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Question about using too many vocal tracks?? (Vocals are muggy..)

Discussion in 'Vocals' started by Maverick87, Dec 21, 2011.

  1. Maverick87

    Maverick87 Active Member

    Before I get into details, heres the gear that I use:

    Computer:
    eMacines W2686
    80GB Harddrive
    512 MB DDR(????)
    1.60 GHz, 992 MB RAM
    Windows XP
    AMD Athalon XP Processor

    Microphone:
    Sterling Audio ST-51 Condensor Mic

    DAW:
    Magix Music Studio 12 Deluxe

    Audio Interface:
    M-Audio Mobile Pre USB

    I record rap/hip hop music and vocals only, yet I stress a lot over getting the sound I'm looking for. Typically we see a lot of verse/chorus/verse n so on with rap music, so heres how I normally lay out my vocals:

    1 Main verse track dry and centered
    2 Main verse tracks, volumes slightly lowered from centered track, I usually throw a slight delay to make it wider and pan these out -35 and 35
    2 Overdub tracks that accent phrases and words I want to stand out, usually panned further at -45 and 45
    2 Adlib tracks to add background and fill gaps and extra things for creativity panned at -55 and 55, these usually have a 1/4 delay blended in
    All tracks are compressed and EQ(Is there a certain order i should add these as i mix?)

    For hooks/choruses I typically do the same layout, but i pan the tracks even further because Ive heard this is good to do to make it stand out, and add a chorus effect on the two panned main tracks.

    I know it's hard to understand without hearing a sample of our music, but it sounds like the vocals are just too muggy. I prefer a thicker sound on my vocals in particular(Im in a duo so I record my partner as well), I dont like how my voice sounds dry and like it layered. We record in Mono so I'm trying to find a way to effectively add depth and make the whole sound wider without overcrowding it.

    I've tried to play with the compression, adding more or less, and the EQ, but i have a feeling it's about how Im laying out my vocals? Any tips on if this is why and/or how i can manage thickening my vocals clearly and effectively?
     
  2. RemyRAD

    RemyRAD Well-Known Member

    You may in fact not be utilizing enough compression and/or limiting? When you say muggy do you mean muddy? Nondescript? Too crowded? Unnaturally layered? Or is it because it's just you and your partner times 6? This may also be caused by being the same distance from the microphone each time. This is a technique that I utilized when overdubbing instrumentalists & group vocalists. Like Rudy Van Gelder, I leave the microphone where it is and move the people further away from the microphones. This creates different sonic architectures along with the proper timing differentials that just works. I don't just add delay but may in fact add some delay utilizing the above mentioned technique. For instance, when I recorded string, Woodwind & brass tracks, I usually didn't have more than 6 musicians at a time. I would set up two rows of chairs. The front six & the rear six. The microphone stayed put. I would track the musicians once. Then I would move them to the rear seats and track them again. This sets up the proper timing differentials along with the proper proximity of an actual section. I would do the same for the vocal tracks but instead of being seated, they would be standing on one line and then have them move back to the next line without moving the microphones. It's crazy how well that works. And I also didn't bother to give them any headphones. Instead, the rhythm tracks would be blaring out of the studio speakers. On the second pass of overdubs, I would take that second track and invert phase. I would then combine that with the first track, together, in Mono. This would cancel out the blaring studio speakers almost completely. While at the same time it also cancels any acoustical aberrations the room had. And for stereo purposes, I would have to do the above times 2. And you don't bump nor move the microphones at all otherwise none of this would work. And that's a specific professional technique I utilized with great results. Impossible results. Remarkable results. And what no one else was even attempting to do.

    Don't forget that SM58
    Mx. Remy Ann David
     
  3. Maverick87

    Maverick87 Active Member

    This may be an error on my part, but I use a very mild compression and a limiter under 'dynamics' on my program as I'm recording the tracks to keep the waves more even. After I record it I add more compression( and the EQ and other effects. Should i just record dry and add everything later? I'm just tryin to balance and make sure my initial recording turns out good so I have something to work with later.

    The last song I recorded, I tried lowering the levels of compresson both before and after the recording, and it sounds a little more natural. Usually when i start to raise the gain on the compression or ratio the vocals can start sounding very strange n processed sounding, and also fuzzy and distorted, its hard to describe. But thats the reason I thought I was overdoing it. Ive also started lowering the gain on the MobilePre so I can avoid the peaking and distorting more.

    Its hard to pinpoint what I dont like about it, The vocals seem like theyre too loud yet unclear. If i lower the volume to where it should blend with the music better, they're not as clear, yet if I raise them enough to be clearly heard, word for word a bit over the music it sounds like it's overbearing and too much 'in your face'.

    I'm still learning and have been going through a long time of trial and error, and it can be very frustrating lol... I feel like I have the concept of what I should be doing, but am missing the details or knowledge to do it correctly for the best sound I'm looking for. I want the vocals to be clear and crisp and blended in well.

    Maybe I can send you a few samples next time I'm in the studio and you can listen and tell me what you think is making it sound like that, and can give me some pointers if you dont mind. I'm sure you guys get that a lot..lol. I'm very critical of my own music and even my voice, so it's uncomfortable for me to just show people if that makes sense.

    I literally just started experimenting with stepping further back from the mic when I record the overdubs, do i turn the mic volume(preamp gain) up since im stepping back further? Or leave it at same volume. If i step too far back the waves show up so tiny that I'm not sure if it's good or bad. I've seen the invert phase option but have no idea what it does, and dont like to mess with things that I have no idea about. What exactly does it do?

    When looking at my track layout, does it seem like its too many tracks? I like to have an equal left/right ratio, excluding the centered track because I'm not sure if its good to use one type of an effect on one side and a different for the other. Is this good or bad?

    I'm also just beginning to learn the concept of aux buses, which was my first post in this forum. Maybe when I get that part down, it will be a much better way or managing how deep the compression and eq's go? Is aux busing vs. track by track effects a huge difference in sound quality? Or just more convenient and easy?

    And also am I using the effects properly? Should i add the delays to more tracks or less? The ones pan furthest or is where i have em just fine? Should i always leave the centered main vocal track dry or should i try a reverb on it?
     
  4. RemyRAD

    RemyRAD Well-Known Member

    I think we would all love to hear what you are doing. Plenty of people post such examples and have benefited from our feedback.

    Sure, you still want to record your levels at a decent record level volume. Naturally, when you step further away from microphone, a little more gain is usually necessary. This is why a lot of folks recommend recording at 24-bit as opposed to 16 bit. But that doesn't change the internal noise floor & head room of the preamp. It gives you more dynamic range to process with from lower levels that will not exceed the clip point. So sometimes when adding any kind of processing, you sometimes have to lower the level, after recording, before processing, to keep it from peaking out & clipping. That's provided that your record level was already within a good usable range. Not too much & not too little but averaging between -12 to -18. That provides you with 12 to 18 DB of headroom for natural peaks in your vocal range & processing. Even on analog recorders like we used to use, we tried to record around 0 VU. Depending upon the sound source, one could actually slam the meters without undue distortion such as for drums. That didn't quite work the same way for vocalists. And even when we were recording at 0 VU, with a meter that only goes up to +3, we all knew we generally had up to +15 hitting the tape before it would turn into doggy Doo Doo. So our meters and percent recording practices basically had goof proofing already built in to the recorder. Digital of course is different. When your meter in digital goes up to 0 DB FS (full scale), that's it, that's all, there isn't anything more above that one can print without horrible sound & clipping. And this typically also occurs when wild kinds of effects are added. They can produce peaks well beyond what the digital medium can support. So it all becomes a tricky trade-off and balancing act. You're just losing your footing on the high wire. Thankfully, nobody gets killed from bad audio.

    You indicated that you are actually utilizing compression while you are recording? Not sure how you are doing that with software? I'm not sure that you are doing that with software? You may be simply monitoring that from software without adding any dynamic range limiting while recording. Most software packages don't allow for that. Some can most don't. When folks like myself track vocals, I come out of the microphone preamp, with a little high pass filtering (low-frequency rolloff switched on the microphone) and into a hardware compressor/limiter. It's from the compressor/limiter that it then gets routed into the computer audio interface or my standalone, purpose built, digital recorder. And in that respect, I generally go fairly lightly. I also do not set the attack timed to be too fast nor the release time to be too fast. This allows the vocals to have what appears to be more dynamics while I have limited the dynamics to some extent. On playback, I may add additional EQ & more compression and/or limiting? This time, frequently through software. This is the hybrid fashion in which I work. Not everybody has that capability however. So in situations where I cannot add any compression or limiting during recording, I must make certain that my gain structure is such that I do not peak out. And then with that recording, I can include some compression/limiting & EQ and not necessarily in that order, when I begin mixing. I've frequently worked both ways and both ways are fine provided proper attention is paid to record levels. Being too low a level simply results in the slightly less resolution which isn't always horrible. Distortion from too much level, is horrible.

    With most recordings and groups I work with, I'm not into a whole bunch of gobbledygook effects. Because effects are just effects and everything does not need to have effects. A tad of reverb is frequently nice. Some flanged echoes can also be quite cool. Along with some oscillated pitch variations on the very light side. Short reverbs can frequently be utilized more to create an acoustic space than a long trail as heard in a church. And when I say short, I mean it does not come off sounding like reverb at all it's so short. It simply creates the effect of a moderate sized room that will add depth without drowning in reverb trails. Oh no! I think I just stepped in some? Dammit! Now I have to get this reverb off the bottom of my shoe.

    Liquid shoe polish or paste? What do you recommend?
    Mx. Remy Ann David
     
  5. Maverick87

    Maverick87 Active Member

    Very interesting. And how would the invert phase come into play and how does it help? For me reading what it is and understanding the change n difference it makes are two different things. And isn't there something you have to watch for with the eq and phase?

    On my software you can activate the compression n limiters and all that before recording, I always assumed it was being used, but maybe like you said I'm just monitoring it?

    With the compression, what are the milliseconds you would recommend for attack n release?

    Should I leave the main centered vocal track dry and get the delay effects n all that out of everything that's Panned? Or are there certain effects that are very useful for the lead vocal?

    Id love to hear any other tips for recording and applying effects and microphone techniques. Im just now learning and trying to comprehend aux and submix budding effects. I pan on trying to eq the separate vocal tracks differently and making the backup tracks more different sounding.

    Just discovered that people aux different level eq's and somehow combine two tracks into one n all that. It's really hard to learn this stuff with nobody to show you in person or people who don't use the same software, but with the right people breaking it down for me, I'm confident I can learn it successfully.
     
  6. RemyRAD

    RemyRAD Well-Known Member

    Equalization in and by itself can cause certain types of phasing issues by virtue of the way it works. It's slowing down and speeding up different frequencies or is that slowing up and speeding down? Well both. I'm only saying that to confuse you more. Inverting the phase is a whole different issue. Speakers can be out of phase, microphones can be out of phase, microphone cables can be wired out of phase. And phase cancellation will result. This cancels some frequencies but not others or it can cancel everything. It's used sometimes for enhancing sound and at other times to eliminate sound. Sometimes accidents happen and your sound gets all screwed up because of it. The only hard and fast rule to remember is, if something sounds screwy, see if inverting the phase might help it. On an analog audio console there is frequently a simple pushbutton on each input to effect that. In software, it's a simple command from one of your menus or virtual mixer window.

    Regarding compression and limiting, in all likelihood, it's just a monitoring function and not recording that. That has the ability to make it sound better in your headphones without printing it to your track. So maybe you're just not using enough upon mixing down. There is no hard and fast rules for attack and release times except for your ears. That's why we all love twiddling knobs and dials to dial in just the right sound. Sometimes vocals need fast attack and fast release which equates more to limiting. Utilizing slower attack times and slower release times equates more to compression and/or " automatic volume control ". As you increase the ratio of compression/limiting from nothing at 1:1 to light compression at 1:1.5/1:2/1:4 to harder limiting at 1:8/1:10/1:20 up to 1: infinity. The amount it's compressing/limiting is adjusted by your threshold control when/where you have set your threshold level to begin its operation. From just a couple of DB gain reduction to 15-20 DB or more. A complete explanation of information regarding compression & limiting is probably in your help menu already. As most of your questions are probably in your help file and/or manual to your software. This function isn't utilized much in fine arts applications such as orchestral, operatic. But it's used a lot in contemporary pop music because of the nature of the beast.

    Panning is very similar to the balance control on your stereo system. Where sound can be adjusted from left to right and anywhere in between. When you are panning a single track in your stereo image is referred to as panning and not balance as in the balance of a stereo recording. Panning to Mono means it's in the 12 o'clock position. Where are panning to the left would indicate it is from the 7/8/9/10/11 o'clock position. To the right would be 5/4/3/2/1 o'clock position of the " panoramic potentiometer " or, " Pan pot ", hence, PANNER. So it is a stereo mix positioner of independent track sources. So utilizing multiple tracks it allows you to create a stereo panorama.

    Adding delay effects would include, echoes, flanging, phasing, reverb, individually or combined for cool sounding effects. That's another item with no hard and fast rule other than your ears and what kind of sound you are trying to create.

    Import and microphone technique to know is that all directional microphones have what is called " proximity effect ". In simple terms, what this means is, the closer you get to a directional microphone the greater the bass response will become. This is useful sounding on some instrumental recording but can actually turn a vocal into mud. If you're 2 feet from the microphone, there won't be much proximity effect but you'll pick up more of the room and extraneous noise. Getting your vocal closer to the microphone will increase the bass response too much. So many microphones have their own base rolloff a.k.a. high pass filter switch. So it attenuates the bass frequencies below 80-100 Hz while allowing the high frequencies to pass freely hence, high pass filter. This is an important factor when recording vocals especially. The problem with this is that all of that bass response sounds really cool in your headphones and really awful on loudspeakers. That's why speakers are so important when mixing and headphones lie to you. All headphones lie to you in this manner.

    You are already understanding some things about microphone technique without yet completely understanding what you are doing. So you're getting close to the true meaning of microphone technique by how far you are from the microphone and how much proximity effect the microphone develops. Omni directional microphones have no proximity effect at all but they pick up the sound from all over instead of what is directly in front of them. That's why directional microphones are more popular to use than Omni-directional patterns of microphones. You don't have to worry about that as your microphones are strictly directional known as Uni-directional, cardioid pattern. So they don't take up much sound from behind them since there is generally no sound behind them you want to pick up.

    Equalizing each vocal track differently can impart a different tonality for each vocal track. Believe me, this is not necessarily what you may want either but you might. Only you can determine that by listening. When we are putting together choral vocal like parts from a single vocalist, we don't necessarily equalize them differently from each other. Instead, what we might do is record each vocal part without any equalization or compression/limiting. When you begin to mix the tracks together, you may combine all of those vocal tracks to a single compressor/limiter. This technique varies with software packages. For instance, you may not take those vocals and assign any of them to the stereo mix. You might just take those vocals and send them all to the same auxiliary send or bus. You would then take that auxiliary bus and put it through a single compressor/limiter to taste. That compressed group of vocal tracks would then be brought into the stereo mix and placed in the stereo panorama where you want them. This could require many tracks and many layers performed in this manner depending upon your software package. So it's really vital that you read your help files more than once. It takes a long time to get this stuff under your belt.

    There are also helpful books available that go into more depth explaining a lot of this from simple home recording procedures to complex and in-depth ProTools. Many of these multitrack audio programs are referred to as deep. That generally means that they have so many capabilities, it could take you years to become completely proficient & fluent with them. It's not an overnight thing. Some folks even take up to 6 years to attain a Masters degree. I'm not one of those people. I've been in the business so long, I started before there were any kind of college courses for this stuff. So many of us had to rely upon trial and error and if we were lucky, maybe had a mentor to help us along. We are all mentors here at Recording.org because that's why we are here to help folks like you as best we can. And for FREE! Of course the old adage that something for nothing is worth nothing is not always true. We like to give of ourselves and help you make the best recordings you can. We love sound and obviously so do you. This may also be similar to your first car? No 16-year-old should be given a new car, ever. That's because, we already intrinsically know that it will likely get banged, nicked, crunched it in some manner fashion or form. And even years later, hopefully you'll never experience anything more than a little minor fender bender? We all still have fender bender's in our audio on occasion. Thankfully nobody gets killed with bad audio except perhaps their business doing audio. So unlike auto insurance, the more audio fender bender's you have, the lower your rates go since folks may not want to record with you. And if you lessen your audio fender benders, you can charge higher rates of your clients. So audio is really the inverse insurance rule to automobiles LOL.

    The sky's the limit when you finally start to understand your software and what you want to create/attain. I've been doing this over 40 years. I was quite accomplished at this more than 20 years ago. It's those first 5-10 years that's the roughest without any formal education, mentoring or help. So you must read everything you can get your hands upon. There wasn't much to read about this 40 years ago I can safely say. Nothing more than what the equipment was or what it did which certainly didn't give you any real answers as to how to use it properly. It was mostly trial and error for many of us. But that's what makes you strong and the best. Everybody today requires spoon feeding it seems. That's kind of understandable given the heavy involvement of the utilization of computers and all that they can do. We only had rudimentary equipment back in the analog days before anything digital hit the market. So learning the absolute basics without worrying about gobbledygook effects will make you a finer engineer than playing with all the gobbledygook the computer has to offer. That means microphones, their use, their selection, their placement, their types without any use of equalization, delay effects, reverb. And how musical sounding your preamps might be. We didn't even think of that much 40 years ago. We just wanted to make sure that we plugged the microphone in and saw the meter move and press the record button. We kept doing that until we got that procedure right. As we grew so did our equipment & opportunities. So you are taking baby steps right now, as you should be.

    I'm still learning at 56 years of age
    Mx. Remy Ann David
     
  7. Maverick87

    Maverick87 Active Member

    One thing that I can tell is strange about the vocals is that if I listen to the songs at a low level, or positioned not directly in front of the speakers, i can still hear word for word whats going on, but up close and really loud it sounds too in your face and it sounds like some the frequencies maybe be boosted too much n other not enough possibly.What can remedy this imbalance?

    I definitely have to do some more playing with the EQ, since the program has so many different options, though i prefer the EQ knobs on the mixer feature. Not quite like analog but similar i suppose..lol. Maybe invert phasing will help.

    I'm gonna experiment with the compression settings as well and maybe I'll be able to figure out whats too much and too little. I tend to rely a lot on presets when I dont know exactly what each little knob and number means and does. Maybe this, and/or busing the compression with whole groups, in this case the whole verse or chorus, will be the change I'm looking for. Like the EQ, theres also many different compressors, so its hard for me to tell which one is quality and which is garbage. I use a plugin that the program has where its set up and resembles a hardware compressor.

    I use a high pass filter in the program track by track. Around where should i be rolling off the low frequencies? Maybe more since i layer the tracks so deeply?

    Im considering cutting back the amount of tracks I use. But my dilemma is that with three full verse/chorus tracks, i can keep one centered and do whatever i do to the two panned tracks equally. If i cut it down from 3 full verse/chorus tracks down to two, would i slightly pan the two full verse tracks slightly left and right to maintan the stereo effect(i know its just a mono stereo spread n not truly stereo) and adjust them equally? Or would I keep one centered and put some kind of a delay or reverb on the other? I can't imagine just panning one of the full verse tracks. Maybe i can use two full verse/chorus vocal tracks for verses and three for choruses to keep it different and thicker, but I'd still like to know how to do it effectively and not set myself back.

    Ive got a bit of improvement just by experimenting and focusing more on the recording process itself and mic position/gain and technique and all that, before applying the advice youve given me even, and the vocals sound more natural so I just gotta keep experimenting. Its just good to know if Im moving in the right direction and where I need to look for adjustments.

    But at this point my biggest obstacle is the imbalance of clarity/volume and muddyness, so any advice on that would help a ton. Since every post of mine turns out to be a novel, ill sign off by saying thanks for all the advice and input so far. I'm sure I'll post some of our songs soon and get opinions. I suppose most of us are a little insecure of getting negative criticism, especially in a genre like rap thats almost frowned upon by "true" musicians(I play guitar, bass, and drums and have played in bands, so I totally see both sides of the argument).
     

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