Recently, CSR FM have aquired a Soundweb London BLU-100 (which I’ve raved about in the previous post). Amongst the myriad of things which it can do, it can also do some quite complex signal processing. Currently, the CSR FM airchain goes from program output (via a switcher), to a DSPXmini FM processor – which processes the audio and compresses it so that it’s of uniform loudness, then on to a 50W FM transmitter (of which we’re only using 25W).
Unfortunately, the FM processor does not have an audio output on it, only a MPX signal which goes directly to the transmitter (which has fancy things like stereo pilot and RDS encoded into it directly) – as such in order to do our online stream we have to have a FM tuner tuned into the frequency, which then gets distributed to a few places, including a Delta 44 on the encoder machine which then goes out to the online stream.
This isn’t ideal, for a couple of reasons. Firstly, any FM signal gets a bit distorted no matter how close you are to the transmitter (it’s directly below it in the rack, in fact). Secondly, the FM signal adds ‘clipping’, which is essentially cutting the peaks off waveforms flat. You’d normally say this is distortion – and it is! However, over FM, being an analogue medium, this actually sounds good and can add to the loudness of the station if done right.
But when going over a digital medium, it sounds horrible (and sounds like the input on the soundcard has been saturated). It’s especially bad when you encode it into a lossy codec such as MP3 – the square edges of the waveforms have lots of high frequency components (check out this video demonstrating how using the Fourier series a square wave is made up of an infinite number of sine waves of increasing freuquency). An MP3 encoder tries to compress the audio by removing ‘useless’ information, that is frequencies that the human ear is less sensitive to. If you throw in lots of high frequency components, it ends up thinking that these are important frequencies when in fact there is much more valuble information that it can compress in different parts of the spectrum.
So, we can come to the conclusion that using a clipped audio source for lossy digital transmission is not good. This applies to online streaming as well as other digital media like DAB, DVB or XM broadcasting.
Therefore, we need a separate audio processor which does everything that our FM processor does, except without the clipping. If we had a better processor – such as an Orban Optimod – it would have two outputs, one with clipping and one without. Unfortunately we don’t have that much money to spend on processing, especially with the Studio 2 project ongoing. So, therefore I decided that we might try to build our own processor using the BLU-100.
In essence, a broadcast processor has a few general stages which it uses to make the audio output of the station sound uniform and punchy. Namely, these are -
- Wideband processing (gating, AGC, compression)
- Multiband AGC
- ‘Enhancement’
- Multiband limiting/compression
- Final limiting/brickwall limiting
A bit like this:
My mission is to implement this using London Architect. I have pretty much all the components I need in the audio processing section of my BLU-100, so lets give it a shot!
Continue reading “A Walk through a Broadcast Audio Processor”




