RadioBOSS and Vortexbox and MP3 normalization

Hello,

We are a community radio station in Australia and we are evaluating RadioBOSS prior to purchasing it.  We are down to the last few features that we need to sort out prior to going on air.

Our music library and program material is stored on a gadget called a Vortexbox (http://vortexbox.org) which is connected to our 1Gb LAN.  We use the Vortexbox for several reasons:
1. It can be shared with the RadioBOSS box and several other work stations
2. It is fast and reliable and configured with a software RAID-1 array and an external backup HDD
3. It has heaps of HDD storage space - we are using 2TB HDDs
4. It is a standalone device and allows ripping of CDs to FLAC and MP3 in various configurations
5. Ripped CDs are stored as FLAC files and can be re-transcoded to MP3 and the settings changed at any time (example: bitrate changed from 320Kbps to 256Kbps and so on)


We have done some tests with RadioBOSS and played standard MP3 files and MP3 files that have been normalized on the Vortexbox.  Audio levels have been much more consistent for the normalized files and this is exactly what we want because we are running an automated station.  By comparison the MP3 files that have not been normalized can vary considerably from album to album and this variation in level is annoying to the listener.

The above tests regarding normalization were done with the RadioBOSS compressor turned off so that listening tests were easier to do.  Ultimately we want to have our MP3 files normalized AND run the RadioBOSS compressor to keep transmitted audio levels up.

Now the questions:

1. Is it necessary for us to normalize all our MP3 files or does RadioBOSS have a way of reading the MP3 tags and adjusting the playback accordingly
2. What is the reference level for normalization that we should use ?  We have researched this and come up with 96dB.  This has given good overall results.


Thanks in advance ...
 
May I refer you to a discussion point regarding the relative merits of normalisation.

http://mp3gain.sourceforge.net/faq.php

When all said and done, I believe that a terrestrial broadcasting station needs a hardware processor of some sort, as in New Zealand, over-deviation is an offence punishable by a fine if not handled correctly.

What are the rules in OZ?
 
Chris, thanks for the reply and info.  That FAQ was interesting reading.  MP3Gain was the app we used to normalize the MP3 files on the Vortexbox via a PC connected to the LAN.  The MP3Gain FAQ mentioned 92dB in it's test case.  Is that the recommended figure ?

I must check to see what the transmitter has to help prevent over deviation.  Perhaps a couple of back to back diodes as a clipper limiter.  I must check up with ACMA in Australia too.

Thanks for the info
 
As a matter of interest, I built a community radio station back in 1993, and I'd be pleased to give you the benefit of the experience if you like.

http://www.radiosouthland.org.nz/

http://www.facebook.com/album.php?aid=1121&id=100000248474331&l=c07821773e

I get free toll calls to OZ from New Zealand if you feel like making voice contact.

email diack at xtra dot co dot nz
 
The problem with clipping, is the generated harmonics.

If your transmitter doesn't have a final limiter of it's own, once the audio is pre-emphasised and runs through high pass filters the peaks will be all over the place again. 16-bit sound cards can also output up to 3dB more in the digital to analogue conversion when it guesses the output level between samples, and most sound cards aren't flat frequency responsed or have phase issues on their outputs so shouldn't be used  straight in to a transmitter, but rather an FM limiter.

When it comes to normalising audio, be sure to normalise or average set the levels rather than bother worrying about peak levels (so long as the peaks don't clip). I'll explain more on this....

If in the future you should play say a peak normalised audio file through a professional balanced sound card, the output on most balanced sound cards is around +20 to +24dBu (far to hot for normal use).  If you normalised audio to say -3dBFS (-3dB digital peak on the computer), that means on a pro card you'll be running it at +21dBu (-3dB from +24dBu output almost consistently, which they are NOT designed for). This is far to hot for a card designed for a norminal output of +4dBu with 20dB of headroom. On average most songs will peak somewhere between -12dB for heavily compressed new CD's, to -8dB for more dynamic 80's and earlier material - in order to achieve the same average volume, or VU based level (not talking about peak level here).

This is more often than not a grey area when it comes to domestic sound cards and low power radio stations. Audio levels are never really given much thought or calibration. If for example, you have an analogue console with mechanical VU meters (volume unit meters, NOT peak meters), then if you drive the console by a 1KHz tone at 0dB on the VU meters, most professional consoles will output +4dBu RMS while allowing up to +24dBu for 20dB of peak headroom (the same way pro balanced cards are designed). If audio is normalised to VU standards, you'll find most of your audio will sound at the same volume on the PC, that way your mix sounds consistent and you don't drive any limiters hard then soft between varying material.

Unfortunately, modern digital peak meters incorrectly take on the name VU meters, when in fact they are not. Neither are peak LED meters on mixers. The difference is quite a lot. So don't plan on using material normalised within a few dB of maximum peak on the computer, on professional gear. You might get away with it on domestic sound cards and consumer mixers, as noise can be a factor. Pro cards thuogh go up to +24dBu and there fore if their signal to noise is say, -80dB, that's 104dB of noise room, unlike a domestic card outputting -10dBu with a noise floor of -80dB (you loose up to 30dB on consumer cards).

Normally to set a broadcast studio, you'd use a -20dBFS 1KHz sine tone on the computer, input that in to a console with true VU meters, align the console input to read 0VU on the console. Likewise then record a 1KHz sinewave from the console at 0VU in to a sound card and have it show -20dBFS on the computer. If you then set those levels up, you'll never need to worry about peak levels or recording levels on the computer again, you just follow the console VU meters and peak headroom is built in to the designed use of pro gear.

Again this doesn't really matter if your using consumer cards and audio gear. But if you ever plan to use it on professional gear, or VU meters intead of peak metered monitoring and mixing, it'll cause you headaches.

That's why audio should always be normalised using VU meters or the human ear, not just peak meters at the same peak level. It makes for an inconsistent mix otherwise and segues are all over the place level wise to the human ear.

Just my 2 cents worth. Just registered on the forum, going to have a look at RadioBOSS. I've used Master Control, Simian and a few other smaller systems. At the moment I prefer Rivendell (a linux based system) it's been such a workhorse but I enjoy looking at where others are going and what I haven't learn't up on.

Cheers,
Gavin,
New Zealand.

 
radiodungog said:
The above tests regarding normalization were done with the RadioBOSS compressor turned off so that listening tests were easier to do.  Ultimately we want to have our MP3 files normalized AND run the RadioBOSS compressor to keep transmitted audio levels up.
You can also consider using AutoAmp (automatic gain control) feature in RadioBOSS instead of Compressor. It will just make all the files sound on the same volume level. But result depends on what kind of music you play... It works fine in most cases.
RadioBOSS doesn't read the Replaygain information from tags.
 
Hello,
@Chris: Chris, thanks for the links and offer for a chat.  I'll take you up on that soon but I'll have to get my breath back first.  Our local area has a population of 8,000 people and only three of us are working on the technical stuff at the moment.  As soon as we get those techy things done in the next few weeks I'll be in touch.  Thanks ...

@kiwi_rock: Gavin, thanks for the comments and details on harmonics, audio levels and the way things can be set up and tested.  As mentioned above we have our hands full at the moment but your test procedures will be very valuable to us after we get the last few bits of equipment sorted out and working.  I'll be in touch soon for sure.  Thanks ...

@djsoft: Thanks for the suggestion about AutoAmp and the compressor.  We will probably stick with normalising the music files in the music library on the Vortexbox, and turn on the Compressor so we can get a little bit more audio punch.  The music we are playing will handle some compression and we have been experimenting with 6dB of compression with reasonable results.

I might add that we are designing our studio and station to have minimal equipment so that it is simple, reliable and has redundancy built in by means of duplicated equipment.

Essentially our station consists of:
1. Production PCs with audio consoles for the production of jingles, promotion tracks, community service notices, news and pre-recorded programs.
2. A Vortexbox for our music and audio library
3. A RadioBOSS PC as the automatic scheduler and audio feed to the transmitter
4. A WiFi Ethernet link over line of sight 3Km path to the transmitter hill
5. A streaming audio player and line matching system feeding to an FM broadcast transmitter running 100 watts
6. A simple folded dipole antenna adjacent to the transmitter hut

The above system is low purchase cost but has a high initial labour cost as we are doing all the work ourselves rather than buying an off the shelf solution.

Wish us luck :)

Thanks for all the answers and suggestions guys.

Jamie C.
 
Good Luck...  hope this never happens to you...
http://www.facebook.com/photo.php?pid=561875&l=02b05c8883&id=100000248474331
 
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