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Broadcast quality voice recording/processing


Beedster
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I've done a couple of things for BBC Radio and the voice quality compared to what I'm getting in my studio was noticeably better. I'm interested in knowing what sort of processing is routinely used in radio, the rooms themselves and the mics are pretty standard, and there doesn't appear to be a standard mic/room setup (for example in one studio I had a ceiling/shock mounted SM7B I think and in another a much smaller and desktop mounted unit) so this isn't about acoustics or mics, placement or any of that stuff. Keen to hear thoughts and experience

Cheers

Chris

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Is it to do with the baked in multiband compression they use? I know that different stations have different settings. Welsh BBC Radio is classified as "Pop" station so has a specific multiband setting across all the output. But it is actually an all things to all people station. On a Sunday morning my clock radio used to wake me to a light classical programme on said station. For years I was convinced that the DJ was spreading a specific reverb over everything until it was explained to me that it was the result of pop multiband compression on classical music.

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Rooms are acoustically treated, mics are top notch, preamps and dynamics are (usually) analogue and classy, most of the audio is in the analogue domain I suspect, plus the engineers are very experienced at voice recording.

The output from the production will be in line with whatever multi band processing the broadcast chain has, rather than fighting with it, so level and loudness content will be tailored to the broadcast output.

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3 hours ago, owen said:

Is it to do with the baked in multiband compression they use? I know that different stations have different settings. Welsh BBC Radio is classified as "Pop" station so has a specific multiband setting across all the output. But it is actually an all things to all people station. On a Sunday morning my clock radio used to wake me to a light classical programme on said station. For years I was convinced that the DJ was spreading a specific reverb over everything until it was explained to me that it was the result of pop multiband compression on classical music.

That's really interesting, must admit it didn't occur to me that it might be something to do with the broadcast process. having said that, a couple of the pieces in question are online - and therefore not broadcast per se - so would that same baked in process apply? 

13 hours ago, Erax Sound said:

Compression, basically. Not too much else.

I think what I'm looking for might be in the 'not too much else' :)

There's certainly a smoothness to the voice track that suggests a few processes - compression, de-esser (I'm very essy), zero plosives (I'm also very plosive) - without sounding overly processed. I'm struggling to achieve the same in my studio, at least without the voice track sounding overly processed, so I'm guess it's just the need for more experimentation. I guess the core of the above question is whether there's a recognised processing model for broadcast that might serve as a staring point for me; I'm pretty confident for example that my voice wasn't treated to any bespoke FX, as I doubt the Beeb feel the need to make my voice sound any better than it does through it's normal signal path?

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1 minute ago, WinterMute said:

Rooms are acoustically treated, mics are top notch, preamps and dynamics are (usually) analogue and classy, most of the audio is in the analogue domain I suspect, plus the engineers are very experienced at voice recording.

The output from the production will be in line with whatever multi band processing the broadcast chain has, rather than fighting with it, so level and loudness content will be tailored to the broadcast output.

Thanks, more food for thought :)

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I'm happy to do a before and after for you @Beedster. Send me a file.

There are many things at play like the signal chain, mic, timbre of voice. There will be small parts of everything going on of course.

Mind you, I've edited podcasts that were really badly recorded and no amount of creative effects would have made it sound like the smooth product you want.

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2 hours ago, Erax Sound said:

I'm happy to do a before and after for you @Beedster. Send me a file.

There are many things at play like the signal chain, mic, timbre of voice. There will be small parts of everything going on of course.

Mind you, I've edited podcasts that were really badly recorded and no amount of creative effects would have made it sound like the smooth product you want.

That's very kind, I might take you up on that. I'm going to do some playing first :)

Anyone able to recommend a resource re multi-band compression for voice work? 

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When we recorded voiceovers or ADR (I'm an ex BBC TV sound recordist) we'd use Schoepps or Neumann mics going direct into Neve or SSL mic amps in Neve or SSL desks. Any compression (which was minimal) or eq tended to be from the same companies. If it was for a film dub we'd record onto 16mm mag film with Dolby SR, video would be onto Sony DAT.

Later, recordings would be digital only into Lightworks, which had very little in the way of plug ins, then when we'd finally got Avid and ProTools to work together, onto that.

As far as I know, the ProTools route is still the way for TV (it still is on my old programme), and it was only by about 1999 that we finally started with heavy compression, simply because production people were getting fed up of how loud trails would be between programmes; we went into more subtle multiband as we wanted at least some semblance of dynamic range!

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On 25/04/2021 at 15:17, Beedster said:

That's very kind, I might take you up on that. I'm going to do some playing first :)

Anyone able to recommend a resource re multi-band compression for voice work? 

If you are on Reaper, you will have ReaXcomp in the toolbox 👍

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On 25/04/2021 at 12:39, Beedster said:

...de-esser (I'm very essy), zero plosives (I'm also very plosive) - without sounding overly processed...

A lot of these phonic artifacts can be greatly reduced, or eliminated, by the simple expedient of having the mic at a greater distance. The downside is, naturally, capturing more of the 'room'. A 'dead' room will help with that; try recording the voice with the mic behind your head, 'looking' over your shoulder. With a heavy blanket suspended in front of you, there'll be less sibilance and pops. It's possible to have a 'normal' second mic, and mix the two, if phasing can be mastered, too.
Just a thought. :friends:

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Don't forget that radio presenters are often chosen as much for how their voices sound as anything else.

In the days when I did student radio (a long time ago, but the technical standards were very high even if the program content wasn't always), studio voices were done with a good quality microphone mechanically isolated from the room, a dead-ish room and a decent distance between the presenter and the mic, then some compression and maybe a touch of added reverb (the tiniest amount so that it's only just obvious when you A/B between reverb and no reverb).

Ultimately though, those presenters blessed with a "radio voice" always sounded better than the rest of us irrespective of the processing.

Edited by BigRedX
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On 26/04/2021 at 20:44, Dad3353 said:

A lot of these phonic artifacts can be greatly reduced, or eliminated, by the simple expedient of having the mic at a greater distance. The downside is, naturally, capturing more of the 'room'. A 'dead' room will help with that; try recording the voice with the mic behind your head, 'looking' over your shoulder. With a heavy blanket suspended in front of you, there'll be less sibilance and pops. It's possible to have a 'normal' second mic, and mix the two, if phasing can be mastered, too.
Just a thought. :friends:

Thanks Dad, I have a vocal booth so room is not a problem, but the dynamic range of my voice using an SM7B is reduced the further away the mic from my mouth. I'm doing a bit of trial and error on this at present :)

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On 27/04/2021 at 13:34, BigRedX said:

Don't forget that radio presenters are often chosen as much for how their voices sound as anything else.

In the days when I did student radio (a long time ago, but the technical standards were very high even if the program content wasn't always), studio voices were done with a good quality microphone mechanically isolated from the room, a dead-ish room and a decent distance between the presenter and the mic, then some compression and maybe a touch of added reverb (the tiniest amount so that it's only just obvious when you A/B between reverb and no reverb).

Ultimately though, those presenters blessed with a "radio voice" always sounded better than the rest of us irrespective of the processing.

Hadn't thought about adding an FX to be honest, might try a few over the weekend. My voice is actually pretty good (vocal tone a whole lot better than bass tone), something that's been picked up by a few of the radio people I've worked with. Guess this thread is about trying to get the best out of the gear I have. So wish that terms such as Neumann, Neve or SSL weren't mentioned :( 

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14 minutes ago, Beedster said:

...the dynamic range of my voice using an SM7B is reduced the further away the mic from my mouth...

For reducing sibilance and 'pops', that's probably a Good Thing. Yes, experimentation is, indeed, the key to optimising. It seems as if you have it covered. :i-m_so_happy:

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Get nice and close for the proximity effect for starters. I can highly recommend the DBX 286S for an all in one preamp, compressor, de-esser and ‘enhancer’ (it has low and high controls that sound a bit similar to BBE processing, I have no idea what they’re really doing but they sound good!).

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12 hours ago, dannybuoy said:

Get nice and close for the proximity effect for starters. I can highly recommend the DBX 286S for an all in one preamp, compressor, de-esser and ‘enhancer’ (it has low and high controls that sound a bit similar to BBE processing, I have no idea what they’re really doing but they sound good!).

There's a trade-off between the proximity effect and having to deal with excessive sibilance and plosives. It will vary depending on the voice and the microphone. You'll need to experiment to find the optimum distance. Generally you want to be less close to the mic then you would for singing.

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37 minutes ago, dannybuoy said:

The SM7B is really good at preventing plosives up close also, since the capsule is buried deep down within, which prevents you getting too close to it.

Yes it is, needs a high gain preamp though, as the output is tiny!

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I have learnt many things during lockdown.  One of those is that Izotope RX8 Audio Editor can do fantastic things to the videos recorded in kitchens which people insist on sending to me as their contributions to online church every week. It can make it sound like something that was at least not recorded in a reverberant kitchen - in a really good way.

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  • 1 month later...

Get/make a good two ply pop shield.

Thats the plosives sorted.

Get the mic at least 2 inches behind the pop shield off axis by between 30 and 60 degrees -adjust to taste.

Download Tokyo Dawn Labs Nova dynamic eq, watch a few videos on how to use it for de-essing, Dan Worrall is yer man btw.

Add some very very subtle ambience with a vst, very very subtle mind!

One other thing that the nice analogue part of a BBC chain is adding is some very subtle saturation, try an analogue obsession vst for a tiny smidge or the Tokyo Dawn Labs Slick EQ which is the canine undercarriage.

 

 

Edited by 51m0n
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If you want to do multiband - and its a worthwhile thing to experiment with - then might I siggest not going to your nearest multiband compressor to do it.

If you use Reaper its perfectly possible to set up a linear phase crossover with the supplied plugins and its very clever routing, as described here:-
 


Personally I would be looking at using this with compressors/saturators on each band of my choice. Molotok might be excellent in this role for the low end or mids.

I would put Nova on the main input channel to de-ess first, then go from there into the crossover, apply compression/saturation to the different bands as required, back out to the combined channel and then add an aux to the ambience.

I would also recommend Kotelnikov for a final extra light compression if required. 

Its sounds more complex than it is in practice.

https://www.tokyodawn.net/tdr-nova/
https://www.tokyodawn.net/tdr-kotelnikov/
https://www.tokyodawn.net/tdr-molotok/
https://www.tokyodawn.net/tdr-vos-slickeq/

Definitely watch Dan Worrall's superc 'Introducing' vodeos on those plug ins. They are superb.

Some of those Analogue Obsession plugins for a different flavour:-
https://www.patreon.com/posts/nos-bundle-49184069
https://www.patreon.com/posts/britpressor-44141645
https://www.patreon.com/posts/specomp-51285294


I would be very surprised if you couldn't get a huge way toward the sound you are looking for with this lot to help.

Might be worth looking at the output on voxengo span of the podcasts you like the sound of to help figure out what the 'magic' is?
Yet another Dan Worrall video, I come across like a fan boy, but his delivery and engineering nouse are equally superb:-


If you want you could send me a wav and I'll set up a project to show you a bit of the processing chain I am talking about - assuming you use Reaper still???

 

 

Edited by 51m0n
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