Change is Good

From the very beginning, the developers of the Omnia.11 were determined to create a processor which would stand head and shoulders above the legendary Omnia 06. 

So, Team Omnia started with a fresh canvas. 

Everything from the firmware platform, the GUI, to every algorithm was re-thought, and in most cases completely re-engineered or re-designed.  The result is unlike any existing product of its kind to date.   Omnia.11 has literally defined the new state-of-the-art for FM audio processing, as evidenced by a trifecta of awards from the three leading industry publications:  2010 Cool Stuff Award from Radio World Magazine; 2010 Pick Hit from Radio Magazine; 2010 TechInk Award for Innovation by Radio Ink Magazine.   

For the first time, listeners will hear the programming, not the processor.   Effortlessly loud.   Thunderous bottom end, sparkling highs, and crisp, clear voice reproduction.  All with that trademark punch and clarity which made Omnia the required audio processor of the highest-rated radio stations in the world.    

So dramatic is the advancement of Omnia.11, that initial tests have revealed that many early adopters were genuinely startled by the lack of a traditional “processor sound” when the unit was first deployed.  The low level distortion and artifacts, long accepted as part of the fundamentals of processing, are now almost completely gone and certainly not perceptible to the ear.

The firmware in Omnia.11 takes advantage of software capabilities never before possible.  The results are dynamics algorithms that were once only a dream of the processing enthusiast.   AGCs, Compressors, and Limiters analyze music in real time and adjust internal parameters for optimum performance across a broad range of material.

Available in two models:

Omnia FMHD: Featuring dual, independently controlled processing paths for FM and HD/DRM

Omnia FM: Without HD/DRM capability, but can be upgraded to FMHD at a later date.


Chameleon Processing Technology

A major part of this technology, the new Density Detector, enables Omnia.11 to properly handle hyper compressed content. The AGC system cannot be fooled due to heavy density, or by older source material which contains high peak-to-average levels. The density-detector keeps Omnia.11 operating on-target, at all times.


Ultra- Multiband Limiter System

Traditional limiting technology has often resulted in various forms of audio corruption.  Omnia.11’s new LoIMD technology coupled with smart gain reduction algorithms, now have limiters which sound amazingly transparent.

All AGC and limiting algorithms employ an auto acceleration/deceleration mechanism, which tunes out perceptible intermodulation distortion. The attack/release functions adjust themselves based upon content density. This breakthrough method literally analyzes the audio content in both the amplitude and frequency domain, then adapts the timing networks - on the fly - to transparently control the signal, without the control being heard. The result is revealed in added detail, clarity, and quality, yet maintaining the desired competitive loudness level.

Special attention was paid to the behavior of live voice quality.  The improved performance of the AGC and limiter functions generate live voice clarity and impact far beyond that which was previously possible.


Bass Management

The bass enhancement algorithm is a key feature of the Omnia.11.  Low end is now broadcast with recording studio-like punch and impact, with no traditional side-effects whatsoever.  

Omnia.11’s exclusive bass-management method is a mixture of innovation, as well as a rearrangement of the system topology.  Achieving great sounding bass requires the most effort, partly due to the fact that the bass spectrum has the most number of harmonics, and all of these must be kept properly accounted for in the time domain. Also, any additional spectra created (enhancement) must have its harmonic content managed, or the bass region begins to sound distorted and unnatural.  This process requires much more than just traditional EQ, bass clipping/filtering, or any ordinary attempt at bass enhancement. Even the location where the function is inserted matters, as well as how it maintains its frequency range along with the rest of the system. An entire dissertation could be done on the bass enhancement/management system alone. The classic Omnia dynamically flat & time aligned crossover system has been further refined to produce smooth, rich, and full tonality. The AGC and limiter sections cannot be fooled into false gain control due to spectral density (or lack thereof) from the crossover network.


New Ultra LoIMD Distortion Controlled Clipper System

Audio processing for conventional broadcast (FM and AM) has, in some applications, reached extreme levels. Various methods are available today capable of creating LOUD competitive signals, but at the expense of perceptible quality. Through critical listening, extensive research, and evaluation of processing methods, it has been determined the single most annoying quotient is due to intermodulation distortion (IMD) induced by aggressive functions within the processing system. The algorithms are pushed to the limits, and beyond. One of the most crucial, aggressively used algorithms in the FM processor is the pre-emphasized final limiter/clipper. Omnia Engineering has developed the new Ultra LoIMD Distortion controlled clipper system specifically to reduce IMD in this critical stage of the processing.  An explanation of the new Ultra LoIMD clipper system follows shortly.

For those who feel the need to use it, there’s also a composite clipper embedded in the stereo generator.  However, to date, all of our testing has been done without any composite clipping. Pilot protection is on the order of magnitude close to 90 dB,  which is considerably more protection than necessary for even the best FM receiver.   Integrated laboratory-grade stereo generator with dual MPX outputs, 19 kHz reference output for external RDS/RBDS systems and pilot protection that provides >80 dB pilot protection - with or without composite clipping. MPX spectral low-pass filter to protect RDS/RBDS and SCA signals if composite clipping is employed.  Multiple ways to adjust the system to achieve the exact sound you’re looking for. An installation wizard will guide anyone through a simple step-by-step setup to on-air operation. Using the answers to a series of simple questions, Omnia.11 adapts itself, based upon the answers, to craft a preset which delivers the desired end result quickly for an effortless out of the box experience.


In Addition

A front panel touch screen GUI, on a 10.5" diagonal screen, provides ease of use and enhanced metering and diagnostics. Remote access is via any web browser, as well as a local onboard WI-FI connection.  Laptops to iPads will have access.

Livewire+, AES/EBU digital and analog I/O is standard. Headphone soft "patch points" are available for listening through the processing chain.


Fanless cooling design built into a rugged 4 RU chassis.


Ultra LoIMD Clipper System Explained

Audio processing for conventional broadcast (FM and AM) has reached extreme levels. Various methods are available today capable of creating LOUD competitive signals, but at the expense of perceptible quality. What causes this, and what can be done to retain audio integrity in the face of competitive pressure?

Adding to the dilemma is the music industry adopting the same traditional radio mind set.  Recordings which are so heavily processed, that they sound as if they’re on-the-air, before being on-the-air.  To say we live in a dynamically processed world is an understatement.

Being loud is not the problem.  The problem is the unfriendly annoying artifacts generated by the common processing practices used by broadcasters and the music industry.  The combination of hyper-compressed content and aggressive on-air processing results in not only audio which lacks definition and quality, but audio which contains annoying side effects.   The challenge for Team Omnia was to retain quality and definition, yet retain the competitive loudness level that many broadcasters demand.

Modern music mastering practices generate content that is noticeably rich in deep bass, presence, and treble. When processed aggressively, especially for FM-Stereo, the resulting audio appears synthetic in tonality and quality.  Treble frequencies appear overly bright, and sometimes harsh, even with additional application of dynamic high frequency processing.  Bass sounds tight and defined, but depending upon the processor’s spectral limiting system, it can also sound distorted.

Listening to current music, with aggressive processing, produces a distinct annoyance: the appearance of a sizzling or frying sound to midrange, presence, and treble spectra. This was noticeable on all the processors used for evaluation. Reducing the final limiting, or clipping, helped ease the pain. This indicated that the problem is harmonic, related to the clipping process. Significant reduction of clipping removed the annoyance, but the loss of loudness was on the order of 6 dB or more. Not suitable for the needs of competitive audio.

Was this the common issue of too much bass forcing the rest of the spectrum into the limiter? This is known as bass induced intermod.  At first, it would seem so, but the test segments did not have any bass content, and the frying was still present. Was it in the original source, and the processor was magnifying it via multiband dynamics control?   Careful evaluation of the source audio revealed the answer to be no.  Something else, apparently.

By example, this is easily heard in the song "Because Of You" by Kelly Clarkson. The opening of the selection is a piano solo, and the vocalist begins to hum along with the piano, a few seconds later. Present day audio processors, set up aggressively, cause the humming in the vocal to sound as if bacon is being fried.  This was a high rotation song on CHR (Contemporary Hit Radio) formatted radio stations.  Since most of those require aggressive processing, this test case replicates the real world. This example is just one of many which illustrate the challenges in current processing technique.

Since the aforementioned bacon frying annoyance was affected by the action of the clipping function,  a probe into that algorithm was in order. Most final limiting/clipping systems in modern audio processors employ some form of proprietary means to control perceived distortion. The methods for these vary. While open for subjective discussion, the end result is still the same: absolute peak control is performed and a minimal level of harmonic distortion is acceptable within a specified operating range. Basically, some form of masking method is used to hide the most annoying clipping side effects from the ear. Although, it appears now, we’ve pushed these methods to the point where modern recordings generate distortion annoyances when aggressive processing is used.

For the processing novice, a clipper - by design - will generate harmonics of the fundamental audio frequency. Using a sine wave for an example, if the upper and lower peaks of the waveform are chopped off (clipped), harmonics are born out of the clippings and show up within the spectrum space as harmonic multiples of the original frequency. An example frequency of 1 kHz, with 3 dB of clipping generates odd-order harmonics at 3kHz, 5 kHz, 7 kHz, etc, out to infinity. Figures 1 - 2 illustrate examples of this.


Figure-1, 1kHz Square Wave

Figure-2, 1kHz Square Wave Spectra


Note: Broadcast audio processors band-limit frequency spectra within a specified range, for their respective transmission paths.FM-Stereo bandwidth is 15 kHz, and AM is between 4.5 kHz and 10 kHz, depending on location. As such, clipping harmonics are limited via non-overshooting filtering methods in order to properly maintain operating legal bandwidth.

The challenging problems stated here are not based upon clipping functions of singular frequencies. Modern clipping methods, with distortion management, reduce clipping side effects over a preset range and only up to a specified level. It appears modern recording techniques either overload the present distortion mechanisms, or they cannot process this content aggressively without generating this frying/sizzling-like distortion. Since this problem exhibits itself with full range linear recordings, data reduced content (mp3 audio files) is even more distorted.


What Happens When Additional Spectra Is Added

When additional audio is added to a fundamental, things get complicated.   Sum and difference frequencies are created along with another component known as intermodulation, or IMD for short. Simply stated, this is where one signal will ride alongside, on top of, or modulate another. Sometimes this is done for specific effect. Music synthesizers use various intermodulating functions to create desired sounds.

In an audio processor, the dynamic action of compressors and limiters are examples of modulators, as they generate a level controlling signal to change the gain of the audio. The level controlling signal and audio is routed to a multiplier function, and the audio is multiplied by the controlling signal. Through this action, the level is dynamically adjusted. This is an example of intermodulation, as the audio is modulated by the control function. When the control signal starts to operate too fast, it generates a controlling rate with an additional frequency of its own. This operating frequency will possess additional harmonics and those get factored (multiplied) into the audio during the multiplication stage. The resultant contains the level adjusted audio along with harmonics from the controlling signal that were intermodulated into the final product. This is what happens when the control signal operates in an overly aggressive manner: the sonic quality becomes fuzzy, dull, and lifeless. We refer to this as dynamic intermodulation distortion.

With the above example in mind, let’s consider what happens within a clipper, when multiple audio signals are present and clipping is applied. A clipper, in reality, is a zero-attack/zero-release time limiter operating with a ratio of infinity-to-one. When multiple frequencies are present and clipping is active, the lower fundamental frequency will push the higher fundamental frequency into, and out of, the clipper at the rate of the lower frequency. This is known as clipper induced IMD.  An easy example of this would be music with deep defined bass and a solo guitar or vocal. When clipping is active, the guitar or vocal will warble at the rate of the bass frequency due to the action of the bass signal pushing the guitar/vocal signal in and out of the clipper. Some audio processors employ bass processing techniques to reduce - and in some cases - remove this annoyance. On account of this, IMD components are amplified in level and spectra. Even modern distortion canceling clippers (or whatever other marketing name is given to them) generate IMD.

Up until now,  it has been an accepted notion that clipper induced IMD was a by-product of deep bass and enhanced midrange/presence/treble content. When studying the example of the Kelly Clarkson track, it became evident the problem was related to clipper induced IMD, except the example does not possess any bass spectrum of any significance.


Figure-3, Kelly Clarkson Segment


Notice inFigure-3, a segment taken from the Kelly Clarkson track, the dominance of signal centered at 500 Hz, and the range between 10 kHz - 15 kHz.  Witness what happens if some IMD tests were run on present clipping systems.


Under The Microscope

Performing an IMD test on a clipping system is quite easy. Two audio frequencies are mixed together, then passed through the system under test and the output is observed on a scope and spectrum analyzer. In this instance, the clipping systems all employed the required 15 kHz low pass filtering and zero-overshoot control mechanisms found in broadcast processors.

For the test, 100 Hz was inserted at a level, which generated 3 dB of clipping. A high frequency component was mixed in at the same level and 75 s pre-emphasis was applied. The tests were run over the range of5 kHz up through 15 kHz, while 100 Hz was used as a constant low frequency source. Figures 4 - 8 are the results of the tests.



Figure-4, Clipper Induced IMD: 100Hz & 5kHz

Figure-5, Clipper Induced IMD: 100Hz & 7.5kHz

Figure-6, Clipper Induced IMD: 100Hz & 10kHz

Figure-7, Clipper Induced IMD: 100Hz & 12.5kHz

Figure-8, Clipper Induced IMD: 100Hz & 15kHz


Notice as the upper frequency is increased; there is significant difference spectra that falls between the two fundamentals. This is extremely severe at 10 kHz, 12 kHz, and 15 kHz. If you recall the music example, this is very close to the spectral illustration in the Kelly Clarkson track.

It is clipper induced IMD.


Clipper Systems, Distortion Cancelling, And Multiple Bands.

As stated, all present day clipping systems employ methods to control distortion. Of interest is that each of these use a static method to mask harmonic distortion when clipping is active. As the Kelly Clarkson example clearly illustrates, harmonic distortion is not the concern as it once was. Intermodulation, due to added presence and high frequency spectra, has overtaken the problem that once was dominated by harmonic distortion. Suffice it to say, all clipping methods must employ some form of harmonic distortion control, or they will not operate sufficiently enough to generate competitive sounding on-air audio. Modern content now requires additional processing means to reduce induced IMD.

Suppressing IMD is significantly more difficult, as the constantly different frequency components are a non-stop moving target. Whereas suppressing harmonic distortion can easily be predicted and controlled through a static filtering system.

Proof of this is demonstrated with an evaluation of present day distortion canceling systems. All of them employ static filtering to mask distortion components. They vary in range from broadband to 5-6 band, or more. All of these fail with aggressive processing. The broadband method suppresses harmonics and some IMD at specific frequencies. The multiband methods are designed to insert gentle low pass filters after multiband clippers in each audio band. This works over a narrow range, but falls apart with aggressive levels of clipping. Multiband clipper/filtering is done in parallel architecture and each singular band clipper is not able to understand what the others are doing. Therefore, the resulting filtered harmonics of each band interact in unpredictable ways - some of which exaggerate IMD.   Adding more bands or steeper filters does not improve or fix the problem.



The answer lies not in additional bands, but in an understanding of the range of frequencies that generate both harmonic and intermodulation distortion, then applying various masking means to suppress both simultaneously as they are generated.   It’s a combination of breaking down the audio spectrum by octaves and interaction with the Gibbs Phenomenon.  Suffice it to say, the prior statement - along with technology - enables a clipping system in Omnia.11 which suppresses BOTH harmonic and IMD distortion components when aggressive processing levels are required.  Additionally, and more importantly, this new clipping method does not employ the use of dynamic compressors or limiters to control depth of clipping in order to minimize clipping induced IMD.   There have been, and remain, a few proponents who utilize this method to reduce generated IMD, but it is at the expense of added dynamic intermod which manifests itself as audio pumping and hole punching.


See For Yourself

Running the same IMD tests, as mentioned earlier, now offer the following results. Compare figures 9 - 13 to those of figures 4 - 8 of both the old and new methods.


Figure-9, Clipper Induced IMD: 100Hz & 5kHz

Figure-10, Clipper Induced IMD: 100Hz & 7.5kHz

Figure-11, Clipper Induced IMD: 100Hz & 10kHz

Figure-12, Clipper Induced IMD: 100Hz & 12.5kHz

Figure-13, Clipper Induced IMD: 100Hz & 15kHz 


It is easy to see. For the exact same amount of clipping employed, midrange, presence, and treble IMD is gone.  With the new Omnia.11 method,  Kelly Clarkson’s test segment does not possess any of the bacon frying sizzle annoyance as heard prior with all other clipping systems. As a matter of subjective observance, all audio auditioned through this new method sounds cleaner for the same given level of loudness. It does not matter if the content source contains deep-rich bass or not: the audio signal is subjectively cleaner for the same level of loudness.

Multiband clipping does not take into consideration any interactivity of outlaying spectra. That’s where the method eventually fails. The proof is in the audio performance with critical content.

So, in the end, with Omna.11, everything you hear is true.

Prove it yourself, with your own critical content.



  • Non-linear Crosstalk: > -80 dB, main to sub or sub to main channel (referenced to 100% modulation).

  • 38 kHz Suppression: > 70 dB (referenced to 100% modulation).

  • 76 kHz Suppression: > 80 dB (referenced to 100% modulation).

  • Pilot Protection: > -65 dB relative to 9% pilot injection, ± 1 kHz.

  • 57 kHz (RDS/RBDS) Protection: better than -50 dB.

  • Connectors: Two EMI suppressed female BNC, floating over chassis ground Maximum Load Capacitance: 5nF (at 10 ohms source impedance).

  • Maximum cable length: 100 feet/30 meters RG-58A/U.


Analog Audio Input

  • Left/Right Stereo. Electronically balanced.

  • Input impedance 10k ohms resistive.

  • Maximum Input Level: +22 dBu.

  • Nominal Input Level: +4dBu, which nets a -18dBFS input meterreading on a steady-state signal when the Input Gain controlis set to 0.0dB. Program material with a nominal average level(VU reading) of +4dBu will typically produce peak readings on the input meter in the range of -12 dBFS to -6dBFS. This is thecorrect operating level.


A/D Conversion

  • Crystal Semiconductor CS5361, 24 bit 128x over-sampled deltasigma converter with linear-phase antialiasing filter. Pre-ADCanti-alias filter, with high-pass filter at <10 Hz.

  • Connectors: Two, EMI-suppressed XLR-female. Pin 1 chassis ground, Pin 2 “Hot”.


Analog Audio Output

  • Left/Right Stereo. Electronically balanced.

  • Output Impedance 20 ohms.

  • Minimum load Impedance: 600 ohms.

  • Output Level adjustable from -2 dBu to +22dBu peak in 0.1dB steps.


D/A Conversion

  • Crystal Semiconductor CS4391, 24 bit, 128x oversampled.

  • Connectors: Two, EMI-suppressed XLR-male. Pin 1 chassis ground, Pin 2 “Hot”.


Frequency Response

  • Complies with the standard 50 or 75 microsecond preemphasis curve within ± 0.5 dB, 30 Hz to 15 kHz. The analog left/right output and AES/EBU Digital outputs can be configured forflat or pre-emphasized output.


System Distortion

  • Less than 0.01% THD, 20 Hz – 7.5 kHz. Second harmonic distortionabove 7.5 kHz is not audible in the FM system.

  • Signal-Noise Ratio: > -80 dB de-emphasized, 20 Hz – -15 kHzbandwidth, referenced to 100% modulation.

  • The measured noise floor will depend upon the settings of the Input and Output Gain controls and is primarily governed by dynamic range of the Crystal Semiconductor CS5361 A/D Converter which is specified as >110 dB. The dynamic range of the internal digital signal processing chain is >144 dB.


Stereo Separation

  • Greater than 65 dB, 20 Hz – -15 kHz; 70 dB typical.



  • > -70 dB, 20 Hz -- 15 kHz.


System Latency

  • 36ms. “FM” channel, as measured from the analog inputs through the composite MPX output.


Composite Outputs

  • Source Impedance: 5 ohms or 75 ohms, jumper-selectable. Single ended and floating over chassis ground. Output Level: 0V to 10V in 0.05V steps, software adjustable.


D/A Conversion

  • Texas Instruments/Burr Brown PCM1798, 24-bit sigma-delta converter.



  • Two electrically independent outputs. Software based level adjustment.


Load Impedance

  • 50 ohms or greater load is suggested.


Pilot Level

  • Adjustable from 4.0% to 12.0% in 0.1% steps and OFF.


Pilot Stability

  • 19 kHz, ± 0.5 Hz.


Signal-to-Noise Ratio

  • -85 dB typical, 75 _S de-emphasized, 15 kHz bandwidth, referenced to 100% modulation).



  • < 0.02% THD 20 Hz – 15 kHz bandwidth, 75 _S de-emphasized, referenced to 100% modulation.

  • Stereo Separation: > 65 dB, 30 Hz – 15 kHz.

  • Linear Crosstalk: > -80 dB, main to sub or sub to main channel, referenced to 100% modulation.



  • XLR-female, EMI-suppressed. Pin 1 chassis ground, Pin 2-3 transformer isolated, balanced, and floating. Standard AES3 specified balanced 110 ohm input impedance.


External Sync Range:

  • Automatically accepts sample rates between 32kHz and 96kHz. Connector: XLR-female, EMIsuppressed. Pin 1 chassis ground, pins 2 and 3 transformer isolated, balanced, and floating – AES3 standard 110 ohm impedance.


Remote Control:

  • Via Ethernet using built-in Java (TM) based remote control program integrated into web page interface. All software is served from the built-in web server to any standard web browser; there is nothing to install on the user’s computer.



  • Ethernet - Industry standard EMI-suppressed RJ-45 connector.


GPI Interface:

  • Connector: EMI suppressed DB-15 female connector.


Power Requirements:

  • Voltage: 100-250 VAC, 47-63 Hz.


Power Connector:

  • EMI suppressed IEC male. Detachable 3-wire power cords supplied for US and European use.


Power Supply:

  • Internal. Overvoltage and short circuit protected.


Digital Audio Input:

  • Configuration: Stereo per AES/EBU standard, CS8420 Digital Audio Transceiver with 24 bit resolution, software selection of stereo, mono from left, mono from right or mono from sum.

  • Automatically accepts and locks to input sample rates between 30 and 108 kHz.

  • Connector: XLR-female, EMI-suppressed. Pin 1 chassis ground, pins 2 and 3 transformer isolated, balanced, and floating – AES3 standard 110 ohm impedance.


Digital Audio Output #1:

  • Stereo per AES3 standard. Output can be configured in software for flat or pre-emphasized response at 50 or 75 microseconds.

  • Digital Sample Rates: Output sample rates software selectable for 48kHz, Sync to Input or Sync to External.

  • Connector: XLR-male, EMI-suppressed. Pin 1 chassis ground, pins 2 and 3 transformer isolated, balanced, and floating. Standard AES3 specified 110 ohm source impedance.

  • Digital Output Level: -22.0 to 0.0 dBFS software adjustable.


Digital Audio Output #2:

  • Stereo per AES3 standard. Output can be configured in software for flat pre-emphasized response at 50 or 75 microseconds.

  • Digital Sample Rates: Output sample rates software selectable for 48kHz, 44.1kHz or Sync to External.

  • Connector: XLR-male, EMI-suppressed. Pin 1 chassis ground, pins 2 and 3 transformer isolated, balanced, and floating. Standard AES3 specified 110 ohm source impedance.

  • Digital Output Level: -22.0 to 0.0 dBFS software adjustable.


External Sync Input:

  • External Sync: Output sample rate can be synchronized to the signal present on the AES/EBU input, or to an AES3 signal applied to the Ext. Sync input connector. (Does not accept Word Clock Inputs)



  • Click here to view the current regulatory compliance.