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SPL Meter - is this one OK ?


davd_indigo

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I'm finally realizing that I need some sort of way to set some levels for recording. Right now I'm only recording on my Sony PCM-D100 - its on a boom stand about 70 inches from the highest C on the piano. And it occurs to me that I need to establish some standard levels. And it seems like a SPL meter would be the tool.

 

Is this one a good tool ?

 

https://www.amazon.com/gp/product/B01A81RLYI/ref=ox_sc_act_title_3?ie=UTF8&psc=1&smid=A3FD19ALDFYBO5

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I'm finally realizing that I need some sort of way to set some levels for recording. Right now I'm only recording on my Sony PCM-D100 - its on a boom stand about 70 inches from the highest C on the piano. And it occurs to me that I need to establish some standard levels. And it seems like a SPL meter would be the tool.

 

The meter that you're looking at is sold by a few different companies. It's OK, not fabulous, but it's not the tool you need. An SPL meter of that grade is useful if you're running the sound at a live show and you want to know if it's getting up to a dangerous level. It's not accurate or complete enough to, for example, use to get an airport to change flight paths or for a county to build a sound barrier between your house and the new 8 lane highway they built right up to your property line. It's probably OK to calibrate your monitoring system so that you know that, for example, that pink noise recorded at -20 dBFS plays back at 85 dBA at your listening position (and that's pretty loud).

 

What you need to do is set the record level on your PCM-D100 correctly. With your recorder placed 6 feet above your piano, you'll have a hard time seeing its meters, and even if you can, you won't be able to adjust the level while you're playing. Get a ladder and a friend to help you, jot down the number on the record volume knob, and use that setting when you're recording your piano. Alternatively, consider using external mics and putting the recorder where you can see and adjust it.

 

I'd suggest trading your D100 for a TASCAM DR-44WL which you can control remotely, including seeing the meters, via WiFi from a phone or tablet, but the D100, as a recorder, is probably marginally better.

 

I have the Audio Control SPL meter (costs a buck, and it's styled after the famous and no longer available Radio Shack meter) on my cheap Android phone, and I use it to check the level at a live event or, if it's annoyingly loud, in a restaurant. I have a decent dedicated SPL meter and I used that to calibrate the meter on the phone. It took a fairly substantial change from the default calibration setting to get it to agree with the real SPL meter. Some apps, including the Audio Control one, attempt to turn off the limiter on the mic so that it won't flatten out above 90 dB SPL. Some Android phones will allow turning the limiter off, some won't. Since the Audio Control meter, part of a whole suite of audio measuring tools, was originally designed for the iPhone, I'm sure it's reasonably accurate and "unlimited" up to painful SPL.

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I don't understand how you can set recording levels using a sound level meter. Can you please explain your thinking?

 

Seems to me you need to set your recorder to your playing and not your playing to your recorder but maybe I'm not understanding.

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Thanks guys (Mike especially). I also had the idea to get a friend over to check (whatever) levels while I play. I plead ignorance in audio technical matters. I have played piano for 55 years and study music daily in a variety of ways. I am about to finish a course called "Fundamentals of Sound" at the local community college (they have a 2 year certificate program in Sound Engineering). This course has equipped me more with what I will call "informed questions" rather than solid understanding. I plan, after finishing this course, to do some remedial learning - I will next take a course, on coursera , in Basic Electronics.

 

I am retired and my goal in this area is to simply learn to make better (hopefully eventually excellent) home recordings. I am engaged in what I like to call a "slow burn" effort in this. Learning slowly over time. But I did have the idea that an SPL meter would enable me to develop some standard levels - like a standard volume to set my digital piano.

 

I won't be getting rid of my little Sony tank any time soon though. I love it.

 

 

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Seems to me you will be going about this kinda backwards. I suggest you rethink your approach.

 

Forgetting recording for a second, you need to be able to play your instrument at a level that both works for your environment and that will be inspiring for you as a musician. As long as you do that it doesn't matter what that level measures on a sound meter.

 

Then after you have done the above you set the levels on your recorder. There are two things to watch for when setting your recording levels. If you set them too high (allowing too strong a signal) you will get distortion. If you set them too low you will begin to hear a little hiss and noise. If you don't hear distortion or noise then you are in a good place. Digital recorders have far more headroom than we did in the old days with tape where setting levels was very important.

 

The technical ideal would be to record as hot as possible and end up just a hair below distortion. That will give you the best possible signal to noise ratio. But like I said, if you don't hear distortion or noise then there really isn't much of anything to worry about. You can always make adjustment after you record to make any improvements needed.

 

So save your money. This isn't a job for a sound level meter.

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I recently recorded 4 or 5 pieces and made a 20 minute CD. I gave several copies to friends and family at Thanksgiving. When I heard my CD played next to a Shawn Colvin Christmas CD - mine was way too low. I haven't noticed any hiss - this recorder seems really clean. Anyway, the only thing I know to raise the level is to normalize. But isn't that just a generic sort of whack - everything the same ? Is there something I should do to get a reasonable sound level ? I know you guys always talk about compression - negatively.

 

I have raised the recording level on the recorder, so that takes care of future recordings - hopefully. But the recordings I did previously that I liked are the ones I want to bring up.

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Normalizing may bring your "apparent loudness" up (or not). Depends mostly on the dynamic level of your recording. However most commercial recordings are made "louder" in the mastering process using a brick wall limiter/maximizer.

 

I'll bet if you listened to the mix before mastering on that Shawn Colvin record it would also be much lower.

 

Btw... if you record soft and also recorded loud after normalizing you would mostly end up the same. You can always go up. But if you get clipping during the recording you can't really fix it. It's pretty safe (and nearly standard practice) to record way low when using digital recording equipment ... especially if you are recording 24 bit.

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I am about to finish a course called "Fundamentals of Sound" at the local community college (they have a 2 year certificate program in Sound Engineering). This course has equipped me more with what I will call "informed questions" rather than solid understanding.

 

It's really good that you're learning how to ask questions, and what questions to ask. If, in the class, you were told "do it this way" you might get to what works right for you.

 

I did have the idea that an SPL meter would enable me to develop some standard levels - like a standard volume to set my digital piano.

 

I didn't realize you were using a digital piano. You can probably rely on the repeatability of the piano's volume control to give you the same nominal level each time you play. Also, your ears will get accustomed to what the "right" volume it. A piano has a lot of dynamic range, though, and the meters on your recorder read the peak level, so that's what you should be watching, at least until you get some practice in interpreting the meters and setting the record level accordingly. And on that topic:

 

The technical ideal would be to record as hot as possible and end up just a hair below distortion. That will give you the best possible signal to noise ratio. But like I said, if you don't hear distortion or noise then there really isn't much of anything to worry about.

 

and further

 

f you record soft and also recorded loud after normalizing you would mostly end up the same. You can always go up. But if you get clipping during the recording you can't really fix it. It's pretty safe (and nearly standard practice) to record way low when using digital recording equipment ... especially if you are recording 24 bit.

 

In the days of tape and 16-bit analog-to-digital converters with everything above 13 or 14 bits being masked with noise, there was an advantage to recording at as high level as you could get away with without getting distortion. There was more signal that could be heard above the noise level. But with today's 24-bit converters (that still have noise down in the last couple of bits), you can record with no peak going above -20 dBFS (that's dB below full scale - all the bits turned on) and still have about the same signal-to-noise ratio as an honest 16-bit recording even when you boost the level in playback (or normalizing) to near full scale. Keeping all peaks below -20 dBFS is pretty conservative, but if you play as loudly as you expect to play on the piece you're recording and set the record level so that you hit around -10 dBFS, you'll have some headroom in case you get a little more enthusiastic when you start recording.

 

Oh, and one other thing. Since you have an electronic piano, have you experimented with recording its outut directly into the line inputs of your recorder? Recording with the mics allows you to record the sound of the piano in the room, but this usually isn't very effective, or rather, tends to be more of a special effect, when the sound source is a loudspeaker. Mics are, of course, the way to record an acoustic piano, and it takes some experimenting to find where the mics pick up the best combination of the direct and room (or reverberant) sound. You might find that you get more consistent recording by going "direct" and adding a touch of artificial reverberation afterward.

 

 

 

 

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I recently recorded 4 or 5 pieces and made a 20 minute CD. I gave several copies to friends and family at Thanksgiving. When I heard my CD played next to a Shawn Colvin Christmas CD - mine was way too low.

 

I have raised the recording level on the recorder, so that takes care of future recordings - hopefully. But the recordings I did previously that I liked are the ones I want to bring up.

 

This is Standard Question #1 when people start to compare their recordings with commercially issued recordings. The simple answer is that those recordings have been "mastered" and yours haven't. Mastering almost always involves some compression and limiting which brings up the averagel level while holding peaks to just below clipping. Since our ears perceive average level as volume, not peak level, this results in a louder sounding playback. Sometimes in mastering peaks are allowed to clip a bit so that it sounds even louder due to the distortion, but that's a different perceptual effect.

 

So let me ask here - You must be using a computer to assemble the CD. What program are you using for that? There are a lot of computer-based tools that you can use to increase the apparent loudness of your finished recordings and you can use your own judgement as to when you've gone too far. Audacity is a free audio editing program that has a lot of useful sort-of-mastering tools for you to experiment with. If you're just using a simple CD burning program that just gets the Wave files from your recorder on to a CD, you might want to try to do some massaging.

 

Another thing that you might try is turning on the limiter in your DR-100, pushing the record level up so that the meters are hitting peak level frequently (though not all the time). I think the DR-100 has the same limiter function as the DR-50, and this works differently from a more conventional limiter. What it does is actually records a second stream about 10 dB lower than what goes directly into the final recording. When it detects clipping during the recording, it automatically replaces the clipped section with the "back-up" duplicate, normalizing the spliced-in segment to full scale, but without clipping. It works really well as long as you don't overwork it, and you can usually increase the average level of your recordings by 6 dB or so without losing any quiet parts.

 

Play with it. That's what real recording engineers do. ;)

 

 

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I have been using Sony Sound Forge Audio Studio. Mostly to chop off unwanted noise at the beginning and end of a recording - before and after the recorded song. I just received Corel's Roxio Creator NXT 5 in the mail this morning. It's loading on another computer right now. I will be trying it out for the same Wave files I used to create the CD I referred to earlier. I guess I will normalize the files to get the sound levels up.

 

I also have Mixcraft 6 which I've used on a limited basis to work with a few multitrack songs recorded on my Tascam DP32SD.

 

I am open to looking into another software is it would be beneficial to making higher quality CD's. Maybe I should look into Audacity, it seems to have good support.

 

Do you have any recommendations (Mike or anyone else).

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Sound Forge has been my go-to program for audio editing and CD burning. For multitrack work, mostly I record on my Mackie hard disk recorder and mix through an analog console, but when I need a computer-based DAW, Reaper has been my workhorse, though I have a copy of Pro Tools 10 in case I have to use Pro Tools. Lately I've been playing with Mixbus from Harrison Consoles. It's more like working on a real console than other DAWs and its built-in EQ and dynamics sound fine.

 

To me (and this should be the case for everybody), when it comes to software, the most important thing is how easy it is for you to work with, and that differs from person to person. Given your age and background, you might want to give Mixbus a look. There's a free annoyware demo version, and the program officially sells for $79, but they frequently have deep discounts. I just got Version 3 for $29.

 

 

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Regarding the limiter, it has a 150 ms, 1 sec, and 1 min setting. I'm thinking 150 ms would cover a hard hit accent on the piano. BTW, I've had the "S/N 100 dB" set - which precludes using the limiter. I have no idea which is "better" to use. I made some recordings tonight with the S/N set - maybe I'll try doing a recording tomorrow (with the mics in the same position on the boom) with the limiter set. Like you said trial and error.

 

I loaded the Roxio Creator 5 NXT software today. I will try burning some CD's tomorrow.

 

When you say Sound Forge, I'm thinking you must mean something like Sony Sound Forge Pro. Version 11 is on Amazon for $369. I might look into something like that later down the line - maybe to use in a different PC. I'm hoping to do little mico-runs of personal CD's for family and friends maybe a couple of times a year.

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I don't know what the "S/N 100 dB" setting does, so I can't comment on that.

 

I've stopped my Sound Forge upgrading at Version 10, but if I didn't have it, I could easily live with the $60 Audio Studio version. Since it supports the VST plug-in format, if you need more signal processing than the basic tools that come with it, or want to try alternatives, the door is open. And there are plenty of inexpensive (or even free) DAW plug-ins to play with.

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I found an academic version of Sound Forge Pro 11 for about $200. I'm about to take my final and won't be a student next term, but hope to take Sound Recording 1 in either June or September. And they are in the process of constructing a new building with a new studio. My current teacher said that Raul Valery, the teacher in SR1 will be teaching us how to calculate the Q. I still can't quite understand what the Q is. But I do know that with a narrower bandwidth you have a higher Q and wider bandwidth a lower Q. Just one example of a concept I"m hazy about, but will gradually grasp a bit more over time.

 

I became mildly obsessed last night with recording a Christmas CD. I moved the D100 on the boom behind my right shoulder. The mics are about 70 inches off the floor. I did "Blue Christmas" last night. Kept working on recording worthy versions of songs today. I'm having to record from beginning to end, so I allow some blemishes in the recordings. If I have some sort of train wreck, I stop the recording. But what excites me is the turn around time. I recorded 9 songs (Blue Christmas twice) and was able to crank out a giftable CD. I love it. Oh , I got an "Ultimate Christmas Fakebook" by Hal Leonard which gave me the basic guide for the tunes.

 

Oh and when listening to the songs in my living room, I thought there was a booming frequency ( is that called a resonant frequency ?) maybe B or A and octave above middle C (on the piano - sorry). When I listenened on my home stereo I didn't notice the boom.

 

I will probably put the recordings for below on Soundcloud. But I love having an easily cranked out CD.

 

The songs I recorded:

PleaseBeHomeForChristmas (Charles Brown)

BlueChristmas

SilentNight

JoyToTheWorld

WinterWonderland

LittleDrummerBoy

SilverBells

AwayInAMangerMaybe

BlueChristmasRevisited

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Q is just a number. It's calculated by dividing the frequecny of the peak or dip in the frequency response of the filter devided by the bandwidth of that peak or dip. The bandwidth is defined as the difference between the frequencies above and below the peak where the level changes by 3 dB.

 

I just happen to have an article in the current (January 2017) issue of Recording magazine all about the terminology and specifications associated with filters and equalizers. It's the latest edition of my occasional series "Demystifying The Spec Sheet."

 

"Boominess" in a room is usually a result of a specific frequency bouncing off the walls and arriving at your ears in phase with that same frequency arriving from a different path. The two add, making that frequency sound louder. The opposite can (and often does) occur where the sound from two paths arrive at your ears out of phase, causing them to subtract and causing a weakness of that frequency. Since the phase relationship depends on the path length of the two waves, it will be different in every point in the room. If you find a boom, there will be a place where that frequency will be weak.

 

It is, in essence, a resonance since standing waves result at frequencies with a wavelengh that's a multiple of the distance between two opposite and reflecting walls. It's the same principle that makes an organ pipe play the note that it does.

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Here's the Christmas collection. It's only piano and voice. My "mix" consisted of trying to have the mics (up on the boom stand on the D100) somewhat in proximity (i.e. sort of facing) my mouth. Very technical (insert eye wink). I recorded at 16 bit/44.1 kHz. It occurred to me that I may try (in future recordings) higher resolutions.

 

I will ponder your quote about boominess. Thanks Mike.

 

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