIS_ENABLE Disable/Enable PCM Deemphasis. The filter used is identical to CS4334 (50/15 uS). 1 - Enables Deemphasis 0 - Disables Deemphasis For systems that plan on using a "Non-IBA-Based" PCM Processing application code such as ac3_3265.ld (that has been configured for PCM Pass-Through) and want to support digital deemphasis for the sampling frequencies of 32 kHz and 48 kHz, as well as 44.1 kHz, PCM, the host only needs to inform the DSP the setting for the current PCM sampling frequency, which is set via the Audio Manager (0x14) before Kickstart.
Note: For systems that plan on using an "IBA-Based" PCM Processing application code such as ac3i326x.ld (that has been configured for PCM Pass-Through) and want to support digital deemphasis for the sampling frequencies of 32 kHz and 48 kHz, as well as 44.1 kHz, PCM, the host, in addition to communicating the setting for the current PCM sampling frequency, which is set via the Audio Manager (0x14), must also download the following commands before Kickstart: For 32 kHz support: 848ff9 bb82fa 848ffe 42aef3 848fff f8d406 For 44.1 kHz support: 848ff8 aeacda 848ffc 3b6543 848ffd f34797 For 48 kHz support: 848ff7 ab9ed7 848ffa 39be36 848ffb f1e0a1 For 64 kHz support: 80 848ff9 a25a7b 848ffe 34f310 848fff ed676b For 88.2 kHz support: 848ff8 99f80b 848ffc 30eb8a 848ffd e90c82 For 96 kHz support: 848ff7 981257 848ffa 300d08 848ffb e8054f
Digital de-emphasis support does not exist for sampling frequencies other than 96 kHz, 88.2 kHz, 64 kHz, 48 kHz, 44.1 kHz, or 32 kHz regardless of what code is being downloaded. Bit 1: PCM_ENABLE Disable/Enable PCM. This bit should only be enabled if AC-3 is disabled. 1 - Enables PCM Pass-Through 0 - Disables PCM Pass-Through
5.2.2 Karaoke Capable Function:
Karaoke Capable functionality is enabled when Bit 8 of Output_Mode_Control in AC-3_Manager is set to 1. If Karaoke_Capable_Enable is not set, then CS4932X defaults to a Karaoke Aware downmix, using levels indicated in the stream. Karaoke Capable functionality is enabled when Bit 8 of Output_Mode_Control in AC-3_Manager is set to 1. In a karaoke bit stream (bsmod=7, acmod=2...7), 3 special channels have to be downmixed according to user specification. They are M (guide melody), and V1, V2 (vocal tracks one and two). In Karaoke Aware mode (default mode where Karaoke_Capable_Enable=0), the CS4932X
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downmixes these channels according to the clev/slev values indicated in the bitstream. This Karaoke Aware downmix reproduces all the program content in the front speakers according to the programmed levels and audio configuration of the bitstream. In Karaoke Capable mode, the downmix coefficients are set up by the user according to the number of output front channels: 1, 2, or 3. Independent settings are available for each of these 3 cases as indicated in Table 29 on page 78. The following equations define the downmixing performed for each of the 3 cases: * 1 front channel: Ck = mlev * M + v1lev * V1 + v2lev * V2. Co = 0.707 * L + Ck + 0.707 * R. 2 front channels: Lk = mllev * M + v1llev * V1 +v2llev * V2. Rk = mrlev * M + v1rlev * V1 + v2rlev * V2. Lo = L + Lk Ro = R + Rk 3 front channels: Ck = mcpan * M + v1cpan * V1 + v2cpan * V2. Lk = mlpan * M + v1lpan * V1 + v2lpan * V2. Rk = mrpan * M + v1rpan * V1 + v2rpan * V2. Lo = L + Lk Co = Ck Ro = R + Rk applications. The host must treat them as such and only enable either AC-3 or PCM at any one time. Transitions from AC-3 to PCM or vice-versa should be done with a soft reset followed by complete Hardware Configuration and Application Configuration. Conversely, an Application Restart message could be used, which would only require that a new input port configuration and application configuration be sent -- output clocks would not be affected.
5.2.4 Special Considerations For Pink/White Noise
Pink/White Noise is controlled by three parameters. These parameters are the currently configured application (i.e. PCM), the channel selected by the Noise_Control variable:, and the state of the Autodetect Enable. In order to use Pink/White Noise, the code should be configured strictly for PCM before Kickstart. If Pro Logic is enabled, the host loses independent control of Pink/White Noise on the Ls and Rs channels -- the surround channels operate in unison. White noise can be generated by additionally setting Bit 4 of Noise_Control in the messages above. In the same message, if Bit 5 is set, a 10.5 dB boosted noise is received.
*
*
5.2.5 PCM Pass-through (Using AC-3 Application Code)
In this mode, the input should be 44.1 kHz PCM data. The data will be taken from the PCM input port, as designated by the Hardware Configuration message. (See Table 30.)
5.2.3 Special Considerations For PCM
When using the AC-3 code, it should be noted that AC-3 and PCM are completely independent
Module PCM Audio Manager Audio Manager
Index PCM Control Sampling Frequency Audio Manager Control
Description Enabled 44.1 kHz Kickstart Application
Opcode and Index 0x9c0000 0x880014 0x880000
Value 0x000001 0x000001 0x001011
Table 30. Enabling PCM Pass-through of 44.1 kHz PCM AN163REV3 81
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5.2.6 Pink/White Noise Generation
Module PCM Audio Manager Audio Manager Index PCM Control Noise Control Audio Manager Control Enabled Pink Noise On Left Channel Kickstart Application Autodetect Disabled Description Opcode and Index 0x9c0000 0x880001 0x880000 Value 0x000001 0x000000 0x000011
Table 31. Enabling Pink/White Noise Generation
To cycle through the channels (after sending the above messages), the host would send the following sequence of commands, pausing
Module Audio Manager Audio Manager Audio Manager Audio Manager Audio Manager Audio Manager Index Noise Control Noise Control Noise Control Noise Control Noise Control Noise Control
between each command to hold the pink noise on a particular channel (See Table 32).
Description Pink Noise On Left Channel Pink Noise On Center Channel Pink Noise On Right Channel Pink Noise On Left Surround Channel Pink Noise On Right Surround Channel Pink Noise On LFE Channel
Opcode and Index 0x880001 0x880001 0x880001 0x880001 0x880001 0x880001
Value 0x000000 0x000001 0x000002 0x000003 0x000004 0x000005
Table 32. Pink/White Noise Generation for All 5.1 Channel
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6. DTS MANAGER AND OPERATIONS
The module described in this section is specific to DTS code. In addition to the DTS Manager, certain operations such as enabling Autodetection of a DTS stream is also included in this section. DTS application code (dts_32xx.ld) supports the following types of DTS data: DTS-1 data (512sample bursts) (Pc = B); DTS-2 data (1024-sample burst) (Pc = C); DTS-3 data (2048-sample bursts) (Pc = D). This application code also supports DTS CDs (DTS Format-16 elementary stream) and DTS LDs (DTS Format-14 elementary stream). Other DTS data formats (not currently available or defined) may be able to be decoded, but will not be able to be autodetected.
Note: For DTS certification, your system should be able to enable DTS decoding with Autodetection disabled in order to detect and decode certain streams which are only available on the DTS certification test disc.
6.1 DTS Manager
Write Opcode = 0x90; Read Opcode = 0x11; Read Response Opcode = 0x91 Write Message = 0x9000HH 0xhhhhhh Read Request Message = 0x1100HH; Read Response Message = 0x9100HH 0xhhhhhh 0xHH = index, 0xhhhhhh = data value
Index
0x00
Variable
DTS_CONTROL
Dataword Content
Bit 8 = DRC_Enable = 0/1 = Disable / Enable Dynamic Range Compression (DRC) Note: DRC coefficients must be present in the data stream for DRC to take effect. Bits 7:4 = Output_Mode = 0...7 0 = 2/0 Lt, Rt Surround Encoded. (Refer to Section 6.1.2 "Downmix Equation for Lt, Rt" on page 86 for the Equation). 1 = 1/0 C. 2 = 2/0 L, R. 3 = 3/0 L, C, R. 4 = 2/1 L, R, S (Ls = Rs = S - 3dB). 5 = 3/1 L, C, R, S (Ls = Rs = S - 3dB). 6 = 2/2 L, R, Ls, Rs. 7 = 3/2 L, C, R, Ls, Rs. Bit 0 = DTS_Enable = 0/1 = Disable/Enable DTS decoding. Note: Undesired Channel outputs should be explicitly muted by setting the corresponding Ch_Vol (see Section 4.1 "Audio Manager for Dolby Digital, DTS, MP3, MPEG Multichannel, AAC and Crystal Original Surround" on page 33) to 0.0
0x01 0x02
FTYPE SHORT
Default* = 0x000070 0..1 = Frame Type Identifier. READ-ONLY 0..31 = Deficit Sample Count. READ-ONLY Table 33. DTS Manager
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Index
0x03 0x04 0x05 0x06 CPF NBLKS FSIZE AMODE
Variable
0..1 = CRC Present Flag.
Dataword Content
READ-ONLY 5..127 = Number of PCM Sample Blocks. READ-ONLY 96..8192 = Primary Frame Byte Size. READ-ONLY 0..63 = Audio Channel Arrangement (AMODE). 0x0 = A 0x1 = A + B (dual mono) 0x2 = L + R (stereo) 0x3 = (L+R) + (L-R) (sum-difference) 0x4 = LT + RT (left and right total) 0x5 = C + L + R 0x6 = L + R + S 0x7 = C + L + R + S 0x8 = L + R + SL + SR 0x9 = C + L + R + SL + SR 0xa - 0xf = Reported, but not supported by this code. 0x10 - 0x3f = User defined. Note: Changes in this stream variable will result in an Unsolicited Message from the DSP if Audio_Config_Change_Notification has been enabled in the Audio Manager. Please refer to Section 4.1.1 "Audio_Control: (Index 0x00)" on page 37 for more information.
0x07 0x08 0x09 0x0a 0x0b 0x0c 0x0d 0x0e
SFREQ RATE MIX DYNF TIMEF AUXF EXT_AUDIO ASPF
READ-ONLY 0..15 = Source Sampling Frequency. READ-ONLY 0..31 = Transmission Bit Rate. READ-ONLY 0..1 = Embedded Down Mix Enabled. READ-ONLY 0..1 = Embedded Dynamic Range Flag. READ-ONLY 0..1 = Embedded Time Stamp Flag. READ-ONLY 0..1 = Auxiliary Data Flag. READ-ONLY 0..1 = Extended Coding Flag. READ-ONLY 0..1 = Audio Sync Word Insertion Flag. READ-ONLY Table 33. DTS Manager (Continued)
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Index
0x0f LFF
Variable
0x0 = Not Present 0x1 = Present, 128 0x2 = Present, 64 0x3 = Invalid
Dataword Content
0..3 = LFF Channel Interpolation factor:
Note: Changes in this stream variable will result in an Unsolicited Message from the DSP if Audio_Config_Change_Notification has been enabled in the Audio Manager. Please refer to Section 4.1.1 for more information. 0x10 0x11 0x12 HFLAG FILTS PCMR READ-ONLY 0..1 = Predictor History Flag Switch. 0..1 = Multirate Interpolator Switch. READ-ONLY 0..7 = Source PCM coding Resolution. This variable has also been chosen as the indicator of a DTS stream being DTS-ES encoded. The DTS-ES present flag has been placed into the LSB of the PCMR field. If the PCMR variable is reported as being set to 0x3 or 0x5, then the material is DTS-ES encoded. 0x0 = 16-bits 0x1 = 16-bits (or material is DTS-ES encoded or DTS 96/24 encoded) 0x2 = 20-bits 0x3 = 20-bits (or material is DTS-ES encoded or DTS 96/24 encoded) 0x4 = Invalid 0x5 = 24-bits (or material is DTS-ES encoded or DTS 96/24 encoded) 0x6 = 24-bits 0x7 = Invalid Note: Changes in this stream variable will result in an Unsolicited Message from the DSP if Audio_Config_Change_Notification has been enabled in the Audio Manager. Please refer to Section 4.1.1 for more information. 0x13 0x14 0x15 0x16 0x17 0x18 0x19 SUMF SUMS SUBFS PCHS SSC PSC RANGE READ-ONLY 0..1 = Front Sum / Difference Flag. READ-ONLY 0..1 = Surround Sum / Difference Flag. READ-ONLY 0..15 = Number of Subframes. READ-ONLY 0..7 = Number of Primary Audio Channels. READ-ONLY 0..3 = Subsubframe Count. READ-ONLY 0..7 = Partial Subsubframe Sample Count. READ-ONLY Dynamic Range Coefficient. READ-ONLY Table 33. DTS Manager (Continued) AN163REV3 85
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Index Variable Dataword Content
0..31 = Subband Activity Count. READ-ONLY 0..31 = High Frequency VQ Start Subband. READ-ONLY 0..7 = Joint Intensity Coding Index.
0x1a..0x1e SUBS[0...4] 0x1f..0x23 VQSUB[0...4] 0x24..0x28 JOINX[0...4]
READ-ONLY 0x29..0x32 DOWN[0...4][0...1] 0..127 = Scale Factors for downmixing. READ-ONLY Table 33. DTS Manager (Continued) Notes: "Default" vs. "Default*" See definition page 32.
6.1.2 Downmix Equation for Lt, Rt
The following equation is used to produce a Lt, Rt Surround Encoded downmix. Lt = (((1/3.1)L) + ((0.7/3.1)C) - ((0.7/3.1)Ls) ((0.7/3.1)Rs) Rt = (((1/3.1)L) + ((0.7/3.1)C) + ((0.7/3.1)Ls) + ((0.7/3.1)Rs)
6.1.1 DTS_CONTROL
Bits 8 DRC_ENABLE By enabling this bit the DSP will recognize
Dynamic
Range
Compression
(DRC)
coefficients placed in the data stream by a DTScertified encoder. Bits 7:4: OUTPUT_MODE These bits set up the output speaker downmix configuration. When configuring the Output _Mode, first the individual volumes of each undesired channel should be set to 0. Next, the desired output mode should be specified. This should be done before kickstart of the application. For example, if output mode 3/0 (L, C, R) is chosen, Ls and Rs volume must explicitly be set to 0 in the Audio Manager module to guarantee no output on those channels. Here is a summary of Output_Mode configuration settings. 0 = 2/0 Lt, Rt Surround Encoded. 1 = 1/0 C. 2 = 2/0 L, R. 3 = 3/0 L, C, R. 4 = 2/1 L, R, S (Ls = Rs = S - 3dB). 5 = 3/1 L, C, R, S (Ls = Rs = S - 3dB). 6 = 2/2 L, R, Ls, Rs. 7 = 3/2 L, C, R, Ls, Rs.
6.1.3 DTS Stream Information
Table 33 lists out a complete set of READ-ONLY variables that are reported to the host by the DTS Decoder module. These can be read using the appropriate Read Request, and Read Response sessions. For example, to read the AMODE value, host should issue the 3-byte Read Request 0x110006 and then read the 6-byte Read Response 0x910006, 0x00009h, where the dataword (latter 3 bytes) indicates the reported value of AMODE. Please refer to the DTS Specification for more information on these stream variables.
6.1.4 DTS Decode with Autodetect
In this mode, the input should be compressed DTS. The data will be taken from the compressed input port as designated by the Hardware Configuration message. If the input on this port is not DTS, the Autodetect function will notify the host as described in Section 4.3.1 "Autodetect Operation" on page 55.
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Module DTS Manager Audio Manager
Index DTS Control Audio Manager Control
Description
Opcode & Index
Value 0x000071 0x001011
DTS Enabled (Assumes all 0x900000 5.1 channels are available.) Autodetect Enabled PLL Enabled Kickstart Application 0x880000
Table 34. Enabling DTS Decode with Autodetect
Figure 20 is an example of pseudocode to configure the part DTS decode with autodetect. This pseudocode can be used as a template for
void DTS_AD_Config() { unsigned char DTS_AD_config_message[] =
other configurations by swapping out the values in the message array and the size.
{0x90, 0x00, 0x00, 0x00, 0x00, 0x71, 0x88, 0x00, 0x00, 0x00, 0x10, 0x11}; Write_*(DTS_AD_config_message,12); /* Replace * with I2C or SPI depending on protocol */ }
Figure 20. DTS decode with Autodetect
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6.1.5 Special Considerations For DTS CDs and Autodetection
A DTS CD is treated like a linear PCM CD by all CD/DVD players. During FF/REW ("trick" modes), the CD/DVD player drops audio samples to create an effect audible to the listener. Dropping DTS data, however, causes corruption of the bitstream which makes the data indistinguishable from linear PCM. As a result, the DTS decoder will generate an autodetect message indicating PCM was detected. However, if a user actually changed the CD (either by hand or by CD changer), this would cause the DSP to report silence; also known as "Silent Input Data (Out of Application Sync)." If this is the case, the host must then respond to whatever the DSP reports after this detection of silence. If the disc is switched to PCM, then the host must load the corresponding PCM code. If the disc is switched to DTS, then the host must load the DTS code. An abrupt transition from DTS to PCM should indicate to the system controller that a "trick" mode is being used, hence the PCM code should not be loaded. If the host does nothing, the DTS code will then automute during this FF/REW period and there will be no problems. Please refer to Figure 21 to see a flow chart depicting the DTS autodetection process. However, the recommendation described above does not take into account a sudden change from DTS to PCM due to a source change. It is therefore suggested that if the user selects a source change (e.g. could change input from a DTS CD playing to a DAT playing 44.1 kHz PCM), the host will be aware of this input change and should subsequently perform a Soft Reset, Configure the Hardware, and
Perform Kickstart with Autodetect
Run Time (Autodetect Enabled) INTREQ goes LOW User Reads Unsolicited Response
Last Input = Current Input OR Last Input = Silence
Host Does Nothing Yes
No
Last Input = Compressed Bitstream AND Current Input = Different Compressed Bitstream
Yes
Host Loads Corresponding AC-3 or MPEG Code, Configures Hardware and Performs Kickstart with Autodetect Bypass
No
Last Input = PCM Bitstream AND Current Input = DTS
Yes
Host Loads DTS Code, Configures Hardware and Performs Kickstart
No
Last Input = DTS Bitstream AND Current Input = PCM Bitstream
Yes
Host performs soft mute and waits for stream to change back to DTS, unless host is aware of source input change.
Figure 21. DTS Specific Autodetection Flowchart
perform a kickstart. This will ensure protection from such a scenario.
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7. MP3 MANAGER AND OPERATIONS
The module described in this section is specific to MP3 code. In addition to the MP3 Manager, certain operations such as enabling Autodetection of a MP3 stream is also included in this section. MP3 application code (mp3_32xx.ld) only supports the following types of MPEG data: (1) MPEG-1, Layer 3 (Pc = 5), and (2) MPEG-2, Layer 3 CBR and VBR streams. It also supports the half-sampling frequencies outlined in the MPEG-2, Layer 3 specification. It does not support the "free" format where the bit rate is unspecified (bit_index = "0000"=free). The MP3 code supports IEC60958 Stereo PCM output or IEC61937 packed MP3 in an I2S, left-justified, or S/PDIF format from the AUDATA3 pin.
7.1 MP3 Manager
Write Opcode = 0x8C; Read Opcode = 0x0D; Read Response Opcode = 0x8D Write Message = 0x8C00HH 0xhhhhhh Read Request Message = 0x0D00HH; Read Response Message = 0x8D00HH 0xhhhhhh 0xHH = index, 0xhhhhhh = data value
Index
0x00
Variable
MP3_CONTROL 0 = Reserved. 1 = 1/0 C. 2 = 2/0 L, R. 3 = 2/0 L, R. 4 = 2/0 L, R. 5 = 2/0 L, R. 6 = 2/0 L, R. 7 = 2/0 L, R.
Dataword Content
Bits 7:4 = Output_Mode = 0...7
Note: Undesired Channel outputs should be explicitly muted by setting the corresponding Ch_Vol (see Audio Manager). It should also be noted that unlike the other application code managers in AN163, the MP3 Manager does not have a "Decode Enable Bit." This is intentional. 0x01 ID Default* = 000070 0 - Extension to lower frequencies, 1 - ISO/IEC 11172-3 or ISO/IEC 13818-3. READ-ONLY 2 bits to indicate which layer is used. READ-ONLY 0 - stream is CRC protected, 1 - not CRC protected. READ-ONLY 4-bit index to a bitrate table. READ-ONLY Table 35. MP3 Manager
0x02 0x03 0x04
LAYER PROTECTION_BIT BITRATE_INDEX
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Index
0x05 0x06 0x07 0x08
Variable
SAMPLING_FREQUENCY PADDING_BIT PRIVATE_BIT MODE
Dataword Content
2-bit index to a sampling frequency table. READ-ONLY 0 - no padding, 1 - padding exists. READ-ONLY reserved for non-ISO purposes. READ-ONLY 2 bits to indicate the audio configuration mode (for 2 channels only) 00 - stereo 01 - joint stereo (intensity stereo and/or ms_stereo) 10 - dual channel 11 - single channel Note: Changes in this stream variable will result in an Unsolicited Message from the DSP if Audio_Config_Change_Notification has been enabled in the Audio Manager. Please refer to Section 4.1.1 "Audio_Control: (Index 0x00)" on page 37 for more information.
0x09 0x0a 0x0b 0x0c
MODE_EXTENSION COPYRIGHT ORIGHOME EMPHASIS
READ-ONLY 2 bits to indicate which sub-bands are in intensity stereo. READ-ONLY 0 - no copyright, 1 - copyright protected bitstream. READ-ONLY 0 - bitstream is a copy, 1 - bitstream is an original. READ-ONLY 2 bits to indicate which type of de-emphasis is used. READ-ONLY Table 35. MP3 Manager (Continued)
Notes:
See definition on page 32.
"Default" vs. "Default*" See definition page 32.
7.1.1 MP3_CONTROL
Bits 7:4: OUTPUT MODE These bits set up the output speaker downmix configuration. When configuring the Output_Mode, first set the individual volumes of each undesired channel to 0. Next, the desired output mode should be specified. This should be done before kickstart of the application. For example, if output mode 1/0 C is chosen, R, L volume must explicitly be set to 0 in the Audio Manager module to guarantee no output on those channels. MPEG Stream Information: Table 35 lists a complete set of READ-ONLY variables that are
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reported to the host by the MP3 Decoder module. These can be read using the appropriate Read Request, and Read Response sessions. For example, to read the MODE value, host should issue the 3-byte Read Request 0x0D0008 and then read the 6-byte Read Response 0x8D0008, 0x00000h, where the dataword (latter 3 bytes) indicates the reported value of MODE. Please refer to the MPEG Specification ISO/IEC 11172-3 for more information on these stream variables.
7.2 MP3 Decode with Autodetect
In this mode, the input should be compressed MP3. The data will be taken from the compressed input port as designated by the Hardware Configuration
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message. If the input on this port is not MP3, the Autodetect function will notify the host as described in Section 4.3.1 "Autodetect Operation" on page 55. Figure 22 is an example of pseudocode to configure the part MP3 decode with autodetect. This pseudocode can be used as a template for other configurations by swapping out the values for the message array and the size in the pseudocode.
Note: This same Unsolicited Message of 0x000005 could also indicate that the stream was IEC61937-packed MPEG-1, Audio Layer 2 or IEC61937-packed MPEG-2 without extension. However, the only way to determine that the stream is MP3 is to download the MP3 application code.
7.2.1 Special Considerations MP3 Autodetect
In general, if the Autodetect function senses AC-3, DTS, PCM, or MPEG data, the appropriate Unsolicited Message is sent indicating that the corresponding stream was detected. The host would then need to download the appropriate code. However, if the autodetect function senses a valid IEC61937-packed MP3 stream, it will send the host an Unsolicited Message of 0x000005 (provided the current application code was not MP3). The host must then download the MP3 application code, in order to determine if the stream is MPEG-1, Audio Layer 3 or is another type of MPEG with a reported Pc value of 5.
After downloading the MP3 application code, configuring the hardware and performing a kickstart, the DSP will then generate an Unsolicited Message indicating that the stream has a Pc value of 5 (0x000005). This first Unsolicited Message does not indicate the MPEG stream type. If the input stream is MPEG-1, Audio Layer 3, the DSP will send the host an Unsolicited Message of 0x800005 and begin decoding the audio stream.
Note: This second Unsolicited Message may not occur for up to 26ms after the first Unsolicited Message.
However if the host has not seen a second Unsolicited Message of 0x80005 after 26ms from the first Unsolicited Message of 0x000005, the host then knows that the input stream is MPEG-1, Layer 2 or MPEG-2 without extension. The host must then download the MPEG-2 application code (mpg_2xxx.ld), Configure the Hardware and perform a kickstart.
Module MP3 Manager Audio Manager
Index MP3 Control Audio Manager Control
Description
Opcode & Index
Value 0x000020 0x001011
MP3 Enabled (Assumes 0x8C0000 2.0 channels are available.) Autodetect Enabled PLL Enabled Kickstart Application 0x880000
Table 36. Enabling MP3 Decode with Autodetect
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void MP3_AD_Config() { unsigned char MP3_AD_config_message[] = {0x8C, 0x00, 0x00, 0x00, 0x00, 0x20, 0x88, 0x00, 0x00, 0x00, 0x10, 0x11}; Write_*(MP3_AD_config_message,12); /* Replace * with I2 C or SPI depending on protocol */ }
Figure 22. MP3 Decode with Autodetect
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7.2.2 For System Designers: When to Download MP3 Application code...
The worst-case autodetection scenario is if MPEG Multichannel application code is already loaded into the DSP and the input data stream changes seemlessly (on the frame, and with no frames of silence) from an MPEG-1, Layer 2 IEC61937packed data stream or an MPEG-2 without extension IEC61937-packed data stream to an MPEG-1, Layer 3 IEC61937-packed data stream. Detection of both stream types by the MPEG code will always report 0x000005, however if the stream is actually MPEG-1, Layer 3, the DSP will not be able to decode the stream, hence no audio will be heard. However, we will assume that an MPEG-1, Layer 2 or MPEG-2 without extension stream will be separated by some detectable amount of silence (approximate 500ms) before being followed by an MPEG-1, Audio Layer 3 stream. With this assumption in mind, the two data streams will be separated with an entirely different type of stream, silence. If the current application code is MPEG and it detects silence, it will send an Unsolicited Message of 0x000010. The host must then download the MP3 application code, configure the hardware and perform a kickstart. Again, if the DSP detects anything other than silence, it will generate the appropriate unsolicited message which the host will interpret and then download the appropriate application code. The inverse of the above scenario however does not present a problem though if the MP3 application code was already loaded. The host would have received an Unsolicited Message of 0x800005 when MPEG-1, Layer 3 was detected. Even if the input data stream changed seemlessly (on the frame, and with no frames of silence) from an MPEG-1, Layer 3, IEC61937-packed data stream to an MPEG-1, Layer 2, IEC61937-packed data stream or an MPEG-2 without extension IEC61937-packed data stream, the host would still report an Unsolicited Message of 0x00005. The host need only wait 26 ms and then load the MPEG-2 application code, and audio could then be heard. Again, if another type of application code is loaded, like AC-3, and the DSP sends the host an Unsolicited Message of 0x000005, the host must download the MP3 to determine if the reported Pc value of 5 indicates that the stream is either MPEG1, Layer 2 or MPEG-1, Layer 3 or MPEG-2 without extension. Please see Section 23 "When to Download MP3 Code..." on page 94, for an overview of the previously described summary of when to download MP3 versus MPEG Multichannel application code.
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Download Corresponding Code, Configure H/W, and perform Kickstart with PLL and Autodetect Enabled
Unsolicited Message: 0x000005
Download MP3 Code, Configure H/W, and perform Kickstart with PLL and Autodetect Enabled
Host waits for at least 26ms
No Additional Unsolicited Message of 0x800005
Unsolicited Message: 0x800005
Download MPEG-2 Code, Configure H/W, and perform Kickstart with PLL and Autodetect Enabled
MPEG-1, Audio Layer 2 or MPEG-2 without extension begins to decode...
MPEG-1, Audio Layer 3 begins to decode...
Unsolicited Message: NOT == 0x000005
Unsolicited Message: NOT == 0x800005
Figure 23. When to Download MP3 Code...
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8. MPEG MULTICHANNEL MANAGER AND OPERATIONS
The module described in this section is specific to MPEG code. In addition to the MPEG Manager, certain operations such as enabling Autodetection of an MPEG stream are also included in this section. MPEG-2 Multichannel application code (mpg_32xx.ld) supports the following types of MPEG data: MPEG-1, Layer 2 (Pc = 5); MPEG-2 without extension (Pc = 5); MPEG-2 with extension (Pc = 6). Please refer to MP3 application code for support of MPEG-1, Audio Layer 3.
Note: MPEG-2 Multichannel application code (mpg_32xx.ld) does NOT support the following types of MPEG data: MPEG-1, Audio Layer 1 (Pc = 4); or MPEG-2, Audio Layer 1, Low Sampling Frequency (Pc = 8); MPEG-2, Audio Layer 2 or 3, Low Sampling Frequency (Pc = 9).
8.1 MPEG Multichannel Manager
Write Opcode = 0x8C; Read Opcode = 0x0D; Read Response Opcode = 0x8D Write Message = 0x8C00HH 0xhhhhhh Read Request Message = 0x0D00HH; Read Response Message = 0x8D00HH 0xhhhhhh 0xHH = index, 0xhhhhhh = data value
Index
0x00
Variable
MPEG_CONTROL
Dataword Content
Bits 7:4 = Output_Mode = 0...7 0 = 2/0 Lt, Rt Surround Encoded. (Refer to Section 8.1.2 "Downmix Equation for Lt, Rt" on page 98 for Equation). 1 = 1/0 C. 2 = 2/0 L, R. 3 = 3/0 L, C, R. 4 = 2/1 L, R, S (Ls = Rs = S - 3dB). 5 = 3/1 L, C, R, S (Ls = Rs = S - 3dB). 6 = 2/2 L, R, Ls, Rs. 7 = 3/2 L, C, R, Ls, Rs. Bit 0 = MPEG_Enable = 0/1 = Disable/Enable MPEG decoding. Default = 0x000070 Note: Undesired Channel outputs should be explicitly muted by setting the corresponding Ch_Vol (see Section 4.1 "Audio Manager for Dolby Digital, DTS, MP3, MPEG Multichannel, AAC and Crystal Original Surround" on page 33) to 0.0 0 - Extension to lower frequencies, 1 - ISO/IEC 11172-3 or ISO/IEC 13818-3. READ-ONLY 2 bits to indicate which layer is used. READ-ONLY 0 - stream is CRC protected, 1 - not CRC protected. READ-ONLY Table 37. MPEG Manager
0x01
ID
0x02 0x03
LAYER PROTECTION_BIT
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Index
0x04 0x05 0x06 0x07 0x08
Variable
BITRATE_INDEX SAMPLING_FREQUENCY PADDING_BIT PRIVATE_BIT MODE
Dataword Content
4-bit index to a bitrate table. READ-ONLY 2-bit index to a sampling frequency table. READ-ONLY 0 - no padding, 1 - padding exists. READ-ONLY reserved for non-ISO purposes. READ-ONLY 2 bits to indicate the audio configuration mode (for 2 channels only) 00 - Stereo 01 - Joint stereo (intensity stereo and/or ms_stereo) 10 - Dual channel 11 - Single channel Note: Changes in this stream variable will result in an Unsolicited Message from the DSP if Audio_Config_Change_Notification has been enabled in the Audio Manager. Please refer to Section 4.1.1 "Audio_Control: (Index 0x00)" on page 37 for more information.
0x09 0x0a 0x0b 0x0c 0x0d 0x0e
MODE_EXTENSION COPYRIGHT ORIGHOME EMPHASIS
READ-ONLY 2 bits to indicate which sub-bands are in intensity stereo. READ-ONLY 0 - no copyright, 1 - copyright protected bitstream. READ-ONLY 0 - bitstream is a copy, 1 - bitstream is an original. READ-ONLY 2 bits to indicate which type of de-emphasis is used.
READ-ONLY EXT_BIT_STREAM_PRESENT 0 - extension bit stream not present, 1 - ext. present. N_AD_BYTES READ-ONLY 8 bits: number of bytes used in MPEG-1 base frame for ancillary data field if an extension bit stream exists. READ-ONLY 2 bits to indicate presence and bandwidth of center channel. 00 - No center channel present 01 - Center channel present 10 - Not defined 11 - Center bandwidth limited (Phantom coding) Note: Changes in this stream variable will result in an Unsolicited Message from the DSP if Audio_Config_Change_Notification has been enabled in the Audio Manager. Please refer to Section 4.1.1 "Audio_Control: (Index 0x00)" on page 37 for more information. READ-ONLY Table 37. MPEG Manager (Continued)
0x0f
CENTER
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Index
0x10
Variable
SURROUND
Dataword Content
2 bits to indicate whether surround channels are present or if secondary stereo exists. 00 - No surround 01 - Mono surround 10 - Stereo surround 11 - No surround, but second stereo program present Note: Changes in this stream variable will result in an Unsolicited Message from the DSP if Audio_Config_Change_Notification has been enabled in the Audio Manager. Please refer to Section 4.1.1 "Audio_Control: (Index 0x00)" on page 37 for more information.
0x11
LFE
READ-ONLY 1 bit to indicate whether LFE channel is present. 0 - LFE not present 1 - LFE present Note: Changes in this stream variable will result in an Unsolicited Message from the DSP if Audio_Config_Change_Notification has been enabled in the Audio Manager. Please refer to Section 4.1.1 "Audio_Control: (Index 0x00)" on page 37 for more information.
0x12 0x13 0x14
AUDIO_MIX DEMATRIX_PROCEDURE
READ-ONLY 1 bit to indicate whether the signal is mixed for large listening rooms. READ-ONLY 2 bits to indicate which dematrix procedure is used.
READ-ONLY NO_OF_MULTI_LINGUAL_CH 3 bits: number of multi-lingual channels present in the Multichannel extension. MULTI_LINGUAL_FS MULTI_LINGUAL_LAYER COPYRIGHT_ID_BIT COPYRIGHT_ID_START READ-ONLY 1 bit to indicate if fs is the same as in the main audio. READ-ONLY 1 bit to indicate which layer is used. READ-ONLY 1 bit: part of a 72-bit copyright identification number. READ-ONLY 0 - no start of copyright identification in this frame, 1 - start of copyright identification in this audio frame. READ-ONLY Table 37. MPEG Manager (Continued)
0x15 0x16 0x17 0x18
Notes:
See definition on page 32.
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8.1.1 MPEG_CONTROL
Bits 7:4: OUTPUT_MODE These bits set up the output speaker downmix configuration. When configuring the OUTPUT _MODE, first the individual volumes of each undesired channel should be set to 0. Next, the desired output mode should be specified. This should be done before kickstart of the application. For example, if output mode 3/0 (L, C, R) is chosen, Ls and Rs volume must explicitly be set to 0 in the Audio Manager module to guarantee no output on those channels. 0 = 2/0 Lt, Rt Surround Encoded. 1 = 1/0 C. 2 = 2/0 L, R. 3 = 3/0 L, C, R. 4 = 2/1 L, R, S (Ls = Rs = S - 3dB). 5 = 3/1 L, C, R, S (Ls = Rs = S - 3dB). 6 = 2/2 L, R, Ls, Rs. 7 = 3/2 L, C, R, Ls, Rs.
8.1.2 Downmix Equation for Lt, Rt
The following equation is used to produce a Lt, Rt Surround Encoded downmix. Lt = (((1/3.1)L) + ((0.7/3.1)C) - ((0.7/3.1)Ls) ((0.7/3.1)Rs) Rt = (((1/3.1)L) + ((0.7/3.1)C) + ((0.7/3.1)Ls) + ((0.7/3.1)Rs)
8.1.3 MPEG Stream Information
Table 37 on page 95 lists a complete set of READONLY variables that are reported to the host by the MPEG Decoder module. These can be read using the appropriate Read Request, and Read Response sessions. For example, to read the MODE value, host should issue the 3-byte Read Request 0x0D0008 and then read the 6-byte Read Response 0x8D0008, 0x00000h, where the dataword (latter 3 bytes) indicates the reported value of MODE. Please refer to the MPEG Specification (ISO/IEC 11172-3 and ISO/IEC 13818-3) for more information on these stream variables.
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8.2 MPEG Multichannel Decode with Autodetect
In this mode, the input should be compressed MPEG (of a supported stream type). The data will be taken from the compressed input port as designated by the Hardware Configuration message. If the input on this port is not MPEG, the Autodetect function will notify the host as
Module MPEG Manager Index MPEG Control
described in Section 4.3.1 "Autodetect Operation" on page 55. Figure 24 is an example of pseudocode to configure the part AC-3 decode with autodetect. This pseudocode can be used as a template for other configurations by swapping out the message array and the size.
Description
Opcode & Index
Value 0x000071
MPEG Enabled (Assumes 0x8C0000 all 5.1 channels are available.) Autodetect Enabled PLL Enabled Kickstart Application 0x880000
Audio Manager
Audio Manager Control
0x001011
Table 38. Enabling MPEG Decode with Autodetect
void MPEG_AD_Config() { unsigned char MPEG_AD_config_message[] = {0x8C, 0x00, 0x00, 0x00, 0x00, 0x71, 0x88, 0x00, 0x00, 0x00, 0x10, 0x11}; Write_*(MPEG_AD_config_message,12); /* Replace * with I2C or SPI depending on protocol */ }
Figure 24. MPEG Multichannel decode with Autodetect
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9. AAC MANAGER AND OPERATIONS
The module described in this section is specific to AAC code. The only AAC-specific module is the AAC Manager. In addition to the AAC-specific module mentioned before, certain operations such as enabling Autodetection of a AAC stream is also included in this section. AAC application code (aac_329x.ld) only supports the following type of AAC data: AAC, Low Complexity, and ADTS Format (Pc = 1C).
9.1 AAC Manager
Write Opcode = 0x9E; Read Opcode = 0x1F; Read Response Opcode = 0x9F Write = 0x9E00HH 0xhhhhhh Read Request = 0x1F00HH; Read Response = 0x9F00HH 0xhhhhhh 0xHH = index, 0xhhhhhh = data value
Index
0x00
Variable
AAC_CONTROL
Dataword Content
Bits 7:4 = Output_Mode = 0...7 0 = 2/0 Lt, Rt Surround Encoded. (Equation in Section 9.1.2 "Downmix Equation for Lt, Rt" on page 102). 1 = 1/0 C. 2 = 2/0 L, R. 3 = 3/0 L, C, R. 4 = 2/1 L, R, S (Ls = Rs = S - 3dB). 5 = 3/1 L, C, R, S (Ls = Rs = S - 3dB). 6 = 2/2 L, R, Ls, Rs. 7 = 3/2 L, C, R, Ls, Rs. Bit 0 = AAC_Enable = 0/1 = Disable/Enable AAC decoding. Default* = 0x000070 0x0...0x3 = profile used. See ISO/IEC 13818-7, subclause 2.1. Note: Only Low Complexity (LC) profile == 0x1 is supported by this application code.
0x01
PROFILE
0x02
FS_INDEX
0x03
PCE_PRESENT
READ-ONLY 0x0...0xF: Index to AAC sampling frequency table. Only the following frequencies are supported: 0x3 = 48 kHz 0x4 =44.1 kHz 0x5 =32 kHz READ-ONLY 0 - The num_?_channel elements variables (see 0x04 - 0x06 below) are not valid 1 - The num_?_channel_elements variables (see 0x04 - 0x06 below) are valid READ-ONLY Table 39. AAC Manager
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Index
0x04
Variable
Dataword Content
0x05
NUM_FRONT_CHANNEL_ELEMENTS Number of audio syntactic elements in the front channels, front center to back center, symmetrically by left and right, or alternating by left and right in the case of single channel elements. READ-ONLY NUM_FRONT_CHANNELS Number of front channels. This field is valid only when channel_configuration is zero. NUM_BACK_CHANNEL_ELEMENTS NUM_BACK_CHANNELS READ-ONLY Number of side and front channel elements, for back channels. READ-ONLY Number of back channels. This field is valid only when channel_configuration is zero. READ-ONLY Number of LFE channel elements associated with the program. READ-ONLY Bit 0 = Home bit from the stream. READ-ONLY Bit 0 = Original/Copy bit from the input stream. READ-ONLY Bit 0 = When 1, Index 0x0c to 0x0e carries the last received copyright identification field. READ-ONLY Bits 0:23 = Map to the first 24 bits of the 72-bit copyright identification field. Bit 23 is the first bit of the field. READ-ONLY Bits 0:23 = Map to the second 24 bits of the 72-bit copyright identification field. Bit 23 is the 25th bit of the field, while Bit 0 is the 48th bit of the field.
0x06 0x07
0x08 0x09 0x0a 0x0b
NUM_LFE_CHANNEL_ELEMENTS HOME ORIGINAL_COPY COPYRIGHT_ID_PRESENT
0x0c
COPYRIGHT_ID_WORD_1
0x0d
COPYRIGHT_ID_WORD_2
0x0e
COPYRIGHT_ID_WORD_3
READ-ONLY Bits 0:23 = Map to the second 24 bits of the 72-bit copyright identification field. Bit 0 is the 72nd bit of the field. READ-ONLY 13 bit field: length of the frame in bytes, including headers and error_check. READ-ONLY Number of raw_data_blocks in the ADTS frame. READ-ONLY
0x0f
FRAME_LENGTH
0x10
NUMBER_OF_RAW_DATA_BLOCKS_ IN_FRAME
Table 39. AAC Manager (Continued)
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Index
0x11
Variable
CHANNEL_CONFIGURATION
Dataword Content
0x0...0x7 = channel configuration used. If channel_configuration is greater than 0, the channel configuration is given by the default bitstream index number in Table 41, Implicit Speaker Mapping. If channel_configuration equals 0, the channel configuration is not specified in the header and must be given by a program_config_element following as first bitstream element in the first raw_data_block after the header, or by the implicit configuration (See ISO/IEC 13818-7, subclause 3.5) or must be known in the application. Note: Changes in this stream variable will result in an Unsolicited Message from the DSP if Audio_Config_Change_Notification has been enabled in the Audio Manager. Please refer to Section 4.1.1 "Audio_Control: (Index 0x00)" on page 37 for more information. READ-ONLY
0x12
PROTECTION_ABSENT
0 - stream is CRC protected, 1 - stream is not CRC protected. READ-ONLY Table 39. AAC Manager (Continued)
Notes:
See definition on page 32.
"Default" vs. "Default*" See definition page 32.
9.1.1 AAC_CONTROL
Bits 7:4: OUTPUT_MODE These bits set up the output speaker downmix configuration. When configuring the OUTPUT _MODE, first set the individual volumes of each undesired channel to 0. Next, the desired output mode should be specified. This should be done before kickstart of the application. For example, if output mode 3/0 (L, C, R) is chosen, Ls and Rs volume must explicitly be set to 0 in the Audio Manager module to guarantee no output on those channels. 0 = 2/0 Lt, Rt Surround Encoded. 1 = 1/0 C. 2 = 2/0 L, R. 3 = 3/0 L, C, R. 4 = 2/1 L, R, S (Ls = Rs = S - 3dB). 5 = 3/1 L, C, R, S (Ls = Rs = S - 3dB). 6 = 2/2 L, R, Ls, Rs. 7 = 3/2 L, C, R, Ls, Rs.
9.1.2 Downmix Equation for Lt, Rt
The following equation is used to produce a Lt, Rt Surround Encoded downmix. Lt = (((1/3.1)L) + ((0.7/3.1)C) - ((0.7/3.1)Ls) ((0.7/3.1)Rs) Rt = (((1/3.1)L) + ((0.7/3.1)C) + ((0.7/3.1)Ls) + ((0.7/3.1)Rs)
9.1.3 AAC Stream Information
Table 39 lists a complete set of READ-ONLY variables that are reported to the host by the AAC Decoder module. These can be read using the appropriate Read Request, Read Response sessions. For example, to read the PROFILE value, host should issue the 3-byte Read Request 0x1f0001 and then read the 6-byte Read Response 0x9f0001, 0x00001h, where the dataword (latter 3 bytes) indicates the reported value of PROFILE. Please refer to the MPEG-2 AAC Specification (ISO/IEC 13818-7) for more information on these stream variables.
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9.2 AAC Decode with Autodetect
In this mode, the input should be compressed ADTS formatted AAC. The data will be taken from the compressed input port as designated by the Hardware Configuration message. If the input on this port is not AAC, the Autodetect function will notify the host as described in Section 4.3.1 "Autodetect Operation" on page 55. Figure 25 is an example of pseudocode to configure the part AC-3 decode with autodetect. This pseudocode can be used as a template for other configurations by swapping out the values for the message array and the size in the pseudocode.
Module AAC Manager Audio Manager
Index AAC Control Audio Manager Control
Description
Opcode & Index
Value 0x000071 0x001011
AAC Enabled (Assumes all 0x9E0000 5.1 channels are available.) Autodetect Enabled PLL Enabled Kickstart Application 0x880000
Table 40. Enabling AAC Decode with Autodetect
void AAC_AD_Config() { unsigned char AAC_AD_config_message[] = {0x9E, 0x00, 0x00, 0x00, 0x00, 0x71, 0x88, 0x00, 0x00, 0x00, 0x10, 0x11}; Write_*(AAC_AD_config_message,12); /* Replace * with I2C or SPI depending on protocol */ }
Figure 25. AAC decode with Autodetect
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9.2.1 Implicit Speaker Mapping
Default Bitstream Number of Audio Syntactic Elements, Default Element to Speaker Mapping Index Number Speakers Listed in Order Received
0 1 2 3 4 See Note. 1 2 3 4 See Note. single_channel_element channel_pair_element single_channel_element, channel_pair_element single_channel_element, channel_pair_element, single_channel_element single_channel_element, channel_pair_element, channel_pair_element single_channel_element, channel_pair_element, channel_pair_element, lfe_element single_channel_element, channel_pair_element, channel_pair_element, channel_pair_element, lfe_element Note: If channel configuration is equal to zero, the speaker mapping is given by the PCE. center front speaker left, right front speakers center front speaker left, right front speakers center front speaker left, right front speakers rear surround center front speaker left, right front speakers left surround, right surround rear speakers center front speaker left, right front speakers left surround, right surround rear speakers front low frequency effects speaker center front speaker left, right center front speakers left, right outside front speakers left surround, right surround rear speakers front low frequency effects speaker
5
5
6
5+1
7
7+1
Table 41. Implicit Speaker Mapping
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10. CRYSTAL ORIGINAL SURROUND (C.O.S.) SPECIFIC MODULES AND OPERATIONS
The two modules described in this section are specific to C.O.S. code. They are: the C.O.S. Effects Manager and the C.O.S. PCM Processor. In addition to the C.O.S. specific-modules, certain operations such as enabling Autodetection of a PCM stream is also included in this section, along with a discussion of: reverb, echo, filter techniques, as well as using COEFFMKR.EXE and EFFECTS.EXE in order to help programmers generate their own processing effects configurations. Some sample configurations have been listed to give you an idea of possible techniques you might utilize in your development efforts. The samples given here do not exhaust all the different ways that you might create your own processing effects.
10.1 C.O.S. Effects Manager
Write Opcode = 0x98; Read Opcode = 0x19; Read Response Opcode = 0x99 Write Message = 0x9800HH 0xhhhhhh Read Request Message = 0x1900HH; Read Response Message = 0x9900HH 0xhhhhhh 0xHH = index, 0xhhhhhh = data value
Index
0x00
Variable
EFFECTS_MGR_CONTROL
Dataword Content
Bit 8 = Test Impulse Enable = 0/1 = Disable/Enable replacement of L/R input with periodic impulse signal (repeated every 24000 samples). Bit 4 = Deemphasis_Enable = 0/1 = Disable/Enable PCM Deemphasis. Note: Please refer to the deemphasis section, located after this table, for more information about using digital deemphasis for PCM sampling frequencies other than 44.1 kHz when using IBA-based codes. Bit 0 = Effects Enable = 0/1 = Disable/Enable Surround Effects. Default = 0x000000 -1.0 to 1.0. Default = 0.5 -1.0 to 1.0. Default = 0.5 -1.0 to 1.0. Default = 0.5 -1.0 to 1.0. Default = 0.5 For future use. For future use. For future use. 0...5615 = Delay in samples for each of the 32 taps. Default = 0. Note: Setting is in samples, and will thus produce different delays for different sampling frequencies. Table 42. C.O.S. Effects Manager
0x01 0x02 0x03 0x04 0x05 0x06 0x07 0x08-0x27
L_MONO_MIXLEVEL R_MONO_MIXLEVEL DIRECT_FIR_MIXLEVEL LPF_FIR_MIXLEVEL RESERVED RESERVED RESERVED T0, T1, ... T30, T31
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Index Variable Dataword Content
-1.0 to 1.0 = Coefficients for each of the 32 taps.
0x28-0x47 C0, C1,... C31
Default = 0.0 0x48 DIRECT_REVERB_MIXLEVEL -1.0 to 1.0. Default = 0.875 0x49 FIR_REVERB_MIXLEVEL -1.0 to 1.0. Default = 0.125 0x4a REVERB_TIME (COMB_G) 0.0 to 1.0 = Reverberation time: unitless constant (minimum 0.0 to maximum 1.0). Default = 0.5 0x4b REVERB_LIVENESS (COMB_H) 0.0 to 1.0 = High-frequency attenuation in reverberator: unitless constant. Default = 0.2 0x4c L_L_OUT_MIXLEVEL -1.0 to 1.0. Default = 1.0 0x4d R_L_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.0 0x4e L_C_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.5 0x4f R_C_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.5 0x50 L_R_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.0 0x51 R_R_OUT_MIXLEVEL -1.0 to 1.0. Default = 1.0 0x52 L_LS_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.5 0x53 R_LS_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.5 0x54 L_RS_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.5 0x55 R_RS_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.5 0x56 L_LFE_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.5 0x57 R_LFE_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.5 0x58 FIR_L_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.0 0x59 REVERB_L_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.0 0x5a FIR_C_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.0 0x5b REVERB_C_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.0 0x5c FIR_R_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.0 0x5d REVERB_R_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.0 0x5e FIR_LS_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.0 0x5f REVERB_LS_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.0 0x60 FIR_RS_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.0 0x61 REVERB_RS_OUT_MIXLEVEL -1.0 to 1.0. Default = 0.0 ADVANCED SETTINGS 0x62 COMB_0_DELAY_SIZE 1...1103 samples. Default=887 0x63 COMB_1_DELAY_SIZE 1...1213 samples. Default=947 0x64 COMB_2_DELAY_SIZE 1...1321 samples. Default=1069 0x65 COMB_3_DELAY_SIZE 1...1433 samples. Default=1297 0x66 COMB_4_DELAY_SIZE 1...1543 samples. Default=1439 0x67 COMB_5_DELAY_SIZE 1...1657 samples. Default=1571 0x68 COMB_0_G1 0.0 to 1.0. Default=0.46 0x69 COMB_1_G1 0.0 to 1.0. Default=0.48 0x6a COMB_2_G1 0.0 to 1.0. Default=0.50 0x6b COMB_3_G1 0.0 to 1.0. Default=0.52 0x6c COMB_4_G1 0.0 to 1.0. Default=0.53 0x6d COMB_5_G1 0.0 to 1.0. Default=0.55 Table 42. C.O.S. Effects Manager (Continued) 106 AN163REV3
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Index
0x6e 0x6f 0x70 0x71 0x72 0x73 0x74 0x75
Variable
COMB_0_MIXLEVEL COMB_1_MIXLEVEL COMB_2_MIXLEVEL Comb_3_Mixlevel Comb_4_Mixlevel Comb_5_Mixlevel Allpass_g1 Allpass_g2
Dataword Content
0.0 to 1.0. Default=0.9 0.0 to 1.0. Default=0.9 0.0 to 1.0. Default=0.9 0.0 to 1.0. Default=0.9 0.0 to 1.0. Default=0.9 0.0 to 1.0. Default=0.9 0.0 to 1.0. Default=0.7 0.0 to 1.0. Default=0.51 Table 42. C.O.S. Effects Manager (Continued)
Notes:
See definition on page 32. current PCM sampling frequency setting via the Audio Manager (0x14), also download the following commands before kickstart:
10.1.1 EFFECTS_MGR_CONTROL
Bit 8: Test Impulse Enable Disable/Enable replaces L/R input with periodic impulse signal (every 24K samples). In this mode, disabling of autodetection causes pulse generation to be independent of the input. If autodetect is left enabled, a pulse will only be generated if PCM is presented at the active PCM input. 1 - Enables Test Impulse 0 - Disables Test Impulse Bit 4: Deemphasis_Enable Disable/Enable PCM Deemphasis. The filter used is identical to CS4334 (50/15 uS). 1 - Enables Deemphasis 0 - Disables Deemphasis For systems that plan on using a "Non-IBA-Based" PCM Processing application code such as eff_32x6.ld and want to support digital deemphasis for the sampling frequencies of 32 kHz and 48 kHz, as well as 44.1 kHz, PCM, the host only needs to inform the DSP the setting for the current PCM sampling frequency, which comes via the Audio Manager (0x14) before kickstart.
Note: For systems that plan on using an "IBA-Based" PCM Processing application code such as effi32xx.ld and want to support digital deemphasis for the sampling frequencies of 32 kHz and 48 kHz, as well as 44.1 kHz, PCM, the host must, in addition to communicating the AN163REV3
For 32 kHz support: 848fff bb82fa 848fe8 42aef3 848fe9 f8d406 For 44.1 kHz support: 848ffe aeacda 848fe6 3b6543 848fe7 f34797 For 48 kHz support: 848ffd ab9ed7 848fe4 39be36 848fe5 f1e0a1 For 64 kHz support: 848fff a25a7b 848fe8 34f310 848fe9 ed676b
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For 88.2 kHz support: 848ffe 99f80b 848fe6 30eb8a 848fe7 e90c82 For 96 kHz support: 848ffd 981257 848fe4 300d08 848fe5 e8054f Digital deemphasis support does not exist for sampling frequencies other than 96 kHz, 88.2 kHz, 64 kHz, 48 kHz, 44.1 kHz, or 32 kHz, regardless of what code is being downloaded. Bit 0: Effects Enable Disable/Enable Surround Effects. 1 - Enables Effects 0 - Disables Effects
Note: Surround Effects parameters (listed without the symbol) are sampled by the CS4932X only once just after kickstart (after a download/Soft Reset or Application Restart). These parameters cannot be varied at run-time. Therefore, to change modes, one has to issue a Soft Reset or Application Restart, reconfigure hardware settings, issue the new Effects messages and then kickstart.
After Reflections. Figure 27 illustrates the various modules of the Crystal Original Surround Effects and the corresponding control parameters available to the user. This forms the Surround Effects Processor whose outputs (square nodes with bold legend) are available to the Output Channel Mixer. The Output Channel Mixer also receives the LR stereo pair input. If Deemphasis is enabled, the LR stereo input is first de-emphasized (50/15 uS). This stage is not shown in Figure 27 and is required only if the input LR PCM input has been pre-emphasized at the source. The next stage allows for Equalization of the input LR stereo pair. When EQ is enabled, stereo PCM input to the Effects Module (after de-emphasis if enabled) is first attenuated by the Pre_EQ_Attenuation scale factor (specified in usual 1.23 format), and then processed by the cascade of three EQ biquads specified by the {EQ0...2: b2, b1, b0, a2, a1} coefficients. The output of the last EQ biquad is then scaled by the Post_EQ_Gain scale factor (specified in 6.18 format to allow for up to 30 dB of gain, i.e., scale factor of 32). The EQ-ed stereo pair output then forms the LR input for all further effects processing. The Equalization module affects the signal at the input, while Tone Control (refer to Section 4.5 "Generalized Tone Control Manager for Dolby Digital, DTS, MPEG Multichannel and C.O.S." on page 63) is designed to only boost the bass or treble just before the Bass Manager. The mono input for the surround processing is derived by mixing the input LR stereo pair with user parameters L_Mono_Mixlevel and R_Mono_Mixlevel. No internal protection or checking is applied to prevent overflow. This is also true for all other summing stages in the application. Therefore, the user settings should ensure that the sum of the magnitudes of summing mix levels does not exceed unity.
10.1.2 Basic Crystal Surround Effects Explanation
Crystal Original Surround Effects is based on public-domain research on acoustic reverberation. For a more detailed discussion of the properties of the Cirrus Logic implementation, see "About This Reverberation Business", James A. Moorer, Computer Music Journal 3(2): 13-28, 1979. Figure 26 is an illustration of basic reverb theory. In this figure, E-R should be interpreted as Early Reflections, while A-R should be interpreted as
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In addition, an impulse generator (not shown in Figure 27) is also provided for testing purposes. In this mode, the input L/R stereo pair is zeroed out, and a unit amplitude impulse is inserted in both input channels every 24000 samples (half a second at Fs = 48 KHz). The LPF at the input of the FIR delay structure is a first-order filter with a corner frequency of Fs/6, i.e., 8 KHz for Fs = 48 KHz and 7.35 KHz for Fs = 44.1 KHz. The variable mix levels, Direct_FIR_Mixlevel and LPF_FIR_Mixlevel allow one to select the amount of high frequencies to be passed into the FIR stage of the surround processor. This can be used to vary the amount of "brightness" of the surround effect. The Delay Buffer can hold up to 5616 samples which corresponds to 117 mS at Fs = 48 KHz, and 127.35 mS at Fs = 44 KHz. Thirty two userspecifiable taps T0-T31 are available from this delay buffer. The settings T0-T31 specify the absolute number of samples of delay (0... 5615). The tap delays need not be in monotonic or any specific ordering, and can be arbitrarily sequenced. For example, T0=4000 samples, T1 = 300 samples, T31 = 0 samples are valid settings. An FIR filter can be constructed using these delay buffer taps with coefficients C0-C31 corresponding to the taps T0-T31. This FIR filter can be used to model the Early Reflections of a particular room. The FIR output is available to the Output Channel Mixer directly. Tap 31 is available as an input to the reverberator for further surround effects processing. The mono input to the Reverberator Module is derived by mixing the output of Tap 31 of the delay buffer and the FIR output using user-specified parameters, Direct_Reverb_Mixlevel and FIR_Reverb_Mixlevel respectively. Note that Tap 31 defaults to a delay of 0, i.e., the direct output of Variable LPF structure. Tap 31 can be adjusted to align the impulse response of the Reverberator Module (After Reflections) with the end of the Early Reflections produced by the FIR filter. The Reverberator Module accepts user settings for Reverb_Time and Reverb_Liveness, which are both unitless constants. A larger value of Reverb_Time increases the duration of the reverberation effect, i.e., the period of sustained echo. A larger value of Reverb_Liveness decreases the high frequency content (increased high-frequency attenuation) in the reverberation. The Reverberator Module produces 5 outputs to the Output Channel Mixer, one for each output full-bandwidth channel. A more detailed treatment of the advanced settings possible in the reverberator, as well as information on creating custom effects modes is presented in Section 10.2.2. Figure 28 illustrates the Output Channel Mixer
Note: "Mix level" abbreviated as "Mlv.'" due to lack of space.
Each of the five full-bandwidth output channels (L, C, R, Ls, and Rs) is derived as a user-specifiable mix of the input L/R stereo pair, the FIR output, and the corresponding output channel of the Reverberator Module. The Low Frequency Effects (LFE) channel output carries no surround processing, and is only a user-specifiable mix of the input L/R stereo pair. Disabling the surround effects can be used for simple PCM pass through.
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Figure 26. Illustrated Basic Reverb Theory
FIR L
o
Direct_FIR_Mixlevel C0 C1
FIR_Reverb_Mixlevel
Reverb_Time, Reverb_Liveness, and Advanced Settings
L_Mono_ Mixlevel
...
C30
C31
LPF
R_Mono_ Mixlevel LPF_FIR_ Mixlevel
T0
T1
... Delay ... T30
Buffer
T31
Reverberator Module
o
R
Direct_Reverb_Mixlevel
Reverb_L
Reverb_R
Reverb_RS
Reverb_C
Reverb_LS
Figure 27. Block Diagram of Effects Code
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L R FIR
L_L_Out_Mlv.
o
R_L_Out_Mlv.
L L_Out R FIR
L_C_Out_Mlv.
o
R_C_Out_Mlv.
o
FIR_L_Out_Mlv.
o
FIR_C_Out_Mlv.
C_Out
Reverb_L_Out_Mlv.
Reverb_C_Out_Mlv.
Reverb_L
L_R_Out_Mlv.
Reverb_C
L_Ls_Out_Mlv.
L R FIR
o
R_R_Out_Mlv.
L R_Out R FIR
o
R_Ls_Out_Mlv.
o
FIR_R_Out_Mlv.
o
FIR_Ls_Out_Mlv.
Ls_Out
Reverb_R_Out_Mlv.
Reverb_Ls_Out_Mlv.
Reverb_R
Reverb_Ls
L R FIR
L_Rs_Out_Mlv.
o
R_Rs_Out_Mlv.
o
FIR_Rs_Out_Mlv.
Rs_Out
L R
L_LFE_Out_Mlv.
o o
R_LFE_Out_Mlv.
LFE_Out
Reverb_Rs_Out_Mlv.
Reverb_Rs
Figure 28. Block Diagram of Effects Channel Mixer
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10.2 C.O.S. PCM Processor
Write Opcode = 0x9C; Read Opcode = 0x1D; Read Response Opcode = 0x9D Write = 0x9C00HH 0xhhhhhh Read Request = 0x1D00HH; Read Response = 0x9D00HHh 0xHH = index 0xhhhhhh = data value 0xhhhhhh
Index
0x00
Variable
PCM_CONTROL
Dataword Content
Bit 4 = PCM_Input_EQ_Enable = 0/1 = Disable/Enable 3-band EQ filters on input PCM. Bit 0 = PCM_Divider_FIR_Enable = 0/1 = Disable/Enable FIR filter on PCM output of Ls/Rs channels.
0x010x10 0x11
Default = 0x000000 DIV_C0, DIV_C1, DIV_C15 -1.0 to 1.0 = Coefficients for each of the 16 taps of the FIR filter. Default* DIV_C0 = 1.0 PRE_EQ_ATTENUATION Default* DIV_C1-15 = 0.0 -1.0 to 1.0 = Pre-EQ attenuation scale factor to setup desired headroom for EQ filtering. Default* = 1.0 -4.0 to 4.0 in 3.21 format = Biquad coefficient for EQ Filter 0. Default* = 1.0 (0x200000) -4.0 to 4.0 in 3.21 format = Biquad coefficient for EQ Filter 0. Default* = 0.0 -4.0 to 4.0 in 3.21 format = Biquad coefficient for EQ Filter 0. Default* = 0.0 -4.0 to 4.0 in 3.21 format = Biquad coefficient for EQ Filter 0. Default* = 0.0 -4.0 to 4.0 in 3.21 format = Biquad coefficient for EQ Filter 0. Default* = 0.0 -4.0 to 4.0 in 3.21 format = Biquad coefficient for EQ Filter 1. Default* = 1.0 (0x200000) -4.0 to 4.0 in 3.21 format = Biquad coefficient for EQ Filter 1. Default* = 0.0 -4.0 to 4.0 in 3.21 format = Biquad coefficient for EQ Filter 1. Default* = 0.0 -4.0 to 4.0 in 3.21 format = Biquad coefficient for EQ Filter 1. Default* = 0.0 -4.0 to 4.0 in 3.21 format = Biquad coefficient for EQ Filter 1. Default* = 0.0 Table 43. C.O.S. PCM Processor
0x12 0x13 0x14 0x15 0x16 0x17 0x18 0x19 0x1a 0x1b
EQ0_B2 EQ0_B1 EQ0_B0 EQ0_A2 EQ0_A1 EQ1_B2 EQ1_B1 EQ1_B0 EQ1_A2 EQ1_A1
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Index
0x1c 0x1d 0x1e 0x1f 0x20 0x21 EQ2_B2 EQ2_B1 EQ2_B1 EQ2_A2 EQ2_A1 POST_EQ_GAIN
Variable
Dataword Content
-4.0 to 4.0 in 3.21 format = Biquad coefficient for EQ Filter 2. Default* = 1.0 (0x200000) -4.0 to 4.0 in 3.21 format = Biquad coefficient for EQ Filter 2. Default* = 0.0 -4.0 to 4.0 in 3.21 format = Biquad coefficient for EQ Filter 2. Default* = 0.0 -4.0 to 4.0 in 3.21 format = Biquad coefficient for EQ Filter 2. Default* = 0.0 -4.0 to 4.0 in 3.21 format = Biquad coefficient for EQ Filter 2. Default* = 0.0 -32.0 to 32.0 in 6.18 format = Post-EQ gain scale factor to compensate for Pre-EQ attenuation and EQ filter gains and maximize digital dynamic range usage. Default* = 1.0 (0x040000) Table 43. C.O.S. PCM Processor (Continued)
Notes:
See definition on page 32.
"Default" vs. "Default*" See definition page 32.
10.2.1 PCM_CONTROL
Bit 4: INPUT_PCM_EQ_ENABLE Switches ON/OFF EQ of input PCM. 1 - Enables Input PCM EQ 0 - Disables Input PCM EQ When EQ is enabled, stereo PCM input to the Effects Module (after de-emphasis if enabled) is first attenuated by the Pre_EQ_Attenuation scale factor (specified in usual 1.23 format), and then processed by the cascade of three EQ biquads specified by the {EQ0...2: b2, b1, b0, a2, a1} coefficients. The output of the last EQ biquad is then scaled by the Post_EQ_Gain scale factor (specified in 6.18 format to allow for up to 30 dB of gain, i.e., scale factor of 32). The EQ-ed stereo pair output then forms the LR input for all further effects processing.
Note: All EQ parameters should be downloaded by the host controller before kickstart.
y(n) = x(n)*b0 + x(n-1)*b1 + x(n-2)*b2 + y(n-1)*a1 + y(n-2)*a2
Note: The coefficients b2, b1, b0, a2, a1 are all specified in 3.21 format to allow for coefficient magnitudes of up to 8.0.
Bit 0: PCM_Divider_FIR_Enable Switches ON/OFF the PCM Divider FIR.
1 - Enables FIR Filter on Ls and Rs PCM output 0 - Disables FIR Filter
PCM produced for output at the Ls and Rs channels according to the Effects setup is passed individually through 2 identical 16-tap FIR filters (assuming Control above has been turned ON). This forms the input to the Bass Manager (if it is turned ON). For an input x(n), the output y(n) of the FIR filter is defined by:
y(n) = x(n)*DIV_C0 + x(n-1)*DIV_C1 + x(n2)*DIV_C2 ... + x(n-15)*DIV_C15
For an input x(n) the output y(n) of each EQ biquad filter is given by:
The coefficients DIV_C0...15 are set by the host, using messaging into the indices as specified above.
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10.2.2 Creating Custom Effects Using Advanced Settings
This section describes the Reverberator module. Figure 29 shows a block diagram of the internal implementation of the reverberator module. The coefficients G1 and G2 for a comb should satisfy G1+G2<1 for stability. The scheme for controlling the parameters in the CS493XX guarantees this. The individual default Comb_[05]_g1 can be changed. G1 and G2 are derived for each comb automatically as shown in Figure 29 using Comb_g and Comb_h as well as Comb_[05]_g1. Increasing (or decreasing) the Reverb_Time (Comb_g) parameter increases (or decreases) the length of the after-reflection tail since it scales up (or down) the feedback in the comb. Increasing (or decreasing) the Reverb_Liveness (Comb_h) parameter increases (or decreases) the amount of LPF action in the feedback, thus making the sound more (or less) dampened. This is particularly important to reduce the clicking effect when reverberating short impulsive sounds. While experimenting, it may be found that a particular comb's reverberation (among the six) is audibly predominant (buzzing sound). To overcome this, the corresponding Comb_[05]_Mixlevel for that comb can be reduced to equalize its audible effect with the other combs. These can also be used to overcome overflow issues that may happen with different choices of g1, g, and h.
10.2.2.1 FIR Section: Early Reflections
The FIR filter shown in Figure 29 allows one to simulate Early Reflections and model specific room acoustics. Given a particular set of FIR filter taps and coefficients (generated by actual room measurements and/or listening experiments), various sizes of rooms can be derived by simply scaling the tap locations (not the coefficients). Other effects can be generated by moving back all of the taps to simulate a delay in the Early Reflections. The amplitude of the coefficients can be scaled according to the desired liveness of the room. If the amplitude of the coefficients does not fall off quickly, then the room will appear more live (that is, later Early Reflections will be more present in the output).
10.2.2.2 Comb Filter Section: After-reflections
Following the Early Reflections simulated by the FIR, comb filters are used to simulate the dense After Reflections that are caused by walls and transitions thereof. Each comb filter reproduces the input signal with periodic delay and decaying amplitude. Furthermore, a first order low pass filter is included in the feedback to simulate the absorption of high frequencies by air. This produces a more realistic or `live' reverberation that sounds less `tinny'. Six combs are used in parallel to produce dense After Reflections. The delay sizes of the combs are chosen to be mutually prime to avoid coinciding echoes from different combs. It is recommended to set the delays of each comb as roughly 10% more than the previous one. However comb sizes with arrangements that are too regular can cause a buzzy output.
10.2.2.3 All-Pass Filter Section: More Afterreflections and Decorrelation
All-Pass filters are another means of simulating After Reflections. In addition, they are used here to decorrelate the different channel outputs from each other. In the CS493XX implementation, the delay sizes are fixed (as specified in Figure 29) and all of the All-Pass filters share the same values of G1 and G2. For stability, G2 < = 1 - G1 ^ 2 should be guaranteed by the host if these values are changed. Increasing (or decreasing) the G2 parameter increases (or decreases) the reverberation time (length of after-reflection tail) by scaling up (or down) the reverberated contribution to the output.
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Comb_[0-5]_Mixlevel
Comb_0 Comb_1
FIR + Direct Mono Mix
AllPass_g1, AllPass_g2
Comb_2 AP_Main Comb_3 Comb_4 Comb_5
AllPass Filters
o
AP_L AP_C
AP_R
AP_Rs
AP_Ls
Reverb_L
Reverb_Time (Comb_g), Reverb_Liveness (Comb_h), Comb_[0-5]_g1, Comb_[0-5]_Delay_Size
Reverb_R
Reverb_RS
Reverb_C
Reverb_LS
Comb Filter Module
x(n) y(n)
+ + G2 +
AllPass (AP) Filter Module o
+ G1 G2 + +
o
z^(-Delay_Size)
x(n)
y(n)
o
+ +
z^(-Delay_Size)
o
z^-(1)
G1 = Comb_h * Comb_g1 G2 = Comb_g * (1 - G1) Delay_Size = Comb_Delay_Size
G1
G1 G1 = AllPass_g1; G2 = AllPass_g2 Delay_Size = 293, 127, 139, 151, 163, 173 for Main, L, C, R, Ls, Rs, respectively.
Figure 29. Block Diagram of Effects Reverberator Module
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10.2.3 Crystal Original Surround Effects Modes
This section describes some standard effects modes that are provided along with the application microcode. Each mode is setup by issuing an Application Restart message or Soft Reset (assuming the microcode is already downloaded), sending any required hardware configuration messages, and finally sending the corresponding configuration message set for each surround effects mode. For each mode described, the configuration settings are given. Although one can assume the default values as listed in Table 42, the provided message sets explicitly set up all the parameters in most cases. Thus they can serve as templates for any future custom development by the user.
Note: Surround Effects parameters are sampled by the CS4932X only once just after kickstart (after a download/soft reset or Application Restart). These parameters cannot be varied at run-time. Therefore, to change modes, one has to issue a Soft Reset or Application Restart, reconfigure hardware settings, issue the new Effects messages and then kickstart.
increasing) the coefficients, the liveness of the room can be increased (or decreased). Scaling the tap locations will increase/decrease the length of Early Reflections. Delaying the taps (offsetting all taps by the same amount of time) causes a delay in the Early Reflections. A brief description of each surround effect mode is provided below. Note that the modes have been designed for Fs=48 KHz. When used for Fs=44.1 KHz, the change is largely inaudible and acceptable in most cases. If necessary, the user may scale the delay and reverb time parameters (if used) appropriately. The effects can be broken into two categories, reverberative and non-reverberative. Reverberative effects have both late and early reflections, while non-reverberative effects are composed entirely of early reflections. The next two sections list the different configuration examples sent with the code. The examples are categorized by whether the effect is reverberative or not. Non-reverberative effects are much more subtle, while reverberative effects are more extreme and can sometimes interfere with the actual source material. The naming of each mode is very subjective and is only intended as a suggestion. Section 10.3.1 "COEFMAKR.EXE and EFFECTS.EXE" on page 120 describes a program that can create effects configuration files for the user.
The included effects modes were all derived from one of two basic filter structures: A basic 7-point FIR filter with taps at: T0-6 = 0, 955, 1699, 1867, 1987, 3355, 3821 samples. The corresponding coefficients are C0-6 = 0.9804, 1.0, 0.802, 0.6225, 0.7049, 0.2618, and 0.2373. A 19-point FIR filter with taps at: T0-18 = 0, 206, 1032, 1080, 1286, 1296, 1430, 2198, 2328, 2746, 2818, 2856, 2938, 3394, 3398, 3485, 3557, 3614, and 3826 samples. The corresponding coefficients are C0-18 = 1.0, 0.841, 0.504, 0.491, 0.379, 0.380, 0.346, 0.289, 0.272, 0.192, 0.193, 0.217, 0.181, 0.180, 0.181, 0.176, 0.142, 0.167, and 0.134. By scaling these taps and coefficients, different effects can be created. By decreasing (or
10.2.4 Non - Reverberative Effects
Non-reverberative effects provide a subtle effect through remapping and/or redirecting the stereo input. In this way, the end user can fill all of the speakers in a surround system without significantly altering the content of the source. Delays are used in some modes to simulate a larger listening area.
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10.2.4.1 Passthru
This mode does not engage the Effects Manager and thus has no reverberation or Early Reflections. The default setup of Output Channel Mixer is used, i.e., L_Out = L R_Out = R C_Out = Ls_Out = Rs_Out = LFE_Out = 0.5 L + 0.5 R Ls_Out = Rs_Out = FIR LFE_Out = 0
10.2.4.5 Movie 1-4
These modes have no reverberation and only use FIR and the Output Channel Mixer. Each mode has a different FIR delay tap/coefficient setting, but all have the following generic setup. The mono mix is set up as (L+R)/2 with no LPF contribution and feeds FIR, a 4 or 5-tap filter depending on the mode. The Output Channel Mixer is setup for: L_Out = L R_Out = R C_Out = Ls_Out = Rs_Out = FIR LFE_Out = 0
10.2.4.2 Mono
This mode has no reverberation or Early Reflections and only uses the Output Channel Mixer. The Output Channel Mixer is setup for: L_Out = R_Out = C_Out = Ls_Out = Rs_Out =0.5 L + 0.5 R LFE_Out = 0
10.2.4.3 Classical
This mode has no reverberation and only uses FIR and the Output Channel Mixer. The mono mix is set up as (L+R)/2 with no LPF contribution and feeds FIR, a 5-tap filter. The Output Channel Mixer is setup for: L_Out = L R_Out = R C_Out = Ls_Out = Rs_Out = FIR LFE_Out = 0
10.2.4.6 Music 1-4
These modes are identical to Movie1-4 modes above, except that the Output Channel Mixer is set up for C_Out = 0.
10.2.5 Reverberative Effects
Reverberative effects combine the different filters discussed in sections Section 10.1.2 "Basic Crystal Surround Effects Explanation" on page 108, and in Section 10.2.2 "Creating Custom Effects Using Advanced Settings" on page 114 to create a more obvious effect. These effects can be used to fill out the incoming source if so desired. Care should be taken as some reverberative effects can be so strong that they can interfere with the original stereo source signal. The effects listed below start with the subtle effects (Room) and graduate to the very obvious (Stadium). To simplify the use of the following configuration files, the following naming convention has been used: tErGaKiN.cfg E - This number represents the FIR filter type used for Early Reflections. Type 1 is a 7-Tap, and Type 2 is a 19-Tap, per the discussion in the previous
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This mode has no reverberation and only uses FIR and the Output Channel Mixer. The mono mix is set up as (L-R)/2 with no LPF contribution and feeds FIR, the same 5-tap filter as in Classical mode. The Output Channel Mixer is setup for: L_Out = L - 0.31623 R R_Out = R - 0.31623 L C_Out = 0.5 L + 0.5 R
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section. The 19-Tap filter will make the Early Reflections sound much thicker. This is useful when trying to create an echo pattern that sounds thick. G - This represents the amount of Initial Delay and Early Reflection time. A shorter Early Reflection time will make the room size sound smaller, and a longer time will make the room sound larger. Initial Delay until Early Reflections will cause the room to sound as if the source is further from the walls. G == 1: Early Reflections last for 40 ms, and the Initial Delay until Early Reflections is 20 ms. G == 2: Early Reflections last for 80 ms (default), and the Initial Delay until Early Reflections is 10 ms. G == 3: Early Reflections last for 115 ms, and there is no Initial Delay until Early Reflections (0 ms). K - This represents the amount of time for After Reflections. The After Reflections produced by Reverb Module will produce a dense tail to the effect, and this parameter will influence the length of that tail. K == 1: Reverb time is set to 0.25 ms. K == 2: Reverb time is set to 0.5 ms(default). K == 3: Reverb time is set to 0.8 ms. K == 4: Reverb time is set to 0.98 ms. This will produce a reverb time that is very large.
Note: Only one example with K=4 is presented here (Stadium).
For N == 2:Reverb Liveness (After Reflection Liveness) is set to 0.8 To simplify the use of the filters, a description of some of the filters follows. Note that configuration files are supplied for many more variations of the parameters specified above. The supplied configuration files are intended to span a range of effects that can be created using the coefmakr.exe program. To create intermediate sounding effects, one could simply interpolate the parameters specified to the coefmakr.exe program and use the corresponding configuration files. Section 10.3.1 "COEFMAKR.EXE and EFFECTS.EXE" on page 120 has more details. For all Reverberative Modes, the following mixlevels (default in coefmakr.exe) are used: L_Mono_Mixlevel = 0.5 R_Mono_Mixlevel = 0.5 Direct_FIR_Mixlevel = 0.875 LPF_FIR_Mixlevel = 0.125 Reverb_Direct_Mixlevel = 1 Reverb_FIR_Mixlevel = 0 L_to_L_Out = 0.875 R_to_L_Out = 0 L_to_C_Out = 0.4375 R_to_C_Out = 0.4375 L_to_R_Out = 0 R_to_R_Out = 0.875 L_to_Ls_Out = 0.125 R_to_Ls_Out = 0.125 L_to_Rs_Out = 0.125 R_to_Rs_Out = 0.125 L_to_LFE_Out = 0.5 R_to_LFE_Out = 0.5
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N - This represents the After Reflection (Reverb) Liveness. To absorb more of the high frequency content of the After Reflections (to sound less bright), the number should be higher. To increase the high frequency content (to sound more bright), the number should be lower. For N == 1:Reverb Liveness (After Reflection Liveness) is set to 0.2
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FIR_to_L_Out = 0.0833 Reverb_to_L_Out = 0.0417 FIR_to_C_Out = 0.0417 Reverb_to_C_Out = 0.0213 FIR_to_R_Out = 0.0833 Reverb_to_R_Out = 0.0417 FIR_to_Ls_Out = 0.5 Reverb_to_Ls_Out = 0.25 FIR_to_Rs_Out = 0.5 Reverb_to_Rs_Out = 0.25 and the high frequencies will be present in the After Reflections.
10.2.5.6 t2r3a3i2.cfg -- Hall 2
This effect, similar to Hall1, has the same amount of Early and Late Reflections. In this effect mode, the higher frequencies are less present, resulting in a less bright sound in the reflections.
10.2.5.7 t2r3a4i2.cfg -- Stadium
This effect, utilizing a maximum of Early and Late Reflections, has a long decay time. The sound is highly decorrelate, making the echoes sound muddy. The long decay time provides for an effect of the sound bouncing off many hard walls, and a very bright tone is used. This effect is presented merely to demonstrate the extreme capability of the Crystal scheme, and may require modification before use in a real system that you develop.
10.2.5.1 t1r1a1i1.cfg - Room 1
This effect mode simulates the reflections from a small room. The room is not live, and has small Early and Late Reflection Times. The type 1 filter is used, providing a very mild effect.
10.3 Effects Configuration Files
COEFMAKR.EXE is a program written to assist development of unique effects configurations. It is documented in Section 10.3.1 "COEFMAKR.EXE and EFFECTS.EXE" on page 120. This program can be used to create configuration files for the effects modes presented in the previous sections. The configuration files have been omitted from this document to conserve space. The COEFMAKR.EXE program along with the configuration files are in a zip file available at www.cirrus.com under the CS49300 page. The configuration file contains paired 3-byte words which are the actual commands to be sent to the CS49300. These messages correlate with the parameters in Table 42. In addition comment lines marked by a # symbol are included to help explain the configuration messages. When porting the messages to code for a host microcontroller, the comment lines should be omitted and care should be taken to keep the paired 3- byte words together in the correct order. The configuration files also contain a "Kickstart" as the last command (see Section 4.1 "Audio Manager for Dolby Digital, DTS, MP3, MPEG Multichannel, AAC and Crystal
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10.2.5.2 t2r3a1i2.cfg - Room 2
This effect mode simulates the reflections from a small room, but uses a larger FIR filter spread out over a larger time to thicken the sound of the Early Reflections.
10.2.5.3 t1r1a2i1.cfg - Theater 1
This effect mode simulates the reflections as in a medium size theater. The Early Reflections will be short, but very present. The Late Reflections are bright, but are present for medium amount of time.
10.2.5.4 t2r3a2i2.cfg -- Theater 2
The Early Reflections in this effect are dominant and long, allowing for a very thick sounding Early Reflection. This effect mode has a short Reverb Time but large Reverb Liveness, making the sound less bright.
10.2.5.5 t1r1a3i1.cfg -- Hall 1
This effect has a lot of Early and Late Reflections, with a large Reverb Time, but a small Reverb Liveness. This effect will have a very large sound,
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Original Surround" on page 33 for more information). Here is a short description of the files and their contents: * COEFMAKR.EXE: This file is the main program for the coefficient maker. * FIR.CFG: This file contains the templates for 2 basic FIR (Early Reflection) filters, which are scaled by COEFMAKR.EXE into the FIR section of the effects code. Additional filters can be added to this file, or the file can be replaced by one of the same structure. However, if a different file is used, it must be specified in the command line (-c option). For COEFMAKR.EXE to run correctly, this file must reside in the same directory. MAKETEST.BAT: This batch file will generate all of the default configuration files shown in section (with reverb) using COEFMAKR.EXE. EFFECTS.EXE: COEFMAKR.EXE GUI version of
10.3.1 COEFMAKR.EXE and EFFECTS.EXE
COEFMAKR.EXE (Coefficient Maker) is a simple, easy-to-use program for generating configuration files used in conjunction with Crystal Effects code for the CS49300 series of Surround Sound DSP's. The program takes all of the configuration options on the command line, and all of these options can be displayed with the -? command line switch. Note that since there are 56 command line options (a minimum of 1 must be specified), to display Mix Level options, the -x? switch must be used. For all Advanced Options, the -o? switch must be used. EFFECTS.EXE is a GUI version of COEFMAKR.EXE which allows the user to edit effects configuration files and try them on one of the CRD49300/CDB49300 application boards in real time. The program mirrors the block diagrams shown in Figure 27, Figure 28, and Figure 29 and allows the user control over the parameters discussed in section.
*
*
More files may need to be present. Please read the README.TXT file in the zip file from which the programs were extracted for more information on these programs.
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10.3.2 Command Line Options for COEFMAKR.EXE
Basic Options Settings: ---------Usage: coefmakr [-qfile.out -cfilter.cfg -fx -dy -tz -rw -lb -mu -v -xelevel] -qfile.out : file.out = Config file to be generated(default=file.out) -cfilter.cfg : filter.cfg = Filter parameter file(default=fir.cfg) -fb : b = Sampling Rate (32, 44, 48* KHz) -du : u = Initial Delay of Early Reflections (0*,1,...,115ms) -te : e = FIR Filter Type (1* or 2) -rg : g = Early Reflection Time (0,1,...,80*,... ms) -lh : h = Liveness of Early Reflection (0,...,1*) -mj : j = Delay Until After Reflections (0,1,...,100*,...ms) -ak : k = After Reflection Time (0,...,0.5*,...,1) -in : n = After Reflection Liveness (0,...,1*) -xp : p = Mix Level(s) to Change (? for options) -os : s = Advanced Options (? for options) -v = Run in non-verbose mode -x Options: ----------Settings for Mix level: a = L_Mono_Mixlevel(-1..0.5*..1) b = R_Mono_Mixlevel(-1..0.5*..1) c = Direct_FIR_Mixlevel(-1..0.875*..1) d = LPF_FIR_Mixlevel(-1..0.125*..1) e = Reverb_Direct_Mixlevel(-1..0.875*..1) f = Reverb_FIR_Mixlevel(-1..0*..1) g = L_to_L_Out(-1..0.875*..1) h = R_to_L_Out(-1..0*..1) i = L_to_C_Out(-1..0.4375*..1) j = R_to_C_Out(-1..0.4375*..1)
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k = L_to_R_Out(-1..0*..1) l = R_to_R_Out(-1..0.875*..1) m = L_to_Ls_Out(-1..0.125*..1) n = R_to_Ls_Out(-1..0.125*..1) o = L_to_Rs_Out(-1..0.125*..1) p = R_to_Rs_Out(-1..0.125*..1) q = L_to_LFE_Out(-1..0.5*..1) r = R_to_LFE_Out(-1..0.5*..1) s = FIR_to_L_Out(-1..0.0625*..1) t = Reverb_to_L_Out(-1..0.0625*..1) u = FIR_to_C_Out(-1..0.03125*..1) v = Reverb_to_C_Out(-1..0.03125*..1) w = FIR_to_R_Out(-1..0.0625*..1) x = Reverb_to_R_Out(-1..0.0625*..1) y = FIR_to_Ls_Out(-1..0.375*..1) z = Reverb_to_Ls_Out(-1..0.375*..1) 1 = FIR_to_Rs_Out(-1..0.375*..1) 2 = Reverb_to_Rs_Out(-1..0.375*..1) -o Options: ----------Usage of Advanced Options Settings: a = Comb_0_Delay_Size(1......1103*) b = Comb_1_Delay_Size(1......1213*) c = Comb_2_Delay_Size(1......1321*) d = Comb_3_Delay_Size(1......1433*) e = Comb_4_Delay_Size(1......1543*) f = Comb_5_Delay_Size(1......1657*) g = Comb_0_g1(0...0.0895*...1) h = Comb_1_g1(0...0.0950*...1) i = Comb_2_g1(0...0.1002*...1)
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j = Comb_3_g1(0...0.1055*...1) k = Comb_4_g1(0...0.1106*...1) l = Comb_5_g1(0...0.1160*...1) m = Comb_0_Mixlevel(0...0.9*...1) n = Comb_1_Mixlevel(0...0.9*...1) o = Comb_2_Mixlevel(0...0.9*...1) p = Comb_3_Mixlevel(0...0.9*...1) q = Comb_4_Mixlevel(0...0.9*...1) r = Comb_5_Mixlevel(0...0.9*...1) s = Allpass_g1(0...0.7*...1) t = Allpass_g2(0...0.51*...1) The following are the command line parameters to use with the COEFMAKR.EXE program to create the configuration files for the effects modes presented in Section 10.2.5.1 "t1r1a1i1.cfg - Room 1" on page 119 through Section 10.2.5.7 "t2r3a4i2.cfg -- Stadium" on page 119. coefmakr -qt1r1a1i1.cfg -r40 -d20 -a0.25 -v coefmakr -qt2r3a1i2.cfg -t2 -r115 -i0.6 -v coefmakr -qt1r1a2i1.cfg -r40 -d20 -v coefmakr -qt2r3a2i2.cfg -t2 -r115 -i0.6 -v coefmakr -qt1r1a3i1.cfg -r40 -d20 -a0.8 -v coefmakr -qt2r3a3i2.cfg -t2 -r115 -a0.8 -i0.6 -v coefmakr -qt2r3a4i2.cfg -r115 -l0.5 -m120 -a0.97 -i0.8 -v
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