AIXpressor can optionally be equipped with a Dolby® E decoder and encoder. Please see Adding a Processor for instructions on installing a new processor.
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.
The decoder will decode all standard Dolby E programs.
Important - Unlike legacy Dolby E implementations in which the second program of a 4x2 configuration appeared on outputs 7/8, AIXpressor always shows the audio outputs in ascending order regardless of which program configuration is being decoded.
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).
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.
Note - The Decode page displays considerable additional information A detailed explanation of all metadata parameters is available in the Dolby Metadata Guide.
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.
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.
Important - Any preset saved and stored in the Dolby E decoder will also be available in the Dolby E encoder, providing an easy way to use the same settings and metadata values. Note that you may receive an error message if the metadata in one of the received programs does not match the program type - for instance, if LFE is activated for a 2/0 Channel Mode of a 5.1+2 Program Configuration. The encoder will automatically correct this, and you may dismiss the message.
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).
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.
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.
Note - A detailed explanation of all metadata parameters is available in the Dolby Metadata Guide.
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.
Note - Dolby, Dolby Audio, and the double-D symbol are trademarks of Dolby Laboratories Licensing Corporation. The Dolby E encoder and decoder in AIXpressor are manufactured under license from Dolby Laboratories.
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.
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.
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.
Note - If you choose not to enter a custom name in the Program name field, the program type will be used as the name.
The GUI for the selected processor will be displayed:
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.
Important! As noted in red on the GUI, remember that changing the processor name here also affects the Ember+ processing name and any remote control connections. As well as the appearience in the Routing matrix.
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.
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.
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.
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.
For detailed descriptions of Jünger audio processing parameters, please visit .
For a list of recommended settings forthe Level Magic processor, please visit .
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 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)
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.
Here is an example with an assumed deviation of ~ 12 % of 75 kHz for the extra services:
20*log (75 kHz – 8.8 kHz) / 40 kHz = 4,4 dB
Or -4.6 dBFS
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 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.
It's important to note that very short transients may not be fully mitigated by the Pre-Emphasis Limiter. Nevertheless, this is a fundamental aspect and has no practical significance for FM transmission.
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 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.
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
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 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 Enable - [ON / OFF]
Reference - [-4 ... 4] dBr
MPX Limiter Profile - [Soft / Mid / Hard]
Low-Pass Filter 15 kHz - [ON / OFF]
