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All About Limiting For Web Radio

G

GeneSavage

I'm in over my head, but I'm wanting to learn.

I run an internet-only station and am trying to process the audio to sound similar to 80s CHRs. To accomplish this, I'm running Stereo Tool in free mode as a gentle multiband leveler into Sound Solution 1.2, which gives me my density.

I've received numerous compliments on my sound, but the other most frequent comment is that my output seems low. I don't doubt it; Sound Solution's limiters are neither distortion-canceled nor look-ahead. At my "optimum" settings, I seem to be about -5 to -6 db below 100% output for peaks... and they look VERY uncontrolled in Adobe Audition.

Someone in another post here commented that without that sort of limiter, you have a choice between low output with uncontrolled peaks, or the grit and distortion of hard clipping.

I first through I needed distortion-canceled clippers (and who's going to provide that in a free plug-in?), but reading further it appears those who do this all the time agree that for streaming you can get excellent results with a look-ahead limiter... results equally good to distortion-cancelled clipping.

I've found a free wideband look ahead peak limiter plug-in and am looking at trying putting it as the last thing in my streaming chain before it hits the encoder (mp3 and AAC+ streams).

Several questions:

1) How fast a release can I run without destroying the bass? I've read everything from 100 ms to 40 ms, and this limiter can go all the way down to 3 us, if you can imagine! (It sounds, obviously, awful if driven more than half a db at that speed.) What's my shortest "safe" release time for 20Hz - 15Khz?

2) Where should I brickwall my signal output at? By slamming everything into the red, I was able to get a level of 98.51% (or -0.13 db). I was feeling bad about not being able to peak out at 100% until another webcaster said he peaks out at -3 db because of lousy D/A converters in many stock soundcards listeners are likely to be using... similar, I'm guessing, to the problem of asymmetrical modulation on AM: it works on some radios, but not all are able to reproduce 125% modulation cleanly.

3) With multiband compression and limiting in front of it, would there be any advantage to a multiband look ahead peak limiter? (If you say yes, I'm not sure what I can do about it; the budget for my station, according to my wife, is zero dollars and zero cents... I'm guessing some of you have had similar experiences at stations you've worked for: "can't you just... I don't know... put some Scotch Tape on it or something?!?") ;D

4) Is there a recommended drive level for a final peak limiter? This thing can be driven up to 24 db, but of course it sounds ridiculous. I seem to have seen a number of final limiters which can be driven up to 5db; is 2.5 db a good average, should I hit it a full 5 db, or just 1 db so it's just catching the most occasional of peaks? (Or is this a "your mileage may vary" question?)

5) I have the option of operating it in stereo, "peak mono" (looking at the sum to determine when to dip), and dual mono. Since this processor isn't supposed to do much & so shouldn't mess with the stereo image, would dual mono make the most sense?

Advice, recommendations, suggestions are greatly appreciated! Having this pool of talent & experience available to ask about projects like this was unimaginable when I was growing up and listening to the powerhouse stations both locally and coming in on skip from around the U.S. THANK YOU for your help.
 
Leave it alone. You're already sucking out what little life the music had with the processing you already have. Let alone, it's digital, being reencoded again to digital, then being converted yet again back to analog at the listener's end. When you start compressing the hell out of it, the artifacting just becomes that much more apparent.

Back the peaks down to at least -4 dB to keep from clipping. If your listeners complain, tell them to use a plug in to squash it further at their end, or God forbid, use the friggin' volume control. Digital needs around 5 dB of headroom compared to analog, because it isn't forgiving in the least. Analog can suck up a lot of overshoot before it becomes unlistenable, digital can't absorb any. It clips, period, and when digital clips, it sounds awful.

Search the net for a white paper by Dorrough on "listener fatigue" and then decide if you want to crush it any further, especially in the digital domain. My biggest complaint with radio, OTA or internet, is that everyone thinks they have to have the loudest and most dense audio, even if it ultimately drives the listeners away. Most stations sound like crap because they NEVER, EVER relax as far as dynamic range is concerned. Not to mention the destruction afflicted on the stereo soundfield. It's literally like an assault on my eardrums, and I can't listen for more than a few minutes before I either turn it down below the threshold of awareness, meaning I really couldn't tell you what just played, or usually I'll just turn it off and put an album on.
 
At least from what I've read you didn't mention anything about your encoder software or streaming bandwidth. These are also key areas to keep in mind.

Back in the days when I operated a deep-formatted Oldies (60's - 80's) webstream I used two different audio chains with great success. The first one was a pair of 4 band musician compressor / processors where the first one operated slower to mimic an AGC while the second one operated more as a limiter. The second chain was a barefoot Opitmod 6200 DAB. Both were used with the same encoder, Spacial Audio's Simplecast.

While both sounded great I noticed a bit of improvement with the Optimod as I was able to bandwidth limit the output to better match the streaming bitrate (at the time, 64k). Not giving the encoder extra info to chew on only to have it discarded in the encoding process helped tremendously. Like mentioned above, too much processing will drive your listener cume into the toilet. The biggest "complaint" I got with the Optimod was while it sounded loud it also sounded much more open.

80's music is already better processed than the stuff that was my main playlist so you should have an easier time tweaking. Louder isn't necessarily better.
 
Radio boogie, let me quote myself:

...I've received numerous compliments on my sound...
Click to expand...

Listeners LIKE what they hear me doing to the songs. I'm not clipping it to death, I'm creating a consistent sound. One respected audio compression software designer said (BEFORE I added Stereo Tool to the chain) that my setup was the best he'd ever heard Sound Solution sound.

Indeed, I've heard Sound Solution abused over and over... most of the presets (including the default) pump, breathe, and generally destroy the audio. If Sound Solution is pushed much at all, the sound turns to fuzz.

In Stereo Tool, I ONLY have the 10-band compressors on, and I have both the attacks and releases set to their absolute lowest. It is a very gentle, 10-band dynamic equalizer, basically. The difference is subtle, but it helps Sound Solution not jump and duck when one thin part of a song suddenly gives way to a bass-rich section. It's actually REDUCED audible artifacts.

Those of us who grew up with 80s radio actually LIKE the sound of some of the flamethrowers of the era. Do we want our CDs to sound that way? NO. In fact, today's audio processing tends to be set to clip the snot out of the audio with little gain riding, giving a sound that is very fatiguing to my ears, but apparently is appealing to... listeners? Engineers? More than likely, Program Directors.

I actually use a program called SeeDeClip Duo Pro on a number of my audio files to try to restore some of the original dynamics of the songs I play. I want the processor to have to deal with the original dynamics of the recording, not some buzz-clipped "remastered" version of the song.

My actual average RMS is about 16db... does that sound like I'm squashing the life out of these songs to you?

Additionally, I'm looking at a look ahead limiter for the very reason you mentioned: to PREVENT digital clipping. I want to hit 100%, but not go one sample over.

FWIW, I put the limiter in the chain last night... WHOA! First and foremost, it allowed my output to ACTUALLY be 100%... I've now backed it off to nearly -1 db output. Second, I'm using less than a db of limiting, just to catch the peaks and it is PLENTY. Third, I had the release at the default of 75 and it started to sound "dirty" to my ears after a while. Backed it all the way off to 300 ms, and it's sounding pretty clean, I think. I thought I'd hear pumping at that slow release time, but I'm not aware of it.

Bill, you asked about my encoder software and streaming bandwidth: I'm using Edcast as my encoder, encoding one stream at 96kbps with the latest version of LAME, the other at 64kbps with AAC+, using the FAAC codec.

Thanks for the replies!
 
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