I have just read that link you gave me and sorry sometimes I don’t quite understand what I’m reading but I understood that so thank you for that link. From reading that wouldn’t mounting mics on the grid above the stage so that they pick up the whole stage evenly work much better than badly placed overheads?
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- Thread starterAlexD
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Alex, just to address one issue, a mic being a condenser mic refers to how it operates (condenser, dynamic, ribbon, etc.) and not to the pattern or other characteristics. A condenser mic typically has a much higher signal level output than a dynamic and because the diaphragm can be lighter, often have a more 'open' sound, but both condensers and dynamics can be cardioid, supercardioid, hypercardioid, figure eight, etc. patterns.
Area micing for reinforcement gets tricky as it involves several factors and especially coordination with the sound system. I recently had someone recount a story where they walked into a situation that had flown choir mics and all sorts of gain before feedback problems. They ended up removing some of the mics and aiming those left not right at the choir but rather so that the nulls or minimums in the patterns were pointed at the speakers. So you can't look at just the mics, you have to look at the sound system as an integrated system and that what you do for one thing may affect other areas of the system.
Boundary mics can be very useful but keep in mind that to work effectively they need to be placed against a boundary with some area. I often use boundary mics for ambient mics for recording, in-ear/personal monitoring systems, ALS systems, etc. And they can be effective as footlight mics for picking up general stage audio, just remember that they will pick up all audio.
Probably the primary thing to keep in mind is that as the mic gets further from the source, the level hitting the mic drops off. The level hitting a mic 10' away is roughly 20dB down compared to one 1' away and thus the one 10' away has to be turned up 20dB to have the same level. At the same time, the room and speakers are going to interact much more with a sound traveling 10' than with one traveling 1', thus not only affecting what is picked up but if part of the live sound reinforcement, also affecting gain before feedback for that system. So minimizing the distance from source to mic and selecting mics that minimize the interaction with the environment and sound system for that specific application are usually the keys.
Area micing for reinforcement gets tricky as it involves several factors and especially coordination with the sound system. I recently had someone recount a story where they walked into a situation that had flown choir mics and all sorts of gain before feedback problems. They ended up removing some of the mics and aiming those left not right at the choir but rather so that the nulls or minimums in the patterns were pointed at the speakers. So you can't look at just the mics, you have to look at the sound system as an integrated system and that what you do for one thing may affect other areas of the system.
Boundary mics can be very useful but keep in mind that to work effectively they need to be placed against a boundary with some area. I often use boundary mics for ambient mics for recording, in-ear/personal monitoring systems, ALS systems, etc. And they can be effective as footlight mics for picking up general stage audio, just remember that they will pick up all audio.
Probably the primary thing to keep in mind is that as the mic gets further from the source, the level hitting the mic drops off. The level hitting a mic 10' away is roughly 20dB down compared to one 1' away and thus the one 10' away has to be turned up 20dB to have the same level. At the same time, the room and speakers are going to interact much more with a sound traveling 10' than with one traveling 1', thus not only affecting what is picked up but if part of the live sound reinforcement, also affecting gain before feedback for that system. So minimizing the distance from source to mic and selecting mics that minimize the interaction with the environment and sound system for that specific application are usually the keys.
Yes as an A level music tec student I really should know about cardioid, supercardioid, hypercardioid ect patterns but I can never remember them... and as to the condenser isn’t that what you generally use for sound reinforcement? Or even on that overheads.
So what you’re saying is that you have to place the mic’s in consideration to the speakers, where the actors talk/sing and what eels?
So what you’re saying is that you have to place the mic’s in consideration to the speakers, where the actors talk/sing and what eels?
Condenser mics respond to physics principles just like any other mic, energy drops off with the square of the distance.
The bottom line is you're asking for the impossible. It is simply not reasonable to think you can mic someone from a distance for sound reenforcement and expect to acheive anything more than a few db gain before feedback. Since 3 db is really the smallest change in volume percieveable by the human ear, distance micing for PA use results in quite possibly no noticable increase in volume before feedback occurs.
No matter how you ask the question, the answer is always the same. There is no way to effective mic a source from more than a few feed away and use that signal for sound reenforcement without feedback occuring before any noticable increase in volume.
The bottom line is you're asking for the impossible. It is simply not reasonable to think you can mic someone from a distance for sound reenforcement and expect to acheive anything more than a few db gain before feedback. Since 3 db is really the smallest change in volume percieveable by the human ear, distance micing for PA use results in quite possibly no noticable increase in volume before feedback occurs.
No matter how you ask the question, the answer is always the same. There is no way to effective mic a source from more than a few feed away and use that signal for sound reenforcement without feedback occuring before any noticable increase in volume.
I have just realized something… the reason I have been asking this question is to get a better understanding of cores and I “though” I had seen it been done but I just remembered that they did use wireless mic’s. And also I have just realized that most theaters have the grids really high up and the theater I’m in doesn’t its about 1.5m high and the set up I did could be refined so that the mic’s mounted on the grid could be turned up more without feedback. At least you guys cleared some stuff up I didn’t get so thanks for the info guys.
Anonymous067
Active Member
As I've said 1000+ times in this thread...
The closer the microphone is to the source, the better it will sound. period.
The closer the microphone is to the source, the better it will sound. period.
Another way to improve gain-before-feedback is to place the speakers closer to the audience (on stands or hanging) and farther from the mics.
The SPL or loudness of a conventional speaker's direct sound increases about 6 dB whenever you halve the distance from the speaker to the listener. That's 6 dB of free gain without any increase in feedback. Plus, the increase in direct/reverb sound ratio at the listeners' ears helps intelligibility. Close speakers sound more clear than distant speakers.
Mounting the speakers close to the audience also places the speakers farther from the mics, reducing the feedback loop gain. And it tends to place the speakers toward the
backside of the directional mics, so you get some attenuation of the speakers' sound by the microphones.
Moving speakers closer to the audience requires that you re-think the number of speakers and their dispersion patterns in order to cover the audience evenly. You also may need to delay the signal going to the speakers so that the listener localizes the sound on stage, rather than at the nearest speaker.
Hope this helps,
Bruce Bartlett
Bartlett Microphones in Elkhart, Indiana - home page
The SPL or loudness of a conventional speaker's direct sound increases about 6 dB whenever you halve the distance from the speaker to the listener. That's 6 dB of free gain without any increase in feedback. Plus, the increase in direct/reverb sound ratio at the listeners' ears helps intelligibility. Close speakers sound more clear than distant speakers.
Mounting the speakers close to the audience also places the speakers farther from the mics, reducing the feedback loop gain. And it tends to place the speakers toward the
backside of the directional mics, so you get some attenuation of the speakers' sound by the microphones.
Moving speakers closer to the audience requires that you re-think the number of speakers and their dispersion patterns in order to cover the audience evenly. You also may need to delay the signal going to the speakers so that the listener localizes the sound on stage, rather than at the nearest speaker.
Hope this helps,
Bruce Bartlett
Bartlett Microphones in Elkhart, Indiana - home page
This seems worth perhaps a bit more explanation. Because most indoor venues are not anechoic, while the difference in direct level when you double or halve the distance to the speaker is 6dB, the difference in overall level is usually less than 6dB as the actual level at the listener is a factor of both the direct and indirect sound paths and any reflections and other indirect sound paths add to the overall level received at the listener, thus reducing the actual loss in level obtained.
One result is that decreasing the distance from speaker to listener tends to not only result in greater direct level, but also a lower indirect sound component and thus a higher direct/indirect energy ratio, which does help with intelligibility. In fact, get a sufficient distance from the speaker and there is a point where the indirect sound level may equal the direct level and beyond that distance the level no longer changes with distance as it is dominated by the indirect sound, this is the reverberant field.
The second issue is that reducing the talker to microphone distance can typically be more effective in increasing potential system gain before feedback than trying to reduce the speaker to listener distance. Not only is it often easier to move the microphone closer to the person talking than to move the listeners closer to the speaker, but halving the talker to microphone distance will also typically provide a 6dB increase in gain before feedback while halving the speaker to listener distance will usually provide something less than that since there are likely indirect sound paths involved.
jkowtko
Well-Known Member
Any that's why on Broadway nowadays they almost exclusively use body mics for reinforcing the actors' voices. Unfortunately it tends to me more expensive, especially when the cast is large, but currently it's the winning strategy.
jkowtko
Well-Known Member
Alex, from the photos you posted it looks like your theater arrangement is similar to ours, i.e. curved seating area, thrust stage, no pit, etc. Since there is no separation of stage from audience (i.e. an orchestra pit) the balance between audience coverage and feedback into actor's mics will be delicate.
In our theater we are fortunate to have highly raked seating (rises 8 feet from to back with 10 rows of seating) so the grid-height placement of our L, C and R speakers works out okay.
Similar to what was mentioned already, for performances involving body mcis you might consider placing small downfills on stands in front of the stage to take some of the burden off the mains and get the vocals out to the audience more easily. Professional houses make heavy use of small fill speakers for this purpose, and in my experience (watching shows) it works great
-- John
I haven’t posted any photos of the threat I am in but it sounds the save just not a curved stage. I have seen those fill speakers and they do look like they work might try and see if we can get some, because they will be at the front of the stage wouldn’t you have to place some further up the audience to make sure people at the back can hear well?
I calculate any required delays based on the distance between my acoustic source, and the FOH line. Anything less than 15 to 20 ft I don't even both with. I also don't bother if it's just spoken word or dialog since my experience is that once I mic an actor, they quit projecting and the acoustic component is minimal. Beyond that, the rule of thumb is 1ms per foot. Again, this is measured from the plane of the sound source relative to the plane of the FOH line.
The problem is that by raising the mic off the stage you can then encounter reflections off the stage causing combfiltering at the mic. It is usually best to either keep the mic as close the boundary surface as possible, the concept behind boundary mics, some distance away from the boundary or to have a boundary that is absorptive at the frequencies involved.
The main body of the audience should be covered by the main speakers, front fills are usually to add just what is needed for those seated in the first few of rows due to their being out of the main coverage of the main speakers (a result of locating, aiming and selecting the main speakers to reduce the levels on stage and increase gain before feedback) at some frequencies. Thus fills are usually fairly low level and primarily higher frequencies since the mains are likely less directional at lower frequencies. Front fills are often delayed to the natural sound on stage.
Delays can be important for spoken word since it can affect intelligibility. How much delay is appropriate, and whether it is needed at all, depends on the situation. For one thing, any delay is relative to the distances from the natural source to the listener as compared to the distance from the speaker to the listener. Most theatres have more than one listener which means that these relationships vary for different listeners. If you do have just one listener, then delay time for the speakers is probably well down the list of things to be worried about.
A related factor in setting any delay is that not only do you have a listener area rather than a finite point, but you also often have a source area rather than a finite point. Now the speaker locations become the only fixed reference with a range of natural source and listener locations. This makes determining and setting any delays a sort of "best fit" exercise, what helps overall without negatively impacting specific locations to much. With a band performance some people will delay to the drums since that is often the source of the most impulsive, and fairly high level, natural sounds and thus the ones that may most readily show any differences in timing.
Another factor is that you have to think in three dimensions. If the speakers are overhead then they have some delay from the vertical distance to the listeners, in fact you can readily get to where you have to worry about having too much delay if the speakers are too high. For many applications with speakers mounted over a proscenium, that alone may provide sufficient delay to make the sound appear to come from where the actors are on stage.
Then there's precedence or Haas Effect, where you intentionally delay a secondary source such as a fill speaker by an additional time, typically 10-20ms, relative to the natural sound or primary source so that the natural or primary source takes precedence in establishing localization or the location of the source. This works even when the secondary source level is 10dB or more greater than the natural or primary source level.
The point is that while a simple two dimensional estimate may get you started in assessing potential delay times for speakers, the actual settings for speaker delays are usually something that has to be verified and tweaked in the field.
So you're comparing me to Tom Cruise? Guess there's both good and bad in that.Charlie Babbit made a joke. See everyone, I told you Brad had a sense of humor.
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