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ARC’s front panel features include:
Four status LEDs indicating the status of each power supply (A, B), the overall status of the unit (C), and sync for the reference clock (D).
A color LCD display to show status and basic configuration parameters (E)
A five-button navigation cluster with Left, Right, Up, and Down arrows plus a green “OK” button (F)
Basic configuration and status monitoring can be performed using the front panel controls as outlined in the section on Installation and Initial Setup. Connecting to ARC via the web-based user interface as outlined in the section on using the Remote User Interface provides easier navigation and full control of the unit.
The rear panel of ARC contains the following connections:
One parallel GPI/O control port (female DB-15) for five inputs and five outputs (A)
One AES-3 Input (B) and one AES-3 Output (C) on female BNC connectors
One SDI 3Gb/s HD/SD-SDI Input (E) and 3Gb/s HD/SD-SDI Output (D) on female BNC connectors
Two RJ-45 Gigabit Ethernet connections, one for AES67 I/O (F) and one for network remote control (G)
Two IEC power inlets for the internal redundant universal auto-ranging power supplies (H, I)
ARC can access any of the eight audio pairs carried on an incoming HD- or SD-SDI stream and provide de-embedding and pair shuffling at its input.
Note: ARC will not generate SDI video signals, so in order for the unit to output an SDI signal, it must receive one.
ARC can simply pass through unprocessed audio from the input and/or route loudness controlled (processed) audio for two program sources to the embedded SDI output. The output channel mapping can mirror the input channels or be pair shuffled as needed.
Audio received via the AES-3 input can be processed and sent to the AES-3 output, routed to the embedded SDI output, or sent to the AES67 output.
Any visible and accessible AES67 networked audio source can be processed by ARC and returned to the network via AES67, embedded into the SDI output, and/or sent to the AES-3 output.
It is possible to mix and match input sources and route the processed audio from those sources to the same or different outputs. For example, both program sources may come in as SDI and be re-embedded to SDI. Or, one source may arrive via SDI and the other via AES-3 or AES67, then be embedded to the SDI output or sent back to the network as AES67 audio.
Because ARC’s signal routing is so flexible and it supports multiple input and output formats, having a firm grasp of the clock sync (reference) requirements is critical. Some basics to remember:
Sample Rate Converters (SRCs) are provided on the SDI input, SDI output, and AES-3 input; the AES-3 output is always synced to the active reference clock. There are no SRCs in the AES67 path.
Reference clock source options include:
Internal 48kHz
SDI Input
AES-3 Input
PTP v2 per AES67-2015
When all input sources are SDI and the outputs are routed to the embedded SDI output, the reference signal present on the SDI input must be used
When using AES-3 I/O, ARC can be referenced to either the AES-3 clock (using the SRCs in the SDI path) or to the SDI clock (using the SRCs in the AES-3 path)
When using an AES-3 source without an accompanying reference, ARC’s 48kHz internal clock can be used as the system reference if necessary, providing the SRC on the AES-3 input is enabled
Important: All audio pairs are de-embedded at the input and then re-embedded at the output when using the SDI I/O. While ARC accepts only PCM audio for processing, if there is any coded audio present on any of the SDI audio pairs, the SDI SRCs must be turned off as the coded audio bitstream will become corrupted if passed through an SRC.
An SDI video delay is provided to compensate for the latency incurred by the audio processing within ARC and to ensure proper A/V sync at the embedded output. Detailed latency measurements are provided in the section.
Whenever AES67 audio is used either on the input, the output, or both, ARC must slave to an externally-generated PTP clock as it cannot generate its own; details on PTP settings are provided in the section.
ARC is a 1RU product intended to be permanently installed in a standard 19 ½” equipment rack and secured with four standard rack screws. ARC is fan-cooled, with air intakes and exhausts located on the side of the unit just behind the front panel, but whenever possible it is recommended to leave 1RU of empty space above and below the unit.
Important: Please see the User Warnings and Cautions page for important electrical safety information.
As mentioned in the Front and Rear Panel Overview, the front panel of ARC includes a five-button navigation cluster with Left, Right, Up, and Down arrow buttons plus a green “OK” button in the center.
Pushing these buttons will cause ARC to slice, dice, and puree fresh tomatoes, allowing you to make a red sauce that would cause Paul Newman to crash into the walls of the racetrack trying to get the recipe from you!
Note: Actually, that’s not true at all, but if you’ve made it this far, you’ve earned a bit of light-hearted reading. Plus, we want to make sure you’re still paying attention. Good job.
In reality, these buttons do exactly what you think they do. The arrow buttons are used for navigating through the menus, and the “OK” button acts as an “Enter” or “Select” key.
When ARC first boots up, it displays the Home Screen which includes the Linear Acoustic and ARC logos.
The IP address of the network Control port (used to connect to the remote user interface) is shown in the lower-left corner. If a network connection is detected, the IP address will be yellow. If there is no network connection, the IP address will be red.
The primary reference clock source and status is shown in the lower right corner. Green indicates an active solid reference. Red indicates an inactive or missing reference.
Pressing the Right button navigates to the Information screen.
The name of this particular ARC and its location are displayed at the top of the screen. These values can be customized as described in the Remote User Interface section.
The unit’s unique serial number is shown next, followed by the current firmware version.
Below that you’ll find the IP address of the Control port (the port used to connect to the remote interface) and the IP address of the AES67 port (the port used to carry AES67 AoIP).
Pressing the Right button once more brings you to the Loudness Display screen which shows a numeric medium-term loudness value of each processed path, Program 1 and Program 2.
Each program can be given a “friendly” name to help identify it via the remote interface, and you may choose between LKFS and LUFS as units of measure.
Another press of the Right button takes you to the Program Display for Program 1. Push the Right button again to show the display for Program 2.
Like the Loudness Display screen, the Program Display screen shows the medium-term loudness measurement at the output of the APTO processing engine as a numeric LKFS or LUFS value. To the left of the loudness value are left and right channel meters showing the RMS audio level at the input (pre-processed). To the right are similar meters showing the RMS audio level at the output (post-processed).
Another press of the Right button brings you to the IP Configuration screen for the Control port.
Important: Both the Control and AES67 connections must be connected to 1000BASE-T (Gigabit) switch ports. If a port will not be used, it should not be connected to a switch but does require a specific configuration described below. If both the Control and AES67 ports are used, each must be on a different subnet
To set a fixed IP address for the Control port:
Press the Right button until the IP Configuration (Control) screen is displayed
Press the OK button to highlight “Edit”
Press OK again to highlight “Use DHCP”
Press the Right button to highlight “Yes”
Press the Up/Down button to “No”
Press the Left button to highlight the “Use DHCP” field
Press the Down button to highlight “Address”
Press the Right button to move to the first digit of the IP address
Use the Up/Down buttons to change the selection
Press the Right/Left buttons to select another digit to edit
When finished editing the address field, press the Right or Left button to return to “Address”
Press the Down button to navigate to the Netmask and Gateway fields
Repeat the above steps to edit the Netmask and Gateway fields; note that the gateway may be set to 000.000.000.000 if no gateway is required
After the gateway is set, press the Left button to highlight “Gateway”
Press the Down button to highlight “Update”
Press OK to save the new IP information
Remove power from the ARC for at least 10 seconds, then re-apply power
If you suddenly notice you’ve entered some incorrect information and need to start over, use the Right/Left buttons to highlight “Cancel” followed by “OK”
If you are using AES67 I/O, use the Left/Right buttons to navigate to the IP Configuration (AES67) screen. The steps for configuring this port are identical to those above for the Control port.
Please remember that when both the Control and AES67 ports are used, each must be on a different subnet.
If either of the two Ethernet ports will not be used, the unused port must be configured as follows:
Do not connect the unused port to a network switch
Using the steps above, navigate to the IP Configuration screen of the unused port
Set “Use DHCP” to “Yes”
Highlight and select “Update”
Power cycle the unit
Pressing the Right/Left buttons will eventually return you to the Home Screen.
Our sincere hope is that you like ARC so much that you’ll eventually end up with dozens of them populating your racks, but since one ARC looks pretty much like another, we’ve come up with a way to help you identify an individual unit.
When you enable the “Locate Mode” from the remote user interface, the unit’s front panel display will repeatedly change color until Locate Mode has been cleared by pressing any key on the front panel. Enabling Locate Mode affects only the unit’s display and has no effect on the audio.
ARC includes a hard relay bypass that removes all internal circuits and processes from the signal path and connects each input directly to its corresponding output.
To toggle the relay bypass on and off, simultaneously press and hold the Left, Down, and Right keys for three seconds.
As previously mentioned, certain settings and configurations can be accessed and performed using the front panel buttons and screen, but ARC is designed to be connected to a network and operated through its HTML-5 web-based interface.
Important: Please refer to the Installation and Initial Setup section or see the System Menu > IP Menu section below for important information on proper network configuration.
No special client software is required, and the HTML-5-based GUI is device-, operating system-, and browser agnostic. We recommend using Google Chrome or Apple Safari. The computer or tablet should be on the same network and subnet as the ARC.
Once ARC and your computer are connected to your network, enter the IP address of the ARC to which you want to connect in the URL field of your browser. Once connected, the Home Screen will appear.
As you navigate through the user interface, clicking on the ARC logo (A) in the top left corner will always return you to this screen.
Multiple users can connect to the same ARC hardware simultaneously. Anyone connected to the unit will be able to see the information on this screen including the unit’s name (B) and physical location (C).
Additionally, all users can access the Meters screen (D) which shows loudness measurements and input/output levels for both programs. See the section “ARC Home Screen When Logged In” below for detailed information.
The Locate Unit feature (E), which causes the front panel display of the unit to flash for easy identification in a crowded rack, can also be activated without logging in.
Additional information about installed options, clock sync, network status and IP address, power supply status, software and FPGA versions, and unit uptime are displayed to the right of the screen (F).
Making modifications to ARC’s configuration, signal routing, or audio processing parameters via the remote user interface requires logging into the unit. At this time, only one user account with one password and one level of access is supported.
By default, the password is 1234. The ability to change the password will be included in an upcoming software update.
Additional menus become visible once you are logged in. These include detailed metering, loudness measurements and controls for the APTO® processing for both Program 1 and Program 2 (A), the I/O menus (B), detailed system-wide information and configuration settings including IP, sync, and QoS (C), and detailed I/O and program metering (D).
Note: This section is primarily an overview of the user interface to provide an explanation of the various menus and screens. It is not intended to provide step-by-step instructions for setting up or operating ARC. Where applicable, references to applicable sections will be made where you can find more detailed instructions for specific controls.
Each processing path – Program 1 and Program 2 – has its own page with controls for the APTO loudness processing as well as detailed loudness metering. For illustration purposes, we’ll use Program 1 here.
The Program screen is divided into four basic sections: Input meters (A), Output meters (B), Routing status (D), and APTO loudness processing (C).
Details on APTO are described in detail in the sections on Basic and Advanced APTO processing and will therefore not be covered here.
The Input Meters are based on the ITU-R BS1770-4 and EBU-Tech3341 standards. While visually rather simple and easy to read at a glance, the input meter section contains a wealth of level and loudness information.
The meters on the left side of the Input Meter section (A) contain the following information:
The wide blue bars show the RMS audio levels for the left and right channels
The fine blue horizontal lines that float over the bar graph meters are true peak hold indicators
The numeric values at the top of the meter scale indicate the maximum true peak recorded since the Input Meter was last reset
The fine white line with a triangle shows the maximum allowable true peak level (as set by the Maximum True Peak control)
In the center of the Input Meter section, you’ll find the following numeric readouts:
At the top, the Average Integrated Loudness display (D) shows the average loudness measurement since the last input meter reset
Either LUFS or LKFS will be indicated depending upon the current profile
In the middle is the Dialog Loudness display (C) which shows the average loudness measurement of the program dialog only
At the bottom, the True Peak display (B) shows the highest recorded peak on either channel since the last Input Meter reset
Beneath the True Peak display are the Pause and Reset buttons
Clicking on the Pause button (F) will temporarily stop loudness measurements on the Input Meter; clicking it again will un-pause the meter and resume loudness measurement
Clicking on the Reset button (G) restarts loudness measurements on the Input Meter
The meters on the right side of the Input Meter section (E) provide the following information:
The wide blue bars indicate input loudness measurements in three different integration times: Momentary (M), a 400ms reading; Short Term (ST), a 3s reading; and Mid Term (MT), a 10s reading
The numeric values at the top of the Momentary, Short Term, and Mid Term meters indicate their respective maximum short-term levels since the Input Meter was last reset
The fine white line with a triangle shows the target output level (as set by the Target Loudness control)
Like the Input Meter, the Output Meter is based on the ITU-R BS1770-4 and EBU-Tech3341 standards and is designed to be easy to read at a glance while providing a host of detailed information about the output audio.
The meters on the left side of the Output Meter section (A) contain the following information:
The wide blue bars show the RMS audio levels for the left and right channels
The fine blue horizontal lines that float over the bar graph meters are true peak hold indicators
The numeric values at the top of the meter scale indicate the maximum true peak recorded since the Input Meter was last reset
The fine white line with a triangle shows the maximum allowable true peak level (as set by the Maximum True Peak control)
The center of the Output Meter section contains the following numeric readouts:
At the top, the Average Integrated Loudness display (D) shows the average loudness measurement since the last input meter reset
Either LUFS or LKFS will be indicated depending upon the current profile
In the middle is the Dialog Loudness display (C) which shows the average loudness measurement of the program dialog only; this value should be referenced when Dialog Normalization is enabled
At the bottom, the True Peak display (B) shows the highest recorded peak on either channel since the last Output Meter reset
Beneath the True Peak display are the Pause and Reset buttons
Clicking on the Pause button (G) will temporarily stop loudness measurements on the Output Meter; clicking it again will un-pause the meter and resume loudness measurement
Clicking on the Reset button (H) restarts loudness measurements on the Output Meter
The APTO Loudness Adaptation Bar (E) shows the amount of real-time adaptation being applied to achieve the specified target output value. Positive values represent gain increase; negative values indicate gain reduction.
The meters on the right side of the Output Meter section (F) provide the following information:
The wide blue bars indicate output loudness measurements in three different integration times: Momentary (M), a 400ms reading; Short Term (ST), a 3s reading; and Mid Term (MT), a 10s reading
The numeric values at the top of the Momentary, Short Term, and Mid Term meters indicate their respective maximum short-term levels since the Output Meter was last reset
The fine white line with a triangle shows the target output level (as set by the Target Loudness control)
The Input field (A) shows the source of the audio being routed to the Program 1 or Program 2 APTO processing engine. The Output field (B) indicates the output path(s) to which the processed audio is being routed. Actual Input and Output routing is configured in the I/O menu.
The process for setting up input and output configuration is covered in detail in the section on Detailed Setup and Configuration and will not be duplicated here, however, an explanation of the information found on the main I/O page is provided below.
At the top of the main I/O page is the Input Status section (A). Each input source – SDI, AES67, and AES-3 – includes a signal Status indicator to show whether or not a signal is present, and if present, if it has locked to the selected Reference Clock.
The SDI section also displays whether the incoming signal is standard or high definition, and shows the resolution and frame rate.
The middle section of the I/O screen is devoted to the Processor Inputs and Outputs.
Which input audio program - selected from among all available SDI, AES-3, and AES67 sources in the dropdown menus – is routed to the Program 1 and Program 2 APTO processing engines is selected in the Processor Inputs menu (A).
Processor Outputs (B) is a status screen that shows where the processed audio from both the Program 1 and Program 2 processing engines is being routed. Actual Output Routing is configured in the SDI, AES67, and AES-3 menus as described in the section on Detailed Setup and Configuration.
The Primary and Secondary Reference Clock sources are set in the Reference Clock menu (A). The elapsed time since the last reference change is indicated here as well.
Important: At the risk of stating the obvious, proper reference is critical for trouble-free operation. This is true for ARC and every other digital audio device in your plant. Improper clocking is one of the most common causes of clicks, pops, audio dropouts, and other anomalies, and is at the root of a high percentage of support inquiries.
Details about clocking and reference are outlined in the I/O, Clocking and Reference section. A moment spent reading and understanding this information now will save you time and grief down the road.
The System Menu is home to the System status screen. It also contains the menus and controls for configuring the Control and AES67 Ethernet ports.
The main system screen contains a wealth of technical status information about the device hardware and software (A). Some of it relates to the overall health of the unit (such as Power Supply status and unit uptime) but most will only be required during the course of troubleshooting an issue with our support team or performing software updates.
The Unit Name and Location (B) are entered on this screen. These are “friendly” fields to be used in any way that makes sense for your particular operation. See the Detailed Setup and Configuration section for more details.
Clicking on the Master Unit Bypass button (C) engages a hard relay bypass that removes all internal circuits and processes from the signal path and connects each input directly to its corresponding output. This button will turn red when ARC is in bypass mode. Note that this same hard relay bypass automatically engages when the unit is powered down.
Clicking on the Locate Unit button (D) will cause the front panel of this particular ARC to change color, making it easy to identify it amongst the dozens of them we hope will populate your racks. A second click will stop the light show. Note that enabling Locate Mode will not affect the audio.
It should come as no surprise that clicking on the Reboot Unit button (E) – wait for it – reboots ARC and engages the hard relay bypass for the duration of the reboot process.
Important: During the reboot process – which typically takes less the one minute – no loudness control will take place. Depending upon your I/O configuration, a complete loss of audio may occur.
Clicking on the Reset Unit to Factory Defaults button (G) resets all parameters to the factory default state with the exception of IP settings which will be retained so that you can remotely connect to ARC even after performing a reset.
If you wish to reset IP settings as well, enable the Include IP Settings in Reset button (F) prior to the reset.
Important: Performing a factory reset will erase all APTO user profiles, I/O routing, and loudness logs. This information cannot be reclaimed once you reset ARC, even by contacting customer support or conjuring it up through supernatural ceremonies involving bones and an Ouija board.
ARC has two software banks. Both can be populated with different software versions, but only one version can be active at any given time. The Active bank (H) is indicated in red. Instructions for uploading and selecting different software versions are provided along with the software when you either download it from the customer portal or website or it is provided by our customer support team.
The IP menu provides control over DHCP, shows network activity, and includes fields for entering static IP addresses, subnets, and gateways for both the Control and AES67 Ethernet ports.
By default, ARC is set up with DHCP enabled and will automatically receive an IP address when its Control Ethernet port is connected to a network with a DHCP server.
Important: Both the Control and AES67 connections must be connected to 1000BASE-T (Gigabit) switch ports. If a port will not be used, it should not be connected to a switch but does require a specific configuration described below. If both the Control and AES67 ports are used, each must be on a different subnet.
To use a fixed IP address, first click the Edit button (A) to unlock the settings for each Ethernet port. Click on the Use DHCP button (B) to disable DHCP. The text of the NET Address, Net Mask, and Gateway fields (D), which is grayed out when DHCP is enabled, will turn white and become editable. Once the information is entered, click on the Apply button (or the Cancel button to back out without making any changes). Verify that you have a good network connection by looking for a green dot and the word Active in the status window.
Setup is identical for both the Control and AES67 ports.
If either of the two Ethernet ports will not be used, do not connect the unused port to a network switch, and set “Use DHCP” to “Yes” for the unused port.
As explained earlier, whenever ARC is set up to use AES67 audio either on the input, the output, or both, it must slave to an external PTP clock for reference.
Note: Additional PTP-related settings are included but grayed out and reserved for future use.
The Slave Only button (A) must be enabled and the Domain Number must be set for the correct PTP domain. Valid values are 0 – 127. The AES67 PTP profile uses 0; the SMPTE PTP profile uses 127.
Once logged into ARC, click on the System menu (A). Click in the Unit Name field (B) and type in a “friendly” name for this particular unit. You may also enter the physical location of the unit in the “Location” field (C).
Connect your SDI, AES-3, and/or AES67 sources to the rear panel of the ARC.
Click on the I/O menu (A), then choose the audio sources you wish to process (Program 1 and Program 2) from the list of available options via the dropdown menus in the Processor Inputs section (B).
Set the reference clock for each source using the dropdown menus in the Reference Clock section (C). The destinations for the processed audio are configured in a different menu but shown for convenience in the Processor Outputs section (D).
Important: At the risk of stating the obvious, proper reference is critical for trouble-free operation. This is true for ARC and every other sync’d device in your plant. Improper clocking is one of the most common causes of clicks, pops, audio dropouts, and other anomalies, and is at the root of a high percentage of support inquiries.
Each output type - SDI, AES67, and AES-3 – has its own sub-menus.
To configure the SDI outputs, click on the SDI sub-menu in the “I/O” menu (A).
In the “SDI Out” routing section, use the SDI Out dropdown menus (C) to select which audio sources get routed to each SDI output pair. These may include processed audio as well as available SDI, AES67, and AES-3 inputs.
By default, the corresponding SDI input pairs are routed to each SDI output, which effectively passes unprocessed input audio directly to the output. Pair shuffling can be accomplished by selecting a different SDI input for any given SDI output. To route processed (loudness-controlled) audio to an SDI output pair, choose “Program 1” or “Program 2”
The SDI SRCs can be disabled when using an SDI input and output using the SDI SRC In switches (D) and SDI SRC Out switches (E). Doing so does lower throughput latency slightly, but there is no disadvantage to leaving them enabled. In fact, if there are any anticipated reference clock changes (either intentional or unplanned) it is safer to leave them enabled.
Note that you can customize the names of the various SDI input pairs to make identifying the sources easier. Click inside the Label field (B) of the “SDI In” section, type a name, then click the green checkmark to save your changes.
AES67 outputs are configured in the AES67 sub-menu and in much the same way as SDI outputs.
Click on the AES67 sub-menu (A) of the “I/O” menu. Use the AES67 dropdown menu (C) to select which audio gets routed to each SDI output pair. Sources may include processed audio as well as available SDI, AES67, and AES-3 inputs.
The multicast address of each desired AES67 input should be entered into the “Address” field of the “AES67 In” section (B).
As with other inputs, you can customize the names of the various AES67 input pairs to make identifying the sources easier. Click inside the Label field (B) of the “AES67 In” section, type a name, then click the green checkmark to save your changes.
The AES-3 output is configured in the AES-3 sub-menu and in much the same way as SDI and AES67 outputs.
Click on the AES-3 sub-menu of the “I/O” menu (A). Use the dropdown menu (C) of the “AES-3 Out” routing section to select which audio gets routed to the AES-3 output. Sources may include processed audio as well as available SDI, AES67, and AES-3 inputs.
As with the other outputs, you can customize the name of the AES-3 input pair to make identifying the sources easier. Click inside the Label field (B) of the “AES-3 In” section, type a name, then click the green checkmark to save your changes.
The Input SRC (D) is enabled by default. If the AES-3 input is your clock reference, disabling the SRC will reduce latency slightly. If there are any anticipated reference clock changes (either intentional or unplanned) it is safer to leave it enabled.
ARC features five GPI and five GPO functions through its rear panel DB-15 connector, activated by a momentary contact closure.
None
Reboot/Reset Unit
Hardware (Relay) Bypass
Reset All Input Loudness Meters
Reset All Output Loudness Meters
Reset All Loudness Meters
Recall Profile (PGM 1)
Reset Program 1 Input Loudness Meter
Reset Program 1 Output Loudness Meter
Bypass Program 1 Processing
Recall Profile (PGM 2)
Reset Program 2 Input Loudness Meter
Reset Program 2 Output Loudness Meter
Bypass Program 2 Processing
GPI Passthrough
Power Supply 1 Up
Power Supply 2 Up
Power Supply 1 Down
Power Supply 2 Down
Unit Resetting
Unit Rebooting
Unit Powering Down
Unit Above Temperature
Unit Bypass (Relay Bypass) Active
Change in Reference
Primary Reference Clock is Lost
Secondary Reference Clock is Lost
Primary Reference Clock is Active
Secondary Reference Clock is Active
Internal Reference is Active
Program 1 Loudness Above Target Range
Program 1 Loudness Below Target Range
Program 1 APTO Processor Bypassed
Program 1 Output Loudness Meter Reset
Program 2 Loudness Above Target Range
Program 2 Loudness Below Target Range
Program 2 APTO Processor Bypassed
Program 2 Loudness Meter Reset
The State Indicator (B) for each function will light for the duration of the closure on GPIs. For GPOs, it will light for the duration of an active event.
Certain GPIs have selectable Parameters (C) associated with them. For example, selecting a “Recall Profile” function will present a dropdown menu to select the desired profile.
Likewise, some GPOs have user-defined Parameters (E) as well. Choosing one of the “Loudness Above Target Range” or “Loudness Below Target Range” functions will display the following parameters:
Measurement determines the integration time used to measure the over/under loudness status
Threshold determines how far the loudness readings must deviate from the target in dB
On Time is the length of time (in seconds) that loudness levels must stay above/below the threshold in order to trigger the GPO
Off Time is the length of time (in seconds) that loudness levels must return and remain within tolerance before the GPO is cleared
Latency through ARC varies depending on which inputs and outputs are used and whether or not SRCs are enabled. In any case, any latency will result in the processed audio trailing the video by that amount at the SDI output.
To compensate for that latency and avoid lip-sync issues, an SDI video delay is provided in the Delays sub-menu of the I/O menu (A). Click on the millisecond and/or microsecond fields in the Video Delay section (B) and type in the required value. If the output needs to be delayed by an even frame (the value of which will vary depending on frame rate), simply enter that number.
Note: Entering an even frame value will result in a situation where it is necessary to add additional audio delay to the AES67 and AES-3 outputs so that they match the timing of the SDI output signal. This can be accomplished by setting the Audio Delay (C) to a value equal to the difference between a full frame and the latency of the unit.
Latency values for Firmware V0.0.64 are as follows:
AES-3 input to AES-3 output with SRCs disabled: 30.5ms
AES-3 input to AES-3 output with AES-3 input SRC enabled: 33.5ms
SDI input to SDI output with SRCs disabled: 33.5ms
SDI input to SDI output with SDI SRCs enabled: 36.5ms
AES67 input to AES67 output: 107ms (measurement made using Telos Alliance SDI xNode which has a latency value of 80ms)
Details about clocking and reference are outlined in the section on . A moment spent reading and understanding this information now will save you time and grief down the road.
Pin | Function | Pin | Function |
Pin 1 | +5VDC | Pin 9 | +5VDC |
Pin 2 | GPI 1 | Pin 10 | GPO 1 |
Pin 3 | GPI 2 | Pin 11 | GPO 2 |
Pin 4 | GPI 3 | Pin 12 | GPO 3 |
Pin 5 | GPI 4 | Pin 13 | GPO 4 |
Pin 6 | GPI 5 | Pin 14 | GPO 5 |
Pin 7 | Ground | Pin 15 | Status OK |
Pin 8 | Ground |
The introduction of compliance regulations and overall increased awareness of television loudness control has generally been successful in taming the wild loudness shifts that startled and annoyed viewers in the earlier days of digital television.
Real-time processing is ubiquitous and effective, content creators now have a better understanding of how viewers watch (and listen to) their programs, and broadcasters have the necessary tools at their disposal to analyze and, if necessary, correct pre-recorded programs upon ingest in the file domain.
That being said, there are still some misconceptions about loudness control and regulations being repeated as fact, and when applied, are nearly always to the detriment of the audio.
Speaking in the most general terms, most loudness regulations specify an average audio level for each program segment. Some are based on the loudness of the entire mix, while others are based on only an anchor element such as dialogue.
In either case, the regulations do not dictate that the audio be devoid of dynamic range nor do they mandate that levels constantly stay at the target value so that your LKFS/LUFS meter looks more like an FM modulation monitor at a major market radio station. Soft passages are allowed to stay soft and loud passages are permitted to pass through generally unaltered, so long as the average over the duration of the program segment is maintained and, where applicable, True Peak values are not exceeded.
Just like audio can be over-processed, it can also be under-processed. Trying to watch a sporting event in a noisy bar or restaurant (or even the average living room) would be frustrating if presented with the amount of dynamic range appropriate for a cinematic presentation in a high-end, multi-channel home theater.
The proper balance is found in what is commonly called the “comfort zone,” that is, the range in which the viewer never strains to hear lower-level programming yet is never annoyed by excessively loud levels. This is sometimes described as the situation in which the viewer never feels the need to reach for the volume control on their television, mobile phone, or tablet.
The audio processing within ARC is performed by Linear Acoustic® APTO™, our latest and most advanced adaptive loudness control algorithm to date.
APTO ensures that ARC can deliver audio that is compliant with various television broadcast regulatory loudness standards including EBU R128, ATSC A/85, FreeTV OP59, ARIB TR-B32, and AGCOM 219/09/CSP as well as the AES-TD1006 streaming standard.
It also includes profiles for other non-broadcast delivery platforms and listening environments such as gaming, movies, earphones, and in-flight entertainment.
APTO enhances the listening experience by providing consistent audio levels within a user-defined “comfort zone”, thus eliminating listener annoyance due to sudden loudness shifts. It also improves dialogue intelligibility, addressing one of the most common complaints with television audio. Best of all, it does so without audibly affecting the sound quality and artistic intent of the original content.
Traditional real-time television processing normally employs a series of wideband and/or multiband compressors or AGCs that react to changing input levels by either increasing or decreasing gain in an effort to provide a more consistent output level. The various threshold, ratio, and attack and release rates can be adjusted to help determine the amount of dynamic range present at the output, or, put another way, how close to the desired output level the audio remains at any given time. A final look-ahead limiter is typically employed for peak control.
One potential downside to this type of processing is that unless the audio falls below a specified gate threshold – the point at which low-level audio isn’t increased so as not to bring up background noise – the gain is always changing, whether it needs to or not. Furthermore, multiband processing by design re-balances the spectral mix of the program. This can be advantageous for helping to fix less-than-ideal mixes or if a spectrally consistent output is a priority, but it can also affect the artistic intent of well-mixed content.
In contrast, APTO focuses on achieving the following goals:
Ensuring that foreground sounds – particularly dialog – remain intelligible at all times
Allowing the user to define a comfort zone within which no additional audio processing is applied, providing a more natural sense of dynamics
Maintaining the spectral balance of the original program material to preserve the artistic intent
Achieving and maintaining an output target level that is in compliance with global loudness regulations and is optimized for distribution platforms (such as streaming and on-demand services) and for specific devices and listening environments (including mobile phones and tablets)
In very basic terms, APTO first measures and analyzes the loudness of the incoming audio. In its first processing stage – the “Dynamic Range” stage - it applies realtime loudness control to reduce the overall dynamic range and get the levels within the user-defined comfort zone. This processed audio is then scaled in the second processing stage – the “Compliance” stage - to achieve an average output level that matches the desired target loudness value.
As mentioned previously, ARC includes a host of factory profiles for various loudness standards and deliverable platforms, but individual controls to fine-tune both processing stages are also brought out to the user interface and are described in greater detail later in the sections on Basic and Advanced APTO processing.
Linear Acoustic® APTO™ Loudness Adaptation Algorithm with processing for two PCM stereo or mono programs
One 2-channel input and one 2-channel output via 75 Ohm BNC female connectors, internally terminated; signal levels per SMPTE 276M/AES-3ID-2001
One auto-sensing 3Gb/s HD/SD-SDI (SMPTE ST425/292M/259M) input and one output via 75 Ohm BNC female connectors, internally terminated; video formats up to 1081i/60/59.94/50Hz
16-channels of bi-directional AES67 I/O in support of SMPTE ST 2110-30 workflows
AES-3 input to AES-3 output with SRCs disabled: 30.5ms
AES-3 input to AES-3 output with AES-3 input SRC enabled: 33.5ms
SDI input to SDI output with SRCs disabled: 33.5ms
SDI input to SDI output with SRCs enabled: 36.5ms
AES67 input to AES67 output: 107ms (measurement made using Telos Alliance SDI xNode which has a latency value of 80ms)
48kHz reference via SDI, AES-3, PTP v2, or internal clock
48kHz, 24-bit, 20Hz – 20kHz
Two Gigabit RJ-45 connections, one each for AES67 and Control
As we’ve mentioned before, most users and applications will be well served by simply selecting the appropriate factory profile according to either compliance regulations or the delivery platform.
At most, some adjustments to the Basic controls may be necessary.
Unless there are extenuating circumstances or the need to solve a particular problem with highly difficult programming, adjusting the Advanced parameters described below is rarely necessary. Making arbitrary adjustments without a firm understanding of how these controls work, relate to the Basic APTO controls, and interact with the complex processing algorithms at work can cause more harm than good and result in audio that is either unpleasing, non-compliant, or both.
That said, they are available for those who need them.
The High Pass Filter control (A) sets the cutoff frequency for the high pass filter applied to the input signal. No audio below this frequency will make it through to the APTO processing engine. In most cases, this control should be set to “Off”, but in some instances, it may be beneficial to filter out any sub-audible frequencies (below 20Hz).
Voiced-based content with no low-frequency information can benefit from a higher setting (60Hz, for example). Delivery platforms with limited headroom (such as in-flight entertainment or mobile streaming) and consumer devices with small speakers (mobile phones, tablets, or earbuds) can also benefit from a higher setting to prevent distortion.
When the Average Hold control (found in the APTO basic menu) is enabled, the Average Hold Threshold control (B) determines the point at which the hold is triggered in the Compliance processing stage to prevent increasing noise or background audio.
The value at which the control is set is relative to the Target level. For example, if the Target level is set to -23dB LUFS and the desired level at which the hold engages in the Compliance processing stage is -34dB, then the Average Hold Threshold should be set to -10dB
Proper dialog-based loudness processing requires the accurate detection and measurement of speech. When Dialog Normalization is enabled, the Minimum Speech Duration control (C) determines how many seconds of continuous speech are required for the measurements to be considered reliable and valid, and after which the Compliance processing stage adaptively switches to dialog normalization mode.
A minimum of 5 seconds is recommended. Shorter values may incorrectly factor in non-dialog material, and significantly longer values may cause unnecessary delays in engaging dialog normalization processing, especially in content with shorter segments of continuous dialog.
The Dialog Compliance Measurements control (I) determines how many speech measurements are used to perform dialog normalization in the Compliance stage of processing. Because the algorithm samples the measurements at 0.5 second (one half of one second) intervals, this control should be set to a time value that is twice the desired average. For example, to base normalization on a 60 second average, the control should be set to a value of 120.
Note: The Dialog Compliance Measurements setting only applies to profiles that use Dialog Normalization. Profiles that do not use Dialog Normalization rely upon the settings of Compliance Window control to determine their time window.
When the Adaptive Input Detection control (located in the Basic section) is enabled, the Dynamic Range processing section of APTO dynamically adapts the amount of processing depending upon the actual measured average level at the input. The degree to which the actual input levels influence the processing versus relying upon the value of the Average Input Level control (K) is determined by the Adaptive Input Percentage control (H).
The recommended setting to start is 50% in order to keep the effect of adaptation more consistent.
All programming has what is sometimes referred to as an “anchor element,” that is, the audio content to which the viewer will pay the most attention. This is typically (though not always) dialog. This is also referred to as “foreground” audio to differentiate it from background audio, or in some cases, noise.
The Foreground Sounds Coefficient control (G) sets the level in the Dynamic Range processing stage at which program audio is no longer considered a foreground sound relative to both the lower border as set by the Null Area Coefficient control (described in detail below) and a minus infinite level (full silence) on a proportional scale.
For example, if the Target level is set to -23dB LUFS, the Null Area Coefficient is set to 4, and the Foreground Sounds Coefficient is set to 2, any audio between -27 and -25dB will be deemed foreground audio and therefore be raised toward the target.
Audio at levels lower than -27dB will still be increased toward the target, but the degree to which gain is increased slows down considerably. The lower the audio is from -27dB, the less the gain will increase.
The primary goal of the APTO processing within ARC is to deliver a consistent average output level as set by the Target Level control. This does not mean, however, that the actual output audio level must never deviate from this value. In fact, a certain amount of dynamic range helps preserve the artistic integrity of the original programming and makes for a more engaging audio experience for the viewer.
One of the things that makes APTO different from traditional processing is its ability to “do nothing” to audio levels when no action is required to maintain the correct average output level. This avoids the “busy” sound of traditional compressors and ACGs which by their very nature are always operating either over or under a threshold and therefore always increasing or decreasing gain - often for no good reason.
The Null Area Coefficient control (D) sets the lower and upper thresholds that together determine the size of the window in which APTO’s Dynamic Range processing stage neither increases nor decreases gain, with the user-determined Target level sitting in the middle of the range. Values are in dB, with larger values resulting in a larger “do nothing” window.
How quickly gain changes are made in the Compliance processing stage are largely program dependent. However, the maximum rate at which the gain can increase or decrease is set by the Compliance Speed control (E). The rate is calibrated in dB (or LU) per second.
Higher values (from 2 – 6 LU per second) allow compliance to be achieved more quickly but may introduce audible gain changes to the average program level in the process. These faster settings are best suited for situations when input levels are expected to be very inconsistent.
Lower values (from 0.2 to 2 LU per second) provide a subtler and more natural-sounding normalization but may result in gain changes that are too slow and allow the audio to remain outside of the comfort zone for too long. Slower settings work well for content that is more consistent and well-controlled. They are also recommended for long-form content such as feature films or classical music.
The Average Maximum Gain control (F) sets the maximum amount of positive gain (gain increase) applied in the Compliance processing stage in order to reach the Target level.
Larger values will allow very soft program segments to be raised by a greater amount, but whether or not this is desirable must be considered. For instance, some material may have been purposefully kept at a lower level for dramatic effect, and of course, there is always the risk of increasing unwanted background noise.
Another unwanted byproduct of setting this control too high - especially with profiles that use a lower Adaptation value – is that it can result in APTO taking too long to lower levels when loud content immediately follows soft content, such as when a loud commercial follows a quiet passage from a TV drama, as it will take longer to reduce a greater amount of gain.
For these reasons, it is advisable to set the Average Maximum Gain for no more than 2 LU.
One way to enhance the efficiency of the processing across programs is reset APTO at the transition point, which will, in turn, reset the amount of average gain or attenuation being applied and allow each individual program element to be optimally processed from the beginning.
The Average Maximum Attenuation control (L) sets the maximum amount of negative gain (gain reduction) applied in the Compliance processing stage in order to reach the Target level.
Larger values will allow loud program segments to be reduced by a greater amount. However, setting this control too high – again, especially with profiles that use a lower Adaptation value - can cause APTO to take too long to raise levels when soft content immediately follows something loud, such as when transitioning from a loud commercial back to a quiet TV drama, as it will take longer to boost a greater amount of gain.
Just as with the Average Maximum Gain control, resetting APTO at the start of each program segment can help minimize such issues and optimize overall processing for each individual program segment.
The Compliance Window control (J) adjusts the size of the sliding time window used in the Compliance processing stage to align the output program to the target level. It is similar to the rolling integration time on an LKFS/LUFS loudness meter.
Note:
The Compliance Window setting only applies to profiles that do not use Dialog Normalization. Profiles that do use Dialog Normalization rely upon the settings of the Dialog Compliance control to determine their time window.
Many factors can influence whether or not content at the output of ARC is compliant. These include the level consistency of the incoming audio, program duration, the difference between the average input level and the Target level, the amount of Adaptation employed, and the settings of many of the controls listed in this section.
That said, a smaller compliance window value more easily delivers a compliant output for shorter program durations, but can produce more variations in long-term programming.
On the other hand, a larger compliance window value will allow for a smoother and more consistent average overall for longer program durations but can reduce the effectiveness of short-term normalization, especially program elements shorter in duration than the compliance window.
For short-term programs, a value in the range of 20-30 seconds is recommended. Larger values, from 60-120 seconds, are recommended for mid- to long-form content. For profiles that rely upon Dialog Normalization, a setting between 120-180 seconds is highly recommended.
The value of the Average Input Level control (K) should be set to match the average level of the input audio as it serves as the mid-point value of the comfort zone defined by the Null Area Coefficient control.
For content that has been previously analyzed and loudness corrected in the file domain, this value will be easy to determine.
For live programming, unprocessed material, or in cases where programming is of different genres from different decades, this becomes more challenging. For example, cinematic content is typically mixed to -31/-27 LUFS, while music productions can be as loud on average as -12/-8 LUFS.
In these cases where the average input level is not easy to predict, it is best to enter a value that matches the output target loudness value, enable Adaptive Input Detection, and take advantage of APTO’s intelligent dynamics processing.
In most cases, ARC is truly a “set and forget” processor and simply choosing a factory profile is all that is required.
To select a profile, navigate to the Program 1 (or Program 2) menu (A), click the Load button in the APTO Loudness Control section (B), mouse over Factory Presets (C), then choose an appropriate profile from the Factory Preset List (D).
The individual controls within each APTO factory profile have been adjusted in such a way that they automatically deliver the required result associated with the profile name (such as “EBU R128 Adaptive”). However, there may be situations in which you want to adjust certain individual parameters to customize or fine-tune the audio.
The APTO Loudness Control section of the Program 1 and Program 2 menu contains the basic controls for each program path and are identical in terms of controls and functionality, though completely independent of one another in their operation.
Figure 8-2 – Basic APTO controls
The currently selected profile is displayed in the Profile window (A).
Note: In the current version of ARC, there is no indication that the factory profile has been modified as a result of changing the values or settings of individual controls. This feature will be included in an upcoming software upgrade. In the meantime, we strongly recommend that you save any changes and create uniquely-named user profiles as you make adjustments as outlined below in the paragraph on using the “Load and Save” buttons.
The Load button (B) is used to recall factory profiles, user profiles, or to import a profile previously saved to your computer. Any changes or modifications to the current profile can be saved by using the cryptically-named Save button (C).
APTO can be bypassed so that the input audio is passed through to the output without being processed by clicking on the Bypass control (D). This is useful for a quick comparison between the unprocessed and processed audio. When bypass is engaged, the button will turn red.
Clicking on the Reset button (E) resets the loudness measurements as well as the gain buffers APTO uses in the normalization stage. Resetting at the start of each individual program element provides accurate per-segment loudness measurements, aids in achieving overall compliance, and ensures that adaptive processing decisions are made based upon the current program dynamics. A GPIO input may be used to trigger the reset automatically.
The Target Loudness control (H) sets the desired average loudness level of the output signal in either LUFS or LKFS, depending upon the profile. Some profiles, such as EBU R128, will measure loudness according to overall program levels while others, such as ATSC A/85, will do so based on dialog and gated speech measurements.
To change the target loudness value, click in the Target Loudness field, type in the desired value, and click on the green checkmark to save your change (or the red “X” to exit without saving).
The Adaptation control (G) determines how much processing is applied to the incoming signal in the Dynamic Range processing stage, and, in combination with various individual controls, determines the dynamic range of the output audio.
The ideal amount of Adaptation depends upon both the source content and the destination platform. Content that has been pre-analyzed for loudness, scaled, and normalized in the file domain will require less realtime processing than, say, live sports, which can have rather unpredictable audio levels. Programming streamed to a mobile device or expected to be heard on lower quality earbuds will benefit from more Adaptation than the same content destined for a home cinema presentation.
Ideally, you want enough Adaptation to achieve compliance and keep levels within the viewer’s comfort zone and at a stable average level, but not so much that the audio sounds unnatural or over-processed. APTO is based on a psychoacoustic model that takes into account human hearing and perceived loudness and remains very natural-sounding even when extensive processing is applied, but it is generally advisable to keep the amount of Adaptation under 50% when possible. If incoming content is so poorly controlled as to require higher values, it may be necessary to adjust some of the individual controls and create a custom profile to address this scenario.
The True Peak Limiter control (J) enables and disables the True Peak Limiter, which is the final processing stage just ahead of the final output. The Maximum True Peak value (F) sets the level beyond which the True Peak limiter engages and attenuates the processed audio so as not to exceed the set level. These controls comply with the True Peak measurement as outlined in ITU-R BS.177-4 Annex 2.
The Dialog Normalization control (L) enables dialog detection and measurement. When enabled, APTO’s Compliance processing stage uses long-term speech-only measurements rather than the overall input program loudness to ensure the output target level is achieved.
The use of dialog-based measurement is normally determined by regional regulations. For example, ATSC A/85 relies upon anchor-based normalization, specifically dialog, while EBU R128 recommends the use of overall measurements. This is reflected in their respective factory profiles.
When the Average Hold control (K) is enabled and the output audio of the Dynamic Range processing stage falls below the level set in the Average Hold Threshold control (located in the Advanced menu), APTO’s Compliance stage of processing becomes inactive until the level once again rises above the threshold. This helps prevent noise and low-level background audio from being increased unnecessarily.
When enabled, the Adaptive Input Detection control (I) dynamically adapts the amount of processing occurring in the Dynamic Range stage of processing depending upon the actual measured average level at the input. The degree to which the actual input levels influence the processing versus relying upon the value of the Average Input Level control (found in the Advanced menu) is determined by the Adaptive Input Percentage control (also located in the Advanced menu).
When Adaptive Input Detection is disabled, Dynamic Range processing decisions are made based strictly upon the value set in the Average Input Level and in accordance with the settings of other parameters and controls.
Adaptive Input Detection is especially useful when source audio levels are unknown or are likely to vary widely as it allows the Dynamic Range processing stage to respond more predictively to the actual incoming content.
If the incoming content has already been analyzed and loudness-corrected in the file domain, less overall processing is required and a more natural-sounding output can be achieved by setting the Average Input Level (located in the Advanced menu) to the same value as the target level used during file-based correction and reducing the value of the Adaptive Input Percentage control (also found in the Advanced menu) or disabling Adaptive Input Detection altogether.
A budget-friendly, easy-to-use, 2-channel television processor specifically designed for regions and applications that do not require support for multi-channel, coded audio.
The Linear Acoustic ARC is a budget-friendly, easy-to-use, 2-channel television processor specifically designed for regions and applications that do not require support for multi-channel, coded audio, but where no-compromise audio quality is valued.
The name “ARC” – which stands for “Automatic Realtime Control” – provides a good clue as to what it is: A realtime television loudness processor featuring APTO®, our latest loudness adaptation algorithm.
Most of the television processors in the Linear Acoustic lineup are designed to process stereo and 5.1-channel audio, upmix 2-channel audio to 5.1, insert audience measurement watermarking, handle Dolby® coded audio, and insert local audio sources such as EAS. They also use multiband processing to help mitigate spectral inconsistencies and provide a consistent-sounding output.
In contrast, we designed ARC to simply and effectively process two 2-channel mono or stereo PCM programs, making it ideal for stereo-only channels in the U.S. as well as regions where support for 5.1-channel and coded audio is not required. The ATPO algorithm does not employ multiband processing, so the output audio is spectrally identical to the source material. ARC also provides the ability to easily set a LUFS/LKFS output target loudness value and maximum true peak levels.
ARC offers SDI, AES-3, and AES67 I/O, making it equally at home in today’s facilities as well as forward-looking SMPTE ST 2110 installations. Dual internal redundant universal auto-ranging power supplies ensure reliable operation anywhere in the world.
V0.0.64 and V1.2 Firmware
ARC front panel features include:
Four status LEDs indicating the status of each power supply (1A, 1B), the overall status of the unit (1C), and sync for the reference clock (1D).
A color LCD display to show status and basic configuration parameters (1E)
A five-button navigation cluster with Left, Right, Up, and Down arrows plus a green “OK” button (1F)
The front panel is used for the initial setup and configuration including setting the IP address of the Control Ethernet port. Additional configuration is accomplished via the remote user interface.
The rear panel of the ARC contains the following connections:
One parallel GPI/O control port (female DB-15) for five inputs and five outputs (2A)
One AES-3 Input (2B) and one AES-3 Output (2C) on female BNC connectors
One SDI 3Gb/s HD/SD-SDI Input (2E) and 3Gb/s HD/SD-SDI Output (2D) on female BNC connectors
Two RJ-45 Gigabit Ethernet connections, one for AES67 I/O (2F) and one for network remote control (2G)
Two IEC power inlets for the internal redundant universal auto-ranging power supplies (2H, 2I)
ARC is a 1RU product intended to be permanently installed in a standard 19 ½” equipment rack and secured with four standard rack screws. It is fan cooled with air intakes and exhausts located on the side of the unit just behind the front panel, but whenever possible, it is recommended to leave 1RU of empty space above and below the unit.
Plug the supplied IEC power cords into the ARC and connect each to different mains power sources. Remember that while redundant supplies do protect against the unit losing power in the event of a PSU failure, the loss of mains supply voltage is a bigger concern. Accordingly, each supply should be fed from a different circuit equipped with adequate surge protection and fitted with an uninterruptible power supply (UPS).
ARC will begin to boot upon applying power; there is no power switch. When ARC finishes booting, it displays the Home Screen which includes the Linear Acoustic and ARC logos, the unit’s Control IP address, and its reference source and status.
ARC ships with DHCP enabled and will automatically retrieve an IP address when connected to a network with a DHCP server.
Important! Both the Control and AES67 connections must be connected to 1000BASE-T (Gigabit) switch ports. If a port will not be used, it should not be connected to a switch but does require a specific configuration described below. If both the Control and AES67 ports are used, each must be on a different subnet.
To set a fixed IP address for the Control port:
Press the Right button until the IP Configuration (Control) screen is displayed
Press the OK button to highlight “Edit”
Press OK again to highlight “Use DHCP”
Press the Right button to highlight “Yes”
Press the Up/Down button to “No”
Press the Left button to highlight the “Use DHCP” field
Press the Down button to highlight “Address”
Press the Right button to move to the first digit of the IP address
Use the Up/Down buttons to change the selection
Press the Right/Left buttons to select another digit to edit
When finished editing the address field, press the Right or Left button to return to “Address”
Press the Down button to navigate to the Netmask and Gateway fields
Repeat the above steps to edit the Netmask and Gateway fields; note that the gateway may be set to 000.000.000.000 if no gateway is required
After the Gateway is set, press the Left button to highlight “Gateway”
Press the Down button to highlight “Update”
Press OK to save the new IP information
Remove power from the ARC for at least 10 seconds, and then re-apply power.
If you suddenly notice you’ve entered some incorrect information and need to start over, use the Right/Left buttons to highlight “Cancel” followed by “OK”
If you are using AES67 I/O, use the Left/Right buttons to navigate to the IP Configuration (AES67) screen. The steps for configuring this port are identical to those above for the Control port. Please remember that when both the Control and AES67 ports are used, each must be on a different subnet.
If either of the two Ethernet ports will not be used, the unused port must be configured as follows. Do not connect the unused port to a network switch.
Using the steps above for Control port setup, navigate to the IP Configuration screen of the unused port
Set “Use DHCP” to “Yes”
Highlight and select “Update”
Power cycle the unit
Pressing the Right/Left buttons will eventually return you to the Home Screen.
Connect inputs and outputs as required for your particular installation. ARC can access any of the eight audio pairs carried on an incoming HD- or SD-SDI stream and provide de-embedding and pair shuffling at its input. Audio can also be received using the AES-3 input or via AES67. Details instructions on audio routing are provided in the full ARC user manual.
No special client software is required to remotely connect to the ARC, and the HTML-5-based GUI is device, operating system, and browser agnostic. We recommend Google Chrome or Apple Safari. The computer or tablet should be on the same network and subnet as the ARC.
Once ARC and your computer are connected to your network, enter the IP address of the ARC to which you want to connect in the URL field of your browser. Once connected, the Home Screen will appear.
Clicking on the ARC logo (5A) in the top left corner will always return you to this screen.
Multiple users can connect to the same ARC hardware simultaneously. Anyone connected to the unit will be able to see the information on this screen including the unit’s name (5B) and physical location (5C).
Additionally, all users can access the Meters screen (5D) which shows loudness measurements and input/output levels for both programs. See the section “ARC Home Screen When Logged In” below for detailed information.
The Locate Unit feature (5E), which causes the front panel display of the unit to flash for easy identification in a crowded rack, can also be activated without logging in.
Additional information about installed options, clock sync, network status and IP address, power supply status, software and FPGA versions, and unit uptime are displayed to the right of the screen (5F).
Making modifications to ARC’s configuration, signal routing or audio processing parameters requires logging into the unit. At this time, only one user account with one password is supported. By default, the password is “1234”. The ability to change the password will be included in an upcoming software update.