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VO-BB - 20 YEARS OLD!
Established November 10, 2004
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louzucaro
The Gates of Troy
Joined: 13 Jul 2006
Posts: 1915
Location: Chicago area
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 Posted: Wed Apr 11, 2007 6:14 pm Post subject: |
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| Frank F wrote: | | Check how your inputs are driven and the output power of your soundcard if these typcial results are not achieved. Many soundcards or pre's are very low level and force using "normalization" or "compression" to achieve the audio level desired. Both of these processes may result in undesired noise and/or artifacts. |
Frank, any advice on how to check / correct this type of situation? I'm struggling a bit with this currently as I have my setup set up but it seems to produce very low levels. The chain is:
Mic > PreSonus Bluetube DP > E-MU 1616m (breakout box connected directly to the sound card via an ethernet-type cable).
I'm super new at all this, and I'm not sure how to set the mic gain & tube drive on the PreSonus vs how to set things on the E-MU breakout box. I've also tried both with an XLR and a 1/4" cable going from the PreSonus to the 1616m to see if that made a difference, but honestly I couldn't tell.
Sometimes even when listening in the headphones, it sounds distorted, but then when I check the resulting file in Audacity, f'rinstance, the peaks are at around 15% - 20%...sometimes not even that high.
The 1616m has a DSP Mixer thingie that loads when Windows starts and I believe it's controlling hardware effects that the product has built-in (there are around 600 of them I think).
I played around a little bit with using the default "Compressor" effect from that batch of stuff, which definitely helped, but I still can't seem to get levels that are high without sound that sounds distorted.
Reading through the various manuals that came with all my hw/sw, to be honest, nothing is very informative from the viewpoint of somebody like me who doesn't know what they're doing yet 
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Lou Zucaro
http://www.voicehero.com
"Well, yeah, there's my favorite leaf!" |
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Frank F
Fat, Old, and Sassy
Joined: 10 Nov 2004
Posts: 4421
Location: Park City, Utah
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| Posted: Wed Apr 11, 2007 6:45 pm Post subject: |
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There are some possibilities... much too long and too single user dependent to discuss on the VO-BB directly. Please check your PM for some instructions.
Frank F
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Be thankful for the bad things in life. They opened your eyes to the good things you weren't paying attention to before. email: thevoice@usa.com |
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louzucaro
The Gates of Troy
Joined: 13 Jul 2006
Posts: 1915
Location: Chicago area
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| Posted: Wed Apr 11, 2007 7:01 pm Post subject: |
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Grazie!
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Lou Zucaro
http://www.voicehero.com
"Well, yeah, there's my favorite leaf!" |
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asnively
Triple G
Joined: 17 Jun 2006
Posts: 3204
Location: Los Angeles
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| Posted: Wed Apr 11, 2007 7:47 pm Post subject: |
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| sdelgo wrote: | Proper recording level is -18dbFS which equals 0dbVU.
Your signal should average 0dbVU from your preamp which is -18dbFS (digital) on your AA program.
looks like Amy wins. |
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Last edited by asnively on Fri Feb 27, 2009 1:57 am; edited 1 time in total |
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MelissaVoicer
Guest
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| Posted: Thu Apr 12, 2007 5:51 am Post subject: |
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| TheVoiceOfBob wrote: | | I used to normalize, but I started messing with the "Compander" setting in the "Dynamics Processing" in Audition. That gives my voice level some added punch, brings it up a bit, and drops the floor down some to where the silent portions are truly SILENT. |
Hi, I, too, would love to boost my levels..and am using Audition (currently 1.5) Could someone expand on what Bob said here? And pleeeez speeek sloooowly to the teknikly challenged. 
I'm currently getting about-6, BUT that is resulting in hot cilps on "upbeat" reads unless I pull back from the mic...which I don't want to have to do (dont fear the mic..the mic is your friend....)
Thanks! |
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sdelgo
Contributor IV
Joined: 04 Dec 2006
Posts: 143
Location: Milwaukee
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| Posted: Thu Apr 12, 2007 7:52 am Post subject: |
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Melissa,
Companding is COMPression and exPANDING and audio signal. Basically the compander is used to compress an audio signal that is too dynamic to fit within the target's bandwidth and then expanded upon arrival. This is only one use... others include the DBX and Dolby C noise reduction schemes.
As for vocal applications it would give a boost to voice and squash the low parts BUT you run the risk of "dropout" usually because the make-up gain amplifier can't keep up with the peaks and valleys of a vocal recording using it "live". Software versions take the total program material and applies it's algorithms to the material so you might have more control and a less chance of "dropout".
As for your problem with the -6db and hot reads... try this
1. record your material at or near -18dbFS (that would be the meter on your AA or 0dbVU the meter on your mixer)
2. run your corrective eqing (control any unwanted or wanted freqs)
3. compress the material as needed ( to control any wild peaks- compression is not used to bring the material up in volume, that is actually a by-product)
4. Run your shaping eqing(to get the overall tone you want)
5. Here is where you boost the sound... Run a limiter that peaks at
about -15dbFS to -12dbFS which is +3dbVU to +6dbVU and that boosts the quiet parts of your material. +3dbVU to +6dbVU is plenty loud (0dbVU is line level for recording gear)
This is just a basic recording method... if you use reverb, echo etc... you'll have to adjust the levels to keep everything low in tracking. Remember the more tracks you use the higher the program material is in dbFS/dbVU.
There isn't one setting for all recordings (that would be too easy) there are presets and those should be used as starting points.
Steve
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MelissaVoicer
Guest
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| Posted: Thu Apr 12, 2007 8:30 am Post subject: |
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Steve, thanks, and I think I understand about 20% of your post!  I've always heard it's better to get the levels and compression you want going in...and not fixing it after by software. Right now I have A LOT of peaks and valleys on my raw...and I have to normalize to 95%...then usually hard limit to -.5 and then reduce the amplification by 3db to get a nice, more "rectangular, flat" wave form. I know that's not ideal...and I've been told my compression settings are off as evidenced by my voice not sounding the best it can and by the shape of my resulting waveform.Here is a sample of uncompressed and compressed audio coming out of my system.
http://www.melissavoiceover.com/AudioFiles/Samples.wav |
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sdelgo
Contributor IV
Joined: 04 Dec 2006
Posts: 143
Location: Milwaukee
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| Posted: Thu Apr 12, 2007 8:47 am Post subject: |
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Melissa,
Don't judge your final sound by the waveform, that is not the way to work.
Instead of normalizing (which you should never do in recording-it's just a volume knob) when you have your peaks and valleys compress the material to tame the peaks and even everything out a little bit then limit it to bring up the volume.
Use minimal compression "on the way in" if any at all.
Tell you what... send me a sample recorded at -20dbFS to -15dbFS so I can hear what it sounds like.
Steve
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you'll always have something on your plate... if you keep your bearings straight.
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Yoda117
M&M
Joined: 20 Dec 2006
Posts: 2362
Location: Philadelphia, Pennsylvania
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| Posted: Thu Apr 12, 2007 12:19 pm Post subject: |
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| JeffreyKafer wrote: |
Yep, me too. I send a true dry and then a compressed. They always use the compressed. |
Same here. I freaking love these Trakkers for compression, as they can give me as much, or as little as I want. It's a true swiss army knife of compressors and yet does all jobs really well.
/end shameless advert for Crane Song products
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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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mediaking
Contributor
Joined: 20 Aug 2005
Posts: 33
Location: Earth...this week
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| Posted: Thu Apr 12, 2007 2:27 pm Post subject: |
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Correct me if I'm wrong, but shouldn't you send the client a recording with no EQ and little, if any compression, so they have complete control?
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Hear It Right. Now. |
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Yoda117
M&M
Joined: 20 Dec 2006
Posts: 2362
Location: Philadelphia, Pennsylvania
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| Posted: Thu Apr 12, 2007 3:21 pm Post subject: |
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| mediaking wrote: | | Correct me if I'm wrong, but shouldn't you send the client a recording with no EQ and little, if any compression, so they have complete control? |
Yep.
Jeff and I both send our respective clients a clean take, but also take the time to use a bit of compressor and/or EQ to suit the copy based on our experience. If the client likes our interpretation of things, awesome... if not, then there's the clean take for them to play with till their heart's content.
While I can't speak for Jeff, my opinion and experience is that the clean take is the best one to send to the client, but he and I were also remarking that more often than not, the processed track that we sent in addition is the one that gets used.
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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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Jeffrey Kafer
Assistant Zookeeper
Joined: 09 Dec 2006
Posts: 4931
Location: Location, Location!
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| Posted: Thu Apr 12, 2007 3:32 pm Post subject: |
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| Yoda117 wrote: | | ..but he and I were also remarking that more often than not, the processed track that we sent in addition is the one that gets used. |
In my limited experience, Yep.
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Jeff
http://JeffreyKafer.com
Voice-overload Web comic: http://voice-overload.com |
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