Introduction

Audio processing for broadcasting is in a near-constant state of evolution, and our industry finds itself in a time of "in-between" in many ways. One of the most obvious is the concurrent availability of proprietary hardware platforms with traditional I/O alongside software running on off-the-shelf servers using networked audio.

Such is the case with the two products that utilize flexAI as the underlying software platform to provide audio processing and encoding:

• AIXpressor is a "traditional" integrated standalone hardware product that supports analog, AES3, MADI and SDI I/O in addition to AoIP.

• flexAIserver is a hardware server running the same software but relying upon various Audio over IP (AoIP) protocols for its audio I/O.

Despite their obvious differences, the underlying operating system and software are very similar, enough so that it made more sense to create one set of documentation for both variations.

Where there are differences - most notably in the I/O configurations and the technical specifications - we've provided separate sections for each.

User Interface

While the AIXpressor has a touch-capable front panel control system, both it and flexAIserver are accessed via a browser-based TCP/IP web interface.

For 3rd party remote control, Junger highly recommends using the LAWO Ember+ protocol, which is widely distributed in the broadcast industry and which enjoys a robust and popular user community worldwide. More information about Ember+ is available at https://github.com/Lawo/ember-plus.

Connecting to an IP Network

There are also some differences in network setup between AIXpressor and flexAIserver.

AIXpressor has four integrated 1GB network ports. flexAIserver, on the other hand, may be configured with discrete NICs and will have a very different configuration when used on a VM.

It either case, however, DHCP is enabled by default on the port recommended for use as a control port to connect via a web browser, allowing more advanced network configuration to be done remotely.

On the AoIP side, users may choose between SMPTE ST 2110 (including support for AMWA IS-04 and IS-05), Livewire+, AES67 (Ravenna is on the road map).

AIXpressor is a 1RU product intended to be permanently installed in a standard 19” equipment rack and secured with four standard rack screws. It is fan cooled but whenever possible, it is recommended to leave 1RU of empty space above and below the unit.

Power

Plug the IEC power cords into the AIXpressor 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). Each power supply has its own On/Off switch.

Standards

• Video complies with SMPTE 424/425M (3G, Level A and B), SMPTE 292M (HD), or SMPTE 259M (SD)

• Automatic format detection

• Audio embedding and de-embedding complies with SMPTE 299M (3G, HD) or SMPTE 272M -AC (SD)

• Metadata embedding and de-embedding complies with SMPTE 2020-2

Video Data Rate

• 2970/296Mbps (3G), 1485/1483.5Mbps (HD), 270Mbps (SD)

Video Formats

• 1080p - 23.975, 24, 25, 29.97, 30, 50, 59.94, 60

• 1080i - 50, 59.94, 60

• 720p - 23.975, 24, 25, 29.97, 30, 50, 59.94, 60

• 625i - 50

• 525i - 59.94

Video Delay

• User selectable 0 ... 15 frames, can be disabled

Audio

• 24-bit, transparent forwarding of PCM and compressed audio

• Sample rate = 48kHz (SDI compliant)

• Channels = 16 inputs and 16 outputs (4 groups of 4 channels each)

• Delay = Embedder audio delay selectable 0 ... 320ms per channel

Metadata (RDD6)

• 1 channel input, 1 channel output, SDID selectable

BNC Input

• Impedance = 75Ohm

• Return loss = >15dB 5 ... 1485MHz; >10dB 1485 ... 2970MHz

• Maximum cable length = 250m (SD), 230m (HD), 140m (3G) using Belden 1694A cable

• Jitter tolerance = >0.7UI (alignment)

BNC Output

• Impedance = 75Ohm

• Output voltage = 0.8Vpp (typical)

• Return loss = >15dB 5 ... 1485MHz; >10dB 1485 - 2970MHz

• Output jitter = <0.2UI (alignment), <0.5UI (timing)

Latency

• Video latency input to output = 120 ... 200 pixel, depending on video standard

• Audio latency input to output, embedder and de-embedder combined = <0.6ms (HD, 3G), <2ms (SD)

General Features

• Power fail relay bypass (may be enabled/disabled via GUI)

• Lip-sync compensation for processed and non-processed audio signals via video delay

• Dedicated routing for non-processed channels; all channels (maximum 16) can be routed to/from the device or looped through

• Test pattern generator (no genlock)

• Master-sync capable

• ITU-R BT.1685 / ARIB STD-B39 metadata support

General

• Multi-channel audio processor

• Rear panel AES3, analog, MADI, tieLight I/O

• 3k x 3k cross-point audio router

• Front panel headphone jack

• Expandable by hardware option boards

• Versatile audio processing via licensed algorithms

Audio Sample Rate

• 48kHz

• +/- 150ppm sync input capture, +/- 25ppm master-sync stability

AES/EBU Inputs

• Relevant specifications comply with AES3-X-2009, IEC 60985, and AES11-2009

• 4 inputs (8 input or 4 input /4 output channels switchable) on D-Sub 25 connector

• 24-bit transparent forwarding of PCM and compressed audio without SRC (sample rate converter)

• 24-bit PCM with SRC activated

• Impedance = 75Ohm single-ended

• Input level = 0.3 ... 5Vpp @ 75Ohm single-ended

• Sample Rate Converter (SRC) = THD+N -120dB @ 0dBFS, 1kHz, latency <0.3ms

AES/EBU Outputs

• Relevant specifications comply with AES3-X-2009, IEC 60985, and AES11-2009

• 4 outputs (8 output channels) on Sub-D 25 connector

• 24-bit transparent forwarding of PCM and compressed audio

• Impedance = 75Ohm single-sided

• Output Voltage = 1Vpp (typical) @ 75Ohm single-ended

• Power fail relay bypass between AES/EBU inputs and outputs (can be deactivated by internal jumper)

SFP Cages

• SFP spec, typically 125MBps

SFP MADI Module

• SFP spec, typically 125MBps

SFP tieLight Module

• SFP spec, typically 2.5GBps

Sync Input

• Multi-standard synchronization interface for AES/EBU, Wordclock, or video sync (including black burst and tri-level); complies with AES11-2009 and relevant audio or video standards.

• Connector type = BNC

• AES/EBU Input = 0.3 ... 5Vpp @ 75Ohm single-ended

• Wordclock input = 1 ... 5Vpp @ 75Ohm single-ended

• Video-sync input = 1Vpp (nominal) @ 75Ohm single-ended

• Supported rates = 23.975, 24, 24.975, 25, 29.97, 30, 49.95, 50, 59.94, and 60 fps (SD and HD)

Network Interfaces

• 4x RJ45 connectors, 1Gbit/s Ethernet auto sense, full duplex, audo MDI/X

USB Interfaces

• Front and rear USB A 3.0 connectors

• Internal USB A 2.0 (for license keys)

• Rear USB 2.0 connectors

Expansion Slots

• 4 general-purpose expansion slots for Jünger option boards

Power Supplies

• Dual power supplies with automatic failover

• 85 ... 264VAC, 50 ... 60Hz, 58W (max)

Environmental

• Operating temperature = 0 ... 50°C, fan cooled

• Non-operating temperature = -20 ... 70°C

• Humidity < 90% non-condensing

Physical

• 19" x 1RU, 27cm depth

• Net weight ca. 5kg

• Shipping weight ca. 7.5kg

Standards

• Video complies with SMPTE 424/425M (3G, Level A and B), SMPTE 292M (HD), or SMPTE 259M (SD) with automatic format detection

• Audio embedding and de-embedding comply with SMPTE 299M (3G, HD) or SMPTE 272M-AC (SD)

• Metadata embedding and de-embedding complies with SMPTE 2020-2

Video Data Rates

• 2970/2967Mbps (3G), 1485/1483.5Mbps (HD), 270Mbps (SD)

Video Formats

• 1080p - 23.975, 24, 25, 29.97, 30, 50, 59.94, 60

• 1080i - 50, 59.94, 60

• 720p - 23.975, 24, 25, 29.97, 30, 50, 59.94, 60

• 625i - 59.94

• 525i - 59.94

Video Delay

• User selectable 0 ... 15 frames; can be disabled

Audio

• Bit depth = 24 bit, transparent forwarding of PCM and compressed audio

• Sample rate = 48kHz (SDI compliant)

• Channels = 16 inputs and 16 outputs (4 groups of 4 channels each)

• Delay = Embedder audio delay selectable 0 ... 320ms per channel

Metadata (RDD6) not activated

• 1 channel input, 1 channel output, SDID selectable

BNC Input

• Impedance = 75Ohm

• Return loss = >15dB 5 ... 1485MHz; >10dB 1485 ... 2970MHz

• Maximum cable length = 250m (SD), 230m (HD), 140m (3G) using Belden 1694A cable

• Jitter tolerance = >0.7UI (alignment)

BNC Output

• Impedance = 75Ohm

• Output voltage = 0.8Vpp (typical)

• Return loss = >15dB 5 ... 1485MHz; >10dB 1485 - 2970MHz

• Output jitter = <0.2UI (alignment), <0.5UI (timing)

Latency

• Video latency input to output = 120 ... 200 pixel, depending on video standard

• Audio latency input to output, embedder and de-embedder combined = <0.6ms (HD, 3G), <2ms (SD)

General Features

• Power fail relay bypass (may be enabled/disabled via GUI)

• Lip-sync compensation for processed and non-processed audio signals

• Dedicated routing for non-processed channels; all channels (maximum 16) can be routed to/from the device or looped through

• Test pattern generator (no genlock)

• Master-sync capable

• ITU-R BT.1685 / ARIB STD-B39 metadata support

Audio

• 24-bit D/A converter

• 8 output channels (for speakers)

• Sample rate = 44.1, 48, 88.2, 96kHz

Analog Outputs

• 8 channels

• Connector type = D-Sub25 female

• Maximum output level (0dBFS equivalent) = 0 ... 24dBu, adjustable in 0.5dB steps

• Impedance = 50Ohm (typical), differential

• THD+N = -91dB @ 0dBFS = 15dBu, 1kHz

• Dynamic range = >103dB (RMS)

• Crosstalk attenuation = >103dB @ 0dBFS = 15dBu, 1kHz

• Frequency response = 20Hz ... 20kHz (< +/-0.3dB) @ 48kHz; 20Hz ... 43kHz (< +/-0.3dB) @ 96 kHz

General Features

• Power faillure glitch prevention

• Balanced analog outputs

• Electrical isolation between outputs and device

Technical specifications for AIXpressor and all available option boards plus D-Sub pin assignments are provided the following sections.

Standards

• Relevant specifications comply with AES10-2008 and AES11-2009

Audio

• 24-bit, transparent forwarding of PCM and compressed audio

• Sample rate = 44.1, 48, 88.2, 96kHz (88.2, 96kHz short framing)

Optical Input, LC

• 64/56 channels @ 44.1 and 48kHz; 32/28 @ 88.2 and 96kHz

• Connector type = LC (IEC 61754-20)

• Center wavelength = 1310nm (typical), 1270 ... 1360nm

• Input optical power

  • O DAP MO MM a = -31 ... -8dBm, OM2 multimode (50/125um)

  • O DAP MO SM a = -23 ... -8dBM, singlemode (standard values, others upon request)

• Cable length maximum

  • O DAP MO MM a = 1.5km, OM2 multimode

  • O DAP MO SM a = 2km, singlemode (standard values, others upon request)

BNC Output

• Optical and BNC outputs carry the same signal

• Impedance = 75Ohm

• Output voltage = 0.6Vpp (typical) @ 75Ohm

General Features

• Field-replaceable optical module (SFP)

• Reference-grade Wordclock recovery, master-sync capable

• Dedicated routing for non-processed channels

• All channels (maximum 64) can be routed to/from the device or looped through

• AES3 channel status management, non-audio detection

• Parallel outputs (BNC/LC) for media conversion

Standards

• Relevant specifications comply with AES3-X-2009, IEC 60985, and AES11-2009

• Electrical isolation between inputs, outputs, and device (if configured for differential mode, 110Ohm)

• Master-sync capable

• Connector type = D-Sub25 female connector

Audio

• 24-bit, transparent forwarding of PCM and compressed audio (without SRC)

• 24 bit PCM with sample rate converter activated

• Sample rate = 44.1, 48, 88.2, 96kHz (32 ... 196kHz @ inputs with SRC)

• Non-audio detection

Inputs

• 4 inputs

• Impedance = 110Ohm or 75Ohm, jumper selectable (110Ohm default)

• Input level = 0.3 ... 5Vpp @ 110Ohm differential

Outputs

• 4 outputs

• Impedance = 110Ohm or 75Ohm, jumper selectable (110Ohm default)

• AES3 channel status management,

Audio

• 24-bit sigma-delta A/D converter, 24 bit D/A converter

• 4 input channels, 4 output channels

• Sample rate = 44.1, 48kHz

Analog Inputs

• 4 channels

• Connector type = D-Sub25 female connector

• Input Level (maximum) 0dBFS equivalent = 0 ... 24dBu, adjustable in 0.5dB steps

• Impedance = 20kOhm (typical), differential

• THD+N = -93dB @ 0dBFS = 15dBu, 1kHz

• Dynamic range = >110dB (RMS)

• Crosstalk attenuation = >93dB @ 0dBFS = 15DBu, 1kHz

• CMRR = >71dB @ 0dBFS = 15dBu, 1kHz

• Frequency response = 20Hz ... 22kHz (< +/-0.1dB) @ 48kHz; 20Hz ... 43kHz (<+/-0.1dB) @ 96kHz

Analog Outputs

• 4 channels

• Connector type = D-Sub25 female connector

• Output level (maximum) 0dBFS equivalent = 0 ... 24dBu, adjustable in 0.5dB steps

• Impedance = 50Ohm (typical) differential

• THD+N = -91dB @ 0dBFS = 15dBu, 1kHz

• Dynamic range = > 103dB @ 0dBFS = 15dBu, 1 kHz

• Frequency response = 20Hz ... 22kHz (< +/-0.1dB) @ 48kHz; 20Hz ... 43kHz (<+/-0.1dB) @ 96kHz

General Features

• Power fail relay bypass between inputs and outputs

• Balanced analog inputs and outputs

• Electrical isolation between inputs, outputs, and device

Standards

• Relevant specifications comply with AES10-2008 and AES11-2009

Audio

• 24-bit, transparent forwarding of PCM and compressed audio

• Sample rate = 44.1, 48, 88.2, 96kHz (88.2, 96kHz short framing)

BNC Input

• 65/56 channels @ 44.1 and 48kHz; 32/28 @ 88.2 and 96kHz

• Impedance = 75Ohm

• Input level = 0.15 ... 0.8Vpp @ 75Ohm

• Cable length maximum = 150m using Belden 1694A cable

BNC Output

• 64/65 channels @ 44.1 and 48kHz; 32/28 @ 88.2 and 96kHz

• Impedance = 75Ohm

• Output voltage = 0.6Vpp (typical) @75Ohm

General Features

• Input cable equalizer for extended range and robustness

• Reference-grade Wordclock recover, master-sync capable

• Dedicated routing for non-processed channels; all channels (max 64) can be routed to/from the device or looped through

• AES3 channel status management, non-audio detection

When you buy a flexAIserver it comes pre-configured with all components and the applications you have ordered. In that case the flexAI core license is included.

If you insist to to install a flexAIserver on your own hardware (bare metal) contact support for compatibility pls. In this case you must buy a flexAI core license.

Selecting Interfaces from the Main Menu adds a column next to the Main Navigation menu which shows all available I/O interfaces and, depending on the particular interface, sub-menus for settings and status.

Each interface is explained in detail in the dedicated sub-sections below.

The System menu includes sub-menus for:

• Setup - Includes "friendly" fields for the Hostname, system location, and system contact.

• Network - Displays and configures settings for the various LAN interfaces.

• Certificate - System cetificate for Transport Layer Security by using SSL based communication

• Update - Facilitates system updates.

• Logging - Displays a variety of system and event logs.

• Licensing - Lists the currently installed licenses and facilitates downloading and uploading license files.

• Info - Displays a variety of system statuses of hardware, software, system load and GNU licenses

• Users - User management, password and SSH key settings

• Storage - Display of the usage of the storage devices of the system

• Terminal - Systm terminal via the UI

• Backup / Restore - Used to create a backup file or restore a unit from an existing backup.

• Power - Provides a means of shutting down or power cycling the unit from the remote interface.

Standards

• Audio-over-IP (AoIP) by Dante™ Digital Audio Networking Standard; parallel operation in AES67 mode

Audio

• 24-bit, transparent forwarding of PCM and compressed audio

• Sample rate = 44.1, 48, 88.2, 96kHz

Inputs and Outputs

• 2x RJ45 1Gbit/s Ethernet

• Primary and secondary ports configurable as redundant or switched

• Inputs = 64 channels @ 48kHz to AIXpressor

• Outputs = 64 channels @ 48kHz from AIXpressor

General Features

• AES67 compliant

• SMPTE 2110-30 compliant

• PTP v2 / v1 sync

• Non-audio detection for input channels

• Glitch-free Dante / SMPTE 2022-7 audio redundancy

Electrical

• Safety classification: Class 1 grounded product / Schutzklasse 1 corresponding to EN 60065:2002.

• Power connection: The device must be connected to a power socket that provides a protective grounding conductor. Be sure the hardware is installed so that you can quickly and easily disconnect the power cords if necessary.

• Power switch: On/Off toggle switch on the rear of the device near the power inlet. An engraved "I/O" indicates the ON/OFF position on the switch. Devices with dual power supplies will include two power cords and two switches. Be sure the hardware is installed so that the switch is easily accessible and familiarize yourself with their location and operation.

• Water protection: The device must not be exposed to dripping or splashing water.

Service Safety

• Only qualified personnel should perform service procedures.

• Do not service the unit alone.

• To avoid electric shock, switch off the device, then disconnect the power cord(s) from the mains power.

• Do not block access to the power cord during service.

To Avoid Fire or Personal Injury

• Mount the hardware on a flat surface or into a standard 19 1/2" equipment rack; when rack-mounting, the use of sheet metal brackets is recommended for additional support.

• For fan-cooled devices, a gap of at least 1cm must be left between the device and any supports.

• Leaving a 1RU gap above and below the device is recommended.

• Use only a power cord specified for this product and certified for use in your region or country.

• Do not operate the devices with covers or panels removed.

• If you suspect the product is damaged in any way, do not operate it, and have it inspected by qualified service personnel.

• This device contains a lithium battery which may explode if installed incorrectly or replaced by a different type of battery.

The AoIP implementation supports Livewire (including Livewire Clock and Livewire Routing Protocol), SMPTE ST 2110-30 / -31 Level A, and AES67. It is capable of generating multicast streams with between 1 - 8 audio channels per stream. Support for up to 64 audio channels per stream is planned for a future release.

When setting up a large number of receivers and senders, entering the information once in the New Stream Defaults section pre-populates the most common fields and eliminates the need to repeatedly enter the information with each new addition.

Protocol (A)

Selects the type of AoIP stream. Choices include Livewire, AES67, and ST2110.

Receiver (B)

Some values may ybe pre-selected based on the protocol, while others are initially blank to allow for a customized value. When you anble the settings they will appear in the Receivers and Senders stream setups in the following chapters. To ease the assignmment of multicast addresses for receiving streams you can set a Start Destination Address Prefix.

If Seamless Protection Switching is used you can set this address for streams for Media Interface 2 as well.

Sender (C)

• Seamless Protection Switching - ON/OFF

• Codec - Select the type of codec for 2110 / AES67 streams

• Auto RTP Payload Type - the paylaod type will b eautomatically set depending on the selected protocol

• Start Destination Address Prefix

Expand the AoIP portion of the Interfaces menu, then select Setup.

AoIP (A)

• Enable - Enables or disables the AoIP interface.

• Sync Source (B) - Displays the sync source and indicates if the clock is locked (green dot) and shows the name of the interface where PTP is received.

• Leader ID (C) - shows the MAC address of the current PTP leader

Management Interface (D), Media Interface 1 (E), Media Interface 2 (D)

All three interfaces work identically.

• Interface - Selects the respective LAN interface by its system name.

• Link State - Indicates the status of the selected LAN interface (green = Up).

• IP Address - Displays the IP4 address of the selected LAN interface.

Livewire (G)

• LWRP (Livewire Routing Protocol) Password Status - Indicates whether or not a Livewire password has been set or not set.

• LWRP Password - The field in which a password may be entered. Toggling the "eye" symbol to the right of the text field allows the password to be visible. Click <APPLY> to save the password. If done the Apply button changes to <CLEAR PASSWORD>.

• 256 Ch Extension License (I) The flexAI core license limits the number of audio channels to 256 for the receivers and the transmitters. If you must use more that this number you must acquire another such license. The field Claimed Extensions below shows the respective number.

• Packet Processing With the introduction of the new High-Efficiency Audio-over-IP (XDP), the user can optimize the system's performance. In general, once the XDP mode is enabled (J), the system load reduces significantly, and packet sizes of 0,125 ms, 0,25 ms and 0,5 ms are unlocked on the transmitter end. However, as the load on an AIXpressor can still be high due to the high packet density, we have introduced a switch <Interval> that allows the user to balance the system between speed and CPU load. The decision is influenced by whether the short packet sizes are needed to optimize latency or if it's solely about transmitting a high number of channels per stream.

  If the goal is a high number of channels, it may be sufficient to transmit packets every 1 ms (several packets in burst mode), significantly relieving the system. On the other hand, if low latency is crucial, the packets must be sent at the same interval as the set packet size, but this simultaneously increases the system load. Rule of thumb for Packet Sizes below 1 ms:

  Small Packet Size + Small Packet Processing Interval = Low Latency + Higher System Load

  Small Packet Size + Large Packet Processing Interval = Slightly Higher Latency (corresponding to interval setting) + Lower System Load. For Packet Sizes equal or above 1 ms the interval setting has no relevance.

NMOS Registry (K)

This colume is meant to set the NMOS IS-04 Discovery Mode. I.e. how the device finds a NMOS registry in the network.

Status (L)

• Connection - Connected (green dot) / Unconnected (red dot)

• Host - IP address of the NMOS registry host

• Port - of the IP connection to the host

• Version - of the registry implementation

Settings (M)

• Discovery Mode - - Multicast DNS-SD (mDNS) flexAI is looking for a NMOS Registry Server by using mDNS. - Unicast DNS-SD flexAI is looking for a NMOS Registry Server by using Unicast DNS within the selected Domain - Manual You may set the host IP4 address and the port number of the Registry Server to query

• Domain Domain name of the network for manually connecting to a NMOS regestry

• Host IP address of the NMOS regestry for manually connecting

• Port Port number of teh NMOS regetry for manually connecting

• Query Registry ON / OFF

Receiver (N)

• Remaining Licensed Channels

• Remaining Licensed Streams

Auto Link Offset / Delay (O)

Will be used by the receiver when Auto Link Offset is turned on, to automaticaally deal with the network latency. Should be set in relation to the duration, in microseconds, of the latency present in the network. See AES67-2018 (page 20) that illustrates the link offset definition. Generally speaking, setting a value of three times the packet time selected in the Sender will provide more robust timing though with a small increase in overall latency. Separate fields are provided for each of the supported packet time options.

Stream Naming (P)

One may use the name from the sender stream or set an own stream name

Discovery (Q)

• Discovery Protocol Filter - The Discovery Protocol Filter can be set to either look for Livewire, SAP/AES67/ST2110, or NMOS/AES67/ST2110 when searching for streams in the network.

• Default Name Filter field is provided to assign the filter a unique name to be used in the stream setup process.

Sender (R)

• Remaining Licensed Channels

• Remaining Streams

Auto DSCP (S)

Allows for selecting the appropriate QoS - Differentiated Services Code Point for each of the supported packet time options.

Selecting Sync (A) from the Main Menu allows the sync settings to be viewed and changed. AIXpressor provides three priority tiers: Choice 1, Choice 2, and Fall Back for automatic failover.

Sync Source Priority

The Sync Source Priority section (B) is used to define the sync source for Choice 1, Choice 2, and AES3.

• Choice 1 and Choice 2 - Options include Internal, tieLIGHT/MADI 1, tieLIGHT/MADI 2, Sync-In WCLK/BB/Tri-Level, Interface Slot 1, Interface Slot 2, Interface Slot 3, Interface Slot 4, AES Select, and PTP. If none of the pre-selected sources in Choice 1 or Choice 2 are present, sync will fall back to Internal in order to maintain audio transport, even if this causes audible clicks and pops due to asynchronous conditions.

• AES3 Select - The drop-down menu is used to pre-select the sync source when Choice 1 or Choice 2 are set to use AES Select as their sync source. Options include Sync-In AES3 (Sync-In BNC) and Inputs AES3 1/2, 3/4, 5/6, 7/8.

Current Sync Source Status

The Current Sync Source Status section (C) provides information about the sync source currently in use.

• Current Sync Source - Displays the current sync source.

• Current Sample Rate - Indicates the current sample rate, which in this firmware version is fixed to 48kHz.

• Current Video Rate (fps) - Shows the current video rate (23.98, 24, 25, 29.97, 30, 50, 50.94, and 60).

System Clock

The System Clock section (D) sets the primary and fallback sample rates and video rates.

• Sample Rate (kHz) - Sets the preferred sample rate (currently limited to 48kHz).

• Internal / Fallback Sample Rate (kHz) - Sets the internal (fallback) sample rate (currently limited to 48kHz).

• Video Rate (fps) - Sets the preferred video rate (Follow Input, 23.98, 24, 25, 29.97, 30, 59.74, 60).

• Internal / Fallback Video Rate (fps) - Sets the internal (fallback) video rate (23.98, 24, 25, 29.97, 30, 59.74, 60).

Sync Source Information

The Sync Source Information table (E) provides an overview of the available sync sources and their values. Information includes Sample Rate and Video Rate (fps) for each source.

Front Panel

Front panel features include:

• USB 3.0 Type A connector (A)

• 6.3mm (1/4") stereo headphone jack (B)

• 1440x240 TFT touch display (C)

Rear Panel

Rear panel features include:

• Dual 110V - 240V auto-ranging redundant load-balancing power supplies with individual power switches (A)

• Four RJ45 1GB Ethernet connections for control and AoIP; LAN 3/4 internally connected to a switch (B)

• USB 3.0 Type A connector (C)

• Status LED; green indicates normal operation (D)

• INIT/RESET button; pressing briefly will shut down the CPU; pressing and holding until the Status LED flashes five times will return the unit to factory default settings (E)

• USB 2.0 Type B connector for connecting to consoles, audio interfaces, etc. (F)

• MADI I/O; SFP cages to hold modules for optical connections as MADI or tieLight interfaces (G)

• 75Ohm BNC external sync connector, supporting Word Clock, AES, Black Burst, and Tri Level (H)

• AES I/O via 25 pin D-Sub female connector; provides 4x AES3 In and Out; may be configured for 2x In and 6x Out (I)

• Analog Output via 25 pin D-Sub female connector; provides 8x balanced analog outputs (K)

• Analog Input via 25 pin D-Sub female connector; provides 8x balanced analog inputs, or may be configured for 4x In and 4x Out for a total of 12x analog outputs (L)

• IF 1 - IF 4 slots to accommodate optional SD/HD/3G-SDI, 4x AES3 I/O, 4x analog I/O, 8x analog out, MADI I/O (optical or BNC) or Dante/AES67 interface boards (J)

General Network Considerations

By default, AIXpressor has DHCP enabled and will automatically retrieve an IP address if connected to a network with an active DHCP server.

After connecting the LAN 1 CTRL RJ45 port, a DHCP-assigned IP configuration, if available, will be displayed on the front panel. If no DHCP server is available, a static IPv4 IP address can be configured using the front panel, or the unit may be reached via zeroconf (mDNS/Bonjour) technology.

The remote GUI is web-based using TCP/IP over Ethernet. While any web browser may be used, the interface is optimized for use with Google Chrome V90.x or higher. Browsers on mobile devices running iOS or Android may also be used.

An SNMP v2 agent is available for use with an external SNMP monitoring system.

For 3rd party remote control, we highly recommend the use of the widely-available LAWO Ember+ protocol, which is in wide use in the broadcast industry and enjoys a robust user community. Please visit https://github.com/Lawo/ember-plus to learn more.

LAN CTRL 1 Setup

Setting the IP Address and Subnet Mask via a Web Browser

Enter the unit's IP address (as assigned via DCHP or manually entered via the front panel) into a web browser in the following format: http://<IP-address>.

Alternatively, if your computer supports zeroconf, take the device name (as displayed on the front panel), add the serial number separated by a hyphen, and add the suffix ".local" in the following format: <AIXpressor-00080.local>.

Once connected, the main Menu page will appear.

Using the menu to the left side of the screen, navigate to System > Network > Interfaces > LAN 1 CTRL > IP (A).

In Figure 2 below, DHCP is enabled and the assigned IP address is displayed. To assign a static IP address, click on the Add IP Address button (B), choose the address type (C), enter the address (D), choose an appropriate Subnet mask (E), then press the OK button (F) to save.

Setting a Static IP Address and Subnet Mask via the Front Panel

A static IP address can be set via the touch-enabled front panel to make an initial connection with the AIXpressor. Referencing Figure 4 below:

• Touch the Menu button (A) to enter the main menu.

• Touch the Network button (B) to show the LAN 1 settings.

• Touch the Edit Static IP Address button (C).

• Scroll up and down on slot machine-style wheels to set an IP Address (D) and Subnet mask (E).

• Touch the Apply button (F) to save and apply the changes.

Setting Up a Gateway

Navigate to System > Interfaces > LAN 1 CTRL > Gateway (A).

In Figure 3 below, DHCP is enabled and the assigned Gateway is displayed. A static Gateway may be added by clicking on the Add Gateway button (B), entering an address in the IP Address field (C), then clicking the OK button (D) to save.

Configuring PTP

The LAN CTRL 1 interface can be used as a PTP (Precision Time Protocol) clock sync source and supports PTPv2 per IEEE 1588-2008.

Navigate to System > Interfaces > LAN 1 CTRL > PTP (A) and use the PTP Mode dropbown (B) to disable PTP or enable PTP in slave mode.

Once PTP is enabled, navigate to the Sync menu (A). Use the Choice 1 dropdown menu (B) to choose the primary sync source. While not required, specifying a second failover source using the Choice 2 dropdown menu (C) is recommended.

Introduction

AIXpressor is a flexible hardware platform that provides a virtualized environment for many of the well-known Jünger audio algorithms and is capable of simultaneously running multiple processing instances.

In addition to various audio processing options, AIXpressor offers traditional physical I/O and integrates AoIP (Audio over IP) including Livewire, Dante (optional), SMPTE ST 2110-30 and -31, and acts as a versatile "edge server" to ease the migration from traditional analog, AES, and SDI I/O to an IP-based infrastructure.

Hardware

The engine of the AIXpressor is an x86 Intel Atom multi-core CPU-based System on Module (SoM) using the SMARC form factor. The CPU manages all audio-related functions including processing and loudness measurement, provides AoIP connectivity, and hosts the web server that provides remote access through a browser-based GUI. A high-performance FPGA completes the system and handles internal hardware I/O routing.

• Chassis: 1RU (1.75") x 19" high-density multi-channel I/O and audio processing unit

• Display: 8" front panel high-resolution TFT touch display

• Status Indicators: Multicolor status LED

• Power Buttons: Individual power buttons for each power supply

• Headphones: 6.3mm (1/4") front panel headphone jack

• RJ45 Network Connectors: 1GB Ethernet ports for control and AoIP

• USB A Connectors: One front panel and one rear panel USB 3.0 type A connectors

• USB B Connectors: One rear panel USB 2.0 type B connector

• Expansion Slots: Four expansion slots, each capable of handling 64 bi-directional audio channels for the following optional I/O boards:

  • SD/HD/3G-SDI

  • MADI

  • Dante/AES67

  • 4-channel AES3 I/O

  • 4-channel analog I/O

  • 8-channel analog out

• On-board I/O: Native I/O includes:

  • 4-channel AES3 I/O

  • 8-channel analog I/O

  • 2 SFP Cages: 2x MADI or tieLight I/O

• External Sync: BNC input for Wordclock, AES, Black Burst, or Tri-Level

Software

• Jünger Level Magic loudness management per ITU BS.1770-1, -2, -3, and -4, EBU R128, ATSC A/85, ARIB TR-B32, Free TV OP-59, Portaria 354, Voice Leveler

• Spectral Signature™ dynamic filter

• True Peak sample-accurate peak limiter

• 5-band parametric EQ with high-pass and low-pass filters

• Phase rotator

• De-esser

• Expander/Gate

• Voice Over for stereo and 5.1-channel input

• Commentator Automix (3x2 commentators over stereo or 5.1 IT, loudness-controlled program mix, n-1 feeds, clean feeds)

• Upmixing to 5.1 with automatic failover

• FM Conditioner

• Dolby E encoding and decoding

I/O

AIXpressor natively supports four AES3, eight analog, and two MADI fiber I/Os. Up to four additional interface modules - the same as those made popular in the Jünger D*APx series of products - are optionally installable. AES3 and analog channels may be configured as 8 In / 8 Out or 4 In / 12 Out to support immersive audio formats such as Dolby® Atmos and Fraunhofer MPEG-H 3D Audio, allowing for configurations up to 7.1.4.

MADI ports may alternately be configured to use the proprietary Jünger tieLight interface, providing high channel-count and low latency connections between multiple AIXpressors and/or servers running flexAI software.

Comprehensive routing setups permit a nearly unlimited combination of signal flow from hardware and/or AoIP inputs, to/from optional Dolby decoders and encoders, and multiple audio processors to hardware and/or AoIP outputs.

Control

The use of presets helps facilitate routing paths and audio processing parameters. Presets may be recalled on demand by the operator via the web-based GUI or simple external control systems, but may also be integrated into complex automated scenarios and executed by an event manager. Ember+ and NMOS protocols IS-04, -05 are supported, as is SNMP (a REST API is on the road map).

Applications and Use Cases

The operational needs of today's broadcasters vary so widely that it would be impossible to provide detailed scenarios of how AIXpressor can be used. Its flexible design makes it suitable as a dedicated, stand-alone audio routing and processing tool, or as an integral part of a facility-wide connected audio infrastructure. It is equally at home at ingest, in studios and master control rooms, as part of on-air play-out systems, or in rebroadcasting and distribution facilities.

woks audio and its sales partner Telos Alliance stand ready to help you discover how AIXpressor and other Jünger Audio processors can meet your needs and integrate into your particular facility.

The AES3 inputs and outputs provide a relay bypass feature that, when enabled, activates in the event of a power failure or upon a deliberate command from the GUI.

In order to enable the relay bypass feature, the jumpers for each AES3 pair (A) must be in place as shown in Figure 1 below. Removing the jumpers prevents relay bypass from occurring.

Introduction

The flexAIserver is a bare metal COTS machine that allows instantiating multiple well-known Junger Audio algorithms and other 3rd party audio processors that run on x86 processors.

Users may supply their own COTS server, or purchase pre-configured hardware from Telos Alliance together with the processing licenses to suit their needs.

Pre-configured Hardware

The pre-configured server meets all of the hardware requirements to run the flexAI software, including an available PCIe card for tieLight/MADI, an available Intel I350 Ethernet card with 4x 1GBit or 2x 10Gbit for independent redundant control network and AoIP connections, and a standard native NIC with independent 1GBit/10GBit ports for control and AoIP networks.

Customer-provided Hardware

Junger recommends hardware meeting the following specifications:

• Intel Xeon Silver 4310 2.1GHz, 18MB cache, 12 cores, 24 threads

• 32GB RAM

• RAID-Controller supporting Linux Debian 10

• 2x 480GB SSD SATA drives

• 1x PCIe slot (full height or low profile)

• Dual, hot-pluggable, redundant power supplies

• Dual Ethernet NICs with IEEE 1588 and Linux Debian 10 support

• Hardware management with openIPMI support and Linux Debian 10 compatibility

Software

• TP Limiter: True Peak sample peak limiter

• Level Magic: Loudness management per ITU Bs.1770-1/-2/-3/-4, EBU R128, ATSC A/85, ARIB TR-B32, Free TV OP-59, Portaria 354, Voice Leveler

• Dynamic Filter: SPECTRAL SIGNATURE™ dynamic filter

• EQ/Filter: 5 band parametric EQ, HP, LP

• Phase Rotator: Makes asymmetric signals more symmetrical

• De-Esser: Dynamic notch filter to suppress sibilance from human voice recording

• Dynamics: Compressor, expander / gate

• Failover/Upmix: automatic fail over, Upmix

• Voice Over: Stereo, 5.1, 7.1 program input

• Commentator Automix: 3x2 commentators over stereo IT, loudness controlled program mix, N-1 feeds, clean feeds

• FM Conditioner

• Dolby® Decoder: Dolby E decoder licenses

• Dolby® Encoder: Dolby E encoder licenses

I/O

flexAIserver supports several AoIP protocols, including SMPTE ST2110, Livewire, and Ravenna, with SMPTE 2020-7 redundancy. With the available PCIe card, it also supports two independent 64-channel MADI I/O streams, Junger tieLight, or a combination of both.

AES I/O 4x in and 4x out or 2x in 6x out

ANALOG IN 8x in or 4x in and 4x out

8x Analog Outputs

jjjj

The pre-configured flexAIserver hardware is a 1RU product intended to be permanently installed in a standard 19” equipment rack and secured with four standard rack screws. It is fan cooled but whenever possible, it is recommended to leave 1RU of empty space above and below the unit.

Power

Plug the IEC power cords into the flexAIserver 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). Each power supply has its own On/Off switch.

Licensing and Hardware

The type of audio processing and the number of processors available is determined in part by which features have been licensed at the time of purchase. Additional processing solutions and features can always be activated by purchasing the respective license.

The available capacity of the SoM (System on Module) x86 processor installed in an AIXpressor is another determining factor. This limitation can be overcome by daisy-chaining an AIXpressor with a COTS (Commercial Off the Shelf) server equipped with a Jünger PCIe card that runs flexAI and connecting it with the proprietary Jünger tieLight low-latency fiber interface.

Processing Structure

Audio processing is based on the Jünger flexAI engine (flexible audio infrastructure) which renders the audio processors and conects to the various audio I/Os and the system OS settings.

The processing itself is "program-oriented" in that the number of audio channels determines what types of processing blocks are involved (Mono, Stereo, 5.1 etc.). Each block appears in the Routing matix by its name where "x" represents the progarm number. For example, a stereo processor will appear as Program x: L and Program x: R while a multi-channel surround processor will appear as Program x: L ... Program x: Rs. A surround processor with voice-over channels would add two more inputs labeled Voice A and Voice B.

Adding a Processor

Select Audio Processing in the Main menu (A), then click the +ADD button (B) to open the "Add new Program" window. Use the Program Type dropdown menu (C) to open the list of available processing options, enter a "friendly" name into the Program name field (D) if desired, then click the OK button (E) to save.

The GUI for the selected processor will be displayed:

Editing a Processor Name

To change the name of a processor once it has been added, click the EDIT NAME button (B), type the new name in the Name field (C), tick the check box, then click the Close button (A) to save and exit.

Deleting a Processor

To delete a processor, click the DELETE button (B), click the checkbox of the processor(s) you wish to delete (E), then click the Delete Selected button (D). Clicking the checkbox a second time de-selects the processor. Clicking the Clear Selection button (C) de-selects all processors. Click the Close button (A) to save and exit.

Processing Parameters

Clicking on the name of an audio processor from the Main menu will open a window to reveal its individual processing stages and parameters.

Clicking on any of the processing stage buttons (A) will cause the window to scroll to the relevant displays and controls section (C) which, like the selected button, will be highlighted in blue. By default, all of the sections are visible but can be individually collapsed by clicking the collapse/expand arrow (B). Depending upon the size of your browser window, it may be necessary to use the scroll bar (E) to view the entire section. Clicking on the Collapse button (D) collapses all sections and changes the button name to Expand.

Parameters and Controls

Presets for Individual Processing Section

Changes made to each processing section can be saved as user presets.

Click on the Save Preset icon (F) to open the Save Preset window. Use the Save Preset As dropdown (A) to choose whether the changes will be saved as a new preset or will overwrite an existing preset. Type a name into the Preset Name field (B) and a description, if desired, in the Preset Description field (C). Click the OK button (D) to save your changes and exit, or the Cancel button (E) to exit without saving.

Once presets have been saved, they can be recalled by clicking on the Open button (A). Clicking on the Preset Operation button (B) provides a convenient means to open, save, and delete presets.

Global Processing Presets

Presets can also be used for entire processing strips. The Open (A), Save (B), and Preset Operation (C) buttons work here exactly as they do for individual processing sections.

FM Conditioning - A Cool Guide To Optimized Deviation

FM radio broadcast is not only about audio. Instead the signal consists of different services that share the ‘space’ available on the FM carrier. A typical stereo radio signal spectrum may look like this

• Mono audio signal (M=L+R) - 30 Hz to 15 kHz base band

• Stereo pilot tone at 19 kHz - approximately 9 % of 75 kHz deviation

• Stereo audio signal (S=L-R) - 30 Hz to 15 kHz base band

• DSB-SC carrier - Double-sideband suppressed carrier

• RDS signal - Radio Data Signal at 1 187,5 Bit/s

• DARC signal - Data Radio Channel at about 16 000 Bit/s

• SCA signal - 14 kHz (narrow) or 26 kHz (wide) bandwidth for auxiliary audio services

To calculate the overall MPX Power the power spectrum of all consisting signals needs to be considered.

Please note that within the FM Conditioner Web UI, only RDS and SCA Deviation can be set as additional services. As SCA and DARC normally cannot be used simultaneously due to their overlapping frequency bands, the SCA Deviation parameter can also be used for DARC. To calculate the overall deviation, all of the services in use must be taken into account to ensure that they do not exceed the modulation limits defined by the ITU (as shown below). After setup, this process happens internally and is not a concern for the FM Conditioner user.

When dealing with processing for FM broadcast, four main parameters come into focus:

• Deviation Δf of the transmission frequency (carrier) fc

• MPX Power of the modulating signal (modulator)

• Pre-Emphasis to enhance the signal-to-noise ratio of FM transmission

• Baseband bandwidth of all involved services (audio signals and auxiliary data)

ITU-R BS.412

ITU-R BS.412 has standardized the maximum values for these parameters. Broadcasters must comply with these limits to not exceed the planned coverage or interfere with adjacent programs. They are:

• Maximum peak deviation of +75 kHz

• Maximum MPX Power of 0 dBr

• A typical audio baseband cutoff at 15 kHz to ensure undisturbed transmission of the 19 kHz stereo pilot tone

• For mono operation a typical audio baseband bandwidth of 17.5 kHz is utilized (no pilot tone is necessary)

Calibration

MPX Power is measured at random intervals of 60 seconds. An MPX Power level of 0 dBr should be equivalent to the modulation power of a stationary sine signal that induces a deviation of +19 kHz. A stimulus frequency of 500 Hz is recommended.

The tasks required to comply with this rule may sound 'simple' on the surface: take your pocket power measurement instrument, connect it to your readily accessible reference antenna, tune it to your transmitter, and take measurements. Then, adjust the relevant audio parameters if necessary. However, as this approach is not applicable for studio equipment, we must calculate MPX Power before modulation and then translate it to the studio output. To ensure precise calculations, all technical equipment must be gain-matched and calibrated.

The crucial step in calibration is setting the Operating Level. A stationary sine signal at this level should induce a 40 kHz deviation in the FM carrier. If the input level (at the FM HPA or uplink line) for this reference modulation is known, simply configure the Operating Level in the FM Conditioner accordingly. This is applicable in most installations.

In many stations, the reference level for a 500 Hz tone is +6 dBu (analog) or -9 dBFS (digital). It may be designated as the operating level and defined at 0 dB relative (as displayed on a peak level meter). However, please exercise caution with this type of reference level scale, as this analog operating level of 0 dBr is not the same as 0 dBr MPX Power.

If the reference modulation is unknown, you need to apply a sine test tone and measure the frequency deviation of the FM carrier over the air. Start with a generator level of -9 dBFS and adjust this value until you achieve a 40 kHz deviation. It's important to note that any processing in the signal chain between the generator and FM HPA must be bypassed during calibration. The calibration process should be carried out without considering any processing, additional services, or pilot signals.

If the Reference Level of your setup differs from -9 dBFS, you can use the Setup Gain of the FM Conditioner for level matching.

The second step of calibration involves configuring the values for the Pilot Tone, RDS, and SCA (DARC) Deviation. The required values depend on the settings of the respective encoders. Please refer to their manuals.

After completing the calibration process, the FM Conditioner will display the available audio headroom.

Operation

All calculation is performed internally and updates automatically whenever any of the involved parameters change. The resulting value is referred to as the 'Ceiling.' It's essential to understand that the Ceiling is calculated with the Pre-Emphasis filtering of the FM transmitter included. Therefore, the wideband true peak level of the audio signal before Pre-Emphasis must be lower. To better grasp this concept, you can refer to the level relation diagram:

Pre-Emphasis

Pre-Emphasis is a filtering system in which higher frequencies are boosted by a shelving filter at the transmission stage and conversely reduced at the receiver end. The Pre-Emphasis filter employs a time constant of 50 µs (or 75 µs in the USA), resulting in a 10 dB gain at 10 kHz. This process significantly improves the signal-to-noise ratio. However, as the increased high-frequency energy contributes to the MPX Power, it must be taken into account within the FM Conditioner.

There are two mechanisms to manage Pre-Emphasis. First, a Pre-Emphasis Headroom parameter reduces the maximum wideband level by lowering the true peak limiter threshold. This results in lower overall audio levels but enhanced high-frequency transparency. Second, a process called Pre-Emphasis Limiter dynamically reduces the high-frequency component of the audio signal, creating 'space' for the additional Pre-Emphasis shelving. The Pre-Emphasis Limiter is always active and prevents high-frequency overmodulation. To lessen its impact, the Pre-Emphasis Headroom should be increased. The Pre-Emphasis Limiter is based on sophisticated dynamic filter algorithms, well-known from the state-of-the-art Jünger Audio De-Esser.

True Peak Limiter

The Maximum True Peak value cannot be manually set by the user since it is automatically calculated and set to the Ceiling Level minus the Pre-Emphasis Headroom. When there's no Pre-Emphasis Headroom, the Maximum True Peak equals the Ceiling.

The MPX Limiter algorithm

The most crucial component of the FM Conditioner processor is undoubtedly the MPX Limiter. Since MPX Power is a value calculated over one minute of integration time, limiting can be a highly intricate task. In theory, a 60-second look-ahead time may seem appropriate, but it's not practically feasible for a real-time processor. Therefore, the Jünger Audio MPX Limiter employs a complex prediction algorithm that adapts to the incoming signal structure. Nevertheless, the limiter reference level remains an absolute brickwall threshold and is considered inviolable. In the event of an 'emergency,' the MPX Limiter will drastically reduce the signal level to prevent any threshold violation. The MPX Limiter in the FM Conditioner can be considered the most effective MPX brickwall limiter available today.

Please be aware that the MPX Limiter Reference can, of course, be exceeded when incoming levels are high and the MPX Limiter has just been activated. According to the measurement principle, it may take up to one minute for the MPX Limiter to stabilize

The MPX Limiter Profile impacts the speed and extent of the process and, consequently, its neutrality toward incoming sound quality. When using softer settings, the system requires a buffer zone between the MPX Limiter reference and the measured MPX of the audio signal. Although this buffer zone is always very small, with harder settings, it becomes even smaller, allowing for higher MPX Power transmission. The optimal setting depends on the type and style of the program being broadcast.

Parameters

Current Measurement

• MPX Power 60 s (dBr) - Currently measured MPX Power

• Duration - Past time since reset

• MPX 60 s Max (dBr) - Maximum MPX value since last reset

• MPX True Peak Max - Maximum True Peak value since last reset

• Gain Reduction Max - Maximum MPX Limiter Gain Reduction since last reset

• Reset Max - Resets Current Measurements and stores last values in Recent Measurement

Recent Measurement

• Duration - Past time for last measurement period

• MPX 60 s Max (dBr) - Maximum MPX value for last measurement period

• MPX True Peak Max - Maximum True Peak value for last measurement period

• Gain Reduction Max - Maximum MPX Limiter Gain Reduction for last measurement period

FM Conditioner

• FM Conditioner Enable - [ON / OFF]

• Setup Gain - [-4.0 … 10.0] dB

  Can be used to adapt loudness processed signals to MPX criteria or level matching

• Pre-Emphasis Headroom - [0.0 … 15.0] dB

• True Peak Limiter Profile - [0 ... 9]

• True Peak Limiter Threshold - No user parameter

• Pre-Emphasis - [OFF / 50 µs / 75 µs]

• Operating Level - [-16 ... -6] dBFS

• Peak Deviation Target - [35 ... 80] kHz

• Pilot Deviation - [0 ... 15] kHz

• RDS Deviation - [0 ... 4] kHz

• SCA Deviation - [0 ... 15] kHz

• Resulting Ceiling - No user parameter

MPX Power Limiter

• MPX Power Limiter Enable - [ON / OFF]

• Reference - [-4 ... 4] dBr

• MPX Limiter Profile - [Soft / Mid / Hard]

Pilot Tone Protection

• Low-Pass Filter 15 kHz - [ON / OFF]

Network Connection

Depending on the network infrastructure, connect one Ethernet interface to the control network and the other to the AoIP network. The network must be able to handle multicast traffic with low latency and must provide a PTP master. Pay careful attention to DSCP settings.

Network Configuration

Connecting to a DHCP server is recommended for the control network, at least for the initial configuration. If this is not possible, please use the server-specific management/deployment tool(s) to perform the initial setup of the IP interfaces.

For Dell servers, Junger recommends the iDRAC (integrated Dell Remote Access Controller) which provides a virtual console to access the Junger service menu by pressing <Alt> + <F1>. Note that the service menu program hides system (kernel) messages. It will re-start automatically when the system itself has closed it, or if it is closed by the user by pressing <F10>.

Alternatively, you may connect a VGA monitor and a keyboard to the server to access the console and navigate by using the keyboard arrow keys.

This will display the low-level System Status and the installed network interfaces. Choose <Network> to show the available NICs:

Highlight an active interface (that is, one connected to a switch or router on the network) and press <ENTER>. You may now manually assign an IP address.

If you have managed to reach the machine via TCP/IP, you can simply launch the GUI by using its IP address as the URL and perform an in-depth IP configuration as described under System > Network > Interfaces.

MADI / tieLight Connection

You may connect and daisy chain the flexAIserver with other flexAIsevers and/or AIXpressors via MADI or tieLight (Junger's proprietary high channel count, low latency fiber interface. It moves 1024 audio channels in both directions) using a Junger PCIe card, making it the ideal solution for edge processing as it provides the ability to add more processing power over time.

Synchronization

As with all networked AoIP devices, timing and synchronization are critical for proper operation, and it is assumed that a PTP stream synced with baseband video (such as BB, TriLevel, or SDI) and audio (such as AES3 or MADI) is in place.

Specific requirements will differ depending upon whether MADI/tieLight is used in conjunction with AoIP.

PTP Synchronization

PTP synchronization must be enabled via the System > Network > Interfaces pane where all network interfaces are listed. Select the network interface that is connected to a network with a PTP master and set the unit to "slave" mode to enable PTP synchronization.

• Hardware PTP Capable: Grey/green

• PTP Mode: Slave/disabled

• Status: Slave/disabled

• Locked to grandmaster: Red/green

• Grandmaster ID: MAC address of the grandmaster

• Distance to grandmaster: Number of boundary clocks in between; -1 means "no"

• Offset from master (us): Error of the control loop should be less than 1us

• Mean path delay: Measured by the PTP client

If the “Lock to grandmaster” soft LED lights up green, the flexAIserver is synchronized to PTP, indicating that the audio processing, the router, and the video reference for Dolby E encoders or similar are in perfect sync.

The Sync Source Priority setting at the Sync pane shows two more options, sync link1 and sync link2.

These are physical sync connections between multiple PCIe cards on additional flexAIservers to facilitate synchronizing the baseband audio interface on those cards using a sync link cable. Note that only one of the cards needs an external reference.

The three icons in the bottom (I) (see screenshot at Routing) manage the routing presets. There are three options:

The presets are related to their respective collection as explained in the next section on Routing Collections.

AIXpressor can optionally be equipped with a Dolby® E decoder and encoder. Please see for instructions on installing a new processor.

Dolby E Overview

Dolby E was introduced in 1999 as a professional audio coding system that allows the distribution of multichannel digital audio within two-channel infrastructures. Up to eight channels of broadcast-quality audio, plus metadata, can be distributed via a single AES3 pair.

Dolby E Decoder

The decoder will decode all standard Dolby E programs.

Input Page

Click on the Dolby E Decoder button (A), then on the In button (B) to display a running total of the number of frames received (C).

Decoder Page

Click on the Dec button (A) to display a list of all parameters (B), including (but not limited to) the following:

• Library Version - The version of the installed Dolby library.

• Input Signal - Green indicates a Dolby E signal; yellow indicates PCM audio; red indicates mute.

• CRC Errors - Displays the number of CRC errors in the incoming Dolby E stream. The counter can be reset with the Execute button at the bottom of the metadata display.

• Frame Rate - Indicates the frame rate of the incoming Dolby E signal.

• Frame Offset - Shows the offset between the incoming Dolby E signal and the system frame reference.

• Bit Depth - Indicates the bit depth of the incoming Dolby E signal.

• Program Configuration - Displays the Program Configuration metadata parameter, which determines how the audio channels are grouped within the Dolby E bitstream.

• Program Description - Shows the Program Description metadata, typically the name of the program or a description of the program source.

• Channel Mode - The Channel Mode (also known as the Audio coding mode) indicates the active channels within the encoded bitstream. In the decoder, this ensures the proper routing of the audio channels to downstream devices.

• LFE Channel - Indicates whether or not there is an LFE Channel present in the incoming bitstream.

• Bitstream Mode - Displays the audio service contained within the incoming Dolby E bitstream.

• Dialog Normalization - Also referred to as Dialog Level or dialnorm, this indicates the average level of dialog in the presentation and determines the proper level shift in the decoder to ensure proper normalization within and between program sources.

Meter Page

Click on the Mtr button (A) to view the decoder meter page. The Peak Level meters (D) show sample-accurate peaks for all eight incoming audio channels with a hold function. Peak Accuracy (B) is currently fixed at sample-accurate resolution. The Peak Hold control (C) allows for Automatic or Manual control of the hold feature. Clicking the Reset Measurement Execute button (E) resets the peak hold indicators.

Output Page

Click on the Out button (A) to show the number of clients (B) currently connected to the unit and a running total of the number of frames emitted (C) from the decoder.

Dolby E Encoder

Input Page

Click on the Dolby E Encoder button (A), then on the In button (B) to display a running total of the number of frames received (C).

Meter Page

Click on the Mtr button (A) to view the encoder meter page. The Peak Level meters (D) show sample-accurate peaks for all eight encoded audio channels with a hold function. Peak Accuracy (B) is currently fixed at sample-accurate resolution. The Peak Hold control (C) allows for Automatic or Manual control of the hold feature. Clicking the Reset Measurement Execute button (E) resets the peak hold indicators.

Encoder Page

Click on the Enc button (A) to display a list of all parameters (B), including (but not limited to) the following:

• Library Version - The version of the installed Dolby library.

• Encoder Status - Green indicates the encoder is active; red indicates it has stopped.

• Video Sync Frame Rate Status - Green indicates proper sync; red indicates a mismatch.

• Video Sync Frame Rate - The actual measured reference frame rate extracted from the sync signal.

• DSP 1/2 - DSP 7/8 - A label of the encoder inputs based on the signal type to ensure proper encoding for the chosen Program Configuration.

• Frame Rate - The Frame Rate of the Dolby E-encoded bitstream.

• Video Sync Shift Offset - Sets the guard band relative to the video switching point (see SMPTE RP 168) to ensure proper Dolby E alignment. Shift steps are measured in audio samples using a 48kHz sample rate.

• Program Configuration - Sets the Program Configuration metadata parameter, which determines how the audio channels are grouped within the Dolby E bitstream.

• Program Description - Sets the Program Description metadata, typically the name of the program or a description of the program source.

• Channel Mode - The Channel Mode (also known as the Audio coding mode) sets the active channels within the encoded bitstream.

• LFE Channel - Informs the downstream decoder as to whether or not an LFE channel is present.

• Bitstream Mode - Sets the audio service contained within the incoming Dolby E bitstream.

• Dialog Normalization - Also referred to as Dialog Level or dialnorm, this sets the average level of dialog in the presentation and determines the proper level shift in the downstream decoder to ensure proper normalization within and between program sources.

• Line Mode Profile - Sets the dynamic range metadata profile for line outputs of the downstream decoder.

• RF Mode Profile - Sets the dynamic range metadata profile for the RF outputs of the downstream decoder.

• DC Filter - Determines whether or not a DC-blocking 3Hz highpass filter is applied prior to encoding to remove DC offsets in the program audio.

• Lowpass Filter - Determines whether or not a lowpass filter is applied to the main input prior to encoding to remove high frequencies that would not be encoded, thereby preventing aliasing upon decoding.

• LFE Filter - Determines whether or not a 120Hz eighth-order lowpass filter is applied to the LFE channel prior to encoding, thereby removing frequencies that would cause aliasing when decoded.

Output Page

Click on the Out button (A) to show the number of clients (B) currently connected to the unit and a running total of the number of frames emitted (C) from the encoder.

Expand the Senders (TX) (A) portion of the AoIP menu, revealing the ADD/DEL buttons (B). Clicking ADD generates a new stream with the default settings and name. Clicking DEL deletes the selected individual stream. Each Sender (C) has its own configuration and status page.

Protocol (D)

Selects the type of AoIP stream. Choices include Livewire, AES67, and ST2110.

Status (E)

• Session Status - Displays the status of the current session.

• RTP Status - Displays the status of the Real-time Transport Protocol.

• SDP - Shows the Session Description Protocol (AES67 and ST2110 only).

• Provisional Stream Status - Displays live status information for the stream.

• Reset Sender Status - Resets the live status information shown in the Provisional Stream Status window.

• Interface 1 and Interface 2 - Displays the RTP status of the individual interfaces.

Settings (F)

• Enable - Enables or disables the stream.

• Stream Name - Allows for a unique name for the stream.

• Channel Count - Displays the number of enabled audio channels present in the stream.

• Packet Time - Selects the packet time; options vary depending upon the selected protocol.

• Seamless Protection Switching - When enabled, specifies the reconstruction of the original stream in case packets are lost in any path per SMPTE 2022-7.

• Codec - Selects the desired codec (AES67 and ST2110 only).

• DSCP - Selects the Differential Code Service Point.

• Auto RTP Payload Type - When enabled, automatically selects the RTP payload type regardless of what is specified in the RTP Payload Type field.

• RTP Payload Type - Allows the RTP payload type to be specified (AES67 and ST2100 only)

• RTP Port - Specifies the port used for RTP.

• TTL - Sets the Time to Live (packet lifetime) value.

• Media Interface 1 / Media Interface 2 - Enter the multicast destination address of the sender in the Destination IP Address field. When the Livewire protocol has been selected, an additional field for the Livewire Channel number appears.

Understanding the Routing Matrix

The routing matrix consists of three primary sections: A list of sources (A) in the left column, a list of destinations (C) in the right column, and a cross-point matrix (B) used for connecting sources to destinations in the middle column.

A Source can be any of the following:

• A physical input from the main PCB, such as AES3, analog, MADI, tieLight, and AoIP

• A physical input from interface slots 1-4, such as SDI, AES-3, analog, MADI, and Dante/AES67

• The output of one of the DSP processing blocks, such as Level Magic Channel Strip - Stereo or Program + Spectral Processing Strip - 5.1

A Destination can be any of the following:

• A physical output from the main PCB, such as AES3, analog, MADI, tieLight, and AoIP

• A physical output from interface slots 1-4, such as SDI, AES-3, analog, MADI, and Dante/AES67

• The input to one of the DSP processing blocks, such as Level Magic Stereo or SpectralSignature Level Magic 5.1

Example routing of a processor from and to physical I/Os

Getting a signal from one of the physical inputs to a DSP processing block and then ultimately to a physical output is a two-step process.

First, one of the physical inputs must be connected to the input of a DSP processing block.

This is accomplished by selecting audio channels of a physical input from the source side of the matrix (A) and then selecting audio channles of DSP processing blocks from the destination side of the matrix (B). Pressing the Link (New) button (C) will create the connection.

In Figure 2 below, channels AES 1 and AES 2 of the AES3 input are being routed to the Level Magic Channel Strip - Stereo processing block L and R for loudness control:

Second, the output of a DSP processing block must be connected to one of the physical outputs.

This is done by selecting the DSP processing block from the source side of the matrix and selecting the physical output from the destination side of the matrix.

In Figure 3 below, the processed output of the Level Magic Channel Strip - Stereo L and R is being routed to EMB1 and EMB 2 of the Slot 2: SDI embedded SDI output. Pressing the Link (New) button (C) will create the connection.

Useful Tools for Easy Routing

The Routing pane provides tools needed to make connections. They span from organizing the source and destination columns to several methods of creating connections (crosspoints).

If you have marked audio channels of a specific source (C), you can move them to the matrix by pressing the bluish arrow (D). This will prepare them to be assigned to their destinations.

AIXpressor has four AES3 interfaces proving a total of eight I/O audio channels. It may also be configured as four inputs and twelve outputs to support the monitoring of immersive audio formats.

Select AES3 (A) from the Interfaces menu.

Displayed information includes:

• Relay Bypass (B) - Enables or disables the hard relay bypass for the AES3 input. When enabled, AES3 I/O routing will default to the configuration set by the jumpers near the I/O connector. See the portion of the Technical Data section for details.

• AES Status Pair (C) - Indicates the status of the AES3 receiver (OK or Fail).

• Signal Status Pair (D) - Shows the signal status for each stereo pair.

• SRC Stereo Pair (E) - Enables or disables the sample rate converters for each of the four stereo pairs.

• I/O Mode (F) - Selects either the eight input/eight output mode or the four input/twelve output mode.

• Output Force Channel Status (G) - Options include Transparent, Auto, Prof PCM, Prof Non-PCM, Cons PCM, and Cons Non-PCM.

AIXpressor's built-in analog interface provides a total of eight inputs and eight outputs. It may also be configured as four inputs and twelve outputs to support the monitoring of immersive audio formats.

Select Analog (A) from the Interfaces menu.

Displayed information includes:

• Headphone Status (B) - Indicates whether or not headphones are connected.

• I/O Mode (C) - Selects either the eight input/eight output mode or the four input/twelve output mode.

The SDI Interface menu contains sub-pages for Status, Setup, the De-Embedder, and the Embedder.

Status

Select Status (A) from the Interfaces > SDI menu.

Displayed information includes:

• SDI Status (B) - Shows the status of the internal clock as Locked or Unlocked.

• Video Format (C) - Indicates the video format as SD, HD, 3G, or N/A.

• Video Standard (D) - Displays the actual decoded standard (e.g., 1080i50), no SDI lock, or TAG last valid.

• Overview (E) - Graphically shows the status and potential issues within the SDI signal path. For example, If there is no SDI reference present, the input portion of the graphic will turn red.

• Audio De-Embedder Status (F) - Indicates the format of the de-embedded signal as PCM, Dolby E, Dolby Digital, Dolby Digital Plus, MPEG-4 HE AAC, MPEG-4 AAC, or N/A.

• VANC Metadata De-Embedder Status (G) - The indicator turns green when the SDID is detected in the stream; the pre-selected stream as established in the de-embedder setup stream is also identified.

• Audio Embedder Status (H) - Indicates the status of the embedder as follows:

  • AUTO Embedding - A new group will be built.

  • AUTO Replace - The structure of the group from the input is retained and the audio content is simply replaced.

  • Delete - The group from the input is deleted.

  • OFF - The embedder for the group is disabled.

• VANC Metadata Embedder Status (I) - Shows the status of the embedder as Enabled or Disabled (for details, see the SMPTE ST 2020-2 standard).

• ARIB STD-B39 Control Data Status (J) - Indicates the status as Available or Not Available along with the Audio Mode; for more information on the ARIB Japanese standard, see "Structure of Inter-Stationary Control Data Conveyed by Ancilliary Data Packets" at

• Embedder Metadata Errors (K) - Indicates metadata errors including No Show, Overwrite, Overwrite Block, Overflow, and Audio Sync errors plus the status of the test pattern generator.

Setup

Select Setup (A) from the Interfaces > SDI menu.

The SDI Bypass section (B) contains two controls:

• The SDI Relay Bypass, when enabled, de-activates the bypass relay which serves to connect SDI-IN directly to SDI-OUT1 in the event of a power failure. It also disconnects the de-embedder from the SDI input.

• The SDI Embedder Bypass, when enabled, will pass embedded audio data from the de-embedder directly through to the embedder 1:1, preserving the original Ancillary Data structure.

The Video Delay section (D) sets the video delay in whole frames to compensate for any latency incurred in the audio chain. "0" turns off the delay function.

The Level B Stream Select dropdown in the 3G SDI Mode section (E) allows Stream 1 or Stream 2 to be used for the embedded audio per SMPTE 425M for details should the 3G-SDI signal contain two HD sub-streams.

The Test Pattern Generator section (C) offers a test generator to either check downstream connections during installation or for use when there is an input source failure. It can also be used to move 16 independent audio channels over a single coax cable from point to point.

The Mode menu includes OFF, AUTO (Input Loss), and Always ON.

The Video Format menu sets the generator's desired output format (resolution and frame rate) and whether it displays color bars or a black frame.

De-Embedder

Select De-Embedder (A) from the Interfaces > SDI menu.

The HD SDI standard allows for asynchronous audio, which may require the use of an embedded Wordclock, and is critical in applications that synchronize the audio transport from the SDI de-embedder.

The Audio Sync Source section (B) sets the Embedded Wordclock source.

• Auto - Asychornous audio is automatically sync'd to the Wordclock of De-Embedded Channel 1.

• DEM1 - Asynchronous audio is manually sync'd to the incoming SDI signal.

The VANC Metadata De-Embedder section (C) enables or disables the metadata de-embedder; when enabled, the Stream Select menu chooses the individual stream source for the metadata (SSID 1 - SSID 9).

Embedder

Select Embedder (A) from the Interfaces > SDI menu.

The Audio Embedder Section (B) contains the following controls:

• Delete Existing Data - Choices include ALL, New HANC Structure, and OFF.

• Each Group, 1 - 4, has its own Embedding Mode menu. Choices include:

  • OFF - Disables embedding altogether.

  • AUTO - Embedding

  • AUTO - Replace Audio

  • Delete

• The AES Channel Status menu offers a choice of Transparent or Professional.

VANC Metadata can be embedded using the following controls in the VANC Metadata Embedder Section (C).

• Enable - Enables and disables the metadata embedder.

• Delete Existing Metadata - All / OFF.

• Stream Select - Determines into which stream the metadata is inserted (SSID 1 - SSID 9).

• Vidoe Line - Selects which line number receives the metadata, which varies depending upon the video standard employed and how many lines are available for data insertion.

Each embedded audio signal can be delayed independently in the Embedder Audio Delay section (D). This feature is useful for maintaining lip sync when if a video delay is used. Values are in milliseconds (ms) with a range of 0 - 340.

Device

Select Device (A) from the Interfaces > SDI menu.

The Parameter Version (B) and Firmware Version (C) of the SDI interface are displayed.

AIXpressor has two independent optical interfaces which can be configured for tieLight and/or MADI.

tieLight is a proprietary Jünger interface providing high channel count and low latency - up to 1,024 bi-directional audio channels with a total latency equalling four samples. It is designed to daisy-chain multiple AIXpressor units and/or flexAI servers.

AIXpressor has a built-in tieLight interface, while flexAI servers use an optional PCIe card available from Jünger.

Select Setup (A) under tieLight / MADI from the Interfaces menu. Use the Interface Transmission Mode dropdown (B) to set the mode for both interfaces. Choices include:

• MADI independent - Both interfaces use MADI but operate independently from one another.

• MADI redundant - Both interfaces use MADI but are configured identically to provide redundancy.

• tieLight independent - Both interfaces use tieLight but operate independently from one another.

• tieLight redundant - Both interfaces use tieLight but are configured identically to provide redundancy.

• IF1 MADI, IF2 tieLight: Interface 1 uses MADI; Interface 2 uses tieLight.

• IF1 tieLight, IF2 MADI: Interface 1 uses tieLight; Interface 2 uses MADI.

When using tieLight, PTP sync distribution across multiple devices can be set in the tieLight PTP Role dropdown (A). Choices include Master, Slave, and Passive.

SFP Status Pages

SFP 1 and SFP 2 each have independent but identical status pages (A).

Displayed information includes:

• RX Status (B) - Shows the current status of the selected optical input as reported from the SFP module.

• Signal Status Stereo Pair (C) - Indicates the presence of a signal for each pair as well as the type of signal (PCM, Dolby E, Dolby Digital, Dolby Digital Plus, MPEG-4 HE-AAC, MPEG-4 AAC, and N/A). This information is gathered and analyzed by the core FPGA router using the signal headers of adjacent odd and even signals (stereo pairs). In addition to simply showing status, the information can be used globally for other functions including event management or muting processor inputs.

The Routing matrix can be subdivided into collections of cross points, each having its own preset. This allows you to create groups of connections.

The names of the Collections appear in the top of the cross-point matrix (A).

Connections are managed by the little icons in the upper right-hand corner (B). The Edit icon opens a menu where you can + ADD, Name / Rename, and Delete collections from a list. The eye symbol toggles between the view of one or all connections, selectable by highlighting and underlining their names when selected (A).

Expand the Receivers (RX) (A) portion of the AoIP menu, revealing the ADD/DEL buttons (B). Clicking ADD generates a new stream with the default settings and name. Clicking DEL deletes the selected individual stream. Each Receiver (C) has its own configuration and status page.

Protocol (D)

Selects the type of AoIP stream. Choices include Livewire, AES67, and ST2110.

Status (E)

• Session Status - Displays the status of the current session.

• RTP Status - Displays the status of the Real-time Transport Protocol.

• Provisional Stream Status - Displays live status information for the stream.

• Interface 1 and Interface 2 - Displays the RTP status of the individual interfaces.

Settings (F)

• Enable - Enables or disables the stream.

• Stream Name - Allows for a unique name for the stream.

• Channel Count - Displays the number of enabled audio channels present in the stream.

• Seamless Protection Switching - When enabled, specifies the reconstruction of the original stream in case packets are lost in any path per SMPTE 2022-7.

• SDP - Allows the Session Description Protocol information to be entered (AES67 and ST2110 only). SDP information will be gathered and displayed in the Discovery section and may be copied and pasted here.

The parameters below are available when Show Expert Settings is enabled:

• Codec - Selects the desired codec (AES67 and ST2110 only).

• Auto Link Offset - Enables or disables auto link offset.

• Link Offset - When Auto Link Offset is disabled, allows a specific link offset value to be added to the minimum RTP offset to keep the RTP offset value positive.

• Ignore RTP Payload Type - When enabled, the RTP payload type will be ignored.

• RTP Port - Specifies the RTP port.

• Interface 1 and Interface 2 - Enter the multicast destination address of the receiver in the Destination IP Address field. When the Livewire protocol has been selected, an additional field for the Livewire Channel number appears.

Discovery (G)

The Discovery section shows the senders available via Livewire, AES67, and ST2110. When highlighted, the details of each stream will be displayed. A Filter field provides a means of entering a stream name or keyword to minimize the amount of scrolling required on a large network. Clicking the Apply button will copy the announcement information to the current receiver.

Select Setup (A) from the System menu. Setup fields include:

• Hostname - The hostname for this particular AIXpressor unit.

• System Location - A "friendly" field for identifying the physical location of the unit within the facility.

• System Contact - A "friendly" field for listing an administrator's name.

AIXpressor has a 6.3mm (1/4") front-panel analog headphone jack.

Select Headphones (A) from the Interfaces menu to adjust the output volume. Set the output volume by clicking and dragging the Headphone Gain slider (B). Enabling High Gain (C) boosts the headphone output to drive high impedance headphones.

Headphone volume can also be adjusted on the front panel by touching the Headphone symbol (A) from the home screen, then using the slider.

If your flexAI system is connected to an open network, secure communication is a must.

Click on Certificates (A) to open the page

The certificate (B) you initially find on a flexAI system is a so called self signed certificate. If you have a CA (certificate authority) in our network you can register this certificate by generating a CRS (D) (Certificate Signing Request) file that is used for that process. But you can also import (C) a certificate that includes the IP address and/or the DNS name of the flexAI system. If it is registered with your browser as well, this will avoid error messages when you start a new HTTPS session (if your browser automatically deletes cookies and the history). You are able to dowload this certificate (E) or simply display it (F) FYI.

The flexAI engine within AIXpressor and all audio processing blocks are license-bound. Stand-alone units will handle licensing through a USB dongle, while larger installations use a WiBU license server.

Select Licensing (A) from the System menu to view the license page.

The Product Name and Product Code are listed in the License section (B). Quantity indicates the number of installed licenses for a given product. Available indicates the number of available licenses not currently in use.

Should you purchase additional licenses in the future, you may be asked to provide a license context file by clicking the Download License Context File (C). New licenses can be uploaded using the Update License Update File button (D).

Since flexAI is a Linux based system, the firmware consists of multiple packages. So one my either update individual packages (meant for experienced users, developers and service) or all relevant packegaes for a complete release.

flexAI firmware can be updated in three ways:

• From a firmware image on a USB stick plugged into a USB port

• From a local image on a computer that is connected to the same network the flexAI hardware is part of.

• From our online server - This is currently not available in the field.

Default Update Page

The default view of the Update Page allows for the three steps of an update to a new flexAI release. From here you can select an update source (A) and start the update (B) and finally Reboot (C) the device to restart all processes.

Extended Update Page

Clicking the Setup icon (C) opens a window with options to:

• Automatically update packages when a new release is installed (default = on)

• Show advanced package management (meant for experienced users, development and service - default = off)

• Only show Junger packages (default = on)

When Show advanced packet management is turned on it shows the installed packages (B) in detail. Use the Display column (A) to select whether All available packages are displayed at once or just the content from a specific section.

Selecting an Update Source

To begin an update, click on the Change Update Source button (A). A Choose Package Source window (B) will open. Click on the desired source. When the pop-up window appears, follow the on-screen instructions, which will vary slightly depending on the selected update source.

When pressing the EXECUTE UPDATE buton a "Refreshing Repository" progress window will appear:

After all files are copied to the internal memory of te flexAI device a "CAUTION" message appears:

Press PROCEED and be patient. Once the update is complete and the "Update Successful" window appears, reboot the hardware:

When connecting to a flexAI device by a browser you must now log in to it. A Sign In po-up appears:

When connecting to a flexAI device by a browser you must log in to it. A Sign In po-up appears:

The default credentials are User name: "admin" and Password: "admin".

After signing in, the initial page of the UI appears and the overall status of the system enters the ERROR sate. Clicking on the System Status icon in the upper right corner (A) :

It says that the password for the user admin is not set so you should do it as the very next step!

Cockpit shows all users of the system on this page. For the moment, only user admin (B) is relevant.

Click on the <Users> button (A) in the navigation bar at the left side:

For now only the password for the user admin (B) can be changed via the Users pane:

The other users are simply presented here FYI. They are generic system users meant for service and development applications and will used in a wider security concept by your IT team.

The Dante interface provides 32 AoIP streams with a total of 64 audio channels to and from the Brooklyn II module on the Jünger DT-100 board. All settings can be performed via the Dante Controller software available for download from the Audinate website:

Figure 13 below shows an example of the routing matrix of the Dante Controller where transmit and receive channels from various Dante interfaces are connected. The interface is color-coded. Red indicates the module resides in a different subnet. Blue represents non-Dante interfaces discovered on the AoIP network that provide AES67 multicast streams (such as Ravenna). Black represents native Dante interfaces operating in either Dante or AES67 mode. Note that if two Dante modules operating in AES67 mode exchange audio signals, the streams will default to Dante Unicast mode.

We strongly recommend studying the numerous documents available from AIMS and the Internet Engineering Task Force to familiarize yourself with the terms used in AoIP networks from PTP over mDNS to Layer 3 routing. For Dante-specific documents are available on the , and imparting such knowledge is beyond the scope of this manual.

When a Dante interface is installed, it will be automatically named by AIXpressor.

Status

Select Status (A) from the Interfaces > Dante menu. The following information is displayed in the status screen (B):

• Device Name - The name of the Dante interface used by the Dante controller software to locate it.

• Sync Source - The AoIP network to which the Dante interface is connected.

• Sync Status - Indicates whether the interface sync is locked or unlocked.

• Preferred Master - Indicates whether this Dante interface is set to be a preferred master for other Dante devices.

• Network Audio Sample Rate - Indicates the sample rate for AoIP audio.

• Device Latency Setting - Defines the size of the buffer size in milliseconds (ms) required to handle network-related latencies.

• AES67 Enabled - Indicates whether or not the interface can connect to both Dante and AES67 networks.

• Device Access Lock - Shows whether the interface is unlocked or locked to prevent authorized access to the module.

Receiver

Select Receiver (A) from the Interfaces > Dante menu. The following information is displayed:

• Channel Label (C) - These are the labels for the 64 transmit channels; they display as numbers from 01 to 64 by default but can be customized using the Dante Controller.

• Connection Status (D) - Shows the connection to each channel as OK, Fail, or N/A.

• Signal Status Stereo Pair (E) - The routing FPGA analyses the signal headers of adjacent odd and even signals that may form a stereo pair. This status information can be used globally for event management or to mute processor inputs. The type of signal is also displayed (PCM, Dolby E, Dolby Digital, Dolby Digital Plus, MPEG-4 HE-AAC, MPEG-4 AAC, or N/A).

• Connected to (F) - Receiver channels must be routed (connected) to transmit channels within the AoIP network. The Dante Controller must be used for signal routing. The receiver shows the transmission channel label and the device name separated by an "@".

• Subscription Status (G) - This indicates whether the receiver is subscribed and if so, to which type of connection. Options include Connected (Unicast), Connected (Multicast), No Subscription, and Subscription Unresolved.

Transmitter

Select Transmitter (A) from the Interfaces > Dante menu. The following information is displayed:

• Routing Label (B) - The routing label is automatically assigned by AIXpressor and serves as a reference in the Routing matrix of the Dante Controller.

• Channel Label (C) - The channel label must be specified and assigned in the Dante Controller.

Primary Network

The Dante interface has two network connectors that may be configured as redundant or switched.

In redundant mode, the interface can be connected to two independent AoIP networks to provide "Seamless Protection Switching" as defined in SMPTE 2022-7.

In switched mode, both networks ports combine to build an Ethernet switch using the built-in Brooklyn II Ethernet interface.

Select Primary Network (A) from the Interfaces > Dante menu.

The Connection Status (B) section shows the following information:

• Network Status - Indicates if the interface is connected to an active network and at what speed (Connected 1G, Connected 100M, Offline, Pending, N/A).

• PTPv1 and PTPv2 Status - Shows the status of the PTP clocks (Startup, Initializing, Faulty, Disabled, Listening, Pre Master, Master, Passive, Uncalibrated, Slave, NA).

The Current Network Status (C) contains the following information:

• Current Mode - Redundant or Switched.

• Current Enable DHCP - On or Off.

• Current IP Information - Including IP address, Netmask, DNS Server, Gateway, and MAC Address.

The Change Network Settings(C) section allows the following information to be changed:

• Request New Mode - Redundant (the default setting) or Switched.

• Config IP Information - Allows changes to DHCP (enabled by default), IP Address, Netmask, DNS Server, and Gateway values.

Secondary Network

Settings for the Secondary Network are identical to the Primary Network with one exception in the Request New Mode setup. If the Secondary network port is not connected or the Primary port is set to "Switched," the network status changes to "Offline" and no PTP status is available.

The Network menu includes setup and status information for DNS, NTP, PTP, Livewire Clock, SNMP, and each of the available LAN interfaces.

DNS

Select Network (A) from the System > Network menu to begin, then choose DNS (B) to open the DNS page. Displayed information (C) includes:

• DNS State - Displays the IP address of the actual DNS server(s).

• mDNS Node Name - Shows the name used for mDNS (Bonjour, Avahi)

• DNS - Field for manually entering the IP address of a DNS server.

NTP

Select NTP (A) from the Network menu to open the NTP page. Displayed information (B) includes:

• NTP State - Displays the current NTP server.

• Enable NTP - Enables and disables NTP.

• Use Fallback NTP - When enabled, uses the internet NTP servers at debian.pool.ntp.org if the unit cannot connect to the specified NTP servers.

• NTP Server - Fields for entering up to four local NTP server addresses.

PTP

Select PTP (A) from the Network menu to open the PTP page. Displayed information (B) includes:

• PTP Clock Domain - The field for entering the proper PTP clock domain.

• PTP DSCP Class - Options include 0, 8 CS1, 10 AF11, 16 CS2, 18 AF21, 24 CS3, 26 AF31, 32 CS4, 34 AF41, 40 CS5, 46 EF (AES67), 48 CS6 (Ravenna), and 56 CS7. 48 CS6 (Ravenna) is the default setting.

Livewire Clock

Select Livewire (A) from the Network menu to open the PTP page. Displayed information (B) includes:

• Mode - Choose "Follower" or "Disabled," as Livewire cannot act as a Leader.

• Network Interface - Selects the network interface used for Livewire.

• Locked to Leader - Indicates when Livewire is locked to the Leader clock.

• Leader ID - The IP address of the Leader clock.

• Offset - Displays the difference (delay) between the local clock and the Leader in microseconds.

• Jitter - Displays the variations in delay between the local clock and the Leader in microseconds.

SNMP

Select SNMP (A) from the Network menu to open the SNMP page. Displayed information (B) includes:

• Enable - Enables and disables SNMP.

• Community - Identifies the SNMP Community string.

• Host Name, System Location, and System Contact - Populated fields from the System Setup menu.

• Trap Sink IP Address - The IP Address of the SNMP trap receiver.

• Trap Sink Port - The port of the SNMP trap receiver.

LAN Interfaces

A total of four LAN interfaces are available. The LAN 1 CTRL and LAN 2 interfaces operate in a similar fashion to one another, so the information below applies to both.

LAN 3/4 differs from LAN 1 and LAN 2 in that they are internally connected by a switch to the network chip which is in turn connected to the SoM via a PCIe link. Therefore, they share the same IP Address, Gateway, MAC Address, and PTP settings.

Interface Menu

Choose the individual Interface (A) for each LAN under the Network menu. In the example below, we've chosen the LAN 1 CTRL interface. Displayed information (B) includes:

• Link Up - Indicates the presence of an active, working link (True or False).

• Speed - The current link speed and type (100Mb/s, Full Duplex).

• RX Bytes - Indicates the number of bytes received since the link was made active.

• TX Bytes - Indicates the number of bytes transmitted since the link was made active.

• RX Throughput Bps - The actual measured receive throughput.

• TX Throughput Bps - The actual measured transmit throughput.

• MAC Address - The MAC address of the interface.

IP Menu

Select IP (A) for the selected LAN under the Network menu.

The Current IP Addresses section (C) displays the current dynamic and static IP addresses.

When Use Dynamic Addresses (B) is enabled, the unit will automatically retrieve an IP address from the network's DHCP server. When disabled, the specified static IP address is used instead.

Static IP Addresses are listed in the Static Addresses window section (D). Clicking the Edit button (E) will open a pop-up window where the address type (IPv4 or IPv6) and subnet mask may be defined. Clicking the Delete button (F) removes the selected static IP address.

To add a new static IP address, click on the Add IP Address button (G), which will open a pop-up window where the address type, IP address, and subnet mask information can be entered.

Clicking the Restore Factory Defaults button (H) will set the LAN to DHCP and remove all static IP addresses.

Gateway Menu

Select Gateway (A) for the selected LAN under the Network menu.

The Current Gateways section (B) lists the current gateways.

The Static Gateways section (C) lists the IP addresses of any manually-added gateways along with any specific actions. To add a new gateway, click on the Add Gateway button (D) which will open a pop-up window to enter the new address.

PTP Menu

Select PTP (A) for the selected LAN under the Network menu.

The Status section (B) contains the following information:

• Hardware PTP Capable - Indicates when the Ethernet hardware is capable of extracting a PTP sync.

• PTP Status - Options include None, Initializing, Faulty, Disabled, Listening, Pre-Leader, Leader, Passive, Uncalibrated, and Follower.

• Locked to Leader - Indicates when PTP is synced to a Leader clock on the network.

• Leader ID - Displays the MAC address of the PTP Leader.

• Distance to Leader - Shows the number of hops between the Leader and flexAI unit.

• Offset from Leader (us)

• Mean Path Delay (us)

The Configuration section (C) contains the PTP Mode control. Options include Disabled and Follower, as flexAI cannot be used as a PTP Leader device.

Select Power (A) from the System menu to view the power page.

To reboot AIXpressor without powering the unit completely down, press the Reboot button (C). To completely power down the unit, press the Shutdown button (B).

This is a display provided by Cockpit to show the storage usage in the system:

For hardware and software support Telos Alliance offers the TelosCare PLUS program:

Select Backup / Restore (A) from the System menu to view the backup and restore page.

To create a backup file containing all audio processing-related settings, click on the Download a Backup File button (B).

To restore the unit from a previously saved file, click on the Select a Backup File button (C), choose the appropriate file from your computer or network, then click the Restore button (D).

The Info menu provides a comprehensive list of various statuses. It is broken into five categories: SYSTEM, HARDWARE, SOFTWARE, SYSTEM LOAD and ABOUT.

SYSTEM STATUS

Select Info (A) from the System menu to view the SYSTEM STATUS, which displays by default.

The Overall Status indicator (B) provides an overall indication of the system. A green "OK" doesn't necessarily mean that there are no abnormal statuses anywhere in the system, but rather that AIXpressor is functioning well overall.

Use the Type buttons (C) to view or filter specific types of statuses, including All, OK, Warning, Error, Info, and Inactive. Multiple buttons may be enabled simultaneously.

The Component dropdown (D) allows you to select which components are included in the detailed status report. Options include All, Service, System, Hardware, Audio Processing, Routing, Sync, Network, and Interfaces.

Clicking the Edit button (E) allows you to exclude certain groups of components from the system status page.

HARDWARE STATUS

The HARDWARE STATUS page (A) is primarily meant for diagnostic and service purposes. It lists the current firmware version along with various temperature, fan, and power supply statuses. Example from an AIXpressor.

SOFTWARE STATUS

The SOFTWARE STATUS page (A) provides details about the software versions currently installed on a flexAI system.

SYSTEM LOAD

The SYSTEM LOAD (A) page provides an aggregated view of the variuous system components that may have influence on the performance. When <Show Expert View> (B) is enabled it will provide a time weighted (integrated) display. In the bottom of the moving sections you see the legend of the various colors.

ABOUT

The flexAI systen is based on a Debian Linux distribution. It contains various software packages which are bound to GNU GPL (Genral Purpose Licenses). This page (A) lists all common and Debian package relevant license information. You may click on one of the <DETAILS> buttons (B) to download and read the respective license information.

is mandatory for flexAI. For full warranty information, please see

AIXpressor provides a comprehensive logging system based on information available in the Linux operating system. Logged information from the Kernel, running daemons, I/O operations, and user applications are stored in internal flash memory for future analysis.

Select Logging (A) from the System menu to view the log files.

Although the log files may be viewed by the end-user, the information is primarily meant for troubleshooting with customer support. Additional access to the log information may be included in future versions.

You can sort the logs by Severity and view All or just applications specific logs selected by the App: pull down (B).

This is a graphical emulation of a SSH (Secure Shell) terminal. When you click on Terminal in the navigatin column (A) it opens in your (admin) home directory (B):