Topic: Advice on Using a Zoom H5 for AUD recording

[cmstewart]

I'm new here! And I'm also a first-timer with recording concerts (so bear with me if you can). I have a Zoom H5 with the SSH-6 mid-side pattern stereo shotgun mic capsule. I also have a Beyerdynamic MCE72 Stereo shotgun mic (crossed pair pattern mic). Both mics in the zoom will be recorded on separate stereo tracks, so I will essentially be making 2 recordings. I have a compact handheld mount (with shockmounts for each mic) to carry it all, and may use a tripod if i can get one before Sunday. These are all recent additions, so I'm still figuring out what else would be necessary (preamps? attenuators? etc.) to make a great recording.

It's a Counting Crows show at the Woodlands Pavilion. My seat is about 13 degrees off-center toward the back of the front/middle section. Pretty nice distance in my (inexperienced) opinion. Does anyone have any advice on Zoom's settings at a live show (limiter/compressors for instance) that have helped you? The great news is that CC allows pretty much anything at their shows. So i don't have to be sneaky. Given that I have that freedom, I feel I need to ask experts in order to make the most of it.  I'll also be mounting my V20 phone to video the concert, but my priority is the audio. And I'd love to have it turn out great. I'm willing to learn as much as I can.

[cmstewart]

Well.... I tried!!!!!!

[seethreepo]

While I don't have first hand experience with your exact equipment , I highly recommend practicing on a local bar band (to estimate levels) get used to starting and stopping the deck etc.
Bring Extra batteries
 you might want to read this thread since it covers "shotgun" mics  which in many opinions are not idea for what we do but YMMV.   http://taperssection.com/index.php?topic=186891.0

If your gear has a "line in" it might be advisable to use that  to avoid brickwalling or "distortion. 
lastly you might inquire with the venue before hand to see if they will allow a mic stand  (tripod, lighting stand)  , print a copy of the CC' taping policy if there is one otherwise you might get denied at the door. 

having taped a ton of bands (some which allow it some who dont) , Its best to clear things ahead of time  rather than assume. People sometimes have isses taping the Grateful dead and they invented audience taping !   also show up early to have a buffer to overcome any hassles at the door.  Tapers to venues are not normal customers.  (we are a nuisance to some and Verboten to others.)   

I'm sure others will chime in  as we are mostly a helpful bunch 

best of luck

[Fatah Ruark (aka MIKE B)]

For levels I'll start with the pre-show music at about -30 db. Likely this will be a little low when the band comes on, but it's better to have your levels a little low, than a little high.

I'm usually happy with peaks somewhere between -12 and -6.

CC don't have a huge taper following so it's likely that you will get denied at the door. Be polite and ask if they can double check with the bands MGT. Also get there early so you have plenty of time to have them check. I was at a shot last week (open taping band). Took about 30 minutes of waiting to get let in.

[cmstewart]

As for the taping policy/venue issue -- I already emailed Cynthia Woods Mitchell Pavilion Thursday and got written approval of the use of recorders, mic, tripods, and video cameras. I'll be carrying that email with me when I go. So, being denied is still possible but I will remain confident. Thanks for the heads up though, hopefully I won't have too many troubles. I'm more worried about where/how I'll set up, where I don't feel like I'm in other people's way!

As for the use of shotguns, I don't really have a choice as it's all I have, and I'm just doing this for the fun and enjoyment of the music, not to be a professional. At the same time, I'm willing to learn and do it the best I can with what I have. I just got the H5 so I'm a novice, but I've run a lot of tests to get a feel for it in the last 3 or 4 days. It's hard for me to simulate the volume of a live show though in my apartment (without getting the cops called at least).

After reading the first page of that thread you linked.... it seems my gear may not be a terrible idea? (perhaps not ideal still). One suggestion seemed to be a M/S pattern shotgun mic configuration in the center, with 2 spaced out omnis on the L and R. Now, I did make one small mistake - the Beyerdynamic MCE72 is actually (although shaped like a shotgun) simply a stereo condenser mic set in a crossed pair pattern at a 120 degree angle. My mistake. But my other shotgun is a stereo M/S pattern shotgun. So I'm hoping those with the right spacing can get a nice L/C/R type of recording?

The H5 does have a line in, but using it cancels out the M/S mic capsule. And I'd need to get the correct cable for the other mic (with a 3.5mm plug) to plug that into it. So I'll have to use the XLR inputs tomorrow. I definitely don't want any brickwalling. I'll start with the levels low and work up to what sounds best to my ears. What height range on a tripod (for a large outdoor pavilion) would you guys think is ideal? I may not have the room for the "ideal", but I'm just looking for general ideas/standards to work from.

Thank you guys for all the input.

[jefflester]

What height range on a tripod (for a large outdoor pavilion) would you guys think is ideal? I may not have the room for the "ideal", but I'm just looking for general ideas/standards to work from.

You pretty much want to get the tripod as high as you can, but people behind might complain and you might get forced to bring it down to head height.

[restevezes]

+1 to the practising tip. It is by far the best source for learning. reading forums is necessary but without practising you will not make any progress. (would you read about how to have sex???  )


Go and tape couple of random bands in a loud environment.
Experiment with different settings, configurations, and write them down somewhere.
Listen at home and try to figure out differences when you changed things and learn why this happened
Then come here asking for any potential questions you might have, and start again. After few rounds of practising you should be ready to go
never go to your wanted show without trying first!

[cmstewart]

+1 to the practising tip. It is by far the best source for learning. reading forums is necessary but without practising you will not make any progress. (would you read about how to have sex???  )


Go and tape couple of random bands in a loud environment.
Experiment with different settings, configurations, and write them down somewhere.
Listen at home and try to figure out differences when you changed things and learn why this happened
Then come here asking for any potential questions you might have, and start again. After few rounds of practising you should be ready to go
never go to your wanted show without trying first!

Haha well that's obvious. I'm not asking to make me an expert, unfortunately CC plays tonight and I just got the idea a couple weeks ago. So I don't care if I suck, I just like having tips to build on. it's better than walking into something blind right? I really don't care, it's fun. I want to enjoy it. I thought that was the purpose. Since Live plays first I'll get about an hour to test things with them before CC goes on stage. Next year they come around I'll have more experience. Thanks.

[cmstewart]

Well - to my ears this recording came out really great, especially for a first attempt. As I mentioned before, I used 2 stereo mics so I have two separate stereo recordings. When I played it back to listen thru, I was listening to both simultaneously. Now I need to blend the two into a single recording... and "master" it? This is an area I'm less experienced with. Would anyone be willing to help guide me through what's necessary (or link me to something with advice)? I'll be searching around the site for advice already here.

[goodcooker]

Well - to my ears this recording came out really great, especially for a first attempt. As I mentioned before, I used 2 stereo mics so I have two separate stereo recordings. When I played it back to listen thru, I was listening to both simultaneously. Now I need to blend the two into a single recording... and "master" it? This is an area I'm less experienced with. Would anyone be willing to help guide me through what's necessary (or link me to something with advice)? I'll be searching around the site for advice already here.

I would seriously consider choosing one stereo recording or the other. Two stereo shotgun recordings are not likely to benefit from being mixed together. It might - and I haven't heard them - but for your first go at it I would just pick one and work on it.

My workflow for "mastering" a stereo recording is -

1. Import file into audio editor (I use Wavelab6) - reduce any unwanted peaks using gain adjustment tool, increase overall gain of file to -.1dB, make any EQ adjustments (I often will remove some lowest bass frequencies) and add any dynamics processing. Add fades if you want them. Export as a .wav file.

2. Use CD Wave Editor to cut the .wav file into individual tracks. I remove long periods of non musical between song bits at this point. I create my info text file at this time with source info, notes and setlist and save it in a folder where I place the FLACS for uploading.

3. Convert files to FLAC - CD Wave Editor does this along with other program like Trader's Little Helper.

4 Upload to LMA and add FLAC files to my tablet so I can jam out in the car - all done!

[cmstewart]

Thanks Goodcooker. Just to clarify... it's not 2 shotgun mics. There's only one. The other is a stereo condenser. I'll let you hear it when I'm through, but there does seem to be a benefit as it's made a L/C/R recording. I lowered the figure 8 (the side mic on the M/S shotgun) down a bit to keep relatively center. The stereo condenser is more wide sounding... much more distant and more crowd noise. The shotgun is far more focused on the band, more mono-sounding, and less crowd noise. Matrixed it has the detail of the shotgun and the openness of the condensers and keeps the band cutting through in the middle. Just my opinion but I like it.

*The thing I personally love about it is it sounds SO MUCH like it really did at the Pavilion. I really love the venue, and when I listen to it the ambience around the band, when I hear it play, I can visually see what I really saw being there. Real clean, the band sounds crisp, and it brings me back to that night when I listen. So maybe I'm selfish mixing this how I like it?  but it is what it is!

Neither really clipped at all but I know if mixed down to one they will at points, so I'll just decrease the level of each and experiment with how much to keep it below 0, adjusting  peaks like you said.

Is Wavelab free? I thought it was only for purchase. I have used audacity, but I have a question about that (if you are familiar). I recorded at 24-bit 48khz. When exporting as WAV, it wants to export as 16-bit. But I can export as 24-bit FLAC. What's the benefit of exporting in WAV then using another program to convert to FLAC. Is CD Wave Editor better at doing it? Cause I can also add fades and track it with Audacity. But maybe there's a quality improvement by using Wavelab or some other programs? I don't really plan on using any EQ, no frequencies stick out and I'd like to keep it pure if I can.

What is LMA? Just FYI - I'm a devoted member at Crows-Town and will also be sharing a download link there as well. I'm a brand new taper, but have collected boots for years. We have a database of hundreds of CC boots if you ever want more. Hope you got a great sound system in your car 

[nulldogmas]

I have used audacity, but I have a question about that (if you are familiar). I recorded at 24-bit 48khz. When exporting as WAV, it wants to export as 16-bit.

You can export from Audacity in 24-bit WAV. You have to select "Custom format" or something like that, then it gives you the option.

Mixing the two sources in Audacity is dirt easy, so if you're familiar with that, I highly recommend it.

And LMA = Live Music Archive = archive.org's music site. It doesn't look like CC is authorized for upload there, though.

[Gutbucket]

From your description of your mixing it sounds like you have a handle on things and will move more quickly toward advanced multichannel techniques than most novice tapers do.  Follow your ears.

As goodcooker suggests, its often simplest and least problematic to choose the better of two stereo recordings rather than mixing them together.  Yet if it works and sounds good to you to mix them then there is no reason not to go for it!

The "problematic or not" issue basically boils down to mixing two recordings which weren't made in such a way that they were designed to combine well without problems.  In that case it's pretty much a crap-shoot as to whether mixing them will make things better or worse.  Unfortunately with tapers that's pretty much the general rule rather than the exception - making two stereo recordings each of which makes sense on its own in isolation, yet are sort of haphazardly setup with respect to each other, both attached to the same stand without much thought given to the spacing, angle, directional pattern and other relationships between the two setups other than having both of them point toward the stage.

There are things one can do with regards to microphones and their setup to help make mixing two sources made in close proximity work more successfully more often, if that's what you like doing.  It helps to consider of all the mics you are going to mix together as essentially a single multichannel stereo array, which may or may not consist of two separate stereo recordings which will be combined.

Your description of how you ran the two stereo microphones in one hand-held mount sounds like you are using a mount which essentially places the two stereo mics in close coincidence with each other, and that's one way of avoiding weird mixing problems.  Another is spacing the pairs or some of the microphones far enough apart from each other.  Unfortunately, the more common and oftimes problematic combination is two seperate near-spaced stereo pairs, placed on a stand so as to be near-spaced to each other.  Too many microphones that are neither coincident nor far enough apart from each other, all pointing in about the same direction and picking up the same content = a recipe for conflicts.

..it's made a L/C/R recording. I lowered the figure 8 (the side mic on the M/S shotgun) down a bit to keep relatively center. The stereo condenser is more wide sounding... much more distant and more crowd noise. The shotgun is far more focused on the band, more mono-sounding, and less crowd noise. Matrixed it has the detail of the shotgun and the openness of the condensers and keeps the band cutting through in the middle. Just my opinion but I like it.

*The thing I personally love about it is it sounds SO MUCH like it really did at the Pavilion. I really love the venue, and when I listen to it the ambience around the band, when I hear it play, I can visually see what I really saw being there. Real clean, the band sounds crisp, and it brings me back to that night when I listen. So maybe I'm selfish mixing this how I like it?  but it is what it is!

You are already doing things which work in a way similar to what I describe with regards to microphone setup and mixing choices for a multichannel array.  My suggestion in the other thread to use a shotgun in the center between a pair of spaced omnis is a real-world adaptation to the oddity of audience perspective recording. It produces a type of L/C/R type of stereo I think you would really like.  The omnis provide the wide openness, you-are-there-ness, the ambience, while the shotgun provides the direct clarity, detail and focus.  The two parts work together to help each other rather than fight with each other trying to provide the same thing.

Using a Mid/Side shotgun in the center makes this better by allowing adjustment of the Side channel level when mixing.  That essentially does two things- blend the shotgun center more more seamlessly into the open ambient "bed" provided by the spaced omnis; and provide some tight Left/Right coincident imaging across the center of the stereo image.  Typically, just as you describe, you'll use much less Side channel than you would if you were using that Mid/Side shotgun alone.  That's because the omnis are providing most of the width and diffuse stereo interest.  The Mid/Side shotgun then can focus on providing a touch of stereo-ization to the direct sound.   Each pair providing something different enough that they aren't competing but rather help each other.

If you have a pair of cheap miniature omnis give it a try, spacing the omnis as far apart as you are able to.  Shoot for 3-6 feet apart but that's not always doable so just do what you can.

[cmstewart]

Thank you Gutbucket for expanding on the L/C/R idea. It was your post that led me to try it, though I feel I was just lucky that that was the equipment I had. I figured, well this seems like the only way to make a shotgun useful, so why not! The big concern I had was that the "side" mic would overlap too much with the L/R condensers and create some phase issues, so I lowered it a few decibels. As far as I can tell though, there's no issues I can hear. What you described about the concept of the whole setup is exactly like I was envisioning. I didn't realize it, but you're right about the side mic helping the shotgun blend in with two condensers. It's really makes it seamless, and the different frequency patterns of each I think help bring out different details that one mic may not have been able to catch. I'm sure I could improve my gear some more though to make it even better.

I love experimenting so if I can get a couple omni's it'd be fun to try. The only issue I see with it is how to get all the mics high enough if they're that spaced out, without picking up too much annoying chatter. When all the mics are nearby, you can just find a nice spot and raise them up to a good level. I only have one tripod for it so I'd need to figure that out. Of course, this also depends on the venue and stuff as well.

You can export from Audacity in 24-bit WAV. You have to select "Custom format" or something like that, then it gives you the option.

You're right, it's called "Other Uncompressed Formats". I've been wondering, what's the benefit of exporting a WAV file and then using another program to convert to FLAC? Is exporting in FLAC doing too much at once? Or are other programs better than Audacity at doing it? Or is it just the "right way" to do it so you have a clear lineage from one step to the next? It's things like this that really expose my inexperience in taping!

[nulldogmas]

You're right, it's called "Other Uncompressed Formats". I've been wondering, what's the benefit of exporting a WAV file and then using another program to convert to FLAC? Is exporting in FLAC doing too much at once? Or are other programs better than Audacity at doing it? Or is it just the "right way" to do it so you have a clear lineage from one step to the next? It's things like this that really expose my inexperience in taping!

FLAC is lossless, so there's zero reason to save first as WAV and then convert later to FLAC that I can think of.

[cmstewart]

You're right, it's called "Other Uncompressed Formats". I've been wondering, what's the benefit of exporting a WAV file and then using another program to convert to FLAC? Is exporting in FLAC doing too much at once? Or are other programs better than Audacity at doing it? Or is it just the "right way" to do it so you have a clear lineage from one step to the next? It's things like this that really expose my inexperience in taping!

FLAC is lossless, so there's zero reason to save first as WAV and then convert later to FLAC that I can think of.

Okay cool. I just see a lot of people exporting from wavelab (or other programs) as WAV then using CD Wave to convert. I guess maybe Wavelab can't export FLAC, or it can't divide up the tracks? I don't have it so I'm clueless as to why. Thank you!

[jefflester]

You're right, it's called "Other Uncompressed Formats". I've been wondering, what's the benefit of exporting a WAV file and then using another program to convert to FLAC? Is exporting in FLAC doing too much at once? Or are other programs better than Audacity at doing it? Or is it just the "right way" to do it so you have a clear lineage from one step to the next? It's things like this that really expose my inexperience in taping!

FLAC is lossless, so there's zero reason to save first as WAV and then convert later to FLAC that I can think of.

Okay cool. I just see a lot of people exporting from wavelab (or other programs) as WAV then using CD Wave to convert. I guess maybe Wavelab can't export FLAC, or it can't divide up the tracks? I don't have it so I'm clueless as to why. Thank you!

Now you get to learn about SBEs. :-)

[cmstewart]

Now you get to learn about SBEs. :-)

Hmmmmmm......... you stumped me.   

[nulldogmas]

It's a CD burning thing:

http://www.audiofile-engineering.com/support/helpdesk/index.php?pg=forums.posts&id=7984

[Gutbucket]

This is advanced stuff, but your ear for it is already there as is your predilections, so I'll jump right into it-

Perhaps the biggest realization I've had which affects how I think about live music recording, how I go about doing it these days, and which has dramatically improved the listening experience of the recordings more than anything else is the use of separate microphones, microphone pairs, or different portions of a single multichannel array to record different aspects of the sound, and then combining those things afterwards to form a well balanced recording where each aspect is well optimized.   This approach eliminates a number of compromises always inherent in using a single stereo pair, while introducing a loads of control over getting each aspect of the recording balanced and sounding its best.

Your current setup using two stereo microphones is essentially providing you the ability to split the job into two separate but overlapping parts- Gathering clear, direct, detailed sound from the rather distant stage and PA via the M/S shotgun, and gathering ambient, immersive, reverberant sound arriving from all directions around you via the Beyerdynamic 120 degree X/Y microphone.  I like spaced omnis for that second role because they are optimally suited for recording and conveying a diffuse, reverberant, immersive sound, and also because they are sort of the polar-opposite of a coincident stereo setup such as your M/S Shotgun, which helps with the "focused on different aspects without too much conflicting overlap" thing.  In addition, the spacing used between the omnis places them far enough away from your center shotgun that mixing problems are further minimized.

But you don't have to use spaced omnis.  I love them but they are not always practical to run, mostly due to the spacing required.  I'll follow this post with a couple suggestions you might try with your current setup if you'd like to do some experimentation, and the first you can try with your existing Counting Crows recording.

[Gutbucket]

Okay, here's a couple suggestions you might try with your current setup-

1) You can use a pair of Mid/Side encode/decoders to change the effective stereo width of the 120 degree X/Y Beyer microphone, essentially making the inclusive angle narrower or wider than 120 degrees.  To do so with two Mid/Side instances, you'd first convert the Left/Right stereo signal back to Mid/Side, make the adjustment to the M/S ratio, then convert back to L/R again.  Or you can use a stereo-width adjustment tool which is basically doing the same in a single step.

My suggestion is to play with making the Beyer mic contribution wider than 120 degrees.  On its own that will probably sound less focused and more reverberant (not what you'd want if making the recording using that microphone alone), but in combination with the center shotgun that will give you greater differentiation between front focus and immersive width.  You'll probably want to readjust the M/S ratio of the stereo shotgun, likely making it a tad wider, to get the best blend and to get a bit more of the sharp coincident imaging across the front.

If you want to play around with taking the mixing to the next level using EQ, you already have lots of additional options available to you with this arrangement.  Besides general EQ of the resulting stereo mixdown, you can EQ each pair separately, making each sound its best for the sonic attributes it is contributing.  You can EQ the center shotgun for clarity and the X/Y ambient pair for smooth naturalness, or whatever works best.  You can even push that a bit, trying different curves which compensate one pair against the other such that the overall EQ remains the same, but the emphasis within the overall sonic scene is altered one way or another.  Each pair listened to in isolation may sound less than optimal that way, but it balances out when combined.  Best to start with simply making each pair sound best on its own in isolation prior to combining them, before going to far with those kinds of compensating EQs.

You can go deeper than that, and play with EQing the Mid and Side signals of each pair differently.  This provides control over the stereo width of each pair by frequency range.  Say you boost the low frequencies of the Side signal while reducing the same frequencies in the Mid signal by the same amount.  The L/R stereo output will have the same overall EQ balance as before, but the stereo width at low frequencies will be widened while the stereo width at mid and high frequencies remains unchanged.  This technique is powerful and can be applied to to any stereo recording by converting it to Mid/Side and back as described above.

2) Try pointing your Beyer X/Y microphone backwards instead of forwards.  This sounds crazy but its really a further extension of making the ambient collection portion more differentiated from the front focused part.  It is also likely to make the ambient portion sound wider and more immersive and will provide additional control over the direct/reverberant balance of the recording by adjusting the relative levels of the two.  As above you may want to use a bit more stereo width from the M/S stereo shotgun in combination with doing this, but perhaps not.  Best to decide that kind of thing by listening as those choices will vary given the recording situation.  Note that you'll want to route the Left channel to Right and Right channel to Left if pointing your Beyer stereo microphone backwards, or probably easier to simply rotate the microphone so that it is upside down when pointing backwards so that the Left capsule side is still pointing Left, and vice-versa.


Other things to try with all this in mind-
If you have a pair of cardioids and want to try using them instead of the Beyer X/Y mic for the ambient portion of the recording, try using those near-spaced (easier to setup than wider spaced omnis) but pointed 180 degrees away from each other to each side instead of forward.  Again, probably not what you'd want if using that pair alone, but more optimal for combination with the forward facing stereo shotgun.  Push them apart as wide as is practical for a single stand and whatever mic bar you are using.  A good target would be twice the width of a standard near-spaced arrangement if you can manage that.  With regards to the microphones forming a single multichannel stereo array, think of the of the center shotgun and each cardioid as forming adjacent near-spaced pairs with a 90 degrees inclusive angle, which is why twice the spacing of a standard near-spaced cardioid pair is probably a good starting point if you can manage that.

[Gutbucket]

As I mentioned in my earlier post, your current setup using two stereo microphones is essentially providing you the ability to split the job into two separate but overlapping parts.  That in turn provides more degrees of freedom in manipulating the resulting recording which you can choose to take advantage of or not.  I run an array of 6 microphones which essentially splits the job into four separate but overlapping parts and I generally balance level and EQ between those parts to build the resulting recording.  I'm still at the mercy of the venue acoustics, audience, sound guy, and band, but I maximize my options this way given the hand I'm dealt.  It's a lot of fun to play around with afterwards, and I have very much enjoyed the learning and experimentation process in getting to this point.

I hope all this is all more helpful than overwhelming.  You certainly don't need to do any of this stuff and I mostly wanted to outline possibilities which are open to you.  Even though my way is not the way most folks approach recording, it's rewarding for me to share these insights and discuss them at TS, in the hope that it may help others along their own path.  Welcome aboard!

[cmstewart]

I hope all this is all more helpful than overwhelming.  You certainly don't need to do any of this stuff and I mostly wanted to outline possibilities which are open to you.  Even though my way is not the way most folks approach recording, it's rewarding for me to share these insights and discuss them at TS, in the hope that it may help others along their own path.  Welcome aboard!

I would say about 80% of that I have a good grip on. I'm relatively new to taping, but from what you said it sounds like this sort of setup isn't common or conventional, which I kinda like. It's part of the fun of doing something to find what "you" like, and as long as it works for what you want there's no reason not to build on it. The Zoom H5 can record the M/S shotgun in "raw", which means it's just placing both mics on different L/R channels so you can adjust them separately after recording, and then matrix them with another program. I didn't do that this time because A) I've never used a program that does that and B) I wanted to monitor the M/S already mixed into one stereo track while it was recording, so I could get a feel for how the actual recording sounded. But if I did have them in raw, I can see a lot of benefits of being able to manipulate both tracks independently to compliment the other.

It's interesting you mentioned turning the Beyerdynamic backwards, because the first time i heard that mic someone used it for voiceover, with a mono mic pointed at them and the Beyer turned around to pick up ambiance of their voice. It sounded really good, just gave a slight stereo-feel to it. I'm sure doing that would allow me to (as you said) increase the side mic on the shotgun since they would overlap less.

When I listened to the whole show through, I was blown away cause I didn't expect it to sound as great as it did. Since it already sounded great, I'm hesitant to manipulate it much with EQ unless something specifically sticks out, which in a few instances has happened. I just don't have a great grip on how to analyze a track to see what's out of balance, and how specifically to correct that problem once you do pinpoint it. So I tend to just want to leave it uncolored as much as I can. Maybe over time I'll begin to understand all that better and can utilize it. I mainly just like using my ears, but that means when I tweak the EQ I'm just taking shots in the dark, then listening to see if it helped or not. Maybe that's how most do it? But I'm sure there's more of a science to it than that.

I've never used an Omni for anything I don't think. So I'm curious how different that would sound. I think the part that's kind stumped me is the encoder/decoder you mentioned to make the Beyerdynamic wider or narrower than 120 degrees. I didn't know I could do that! I also find it interesting that you use 6 mics in your setup. I'd have to upgrade to a recorder with more inputs to be that ambitious, the H5 only has 2 in addition to the built-in capsule. I may see if I can get my hands on a couple omnis and run some experiments to hear the difference they make. Thanks so much for the welcome! And I really appreciate all the input!

[cmstewart]

It's a CD burning thing:

http://www.audiofile-engineering.com/support/helpdesk/index.php?pg=forums.posts&id=7984

That is a really interesting article. Since I recorded in 24-bit, I'm curious.... is there a preferred way to convert it to 16-bit for burning?

[Gutbucket]

..it sounds like this sort of setup isn't common or conventional, which I kinda like. It's part of the fun of doing something to find what "you" like, and as long as it works for what you want there's no reason not to build on it.

Contrary to the way most tapers are doing it.  Most are pointing a near-spaced stereo pair of microphones at the stage/PA, making relatively basic variations on that, using mics with different polar patterns and making minor (though important) setup variations of microphone angle and spacing.  If running multiple microphones and mixing them, it's usually two or more of those near-spaced stereo pairs pointing at the PA, maybe a pair of somewhat wider-spaced omnis.  Or one near-spaced stereo pair mixed with a soundboard recording. Some run an X/Y or Mid/Side pair, but few seem to be mixing those with other mics (other than a soundboard feed if available).   I understand why folks do that, they like the sound of near-spaced stereo configs (as do I).  The problem is that a near-spaced config represents an optimized 2-microphone setup.  Mixing in more microphones or other pairs does not tend to make that more optimal, and frequently makes it less so. There are ways of making optimized multimicrophone stereo recordings which are designed so as to mix correclty, but that generally requires abandoning the optimized near-spaced stereo configurations that are most popular, and most tapers don't wish to take that leap.

The Zoom H5 can record the M/S shotgun in "raw", which means it's just placing both mics on different L/R channels so you can adjust them separately after recording, and then matrix them with another program. I didn't do that this time because A) I've never used a program that does that and B) I wanted to monitor the M/S already mixed into one stereo track while it was recording, so I could get a feel for how the actual recording sounded. But if I did have them in raw, I can see a lot of benefits of being able to manipulate both tracks independently to compliment the other.

Not sure of the H5, but many/most? recorders which feature a M/S function allow you to switch on the M/S>L/R function for monitoring (and playback) while recording the Mid and Side channel signals in their non-encoded state.  That's intended for doing exactly what you want- monitoring L/R while recording M/S.

When I listened to the whole show through, I was blown away cause I didn't expect it to sound as great as it did. Since it already sounded great, I'm hesitant to manipulate it much with EQ unless something specifically sticks out, which in a few instances has happened. I just don't have a great grip on how to analyze a track to see what's out of balance, and how specifically to correct that problem once you do pinpoint it. So I tend to just want to leave it uncolored as much as I can. Maybe over time I'll begin to understand all that better and can utilize it. I mainly just like using my ears, but that means when I tweak the EQ I'm just taking shots in the dark, then listening to see if it helped or not. Maybe that's how most do it? But I'm sure there's more of a science to it than that.

If it sounds good it is good.  No need to mess with it, except if you want to play around with things in order to learn what's going on and further hone your ear and audio manipulation skills and get a better handle on what's going on such that you can use that to inform further variations on your recording technique.  And that's a good reason for me.

I think the part that's kind stumped me is the encoder/decoder you mentioned to make the Beyerdynamic wider or narrower than 120 degrees. I didn't know I could do that!

You can do this with any stereo recording!  It's a powerful mastering technique to convert L/R to M/S, then adjust the stereo width or do other things to the Center versus the Sides, and then convert back to L/R again.  It's least problematic to do so with a coincident stereo recording where there are no phase differences between channels, but it can still be done with near-spaced directional mic and wide-spaced omni recordings, if with a somewhat more narrow range of useful readjustment.

I've never used an Omni for anything I don't think. So I'm curious how different that would sound. I also find it interesting that you use 6 mics in your setup. I'd have to upgrade to a recorder with more inputs to be that ambitious, the H5 only has 2 in addition to the built-in capsule. I may see if I can get my hands on a couple omnis and run some experiments to hear the difference they make. Thanks so much for the welcome! And I really appreciate all the input!

I love omnis, they are the most-basic pattern, the least wind and handling noise sensitive, usually more natural sounding, and relatively less complicated to manufacture - so in general less costly omnis tend to sound better than similarly less costly directional microphones.  But they'll pretty much pickup everything around them in that natural way, which can be both good and bad, depending.  They can make for a more natural "you are there" sounding recording, and perhaps not surprisingly that can sound less detailed, close and intimate given a distant recording position.  For me they are the basis upon which I build the recording I want to hear via a mindful addition of other directional microphones.  Below is a link to a long-running thread which more-or-less blogs my path over the past 12 years from a single pair of spaced omnis to the multichannel arrays I'm currently using.  It's called the Oddball Microphone Techniques Thread for good reason, and suspect that may connect with your inherent preference for non-convention. https://taperssection.com/index.php?topic=96009.0

[Gutbucket]

It's a CD burning thing:

http://www.audiofile-engineering.com/support/helpdesk/index.php?pg=forums.posts&id=7984

That is a really interesting article. Since I recorded in 24-bit, I'm curious.... is there a preferred way to convert it to 16-bit for burning?

Apply dither when reducing bit-depth.  Doesn't need to be fancy noise-shaped dither, basic "triangular dither" works fine and is what I use.  Most modern editing programs will automatically dither when outputting to a lower bit-depth file, just check setup.  This should be your last processing step, other than maybe tracking (and tagging).

[cmstewart]

Hey guys, I've edited the tracks in Audacity and am pleased with where it's at (I have both recordings in Audacity in full right now). So, I'm wondering the most efficient/effective way to track it (split the songs up). I've seen some people use Audacity for tracking. Some CD Wave. And others of course. In addition, I'm wondering why most people track it first, then convert to flac later. You can export from Audacity in Flac so I'm wondering if there's a reason to export in Wav then convert later? I haven't used CD Wave or any other program besides audacity, so if you have a suggestion for me with one of the others I may need a little help figuring it out. I appreciate all of yall's help so far.

[morst]

Hey guys, I've edited the tracks in Audacity and am pleased with where it's at (I have both recordings in Audacity in full right now). So, I'm wondering the most efficient/effective way to track it (split the songs up). I've seen some people use Audacity for tracking. Some CD Wave. And others of course. In addition, I'm wondering why most people track it first, then convert to flac later. You can export from Audacity in Flac so I'm wondering if there's a reason to export in Wav then convert later? I haven't used CD Wave or any other program besides audacity, so if you have a suggestion for me with one of the others I may need a little help figuring it out. I appreciate all of yall's help so far.

I used to actually "split" tracks in Audacity but then I found out about using Ctrl-B (Command-B on mac) to add "Break" points, so you can Export Multiple with those.


If you want to be CD compatible to the max, Audacity will let you snap to Hours/Minutes/Seconds + CDDA Frames (75 fps.) You can change the time units on the bottom of the window to that. There are meters for Selection Start, End/Length, and Audio Position. Each of those has a little downward-pointing triangle which will let you pick what time units to Snap To when you turn that on. Be sure to do this BEFORE you add track points with Ctrl-B or if you split the tracks.


I tried to get Audacity to export FLAC files, but it made a mess of the tags, so I had to use my favorite utility (xACT for mac) to fix stuff anyhow. Be sure to set Audacity's parameters when you export so you get the word length and file type you intend.

[cmstewart]

I used to actually "split" tracks in Audacity but then I found out about using Ctrl-B (Command-B on mac) to add "Break" points, so you can Export Multiple with those.


If you want to be CD compatible to the max, Audacity will let you snap to Hours/Minutes/Seconds + CDDA Frames (75 fps.) You can change the time units on the bottom of the window to that. There are meters for Selection Start, End/Length, and Audio Position. Each of those has a little downward-pointing triangle which will let you pick what time units to Snap To when you turn that on. Be sure to do this BEFORE you add track points with Ctrl-B or if you split the tracks.


I tried to get Audacity to export FLAC files, but it made a mess of the tags, so I had to use my favorite utility (xACT for mac) to fix stuff anyhow. Be sure to set Audacity's parameters when you export so you get the word length and file type you intend.

Huh... I never knew about the Ctrl-B command either. I just worked around with it and I see what you're talking about. So I'll just position the labels for each track, then export multiple. It seems there may be a way to make a cue sheet (I googled it a bit - Audacity lets you "export labels" in a text file, and then you can convert the txt file to a cue sheet if you want to track the source file again later on).

I haven't had many issues with the tags/metadata getting screwed up. I also use Mp3tag with most of my stuff anyway, so I can use that if need be to add/edit the tags afterward. Just have to try it and see what happens.

[morst]

I haven't had many issues with the tags/metadata getting screwed up.

I think it does things "right," but I don't work with the same work flow it expects. I often keep the audio from before and after the part I want to share, but I don't want it numbered in the scheme of the tracks I'll share.


Letting audacity number tracks before I have had a chance to cull the intros and set break gaps doesn't help me. I have to rename tracks later anyhow...


Yeah the EXPORT LABELS feature is handy. I save it as info.txt and then search and replace the digits and tabs (with nothing) so I generate my track list, without having to ever type titles twice.

[cmstewart]

One issue I'm having (which I kind of anticipated) is I recorded in 24/48.... and I did extensive peak editing to avoid any clipping in all 150 minutes of the show. When I export and dither the full project from 32-bit float to 24/48, everything comes out perfect and clean. But when I export/dither the entire project to 16-bit, 44.1 it comes out with a ton of tiny instances of clipping. REALLY pisses me off since i took days of work to avoid any of that.

But if I take the already-exported 24/48 WAV file that's clean and re-import it, and then export it again as 16/44.1 it comes out (almost) clean. I know that means I'm dithering it twice, but I seriously hate the thought of having to re-edit everything to get the 16-bit one clean as well. Maybe I'm being perfectionistic, since the clipping is clearly just barely touching the line. But idk. Have you ever run into an issue like that? Would it be a bad idea to just use the converted 24/48 WAV file over the full project 16/44.1 exported file?

Hope all that made sense.

[morst]

Would it be a bad idea to just use the converted 24/48 WAV file over the full project 16/44.1 exported file?

When I record at 24/48, I release at 24/48. ¯\_(ツ)_/¯

[cmstewart]

Would it be a bad idea to just use the converted 24/48 WAV file over the full project 16/44.1 exported file?

When I record at 24/48, I release at 24/48. ¯\_(ツ)_/¯

Well, I was thinking of releasing both. As a courtesy to Crows fans, I know many still listen by burning to CDs. And I probably will do the same when my car gets outta the shop.

[morst]

Well, I was thinking of releasing both. As a courtesy to Crows fans, I know many still listen by burning to CDs. And I probably will do the same when my car gets outta the shop.

That's very nice of you. I figure that most CD burning software can do it from 24/48 files, but I know there is some legacy stuff that won't. It was not so long ago that I was the one asking TaperJoe for 16-bit files suitable for burning!??! Now the shoe is on the other hand. 

[cmstewart]

I've actually never burned 24-bit to a CD honestly. I guess it's cause so much of my stuff is normal 16-bit FLAC. Not sure why but everything seems to end up floating around in that format, so I'm just putting it out there for whoever wants it.

I've kinda solved my issue, as best I can. Should finish it up soon.

[cmstewart]

To finish up this thread I just have one final question (the last step in this whole process I assume). Before I ask I just want to thank EVERYONE on here that helped me navigate this process to make this recording possible.

So, now that I have both a 16-bit and 24 bit version of the show, where so you recommend I share it? I know CC isn't approved for archive.org. Would it be cool to share it on here with you guys? I assume other Crows fans here would want to hear it... and I'd love to hear feedback from y'all !! 

[Fatah Ruark (aka MIKE B)]

If they are open to taping I'd share on etree (bt.etree.org).

If not I usually upload "not taping friendly" bands to either The Traders Den or Dime A Dozen.

I've also been archiving my recordings by:

1. Uploading a zip file of the MP3's to mega.co.nz (I do this for the technically impaired that don't know what FLAC's are and don't know how to torrent).
2. Uploading the full show audio to YouTube by creating a "video" with the finished audio with either a scan of the ticket stub or photo as the "video" part.

[cmstewart]

If they are open to taping I'd share on etree (bt.etree.org).

If not I usually upload "not taping friendly" bands to either The Traders Den or Dime A Dozen.

I've also been archiving my recordings by:

1. Uploading a zip file of the MP3's to mega.co.nz (I do this for the technically impaired that don't know what FLAC's are and don't know how to torrent).
2. Uploading the full show audio to YouTube by creating a "video" with the finished audio with either a scan of the ticket stub or photo as the "video" part.

Thanks man! I'm pretty sure etree is for torrents right? I do have a step-by-step guide to making torrents from a thread on here, so it'll be my first but I can give that a shot. Counting Crows are/have been for years a taping-friendly band. Their official policy allows both audio/video recording at all their shows.

[Gutbucket]

One issue I'm having (which I kind of anticipated) is I recorded in 24/48.... and I did extensive peak editing to avoid any clipping in all 150 minutes of the show. When I export and dither the full project from 32-bit float to 24/48, everything comes out perfect and clean. But when I export/dither the entire project to 16-bit, 44.1 it comes out with a ton of tiny instances of clipping. REALLY pisses me off since i took days of work to avoid any of that.

But if I take the already-exported 24/48 WAV file that's clean and re-import it, and then export it again as 16/44.1 it comes out (almost) clean. I know that means I'm dithering it twice, but I seriously hate the thought of having to re-edit everything to get the 16-bit one clean as well. Maybe I'm being perfectionistic, since the clipping is clearly just barely touching the line. But idk. Have you ever run into an issue like that? Would it be a bad idea to just use the converted 24/48 WAV file over the full project 16/44.1 exported file?

The part in bold shouldn't be a problem.  Odd that you got clipping upon direct export of the 16 bit file though, as it's supposed to only chop off the least-significant bits (the quiet end of the spectrum, which should be nothing but random noise down there at the bottom), and not supposed to affect the most-significant bits (the loud stuff).

Dither affects only the least-significant bit above the target bit depth of the output file.  So the dither which was applied when converting from 32bit (the internal working bit depth of the editor) to 24 bit is thrown away along with the other 8 least-significant bits (bits 24 through 17, if counting downwards from 0dBfs) when further reducing the 24 bit file to 16 bits.


Folks like CDWave because it is super straightforward, quick and easy to use, and often more straightforward for tracking than doing so within the editor.  It can also be advantageous when producing both 16 and 24 bit file sets because you can export both from the editor as single files, do the initial tracking on one file set, save the cue-sheet with the split info, then apply that to the other fi. It automatically places the splits on CDR sector boundaries, so you needn't worry about having your 16 bit tracked files fully CD compatible.  In some other software you'll need to specify split on sector boundaries when tracking 16 bit files intended for CD burning.

[cmstewart]

The part in bold shouldn't be a problem.  Odd that you got clipping upon direct export of the 16 bit file though, as it's supposed to only chop off the least-significant bits (the quiet end of the spectrum, which should be nothing but random noise down there at the bottom), and not supposed to affect the most-significant bits (the loud stuff).

Dither affects only the least-significant bit above the target bit depth of the output file.  So the dither which was applied when converting from 32bit (the internal working bit depth of the editor) to 24 bit is thrown away along with the other 8 least-significant bits (bits 24 through 17, if counting downwards from 0dBfs) when further reducing the 24 bit file to 16 bits.


Folks like CDWave because it is super straightforward, quick and easy to use, and often more straightforward for tracking than doing so within the editor.  It can also be advantageous when producing both 16 and 24 bit file sets because you can export both from the editor as single files, do the initial tracking on one file set, save the cue-sheet with the split info, then apply that to the other fi. It automatically places the splits on CDR sector boundaries, so you needn't worry about having your 16 bit tracked files fully CD compatible.  In some other software you'll need to specify split on sector boundaries when tracking 16 bit files intended for CD burning.

Thanks for taking the time to help with that. I too am a bit baffled by the clipping after dithering problem. I just resolved it by editing a couple sections before exporting the 16-bit version.

I also had a similar problem when decided to cut off the first minute of the pre-show music, and then shifting the tracks back. When I exported it after moving them, they summed together a bit differently and a couple peaks suddenly were clipped. In audacity, it shows you with red vertical lines where the clipped bits. I just edited those bits again and export again, keeping it clean. But I can't quite explain it. I kind of just assumed that the dither (which I believe is a low-level noise of some kind used to mask the truncated frequencies) added some fraction of a decibel to those bits and thus pushed them over the edge. Not much mind you, but still it was showing as being clipped. Luckily after all the editing it worked out and sounds really good. Took some work but I feel it was well worth it.

I'd love hear some of your recordings using your omni-setup to hear the possible difference with those verses the condensers I used. Or maybe have you take a listen to mine and tell me what I should possibly tweak or try, or if things sound pretty good as is. Sometimes listening helps more than just talking about it.

[Gutbucket]

I also had a similar problem when decided to cut off the first minute of the pre-show music, and then shifting the tracks back. When I exported it after moving them, they summed together a bit differently and a couple peaks suddenly were clipped.

That seems to me like you are summing multiple channels and may have cut a slightly different length from the start of each in making the shift.  Not enough that you would visually notice a shift or such that they sound obviously misaligned afterward, but just enough that some high momentary peaks sum differently.  This can happen depending on how you specify making selections and cuts, which is somewhat different for each editor.  For example, if you have, select between zero-crossings (or the equivalent) specified in your editor, the cuts to each separate object are made at whatever zero-crossings are closest to the edit points you specify, which is likely to differ by a few samples for each object.  If you then shift both cut objects in time to the beginning of the time-line such that both align with the zero-point of the timeline, they will be misaligned by the same amount.

Not saying that's definitely the case, just outlining one potential way it could have occured.  And to be clear- specifying select on zero crossings is often what you want when a cut occurs within the piece without a fade, as it will eliminate a potential click at the edit point.

An easy way to avoid that is to trim the heads and tails (the unwanted pre-show and after-show portions) prior to alignment and editing, or to simply leave them there and start your tracking not from the very start of the raw audio file trimmed to length, but from wherever you want the initial fade-in for track 1 to occur.  Otherwise you need to be careful in how you select, cut, and move the objects to keep them aligned with each other.

In audacity, it shows you with red vertical lines where the clipped bits. I just edited those bits again and export again, keeping it clean. But I can't quite explain it. I kind of just assumed that the dither (which I believe is a low-level noise of some kind used to mask the truncated frequencies) added some fraction of a decibel to those bits and thus pushed them over the edge. Not much mind you, but still it was showing as being clipped. Luckily after all the editing it worked out and sounds really good. Took some work but I feel it was well worth it.

The dither is way too low in level to add in that way. It is completely swamped by the far higher signal in both sources.  You're only going to get increased peaks where both sources have aligned peaks with similar levels.

I'd love hear some of your recordings using your omni-setup to hear the possible difference with those verses the condensers I used. Or maybe have you take a listen to mine and tell me what I should possibly tweak or try, or if things sound pretty good as is. Sometimes listening helps more than just talking about it.

Definitely does!  Happy to give a listen.  Most of my stuff is not posted, but I'm working on shifting towards doing so, mostly for folks here.  I think most productive is a hand-in-hand combination of discussion and listening to really get a good grip on what's going on.

[cmstewart]

Thanks for the response. Sorry for my delay... the start of the school-year kept me busier than usual. It was a crazy week!

Your assessment on how shifting caused the tracks to sum differently is pretty much exactly what I was thinking... you just worded it better. But I'm not sure it's because of the same reasons that you described. When I cut out a section at the beginning, I first selected to "sync-lock" the tracks, and then cute both at the exact same point. Then deleted the beginning portion, and while still sync-locked, shifted them back. I wasn't worried about the zero-crossing points because I was going to fade in from the beginning in a later step, so the points I cut at were going to be fading in from zero anyway. But, I do suspect that there may have been something (somehow) that got slightly offset which made it sum differently anyway. Audacity has given me some weird things like that when using multiple tracks before.

I know the dither is supposed to be basically in-audible, and at an incredibly low-level. It's just really odd that when I exported the exact same track without any alterations in 24-bit/48k and 16-bit/44.1k the 16/44.1 will suddenly have some clipped peaks, and the other is clean. My best guess besides the dither is when Audacity resamples it, there's some rounding that went up instead of down on those specific peaks, and made the amplitude a "bit" louder? Whatever it was, clearly the resampling did something to cause those clips.

I have some external links to my show. I also want to share it as a torrent (I think I'll use DIME at this point). I haven't shared torrents till now, it'll be my first one. But it seems like it'll circulate better that way. I can PM you those external links if you want them. The things I'd love to know is how the mix sounds - there are some occasions where I think I hear some phasing. I can't be sure though because it's only on vocals, and i know they use a phaser on his vocals sometimes anyway. Also the mix of ambiance and the band. If it would be better to have more ambiance, I may turn my condenser mic backwards next time. It it needs less, I may turn up the shotgun and the condenser down. Your advice would help a lot!