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Author Topic: Thoughts on mic correction, specific to "what we do" and "how we do it"  (Read 2709 times)

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Offline jnorman34

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An excellent discussion.  Thanks gut, for all your efforts and thoughts on this subject.  Fun reading!
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Offline MIQ

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Hi Gut,

Ok, not trying to bog the discussion down with specifics of RoomCapture but you asked early on in the discussion for experiences using automated EQ.  Thought I'd share a few relevant aspects.  But hey, this is your thread, so you lead and I'll follow.   ;D

To get back to your basic premise, I still think you will have a very hard time determining the differences in the mics inherent responses or differences due to mounting/array influences if you are using recorded signals from a specific room.  I don't fully follow the reasoning that since the applause is diffuse and originating throughout the venue that it will somehow negate the influence of being in a room. 

The fact that the applause is diffuse is not only because is is distributed throughout the venue but also because it is reflecting off all the room boundaries.  How is it that the applause "test signal" becomes room agnostic when it is being generated in a room?

Also do you worry that the applause test signal is originating mostly from places in the venue that are different from where the music source will be (stage and PA speaker locations)?  Don't you want to determine the influence the different mics show to sound originating from the same basic location as the sound sources you are truly interested in recording?  Even if the mics are "far" from the performance sound sources and are picking up more than just the direct sound, the room effect on those performance sound sources will be different than the effect the same room has on sounds emanating from locations different than the performance sound sources.  This is especially true at low frequencies. 

To me it seems like you could be pretty successful at determining the basic differences between the mics when positioned in their array by doing more controlled testing, even at home.  A calibrated mic (or even a nice quality Omni made for field recording) used for comparison to the mics in the array mounting and using gated measurements will yield info you can apply to every recording you use this same array for.  Yes it requires some time and care, but so does rotating mics during applause and making freq response comparisons.  You are already using some of the same tools and techniques, why not try to control the test set up more? 

I think your idea is interesting but I'm not convinced your method will yield the results you are looking for.  Maybe (probably) I'm misunderstanding but the goal is a set of corrections you can apply to the mics in the array every time it is used, regardless of the room they are in right?  If you are using signals that are averaged over a long time in a room to determine these differences, it willl be difficult to eliminate the rooms effect.  Is that slight rise from 200-500Hz due to the mic mounting, or due to the room???

Miq

Offline DATBRAD

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What I got from the OP was the result would hold true every time, for each venue, not across a number of different ones. Either way, I think there are always too many variables to take away anything substantial from the type of tests described. Too many variables that can impact sound quality. How could barometric pressure affect the sound waves? I don't know, but I've considered before that since low pressure essentially means thinner air, differences in barometric pressure could be audible. We already know that relative humidity and temperature play a role in how transducers perform. But there could be a difference between a show during the summer months and one on a bitter winter night, from the tons of jackets, fleeces, sweaters, etc, piled up in, on, and between seats. Could that add to the absorption of standing waves, since headcount alone already impacts that directly? I probably asked more questions that I've helped answer in this thread but the main thing is a bunch of tapers are now going to spend the rest of the weekend wrestling with these ideas in their minds, fully engaged. And isn't that one of the things that makes this hobby so continuously fulfilling? Thanks Gutbucket!
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Offline Gutbucket

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Thanks for joining in everyone. Not trying to squelch specifics on RoomCapture, MIQ. I just want to get at the basic outline of how auto-matching EQ functions or those kinds of measurement/analysis tools may apply, rather than getting too deep into the specific details of each at this point.  And as that one in particular is listed as a $900USD tool, it's out of the range of most users here, myself included.

The easy answer to avoid room response specifics is to simply make the applause recording at an outdoor event, completely eliminating most of them.  But I don't think that's necessary, and here's why- Consider close-mic'ing of instruments in comparison to "taper distance mic'ing".  In the close-mic'ed source, the proximity to the source makes the direct sound almost completely dominant over the reflections and reverb tail.  The room sound is in the signal, but at such a lower level that it has little if any influence, being masked by the direct sound.  I see applause being is similar.  I'm suggesting recording from a position within the audience itself, in close proximity to the distributed sound sources - at least those immediately surrounding to the recording position.  Also, the applause signal is more or less constant, it does not energize the room like a single impulse transient then decay away revealing the masked reverberant modal stuff.

More later on this and DATBRAD's thoutghts, gotta run..
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Offline Gutbucket

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Also do you worry that the applause test signal is originating mostly from places in the venue that are different from where the music source will be (stage and PA speaker locations)?  Don't you want to determine the influence the different mics show to sound originating from the same basic location as the sound sources you are truly interested in recording? 

No.  The intent of this corrective step is quite the opposite - eliminating the influence of those particularities as much as possible, precisely because those particularities vary - from venue to venue, by recording position, and by a number of other variables.  This base-line correction is intended to be limited in scope to the particularities of the microphones, microphone array, and recording signal chain only.  Anything beyond that can be addressed by a separate correction step or steps.

Now If one regularly records from the same spot in the same venue with the same PA setup and house EQ, and wants to figure an additional base-line correction specific to that situation, one could figure a following corrective step limited in scope to those venue and recording location specifics.  Doing that might be worthwhile to make homing in on what's needed for correcting reoccurring recordings a bit quicker, since the corrections specific to that recording position in that room may always  be approximately the same.  We already do this to some extent by memory when we think to ourselves "usually when using these mics in this way I need to correct for their excessive of 12Khz empasis with a peak filter", which is a microphone-specific base-line correction which may apply to all recordings we make with those microphones (and in a particular setup), or "when recording in this room, I usually need to notch the lower-mid/upper bass region around 500hz and reduce the bottom end with a shelf-filter around 100Hz", which is a secondary venue-specific correction.

Quote
To me it seems like you could be pretty successful at determining the basic differences between the mics when positioned in their array by doing more controlled testing, even at home.  A calibrated mic (or even a nice quality Omni made for field recording) used for comparison to the mics in the array mounting and using gated measurements will yield info you can apply to every recording you use this same array for.  Yes it requires some time and care, but so does rotating mics during applause and making freq response comparisons.  You are already using some of the same tools and techniques, why not try to control the test set up more? 

Sure, go for it.  I'm somewhat simplifying this process by just adjusting for "natural sound" by ear to keep everything simple (even if my discussion of it is not  :P) and more subjective than analytical.  After all, naturalness of sound and a good starting point for creating a convincing illusion is the true goal here, not a technically measurable flat response.  Sure, with that goal in mind, comparison against a calibrated measurement omni in free space may be valuable (either at home in the studio or at the venue with applause as the test signal).  It's a more abstract approach of inverted curve-matching though, and I suspect the extra variables of using limited bandwidth home stereo speakers within a small reverberant room, in combination with the details of mounting the mics to avoid the room influences will introduce a lot of potential error points.  By contrast, I found it quite quick and easy to dial in a natural applause sound by ear which applied nicely to the music.  But for those who enjoy a more technical approach to, the other path may get you there too. 

I guess it comes down to how much work do you want to put into this and do you find figuring out and implementing the process enjoyable?

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Offline Gutbucket

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What I got from the OP was the result would hold true every time, for each venue, not across a number of different ones. Either way, I think there are always too many variables to take away anything substantial from the type of tests described. Too many variables that can impact sound quality. How could barometric pressure affect the sound waves? I don't know, but I've considered before that since low pressure essentially means thinner air, differences in barometric pressure could be audible. We already know that relative humidity and temperature play a role in how transducers perform. But there could be a difference between a show during the summer months and one on a bitter winter night, from the tons of jackets, fleeces, sweaters, etc, piled up in, on, and between seats. Could that add to the absorption of standing waves, since headcount alone already impacts that directly?

Yes, those things influence whatever further corrections we might want to make after the base-line corrections specific to just the microphones and setup have been applied.  There will always be room for further adjustment and subjective sweetening if we have the inclination to do so and should we feel they are needed.  This should help make that decision and those choices clearer and easier. 

And if one doesn't enjoy the subjective application of EQ or doesn't want to have to commit the time and effort to doing that for each recording, it could serve as a way of getting better sounding recordings with minimal effort each time.  If a good base-line correction can be found which "always applies", those who don't want to do be bothered with excess post work attention could just apply this corrective filter set by rote and get pretty close.  Put in a bit of effort up front to figure out the base-line corrections in order to avoid having to do specific EQ each time.. unless one want's to do so.
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Offline DATBRAD

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Sorry for drifting off topic, but EQ has been something of a touchy subject for me. It's a bias that goes back to my analog taping days when EQ was frowned upon because it wasn't considered "pure" to do anything except hook a pair of high end Nak, Denon, or HK cassette decks together and making as close to an exact copy as possible with the master tape. Dolby was another sensitive subject as some believed in decoding on the playback deck being copied from, and others (myself included) followed the logic that if the master was encoded in Dolby, all subsequent copies should be made with Dolby turned off, since the encoding still passes through to the copy. But I digress.....EQ was something I felt made a recording tailored to the playback system of the person doing it, and that may sound terrible on another person's system.
Today I only use EQ for a recording that is almost unlistenable without it due to extreme bass, or muffled highs. I'm sure many recordings I've made would benefit from some very light EQ that would translate on any playback system. There are tapers that EQ almost every recording they make with great results. I just can't seem to get the old taper in me to make that part of my standard workflow.....oh well...
« Last Edit: May 23, 2017, 07:18:08 AM by DATBRAD »
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Offline kuba e

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Thank you for your posts. It is very interesting.

And if one doesn't enjoy the subjective application of EQ or doesn't want to have to commit the time and effort to doing that for each recording, it could serve as a way of getting better sounding recordings with minimal effort each time.
Great. This is my case.

Is this procedure useful when I record open with one pair of matched microphones? When I record stealth, I will never have exactly the same microphone placement, should I make this correction separately for each recording? What is the exact procedure - to divide a stereo track with applause into two channels and try to eq one channel to sound the same way like second?

Offline Gutbucket

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Thanks guys.  Although it could apply, this probably won't be as useful for matched microphones used in open setups.  And the setup likely would need to be pretty similar each time.  How similar? I'm not sure, but a few different recordings of applause with the same rig incorporating typical variants in the setup could be compared to see how different they are from each other, and conclusions drawn from that.

I'm out in LA getting ready to help a friend drive back across the country.  I'll be offline for about a week or so.  I'll post more thoughts and comments when I get back, including your EQ concerns Brad.  Feel free to keep up the conversation while I'm away if you like.
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Offline kuba e

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I'm out in LA getting ready to help a friend drive back across the country.  I'll be offline for about a week or so.  I'll post more thoughts and comments when I get back, including your EQ concerns Brad.  Feel free to keep up the conversation while I'm away if you like.

I wish you a beautiful trip across the States. I did the same trip 15 years ago, it was great.

Thanks guys.  Although it could apply, this probably won't be as useful for matched microphones used in open setups.  And the setup likely would need to be pretty similar each time.  How similar? I'm not sure, but a few different recordings of applause with the same rig incorporating typical variants in the setup could be compared to see how different they are from each other, and conclusions drawn from that.

Every help and ease of setting the eq is good. I usually record open in small clubs, but if I record in larger spaces, I'll take the process.

I also suppose I should be in the middle of the audience. Perhaps we will be limited by the fact that direct applause will only come in the horizontal plane. The sound from below and from the top may depend on the room acoustics as MIQ mentioned.

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Safe travels Gut. 

Offline Gutbucket

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revisiting this thread..  and taking a step back to view the bigger picture.

Sorry for drifting off topic, but EQ has been something of a touchy subject for me. It's a bias that goes back to my analog taping days when EQ was frowned upon because it wasn't considered "pure" to do anything except hook a pair of high end Nak, Denon, or HK cassette decks together and making as close to an exact copy as possible with the master tape. Dolby was another sensitive subject as some believed in decoding on the playback deck being copied from, and others (myself included) followed the logic that if the master was encoded in Dolby, all subsequent copies should be made with Dolby turned off, since the encoding still passes through to the copy. But I digress.....EQ was something I felt made a recording tailored to the playback system of the person doing it, and that may sound terrible on another person's system.
Today I only use EQ for a recording that is almost unlistenable without it due to extreme bass, or muffled highs. I'm sure many recordings I've made would benefit from some very light EQ that would translate on any playback system. There are tapers that EQ almost every recording they make with great results. I just can't seem to get the old taper in me to make that part of my standard workflow.....oh well..

This is astute and totally reasonable. 

In my way of thinking we should at least conceptually split any corrections made into three separate stages:

The first is whatever corrections we would like to make for the microphones themselves and the way they are mounted.  That shouldn't change from recording to recording as long as the recording setup remains the same and that's what this thread is about. Once determined, it can be applied by rote, to get us to our "base-line-good" response of the recording setup without a lot of corrective effort each time.  I look at these corrections no differently than the choice of what microphone or mic setup to use, or where to setup in the venue.  These are setup questions and setup corrections.

The second stage is the mixing stage were we correct things specific to each particular recording, making subjective decisions about what sounds best.  There are two sub-parts to this one- the first is fixing obvious glitches and problems.  It not glamorous or fun, it's just getting to a problem free starting point. The rest is the more creative part, and mostly about making it sound as good as possible on our own monitoring system.  Maybe we aren't doing anything here but leaving it alone, fading and tracking.  Maybe we are normalizing, maybe mixing two pairs of mics, maybe doing a SBD/AUD matrix, maybe doing some EQ, dynamics manipulation, stereo-processing or whatever.  Separating the first step from this one makes this subjective/creative step faster, easier, more enjoyable.  This is stuff specific to each recording rather than the setup used to record it.

The third stage is the mastering stage.  This is where things relate to the outside world.  This is where decisions need to be made concerning making the recording sound good for everybody else, not just yourself at home. It concerns how others will listen. This one is quite tricky.  It requires very truthful monitoring so that the recording will translate correctly to other systems, as well as truthful listening.  Its also in someways destined to failure from the start- Do we want full range live concert dynamics which might only be appreciated on a big playback system? Do we want less dynamics suitable for listening in the car or otherwise on the go?  Will it sound good on a tiny blue-tooth speaker, earbuds, in the Ford Focus as well as the Lexus, on the big home theater with a subwoofer as well as the clock radio? Well, each of those situations ideally requires different mastering choices.  Format and distribution questions come into play here as well- FLAC, mp3, and for some like me, 2-channel stereo or multichannel audio.

It's that third mastering step I'm posting about here today, mostly because I can't find the other thread I'd started about these opposite-end-of-the-chain corrections which is where this post really belongs.  In that thread we were talking about releasing the raw legacy file (thus preserving the raw master warts and all) along with a corrective "difference file" which when combined with the original would apply all our corrections.  Various corrective files could be made, used and chosen from, acting like different mixes or remasters.  New versions could be made and applied to the original recording at any time. Problem is that each corrective difference file ends up taking up as much space as a new mix, so in the end we don't really save any storage space. Instead we can just store the original file along with each edited version like we do now, which is better because it eliminates the need to recombine them at the listening end.  We only need the edited version to hear what we want.  So the idea although conceptually attractive becomes less compelling in reality. But if instead of a full-sized difference file we could just store metadata along with the raw recording describing how we mixed it, what EQ settings and dynamics manipulations and whatever else we applied to it, we could basically do that without a big storage hit.  The key I think is to only apply that to the mastering stage stuff, not the mixing stage stuff.  We aim to make it sound as good as possible in a no-compromise situation, then the end listener can apply whatever mastering option works for their particular listening situation.  Full dynamics for home, squashed for the gym/subway-commute, whatever.  I would require some processing overhead at the player, EQ, compression, etc, as instructed by the metadata.

This video made me think about this again-  https://youtu.be/KHzD-fR2XUw?t=54m5s  The link points about 54 minutes into a Triangulation 221 podacast interview with Mark Waldrep of AIX records (he operates a label specializing in High Resolution recording) who talks a bit about the potential of this kind of flexible metadata mastering approach.  At first they are talking about re-mixing by the listener (which is solidly within the mixing stage of the 3 stages I've outlined), but they both acknowledge that not many listeners are interested in doing that although it's fun to play around with a few releases where that's possible (Todd Rundgren, Trent Reznor).  The bigger potential is one release which is adapted by the player to the listening situation, environment, and system.  Then one release can be adapted to work everywhere.  Preserving the raw master lineage in our case.

I haven't played further with the mic-setup correction stage this thread is intended to discuss, but plan to do so in the future.
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