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starter home studio setup?

Discussion in 'Recording' started by dj sym, Jan 12, 2013.

  1. dj sym

    dj sym Active Member

    First off, I'm very grateful to have found recording.org. I'm not going to blast any other forums, I'm just glad I found this one. Though I've performed a lot on stage as a spoken word artist with whatever mics they had (sm58s common?), I'm fairly new to recording and definitely a n00b as an engineer. So..the question at hand..

    Here is my current/imminent setup:

    Mac Pro 1,1
    RME Babyface
    Axiom 61
    pair of BX5s
    MXL V88
    AKG K99
    Pro Tools 9, Logic 9, Reason 5, BFD2

    For: recording spoken word/hip hop vocals.
    Voice types: raspy/high, smooth/strong mid,low, raspy/mid, airy/mid

    The monitors will be upgraded to Yamaha HS8's, but for now the BX5's will have to do. So I guess I'm mostly concerned about the mic and pre. I've read a lot about mic pres and that budget just really doesn't get it for this item. I've also read that the Babyface pres are pretty solid. I would like to get an external pre next (before the HS8's) and have read in a thread here that it can be ran as a 1/4 line in through the mic input on the babyface. All of the instruments will be virtual instruments, so I only need 2 channels. So far I am considering the Sebatron vmp2 (is it much different than the 4 other than # of pres of course?) because I really like the reviews about the coloring and especially the versatility. The reviews were about the vmp4, so that's why I asked.

    I know the room is important, and I may have some issue there. I'm recording in my bedroom, and I have some furniture that isn't going anywhere. A 5' dresser and a 6' compartmentalized desk are my big concerns there. I can treat the walls, but will the furniture still give me problems? If so, what mics can I use to compensate and accentuate the voice types? Would a dynamic be better than a condenser considering the room issues? The smooth voice will probably sound good through most mics, but my own voice (the raspy/high or airy/mid depending on what I'm doing) just doesn't cut through like I would like it to. If mic placement alone just can't get it done, is it a mic or pre problem? Or both?

    Also, I seem to become more nasal when recording with playback via closed back headphones (I even do that when talking on the phone with earbuds), but I'd like to avoid any feedback from only having one ear on (which helped, but my voice itself doesn't cut through like I'd like). Would it help to have partially open headphones or should I just suck it up and learn how to compensate? If so, any tips for doing that?

    Thank you all in advance :)
     
  2. RemyRAD

    RemyRAD Guest

    You seem to be a little confused? You're talking about acoustics, microphone preamps, microphones. You don't get what you want by just choosing microphone preamps and microphones. Careful processing has to be utilized also for good spoken word and vocal recordings. This processing will include using your high pass filter either on the microphone or on the preamp or in the software. It will also require careful gains staging and settings so as not to be overblowing your microphone preamp front end. Sometimes the microphone, particularly condenser microphones, put out a very hot output level. Many preamps need to have a pad switch on the preamp to deal with that hot output. The pad switch on the microphone is different from the pad switch on the preamp. Sometimes but rarely, both have to be engaged depending upon the sound source and its physical SPL. There are folks who are screamers and they need both. Spoken word not so much so.

    Then there is the issue of other equalization. Most condenser microphones and other dynamic microphones such as that SM58, already have a pleasant presence boost not requiring much additional equalization to be used.

    And then you want to limit or compress some of your dynamic range. This unfortunately also brings up the bad sounds of your room, ambient noise, HVAC, traffic, kids, dogs, babies crying, all that good stuff. And the only way to really eliminate that barrage of noise is through downward expansion a.k.a. noise gates. Sometimes noise reduction in software can also be utilized but it does not get rid of sounds above the noise level. That's only where a careful adjustment of the threshold of a downward expander or gate comes into play. The threshold dictates when it will be open and when it will close. Had a careful adjustment can easily track a vocal quite effectively. It's used in all movie dialogue, radio and TV commercials, computer games. We can't live without dynamic range limiting and compression when it comes to any kind of spoken or sung vocals. And it really tightens up that vocal sound. Proper adjustment of a threshold can also be utilized to reduce the sound of gasping breaths you are taking their augmented by the compression and/or limiting. Sometimes people will gate or downward expand before the compressor/limiter but I actually prefer it, personally, after the compression and/or limiting. And there's a good reason for doing it that way. The threshold cannot track constantly varying vocal levels. That's why I compress and limit after the downward expander/gate. It brings the local level up, makes it more consistent, increases the mumble and reduces uncomfortable sounding peaks. It then puts the vocal range into a more consistent level at which the downward expander or gate can be more effectively used. Though you do find plenty of outboard hardware compressor/limiter's with dates and downward expanders at the input such as the DBX 166/266 devices. And while that is still usable, I find it less usable that way for vocals. More effective for things like snare drum, bass drum, tom-toms, noisy guitar amplifiers, keyboard amplifiers, et cetera. Because their dynamic range is nowhere near that of the human voice.

    So if one does not use any hardware devices other than the preamp to capture the vocal, all of the above will be done in software. And you would have to chain your devices in a daisy chain, to have EQ followed by dynamic range modifications followed by the noise gate/downward expander. It is also interesting to note that depending upon what multi-tracks software package you are using, many of the downward expansion and noise gate features are actually created with a compressor plug-in. Many of these compressors allow for a GUI, indicating its operation. This GUI was also shown and displayed for the hardware units in their manuals. It's something I do all of the time using the stock dynamic range feature in Adobe Audition from its earliest days known as Cool Edit back in 1996. And it's actually within that GUI, you can create both the compression, limiting and downward expansion to be accomplished in a single step. And I've never quite found any plug-in that does what I do? You find some that indicate compression and limiting. Others are for a very hard gate which can sound extremely unnatural for spoken word on the human voice. And that's why I recommend downward expansion where the downward can be stopped at a level of your choosing. My downward expansion is usually only -10 to -20 db and no further down. This ensures that the microphone will always be on to some extent albeit low. This also ensures that you will not lose anything that way. I generally adjust my downward expansion for about the same amount as the downward amount of compression/limiting gain reduction, in use.

    Unfortunately, there is no one-button plug-in that will get this perfectly right for you. This is what also separates many of the professionals from the amateurs. One needs a full understanding of what needs to be accomplished audibly which only comes from a full understanding of what you're using. Much of this we all got ourselves through many hours, days, weeks, months, years of experimenting. We didn't have anybody to tell us where to set the knobs when we were growing up in the business. So this is all done by ear because we are professional ears. This is what we trained ourselves or were trained by others to do. Some of us I simply got while being a maintenance engineer at Media Sound, NYC. Oh wow! What an education sitting over the shoulder of Bob Clearmountain and others. I was already a good music and commercial broadcast engineer. This just verified and improved my own engineering. So what I learned only indicated that I had identified and listened to things correctly. I was already running my KEPEX-1's after my 1176 limiter/compressor before I saw Bob doing it that way. Or anybody else for that matter at Media Sound.

    When setting gates and downward expanders, generally, the fastest attack time possible should be used. No look ahead which many of these software compressor/limiters now feature. One would think that would improve the gating but it really sounds unnatural. The compressor is already kicking up the leading edge of the peak if you don't use too fast an attack time on the compressor and limiter. And that leading-edge peak will slightly make it through the noise gate better by not only opening the downward expander/gate better while still allowing a few of those peaks to pass. And you can get a very nice natural sounding vocal that is actually highly processed that way. And you don't even really perceive all that process because it just sounds right. I've been doing this to musical and spoken word vocals since 1978 and it makes a huge huge difference. It's something I cannot live without.

    That nasal sound. Yes that nasal sound. That usually indicates you are working your microphone much too closely. Sounds great in the headphones for sure. Lousy on any speakers. So obviously, that's not how it's done. You should be at least 6 inches away from your capsule. That's another reason why people started using lollipop pantyhose sticks. You can only get so close to the capsule then. And that's called proper mic placing. You're still going to need the high pass filter on the microphone engaged. This will make it sound thin on your headphones but oh so right through the speakers. And it will sound professional. And then you add your processing to complete the picture. So it's really kind of simple once you get the hang of it. It will change your world. LOL as it changed mine, so many years ago.

    Sometimes, at the very tail end, after all that processing, you may also need some dynamic sibilance control? Especially if you want a very aggressive and stylistic sound. The sibilance will become brutal without some additional dynamic control that has been frequency weighted in that sibilance area of 5-10 kHz.

    Sometimes, depending upon the room and the environment therein, condenser microphones especially the large diaphragm types, can sound extremely awful. This is where that venerable SM58 can really shine. It's a great vocal microphone. Put another foam pop filter overtop the metal ball and work it at least 3 inches from that additional foam filter and no closer. This microphones will reject a lot of that awful room acoustic signature. And the only EQ you might need other than the high pass filtering is a little boost around 12 kHz via a couple of db and not much more. It's limited frequency response actually works in your favor here. You don't need a microphone that goes down to 20 Hz. You don't want a microphone goes down to 20 Hz. And you don't need a microphone goes out to 20 kHz. Only noises in those frequencies that you do not need, do not want. And that 58 has a response of 50-17,000 Hz just perfect. This is one of the reasons why I love them so much. They sound great in the worst acoustic environments you can think of or find yourself in. And one of the reasons why they are so popular for recording virtually everything in the studio or live. It is only when I am in a very nice acoustic environment that I'll grab at some large diaphragm condenser microphones. If I want a condenser microphone and the acoustics are not ideal, I won't use a LDC but rather, I'll use a small diaphragm condenser microphone. Their off axis response to lousy acoustic environments will not be as obvious sounding if I need to have that condenser sound on a vocal. And there are plenty of SDC's that will sound much better in bad sounding rooms like you have. So I really think you should sell that LDC? It's not going to work right in that environment of yours. Sorry but that's the way it goes. Your recording environment is just not appropriate for these large diaphragm condenser microphones. They tend to accentuate all that you don't want to hear. It's an honest mistake that many people make based on advertising and marketing rhetoric.

    Keep posting any other questions regarding this technique until we can help you dial it in correctly.
    Mx. Remy Ann David
     
  3. dj sym

    dj sym Active Member

    Thank you Remy. I am very grateful for and humbled by all your input. You clearly didn't have to do that, yet you took the time to tell me information that I probably should know but didn't. I'm sure it will help me immensely and give me a much better understanding of what I have and what to do with it. So yes, apparently I'm very confused, but at least I can work toward getting it right now.

    When I was recording at a friend's before, I did notice that the condenser picked up everything. I also relied on LPF in an attempt to change my voice I guess instead of capturing it in the best light? I noticed it simply muddied the vocals, but perhaps because I didn't understand EQing? At the time, I believe I was using an Izotope plug via Pro Tools.

    I haven't received the babyface yet. Before I was using a Presonus Studio Channel with which did notice the -20db pad. Should be a few days. I don't see a pad on the pictures though, so is that something that would be managed via software with this? I don't want to ruin a brand new interface while recording. For any additional preamp I get later though, I will keep that in mind. Before, I was using a Presonus Studio Channel with which I did notice the -20db pad and usually kept it engaged. It was awhile ago and I didn't get to use it much. I don't recall being able to hear a difference with the pad on or off though, so as you mention, not as helpful for vocals? But when expanding the studio (lol with actual separate, well treated rooms) to accommodate recording live instruments, a physical noise gate will be best, but a plug-in is fine for what I'm doing now? So if I understand correctly, my gain staging while tracking will be EQ>compressor/limiter>noise gate/downward expander (Adobe Audition as a send?) because an all-in-one plug-in would not allow manual control over the gating itself. Followed by de-esser for sibilance? I think the attack I had on compression before was between 5-8ms, but as you said my mic placement was off so I should get more out of it now that I have a better understanding. And thanks for correcting me on my terminology. I thought mic placement was where in the room the mic was instead of how close the source was. I used to put the mic in the corner to try to limit room noise (the old room was awful too) before learning about letting the soundwaves dissipate over distance rather than putting foam right behind the mic.

    I would likely to eventually include r&b artists and even rock bands to record in a much better space, so I think I will hold on to the V88 for now and just add an SM58. Like you said, that's how these things go. I'll just take it as a learning lesson. I do like how condensers capture the small nuances in a vocal, so maybe I'll add an SDC some time after the SM58. Depends on how soon I can upgrade my space I guess. Also I, hopefully correctly, assume the high pass filter you were referring to is with the V88. I've been looking online and haven't found anything about one on the SM58. So when I get the 58 I would engage the high pass on the preamp? Or just when EQing through the DAW? Other than that, foam filter on the ball with the pop screen 3 inches away. Gotcha. Thanks again for your help Remy! It looks like I'm getting quite an education myself! That someone with your amount and level of experience would give that much input means a lot. Not blowing smoke up your ass either. I really do appreciate it. I have a lot to say and if someone doesn't like it, I'd much rather it be they just don't like what I have to say and not the quality of the recording.
     
  4. RemyRAD

    RemyRAD Guest

    The cool thing about microphones is that there are only a few basic flavors. And then some variations on those themes. So you really only need four microphones. Eight if you like buying them in pairs like I do. 2-LDC's, 2-SDC's, 2-Dynamics, 2-Ribbons. That's red blue green and purple. And you've covered most of the bases and also the bassists. And you can have large and small capsule dynamic microphones as well as long and short geometry ribbon microphones that can either be passive or phantom powered active. And if you have that, ain't nothing you can't record really well. And you really can't tell what you're going to want until you've lived with all of those guys or girls. I guess the ribbons are the girls LOL? Slender and corrugated. Instead of round and flat or some with a Buddha belly? Which our boy microphones. And you have to have both guys if you want to create beautiful music with them together. Or one from each opposite side. As in the creative world, you find it frequently working both ways.

    The microphone preamps that have those -20 db internal front end pads was the way that most microphone preamps were made. This would also mean that the operational amplifier, doing the actual amplifying, had to traverse a much broader range of gain. And the control that would trim this gain will change the characteristic sound of a more open loop or closed loop amount of negative feedback sent back to the inverting input side of the operational amplifier. And we could use these broad differences in the characteristic sound this would create. And the input could be easily overloaded necessitating the need for a resistant front end Pad.

    In another school of thought, greater consistency was had, Is the first stage microphone operational amplifier was preset for only 20 DB of gain. Well that's not enough sometimes. So after that first stage pre-amplification, there is a secondary adjustable buffer amplifier that is not adjustable over such a broad range of gain as to affect the sound of that amplification stage much. Consoles like the early legendary Neve 1073 modules were built that way and so are Mackie's. My slightly different Neve is like the earlier broader range amplifier allowing me to change the character of the sound.

    This is certainly not true of other highly coveted older pieces like API. That's a broad ranging that requires the pad. Because we paint with sound, I like that variable sonic option. To hell with consistency. I like being consistently inconsistent. Also those 20 DB preset microphone first stage preamps, don't need the pad switch. That's because only having to generate 20 DB of gain means it will generally never overload. And the next stage amplifier only needs to add another 10-30 on average should you have a low output ribbon microphone trying to capture that oboe solo from 30 feet away. Which most home rock and rollers won't have to deal with. And those are known as goof proof preamps in my book.

    So one is in the shallow end and one is in the deep end. Do you know CPR? Clipped Peak Restoration?
    Mx. Remy Ann David
     

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