LET'S TALK MICROPHONE PROCESSORS Favorites AND Unique

[Jay Walker]

I've been involved with AM/FM music radio processing for four decades.

(I'm showing my age )

I strive to set the microphone processor so the talent floats over the song intro to match the intensity and placement of the lead singer in the mix. In other words I try to make the air talent sound like he's a part of the song rather than an overlay. I equate this visually to the music level rising to the chin height on the talent if that visual makes sense. There's nothing worse to my ear than a jock who plays roller coaster with the fader ducking the music up and under...

Over the years I've had the opportunity to use many different models of on-air microphone processors both as a talent and as an engineer. In no particular order here are some of my favorites and/or unique devices. Notice most of these are 30 or more years old. Sadly I've not really found a DSP device that beats these old rack warmers...

Here's the short list in no particular order:

Orange County Vocal Stressor (great in production rooms, smooth to brick wall)
Valley People Gain Brain/Kepex (universal classic combo)
Symetrix 528/528E (cheap but effective all in one)
DBX 900 Series (smooth as silk)
Pacific Recorders (PR&E) 99-415, 99-416 & 99-296 (nice combo tight sound)
Valley 400 (updated gain brain)
CRL "Instant Personality Processor" (Early interesting multi-talent selection)
Universal Audio LA-610 (tube warmth, think Urei LA-2)
Universal Audio 6176 (Urei LA-2 like sound)
Manley VOXBOX Tube Processor (another classic tube sound)
Air Corp Model 500PH (smooth, but does overshoot)
Shure Level-Loc (a cheesy little box from way back in the day junk but cute)


I am under impressed with the new Air Tools 6200 DSP processor and other DSP based devices.
I have not found a sound that I'm looking for. Most get close, but not close enough for the price.
Any recommendations for DSP based microphone processors you've had success using in music radio applications with live air talent?

What's your preference?
Jay Walker

 

[richard.vanderveen]

I really like the Yellowtec Digital VIP, combined with a large membrane condenser. Not east to get the settings right, but once they are... Very transparent and controls the lows very good without sounding muddy. Th fact that you can save your preset on a chipcard and load it into any other VIP is also nice.

 

[bigtom101]

I love using the M1 personally...it has DSP, a traditional front panel design, and latency is very low.

 

[WNTIRadio]

Some places I have talent that can't take any sort of latency. That rules out the digital options. They couldn't even deal with the 5ms or so from the "DJ Headphone" output of the Omnia.11 for monitoring. Ye Olde 8100a to the rescue.

An Orban 422 or 424 works great with a lot of ability to tailor the sound too. They can be as punchy as you want. Need an outboard pre for them, or if your board has an insert point on the mic channels they work great there too. Downside is no expander/gate.

Also liked the Valley 400, but they're almost unobtanium now.

 

[chriscollins]

The Valley 400/401 are still what I consider the best processors ever made.

With that said, I use the Aphex 230D, which I think is discontinued on both of my digital studios. It gives me as good as, or better than Valley. It just takes a little more tweaking.

 

[Jay Walker]

Some places I have talent that can't take any sort of latency. That rules out the digital options. They couldn't even deal with the 5ms or so from the "DJ Headphone" output of the Omnia.11 for monitoring. Ye Olde 8100a to the rescue.

An Orban 422 or 424 works great with a lot of ability to tailor the sound too. They can be as punchy as you want. Need an outboard pre for them, or if your board has an insert point on the mic channels they work great there too. Downside is no expander/gate.

Also liked the Valley 400, but they're almost unobtanium now.

When I built up a CHR "Flame-thrower" I used the Pacific Recorders Processors/EQs in series with the Orban 424 at the insert point on a BMX-3. The talent loved the audio and it was stout without the annoying visits from the "feedback bird" even with talent who insisted on using open ear headphones at full volume.

Still got my Valley People 400 in the home studio...

 

[Dale H. Cook]

I strive to set the microphone processor so the talent floats over the song intro to match the intensity and placement of the lead singer in the mix.

That is difficult to do unless:

1) You are live and the jocks are good at running a board.

or:

2) You are automated and the music library has been normalized to put all of the lead vocals at the same level.

 

[dtube1]

IMO, mic's require analog processing. The preamp circuits and A/D convertors in most broadcast equipment leave a lot to be desired when measured against recording equipment. I am advocate of recording-type, channel-strip processors like the Grace m103 or API Strip units. The problem with those though is price. Plus, you'll need to add a good-quality gate/expander unit on the inserts. I mentioned in the other mic proc thread about having good luck with Presonus Eureka strips with a a Drawmer 4-ch gate. That combo sounds great and has been and has been surprisingly hearty over the years.
-D

http://www.gracedesign.com/products/m103/m103.htm

http://www.apiaudio.com/chanstrip.html

 

[Jay Walker]

I strive to set the microphone processor so the talent floats over the song intro to match the intensity and placement of the lead singer in the mix.

That is difficult to do unless:

1) You are live and the jocks are good at running a board.

or:

2) You are automated and the music library has been normalized to put all of the lead vocals at the same level.

There is a bit of work required for sure. Any station I consult or do day to day, I always stress the GIGO principle "garbage in garbage out". So quality control is the first step to the desired result. The other key element to the "float" is correct settings of the follow on processor and it's recovery time(s). It's a tricky balance but when set correct it sounds wonderful. When the talent hears how full their voice is in the monitor/headphones they tend to not over drive the mix. Usually a 3db difference between the intro and voice gives the perfect result down stream.

I fell in audio love with this mix concept as a baby disc jockey when I worked my first "big" gig on a Top-40 Kansas City FM. It made me sound like I had grown an extra set so to speak. ;D

Story below...

The most intense processed station I worked on air as a talent was KBEQ Kansas City MO. around 1974/75. You could go from the cold ending of Radar Love by Golden Earring into Don't Let the Sun Go Down on Me by Elton John and not hear a level change. The downside was Elton's piano intro sounded like it was being played with a sledge hammer. That station compressed the music onto cart with a stereo Audimax and used Urei LA-3s into an 1176 Limiter. Loud yes, but pretty fatiguing... The microphones used Valley People Kepex/Gain Brain for processing. I don't recall the CE's name I was only the 17 year old over night guy.

One other point I'd like to make regarding music transfers to disc delivery systems.
We NEVER RIP audio into the system or rely on automated normalization, music is always recorded real time with each beginning level set for 0vu. The dubbing personal then rides gain through out the dub process. The result is every song whether live or automated always starts at 0vu and maximum peaks are always 0vu. It takes more effort and attention to detail, but attention to detail is the difference between winning and losing...

 

[oldiesstation]

I was a fan of the Valley processors. think Galaxy audio may still service them. A friend that does lots of national radio and tv put me on the M1.Dial it in and it really sounds good.Also have the M2 on the way.Never cared for the symetrix analog or digital boxes.Most jocks run way too much compression for that big gonad voice.Dbx 286 is a nice value priced box.

 

[rrsounds]

The eight years I was CE at Z100, I never felt the need to touch the mics and processing that Frank Foti had put in: Shure SM-7 with poplesss filter into a UREI LA-4 compressor with roughly 10 dB GR at 8:1, with a UREI octave EQ set to mildly push up the upper mid range and a bit of bass. Maybe not the most pristine setup, but I liked it because I could easily recognize - and appreciate - the jocks' individual voices.

Kind Regards,
David

 

[WNTIRadio]

Jay, I have them rip songs in but use Adobe and the correction tools in there to fix things.

I do this for three reasons:

1. It's cleaner to rip the digital bits in than an unnecessary trip through the console and all of the associated electronics.

2. If a mistake is made, there is the handy "undo" feature.

3. The level control is more precise and repeatable than a hand on the pot. You can see what you're doing on the waveform as well in regards to clipping and being able to max everything out at a certain level.

I use -10dbfs as a reference to give any digital boards or downstream equipment headroom.

 

[Jay Walker]

Jay, I have them rip songs in but use Adobe and the correction tools in there to fix things.

I do this for three reasons:

1. It's cleaner to rip the digital bits in than an unnecessary trip through the console and all of the associated electronics.

2. If a mistake is made, there is the handy "undo" feature.

3. The level control is more precise and repeatable than a hand on the pot. You can see what you're doing on the waveform as well in regards to clipping and being able to max everything out at a certain level.

I use -10dbfs as a reference to give any digital boards or downstream equipment headroom.

Your reasons are all good reasons, but by having the songs dubbed in real time the dub engineer has the ability to manage the intro levels. Since we always start a song at 0vu we don't have the issue of hard level changes from song to song. While ripping allows max levels to be set at 0vu, it does not take into account the lower intro levels (less than max peak) found on the music cuts.

Another advantage at my operation, everything dubbed is, with the exception of analog turn-tables (limited use) is all digital from the time the hand loads the CD until the hand clicks the mouse for playback. So any analog noise/distortion is well below the "bar" so to speak.

I equate Real-time dubbing to having the dub engineer run the song like a good board op on the air. Starting every event at 0vu makes a big difference in transitions IMHO. BUT it is not quickest way to load a library.

I too use ProTools or Adobe to massage the clicks, clips, and other distractions in cuts. Some of the material we use are from vinyl copies (the only available source) and Abobe is essential to clean up the cuts.

One other point, we have "built in" a reasonable amount of headroom to avoid the problem of too hot levels. Also all our cards are Pro cards. There are no semi-pro cards used in the plant that get near the on-air chain. The only time semi-pro cards are used are in traffic for spot preview etc. Real-time versus Ripping to me is an extra advantage that I use to give music transitions a more "real feel". If I did not have the personal to do Real-time then I'd use (begrudgingly) ripping into the system. This is one of those "to each his own" operational items ;D

We are lucky at our operations, since we are able to justify manpower to be hyper focused on Quality Control. The total sum of our efforts are reflected in our ratings. Top 5 in the money demo for many books, which is a direct result of the synergy between programming, operations, and engineering. It's as close to "Radio Heaven" as you can get in today's cluster based operations.

 

[chriscollins]

Jay, I have them rip songs in but use Adobe and the correction tools in there to fix things.

I do this for three reasons:

1. It's cleaner to rip the digital bits in than an unnecessary trip through the console and all of the associated electronics.

2. If a mistake is made, there is the handy "undo" feature.

3. The level control is more precise and repeatable than a hand on the pot. You can see what you're doing on the waveform as well in regards to clipping and being able to max everything out at a certain level.

I use -10dbfs as a reference to give any digital boards or downstream equipment headroom.

I totally agree on this. That is the way we deal with problematic music. Nothing is more precise than a good digital waveform edit. They serve us well. The fact that I am a Program Director and IT Manager also helps, as I get the way I need to use tech to make my station better.

 

[Jay Walker]

Jay, I have them rip songs in but use Adobe and the correction tools in there to fix things.

I do this for three reasons:

1. It's cleaner to rip the digital bits in than an unnecessary trip through the console and all of the associated electronics.

2. If a mistake is made, there is the handy "undo" feature.

3. The level control is more precise and repeatable than a hand on the pot. You can see what you're doing on the waveform as well in regards to clipping and being able to max everything out at a certain level.

I use -10dbfs as a reference to give any digital boards or downstream equipment headroom.

I totally agree on this. That is the way we deal with problematic music. Nothing is more precise than a good digital waveform edit. They serve us well. The fact that I am a Program Director and IT Manager also helps, as I get the way I need to use tech to make my station better.

My only point I wish to make is "Normalization" does not bring the Intro of a song to 0vu, as you know it only sets 0vu at the maximum level in the song which can be anywhere in the song . In our operation, the best way we've found to make music transitions consistent is by recording the songs onto the hard disc system with the beginning level set at 0vu.
Then the dubbing engineer rides gain throughout the rest of the song. It's tedious yes, but like I've stressed the transitions are much much cleaner...

Again we are lucky since we have the manpower to devote to this tedious process. Now if there was an "automated" way to make an intro 0vu without using additional compression... that would be a different scenario.

Bottom line, whatever works in your situation is what's best ;D

I'll quit beating this dead horse ;D

I appreciate all of your quality input. Professional dialog between peers, is for me a great way to learn even better ways of performing my duties. Even after 40 years I strive to learn, and explore new, better ways of doing things.

 

[TomT]

We use Sound Forge, with it you can normalize just a section of the file. Also allows for fade in or fade out in a selected portion of the cut--useful for those "live" recordings, or to shorten certain cuts (especially where you don't have a "radio edit" copy to drop in). E.G. "Mr. Bo Jangles."

 

[WNTIRadio]

I never use auto normalize in anything. But, I do use, and instruct people carefully, to use the fade features of Adobe Audition. I can start a song at 0vu and program the fade to bring it down to 0vu when the song gets loud and vice versa.