First, a question: does mAirList support replay gain information stored in MP3 files? And if so, does it support the information writte to files by the popular MP3Gain utility?
I ask this because my chief engineer just ‘sprung’ a request to use MP3Gain on our library (especially the Cartwall items like jingles etc.). He personally uses and endorses this utility, which he says works very well at his DJ gigs like weddings etc. (come on, we’ve all done that kind of gig at some point!).
He likes the idea that he can playlist any track in his travelling kit without needing to worry about its level, after he sets his initial desk and amp. levels at the start of the night.
Also, what does the panel think of replay gain in general? I understand the concept, and sort of approve of the idea of tracks which are proof against the ‘audio-challenged’ (i.e. deaf and/or clueless presenters); but my personal previous experience of replay gain was Not Good. Admittedly I wasn’t using MP3Gain at that time (some other utility), but I wasn’t too enthused.
I am not sure whether BASS.DLL reads and applies the ReplayGain information (it’s stored in some custom ID3v2 tag, isn’t it?). I haven’t found anthing in the docs or the forum about that topic. Can you please check it? Just significally change the replay gain of a file and then listen to it. (I don’t have mp3gain installed here at the moment.)
If BASS.DLL doesn’t apply the gain itself, this get a little complicated. I would need to know how to identify the tags, and what exactly to do with the samples. And I would need to employ the DSP callback features of BASS.DLL, which let you edit the samples before they get played. I haven’t done that before. Would envolve a little work.
I have seen some other BASS-based things which claim to utilise replay gain, so whether they also had to ‘DIY,’ or whether there is some DLL switch or setting somewhere to do this, I don’t know.
DLL callbacks can be a real PITA to write, so I wouldn’t bother doing that unless other people think it’s a good and worthwhile idea. Especially if this is your first one ;).
Re the tags: ‘it depends.’ If I understand the literature correctly:
MP3Gain will optionally write its changes to a tag (an APEv2 tag, apparently), so it can Undo them later. However, it more usually changes the data directly, but apparently reversibly (?). It changes the gain of each MP3 frame (!) to achieve a nominal specified SPL, for example the movie industry standard of 83dB SPL. Therefore in theory, the change should work on any MP3 player software (it says here ).
Yes, I’ve checked and MP3Gain does indeed change the ‘gain’ byte in each frame within the MP3 file. It also optionally stores ‘undo’ information (specific commands to MP3Gain) in a custom APEv2 tag within the MP3 file, to make it easier to ‘back out’ any change you don’t like.
So to answer my own question :), yes it does work with mAirList and BASS (and indeed any app. which can play an MP3 encoded audio file), because the files themselves are changed.
I’m still not convinced by the idea of ‘replay gain setting,’ though. A fairly heavily compressed station promo (6:1 ratio with hard knee at -9dB) sounds somehow ‘bland and muddy’ after processing. >shrug<
Seems that software processing is really becoming a big thing now, especially as my Nan is able to stream music online and set up her own station To me, and this is purely a personal opinion - sound processing should really be done at the end of the signal chain, not at the start. Peak levels should be watched, but not to the point of obliterating music files just to get a little bit more loudness out of them…
With fairly decent mid-range priced rack units, and free plugins - I can’t see how anybody couldn’t at least manage some sort of AGC+Multiband processor on their output
sound processing should really be done at the end of the signal chain, not at the start.
True, and indeed we [b]will[/b] be slapping one of yon wee Behringer boxes on the end of our chain. The bothersome part about that will be:
a) it's a PITA to feed the post-processing signal back to the presenter's cans so that the presenter can hear how it REALLY sounds, and
b) the Behringer has a (MINIMUM) processing delay of something like 300mS or so, making it an even bigger PITA to make this a cans feed.
Peak levels should be watched, but not to the point of obliterating music files just to get a little bit more loudness out of them...
Well, if we [i]did[/i] have PPMs on our desk (!), I would in general be advising everyone to peak at 6 for speech, 5 for most music, and 4-4.5 for Very Loud Rock tracks. Sadly we have stereo LED ladder 'meters' (bah!). But I know what you mean.
With fairly decent mid-range priced rack units, and free plugins - I can't see how anybody couldn't at least manage some sort of AGC+Multiband processor on their output ;)
1) We are presently a zero budget station: our Behringer processor thingy will be borrowed.
2) I wouldn't use EVER a 'plugin' in ANYthing, even if you DID put a gun to my head. I am an old school 'analogue but uses digital under protest' sort of chap. "Rack-mounted hardware good, dubious software 'plug-ins' bad."
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Leith FM is an FM radio station, not a ‘podcaster.’ We also can’t afford an MCPS/PRS licence to stream our output on to the Internet at present. Our only reason for doing that will be so that Leithers not presently in Leith can listen, but Internet streaming is not a big deal, and is well down our list of priorities. Personally, I don’t regard the Internet as ‘proper radio’ anyway; I don’t listen to podcasts on principle, and only rarely do I use the BBC Radio Player, on an occasion when I’ve missed a favourite show. I certainly would never choose to listen to radio via the Internet. Radio is radio, and Internet is Internet (NOT radio!).
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Sorry to drag this even further off-topic, but Cad - Perhaps you can explain (via PM if you wish) what Behringer processor you’re using that’s running a delay of 300mS (that’s quite a look-ahead delay!) I have an Ultradyne here, it’s quite latent - about 8mS but I’m used to it.
For some coffee-break reading, have a look at my studio page with some ideas and info my setup.
It’s a DSP9024 but I can’t remember precisely why there is yon delay. It’s some specific mode or other that our chief engineer uses on the box, which needs a minimum lookahead of 300mS or thereabouts.
If I remember to find out, I’ll PM you about it—as you said, we’re somewhat, er, >cough< off-topic!
We use an Ultra-Dyne as well for the Internet stream processing. The amount of lookahead can be customized. It claims to yield better results with a larger lookahaed.
The “delay” or lookahead is in the 2nd setup menu (hold it for a second or so). 300mS is a tad long, in my book. The DPS9024 Ultradyne is a nice bit of kit, considering it’s low price - Shame it got discontiued by Behringer. Nice unit for webstreams of home studios, I also use a TC Electronic TripleC (surprise surprise, also discontinued).
But both really need something before to act as an AGC, like a Composter/Multicom. But for the price, they are nice units - If a little hot when running in a cramped rack.
By the way, we have an old DSP9024, which is broken. Light goes on when you switch it on, but it doesn’t boot. Serves as a monitor stand now. In case anybody needs spare parts …
Torben - Does it do what you’d call “level crossing” flashing of the LEDs ? Something like that (and many units have failed in a similar way) is down to the PSU. I just got one back from repair for this - The Behringer site will have details of you local repair shop - Or if you’re close to Uli’s place, you could pop it through his letter-box!
The display hangs at “Welcome to the interactive world of Behringer”, and the MIDI/PROCESS/METER/SET UP LEDs are flashing. Upon my support request, Behringer recommended to send it in for diagnostics, but before I could do so, I had found a cheap replacement on eBay.
Definitely a PSU fault, and cheap to repair. Mine was away for less than a week, and they said they’d uprated the new parts - So it wouldn’t happen again.
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