This is the preset I’ve created for Omnia 6 EX.  It has a distinctive bass sound that took me quite some time to achieve (btw, if you change any of the parameters, you will probably lose its magic). Dynamically, it has more alive or should I say “organic” sound, compared to the factory presets that keep things more in check. Whether you like that or not, is of course a matter of subjective preferences 😉

Download GTCHRBass preset for Omnia 6 (3730)

Here’s an overview of the  iQ console, a new digital console from Axia.

One of the two iQ prototypes

The Axia iQ is the new, lower cost addition to the Axia’s line-up of networked audio consoles and devices. Like all Axia products, it’s based on Livewire audio-over-IP technology.

If you haven’t heard of Livewire yet, it’s a standard developed by Telos and Axia that allows multiple channels of audio to be carried over a standard Cat5/Cat6 cables. And not only audio, but GPIO and control signals as well. At the heart of the Axia network is a multicast switch, which by design makes every Axia system a router/matrix system. Livewire uses standardized protocols such as RTP and UDP, however the synchronization mechanism that enables syncing of all streams in the network and provides low latency, is proprietary. It is available for licensing and there are a number of manufacturers supporting Livewire and equipping their products with Livewire I/O connectivity. Note: Axia recently released limitless Livewire license (unlimited number of devices) for a $500 fee.

The iQ is the latest addition to the Axia family and is targeted towards smaller radio station and smaller installations, which reflects its low price. Typically, an equivalent Axia Element configuration (with roughly the same number of I/O and faders on the console) would cost twice as much.

The basic iQ system consists of the Main Frame (8 faders and a monitor module) and an iQ core. The core is a similar concept to Axia Powerstation, in a sense that it combines a console power supply, DSP mixing and processing engine, a network switch, GPIO ports as well as digital and analog I/O.

The control surface of the console has a clean and easy-to-understand layout and most users will be able to use it immediately, without too many instructions. The Main Frame has 8 fader channels, each with 100 mm conductive plastic faders, high resolution OLED display, an optical rotary ‘Option’ encoder, 4 program buses, preview and soft buttons and two heavy-duty ON and OFF switches. The rotary encoder is used to select which source to assign to a fader, to adjust gain, equalization, select stereo/mono mode (for stereo sources) and provide pan/balance control.

OLED screens provide information at a glance

All this information, as well as the source name and a small pre-fader level meter, is clearly visible on each fader’s OLED screen. EQ is available on up to 6 channels, but there is no dynamics processing available in the iQ (some things had to be sacrificed for the console’s low price point). However, that is not a serious limitation as you can simply add an outboard voice processor to the system, if you want to.

The monitor module features volume controls for the control room and studio speakers as well as headphones, selection of monitoring sources (two external inputs are supported), preview volume control with an option to route preview to headphones. There are also talk back buttons, a profile recall selection (the iQ can store and recall 4 console “snapshots”) and buttons to select metering and adjust timer/clock. The metering bridge has horizontal OLED meters for monitoring two, three or four program busses and a timer/clock display.

The Main Frame is essential, but there are also additional frames available for users who wish to expand the console with more faders/functionality. The maximum number of frames that an iQ core can support is 3, limiting the maximum number of faders to 24. Expansion frames that are available are: additional 8-faders expansion frame, 6-faders expansion frame that has two sets of 5 film-cap configurable push buttons and 6-faders telco expansion frame with built-in controller/dial-pad for the iQ6 telco gateway. All frames can be placed on the table top, recessed in the table, connected together to form a unified console surface or even mounted in a rack.

Available expansion frames

The heart of the system is the iQ core which is a mixing/processing engine with power supply, multicast-enabled switch and I/O interface – all in a 3U enclosure. This time Axia got the number of inputs and outputs right, as there are more inputs than outputs and a sufficient number of mic and analog inputs. To be precise, iQ core has 4 microphone inputs, 16 analog stereo inputs and 8 analog stereo outputs, 2 AES/EBU inputs and 2 AES/EBU ouputs and 8 GPIO ports – more than enough for most installations. All audio I/O (except microphone inputs) are on RJ45 connectors following the StudioHub+ standard. Additionally there are 6 100Base-T ports to connect other Axia/Livewire devices, and 2 Gigabit ports to connect/daisy-chain studios together.

I/O connectivity of the iQ core is substantial

There is another limitation here, however – the iQ supports 16 network (Livewire) streams (12 sources and 4 destinations or 8 sources and 8 destinations). This is limitation one has to be aware of when building a studio with iQ. For example, adding an iQ6 hybrid will take away 2 of your sources and destinations. A playout system using Axia Audio PC driver might take you anywhere from 1 to 4 or more sources and 1 or 2 destinations. Omnia processors such as Omnia.One will use up one destination as well, etc. Adding an AES/EBU node would eat up all your network sources and destinations, if you would want to use all of its inputs and outputs. Luckily, iQ has quite a sufficient number of inputs (especially analog) that a full-size node will rarely be needed. If you need network sources than iQ can provide, that probably means the Element console would be more suited for the size of your installation, which has no limitations of this kind.

The iQ also packs 8 GPIO ports for remote control of devices and red lights, 6 Ethernet ports (4 of which are auto-sensing PoE) and 2 Gigabit SFP ports, so you can use either copper or fiber cables. It’s worth mentioning that like most recent Telos, Axia and Omnia equipment, iQ core is convection cooled and you can put it any studio as it produces no noise at all. An optional backup power supply (2U high) is available, if you want to add power supply redundancy.

Configuration of the iQ system is as easy as connecting a PC to the network, opening a browser and entering the IP address of the iQ core. Everything is done through the web interface – configuring the console, assigning inputs and outputs, defining source profiles, saving show profiles (snapshots)… For Element users, it should be pointed out that iQ has only one virtual mixer and one virtual mode circuit, but also that there is now a check box to mix preview with CR monitoring. You don’t have to use a virtual mixer to achieve this functionality anymore, which is a nice addition.

Some more screen shots of the web configuration options:

As an example, a simplified studio layout based on Axia iQ could look like this:

Example configuration of the studio based on Axia iQ

The 6-fader telco expansion frame is added to the Main frame, for the total of 14 faders. The integrated call controller makes it easy for an operator to talk to the callers off and on-air and also read the status of the 6 phone lines (POTS or ISDN) that can be connected to the iQ6 telco gateway. The iQ6 has two digital hybrids under the hood, both of which are using the advanced 3rd generation Telos hybrid technology with Digital Dynamic EQ and dynamics processing by Omnia, for consistent audio from call to call. It also features AEC (Advanced Echo Cancellation) from Fraunhofer Labs to reduce feedback and echo, especially with mobile and VoIP calls.

iQ6 dual digital hybrid with 6 POTS or ISDN phone lines

Additionally, a VSet12 can be connected to the system to provide a physical phone and hybrid controller in one device. VSet12 has a large LCD color screen to display phone line status, caller information, fader assignment, access built-in address book or call logs etc. A VX Producer software is also available for call screening purposes, but it can also act as a phone as well – just connect the headset to the PC running VX Producer and you can not only talk to callers, but record it and later edit it with built-in audio editor and send to studio for playback. A very nice feature, great for news desks!

On the other side, there’s a PC with playout software and Axia Audio PC driver, delivering digital audio straight into the Axia network. The same PC can receive streams from the iQ, such as EXT1 for logging purposes or PGM4/Record bus for recording voice tracks and other bits. Finally, an Omnia One processor running FM, AM or Multicast software feeds the transmission system for broadcast.

As with all Axia and Livewire installations, the cabling is reduced to minimum as one Cat5 cable will carry many channels of digital audio and control signals, routing is as easy as clicking a mouse in a browser window, mix minuses are automatically generated and with Axia Audio drivers for PC, there’s no need to use sound cards anymore. On top of all that, all Axia devices carry a 5 year limited warranty.

The iQ is a very nice, compact and modern audio-over-IP broadcast console. Perfect for smaller radio stations and other installations, it has substantial I/O capability, enough faders even in the basic configuration and the flexibility and advantages of Axia’s Livewire technology. A basic Main frame + iQ core costs only $7,990, while our example configuration together with iQ6 hybrid and Omnia One processor will cost you $15,275 (prices valid on July, 2010.).

Goran Tomas


An overview of my work of transforming the DSPXmini-FM processor to the DSPXmini-FM SE (Second Edition).

DSPXmini-FM SEDSPXmini-FM SE processor

Anybody who has some experience in processing, knows it’s VERY subjective. People hear differently, are sensitive to different things, prefer different “flavors” and textures of the sound, etc. Describing processing and the sound becomes quite difficult as well. Nevertheless, using some short audio clips I’m going to attempt to illustrate the sound and the thought process behind re-designing the original DSPXmini processor. As they are only short clips, an untrained ear may not perceive as significant difference as you would, had you had an opportunity to listen to these processors and effects for longer time. Keep that in mind while listening.

As with all re-designs, things usually start with the discontent of some aspects of the existing product and an idea that things could be done better… Having spent enough hours carefully listening to the processor, you get to know its character (almost like you would a person). You get to the point where you can anticipate how it will behave in certain situations, what reaction you will you get for certain input. You get to know its good sides and its flaws. And if these flaws are obvious, displayed too frequently and cannot be tamed down sufficiently with available parameters adjustments, this is where a designer starts to consider if things can me improved.

The original DSPXmini had several things that needed to be improved in my opinion. Let me try to illustrate them to you:

Note: It’s best to use headphones (or a hi-fi system) to listen to these clips. The processor was set to 50µs pre-emphasis and the ‘AC’ preset.

  1. The overall frequency balance that the processor always tried to created was too bright. Inherently in the design of the processor, there was a strong tendency to shift emphasis to high frequencies. To the point of the sound becoming too silky and strident:

    Note: After you press play, it may take some time for video to load.

    [flowplayer src=’http://www.gorantomas.com/wp-content/uploads/2010/07/FreqBal1.mp4′ width=368 height=272 splash=‘wp-content/uploads/2010/07/vs.jpg’]
    [flowplayer src=’http://www.gorantomas.com/wp-content/uploads/2010/07/FreqBal2.mp4′ width=368 height=272 splash=‘wp-content/uploads/2010/07/vs.jpg’]

  2. There was an irritating “hole-punching” effect in the high-frequencies, where the processor would distort and duck the high-end on HF transients such as the words starting with an ‘s’ and other sibilants:
    [flowplayer src=’http://www.gorantomas.com/wp-content/uploads/2010/07/HolePunching.mp4′ width=368 height=272 splash=‘wp-content/uploads/2010/07/vs.jpg’]

  3. The overall sound was too busy, too mashed in. More often than not, you could hear the layer of processing rather than being able to listen to the music and enjoy it. Additionally, the processor was breathing too much, slowly changing the sound and the frequency balance unnaturally during the song:
    [flowplayer src=’http://www.gorantomas.com/wp-content/uploads/2010/07/Unstable.mp4′ width=368 height=272 splash=‘wp-content/uploads/2010/07/vs.jpg’]

  4. The low-end texture was too tight and constrained, rather than being open, big and adding power to the sound.

By listening to the processor, observing the meters and noticing what effect the adjustments of the parameters had on the sound, it was obvious to me that the major source of the problems of ducking, changing frequency balance and busy and constrained sound was with the 4-band limiter. After looking at the algorithm structure and trying to modify it to correct these problems, it became clear that a completely different approach was needed. This was when I embarked on the development of the limiter algorithm from the scratch. Most of these ideas came from my home-brew processing experiments in analog. I also wasn’t going to satisfy on the first idea I had, but on the one that sounded the best. That took time, but in the end, the new limiter algorithm with smart program-dependent attack, release and gating circuits and a few tricks to handle the pre-emphasis more naturally, was born.

ToDo listSneak peak on the excerpt from my ToDo list during development

However it did not solve all the problems. Mainly, there was still a lack of bass punch. And in my book, a solid, powerful low-end is the foundation of a good sound on-air! I felt the new limiter could handle much more than it was being taxed with, so I ended up making tweaks to the AGC in order to change the energy balance handled by the AGC vs the limiter. This produced some results, but still not enough, so another major re-design was in order – the crossover. Without going into the complexities of crossovers, the new design really helped the limiter bring out some of the bass thump and low-end up in the overall sound. Finally, a distortion-controlled clipper received some tweaks in order to reduce the aliasing distortion and the transformation was finished with several more weeks of tweaking and fine-tuning of all the processing stages so that they would work synergistically and smoothly with each other.

And here are the results (a sort of before and after comparison):

Note: It’s best to use headphones (or a hi-fi system) to listen to these clips. Both processor were recorded in the same environment, with the same input and output settings, 50µs pre-emphasis and running the ‘AC’ preset.

  1. Frequency balance:
    [flowplayer src=’http://www.gorantomas.com/wp-content/uploads/2010/07/VS-FreqBal.mp4′ width=368 height=272 splash=‘wp-content/uploads/2010/07/vs.jpg’]
  2. Stability of the sound:
    [flowplayer src=’http://www.gorantomas.com/wp-content/uploads/2010/07/VS-HolePunching.mp4′ width=368 height=272 splash=‘wp-content/uploads/2010/07/vs.jpg’]

  3. Low end texture and bass thump:
    [flowplayer src=’http://www.gorantomas.com/wp-content/uploads/2010/07/VS-LowEnd.mp4′ width=368 height=272 splash=‘wp-content/uploads/2010/07/vs.jpg’]

For better comparison, here are two longer clips with mixed music of both the original DSPXmini-FM and the DSPXmini-FM SE:

Note: You can switch between the clips by pressing the Play buttons (only one clip will play at a time). You can also position the playback bar where you want it to start playing so you can easily compare processors at different parts of the music mix. These players will play 128 kbps MP3 encoded audio, but you can download the original WAV files by clicking on the Download (down arrow) buttons.

Source audio

Original DSPXmini-FM (AC preset)

DSPXmini-FM SE (AC preset)

The mini SE is not without flaws… One thing that I wanted to do is tackle the AGC which could be better and more effective, and the overall cleanliness could be improved as the processor will not exactly “burn the dial” with loudness and punch… Personally, I always want to improve more and I’m rarely completely satisfied with something. But then you have to consider the fact that this processor costs only EUR 1200/USD 1500 (at the moment of this writing) and there’s only so much development a company can and wants to afford, with a profit from the sales at that price point.

However, I believe the mini SE really matured from the original processor and improved on most of its previous weaknesses. Aside from comparing the processor with its competitors, the way I judge whether the processor is good, is to see if I can listen and forget about the processor and just enjoy the music… Whatever the processor does it has to be natural, in harmony with the audio and it must not detract my attention by getting in the way and doing something unnatural. And I believe I was able to get very close to that important goal with the DSPXmini-FM SE.

The DSPXmini-FM SE was presented at the IBC 2008.
Visitors could compare it to the original DSPXmini-FM and other processors.

There was quite a bit of skepticism about what will come out of my design. But in the end, pretty much everybody liked the improvements and the new sound of what became the Second Edition of the DSPXmini-FM and the reactions from the customers proved it as well.

Goran Tomas


This is the paper I presented in Graz, at the 3rd Congress of the Alps Adria Acoustics Association in 2007. It answers the question of what happens to the audio quality as you push the audio processing to the maximum, while you are broadcasting digitally (meaning there’s some kind of perceptual coding after the audio processor).


Abstract: Radio broadcasters have been pushing the loudness of FM processing for years in effort to attract more listeners, despite all accompanying negative effects on audio quality. It is likely to expect that such practice will continue with digital broadcasting as well. Our goal was to determine how listeners perceive varying degrees of aggressiveness of dynamics processing for digital broadcasting, and in particular how aggressive dynamics processing influences quality of perceptually coded audio. A listening test was conducted with eight samples of varied aggressiveness of dynamics processing, before and after encoding with HE-AAC v1 at 48 kbps stereo.

Key words: dynamics processing, digital audio broadcasting, audio quality, listening test, perceptual coding


My article for Radio Guide in Jan 2007. explains why you need audio processor specifically designed for digital audio broadcasting, and what are the different requirements between analog and digital transmissions in regards to audio processing.

Abstract: With IBOC (aka HD Radio) gaining momentum, many stations are deciding to include a second channel (HD2) and even third (HD3) as an alternative and/or additional programming to their main channel. Many broadcasters are also looking at streaming to the web as royalty fees are now clearly regulated and more people are using on-line multimedia services than ever before.
When it comes to processing for these new services – especially if they have different programming than the main channel – it is tempting to use your old FM processor that has been sitting as a backup. But that would be a mistake. In this article we will try to explain why and what are the requirements and the differences in processing for analog FM and digital channels.

Key words: digital broadcasting, HD Radio, IBOC, digital audio processing, dynamics processing, clipping, look-ahead limiting, pre-emphasis, codecs, perceptual coding