Source Profiles

As with other Axia consoles, Altus uses “profiles” to categorize audio that is present on the network and to define the behavior of the console. Profile types include:

• Source Profiles – Defines the audio inputs

• Show Profiles – Determines the layout of the console

• Audio Outputs – Defines the audio outputs

Profiles are set up through the Altus Control Center, accessible by launching your preferred web browser on a computer connected to your studio network and entering the IP address of the Altus in the address bar. When prompted for authentication, enter the user name “user”. The password field is left blank.

Source Profiles

To create a Source Profile, first click on Sources (3-1A) in the Profile Manager section of the Altus Control Center menu.

Next, click on the Input Source dropdown (3-1C). Available options include:

• Line Input – Used for any general audio source. A GPIO logic port can be used to provide machine start/stop pulses if desired.

• Computer Player – Similar to Line Input, but with a different logic control commonly found with PC-based automation systems.

• Phone – Defines the source as a hybrid or broadcast phone system input. A summed mono mix-minus is automatically created for the source, and controls for Telos products will be available if defined as part of the profile.

• Codec – Generates a dual mono mix-minus for the source consisting of one PA feed (right channel) and one talent feed with talkback (left channel).

• Operator Microphone – This is the board op’s mic and the source for Altus “talk to” functions. Its logic mutes CR (control room) monitors and Preview audio when on.

• CR Producer Microphone – This is for an in-studio producer’s mic. It has associated GPIO logic to operate “talk to” functions from a remote producer’s panel and mutes CR monitors and Preview audio when on.

• CR Guest Microphone – Used for any other mic in the control room. Its associated logic automatically mutes the CR monitors and Preview audio when on.

• Studio Guest Microphone - Used for any mic in a separate studio. Its logic mutes the Studio monitors when on.

• External Microphone – Used for any mic located outside of the CR or studio. It functions like a Guest microphone, but without any muting logic.

• Studio Feed – Defines a source that receives a backfeed and generates an IFB backfeed in return. This is intended for an external studio source for which you wish to create a talkback channel.

Clicking on the Create button (3-1B) changes the display to show all options associated with the selected source type. Some options are common to all source types, while others are unique to a particular source.

Note: Detailed information on the GPIO logic functions for each Source Profile as well as detailed pin-out information per profile are provided in the section on GPIO

Source Profile Options

Some of the Source Profile options below apply to all Input Source types while others will appear only for certain sources. For example, dynamics processing (noise gate, compressor, de-esser) is included for the various microphone inputs, but not for a Computer Player or Line.

As a result, what is shown on screen may differ somewhat between source types.

Source Name

Each source needs a unique name so that it can be displayed on the console’s channel display. Up to 10 characters (which may include spaces or underscore characters) can be entered in the Source name field (3-2A).

Source Input

The audio input source is set by the Source input dropdown (3-2B). The options available in the Primary Source dropdown are dependent upon this selection. Options include:

• Livewire – Livewire AoIP audio.

• Livewire Backfeed – The backfeed streams created by consoles on the network.

• Livewire Surround – The last two channels of the original Livewire Surround stream are a stereo downmix of the 5.1-channel content which can be made available here as an input.

• AES67 SIP – Provides a unicast option within the AES67 standard.

• AES67 Multicast SAP – The most typical AES67 stream type. If an AES67 source is advertised though SAP, the browse button in the Primary source field will have the source listed.

• AES67 Multicast 16-bit – 16-bit streams are supported but not automatically detected; use this option to manually identify a 16-bit stream.

• Sine 1kHz -20dBFS – An internally-generated tone for analog setup and testing.

• V-Mixer – An internally-created mixer for special use cases.

• WebRTC (WebRTC contribution license required) - establish a webRTC input where a host would be remote from the studio and able to contribute content. Consider it a built in codec. It is suggested to use the codec source type for this option.

Primary Source or Name

The specific AoIP source is entered in the Primary Source field (3-2C). What is entered here is dependent on what is selected for "Source input".

• Livewire – A Livewire channel number (1-32767) is expected; the Browse button will show any sources found through the Livewire discovery channel.

• Livewire Backfeed - A Livewire channel number (1-32767) is expected; the Browse button will show any backfeed streams found through the Livewire discovery channel.

• Livewire Surround - A Livewire channel number (1-32767) is expected; the Browse button will show any surround streams found through the Livewire discovery channel.

• AES67 SIP – A SIP URI scheme is expected (sip:user@host)

• AES67 Multicast – A multicast address is expected. Is the source is advertised through SAP, the browse button will present items that could be selected.

• AES67 Multicast 16-bit – A multicast address is expected.

Signal Mode

The Signal Mode dropdown (3-2D) determines whether the source will be treated as mono or stereo.

• Stereo – Feeds an incoming L/R to the left and right channels of assigned mixes.

• Left – Feeds an incoming left channel to both left and right channels of assigned mixes.

• Right – Feeds an incoming right channel to both left and right channels of assigned mixes.

• Sum L+R – Creates a mono mix of an incoming stereo source and feeds it to both the left and right channels of assigned mixes.

Signal Phase

The Signal Phase dropdown (3-2E) determines whether the source will maintain normal phase or be inverted.

Signal Mode for Record Bus

The Record Bus is a special variation of the Program 4 mix that by default is post-fader but pre-ON/OFF. The Signal Mode for Record Bus dropdown (3-2F) determines how the source will feed this mix.

• Stereo – Source is delivered to the Record mix in stereo.

• Sum to Left – Creates an L+R sum of the source and delivers the summed signal to the Left channel of the Record mix.

• Sum to Right – Creates a L+R sum of the source and delivers the summed signal to the Right channel of the Record mix.

The Sum to Left and Sum to Right options create the ability to provide a split Record, a common setup for phone recorders. The Operator mic can be recorded to one side of a stereo channel while all other sources are recorded on the other channel.

Record Insert Mode

By default, the Record mix is a post-fader feed. It can be switched to a pre-fader feed with the Record Insert mode dropdown (3-2G).

Input Trim Gain

The Input Trim Gain field (3-2H) allows you to specify a gain adjustment of between -25dB and +25dB to the signal path just ahead of the fader.

Panorama Position

The Panorama Position field (3-2I) is a pan (left/right balance) control. “0” is the center position. -24 is a hard pan left, while +24 is a hard pan right. This setting can be adjusted on the fly by the operator if permitted in the Show Profile settings.

Audio Delay

Adjusting the Audio Delay (3-2J) allows audio to be delayed by a specific amount in milliseconds. This gives control of the input buffer which is needed for jittery AES67 sources. Computer sources are notorious for being jittery and incrementing this value can clean up the audio.

Synchronous Mode

Enabling Synchronous Mode (3-2K) strictly uses the incoming AES67 timestamp plus any specified link offset value to compensate for any network delay (which can add many milliseconds over a typical WAN) when receiving audio. Using Synchronous Mode provides the lowest possible latency - typically one packet (1ms) less than if it was disabled – which is especially critical for the mic-to-headphone monitor loop.

When Synchronous Mode is disabled, Altus plays the incoming stream with packet buffering plus any additional link offset. It automatically works for WAN and legacy Livewire streams where the timestamps are not PTP accurate, but the exact alignment of the incoming stream is unknown and there is no consideration for overall network delay.

Dynamics processing, including a Noise Gate, Compressor, and De-esser, is available for Microphone source types in the Microphone processing menu (Figure 3-3). The checkbox to the left of each section enables that particular component.

The Noise Gate automatically reduces the gain of the microphone when input levels decrease to prevent background noise from being heard or increased either by the compressor in the channel strip or in the station’s main audio processor.

• Threshold – When the input signal falls below this level, the Noise Gate activates. Higher values (toward 0dB) will cause the gate to activate sooner, while lower values (toward -50dB) will allow input audio to fall to a lower level before it activates.

• Depth - Determines how much attenuation is applied to the input signal once it crosses the specified Threshold.

Compressor

The Compressor reduces the dynamic range and peaks of the incoming audio to help smooth out audio levels at the output.

• Threshold – The Compressor begins working once the input audio exceeds the value of the Threshold; audio below the Threshold is not compressed.

• Ratio - The Ratio control sets the amount of processing that takes place and controls how “aggressive” the processing sounds. The Radio is the amount of change to the input level in dB that is required to change the output level by 1dB. For instance, a “loose” ratio of 2:1 would require a 2dB change at the input to yield a 1dB change at the output, allowing more dynamics from the mic. A “tight” ratio 16:1 ratio would require a 16dB change at the input to yield a 1dB change at the output and result in a more dense, consistent sound.

• Freeze Mode – When audio falls below a pre-determined threshold, the action of the compressor will freeze to prevent background noise from being increased.

De-Esser

The De-Esser reduces sibilance caused by “s” sounds that can often cause distortion in the signal.

• Threshold – The De-Esser begins working once the input audio exceeds the value of the Threshold.

• Ratio – Works exactly as the Ratio control in the Compressor (see above) but with a maximum ratio of 8:1.

Equalizer

A Three Band Equalizer (Figure 3-4) allows the overall tonal shape of the audio to be customized on each channel strip.

• Frequency – Sets the center frequency of the Low band (20-320 Hz), Mid band (125-2000 Hz), and High band (1250 – 20000 Hz).

• Gain – Determines the amount of boost or cut applied to each band (-25dB to +15dB).

• Mode – Sets the High band to be either a Peak or Shelf EQ; the Low band will automatically switch between Peak and Shelf depending upon the setting of its Gain control.

The Source Availability checkboxes (Figure 3-5) determine where the sources can be assigned to. For example, if you want the Control Room Mic to appear only on the left-most fader, you would check only the “Channel 1” box.

For sources that are monitor-only, uncheck all Channels to prevent the source from being fed into the final mix and creating an audio feedback loop. Use the “All Channel” checkbox for a quick toggle option to select “All” or “None”.

Channels that are not licensed do appear (in italics) so that if the license for fader count changes, the profile isn't missing the new channels. Otherwise nothing could be applied to the new fader channels.

Figure 3-5 - Source Availability control within the Source profile Fader Mode

The Fader Mode dropdown (Figure 3-6A) defines fader start actions and start logic.

• Normal – The On/Off button is exactly that, and on/off toggle when the button is clicked by the operator.

• Fader Start – Turns the channel on and activates any associated machine logic when the fader is raised from the -∞ position.

Preview Mode

The Preview Mode dropdown (Figure 3-6B) determines whether Preview audio is pre-fader, which acts like a traditional “Cue” circuit, or post-fader, which behaves like a “Solo” feed.

Preview Switching

The Preview Switching check boxes (3-6C) controls the behavior of the channel strip with Preview is engaged. Either option can be turned off, used alone, or used together.

• Channel ON turns Preview OFF – If the Channel is turned OFF and Preview audio is turned ON, the Preview audio turns off when the Channel is turned on.

• Preview ON turns Channel OFF – If the Channel is turned ON and Preview audio is turned OFF, the Channel turns off.

Auto-start Timer

When the Auto-start Timer (3-6D) is enabled, the count-up timer will start when a Channel is turned ON and the counter is set to “Auto” in the Show Profile setting.

Logic Port

When the Logic Port control (3-6E) enables or disables GPIO machine logic and, when enabled, selects its mode of operation.

• Disabled – Turns off GPIO control for this source.

• Exclusive Mode – Permits GPIO control and associates it with a single fader.

• Shared Mode – Allows more than one user to send ON/OFF/START controls via GPIO to the source.

Figure 3-6 - Common properties within Source profiles GPIO Hybrid Control (Phone source type)

Phone source type assumes a need to control the phone interface. Some products offer GPIO control of the call. Momentary closures are available on Pin 4 and 5 (per the Phone source type GPIO logic table). The options are to trigger with ON state or ON state and Preview state. Disabled is the default option.

Feed to source (Phone or Codec source types)

The audio returning to a source (also referred to as backfeed) is a function of some source types. What is that audio is defined by this property and the options are.

• Auto (Program 1 / Phone) - When the channel is OFF, the Phone mix will return. If the channel is ON, the Program 1 mix will return. Refer below for what is the Phone mix.

• Program 1 - All audio in Program 1 minus the source will return

• Program 2 - All audio in Program 2 minus the source will return

• Program 3 - All audio in Program 3 minus the source will return

• Program 4 - All audio in Program 4 minus the source will return

• Phone - All audio in the Phone mix minus the source will return. The Phone mix is anything applied to Program 4 pre ON/OFF and Pre Fader. Effectively any channel with the Program 4 button engaged will feed to the Phone mix.

• Talk only - No audio from any mix. Pressing the talkback button on the source will return the operator audio.

• Auto-Rec (Program 1 / Phone) - The return audio is Program 1 mix unless the record mode is enabled, which the return audio is the Phone mix.

Feed to source dim gain (Phone or Codec source types)

The attenuation applied to the normal return feed when a Talkback is initiated.

Hybrid for Telos phone (Phone source types)

Define if the Phone source is controlling a VX Fixed (Euro mode) or Selectable Hybrids. None is the default mode. If the source is from a VX, define if the source is a Fixed line or Selectable line and which number it is in the VX studio definition.

Page Buttons

There are four Page Buttons at the bottom of the Source Profile page.

• Save as Copy – Duplicates the current configuration and creates a copy on the main Source Profiles page. This is a handy shortcut for creating sources that are identical except for one or two fields – typically the name and Livewire channel number – such as multiple CD players.

• Apply – Immediately applies any changes made.

• 'OK – Immediately applies any changes made then closes the page and returns you to the main Source Profiles page.

• Cancel – Discards any changes made since it was last opened and returns you to the main Source Profiles page.