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VO-BB - 19 YEARS OLD!
Where A.I. is a four-letter word.
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Rob Ellis
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Joined: 01 Aug 2006
Posts: 2385
Location: Detroit
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 Posted: Mon Mar 24, 2008 6:41 am Post subject: Test Driving a TLM 103 and getting (even) more confused! |
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I think this mic has good potential for my voice, but at the same time it is often overly BOOMY, (my voice is somewhere between a bass and baritone) especially when I raise my voice or get too close to it.
Side micing helps some, as does using the high-pass filter, but I have mixed feelings about having to do these things. Here's the signal chain:
TLM 103-Focusrite Voicemaster Platinum pre-amp-Mackie Spike (w/ gain down) as audio interface
Here's an audio file to illustrate what I'm talking about:
http://www.vo-bb.com/demos/RobEllisTLM103frontaddress.mp3
Would a different pre-amp maybe remedy this? Or should I just surrender and get a U-87
HELP! |
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BenWils
The Thirteenth Floor
Joined: 08 May 2006
Posts: 1324
Location: In a Flyover State
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| Posted: Mon Mar 24, 2008 8:07 am Post subject: |
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You have EQ settings on that Voicemaster. They are there to be used.
I hear nothing wrong with the audio from that mic. You sound like you have morning voice to me. I bet it would be much the same on another mic. This is where proximity comes into play and your noise floor. If you are having to crowd the mic because your recording area is so noisy, then you will get recordings where the mic is much more bass heavy. Back off the mic and your vocal will not have as much bass response.
Before you question the Neumann mic, question your listening environment and monitor set-up though (standing waves etc.....yeah, I know, WTH are standing waves....I'm just a voiceover).
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Ben
"To be really good at voiceover, you need to improve your footwork and hip snap." |
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Rob Ellis
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Joined: 01 Aug 2006
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Location: Detroit
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| Posted: Mon Mar 24, 2008 8:45 am Post subject: |
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Thanks, Ben. Sometimes it helps just to get an objective ear and viewpoint.
| Quote: | Before you question the Neumann mic, question your listening environment and monitor set-up though (standing waves etc.....yeah, I know, WTH are standing waves....I'm just a voiceover).
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I have wondered if maybe my monitors, which are pretty heavy-duty, may just be overemphasizing the bass. They have a bass filter switch which I usually have in the "cut" position (versus the "flat" position). |
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Yoda117
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Joined: 20 Dec 2006
Posts: 2362
Location: Philadelphia, Pennsylvania
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| Posted: Mon Mar 24, 2008 9:58 am Post subject: |
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Hmmnn...
My issues were already brought up by others in this thread.
Quick question though, what is your recording level set at? Maybe it's me, but it sounds like this was originally recorded a tad too hot for my tastes.
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Philadelphia based Voice Actor
Blog - A man, a martini, and a lot of microphones |
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Rob Ellis
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Posts: 2385
Location: Detroit
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| Posted: Mon Mar 24, 2008 10:15 am Post subject: |
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| It was recorded at +40db, with the Output Fader at -10 I believe |
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BenWils
The Thirteenth Floor
Joined: 08 May 2006
Posts: 1324
Location: In a Flyover State
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| Posted: Mon Mar 24, 2008 12:19 pm Post subject: |
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You may want to increase your output and decrease your input some. -10 doesn't sound like where I would start with the output knob.
I also have the V and when I use it, i am set at just a notch below 0 dB on the output fader.
Also the V has MANY settings, so it can be variety of things when you have issues with your sound. Are you using the Tube Sound?....you have voice optimised (european spelling) EQ settings. What do you set those levels at for Breath, Mid, Absence and Warmth....Vintage Harmonics...... and on and on.
So like I said, before you question the Neumann, check other things. Also, are your monitors setting directly on your mixing desk/workstation? This can cause resonance issues giving you a false sense of bass response. Aurlex MoPads can help. If you want to really do it up, get bass traps for your room to make sure the "squareness" of the room is toned down. LENRD (Low End Node Reduction Device) bass traps will do.
Bass = large sound waves. Once they start bouncing....your mix will sound thin because you are overcompensating for what the room is doing. Adding a couch behind your mixing station can be a fix for most studios with room. It reduces the "square" shape of the room and knocks down reflecting low frequencies.
The easiest fix though.....turn down the Warmth knob on the V if your voice is too bass heavy. And don't use the Tube sound either. It is a very aggressive sound.
Send checks to made payable to me, at my address. Gift cards also accepted. 
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Ben
"To be really good at voiceover, you need to improve your footwork and hip snap." |
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Rob Ellis
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Location: Detroit
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| Posted: Mon Mar 24, 2008 12:34 pm Post subject: |
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Thanks again y'all . I think I finally have this resolved.  Yoda triggered the right thought, on the input level, and Ben brought the idea home with his specific recommendations. I reduced input to line level and increased the Output fader to "0" and the reverberation has been reduced noticeably, and that crisp Neumann sound is coming through much better! I'm beginning to think the 103 may actually like my voice.
I don't really use the Tube sound or Vintage Harmonics. I mainly use the Compression and Expander, both set at 12 o' clock.
All out of checks Ben, but I may have a Starbucks card somewhere around here...seriously, I do appreciate the input from you and Yoda. I can get very emotional about these things, kind of like when my toys didn't work on Christmas morning or didn't have batteries, etc.
BTW, when I was first addressing this issue, my Sweetwater rep (Ryan) recommended a Mojave Audio 200 as a possible alternative. Tried it, and definitely prefer the Neumann sound. |
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Yoda117
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Joined: 20 Dec 2006
Posts: 2362
Location: Philadelphia, Pennsylvania
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| Posted: Mon Mar 24, 2008 2:15 pm Post subject: |
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| Rellis wrote: | | Yoda triggered the right thought, on the input level, and Ben brought the idea home with his specific recommendations. I reduced input to line level and increased the Output fader to "0" and the reverberation has been reduced noticeably, and that crisp Neumann sound is coming through much better! I'm beginning to think the 103 may actually like my voice. |
One major reason for the hate that Neumann gets on a lot of boards is that people record waaaaay too hot when tracking. Save it for the mixdown or (preferrable) during mastering. I've noticed a ton of home VO talent do that, which is a real shame.
I was chatting with Adam from Mercenary Audio this weekend over a bottle of Glenlivet (had to toss it out there... they drank the whole thing in less than 24 hours) and we talked about tracking levels. I have a new appreciation as to just how much it affects a VO after doing some experimentation of my own while the two of us were testing out converters for the room I'm building.
| Quote: | | I don't really use the Tube sound or Vintage Harmonics. |
Good call...
| Quote: | | I mainly use the Compression and Expander, both set at 12 o' clock. |
try tracking clean, or with corrective EQ. THEN use Compression and dynamic processing (dedicated devices are best IME, but try it... I think you'll be surprised by the results compared to adding it during tracking as opposed to adding it later after cleanup).
| Quote: | | I do appreciate the input from you and Yoda. |
I defer credit to Ben. I just made an observation.
| Quote: | | I can get very emotional about these things, kind of like when my toys didn't work on Christmas morning or didn't have batteries, etc. |
The tools are only as good as the craftsman, but I understand. I'm chucking aside a heavily modded Onyx 1220 for some dedicated AD/DA (keeping the mixer on side channel for utility purposes) because it's not giving me what I want... took me two years to figure that out and still realize that the device plays an important role on my desk.
Yeah, I think you'll be glad down the road that you didn't take the Mojave, but if you like the "Neumann bump" (that little frequency shift in the mids around 4.5-6K), then take a peek at Microtech Gefell. They're the original factory for Neumann, and are also highly recommended.
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Voiceovers by Gregory Houser
Philadelphia based Voice Actor
Blog - A man, a martini, and a lot of microphones |
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Rob Ellis
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Joined: 01 Aug 2006
Posts: 2385
Location: Detroit
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| Posted: Mon Mar 24, 2008 2:46 pm Post subject: |
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| Quote: | try tracking clean, or with corrective EQ. THEN use Compression and dynamic processing (dedicated devices are best IME, but try it... I think you'll be surprised by the results compared to adding it during tracking as opposed to adding it later after cleanup).
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I have to be honest, I don't really know how to "mix down" or "master" using a dedicated device (assuming you are referring to using the pre-amp versus doing it with editing software)
I can do it in Adobe audition. I assume it is better to do it using the Focusrite(???) If so, how, in a nutshell do you do it. I have never understood this.
I think it is explained in my Focusrite manual, (although from what I gather, they assume that you already know how to do that I think)...
and how do you "clean-up" the audio before adding the EQ and compression? Is there a post or thread that covers these things?
Last edited by Rob Ellis on Tue Mar 25, 2008 3:02 am; edited 4 times in total |
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Jowillie
Lucky 700
Joined: 20 Aug 2006
Posts: 714
Location: North Carolina
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Yoda117
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Joined: 20 Dec 2006
Posts: 2362
Location: Philadelphia, Pennsylvania
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| Posted: Tue Mar 25, 2008 9:18 am Post subject: |
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| Rellis wrote: |
I have to be honest, I don't really know how to "mix down" or "master" using a dedicated device (assuming you are referring to using the pre-amp versus doing it with editing software) |
I prefer using standalone EQ and compressors, but not everyone has one. I'd use the EQ and compressor of the vocal channel, while bypassing the preamp (which I believe can be done either through DI or, most likely through a bypass input).
| Quote: | | I can do it in Adobe audition. I assume it is better to do it using the Focusrite(???) If so, how, in a nutshell do you do it. I have never understood this. |
Now's as good a time as any to learn. I'm not a fan of Focusrite for this kind of task normally, but it's what you've got, so use it. Try sending the recording back through the vocal channel via a bypass or worst case, try using a DI and then processing the signal. Now do the same through audition.
Listen to both... since you should have used the same "clean" track to start with for both attempts, you should be able to identify which method you preferred. Go with that
| Quote: | | and how do you "clean-up" the audio before adding the EQ and compression? Is there a post or thread that covers these things? |
Depends on a few factors. You might need/want to use some gobos to further reduce the room noise. You might need to use some initial EQ as a "noise filter" to remove some unwanted noise at the 80-100Hz portion of the spectrum. You might need to remove some breathy issues or a trailing sound that you made when addressing the mic. It's general housekeeping really, and makes it easier to work with the signal when you're playing with the processing.
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Voiceovers by Gregory Houser
Philadelphia based Voice Actor
Blog - A man, a martini, and a lot of microphones |
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Rob Ellis
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Posts: 2385
Location: Detroit
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| Posted: Tue Mar 25, 2008 11:42 am Post subject: |
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I can probably figure this out, but still don't know exactly how to do it...
Okay, so I have recorded the track clean in Adobe Audition using the pre-amp, and my Mackie Spike interfaces it with my computer.
How do I put the signal back through the pre-amp, Is it through some of the inputs in the back of the pre? And do I send it from the Mackie, or what?
How do I record the new track as I'm mastering it with EQ and compression. Do I open a second window in Adobe, or do it in the the Multi-Track window, or what?
I know this is elementary to a lot of people, but in radio I just basically recorded the voice and then added SFX and music and that was it.
And I've heard latency can be an issue, and something about Wordclock.
I don't want to take up a lot of your time, if this is beyond the scope of this post. I have the basic audio engineering aptitude to learn this, but I fear that all my questions may be more than you bargained for!
But I do appreciate everyone's input and encouragement.
It's obviously a lot easier to just do it with the software. Is the quality really noticeably better running it back through the pre?
| Quote: | | You might need/want to use some gobos to further reduce the room noise |
My closet/booth is lined with Auralex, but it is rectangular, (approx TWO by EIGHT),
so it can get a little boxy in the corners. I have ordered bass traps for the corners, as well as the corners in my mixing room.... |
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Diane Maggipinto
Spreading Snark Worldwide
Joined: 03 Mar 2006
Posts: 6679
Location: saul lay seetee youtee
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| Posted: Tue Mar 25, 2008 11:50 am Post subject: |
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i especially like your last question, rob, and am curious myself.
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Yoda117
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Joined: 20 Dec 2006
Posts: 2362
Location: Philadelphia, Pennsylvania
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| Posted: Tue Mar 25, 2008 12:06 pm Post subject: |
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| Rellis wrote: |
It's obviously a lot easier to just do it with the software. Is the quality really noticeably better running it back through the pre?
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I'm about to run out for some supplies for my own VO-related chain, so answers will have to wait on most of the questions till tonight.
This one though I want to hit. The answer is that you're not going through the pre, but bypassing it and using the EQ and compressor. If you have to use the pre to get to the EQ and compressor, then your device was poorly designed. There should be a bypass.
As for whether or not it makes a difference, or is noticeable. Let me put it this way... there's a lot of outboard devices in studios for a reason. The reason is that while the processing that is available to you through tools such as PT or Audition has come a long ways, it's still not able to emulate what a dedicated box can do.
If anyone wants to try for themselves, take a Purple 77 and then compare it against the plug-in from Bomb Factory. On the same clean track, you'll notice the artifacts and clips-offs from the plug-ins, opposed to the smooth frequency response from the dedicated device. While they both sound similar, which one do you think will usually sound more natural?
I could have spent, and have spent $ on plug-ins such as Ozone and Izotope. That said, I'm so very glad that I spent the $ I did on dedicated EQs and compressors. They've been expensive, but I made the $ back when it came time to help fix the mixes that other VO talent and ad agencies weren't able to do.
It's also earned me a reputation that's earned me a lot of repeat customers over the years, so I have no complaints on that end either
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Voiceovers by Gregory Houser
Philadelphia based Voice Actor
Blog - A man, a martini, and a lot of microphones |
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Rob Ellis
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Posts: 2385
Location: Detroit
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| Posted: Tue Mar 25, 2008 12:12 pm Post subject: |
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| Quote: | If you have to use the pre to get to the EQ and compressor, then your device was poorly designed. There should be a bypass.
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There is a "process bypass" button on the front... |
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