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Thread: Help choosing mic for unusual studio setup

  1. #11
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    Microphones should last many years and the Rodes have only been around since the mid 90's so they'll be less than abouot 25 years old which means they are mere youngsters compared to many mics still in regular use. It sounds like you could be using them in a damp or dirty atmosphere. Do you cover them between sessions? I'd look into having them repaired as they are still reasonable mics.
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    I recently purchased a new microphone and tried it out as soon as I opened the box it came in. I plugged it into my Zoom H5 number 1 socket.

    I noticed a little background noise as I increased the gain. So I changed the microphone lead and the background noise dropped. I then plugged the lead into my Zoom H5 number 2 socket. The background noise basically disappeared.

    So not only was my lead or connector plug making a difference but also the different sockets and pre amps in the same device made a difference as well.

    Initially I suppose we all blame the most obvious but sometimes problems are a lot more involved than what they seem.

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    Quote Originally Posted by einstein magoo View Post
    Not sure we have a good picture of your recording situation. At least I do not.

    You mention recording conversations and say you are down to one mic. Does that mean you are capturing a discussion in a room with just the one mic and two or more people are contributing to the conversation?

    Reason I am asking this is, a dynamic will capture less background noise because the performer or person speaking is able to get his/her mouth close in to the microphone. If you have two people having a conversation, it would be best to have each person speaking into their own microphone up close. A dynamic won't do what you want if you are just going to have your voice actors sit around it and speak toward it, but still be two or three feet away. Catch my drift? I may not be explaining this well.
    When we had two microphones, each speaker had their own microphone. Now that we have only one, we just do one speaker at a time. They just read their lines and I assemble them into a conversation in editing. We do not have more than one person using one microphone. The speakers are always right next to the microphones with our current setup. The "recording studio" is basically a closet, about 1m x 2m.


    As for the cause of the background noise, we have been using these mics in this studio for years now. We have tried switching around the rest of the hardware, and we have confirmed that it is the microphones themselves that are the source of the problem. We do not have a complicated setup. We are not recording complex sound. We just have people sitting in front of the microphones reading some lines of text directly into them. And the sound was perfect in terms of quality for years, up until that first microphone basically died. Nothing else had changed, just the microphone (after years of being bumped around) stopped recording sound properly and started introducing all kinds of awful static, distortion, and other problems.

    The equipment is not covered when we're not using it, but it is left in this tiny room with the door closed. It only gets used about once per month, for a few hours. It's certainly possible that over the years, the little room has gotten humid at points, at least during the summer (it's certainly bone dry in there now). And as I have said, they have been moved between offices several times, and they get bumped into and occasionally dropped, so they haven't been treated as gently as they should have been. I don't necessarily think they've worn out because they're too old, but they have not been handled gently and they are not working anymore.

    I don't doubt that the Rode microphones are good quality, but the time, effort, and money that would go into repairing them is not worth it if we can just buy new microphones that meet our needs better. Even if we could get these ones working again, they would still pick up just as much of the background noise (and even if the company agreed to the time and expense of properly soundproofing the booth, we don't even have the right to make those kinds of changes to our rented space). We won't throw the old ones in the trash or anything - we'll either keep them around and investigate them more in the future, or give them to someone who can make use of them somehow.

    I still get the impression that dynamic microphones would be more suitable for these conditions. Keeping in mind that the detailed quality of the sound is absolutely not important, as long as it's clear. Again, these CDs we're making are only going to be used by teachers in classrooms on cheap, old CD players with terrible sound. We're not exactly recording music or dramatic radio plays or anything. It's just "listen to the two people talking about where they'd like to eat lunch and answer the questions" type stuff.

    Does anyone else have experience with the Shure SM58? Should that be sufficient for this type of recording? Will it be okay for people with quieter voices (I will try to teach them to speak up, anyway), or does it require a strong pre-amp of some sort? Currently we are just using a Lexicon Lambda audio interface (which is as old as the microphones but still seems to work perfectly) and I usually need to turn the microphone input up to max, or close to it, to capture some people's voices. If we require any more pre-amp than that, I will have to factor that in to the purchase and my approved budget for this.

    Thanks again for all the input!

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    Why not buy another Rode? 10 year warranty 150.00.

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    As I said in a previous post, they are outside my price range. I have a set budget granted to me by my boss. If I can get something cheaper that will meet my needs, it leaves me the option of also buying some other equipment that could use replacing. And I do not want a condenser microphone this time because the recording space just has too many problems with noise, and we don't need that level of quality.

    I get that a lot of people are saying Rode microphones are great, but seriously, I don't need anything fancy. I just need something that will work and not pick up too much background noise. All I'm concerned with at this point is whether a dynamic microphone will need any additional amplification (which would mean I need to buy something else to amplify it) or if they will be fine on their own with people speaking in a normal voice.

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    Possibly a decent lapel mic? Clippy Microphones - micbooster.com

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    Hi Aria and welcome. First of all DO NOT get a microphone with a switch!
    One reason, people will fiddle and another is I have found them a source of unreliability over the years.

    Now, you say you needed the Lambda's gain close to max even for the Rode? The AIs were not noted for high gain and low noise (but to be fair, most early AIs were poor in that respect) so with a dynamic of some 20 dB (x10) lower sensitivity you are going to struggle. The usual solution is an in line, phantom powered preamp such as the Fethead but that will add around 80 per mic to the budget. I have a suggestion, well two actually.

    There is a very decent dynamic mic called the Prodipe TT1 (French) and they are about a third the price of the SM58s. They have slightly more output but not nearly enough to compare with a capacitor. So, what you save on the mics, spend on a small mixer. These can be found from Behringer, Allen & Heath, Soundcraft and many others. They have excellent mic pres these days and the benefit of per channel EQ so you can roll off the bass end (and with it some noises off) before you even hit the Lambda*.

    A downside of dynamics is that folks will have to stay within 30mm or so of the grill and from what you have said they are not very well behaved!

    At the risk of "grannies and eggs" Do you record with 24bits and keep levels down around -20dB(fs) on the meters in the recording software (do we know it?) . If I AM telling you something knew the reason is. At -20dB the noise will all be what you "input" and you can digitally boost, post tracking with impunity. Then, even if someone shouts they are unlikely to hit digital clipping.

    Last thoughts...Do you have signs to put up "RECORDING IN PROGRESS. PLEASE BE AS QUIET AS POSSIBLE!!"

    *I shall look up the spec of the Lambda.

    Ooops! Forgot second suggestion. If you have 20 or so handy, buy a BM-800 and give it a go. You are not you say looking for super quality and they are not but they ARE cheap and the two I have had work pretty well.

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  9. #18
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    Quote Originally Posted by aira View Post
    ....
    Does anyone else have experience with the Shure SM58? Should that be sufficient for this type of recording? Will it be okay for people with quieter voices (I will try to teach them to speak up, anyway), or does it require a strong pre-amp of some sort? Currently we are just using a Lexicon Lambda audio interface (which is as old as the microphones but still seems to work perfectly) and I usually need to turn the microphone input up to max, or close to it, to capture some people's voices. If we require any more pre-amp than that, I will have to factor that in to the purchase and my approved budget for this.
    ....
    The SM58 was and often still is the principal microphone used in live sound events. I have a couple that are 40 years old and I use them all the time. I used them at a band practice I recorded recently because the practice room had much newer Beta58a's but I prefer the SM58. Honestly, if you haven't come across them and you've been in live music or recording for any length of time (other than your own isolated situation), I don't know how that's possible.

    Now, yes, they do require a bit of gain, as most dynamics will need more than a condenser microphone. The Lexicon Lambda is not an interface that is going to do this easily (spec'd at +44dB) and if you've been turning it up "to max" with condenser mics, it's likely to be a problem. However, part of that may be you're attempting to squeeze a level out of the interface that's not required. A 24-bit recording can be boosted quite a bit in post, so you don't have to have an input level that's loud to be able to render and audible output. So, with the SM58's sensitivity at -54.5dB you'd be down about -10dB when maxed, but you can be down -20dB and still render that to a decent signal in post. However, you probably would want to add a headphone amp into your budget so you can monitor whether you're getting a clean enough signal. (Turning up the interface gain to max is almost certain to introduce noise.)

    You could add a simple mixer to the chain to provide additional mic gain, which might be worth doing if you've been having problems with soft-voiced speakers already; i.e., regardless of which microphone you choose. Even something like the Behringer Xenyx 802 (which I have) could be used, and the two microphones panned hard left and right, with the mixer output mains going to the LINE inputs 1 & 2. Then, you use the mixer GAIN to adjust for different mics, and leave it alone, with the fader giving you some control over the voice differences. Keep the interface gain away from the "max" position by a good bit, no more than say 75%.

    https://www.fullcompass.com/common/f...mbdaManual.pdf

    http://cdn.shure.com/specification_s...et-english.pdf
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    Oh, wow, we're getting into stuff that's over my head here. I've been doing this job for 2 years but I've just been teaching myself as I go. I don't really know much of anything about sound, and transferred all my editing skills from video editing using Adobe Premiere. All the hardware was purchased and set up by the person who was doing this job quite a few years ago. So there's so much I don't know, and yeah, there is clearly a lot I need to learn. I didn't realize it would be so relevant to something as (I thought) simple as buying replacement mics, but maybe it is. :P Forgive my ignorance; I am here to learn.

    I should probably also mention at this point, since we're getting into the details of recording, that due to a disability I can't use headphones at all, and I can't turn up any sound very loud (though I hear sounds much quieter than most people can hear). It might sound ridiculous for someone with the job title of "sound master", but in fact, I do my editing work almost entirely visually. While recording, we have the monitor audio playing out over speakers (facing away from the recording booth, and we don't have any problem with echo being picked up, despite how much it picks up other sounds). This is necessary because the people who are responsible for the scripts need to listen and make any necessary changes during the recording. In order for it to be loud enough for them to hear, it is too loud for me, so I spend most of the recording session cringing in pain and sometimes even sticking my fingers in my ears. I can't really listen to the sound while recording. Instead, I monitor the recording visually in Reaper. I know about what level I'm shooting for overall. I know how to spot if the volume has gone too high (a frequent issue with some of the voice actors, who vary the volume of their voices so much that they're often either inaudible or much too loud). I know when the volume is so low that I won't be able to boost it enough to match the other voice actors' audio, and I need to either turn up the gain on the audio interface or ask them yet again to speak up or try to get even closer to the microphone. But I have no idea about anything to do with any of the numbers.

    Then I take the recorded audio home on a USB stick and edit it. Because noise is such a massive problem, the first thing I do to each track is use a noise gate to remove all sound below a certain level (peeking into the program, it's currently set at -39.5 dB). For tracks recorded with quiet-voiced people (who have been recorded with the gain turned all the way up and couldn't be boosted further during recording), I increase the volume of the entire track by around 3-5 dB in order to make it roughly even with the more average speakers. This often means the background noises get too loud to be caught by the gate, of course, so I have to do a lot more work removing sounds from these tracks. Then after cutting everything up and pasting it in the right order, I listen through and watch the volume meter. When the voice actors aren't speaking, there should be no activity there at all. If I see a bump or a blip, it means there's a background sound there (even if I can't actually hear it) and I completely cut that part out of the recording. Then, after exporting the track, I run it through the "Levelator 2" program to even out the levels overall. With this method, I manage to create quite good-quality tracks without having to use my ears very much at all (and I've gotten universally good feedback from everyone involved - they're very satisfied with the results).

    I also have no idea what is meant by a 24-bit recording. Would someone be so kind as to explain? That sounds like the sort of thing I need to learn. Looking through the settings in Reaper, IF I am reading it properly (under Preferences>Audio>Device), it seems "sample format" is set to 16-bit by default, and I certainly haven't touched that (though I'm not at the office recording computer right now). What are the pros and cons of changing that? (Or am I looking at the wrong thing?)

    I don't know how to tell what volume the recording is at, actually. In Reaper, there is a green bar on the master volume meter which is what I shoot for in terms of average volume in the recordings. I think it's set at "0". The quietest voice actors we have tend to be a bit below that line when boosted to maximum during recording, though I can't pinpoint the exact number. It's entirely possible that due to my ignorance, I'm trying to make the recordings far louder than they need to be, and that I could be turning that gain dial down quite a ways without losing anything. I am extremely open to more information here, for those patient enough to help me overcome my quite obvious ignorance! It could well be that we wouldn't have any need at all for amplification on a dynamic mic, once I learn the proper settings - but if we do, and I get one of the cheaper microphones, it shouldn't be a problem to get a cheap mixer. The cheapest one I see in the Kytary shop is the PHONIC AM55 - and I could afford that plus two of the Shure SM58s with my current approved budget, with a little left over for maybe some extra cables. Without any extra cables, the most expensive one I could add would be the PHONIC AM6GE, and there are quite a few in between those two. (Yay, I can finally post links!)

    We do have signs up asking people to be quiet and I spend a good part of these recording sessions SHUSHing people who type like they're angry at their keyboards. (Especially frustrating since it's these same people who later ask me to remove typing sounds from the first drafts of the edited tracks!)

  12. #20
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    Oh wow Aria you do have a mess of troubles! I am staggered and impressed that you mange to turn out the work you do! I too suffer from hearing problems (registered deaf, got the aids, T shirt the bizz) but yours sound like a much greater disability (plus I am very old!)

    You WILL need some amplification for the dynamic mics and I note the Lambda only has a mic amp gain of 44dB* That is over a hundredfold boost which might sound a lot but these days 60dB (X1000 ) is quite common and it would be rare to find an interface with less than 55dB (X abt 300)

    Since you seem to be "riding" the levels on the AI for the different speakers I think you would find the controls of a mixer easier to use. We can go into the setup in more detail later.

    The "bits" a digital recording chain uses sets the Dynamic Range of the system. DR is the ratio between the highest level before serious distortion (aka "clipping") and the noise "floor" the hiss inherent in all electronic systems. The CD gives an entirely adequate 90dB dynamic range a ratio of 36,000 to one for 16 bits. For recording however most of us use 24 bits which gives an effectively infinite DR set only by the quality of the analogue electronics. A DR of 100dB (100,000:1) is easily found for modern AIs and the very best reach 120dB and better.

    However, this massive range is not why I suggest you go for 24 bits, it is because of the variation in levels you are telling us about. You see, set for 24 bits you can record down at -20 even -25dB in the software meters and any noise introduced will be due to the electronics or, in your case, probably mostly by other people!. You will then have a 20 dB (x10) "headroom" for the shouty ones. This should cut down on your "homework" and spoiled "takes".

    BEHRINGER Q802USB Analogue Mixer

    I would suggest the mixer above over the Phonic. I think having just one mic input will come to be a big handicap. The mixer is also a USB interface but not a good one. If you can find a model without USB but two mic ins and cheaper, go for it. I also have the earlier version of that mixer, the Xenyx 802 and the mic amps are better than many AI units. Mine has sat in a bedroom, powered 24/7 for over 5 years amping up wildlife in my garden and has never missed a beat.

    *Sorry about the decibabble jargon but that is the lingua franca of audio and I think you can see why? All those huge numbers are very unwieldy. I am retired BTW so keep asking, always here.

    Dave.

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