Professional Documents
Culture Documents
Harris Corporation Broadcast Communications Division 4393 Digital Way Mason, OH USA 45040
Copyright 2012, Harris Corporation, 1025 West NASA Boulevard, Melbourne, Florida 32919-0001 U.S.A. All rights reserved. This publication supersedes all previous releases. No part of this documentation may be reproduced in any form or by any means or used to make any derivative work without permission from Harris Corporation. Harris Corporation reserves the right to revise this documentation and to make changes in content from time to time without obligation on the part of Harris Corporation to provide notification of such revision or change. UNITED STATES GOVERNMENT LEGEND If you are a United States government agency, then this documentation and the software described herein are provided to you subject to the following: All technical data and computer software are commercial in nature and developed solely at private expense. Software is delivered as Commercial Computer Software as defined in DFARS 252.227-7014 (June 1995) or as a commercial item as defined in FAR 2.101(a) and as such is provided with only such rights as are provided by Harris standard commercial license for the Software. Technical data is provided with limited rights only as provided in DFAR 252.227-7015 (Nov 1995) or FAR 52.227-14 (June 1987), whichever is applicable. You agree not to remove or deface any portion of any legend provided on any licensed program or documentation contained in, or delivered to you in conjunction with, this User Guide. This publication, or any part thereof, may not be reproduced in any form, by any method, for any purpose, without the written consent of Harris Corporation. Contact Harris Corporation for permission to use materials as well as guidelines concerning foreign language translation and publication. Harris Corporation reserves the right to revise and improve its products as it chooses. This publication is designed to assist in the use of the product, as it exists on the date of publication of this manual, and may not reflect the product at the current time or an unknown time in the future. This publication does not in any way warrant description accuracy or guarantee the use for the product to which it refers. The Harris logo and assured communications are registered trademarks of Harris Corporation. D-Series is a trademark of Harris Corporation. All other trademarks are held by their respective owners. This user guide was created for APM6803+, Edition B. Windows is a registered trademark of Microsoft Corporation. AMD and Operton are trademarks of Advanced Micro Devices, Inc. Dolby and the double-D symbol are registered trademarks of Dolby Laboratories. DTS Neural audio products are manufactured under license from DTS Licensing Limited. DTS and the Symbol are registered trademarks & the DTS Logos are trademarks of DTS, Inc. 2008-2010 DTS, Inc. All Rights Reserved. Java is a trademark of Sun Microsystems, Inc. or its subsidiaries in the United States and other countries. All other trademarks are the property of their respective holders. Publication Date: February 2012
iii
Contents
Preface ......................................................................................................................... ix
Manual Information ...................................................................................................... ix Purpose ..................................................................................................................... ix Audience ................................................................................................................... ix Revision History ......................................................................................................... ix Writing Conventions ...................................................................................................x Obtaining Documents .................................................................................................x Unpacking/Shipping Information .................................................................................x Unpacking a Product ..................................................................................................x Product Servicing ........................................................................................................x Returning a Product ................................................................................................... xi Safety ............................................................................................................................. xi Safety Terms and Symbols in this Manual .................................................................. xi References ..................................................................................................................... xi Restriction on Hazardous Substances (RoHS) Compliance ....................................... xiii Waste from Electrical and Electronic Equipment (WEEE) Compliance .................... xiv
Chapter 1
Introduction.............................................................................................................. 1
Overview ....................................................................................................................... 1 Product Overview ......................................................................................................... 2 Architecture .............................................................................................................. 3 Virtual Stream Interface ............................................................................................. 3 Intelligent Metadata Handler ..................................................................................... 3 Resource Utilization ................................................................................................... 4 Licensing ................................................................................................................... 4 Real Time Loudness Control ....................................................................................... 4 Automation Control .................................................................................................. 4 Main Features ............................................................................................................ 4 Monitoring and Operating ......................................................................................... 6 Components .................................................................................................................. 6 Main Module ............................................................................................................. 7 Sub Module ............................................................................................................... 8 Riser Board ................................................................................................................ 8 Standard Back Module .............................................................................................. 9 Relay Back Module .................................................................................................. 10
Copyright 2012, Harris Corporation
iv
Contents
Breakout Cable ........................................................................................................ 11 Pinouts for 44-pin Connector ................................................................................... 12 Serial Port (RS-422/232) Connector .......................................................................... 14 General Purpose Interface (GPI) Connectors ............................................................. 15 Signal Flow .............................................................................................................. 15
Chapter 2
Installation .............................................................................................................. 19
Overview ...................................................................................................................... 19 Unpacking the Module ............................................................................................... 19 Packing List .............................................................................................................. 20 Selecting an External Balun ...................................................................................... 20 Main Module Jumpers ................................................................................................ 21 Sub Module Jumpers ................................................................................................... 22 Maximum 6800+ Frame Power Ratings ...................................................................... 23 Installing 6800+ Modules ............................................................................................ 24 Required Frames and Back Connector Types ............................................................ 24 Installing and Removing APM6803+ Modules .......................................................... 24 Relay Back Module ...................................................................................................... 25 Upgrading Module Firmware ..................................................................................... 26 License Options ........................................................................................................... 26
Chapter 3
Chapter 4
Audio Configuration Examples ................................................................................ 39 Resource Utilization .................................................................................................... 40 AAP Status Parameter .............................................................................................. 40 Processor Utilization (DSP and AAP) ......................................................................... 41 Advanced Audio Processing Utilization and Delay .................................................. 42
Chapter 5
Chapter 6
Chapter 7
vi
Contents
Stereo Mixing .......................................................................................................... 87 V-bit & Non-PCM Auto Correction ........................................................................... 87 Voice Over Functionality ........................................................................................... 88 Video Processing ......................................................................................................... 90 Video Frame Synchronization ................................................................................... 90 Video Delay Ranges ................................................................................................. 92 Auto Match Audio/Video Delay ................................................................................ 92 Audio/Video Align .................................................................................................... 93 AFD/WSS/VI Detection and Insertion ........................................................................ 94 Thumbnail Display .................................................................................................... 95 Fast Video Switch ..................................................................................................... 95 GPI Configuration ........................................................................................................ 95 GPI Input Configuration ........................................................................................... 95 GPI Output Configuration ........................................................................................ 97 Custom GPI Configuration ....................................................................................... 97
Chapter 8
Chapter 9
vii
Chapter 10
Chapter 11
Appendix A
Appendix B
viii
Contents
ix
Preface
Manual Information
Purpose
This manual details the features, installation, operation, maintenance, and specifications for the APM6803+ Multichannel Audio Processing Station.
Audience
This manual is written for engineers, technicians, and operators responsible for installation, setup, maintenance, and/or operation of the APM6803+ Multichannel Audio Processing Station.
Revision History
A B
Initial release. Added the following: New relay back module option (See Relay Back Module) Dolby E and Dolby Digital Encoder and Decoder functionality (See Dolby Processing) Custom GPI and Parameter Scripts (See Custom Scripting) Voiceover functionality (See Voice Over Functionality) Surround Sound Presets (See DTS Neural UpMix Presets, DTS Neural DownMix Presets, and DTS Neural MultiMerge Presets) Dialnorm Translator (See Dialnorm Translator)
Preface
Writing Conventions
This manual adheres to the following writing conventions. Table P-2. Writing Conventions Term or Convention Bold Description Indicates dialog box, property sheet, field, button, check box, list box, combo box, menu, submenu, window, list, and selection names Indicates email addresses, names of books and publications, and first instances of new terms and specialized words that need emphasis Indicates a specific key on the keyboard, such as ENTER, TAB, CTRL, ALT, DELETE Indicates variables or command-line entries, such as a DOS entry or something you type into a field. Indicates the direction of navigation through a hierarchy of menus and windows. Indicates a jump to another location within the electronic document or elsewhere Indicates a jump to a Web site or URL Indicates important information that helps to avoid and troubleshoot problems
Italics
CAPS
Code
> hyperlink Internet address Note:
Obtaining Documents
Product support documents can be viewed or downloaded from our website. Alternatively, contact your Customer Service representative to request a document.
Unpacking/Shipping Information
Unpacking a Product
1 2 3 4 5 This product was carefully inspected, tested, and calibrated before shipment to ensure years of stable and trouble-free service. Check equipment for any visible damage that may have occurred during transit. Confirm that you have received all items listed on the packing list. Contact your dealer if any item on the packing list is missing. Contact the carrier if any item is damaged. Remove all packaging material from the product and its associated components before you install the unit. Keep at least one set of original packaging, in the event that you need to return a product for servicing.
Product Servicing
Except for firmware upgrades, APM6803+ modules are not designed for field servicing. All hardware upgrades, modifications, or repairs require you to return the modules to the Customer Service center.
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Returning a Product
In the unlikely event that your product fails to operate properly, please contact Customer Service to obtain a Return Authorization (RA) number, and then send the unit back for servicing. Keep at least one set of original packaging in the event that a product needs to be returned for service. If the original package is not available, you can supply your own packaging as long as it meets the following criteria: The packaging must be able to withstand the products weight. The product must be held rigid within the packaging. There must be at least 2 in. (5 cm) of space between the product and the container. The corners of the product must be protected. Ship products back to us for servicing prepaid and, if possible, in the original packaging material. If the product is still within the warranty period, we will return the product prepaid after servicing.
Safety
Carefully review all safety precautions to avoid injury and prevent damage to this product or any products connected to it. If this product is rack-mountable, it should be mounted in an appropriate rack using the rack-mounting positions and rear support guides provided. It is recommended that each frame be connected to a separate electrical circuit for protection against circuit overloading. If this product relies on forced air cooling, it is recommended that all obstructions to the air flow be removed prior to mounting the frame in the rack. If this product has a provision for external earth grounding, it is recommended that the frame be grounded to earth via the protective earth ground on the rear panel. IMPORTANT! Only qualified personnel should perform service procedures.
References
AES3-2003 AES Recommended Practice for Digital Audio I/O Interfacing Serial Transmission Format for Two-Channel Linearly Represented Digital Audio Data
xii
Preface References
ANSI/SMPTE 259M-2006 SDTV Digital Signal/Data Serial Digital Interface ANSI/SMPTE 276M-1995 Transmission of AES/EBU Digital Audio Signal Over Coaxial Cable SMPTE 292M-2006 1.5 Gb/s Signal/Data Serial Interface SMPTE 291M-2006 Ancillary Data Packet and Space Formatting SMPTE 337M-2000 Format for Non-PCM Audio and Data in an AES3 Serial Digital Audio Interface SMPTE 346-M 2000 Time Division Multiplexing Video Signals and Generic Data over High-Definition Interface SMPTE 352-M 2002 Video Payload Identification for Digital Interfaces SMPTE 424-M 2006 3Gb/s Signal/Data Serial Interface SMPTE 425-M 2006 3Gb/s Signal/Data Serial InterfaceSource Image Format Mapping SMPTE 2020-1-2008 Format of Audio Metadata and Description of the Async Serial Bitstream Transport. SMPTE 2020-2-2008 Vertical Ancillary Data Mapping of Audio Metadata - Method A SMPTE 2020-3-2008 Vertical Ancillary Data Mapping of Audio Metadata - Method B SMPTE RP 184-2004 Specification of Jitter in Bit-Serial Digital Systems EIA/TIA-422-B 1994 Electrical Characteristics of Balanced Voltage Digital Interface Circuits EN55103-1 EMC emission requirements applies to professional audio, video, audio-visual and entertainment lighting control apparatus EN55103-2 EMC immunity requirements applies to professional audio, video, audio-visual and entertainment lighting control apparatus ITU-R BT.601-5 Studio Encoding Parameters of Digital Television for Standard 4:3 and Wide-Screen 16:9 Aspect Ratios ITU-R BT.709-5 Parameter Values for the HDTV Standards for Production and International Programme Exchange
Copyright 2012, Harris Corporation
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RDD 06 - 2008 Description and Guide to the Use of the Dolby E Audio Metadata Serial Bitstream RDD 19 - 2011 Guidelines on the Use of Dolby E with Video Signals at Frame Rates Greater than 30 Hz TIA/EIA-232-E 1991 Interface Between Data Terminal Equipment and Data Circuit-Terminating Equipment Employing Serial Binary Data Interchange 47 Code of Federal Regulations Part 15 FCC rulesRadio Frequency Devices
xiv
Preface References
1
Overview
Introduction
The following is covered in this chapter: Product Overview on page 2 Architecture on page 3 Resource Utilization on page 4 Licensing on page 4 Main Features on page 4 Monitoring and Operating on page 6 Components on page 6 Signal Flow on page 15
Chapter 1 Introduction
Product Overview
SDI Metadata Mux EXT REF FRAME REF Genlock Audio Mux AFD/WS S/VI Demux Audio Demux 16 x 1 Router 16 x 1 Router Dolby Decoder 1 Dolby Decoder 2 SRC Audio Router Virtual Out 1-4 Audio Router Virtual In 1-8 Audio Gain Virtual In 1-8 8 x 1 Router
Surround Processor 1 (UpMix, DownMix, MultiMerge) Audio Gain Audio Gain Audio Gain Audio Gain
bypass bypass
AFD/WSS/VI Mux
Video Proc
VBI Delete
Deserialize
Serialize
SDI OUT 1
SDI IN
SDI OUT 2
LC 1
bypass
Dolby Encoder 1
bypass
SDI OUT 3
SRC
LC 2
bypass
AES IN 1
AES IN 8
8 x 1 Router
...
Input Tones
LC 3
bypass
LC 4
Output Router
Dolby Encoder 2
SDI OUT 4
Output Tones
...
Monitor 1 Generator 1 Generator 2 SERIAL IN 1/2 Metadata Router MetaSmart 1 MetaSmart 2 Metadata Router Monitor 2
SERIAL OUT 1 SERIAL OUT 2
Surnd Processors and LCs same function blocks as above. GPI IN 1 GPI IN 12
...
Presets Control
LCs (1,2,3,4)
...
The APM6803+ is a multichannel audio processing station for managing loudness and surround sound audio streams. Featuring DTS Neural Surround UpMix, DownMix and MultiMerge and Loudness Control, the APM6803+ combines unparalleled surround sound image placement and stability with loudness management processing to deliver a natural, open quality not found in traditional multiband compression technology. Flexible user-defined workflows and intelligent metadata handling ensure that the right processing is applied at the right time, allowing broadcasters to meet regulations while preserving the artistic integrity of the content. The APM6803+ is an audio processing system on a card and can be configured to support a wide variety of multichannel applications. Dual Dolby encoders and decoders can be enabled, allowing the APM6803+ to interface to virtually any type of signal. The APM6803+ also boasts a feature-rich audio/video frame synchronizer with monochannel routing for embedding and de-embedding. With eight discrete AES inputs and eight discrete AES outputs, the APM6803+ enables any combination of AES and embedded SDI audio input and output to be easily achieved.
Architecture
At the core of the APM6803+ lies an Audio Processing System comprising two high-end Digital Signal Processors (DSP), as well as custom-built hardware, that can process multiple channels of audio (see Figure 1-2). The Audio Processing System is a powerful and flexible signal processing engine. A virtual stream interface and flexible license management have been created to enable users to configure the system architecture dynamically and tailor the processing according to individual needs. For more details, see Chapter 4, Advanced Audio Processing.
Chapter 1 Introduction
Resource Utilization
To take full advantage of the processing power of the APM6803+, users are free to modify the architecture according to their needs. With this approach comes the possibility that a particular configuration may exceed the capabilities of the Audio Processing System. To assist users in implementing achievable configurations, key resources are tracked and reported. For more details, see Resource Utilization on page 40.
Licensing
The Audio Processing System is highly configurable and can support a wide range of Dolby and DTS processing blocks. Software license keys are required to enable individual Dolby encoders and decoders. For DTS functions, a flexible token system has been designed to allow users to change the type of processing without purchasing additional licenses. All licenses are field upgradable, enabling users to start with a minimal setup and grow the system as requirements dictate. For more details, see Chapter 11, Licensing.
Automation Control
Predefined Loudness Control settings are available for instant recall by automation, allowing users to create simple, effective automation profiles. Additional user-defined presets are available for maximum flexibility. The APM6803+ has a large number of GPI ports available for automation control.
Main Features
The APM6803+ includes the following features: 3G/HD/SD-SDI input and output capability SD/HD compliant SDI bypass relay Eight AES inputs and eight AES outputs Virtual audio stream interface for simplified configuration and control Multichannel audio system supporting up to 4 independent output streams Dual Dolby encoders and decoders (Dolby Digital and Dolby E) DTS Neural Loudness Control on each output stream Dual Surround Sound processors for implementing DTS Neural Surround UpMix, DownMix and MultiMerge Surround Field Protection using DTS Neural Surround MultiMerge ensures smooth, consistent Surround Sound output, while input switches between stereo and surround sound sources
Copyright 2012, Harris Corporation
Dynamic DTS license allocation Two individual metadata generators Intelligent metadata processing Dialnorm Translator Each output individually configurable for EAS (Emergency Alert System) override Each input individually configurable for Voice Over mixing Full-featured Harris frame sync technology Automatic audio/video delay alignment for consistent lip synchronization Automation control for dynamic on-air changes to loudness control profile Seamless sound maintains AES embedded audio on loss of SDI input Q-SEE-compliant thumbnails and alarms Custom Surround Sound and Loudness Control Presets Custom GPI and parameter scripting support
Chapter 1 Introduction
Components
The APM6803+ consists of a main module, a sub module, and a riser (interconnect) board assembled together, and a back module.
Sub Module
Back Module
Main Module
Figure 1-3 APM6803+Assembly The following are the main components: Table 1-1 APM6803+ Components Component Main Module Back Module Riser Board 44 Pin connector Description See Main Module on page 7 See Standard Back Module on page 9 and Relay Back Module on page 10 See Riser Board on page 8 See Pinouts for 44-pin Connector on page 12
Table 1-1 APM6803+ Components Component Breakout Cable Serial Port Cables GPI Cables Description See Breakout Cable on page 11 See Serial Port (RS-422/232) Connector on page 14 See General Purpose Interface (GPI) Connectors on page 15
Main Module
Mode select rotary switch Remote/local control jumper Navigation toggle switch LEDs Extractor handle
Figure 1-4 Top view of main module Feature LEDs Mode select rotary switch Navigation toggle switch Description Various color and lighting combinations of these LEDs indicate the module state. See LEDs - Main Module on page 32 for more information. This switch selects between various control and feedback parameters. This switch navigates up and down through the available control parameters: Down: Moves down through the parameters Up: Moves up through the parameters Local/remote control jumper Local: This jumper setting locks out external control panels and allows card-edge control only; limits the functionality of remote software applications to monitoring Remote: This jumper setting allows remote or local (card-edge) configuration, operation, and monitoring of the APM6803+ (this is the default setting)
Chapter 1 Introduction
Sub Module
Riser Board
The riser board connects the main module to the sub module .
SDI IN
RS-232/422 SERIAL
SDI OUT
GPI 1
GPI 2
ETHERNET
Figure 1-7 APM6803+ Standard Back Module Note: To maintain optimal output video signal integrity, terminate unused output video with 75 terminators.
10
Chapter 1 Introduction
APM6803+RLYT
RS-232/422 SERIAL
SDI OUT
GPI 1
GPI 2
ETHERNET
Figure 1-8 APM6803+ Relay Back Module For more details, see Relay Back Module on page 25.
11
Breakout Cable
The standard APM6803+ ships with a breakout cable that has unbalanced (coaxial) audio connectors to expand the number of available connections beyond what would fit on a standard tri-slot back connector.
AES1 IN AES2 IN AES3 IN AES4 IN AES5 IN AES6 IN AES7 IN AES8 IN AES1 OUT AES2 OUT AES3 OUT AES4 OUT AES5 OUT AES6 OUT AES7 OUT AES8 OUT
AES 1 In AES 2 In AES 3 In AES 4 In AES 5 In AES 6 In AES 7 In AES 8 In AES 1 Out AES 2 Out AES 3 Out AES 4 Out AES 5 Out AES 6 Out AES 7 Out AES 8 Out
12
Chapter 1 Introduction
Figure 1-10 Pin Numbers for 44-Pin Connector Table 1-2 Pinouts for 44-pin Connector
Pin No. on DB-44M 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 25 26 27 28 29 30 BNC BNC BNC GND BNC BNC GND BNC BNC GND 162A10019X (DB9.5) BNC GND BNC BNC GND BNC BNC GND BNC GND AES OUT 7 DARS IN 1 DARS IN 1 GND AES IN 2 AES OUT 3 GND AES OUT 3 AES OUT 6 GND RS232_GND (DB9) AES OUT 4 GND AES OUT 4 AES OUT 5 GND AES IN 8 AES IN 8 GND AES IN 5 GND AES OUT 7 DARS IN 1 DARS IN 1 AES IN 2 AES OUT 3 AES OUT 3 AES OUT 6 SERIAL AES OUT 4 AES OUT 4 AES OUT 5 AES IN 8 AES IN 8 AES IN 5 Blue Yellow Yellow White Blue Blue Blue Black Blue Blue Blue White White White Blue Black Black White Blue Blue Blue N/A Blue Blue Blue White White White Connection Type BNC BNC GND BNC GND BNC BNC GND BNC BNC GND BNC BNC GND BNC BNC GND BNC BNC GND BNC GND BNC Not Connected Description GENLOCK GENLOCK GND AES OUT 7 GND AES IN 4 AES IN 4 GND AES IN 3 AES IN 3 GND DATA IO (not used) DATA IO GND AES OUT 2 AES OUT 2 GND AES OUT 1 AES OUT 1 GND AES IN 7 GND AES IN 7 Wire Label GENLOCK GENLOCK AES OUT 7 AES IN 4 AES IN 4 AES IN 3 AES IN 3 DATA IO DATA IO AES OUT 2 AES OUT 2 AES OUT 1 AES OUT 1 AES IN 7 AES IN 7 External Cable Color Black Black Blue White White White White Yellow Yellow Blue Blue Blue Blue White White BNC Color Black Black Blue White White White White Yellow Yellow Blue Blue Blue Blue White White
13
14
Chapter 1 Introduction
2 3 4 5
10
6 7 8 9 10
2 3 4 5
10
6 7 8 9 10
15
2 3 4 5
6 GPI
10
6 7 8 9 10
2 3 4 5
GPI input 9 GPI input 10 GPI input 11 GPI input 12 GPI output 4 GPI output 5 GPI output 6
6 GPI
10
6 7 8 9 10
Signal Flow
Video Path Audio Path Metadata Path
16
Chapter 1 Introduction
Video Path
Metadata Processing SDI OUT 1 Serialize SDI OUT 2 Audio Demux EXT REF SDI OUT 3 2x1 Mux Genlock SDI OUT 4 FRAME REF SDI DEMUX Audio Processing SDI MUX
Deserialize SDI IN
AFD/WSS/VI Demux
Audio Mux
VBI Delete
Video Proc
WITH RELAY BM
Deserialize SDI IN Serialize
SDI OUT 1
SDI OUT 2
SDI OUT 3
SDI OUT 4
Audio Path
SDI MUX
Audio Output Configuration (Router) Audio Input Configuration (Router)
bypass bypass
SDI DEMUX
16 x 1 Router Dolby Decoder 1 SRC
Audio Gain Audio Gain Audio Gain Audio Gain LC 1
bypass
Dolby Encoder 1
bypass
Audio Gain
...
AES IN 1 AES IN 8
16 x 1 Router
Dolby Decoder 2
SRC
8x1 Router
Surround Processor 1 (UpMix, DownMix, MultiMerge ) Surround Processor 2 (UpMix, DownMix, MultiMerge )
LC 2
Dolby Encoder 2
bypass
LC 3
bypass
AES OUT 1
8x1 Router
...
LC 4
AES OUT 8
DARS IN
Genlock
2x1 Mux
DARS Reference
Surround Processors and LCs same function blocks as above .
Output Tones
GPI IN 1 GPI IN 12
...
Presets Control
LCs (1,2,3,4)
...
GPI OUT 6
Figure 1-12 Audio Path Note: Blocks in yellow indicate licensable functions implemented in the DSPs.
17
Metadata Path
Dolby Dec 1 Dolby Dec 2 Generator 1 Generator 2 Metadata Router MetaSmart 1 MetaSmart 2 Metadata Router
18
Chapter 1 Introduction
19
2
Overview
Installation
The following is covered in this chapter Unpacking the Module on page 19 Main Module Jumpers on page 21 Maximum 6800+ Frame Power Ratings on page 23 Installing 6800+ Modules on page 24 Upgrading Module Firmware on page 26 License Options on page 26
20
Chapter 2 Installation
Packing List
Table 2-1 Available Product Packages Ordered Product APM6803+T Content Description One APM6803+ front module One sub module One triple slot back connector (standard) One unbalanced audio breakout cable One riser Three mini-mate header cables (two for GPI connectors and one for serial port) APM6803+RLYT One APM6803+ front module One sub module One triple slot back connector (relay) One unbalanced audio breakout cable One riser Three mini-mate header cables (two for GPI connectors and one for serial port)
21
CJ1 Jumper
Figure 2-1 Jumper Location 1 2 Locate jumper CJ1 on the module (behind the mode select rotary switch). Figure 2-1 shows the location of the CJ1 jumper. Place the jumper on pins 1 and 2 to set the module for Remote control, or pins 2 and 3 to set the module for Local control. See Figure 2-2.
Figure 2-2 CJ1 Settings for Local and Remote Control Note: The white triangle near the jumper pins on the module indicates pin 1.
22
Chapter 2 Installation
Table 2-2 Sub Module Jumpers Jumper Designation J16A/16B RS-422 Serial Port 1/2 Termination Unterminated (default) Pins 1 and 2 open Terminated Pins 1 and 2 shorted Settings
1 2
1 2
1 2
1 2
Pins open
Unterminated
J4/J6 Serial Port 1/2 Duplex Mode Full Duplex (default): Pins 1/2 and 3/4 open Half Duplex: Pins 1/2 and 3/4 shorted
Pins open
4 2 4 2
23
Jumper J2 J5 J8 J9
Jumper Default Positions Pins 1 and 2 shorted Pins 2 and 3 shorted Pins 2 and 3 shorted Pins 2 and 3 shorted
Note: Ensure that all factory jumpers are per the default settings.
24
Chapter 2 Installation
See the 6800+ Frame Installation and Operation Manual for information about installing and operating an FR6802+QXF, FR6822+, or FR6802+ frame and its components. CAUTION Before installing this product, read the 6800+ Series Safety Instructions and Standards Manual shipped with every 6800+ Frame Installation and Operation Manual or downloadable from our website. This safety manual contains important information about the safe installation and operation of 6800+ series products.
Unsupported Frames
APM6803+ modules cannot be installed in: FR6802+DM FR6800+MB 6800/7000 series frames The APM6803+ cannot be put in chained frames or old ICE frames. See your 6800+ Frame Installation and Operation Manual for installation instructions and for details on installing back connectors in a frame.
25
RS-232/422 SERIAL
SDI OUT
GPI 1
GPI 2
ETHERNET
The relay back module functions as follows: When the front module/card is running normally, the relay is closed and the signal is not bypassed. If the front module/card is pulled out of the frame OR the frame loses power, the relay is opened and the signal is bypassed from SDI IN to SDI OUT 1. When the front module/card is inserted into the frame, the signal is bypassed until the card boots up. After the card boots, the relay is closed again.
Notes The relay functionality occurs only on power loss; it cannot be manually controlled from the GUI. Ensure you upgrade to firmware version 1.1.82 or higher for the relay back module. See Specifications on page 119 for more details.
26
Chapter 2 Installation
If all the LEDs work normally or the parameter Sub Board Status reports Normal, the module can be operated. Note: Ensure the ETH card is version 4.5.1 or later. When upgrading, do not remove the module from the frame.
License Options
For details, see Chapter 11, Licensing.
27
3
Overview
The following sections are covered here: Parameter Categories on page 28 Changing Parameter Settings using Card-Edge Controls on page 29 Changing Parameter Settings Using CCS Software on page 30 Changing Parameter settings using a Web browser on page 31 LEDs - Main Module on page 32 LEDs - Sub Module on page 34 Alarms on page 34
28
Parameter Categories
The following table lists the main categories that Parameters are divided into and what you can control from those categories: Table 3-1 General Parameters Licensing and Utilization parameters. GPI Input/Output/Custom Parameters Parameter Control Script AFD / VI / WSS Embedder and De-Embedder parameters Audio Input Configuration (including Dolby Decoder) Surround Processing (including UpMix, DownMix, MultiMerge) Audio Output Configuration (Including Loudness Control, Dolby Encoder) Output Router Configuration (Embedded, AES, and Output Tones) SDI Genlock/DARS Input Video Delay Synchronization Proc Amp VBI Embedded AES Loudness Control Status Serial Port Metadata Input Metadata Ouput Generator MetaSmart Alarm Configuration Video Streaming information Displays information on the Version and Presets. For details, see Chapter 7, Key Parameters. See Chapter 8, Custom Scripting for details on custom parameter scripts. For a complete list of parameters, refer the HTM Parameter list (available with the documentation). You can access APM6803+ parameters through CCS Navigator, an HTTP web browser, or a third-party SNMP-based control application (depending on your host frames options). When you change a parameter, the effect is immediate. However, the module requires up to 30 seconds to save the latest change. After 30 seconds, the new settings are saved and will be restored if the module loses power and must be restarted. If you make changes to certain parameters, other related parameters may also be affected. For example, virtual stream selections can have an impact on routing. General Presets (loading and saving of presets) are not currently supported.
Audio Configuration
Video Configuration
Audio Status
Audio Metadata
29
Rotate the hex switch to the parameter number (1 to 9) or letter (A to F) of the option you want to set. Toggle the navigation switch to select and set the value of the chosen parameter. Do either of the following: Rotate the hex switch to another parameter number/letter in the current bank, and then repeat step 4. Rotate the hex switch to 0 again to select a different bank, and then repeat steps 3 and 4. Use an available 6800+ software control option to aid in viewing, setting, and confirming the parameter value. Note: Refer to the HTM Parameter list (available with the documentation) for more details on changing parameter settings using card edge controls.
30
4 5
31
You can now switch to Control mode by selecting Operational Mode > Control from the main menu. Double-click APM6803+ in the Navigation pane. The Control dialog box opens displaying the modules controls.
32
If the Hardware item displays a version number, then the revised specifications apply to your module. If the Hardware menu item provides no version information, then the original specifications apply.
Figure 3-1 Location of APM6803+ Main Module LEDs Table 3-3 Main Module Card Edge LEDs LED AUTO 720P 1035I 1080P/SF 1080I 525 625 VID PRES ELECT
Copyright 2012, Harris Corporation
Color Indicators Green Green Green Green Green Green Green Green
Description Auto detection of the SDI input standard is active Selected video standard is 720p Unused Selected video standard is 1080p or 1080psf Selected video standard is 1080i Selected video standard is 525 Selected video standard is 625 Electrical input video is present
33
Table 3-3 Main Module Card Edge LEDs LED VID PRES FIBER ELECT IN SEL FIBER IN SEL REF PRES FREEZE DARS PRES AUDIO 1 AUDIO 2 AUDIO 3 AUDIO 4 AUDIO 5 AUDIO 6 AUDIO 7 AUDIO 8 Color Indicators Green Green Green Green Yellow Green Green Green Green Green Green Green Green Green Description Not used Not used Not used Indicates presence of reference signal Indicates the output video is frozen Indicates presence of DARS Indicates presence of AES audio on AES input 1 Indicates presence of AES audio on AES input 2 Indicates presence of AES audio on AES input 3 Indicates presence of AES audio on AES input 4 Indicates presence of AES audio on AES input 5 Indicates presence of AES audio on AES input 6 Indicates presence of AES audio on AES input 7 Indicates presence of AES audio on AES input 8
Note: If the LED is flashing red, please contact your Customer Service representative.
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Alarms
If a major or minor alarm is triggered within your APM6803+ module, the Status LED lights red. Alarms are usually logged and monitored within available software control applications (for example, CCS Navigator). You can only differentiate between major and minor alarms within a software control application. See the appropriate software control user manual or online help for more information.
Alarm Options
The following settings can be made for each alarm within Navigator software. Table 3-5 Alarm Options Alarm Option Enable/Disable Effect This option toggles between Enabled and Disabled. If the alarm is Enabled, an alarm condition generates an alarm; if it is Disabled, the alarm condition is ignored. By default, all alarms are disabled. This setting determines whether a triggered alarm is reported as major or minor. The range is 110. A priority of 6 or higher is a major alarm, and a priority of 5 or lower is a minor alarm.
Alarm priority
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Table 3-5 Alarm Options Alarm Option Trigger (s) Effect This option determines how long an alarm condition must exist (in seconds) before the alarm is triggered. If the alarm level is reached for less time than the Trigger duration, then the alarm will not trigger. Choose any duration from 0 to 7200 seconds (or 2 hours). If this option is set to 0 and the alarm condition exists for any period of time, the alarm is triggered. Determines the amount of time the alarm condition must be in abatement in order for the alarm to be turned off. Choose any duration from 0 to 7200 seconds (or 2 hours). If this option is set to 0 and the alarm condition ceases for any period of time, the alarm is cleared. When an alarm is active, click this option to allow other users on the network to see that you have acknowledged the alarm.
Clear (s)
Ack
Alarm Definitions
Note: Refer to the HTM Parameter list (available with the documentation) for a complete list of alarms.
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Audio Processing
The backbone of the audio processing capability of the APM6803+ is the Audio Processing System (APS). The APS consists of two Digital Signal Processors (DSPs). Each DSP is further divided into two Advanced Audio Processors (AAPs). The AAPs are responsible for processing all of the licensable functions, namely Loudness Control, Surround Sound and Dolby Encoder/Decoders. As you enable functions, AAPs are automatically configured by the system as appropriate.
Outgoing Audio
Port 1
AAP1 DSP1
Incoming Audio
AAP2
AAP1 DSP2
Port 4
AAP2
Smart Routing
As the user configures the various input and output types, along with the required surround processing and loudness control, the systems Smart Routing function will automatically configure each AAP and route the appropriate signals to the AAP ports as required. The user does not need to understand the inner workings of the various blocks and is free to work at the logical level. For example, when enabling Loudness Control on a Virtual Output, simply enable Loudness Control for that output. The system will then find an unused AAP and route the audio to that AAP, then configure the AAP accordingly.
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SDI De-embedder
SDI Embedder
Dolby Decoder 1 AES1 AES2 Mono Channel Audio Router ... AES8 Dolby Decoder 2
Virtual In1
Loudness Control
Dolby Encoder 1
Virtual In2 Virtual Audio Stream Router Loudness Control Loudness Control Dolby Encoder 2 Virtual Out2 Mono Channel Audio Router
AES2 ...
Virtual In3 Virtual In4 Virtual In5 Virtual In6 Virtual In7 Virtual In8
Virtual Out3
AES8
Loudness Control
Virtual Out4
Figure 4-2 Virtual Input and Output streams When a Virtual Input stream is defined, you need to choose the source for each channel in the stream from the input router. For instance, if Virtual Input 1 is defined as Stereo, you will need to select the source for the L and R channels.
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Once a stream is defined, you can simply manipulate the stream as a single entity. Signal routing, proc amp settings and Loudness Control are then applied automatically by the APM6803+ to each sub channel within the stream as appropriate. From that point on, you no longer need to keep track of individual channels within the stream, and can route the entire stream as a single instance when routing to the Surround Processors or to output streams. Proc Amp settings like gain and mute can now be applied to the entire stream under a single parameter.
Loudness Control simply becomes an option that can be enabled or disabled for the stream, instead of having to individually route each channel to a Loudness Control block.
Managing the audio in this way, at the "stream" level, rather than having to manage individual channels, makes it much easier, quicker and less error prone to setup the configuration. Virtual streams unleash the full processing power of the APM6803+ without bogging down the user with complicated routing and configuration menus.
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Resource Utilization
As functions are enabled, the APM6803+ keeps track of the processor utilization and reports the current total. It also reports the available licenses, and how many have been used so far. The values are available under General > Utilization.
For the majority of scenarios, the Virtual Stream interface and the automatic routing and configuration that occurs greatly simplifies system configuration. However, it is possible to exceed the capabilities of the APS. The high level limitations are as follows: The sum of the processing power required for two AAPs in a given DSP must not exceed the full processing power of that DSP. Each AAP has a single port for sending and receiving audio. As such, some error messages are provided to assist in understanding what limitation is preventing the system from implementing the users request.
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Similarly, when defining output audio, you select the Virtual Out 1 Type as Surround5.1+2.0 with loudness control enabled, the Out 1 AAP Status will show as Out1+LC 5.1+2.0.
The AAP Status parameter is just an indicator to confirm whether your selected function was successfully applied. If the selected function was not available or was not applied for some reason, the AAP status will reflect this.
If the request fails, one of the following messages may be displayed (Also see the Troubleshooting section in the APM6803+ Quick Start): Table 4-1 AAP Error messages AAP Status Parameter Router Full Failed-No License Failed-MCBSP Full Description Indicates that all AAPs are utilized. Indicates that adequate licenses are not available for processing. Indicates that the DSP (Digital Signal Processor) does not have enough resources.
There are 2 Digital Signal Processor (DSP) chips that each have 2 Advanced Audio Processing Blocks (AAPs).
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To simplify user configurations, when you configure an audio function (be it Loudness, MultiMerge, Dolby, etc.), the DSP using smart routing automatically routes audio functions to processors as required. The AAP 1-4 Configuration parameters reflect an internal allocation of audio processing functions. For example, if you configure a Virtual Input to a Dolby-E stream, you will see your DSP 1 Utilization go to about 23%. And one of the AAP blocks will report its usage as In1+E5.1.
If you opt to route this to a surround processor and use MultiMerge, you will see the MultiMerge function auto routed to the second available AAP block, and the total DSP utilization will also increase proportionately.
Free or available Processors will show status as None. The Audio DSP utilization should be 0% if you have nothing configured, and the individual AAP blocks will report their usage as None. Although the APM6803+ attempts to combine functions into a single AAP where possible, depending on a particular configuration, it is possible to exceed available resources. If all the Processors are in use, additional handling may not be possible unless some processing is freed up. The following table provides approximate utilization information for audio functions. The percentages are per function enabled.
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Table 4-2 Advanced Audio Processing Utilization AAP Block UpMix (high latency) DownMix MultiMerge (low latency) MultiMerge (high latency) Loudness Control 5.1 Loudness Control 2.0 Loudness Control 5.1 + 2.0 DSP Utilization (%) 27 4 40 40 29 9 39
Table 4-3 Advanced Audio Processing Delay AAP Block Dolby Digital Decoder Dolby Digital Decoder with DownMix Dolby Digital Encoder UpMix (low latency) UpMix (high latency) DownMix MultiMerge (low latency) MultiMerge (high latency) Loudness Control 5.1 Loudness Control 2.0 Loudness Control 5.1 + 2.0 Latency (ms) 69.33 69.33 149.23 108.60 129.50 87.00 132.31 153.64 137.29 135.48 137.29
Table 4-4 Advanced Audio Processing Delay for Dolby E AAP Block Dolby E Decoder Frame Rate 29Hz/59Hz 25Hz/50Hz 30Hz/60Hz 24Hz 23Hz Dolby E Encoder 29Hz/59Hz 25Hz/50Hz 30Hz/60Hz 24Hz 23Hz Latency (ms) 100.77 120.80 100.67 125.83 125.94 108.78 130.40 108.67 135.83 135.95
Copyright 2012, Harris Corporation
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Dolby Processing
Dolby E Decoder
An incoming Dolby stream is treated as a normal audio program in the path, and all you have to do is configure one of the Virtual Input groups as Dolby. For example, if the incoming audio program is Dolby E 5.1+2.0, configure Virtual In 1 as Dolby E-5.1+2.0. When defining the Virtual In Type for a Dolby E stream, the available options are: Dolby E-5.1 Dolby E-5.1+2.0 Dolby E-4x2.0 These 3 options are based on the three most common program configurations; however, free mapping of Dolby channels based on your requirements is supported.
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If the incoming audio program is PCM, it is passed out on the Left channel and Right channel. The options Virtual In 1 Surround 5.1 and Virtual In 1 Stereo in the downstream represent the decoded PCM audio of the Dolby E stream. The Dolby E Dec X Program Config feedback parameter reports the actual program type of the incoming Dolby E stream. Metadata from a Dolby E stream is available for downstream processing and can be sourced to a serial port or MetaSmart. The Metadata Monitor can be used to check the metadata status of an incoming Dolby stream.
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Dolby E Encoder
Configure a Dolby E Encoder on the virtual output side. The options Dolby E-5.1, Dolby E-5.1+2.0, and Dolby E-4x2.0 are available for flexible audio routing. The Dolby E Encoder follows parameters in external metadata, sourced from the metadata generator, SDI input, serial port input, MetaSmart or incoming Dolby E. Refer Dolby Metadata Dependencies for metadata dependencies.
Note: When generating metadata for the Dolby E Encoder (see Generating your own Metadata), the 7.1 or 7.1 Screen Program Configuration is not supported for the Encoder. Ensure that you do not select the 7.1/7.1 Screen Program Configuration option with the Encoder to ensure proper operation.
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Go to Audio Configuration > Output Configuration > Virtual Out 1 > Source Select Set Out 1 Dolby Enc Surround Sel to Virtual In 1 Surround 5.1 Set Out 1 Dolby Enc Stereo Sel = Virtual In 1 Stereo
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L1 L1 L1 L1
R1 R1 R1 R1 R1 R1 R1 R1 R1 R1 M2 R1 R1 R1 R1 R1 R1 M2 R1
C1 C1 C1 C1 C1 C1 L3 L3 M3 M4 M3 C1 C1 C1 L3 M3 M4 M3 C1
L2 M2 L2 L2 L2 M2 L2 L2 L2 M2 M7 L2 M2 L2 L2 M2 -
R2 M3 R2 R2 R2 M3 R2 R2 R2 M3 M8 R2 M3 R2 R2 M3 -
4+2+1+1 (4+2+2x1) L1 4+1+1+1+1 (4+4x1) L1 2+2+2+2 (4x2) 2+2+2+1+1 (3x2+2x1) 2+2+1+1+1+1 (2x2+4x1) 2+1+1+1+1+1+1 (2+6x1) 1+1+1+1+1+1+1+1 (8x1) 5.1 4+2 4+1+1 (4+2x1) 2+2+2 (3x2) 2+2+1+1 (2x2+2x1) L1 L1 L1 L1 M1 L1 L1 L1 L1 L1
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L1 L1 M1
R1 R1 M2
M3
M4
L2 M2 -
R2 M3 -
The Dolby D Decoder applies line mode processing for dynamic range control and enables dialog normalization. Decoded audio levels are always reduced unless the dialnorm parameter is -31. For example, a program with dialnorm value -25 will be reduced by 6 dB after decoding. It is not recommended to re-encode a Dolby D stream, and accordingly, metadata from a Dolby D stream is not available for downstream processing. You cannot output Dolby D metadata through the serial port, or as a source of MetaSmart. The metadata status of an incoming Dolby stream can be viewed from the Metadata Monitor. If a Dolby decoder is configured, Monitor X Source displays a Decoder X option. Control parameters for Dolby D Decoder are under: Audio Configuration -> Input Configuration -> Virtual In X -> Dolby Decoder.
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6
Overview
DTS Processing
The following DTS Neural functions are available as options: DTS Neural Surround UpMix on page 55 DTS Neural Surround DownMix on page 57 DTS Neural Surround MultiMerge on page 59 DTS Neural Loudness Control on page 64
2. 0 input
UpMix processing
5. 1 output
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-100 to 100
Sur Proc x UpMix Surround Width Sur Proc x UpMix LFE Cut off Sur Proc x UpMix Final Limiter Ceiling Sur Proc x UpMix Preset Save
0 to 100 60 Hz to 140 Hz
Specifies the threshold where final -12 dBFS to 0 dBFS limiting on the up-mixed output occurs. Select a preset to save custom settings to. Custom 1 Custom 2 Custom 3 Custom 4 Custom 5
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Save your settings to one of the available custom presets (Custom1-Custom5) by selecting it from the Sur Proc x Upmix Preset Save drop down.
To load settings that you saved to a custom preset, select it from the Sur Proc x Upmix Preset Recall drop down. For example, select Custom1 to load UpMix settings saved to that preset.
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Figure 6-2 shows a DownMix taking a multi-channel audio source. The downmix creates two-channel audio source using the Neural Audio approach of embedding a watermark signal within the stereo audio signal patch. The watermark signal contains spatial and steering positioning information. The resulting stereo audio signal is also known as LwRw.
5. 1 input
2. 0 output
No (select save)
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To load settings that you saved to a custom preset, select it from the Sur Proc x Downmix Preset Recall drop down. For example, select Custom1 to load DownMix settings saved to that preset.
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Figure 6-3 shows how the MultiMerge takes a two-channel audio source (stereo, matrix encoded stereo, LtRt or DTS Neural Surround LwRw) and renders a 5.1 multi-channel mix; in combination with taking original multi-channel content and creating a stereo downmixed signal, depending on the input configuration and content source used.
2. 0 input (L , R) 5.1 input (C , LFE , Ls, Rs) 2. 0 input (L , R) UpMix processing or passthru 5. 1 output
2. 0 output
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Table 6-3 MultiMerge parameters Audio Configuration/Surround Processing/Surround Proc x/MultiMerge Settings Parameter Name Sur Proc x Input Selection Mode Function Specifies how input channels are selected. Mix mixes the auxiliary 2.0 inputs into the multichannel 5.1 inputs. Multi always uses the multichannel 5.1 inputs. Stereo always uses the stereo L/R pair from the multichannel inputs. Aux always uses the auxiliary 2.0 inputs. Multi Detect uses multichannel 5.1 inputs if they are active. If not, the auxiliary 2.0 inputs are used. Stereo Detect uses the stereo L/R pair from the multichannel 5.1 inputs if they are active. If not, the auxiliary 2.0 inputs are used. Aux Detect uses the auxiliary 2.0 inputs if they are active. If not, the multichannel 5.1 inputs are used. Multi Aux Stereo uses multichannel inputs if they are active, or uses the auxiliary inputs if they are active. If neither is active, it uses the stereo L/R pair from the multichannel inputs. Sur Proc x Input Noise Floor Threshold Sur Proc x Input Cross Fade Time Specifies the amount of signal required when detecting active inputs. Specifies the transition time between inputs when switching due to detected level changes. -80 dB to -50 dB 50 ms to 750 ms -60 dB 200 ms Options Mix Multi Stereo Aux Multi Detect Stereo Detect Aux Detect Multi Aux Stereo Default Multi
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Table 6-3 MultiMerge parameters Audio Configuration/Surround Processing/Surround Proc x/MultiMerge Settings Parameter Name Sur Proc x MultiMerge Mode Function Current status of the MultiMerge mode. Auto determines if content is stereo (2.0) or surround (5.1), and adapts to the correct UpMix/Passthrough mode for consistent 5.1 output. When in Auto, the Detect Threshold parameter controls the noise floor level for the detection. Any content above this threshold on the surround channel inputs 3-6 (C, LFE, Ls, Rs) is considered surround and MultiMerge will be in passthrough mode. Passthrough forces MultiMerge to always pass through 5.1 content to the 5.1 outputs untouched, while creating a downmix for the Aux output. UpMix forces MultiMerge to always upmix stereo content on both of the Left/Right 5.1 inputs and the Aux 2.0 inputs. Sur Proc x MultiMerge Latency Specifies the latency profile of the multi-merge. Sur Proc x Noise Floor Threshold Sur Proc x Cross Fade Time Specifies the signal level that must be detected on any. Specifies the transition time between up-mix and passthrough when in auto mode. Specifies the amount of front-to-back bias to apply to the standard soundstage. Specifies the amount of narrowing or widening to perform on the front channels. Specifies the wideness of the surround channels. The cutoff frequency for the generated LFE channel. Specifies the encoding mode for the left and right channels. The cutoff frequency of the input LFE channel. Low Latency High Latency -80 dB to -50 dB 50 ms to 750 ms Low Latency -60 dB 200 ms Options Auto Passthrough Upmix Default Auto
Sur Proc x MultiMerge UpMix Depth Sur Proc x MultiMerge UpMix Front Width Sur Proc x MultiMerge UpMix Surround Width Sur Proc x MultiMerge UpMix LFE Cut off Sur Proc x MultiMerge DownMix Encoding Mode Sur Proc x MultiMerge DownMix LFE Cut off
-100 to 100
-100 to 100
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Table 6-3 MultiMerge parameters Audio Configuration/Surround Processing/Surround Proc x/MultiMerge Settings Parameter Name Sur Proc x Multi Final Limiter Ceiling Sur Proc x Stereo Final Limiter Ceiling Sur Proc x MultiMerge Preset Save Function Specifies the threshold where final limiting on the surround (5.1) output occurs. Specifies the threshold where final limiting on the auxiliary stereo (2.0) output occurs. Select a preset to save custom settings to. Options 20 dBFS to 0 dBFS Default 0
20 dBFS to 0 dBFS
(select save)
To load settings that you saved to a custom preset, select it from the Sur Proc x MultiMerge Preset Recall drop down. For example, select Custom1 to load MultiMerge settings saved to that preset.
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LC x Control
Indicates whether or not to bypass the loudness control algorithm and this CODEC will behave as a delay only. Indicates whether or not bypass the loudness control algorithm and this CODEC will behave as a delay only. Specifies the type of loudness measurement to make prior to performing loudness control. Selects the source of target Loudness Level. See Using MetaSmart.
Always On
LC x Control Feedback
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Table 6-4 Loudness Control Parameters Audio Configuration/Output Configuration/Audio Out x /Loudness Control/Loudness Control Settings Parameter Name LC x Ratio Function Controls the amount of gain or attenuation that is applied when loudness differences are measured. For example: A setting of 1.00 indicates that for every dB of loudness difference measured between the input signal and the target level, 1 dB of gain or attenuation will be applied. For example, a setting of 0.50 indicates that for every dB of loudness difference measured between the input signal and the target level, 0.5 dB of gain or attenuation will be applied. LC x Upper Threshold Represents the allowable distance above the Target Level the input signal can range before attenuation. If an input signal falls between the Upper Thresh and the Lower Thresh in reference to the Target Level, no correction will be applied. For example: A setting of +4dB indicates that audio content can range 4 dB above the Target Level before attenuation is applied. Content arriving louder than 4 dB above the Target will be appropriately attenuated. -27 dBFS (Target) + 4 dBFS (Upper Thresh) = -23 dBFS (Upper Thresh Value) For example: A setting of 0 dB indicates that attenuation will be applied to any input signal exceeding the Target Level. 0 dB to 20 dB 0 dB Options 0.00 to 1.00 Default 0.98
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Table 6-4 Loudness Control Parameters Audio Configuration/Output Configuration/Audio Out x /Loudness Control/Loudness Control Settings Parameter Name LC x Lower Threshold Function The Lower Threshold parameter represents the distance below the Target Level the input signal can range before gain is applied. If an input signal falls between the Upper Thresh and the Lower Thresh in reference to the Target Level, no correction will be applied. For example: A setting of -5 dB indicates that audio content can range 5 dB below the target level before gain is applied. Content arriving quieter than 5 dB below the Target will be appropriately raised. -27 dBFS (Target) + -5 dBFS (Lower Thresh) = -31 dBFS (Lower Thresh Value) For example: A setting of 0 indicates that gain will be applied to any input signal falling below the Target Level. LC x Upper Alarm Threshold LC x Lower Alarm Threshold LC x Upper Warning Threshold Specifies the Loudness control Upper Alarm Threshold. Specifies the Loudness control Lower Alarm Threshold. Represents the allowable distance above the Target Level the input signal would not trigger a warning alarm. The value of this parameter is always no bigger than Upper Threshold. Represents the allowable distance below the Target Level the input signal would not trigger a warning alarm. The absolute value of this parameter is always no bigger than Lower Threshold. 0 dB to 20 dB -20 dB to 0 dB 0 to 20 dB 0 dB 0 dB 0 dB Options -20 dB to 0 dB Default 0 dB
-20 to 0 dB
-20 dB
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Table 6-4 Loudness Control Parameters Audio Configuration/Output Configuration/Audio Out x /Loudness Control/Loudness Control Settings Parameter Name LC x Freeze Window Function Sets the size of a window in dB where small loudness differences that fall within this window are deemed allowable. When loudness differences fall outside of this allowable window then appropriate gain or attenuation is applied to control loudness. The Freeze Window parameter can be effectively used to limit the amount of small range gain variability. For example: A window setting of 4 dB indicates that when the loudness level is measuring within 2 dB of the target level that no further correction is required. When the loudness level becomes further than 2 dB from the target level then appropriate gain or attenuation is applied. LC x Noise Floor Prevents increasing the noise floor level. Input signals below the threshold are considered too low and are not managed by the loudness processing to avoid bringing up the noise level. Input signals above the Noise Floor are considered valid and are managed by the loudness processing. Controls how quickly the processing will respond to sharp onsets in loudness level. Note: A compressor is present after the attack time processing to catch any short-term loudness peaks which may pass through. Controls how quickly the processing will respond to sharp drops in loudness level. A compressor is present to catch any short-term loudness peaks which aren't fully captured by the attack time processing of Loudness Control. The compressor threshold parameter is set as the allowable short-term peak above the Target Level. -80 dBEq to -20 dBEq -55 dBEq Options 0.0 to 10.0 Default 2.0
LC x Attack Time
5 ms to 150 ms
50 ms
LC x Release Time
20 ms to 500 ms
150 ms
LC x Compressor Threshold
0 dB to 16 dB
5 dB
LC x Compressor Ratio
0.00 to 1.00 Controls the amount of attenuation that is applied when short-term peaks exceed the compressor threshold. See the Ratio parameter for further detail.
0.50
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Table 6-4 Loudness Control Parameters Audio Configuration/Output Configuration/Audio Out x /Loudness Control/Loudness Control Settings Parameter Name LC x Shaping Function Specifies the amount of loudness shaping desired. A value of zero is no loudness shaping. A final limiter is present to control the absolute waveform peak level for the stereo output. This parameter controls the output peak level in dBFS. A value of zero disables limiter. Activates real-time meter values in the status output. Indicates whether or not final limiters be applied according to the value of FinalLimiterCeiling_dBFS. Selects to save the current LC setting. Options 0 to 10 Default 0
0 dBFS
Disabled Enabled No Yes (select save) Custom1 Custom2 Custom3 Custom4 Custom5
Enabled
LC x Preset Save
(select save)
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Set the Loudness Upper Threshold. This parameter represents the allowable distance the input signal can range above the Target Level before attenuation. If an input signal falls between the Upper and Lower Threshold in reference to the Target Level, no correction will be applied. For instance, if Target Level is -20 dBFS, Upper Threshold is 12 dBFS, anything over -8 dBFS will have attenuation applied.
7 8
Similarly, set the Loudness Lower Threshold. Set the Upper Alarm Threshold If the long-term loudness average level In Average Loudness falls between the Upper and Lower Alarm thresholds in reference to the Target Level, the alarm LC Alarm Level is not triggered. Otherwise, the alarm is triggered (also affected by the trigger time) and it automatically controls the loudness control working mode if LC Control Mode is set as On Alarm Only. (Also see Loudness Protection)
9 10
Similarly, set the Lower Alarm Threshold Set the Upper Warning Threshold (0-20). This is the allowable (relative) distance between the Target Level and Upper Threshold and this range will not trigger a warning alarm. For instance, if Target Level is 20 dBFS, Upper Threshold is 12 dBFS, and Upper Warning Threshold is 6 dBFS - anything over 32 dBFS will have attenuation applied; anything between 26-32 dBFS will trigger an alarm, but no correction will be applied. (Also see Loudness Protection)
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Similarly, set the Lower Warning Threshold. Set the Freeze Window This is a window size between 0 and 10 dB where small loudness differences are allowed. Anything outside this window has gain/attenuation applied.
13 14 15 16 17 18 19 20
Set the Noise Floor to control the lowest input level where audio correction can still be applied. Set the Attack time to control how quickly processing responds to sharp onsets in loudness levels. Set the Release Time to control how quickly processing responds to sharp drops in loudness levels. Set the Compressor Threshold to detect and allow short term loudness peaks. Set the Compressor Ratio to control the attenuation applied to short peaks that exceed the Compressor Threshold. Set the Loudness Shaping. Set the Output Limiter Threshold. Set the Metering.
Loudness Protection
The APM6803+ can provide certain warnings and alarms to indicate to the user that the incoming loudness exceeds pre-defined thresholds. This is particularly useful when using a GPI and automation to change the loudness control profile dynamically. The idea is to warn the user that the incoming loudness has been consistently low or high for an extended period, and that enabling loudness correction may be desired.
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The first step is to average the incoming loudness on any Virtual Output that has Loudness Control enabled. The user specifies an averaging window size, and the APM6803+ will use it to average the loudness of the audio going into the Loudness Control block. This average is then sent to a series of alarms that have trigger and clear times. Two sets are provided, one to act as a warning, and the other to act as an alarm condition. The alarm condition has the added benefit that it can also trigger a loudness control preset.
Target Level
By setting appropriate warning and alarm upper and lower thresholds, the APM6803+ can provide additional information to help protect against excessive loudness when very light or no real time correction is being applied. In addition, it can take action automatically and turn on a more aggressive loudness correction profile. Care must be taken to ensure the clear time in this case is longer than the trigger time in order to create hysteresis in the system, such that the system does not continuously enable and disable real time loudness correction. The clear time should provide sufficient time for a user to investigate and either keep the loudness correction profile or manually revert to another profile.
Each Loudness Control Status category (Loudness Control 1-4 Status) contains 4 groups (Group 1-4 LC Status) - each group displays status for one audio channel (one group for stereo, one for surround etc). The last two groups are not currently used (will be used for Dolby).
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Status is displayed in groups as follows: Table 6-5 Loudness Control Status Groups Virtual Output Type Surround 5.1 Surround 5.1+2.0 Audio Channel Assigned Status Group 5.1 5.1 2.0 Stereo 2.0 Group 1 LC Status Group 1 LC Status Group 2 LC Status Group 1 LC Status
For example, if Virtual Out 1 is Surround 5.1+2.0 with loudness enabled, Loudness Control 1 Status provides the following: Group 1 LC Status displays information for the 5.1 Surround audio Group 2 LC Status displays information for the 2.0 stereo audio. Groups 3 and 4 LC Status will be disabled.
LC Grp Input Loudness: The input loudness measurement LC Grp Input Peak: Input peak measurement LC Grp Output Loudness: Output loudness measurement LC Grp Output Peak: Output peak measurement LC1 Grp1 Correction: Correction (attenuation/gain) actively applied LC1 Grp1 Compression: Compression actively applied
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Four 1770 presets (BS.1770-Ultra Light, BS.1770-Light, BS.1770-Medium, BS.1770-Aggressive) Five Custom presets are also available. The DTS and 1770 presets do not affect the target loudness level or the target loudness level source - you need to use Custom presets to change those settings. Note that in cases where you choose to use custom presets to control general loudness control settings, target level and target level source are also changed. The following is a list of loudness control parameters affected by DTS/1770 and custom presets. Table 6-6 Loudness Control parameters affected by presets LC Parameter Preset DTS/1770 Preset Used Custom Preset Used Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes
LC Loudness Measurement Type Yes LC Target Loudness Fixed Level LC Target Ratio LC Upper Threshold LC Lower Threshold LC Freeze Window LC Noise Floor LC Attack Time LC Release Time LC Compressor Threshold LC Compressor Ratio LC Shaping LC Final Limiter Ceiling LC Metering LC Control (Bypass) Upper Alarm Threshold Lower Alarm Threshold Upper Warning Threshold Lower Warning Threshold Target Loudness Level Source No Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes No No No No No
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Key Parameters
This chapter explains some of the key parameter settings in detail. Audio Processing Input Audio Rate Sample Rate Convert (SRC) Control Group (1-4) Deembedding Control Audio Synchronization Audio V-Fade ADS Clean Parameter Audio Embedding Modes Audio Embedding Errors Audio Test Tones Audio Delay Ranges EAS Control Channel Word Length Stereo Mixing Voice Over Functionality Video Processing Video Frame Synchronization Video Delay Ranges Auto Match Audio/Video Delay Audio/Video Align AFD/WSS/VI Detection and Insertion Thumbnail Display Fast Video Switch GPI Configuration GPI Input Configuration GPI Output Configuration Note: To locate specific parameter settings referenced in this chapter, access your APM6803+ through a web browser or CCS Navigator and then go to the specified location.
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Audio Processing
Input Audio Rate
When embedding audio in PCM mode CH1A(G1C1) to CH8B(G4C4), the input audio sample rate may be from 32 kHz to 108 kHz. This input audio is sample rate converted to 48 kHz prior to embedding in the video signal; as well, the embedder indicates 48 kHz in the Rate word of the audio control packet for each embedded audio group. When embedding audio in Non-PCM mode AES 1A to 8B, the input audio sample rate must be 48 kHz, and it must be frequency locked to the source video. In this scenario, the sample rate conversion function is bypassed and the embedder indicates 48 kHz in the Rate word of the audio control packet for each embedded audio group.
The APM6803+ has a set of parameters to control the SRC (Sample Rate Convert) chips. The APM6803+ employs 16 SRC for each audio pair from SDI (8 pairs) and AES input (8 pairs). In auto mode, the module monitors audio format, the v bit of each channel, as well as the nPCM bit packed in the C bit stream. If a Dolby stream is detected, or the v bit is asserted or a non-PCM stream is detected in either channel of one audio pair, the relevant SRC is automatically set to bypass mode.
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The following table describes options for Group (14) Deembedding Control. Table 7-1 Deembedding Control Options Item Repeat Mute Description Upon detection of a de-embedding error, the de-embedder repeats the last good AES sample. Upon detection of a de-embedding error, the de-embedder mutes the current outgoing AES sample.
Audio Synchronization
Parameters > Audio Configuration
By default, APM6803+ synchronizes de-embedded or AES input audio with timing information from the video frame synchronizer prior to re-embedding the audio. You can set the Audio LOV Output Mode parameter so that the audio is either passed or muted in the event of loss of video.
Audio V-Fade
Parameters > Audio Configuration
To enable a smooth deembedded audio V-fade transition when switching video sources, make the following settings: Set the Audio V-Fade parameter to Enable. Set the Audio LOV Output Mode parameter to Mute. Set the ADS Clean parameter to Yes.
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Note: For proper operation in this mode, set the user delay parameters for all audio channels to the same value.
Depending on the setting of the ADS Clean parameter, the Audio Group (14) Embedding Mode parameters have different options. Table 7-2 ADS Clean Parameter ADS Clean State Yes No Audio Group (14) Embedding Mode Options Off Append Off Append OverWrt Auto
The audio embedder component in the APM6803+ is composed of several smaller subcomponent blocks: One ancillary data stripper (ADS) - This block removes all ancillary data packets in the input SDI stream, prior to embedding. Four audio embedding subcomponents - Each subcomponent has the ability to operate on only one audio group, either appending or overwriting a predetermined group onto the SDI stream.
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The audio embedding modes are Append, OverWrt, and Auto available from the Audio Group (14) Emb Mode parameter. Table 7-3 Embedding Mode Options Options Append Description Attempts to insert the audio data and control packets immediately following the last existing audio data/control packet in the horizontal ancillary region (see Append Embedding on page 79) Attempts to overwrite existing audio data and control packets of the same group number with the new audio data (see Overwrite Embedding on page 81) Attempts first to overwrite existing audio data and control packets of the same audio group number; failing that, it appends the new audio data and control packets immediately following the last existing audio data/control packet.
OverWrt
Auto
Append Embedding
When you select Append embedding, the APM6803+ attempts to insert the audio data and control packets immediately following the last existing data/control packet in the horizontal ancillary data space (ADS). Append embedding is only valid if the audio group to be embedded does not already exist. Figure 7-1 shows how append embedding appears in the ancillary data space when there is no previous audio or other data.
Figure 7-1 In Append Embedding Mode, Adding Group 1 When No Other Data is Present When auxiliary data exists in the ancillary data space, appended audio appears following that data, as shown in Figure 7-2.
Group 1
Figure 7-2 In Append Embedding Mode, Adding Group 1 When Auxiliary Data is Present
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If you attempt to insert audio into Group 1 when Group 1 audio data already exists in the ancillary data space, an error is returned, as shown in Figure 7-3.
Figure 7-4 Append Embedding Mode, Adding Group 2 Following Group 1 If you insert Group 1 audio when there is pre-existing Group 2 audio in the ancillary data space and no Group 1 audio, the Group 1 audio is inserted following the Group 2 audio, as shown in Figure 7-5.
Figure 7-5 Append Embedding Mode, Adding Group 2 Following Group 1 An audio group cannot be divided. In Append embedding, the audio group is always added following the last block in the ADS. If there is not enough room to append the audio group following the last block of auxiliary data or audio in the ADS, the attempt results in an error, as shown in Figure 7-6.
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Overwrite Embedding
When you select Overwrite embedding, the module attempts to overwrite any existing audio data and control packets of the same group number with the new audio data. This setting is valid only if the audio group to be embedded already exists. If the new sample distribution does not exactly match the existing audio data packet sample distribution, the embedder marks some audio data packets for deletion (DID word is set to 180h). To avoid sample distribution issues, activate the ADS Clean feature (see Channel Word Length on page 83). When you attempt Overwrite embedding and there is no previous audio (as in Figure 7-7), it returns an error because there is nothing to overwrite.
Figure 7-8 Overwrite Embedding Mode When There is Auxiliary Data on Group 1
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Figure 7-9 shows how overwrite embedding appears in the ancillary data space when there is pre-existing Group 1 audio. This operation is successful.
Figure 7-9 Overwrite Embedding Mode When There is Group 1 Audio in the Group 1 Space
To avoid embedding errors, follow these guidelines: Set the embedding mode to Auto. If the group is present, it will be overwritten; if it is not present, it will be appended. The overwrite and append errors are actually warnings that the operation is not being performed. In SD-SDI mode, do not overwrite embedded groups of 20-bit audio samples with groups of 24-bit audio samples. The required extended data packet information appended to that group may overwrite a pre-existing audio group. For 24-bit audio embedding, use the ADS Clean feature provided.
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Table 7-5 describes the frequency and levels of each audio input test tone, available as a selection from each of the Virtual In (18) Source Select parameters. Table 7-5 Audio In Test Tones Test Tone Test Tone 1 Test Tone 2 Test Tone 3 Test Tone 4 Frequency 400 Hz 1 kHz 2 kHz 4 kHz Level -18 dBFS (Fixed) -18 dBFS (Fixed) -18 dBFS (Fixed) -18 dBFS (Fixed)
Output Tones
Parameters > Audio Configuration > Output Router > Output Tone Audio Gain
Table 7-6 describes the frequency and levels of each audio output test tone, available as a selection from Output Tone Audio. Table 7-6 Audio Out Test Tones Test Tone Test Tone 1 Test Tone 2 Test Tone 3 Test Tone 4 Frequency 400 Hz 1 kHz 2 kHz 4 kHz Level -36 dBFS to 0 dBFS -36 dBFS to 0 dBFS -36 dBFS to 0 dBFS -36 dBFS to 0 dBFS
When the incoming video is 3G or 1.5G HD-SDI, the default value for all Audio Ch XX-XX Word Length parameters is 24 bits.
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When the incoming video is SD-SDI, the default value is 20 bits. However, if at least one channel is set to 24 bits, the corresponding embedder enables 24 bit embedding. When the output is sourced from Dolby E Encoder with bit depth 20 bits, Word Lengths 16 bits option is not available.
Dialnorm Translator
The APM6803+ provides a unique way to correct audio levels with dynamic DC-gain control. Parameters to control this are available under Audio Configuration > Output Configuration > Virtual Out X > Gain.
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EAS Control
Note: EAS is an Emergency Alert System used in North America to send emergency information via a single AES stream (stereo) that replaces the audio. EAS control is a high priority control feature that has the ability to replace any audio stream with an EAS source when both the Virtual In 8 "EAS Source Enable" and the Virtual Out "Out x Allow EAS Source Override" parameters are enabled. The audio channels that become active when this feature is enabled are mapped according to Table 7-7. Virtual In 8 is the only available EAS input source path (Audio Configuration > Input Configuration >Virtual In 8). A special parameter EAS Source Enable becomes visible when the Virtual In 8 type is set to either Mono or Stereo.
Note that activation of this EAS control feature does not make the APM6803+ generate an EAS protocol compliant signal (EAS Encoder) nor detect a valid EAS Input signal per FCC Part 11 (EAS). You have to manually select the EAS source signal for activation, routing, and embedding.
EAS
Program In (Virtual In 1-4) Gain EAS (Virtual In 8) Delay Gain
bypass bypass
LC
Dolby Encoder
Once EAS is enabled, you can choose if one or all of the Virtual Outs are switched to the EAS audio stream. Each of the Virtual Outs have an Out x Allow EAS Source Override parameter. When EAS is disabled, the original sources take effect.
Table 7-7 EAS mapping Virtual Out Type Surround 5.1 Original Audio
Sur-L Sur-R Sur-C Sur-Lfe Sur-Ls Sur-Rs
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Table 7-7 EAS mapping Virtual Out Type Surround 5.1+2.0 Original Audio
Sur-L Sur-R Sur-C Sur-Lfe Sur-Ls Sur-Rs Ste-L Ste-R
Stereo Mono
Gain, Delay, and Phase Invert set for Virtual In 8 will also be effective on the EAS audio stream. Loudness Control, Gain, and Phase Invert for Virtual Out are also effective when EAS is enabled. In the Output router, if an audio tone or mute is selected as the source of embedded or AES audio, it is not replaced by EAS. If Dolby is passed out, its content is not replaced by EAS.
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Stereo Mixing
The Stereo channel can be mixed into a single mono channel by doing the following: 1 2 3 4 Configure one Virtual In Type as either Stereo or Surround 5.1+2.0 Configure one Virtual Out Type as Mono Set the source (Source Select) of the Mono to Virtual In x Stereo L+R. Route the Mono channel to output (Output Router).
Virtual Out 1 Type = Surround 5.1 AES1a Source <= Dolby Enc 1 Virtual In 1 Type = Surround 5.1 AES1b Source <= Dolby Enc 1 In 1 Surround L <= AES 2A (IN) In 1 Surround R <= AES 2B (IN) Virtual Out 1 <= Virtual In 1 AES1a Source <=Virtual Out 1 Surround L AES1b Source <=Virtual Out 1 Surround R Force to clear V-bit & Non-PCM bit of Out AES1 to 0 (PCM). Force to clear V-bit & Non-PCM bit of Out AES1 to 0 (PCM). Set V-bit & Non-PCM bit of Out AES1 to 1 (DATA) automatically due to Dolby type. Set V-bit & Non-PCM bit of Out AES1 to 1 (DATA) automatically due to Dolby type.
Auto Correction
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Voice Over
Delay
Gain
Mixing
Voice (Virtual In 7)
Delay
Virtual In 7 (Parameters > Audio Configuration > Input Configuration > Virtual In 7) is supported for configuration as the Voice channel, either in Stereo type (Virtual In 7 Type = Stereo) or Mono (Virtual In 7 Type = Mono) type.
Voice Over functionality is available for Virtual In 1 to Virtual In 6. When the Virtual In Type is Dolby, the decoded PCM channels are mixed with voice channels.
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Table 7-9 Voice over mixing Virtual In Type Surround 5.1 Mixed with Voice Over (Stereo)
Sur-L + VO-L Sur-R + VO-R Sur-C, Sur-Lfe Sur-Ls Sur-Rs
Surround 5.1+2.0
Sur-L + VO-L Sur-R + VO-R Sur-C Sur-Lfe Sur-Ls Sur-Rs Ste-L + VO-L Ste-R + VO-R
Ste-L + VO-L Ste-R + VO-R Mono + VO-L Sur-L + VO-L Sur-R + VO-R Sur-C Sur-Lfe Sur-Ls Sur-Rs
Sur-L + VO-L Sur-R + VO-R Sur-C Sur-Lfe Sur-Ls Sur-Rs Ste-L + VO-L Ste-R + VO-R
Ste1-L + VO-L, Ste1-R + VO-R Ste2-L + VO-L, Ste2-R + VO-R Ste3-L + VO-L, Ste3-R + VO-R Ste4-L + VO-L, Ste4-R + VO-R
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Table 7-9 Voice over mixing Virtual In Type Dolby Digital Decoder Mixed with Voice Over (Stereo)
Sur-L + VO-L Sur-R + VO-R Sur-C, Sur-Lfe Sur-Ls Sur-Rs
Sur-L + VO-L Sur-R + VO-R Sur-C, Sur-Lfe, Sur-Ls Sur-Rs Ste-L + VO-L Ste-R + VO-R
Video Processing
Video Frame Synchronization
Parameters > Video Configuration > Synchronization
The frame synchronizer offers two modes of operation: Delay mode and Synchronizer (Sync) mode that can be chosen using the Frame Sync Mode parameter. In Delay mode, the output video is synchronized to the input video. In Sync mode, the output video is synchronized to the reference video. The reference standard you can use depends on the Output Video Standard you have set, as outlined in Table 7-10.
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Any other combination of reference and output standard triggers a Mismatched Output/ Ref Standards alarm. Table 7-10 Supported Reference and Output Video Standard Combinations in Sync Mode Reference Standard 525i 59.94 Output Video Standard 1080i 59.94 1080p 59.94 1080p 29.97 625i 50 1080i 50 1080i 50(295M) 1080p 50 720p 60 1080p 60 720p 59.94 1080p 59.94 720p 50 1080p 50 1080i 60 1080p 30 1080i 59.94 1080p 29.97 720p 60 1080p 60 720p 59.94 1080p 59.94 720p 50 1080p 50 1080i 60 1080p 60 1080i 59.94 1080p 29.97 1080p 59.94 1080i 50 1080p 25 1080i 50 1080i 50(295M) 1080p 50 1080p 24 1080psf 24 1080p 23.98 1080psf 23.98 1080p 24 1080psf 24 1080p 23.98 1080psf 23.98 1080p 25 720p 50 625i 50 1080p 30 720p 60 720p 59.94 525i 59.94 1080p 25 720p 50 625i 50 720p 59.94 525i 59.94
In both Sync mode and Delay mode, APM6803+ provides several controls to manipulate the output video signal: Horizontal timing Vertical timing Adjustable frame delay Options to determine the output video behavior on loss of input video, including pass, black, grey, and freeze Manually freeze output video on first or second field (interlaced standards), or on the whole frame (all standards).
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As functions are added to a particular audio path, such as Loudness Control, Surround Processing or Dolby encoding/decoding, the audio processing delay will change. In order to simplify system configuration, an Auto Match feature is provided which can automatically delay the video to match the audio delay, and therefore preserve lip sync. The Auto Match feature will: Calculate the audio processing delay. If the card is in Frame Sync mode, it will add additional delay to align the audio delay to the next integer multiple of a frame of video delay. This is called the Audio Compensation Delay. If the card is in Delay mode, the audio compensation delay stays at 0. The video channel will be delayed by the audio processing delay + audio compensation delay.
Example
Assuming the audio signal path is configured as follows: Virtual In 1 (5.1+2.0) -> Surround Proc 1 (MM) -> Virtual Out 4 (5.1+2.0, LC Enabled) The overall delay in this path is 143.18 ms (MM) + 135.66 ms (LC5.1) = 278.84 (ms). Make the following settings: Auto Match A/V Delay to Enabled Video Tracking Channel to Virtual Out 4 The audio processing delay will be added to the video path automatically such that the content of the video signal and the content of audio signal are still aligned. In Delay mode, the compensated video delay is 278.84ms.
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While in sync mode, to guarantee the fixed output phase related to reference signal, the compensated video delay is increased by an integer number of video frames. Therefore, if the system is working in 625i/50 standard, the compensated video delay is 280ms (7 frames). The audio compensation value of 1.16ms would be added on all the audio Virtual In groups, such that the audio processing delay (278.84ms) and audio compensation delay (1.16ms) is also an integer number of frames, and is equal to the video compensation delay (280ms). Since there is only one video channel, but many possible audio channels, the card has to choose a single audio path from which it will determine the necessary video delay. This is set in the Video Tracking Channel. For example, if the Video Tracking Channel is Virtual Out 2, and Virtual Out 2 gets its source from Virtual In 1, the card will analyze the audio delay from Virtual In1 to Virtual Out 2. However, the other three Virtual Out channels may be affected in this case. It is possible to manually adjust the audio delay at the input end to match the A/V delay for all audio channels.
Note: If Virtual Out is multi-grouped, like 5.1+2.0 or 4x2.0, the system will take the first group as the main group. The tracking is only to compensate the delay of the main group. If EAS overwrites Virtual Out 4, the overall delay value will stay on and not be re-calculated.
Audio/Video Align
Parameters > Video Configuration > Video Delay
Audio/Video Align is used for audio to automatically track video delay. When set to enabled, any delay added on the video path (parameters such as Video Delay, Vertical Phase and Horizontal Phase) is automatically added to all audio paths. When disabled, video delay is not added to audio paths.
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AFD, WSS (625 only), and VI (525/625 only) embedded video metadata specify the active area to be displayed from a video input. These codes do not do any aspect ratio conversion themselves, but indicate how the video should appear to systems capable of interpreting the data. Input parameters on APM6803+ modules indicate the presence of AFD, WSS, and VI data when you determine the lines where this data is likely to appear, if present. The Out Aspect Ratio parameter determines the options for the AFD Select and VI Select parameters. Fifteen output parameters (Output > AFD/VI/WSS Embedder) determine the line to embed AFD, WWS, and/or VI codes on, the code(s) to be embedded, and what happens when a code disappears from the input.
When the Out Aspect Ratio parameter is set to 4:3, the AFD Select and VI Select parameters have the following options: 16:9 top 14:9 top >16:9 in 4:3 4:3 full 16:9 l 14:9 l 4:3 a 14:9 16:9 l a 14:9 16:9 l a 4:3 When the Out Aspect Ratio parameter is set to 16:9, the AFD Select and VI Select parameters have the following options: 16:9 full 14:9 p
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>16:9 in 16:9 4:3 p 16:9 prtctd 4:3 p a 14:9 16:9 a 14:9 16:9 a 4:3
Thumbnail Display
When installed in a frame that contains a 6800+ETH resource module, an extra Streaming tab is present in Navigator which displays thumbnail output from the module.
When input video is switched between two sources while both sources are within vertical blanking, use the Fast Switch parameter to enable fast video switching between the sources. In this mode, output video is not frozen when both sources are within the vertical blanking area when the switch takes place.
GPI Configuration
The APM6803+ provides a flexible configuration of GPI inputs and outputs. GPI Input Configuration GPI Output Configuration Custom GPI Configuration
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Table 7-11 GPI Input Configuration Input Event Select the GPI function category. Input Event options are dependent on the input trigger settings. For Rising Edge or Falling Edge (input trigger settings), Input Event Options are Preset and Custom. For Active High or Active Low, Input Event Options are EAS, Voice Over, and Custom. If Input Trigger is set to Off, no event options are available. Input Event Source Applicable in case of selecting Input Event as Preset. Select from one of the following preset sources. LC [1-4] Preset UpMix [1-2] Preset DownMix [1-2] Preset MultiMerge [1-2] Preset Input Event Mode Select the event mode. This selection is dependant on the Input Event Source selected. When the selected Input Event Source is an LC Preset, the available options here are: DTS Neural-Ultra Light DTS Neural-Light DTS Neural-Medium DTS Neural-Aggressive BS. 1770-Ultra Light BS. 1770-Light BS. 1770-Medium BS. 1770-Aggressive Bypass Custom [1-5] When the selected Input Event Source is an UpMix/ DownMix/MultiMerge preset, the available options here are: Custom [1-5] GPI Input Source Status (read only) Displays the input source status (Enabled or Disabled)
For example, to configure GPI Input 2 to turn off Loudness Control 2 (using rising edge), configure as follows: GPI Input To Edit: Input 2. Input Trigger: RisingEdge Input Event: Preset Input Event Source: LC2 Preset (LC2 must be enabled) Input Event Mode: Bypass
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8
Overview
Custom Scripting
The APM6803+ provides custom scripting functionality for finer control over certain operations. Common Scripting Guidelines Custom GPI Scripts Parameter Control Scripts
Statements
Scripts are created from several statements. Each statement consists of one condition and several assignments. In each statement, if the condition is satisfied, the assignments will take effect. The number of characters (including spaces) in one command line is limited to 251
IF Condition
The number of IF command combinations is limited to 30 The number of condition combinations in each IF command is limited to 10 IF [condition] THEN [assignment] [assignment]...
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Comparisons
A condition is created from one or more comparisons. Comparisons can be ANDed together using && and ORed together using || to form a condition. The AND operation always has precedence over the OR operation when AND and OR both exist in a condition. The following comparison operators can be used: == equal > greater than < less than >= greater or equal than <= less than or equal than != not equal
Notes
Physical GPIs are labeled from natural number, from GPI In 1 to GPI In 16, and GPI Out 1to GPI Out 6. However, the IDs of GPIs used in scripting start from 0. For instance, GPI In 3 is called GPI2 in the script and GPI Out 6 is called GPO5. Refer the APM6803+ parameter list for parameter IDs used in scripting.
Error Diagnosis
Whenever scripting is used, ensure the status feedback (Custom Input Status/Custom Output Status/Script Status) is Active. If Parameter scripting is in use, ensure the parameter Activate Scripting (Parameters > General > Parameter Control Script) is set to Enabled. If GPI scripting is in use, ensure the targeted GPI ports are configured as Custom GPI. (Parameters > General > GPI Input/Output)
Keywords
Keywords for the scripting parser are limited to the following and must be in all caps: IF, THEN, PARAM[, ], =, (, ), ==, &&, ||, >=, <=, ==, !=, >, <, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, The following keywords are only acceptable in GPI scripts. GPI0, GPI1, GPI2, GPI3, GPI4, GPI5, GPI6, GPI7, GPI8, GPI9, GPI10, GPI11, GPO0, GPO1, GPO2, GPO3, GPO4, GPO5
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Examples
The following is an example of correct syntax: IF GPI1==0 THEN PARAM[1447]=0 The table below illustrates errors in syntax and how errors are reported. Table 8-1 Error Reporting Incorrect Command IF GPI1=0 THEN PARAM[1447]=0 Status Error:7 Comments == must be used in comparison commands. = is incorrect. = must be used in assignment commands. == is incorrect. There must be no space between PARAM and [ PARAM must be in all caps. pARAM is incorrect. There are 2 errors in this command - GpI is not in all caps and PARRAM has been misspelt. However, only the index of the first error found is reported.
Error:28
IF GPI1==0 THEN PARAM [1447]=0 IF GPI1==0 THEN pARAM[1447]=0 IF GpI1==0 THEN PARRAM[1447]=0
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Comparisons
In a GPI input script, the comparison could be either GPIx == 0 or GPIx == 1, where x is in the range of [0, (number of GPI inputs - 1)], and 0 represents low and 1 represents high. In a GPI output script, the comparison is in the format of PARAM[x]==y, where x is the ID of a device parameter and y is a value for that parameter.
Assignments
In a GPI input script, an assignment is written as: PARAM[x]=y where x is the ID of a device parameter and y is a value for that parameter. In a GPI output script, an assignment is written as: GPOx=0 or GPOx=1 where x is the range [0, number of GPI outputs -1] and 0 represents low and 1 represents high.
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RESULT Parameters Out 1 Surround Sel (PARAM [1448]) Virtual In 1 Sur Virtual In 4 Sur Sur Proc 1 5.1 Out Sur Proc 2 5.1 Out Out 1 Stereo Sel (PARAM [1449]) Virtual In 1 Stereo Virtual In 4 Stereo Sur Proc 1 Aux Out Virtual In 3 Stereo Out 2 Dolby Enc Surround Sel (PARAM [1582]) Virtual IN 1 Sur Virtual IN 4 Sur Sur Proc 1 5.1 Out Sur Proc 2 5.1 Out
ACTION Parameters Out 1 Surround Sel (PARAM [1448]) Virtual In 1 Sur Virtual In 4 Sur Sur Proc 1 5.1 Out Sur Proc 2 5.1 Out
Pre-Requisites
Make the following configuration settings: Go to Audio Configuration > Input Configuration Set Virtual In 1 > Virtual In 1 Type to Surround-5.1+2.0 Set Virtual In 3 > Virtual In 3 Type to Stereo Set Virtual In 4 > Virtual In 4 Type to Surround-5.1+2.0 Go to Audio Configuration > Surround Processing Set Surround Proc 1 > Surround Proc 1 Type to MultiMerge Set Surround Proc 2 > Surround Proc 2 Type to UpMix Go to Audio Configuration > Output Configuration Set Virtual Out 1 > Virtual Out 1 Type = Surround-5.1+2.0 Set Virtual Out 2 > Virtual Out 2 Type = Dolby Digital Go to General > GPI Input Set GPI Input to Edit to Input 2
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Set Input Trigger to Active High Set Input Event to Custom GPI Repeat the above steps to configure Input 3 Go to General > GPI Output Set GPI Output to Edit to Output 1 Set Output Trigger to Active High Set Output Event to Custom GPI Repeat the above steps to configure Output2, Output3, and Output4
Enter the following in the Custom Input Script (continued) text box:
IF GPI1==1 && GPI2==0 THEN PARAM[1448]=8 PARAM[1449]=17 PARAM[1582]=8 IF GPI1==1 && GPI2==1 THEN PARAM[1448]=9 PARAM[1449]=10 PARAM[1582]=9
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Pre-requisites
Set Audio Configuration > Input Configuration > Virtual In 2 > Virtual In 2 Type to Dolby Digital Set Audio Configuration > Input Configuration > Virtual In 4 > Virtual In 4 Type to Surround-5.1+2.0 Set Audio Configuration > Output Configuration > Virtual Out 1 > Virtual Out 1 Type to Surround-5.1+2.0
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Pre-requisites
Set Audio Configuration > Output Configuration > Gain > Dialnorm Translator Source to Follow MetaSmart 1
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Metadata
The APM6803+ offers intelligent metadata processing for both agile and fixed metadata. A high degree of flexibility is provided enabling you to select the source of incoming metadata, monitor and view it, fix any incorrect incoming metadata, define your own metadata, or create fallback metadata. Incoming Metadata on page 108 Metadata through the Serial Port on page 108 Metadata through SDI Input on page 108 Metadata in an Incoming Dolby Stream on page 108 Generating your own Metadata on page 109 Outputting generated metadata on page 110 Using MetaSmart on page 115 Using MetaSmart Example 1 - Creating Fallback Metadata on page 116 Using MetaSmart Example 2 - Agile Metadata on page 117 Using MetaSmart Example 3 - Fixed Metadata on page 117
SDI
De-embedder
SDI Metadata Input
SDI
Embedder
SDI Metadata Output
Metadata Router
Generator 1 Generator 2
MetaSmart inserts a new dialnorm value
Monitor 1 Monitor 2
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Incoming Metadata
Incoming Audio Metadata may be embedded in various sources including the Dolby stream, Serial Port, or SDI Input. First, identify the source of incoming metadata. There are two metadata monitors provided and once metadata is identified, you can view/monitor the details. To check for incoming metadata, you can do the following:
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Chapter 9 Metadata
Note: 5.1 + 2 refers to 1 surround channel and 1 stereo channel; 2 + 2 refers to 2 independent stereo channels; 2 + 2x1 refers to 1 stereo channel and 2 mono channels; 2x2 + 2x1 refers to 2 stereo channels and 2 mono channels and so on. When generating metadata for the Dolby E Encoder (see Generating your own Metadata), the 7.1 or 7.1 screen program configuration is not supported for the Encoder. Ensure that you do not select the 7.1/7.1 Screen Program Configuration option with the Encoder to ensure proper operation. 5 Click Program ID Config to define your metadata.
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Then specify where the selected source should pick its metadata from.For example, if you select Output as Serial Port, select where the Serial Port should get the metadata from.
Generator
The metadata Generator is used to generate standardized Dolby Digital (AC3) metadata or Dolby E metadata in serial format. It can also generate custom metadata, regardless of any dependency. Table 9-1 Dolby Metadata Dependencies
Parameter Default Value Dependencies Available Options Dolby E Dolby AC-3
Frame Rate
29.97 fps
None
NA
Bit depth Note: The parameter "Bit depth" is configurable under Virtual Out -> Dolby Encoder Program configuration
20-bit
None
NA
5.1+2
5.1+2 5.1+1+1 4+4 4+2+2 4+2+1+1 4+1+1+1+1 2+2+2+2 2+2+2+1+1 2+2+1+1+1+1 2+1+1+1+1+1+1 1+1+1+1+1+1+1+1 5.1 4+2 4+1+1 2+2+2 2+2+1+1 2+1+1+1+1 1+1+1+1+1+1 4 2+2 2+1+1 1+1+1+1
Yes
NA
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Chapter 9 Metadata
3/2
None
1/0 (C) 2/0 (L, R) 3/0 (L, C, R) 2/1 (L, R, S) 3/1 (L, C, R, S) 2/2 (L, R, Ls, Rs) 3/2 (L, C, R, Ls, Rs) On Off 56 kbps 64 kbps 80 kbps 96 kbps 112 kbps 128 kbps 160 kbps 192 kbps 224 kbps 256 kbps 320 kbps 384 kbps 448 kbps 512 kbps 576 kbps 640 kbps Complete Main Music and Effects Visually Impaired Hearing Impaired Dialogue Commentary Emergency Voiceover / Karaoke
Yes
Yes
On
Not 1/0 or 2/0 acmod 1/0: 56640 kbps 2/0: 96640 kbps 2/1: 128640 kbps 3/0: 128640 kbps 3/1: 192640 kbps 3/2: 224640 kbps
Yes
Yes
448 kbps
N/A
Yes
Bitstream mode
Complete Main
Karaoke mode should only be available if the acmod is not 1/ 0. Otherwise, selecting 1/0 acmod should display Voiceover or Reserved. None
Yes
Yes
Dialog normalization
27 dB
31 to 1 dB
Yes
Yes
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Bitstream Information
3 dB
Only when Center channel is used, (acmod 3/0, 3/1, 3/2); grayed out if extended BSI enabled. Only when surround channels are used (acmod 2/ 1, 2/2, 3/1, 3/2); grayed out if extended BSI enabled Only when 2/0 mode is selected None
3 dB 4.5 dB 6 dB
N/A
Yes
3 dB
3 dB 6 dB -Inf dB
N/A
Yes
Surround Mode (Dolby Surround mode) Aud. Production Info (Audio production info exists) AC-3 Mix Level (Mixing level)
Not indicated
Not indicated Not surround encoded Surround encoded On (exists) Off Lt/Rt downmix
Yes
Yes
No
always enabled
Yes
105 dB SPL Audio production information exists is on Small room Audio production information exists is on None
Yes
Yes
Room type
Yes
Yes
On
Yes
Yes
Original bitstream On
None
Yes
Yes
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Chapter 9 Metadata
Ext BSI1 Present (Extended bitstream enabled) Ext Stereo Downmix (Stereo downmix preference) Ext LtRt Ctr Mix Lvl (Lt/Rt Center mix level)
Enable
None
Enable Disable
always enabled
Yes
Lt/Rt downmix
Extended BSI 1 enabled; acmod set to 2/1, 3/0, 2/2, or 3/2 Extended BSI 1 enabled; acmod set to 3/0, 3/1, 3/2 (three front channels in use)
Not indicated Lo/Ro downmix Lt/Rt downmix 3 dB 1.5 dB 0 dB 1.5 dB 3 dB 4.5 dB 6 dB 0 (-Inf dB) 1.5 dB 3 dB 4.5 dB 6 dB 0 (-Inf dB) 3 dB 1.5 dB 0 dB 1.5 dB 3 dB 4.5 dB 6 dB 0 (-Inf dB) 1.5 dB 3 dB 4.5 dB 6 dB 0 (-Inf dB) Enable Disable Standard HDCD
Yes
Yes
3 dB
Yes
Yes
Ext LtRt Surr Mix Lvl (Lt/Rt surround mix level) Ext LoRo Ctr Mix Lvl (Lo/Ro Center mix level)
3 dB
Extended BSI 1 enabled; acmod set to 2/1, 2/2, 3/1, or 3/2 Extended BSI 1 enabled; acmod set to 3/0, 3/1, 3/2 (three front channels in use)
Yes
Yes
3 dB
Yes
Yes
Ext LoRo Surr Mix Lvl (Lo/Ro surround mix level) Ext BSI2 Present Ext A/D Type (A/D converter type) Ext Surr EX Mode (Dolby Surround EX mode)
3 dB
Extended BSI 1 enabled; acmod set to 2/1, 2/2, 3/1, or 3/2 None Extended BSI 2 enabled
Yes
Yes
Enable Standard
Yes Yes
Extended BSI 2 Not indicated enabled; acmod Dolby EX encoded set to 2/2 or 3/2 Not Dolby EX encoded
Yes
Yes
115
Input Filtering DC Highpass Flt (DC highpass filter) B/W Lowpass Flt (Bandwidth-limiti ng lowpass filter) LFE Ch Lowpass Flt (LFE lowpass filter) Surround Channel Processing Srnd Phase Shift Flt (90-degree phase shift) Srnd Ch Attenuator (3 dB attenuation) Dynamic Range Control Line Mode Profile (Dynamic range control Line mode profile)
On
None
Yes
Yes
On
None
Yes
Yes
On
Active only when LFE is enabled Active only when surround channel is enabled Active only when surround channel is enabled None
Yes
Yes
On
On Off
Yes
Yes
Off
On Off
Yes
Yes
Film Standard
None Film Standard Film Light Music Standard Music Light Speech None Film Standard Film Light Music Standard Music Light Speech
Yes
Yes
Film Standard
None
Yes
Yes
Using MetaSmart
MetaSmart is an intelligent metadata mechanism that enables you to configure overrides to metadata settings, or to define a fallback metadata source. There are two MetaSmart blocks available for configuration. To use MetaSmart, go to Parameters > Audio Metadata > MetaSmart > MetaSmart 1/2 Configuration and define the following: 1 Select Metadata Source Select the source of your metadata. You have the following options: Serial Port 1 or 2 if your metadata source is the serial port Decoder 1 or 2 if your metadata source is a Dolby E Stream SDI Input if your metadata source is from SDI Generator 1 or 2 if your metadata source is internally defined
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Chapter 9 Metadata
Once you select the source, the MetaSmart x Primary Source status parameter will change to Yes if metadata is present in your selected source.
Select Fallback Metadata Source (Optional) Select the fallback metadata source or select Pass to not define any fallback source. You can opt to fallback to a serial port, SDI Input, Dolby E Decoder, or Generator (internally defined metadata). Again, once you select the fallback source, the MetaSmart x Fallback Source status parameter will change to Yes if metadata is present in your selected fallback source.
Edit Dialnorm Setting Select the Dialnorm Setting Type as Fixed or bypass. Select Fixed to specify a fixed dialnorm level. For example, -27 dB. This value will then reflect in the Dialnorm feedback textbox.
Select bypass to accept the dialnorm value from the Source. The source value will then reflect in the Dialnorm Feedback textbox. For example, if your Generator source specifies Dialnorm value as -11, it will reflect that value.
Edit AC3 data rate This parameter is useful when transferring a metadata stream from Dolby E to Dolby D.
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1 2 3 4 5 6
Go to Parameters > Audio Metadata > Generator > Generator 1/2 Configuration Go to Program ID Config and define (fallback) metadata settings through the Generator Go to Parameters > Audio Metadata > MetaSmart > MetaSmart 1/2 Configuration Define the Metadata Source, in this case, Serial Port 1 Set the Fallback Source to a Generator Set Dialnorm Setting to Fixed or bypass as required and provide the dialnorm value. In case of the above, if the metadata stream from serial port 1 is lost (indicated by the MetaSmart Source Present parameter showing as None), the metadata source will automatically switch to the defined Generator.
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Chapter 9 Metadata
119
10
Inputs
Specifications
1.5G HD-SDI Specification (applicable with standard and relay back modules)
1 1080i/p (SMPTE 274M) 720p (SMPTE 296M) BNC (IEC169-8) 75 1080p: 23.98, 24, 25, 29.97, 30 Hz 1080psf: 23.98, 24 Hz 1080i: 50, 59.94, 60 Hz 720p: 50, 59.94, 60 Hz
Return loss
> 15 dB from 5 MHz to 1485 MHz >10dB from 1485 MHz to 2970 MHz
Equalization
Adaptive cable equalization for up to 459 ft (140 m) (typical) of Belden 1694A coaxial cable
With standard back module: Adaptive cable equalization for up to 656 ft (200 m) (typical) of Belden 1694A coaxial cable With relay back module: Adaptive cable equalization for up to 262.4 ft (80 m) (typical) of Belden 1694A coaxial cable.
With standard back module: Adaptive cable equalization for up to 984 ft (300 m) (typical) of Belden 8281 coaxial cable With relay back module: Adaptive cable equalization for up to 820 ft (250 m) (typical) of Belden of Belden 8281 coaxial cable, 1312 ft (400 m) (typical) of Belden 1694A coaxial cable.
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Chapter 10 Specifications
AES/DARS Input
Table 10-2 AES/DARS Input Item Standard Connector Sensitivity Impedance Return loss Common mode rejection Input audio rate Maximum input signal Bits Channel status and user bit Balanced/DARS Specification AES 3 44-pin connector; or 3-pin connector, female XLR with external balun <200 mV 110 20% (0.1 to 6 MHz) N/A 0 V to 7 V (0 kHz to 20 kHz) 32 kHz to 108 kHz (DARS 48 kHz only) 10 V pk-to-pk 16, 20, or 24 Unbalanced/DARS Specification AES 3, SMPTE 276M BNC (IEC 169-8) <100 mV 75 > 25 dB, 0.1 to 6 MHz N/A 32 kHz to 108 kHz (DARS 48 kHz only) n/a 16, 20, or 24
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Outputs
SDI Video Output
Table 10-3 SDI Video Output 3G HD-SDI Specification Item (applicable with standard back module only) 4 synchronized 1080p (SMPTE 424M) 1080p DL (SMPTE 372M) 1.5G HD-SDI Specification (applicable with standard and relay back modules) 4 synchronized 1080i (SMPTE 274M) 720p (SMPTE 296M) SMPTE 292M with SMPTE 299M embedded audio Frame rate 1080p: 50, 59.94, 60 Hz 1080i: 50, 59.94, 60 1080p: 23.98 (p/psf), 24 (p/psf), 25, 29.97, 30 Hz 720p: 50, 59.94, 60 Hz Connector Impedance Return loss BNC (IEC169-8) 75 > 15 dB from 5 MHz to 1485 MHz >10 dB from 1485 MHz to 2970 MHz Signal level D.C. offset 800 mV 10% 0.0 V 0.5 V 800 mV 10% 0.0 V 0.5 V < 270 ps, within 100 ps of each other < 10% of amplitude >100 kHz: < 0.2 UI (135 ps) pk-to-pk >10 Hz: < 1 UI (675 ps) pk-to-pk 800 mV 10% 0.0 V 0.5 V 0.41.5 ns < 10% of amplitude < 0.2 UI (740 ps) pk-to-pk BNC (IEC 169-8) 75 > 15 dB from 5 MHz to 1485 MHz BNC (IEC 169-8) 75 > 15 dB up to 270 MHz SD-SDI Specification (applicable with standard and relay back modules) 4 synchronized SMPTE 259M-C, 270 Mb/s, 525/625 component
Number Standard
525, 625
Rise and fall time < 135 ps Overshoot Jitter < 10% of amplitude >100kHz: < 0.3 UI (101 ps) pk-to-pk >10 Hz: < 2.0 UI (675 ps) pk-to-pk
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Chapter 10 Specifications
Reference Video
Table 10-5 Reference Video Item Level Signal type Connector Impedance Return loss Specification 1 V pk-to-pk +6 dB/-3.5 dB Analog composite 525/625 or tri-level sync (1080i/p/720p) BNC per IEC 169-8 75 > 40 dB up to 10 MHz (typical)
GPI I/O
Table 10-6 GPI Inputs and Outputs Item Inputs Number of Inputs Connector Trigger Action 12 SAMTEC mini mate header Low, High, Rising Edge, Falling Edge, Active High, Active Low Specification
123
Table 10-6 GPI Inputs and Outputs Item Internal Pull-Up Baud Rate Outputs Number of Outputs Signal Standard Connector Baud Rate 6 TTL Active low or high SAMTEC mini mate header < 10 Kbps Specification +5 V < 10 Kbps
Serial Ports
Two serial ports are integrated into a single SAMTEC mini-mate header and allow encoding dialnorm and other metadata.
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Chapter 10 Specifications
EIA/TIA-232-E
Table 10-8. EIA/TIA-232 Specifications
Item Maximum Output Level (unloaded) Output Level (loaded) Driver Load Impedance Slew Rate Max Driver Current in High-Z Receiver Impedance Receiver Sensitivity Receiver Input Voltage Range Specification < 25 V 5 V to 15 V 3k to 7k < 30 V/ s 6 mA @ 2 V 3 k to 7 k 3 V 15 V
EIA/TIA-422y
Table 10-9. EIA/TIA-422 Specification
Item Input Termination Maximum Output Level (unloaded) Output Level (loaded) Driver Load Impedance Max Driver Current in High-Z Receiver Impedance Receiver Sensitivity Receiver Input Voltage Range Specification 120, jumper selectable < 6 V > 2.0 V 50 100 A 4K 200 mV -7 V to +7 V
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Propagation Delay
Video Propagation Delay
Table 10-10 Video Propagation Delay Video Standard Format 525 625 1080i /59.94 /50 720p /59.94 /50 1080p/59.94 /50 Propagation Delay (s) 43.9 s 43.8 s 9.42 s 9.42 s 4.79 s
Power Consumption
Table 10-12 Power Consumption Specifications Module APM6803+ Power Consumption < 18 W
Start-Up Time
The start-up time for APM6803+ modules is approximately 60 seconds.
Operating Temperature
The operating temperature for APM6803+ modules is 41 to 113F (5 to 45C).
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Chapter 10 Specifications
127
11
Licensing
DTS Licensing
DTS licenses enable a wide variety of DTS Neural audio processing options such as Loudness Control, UpMix, DownMix, and MultiMerge. These options make it possible to offer advanced audio processing for high-definition and surround sound programming using 5.1 and stereo sources (2.0).
DTS Credits
The number of credits determine how many DTS functions are available. The following table provides the number of credits required for each DTS function. Table 11-1 DTS Credits Function UpMix DownMix MultiMerge Loudness Control (5.1) Loudness Control (2.0) Loudness Control (5.1+2.0) Credits Required 3 3 4 3 1 4
You can have a maximum of 16 credits. For example, a combination of UpMix, MultiMerge, and Loudness Control (5.1) will require 10 credits (3+4+3). Credits are flexible. You can use available credits for different functions at different times. For instance, if you have 10 credits, you could use an UpMix+MultiMerge+Loudness Control (5.1) combination at one time and a MultiMerge+Loudness Control (5.1)+Loudness Control (5.1) combination at another time. For more details, see DTS Licenses on page 130.
Copyright 2012, Harris Corporation
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Chapter 11 Licensing
Dolby Licensing
The following types of Dolby licenses are available: Dolby E Encoder licenses Dolby E Decoder licenses Dolby Digital Encoder licenses Dolby Digiti al Decoder licenses A maximum of 2 x Dolby Encoder and 2 x dolby Decoder can be supported on a single APM6803+ module. For more details, see Dolby Licenses on page 130.
Licensing Parameters
Table 11-2 Licensing Parameters Parameter Name Serial Number [RO] License Key DTS Credits Licensed [RO] Dolby E Enc Licensed [RO] Dolby E Dec Licensed [RO] Dolby Digital Enc Licensed [RO] Dolby Digital Dec Licensed [RO] Function Displays the Serial Number. Allows you to input your License Key. Shows number of DTS Credits licensed. Shows number of Dolby E Encoders licensed. Shows number of Dolby E Decoders licensed. Shows number of Dolby Digital Encoders licensed. Shows number of Dolby Digital Decoders licensed. 0 to 16 0 to 2 0 to 2 0 to 2 0 to 2 Options
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Utilization Parameters
Table 11-3 Utilization Parameters Parameter Name DTS Credit Licensed [RO] DTS Credit Used [RO] Dolby E Enc Licensed [RO] Dolby E Enc Used [RO] Dolby E Dec Licensed [RO] Dolby E Dec Used [RO] Dolby Digital Enc Licensed [RO] Dolby Digital Enc Used [RO] Dolby Digital Dec Licensed [RO] Dolby Digital Dec Used [RO] Audio DSP 1 Utilization Audio DSP 2 Utilization Function Shows number of DTS Credits licensed. Shows number of DTS Credits used. Shows number of Dolby E Encoders licensed. Shows number of Dolby E Encoders used. Shows number of Dolby E Decoders licensed. Shows number of Dolby E Decoders used. Shows number of Dolby Digital Encoders licensed. Shows number of Dolby Digital Encoders used. Range 0 to 16 0 to 16 0 to 2 0 to 2 0 to 2 0 to 2 0 to 2 0 to 2
Shows number of Dolby Digital Decoders licensed. 0 to 2 Shows number of Dolby Digital Decoders used. Shows DSP utilization. Shows DSP utilization 0 to 2 % %
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Chapter 11 Licensing
Dolby Licenses
To order licenses for Dolby functions: Provide your existing Serial Number and License Key. You can get this information by accessing your APM6803+ module in a web browser and going to Parameters > General > Licensing. Quote the relevant part number mentioned below and indicate how many licences you need. You can order a maximum of 2 of each kind of Dolby license. Table 11-5 Dolby Software Key Info Part Number 68OPT-DEE 68OPT-DED 68OPT-DDE 68OPT-DDD Description Software Key License for Dolby E Encoder. Software Key License for Dolby E Decoder. Software Key License for Dolby Digital Encoder Software Key License for Dolby Digital Decoder
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A
Overview
The tables in this appendix contain information on the manipulation of bits that occur when using APM6803+ modules. The following items are documented: Manipulating Channel Status Bits (C-Bit) on page 132 Manipulating Validity and User Bits (V-Bit and U-Bit) on page 134 Identifying Audio Characteristics (Audio Sampling Frequency and Word Length) on page 134 Table A-1 Description of Short Forms in the Appendix RX Key N=Not recognized Y=Recognized S=Recognized and stored, passed-through, or both TX Key N=Not transmitted Y=Transmitted Sample Rate Indication Audio Word Length Byte 0 Bits [6,7], Byte 4 Bits [3,4,5,6] [00,0100] = 96 kHz [01,0000] = 48 kHz [11,0000] = 32 kHz [00,1000] = 24 kHz [00,0101] = 88.2 kHz [10,0000] = 44.1 kHz [00,1001] = 22.05 kHz Byte 2 Bits [0,1,2] [000] = Maximum word length 20 bits (auxiliary bit use not indicated) [001] = Maximum word length 24 bits (auxiliary bits used for audio) Byte 2 Bits [3,4,5] [100] = Encoded word length =Maximum word length4 bits [101] = Encoded word length =Maximum word length0 bits
132
RX ignores bits TX sets bit to [0] RX ignores bits TX sets bits to [01]
6 to 7
0 to 3
4 to 7
0 to 2
3 to 5
133
Table A-2 Channel Status (C-Bits) Data Description Byte 2 Bit 6 to 7 Function [00] Alignment level not indicated [01] Alignment to SMPTE RP155 [10] Alignment to EBU R68 [11] Reserved bit 7 = 0: Channel number bit 7 = 1: [0,1,2,3] Channel number [4,5,6] Multi-channel mode [0] Undefined multi-channel mode [1] Defined multi-channel mode [00] Not a reference [01] Grade 1 reference [10] Grade 2 reference [11] Undefined Reserved RX N TX Y Remarks RX ignores bits TX sets bits to [00]
0 to 6
0 to 1
N N N N N
Y N N N Y
3 to 6
[0000] Not indicated [1000] 24 kHz [0100] 96 kHz [1100] 192 kHz [1001] 22.05 kHz [0101] 88.2 kHz [1101] 176.4 Hz [1111] User defined [0] Sample frequency not scaled [1] Sample frequency scaled by 1/ 1.001 Reserved
0 to 7
6 to 9
0 to 7
10 to 13
0 to 7
14 to 17
0 to 7
18 to 21
0 to 7
22
0 to 3
Reserved
22
134
Table A-2 Channel Status (C-Bits) Data Description Byte 22 Bit 5 Function Bytes 6 to 13 reliability flag RX S TX Y Remarks RX ignores bits TX passed bits unmodified 22 6 Bytes 14 to 17 reliability flag S Y RX ignores bits TX passed bits unmodified 22 7 Bytes 17 to 21 reliability flag S Y RX ignores bits TX passed bits unmodified 23 0 to 7 CRC Y Y RX ignores bits TX calculates CRC on output
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Solution
Ensure you have specified the proper module slot. See your 6800+ Frame Installation and Operation Manual for more information about slot identification. Verify whether there is an ICE6800+ or 6800+ETH module installed in the frame. FR6802+ frames only use ICE6800+ modules to communicate with CCS software and hardware. FR6802+QXF and FR6822+ frames use ICE6800+ or 6800+ETH modules to communicate with CCS software and hardware. Remove any legacy 6800 series product that is in the frame. CCS software cannot communicate with legacy 6800 series products, even though these modules may operate with card-edge controls in an FR6802+ or FR6822+ frame. Legacy 6800 products do not have the + symbol on their extractor handles. Check for pressed pins on the back module by following this procedure: Unplug the front module. Unscrew and remove the back module. Inspect the 20- or 30-pin spring connector at the bottom of the back module (Figure B-1 on page 136), and verify that the connector does not have any slightly bent or pressed pins. Carefully reposition any bent pins. If this is not possible, contact Customer Support.
136
Problem
The frame is powered up, but CCS Software does not communicate with the module.
Solution
Ensure you have installed the correct cables between the PC and the frame: A null modem serial cable must be used between a PC and an FR6802+ frame (not a straight-through cable). At minimum, this requires that pins 2 and 3 are crossed, but pin 5 (ground) is straight-through to pin 5. A USB cable must be used between a PC and an FR6802+QXF frame. Verify whether there is an ICE6800+ or 6800+ETH module installed in the frame. CCS Software control is disabled if an ICE6800+ or 6800+ETH control module is installed in the frame. Set the Frame ID DIP switch settings correctly (the Frame ID DIP switch is located at the rear of the frame.) If only one frame is connected, set the DIP switch to 00. If multiple frames are used in a daisy-chain, set the DIP switch numbers consecutively to the following: 00 (frame 1) 01 (frame 2) 10 (frame 3 11 (frame 4) See your Frame Installation and Operation Manual for more information about frame ID switch settings. Check that the correct COM port is configured in CCS Software and that another application is not using that COM port.
137
Problem
CCS software sees the frame, but do not find all of the modules.
Solution
Remove any legacy 6800 series products. Plug your modules in before starting the discovery. Start your discovery after the frame and all modules have fully powered up. Refresh the CCS software and ensure that the installed modules are fully powered up first before discovery.
Problem
CCS Software does not respond after it is launched.
Solution
Close any CCS software that is already launched.
Problem
CCS software shows a module in the Control window, but cannot control it.
Solution
1 2 3 4 Set the modules Local/Remote jumper to Remote. Ensure the module name in the Control window matches the module type in the frame. Gently push the module into its slot in the frame to ensure it is seated properly and powered up. Verify that the Control window indicates the device is ready.
Solution
1 2 3 Wait four minutes for the frames to recover from the power failure. Close the CCS software, and then restart the PC. Restart the software application.
138
Problem
The module does not seem to work.
Solution
Ensure the correct frame is powered up. Verify that all appropriate rear connections are secure. Gently push the module into its slot in the frame to ensure it is seated properly. Ensure the back module does not have bent pins by following this procedure: Unplug the front module. Unscrew and remove the back module. Carefully reposition any bent pins. If this is not possible, contact Customer Support. Note: Pressed pins are ok and will not affect the functionality.
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Index
A
ADS Clean 78 AFD/WSS/VI 94 Alarms 34 List 35 Options 34 APM6803+ Block Diagram 2 Features 4 Frames Supported 24 Unsupported 24 Monitoring 6 Operating Temperature 125 Overview 2 Parameters 28 Card-Edge Controls 29 CCS Software 30
Parameters 56 Audio Synchronization 77 Audio Test Tones 83 Audio V-Fade 77 Audio/Video Align 93
B
Block Diagram 2 Breakout cable 11
C
Card Edge LEDs 32 Card-Edge Controls 29 Components Cables Breakout 11 GPI Connectors 15 GPI I/O 1 15 GPI I/O 2 15 Modules Back 9 Main 7 Sub 8 Overview 6 Riser Board 8 Serial Port RS-232 14 RS-422 14
Complete List 28 Power Consumption 125 Power Ratings 23 Startup Time 125 System Block Diagram 2 Virtual Streams 38 APM6803+RLYT 20 APM6803+T 20 Audio Embedding Errors 82 Modes 78 Append 79 Overwrite 81 Audio Options DownMix 57 Parameters 58 Loudness Control 64 Parameters 65 Working 69 MultiMerge 59 Parameters 60 UpMix 55
D
Discovery 30 Dolby Dolby Digital Decoder 52 Dolby E Decoder 45 Free Mapping 46 Dolby E Encoder 48, 53 Dolby E Start Line 48 Free Mapping 48 Processing 45 DownMix 57
Copyright 2012, Harris Corporation
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Index
Parameters 58
E
EAS Control 85 ETH Card 26
F
Factory Defaults 30 Fast Video Switch 95 Firmware Upgrading 26
Incoming 108 Dolby 108 Serial Port 108 MetaSmart 115 Overview 107 Module Back Relay 25 Module Status LEDs 33 Monitoring 6 MultiMerge 59 Parameters 60
G
GPI Configuration 95 Input 95 Output 97 GPI Connectors 15
O
Out Aspect Ratio 94 Outputs AES Audio 122 Reference Video 122 SDI Video 121
I
Input Audio Rate 76 Inputs AES/DARS 120 SDI Video 119
P
Parameters 30 ADS Clean 78 Audio Processing 76 Audio Delay Ranges 83 Audio Embedding Modes 78 Audio Synchronization 77 Audio Test Tones 83 Audio V-Fade 77 EAS Control 85 Input Audio Rate 76 Sample Rate Convert 76 Out Aspect Ratio 94 Video Processing 90 AFD/WSS/VI 94 Audio/Video Align 93 Fast Video Switch 95 Thumbnail Display 95 Video Delay Ranges 92 Video Frame Synchronization 83, 90 Power Ratings 23 Propagation Delay Audio 125 Video 125
J
Jumpers Main Module 21 Sub Module 22
L
LEDs Main Module 32 Card Edge 32 Module Status 33 Sub Module 34 Licensing Adding License Key 129 Dolby 128 DTS 127 Credits 127 Parameters 128 Loudness Control 64 Parameters 65 Presets 72 Status 71 Groups 71
R
Relay Back Module 25
M
Main Module 7 Main Module LEDs 32 Metadata Generating 109
Copyright 2012, Harris Corporation
S
Sample Rate Convert 76 Signal Flow Audio Path 16 Metadata Path 17 Video Path 16
141
Status AAP Status 40 Failure Messages 41 Processor Utilization 41 Resource Utilization 40 Sub Module 8 Sub Module LEDs 34 System Block Diagram 2
T
Thumbnail Display 95
U
Upgrading firmware 26 UpMix 55 Parameters 56
V
Video Delay Ranges 92 Video Frame Synchronization 90
142
Index