Dealing With Volume Spikes In The Recorded Waveform

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Dr. Varney

Dr. Varney

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Hi again, guys! :) Having taken a break from recording, now coming back to it, I thought it about time I popped in to catch up with everyone here.

I'm digitally recording voice samples for an audio play, so the main bulk of the 'song' (so to speak) constitutes the spoken word.

Most of the time I'm getting good results recording the people who volunteerto act in the play. Some people have more natural ability than others in projecting their voice and in reading aloud from a script - and so, now and again, I'll get someone whose voice trails off at the end of each sentence; perhaps starting the passage with an overly enthusiastic 'attack' and sort of trailing off into the distance.

Now I am wondering about the best way to go about fixing the problem.

A typical waveform will look like this:
fledisonwf001.jpg

(The Edison recorder, as supplied with FL Studio 8 Producer Edition).

The selected part (in red) is the part I have manually but very painstakingly altered using the amp tool - spike by laborious spike. This takes too long. I will be here for the next five years, as I have literally hundreds of wave files to edit.

The green band in the middle is my target envelope; a rough ceiling for the sort of size I want the spikes to fall within.

Can anyone please suggest an efficient method for making the waveform roughly equal in volume all the way through? What sort of fixes and tweaks do YOU apply on your recorded stuff?

Do you try to fix it at waveform level and resave the .wav file, (as I have) or do you tend to run it through some sort of 'gating' device in the mix afterwards?

I have various limiters and compressors and whatknot at my disposal, which I have no idea how to use - hence why I have meticulously selected each spike and applied simple amplitude (both neg. & pos.) But this is not really very time efficient for me. I'm not in a big hurry to finish the piece but now the task is starting to drag on rather longer than I had hoped.

Any advice gratefully received.

Dr. V
 
Hey Varney. Here's a long post I made on compression a few weeks ago, should have everything you need...

You may already know this, but just in case you don't, I thought I'd wreck your head by writing it out :D



The parameters of a compressor affect how it works (duh), and it's very important to know exactly what they do in order to get your desired effect.

Threshold: Sets the level (dB) at which the signal starts compressing. So, if you set your threshold to -20dBFS (probably written as just "-20dB"), when the signal is below -20dBFS it will be left alone by the compressor, but once the signal goes above that level, it will be compressed by the amount you set in your ratio.

Ratio: The amount of compression applied to the signal once it goes above a threshold, indicated by a ratio (funnily enough). So, if you have a threshold of 3:1, for every 3dB the signal goes above the threshold, there will be a resulting 1dB output. I've seen some compressors where the ratio goes backwards (such as 1:3), and I'm assuming it acts like an "expander" which is the opposite of a compressor.

Knee: This is the one I'm not entirely clear on. Some compressors offer you a setting in dB, other compressors offer a setting of hard/soft knee. If the compressor has a hard knee, then it won't compress until the signal hits the threshold exactly. If it has a soft knee, it will start compressing slightly once it nears the threshold, and increase the amount of compression up to (and slightly above) the threshold. If anybody knows how the "dB" option of a knee works, I hope they chime in and explain it because I want to know :D

Gain reduction: This isn't a parameter that you can adjust, it's a meter of how much compression you're applying, or in other words: how much you are turning the signal down with compression.

Attack: Sets the amount of time it takes the compressor to start fully compressing once the signal goes above the threshold. It's usually measured in milliseconds, and sometimes it goes down to microseconds. (I'd beware of microseconds, they can sometimes cause clicks and pops when set too short). So, if you have an attack time of 10ms: once the signal goes above your threshold, it will start gradually reducing gain more and more (to the amount you have set) over that amount of time.

Release: Kind of the opposite of attack. This sets the amount of time it takes the compressor to "release" from gain reduction once the signal falls back below the threshold. The release time is normally much longer than the attack time, and I've never seen it measured in microseconds (usually 50 milliseconds up to about 3 seconds). If you have a release time of 200ms: once the signal falls back below the threshold, it will take 200ms for the compressor to release gain reduction. Remember to set the release short enough so that the compressor releases gain reduction before the next transient comes in, otherwise the signal will be in constant compression and thats not good (or it could be - play around with it). It releases gradually over the release time - this is what makes it different from "hold."

Hold: This is almost the same as release. The difference is that it doesn't release gradually. Once the signal falls below the threshold, it will "hold" the amount of gain reduction in place for the amount of time you set, and then the compressor will go into the release time. So, if you have a hold time of 50ms and a release time of 100ms: once the signal falls below the threshold, the compressor will hold the gain reduction in place for 50ms, and then release the gain reduction gradually over 100ms. The "hold" parameter isn't as common on compressors as attack and release times. At least if you come across one, you'll know what it does ;)

Makeup Gain: Sometimes referred to as "Output" - This one is simple. Normally, what you're trying to do with a compressor is even out the peaks, but you don't want to make everything quieter. So, you use your makeup gain to turn up the signal after it has been compressed - effectively making the quiet parts louder (pretty much). I always find it's a good idea to try to match the average level of a signal with makeup gain before and after compression, so when you're A/B'ing the effect to see if it sounds better, it'll be just as loud as before (so you can see if it sounds better because it is better, and not just louder).


Sidechain: This one's a little tricky to explain (especially when I don't know what software you're using). Basically, you can set your parameters for compression, and then you can send another signal to the sidechain to trigger the compressor - so the compressor won't do squat until it receives signal from the sidechain. This is commonly done with the kick and bass in dance music. For example: The kick is left untouched through the song, but the bass can get in its way. So, you put a compressor on the bass, and insert the kick into the sidechain. So, when the kick isn't playing the bass is left untouched, but when the kick hits, the compressor will clamp down on the bass and turn it down to let the kick pass through. It can be a little tricky to get right, but when you do it can add a lot of excitement to a song.

Some compressors don't have a threshold or makeup gain setting. Rather, they have an input and output. This was very confusing to me at first but it's rather simple. Basically, the threshold is automatically set within the compressor, and you can't change it. Rather, you can turn up/down the input more/less until the signal hits the threshold where you want it to (you'll see gain reduction on the meter). And the output is just your output level :)



Now, on to what you originally asked (sorry about all that :D)

There are a few different uses for compressors.

Sometimes they're used to shape the "envelope" of a sound - if you're not familiar with envelopes, just ask and I, or another member here will be more than happy to explain. I normally use them on snares for this reason. I'll set a high ratio (about 7:1) and a fairly low threshold until it sounds right. Then, I'll set the attack time to between 30ms-50ms and a quick enough release. That way, when the snare hits, it should be going over my low threshold, and it'll take the compressor a few ms before it kicks in, and it allows the initial strike of the drum to come through loud and clear before it clamps down on it. It's a great way of getting snares to POP!

On the flip side, for an acoustic guitar (for example), you could set a quick attack time (about 3-5ms) and release time, set a low ratio (about 3:1), and threshold accordingly. If the initial pluck of a string is really loud, and the sustain is much quieter, this should do the job fairly well. It will clamp down on the pluck quickly, and then release to let the sustain of the string pass through nice and loud enough.


You could use a compressor to get a creative sound. One common technique with drums is to put a "room" mic in the room when recording drums, and then squash it with a compressor afterwards. The Urei 1176 is very common for this (I doubt you have one of those though :p). Set a low threshold and high ratio (10:1 or above), and play around with the attack and release til it gets really "pumpy", so that nearly everything on the signal gets compressed. Then, you slowly blend it in with the rest of your kit, low enough so that it's not really noticed when it's there, but loud enough so that it's definitley noticed when it's gone. It can add a lot of thickness to the drums. Alternatively, you could compress one or two room mics on the drums, and use them as your main kit sound. I can't think of any examples right now but I'm sure someone else in here will.



You could use a compressor for it's most basic use: Dynamic range control. Let's say you have a vocal that's really dynamic - going from quiet to loud in the same phrase. You don't want to kill the dynamics of the vocal, because that's what gives it it's energy (well, not all the time). But, you might want to control some of that level change. So by compressing the louder parts, you can use makeup gain to "make up" for the reduction in level, and the transition between quiet to loud isn't as big. Just remember, that when using it for this (common) purpose, you normally don't want to "hear" the compression. It should be transparent and smooth, not pumpy, so don't overdo it.

Although, compression isn't always the best thing for that job. I've gotten into a habit of manually gaining up/down individual phrases and words to a relatively similar level before compressing. That way, the compressor has less work to do, and it runs a lot smoother. Then again, after doing that you mightn't even have to apply compression.
Also, just to mention, bass guitar is probably the most commonly compressed instrument in mixing. The bass (along with the drums) usually provides the foundations to a song, and like any building, it's good to have a "steady" foundation. A good solid bassline, consistent in level, keeps the energy throughout a track and provides strength to the sound. Also, I myself find it very hard to compress a bass so much that it sounds bad within a mix, although I'm sure it's possible.


Last but not least: Parallel compression - This is common technique. You double the signal, and compress one and leave the other untouched. Sometimes the compression is heavy (I use heavy parallel compression on my drums sometimes, it's alot like the "room mic" thing I mentioned earlier). Sometimes the compression is light. It can be a great way of leaving the dynamics of a signal untouched, and having it sound more natural, but at the same time, the compressed signal provides a kind-of "backbone" so that if the natural one falls too quiet, the compressed signal is there to back it up and remain heard.


That's all I can think of right now, but just a couple of points:

Always play around with the settings - Play with the attack and release times and listen to what sounds best. Also, sometimes a low threshold and low ratio will work better than a high threshold and high ratio will. Things like that.

Don't hear the compression, unless you're going for a creative effect. There are few things that sound worse on a track than bad compression, or "pumping", unless it's done right and on purpose.

EQ before or after compression? Firstly, listen and decide if your track NEEDS EQ. If it does, then EQ it. Try EQ before compression, try it after, and see what sounds best. I personally like cutting all the crap out with EQ before compression, and then I'll EQ more creatively after compression. This way, by cutting out the crap before compression, the signal isn't being compressed by something that's not being heard. Then, by EQ'ing after compression, I'm not losing any of my tasteful EQ to compression. Also, when you compress, you can change the tone of a signal. You might want to compensate for that with EQ after compression. Once again, play around, see what works ;)

Does it need compression? - Common mistake by n00bz :) Listen to the signal before you touch it. Is it too dynamic? Does it need envelope shaping? Do you want it to have a creative, compressed effect? If it doesn't need compression, leave it alone. Don't compress something just because you think you should. Only compress it if it needs it.

Don't kill your dynamics - As mentioned earlier, you can suck the life out of a song by over-compressing everything, even if you don't hear pumping. It's not natural at all. How much dynamics are left in your music depends on the song, genre, instrument etc. Listen and decide.

Finally: Use your ears - This is the best advice for anyone when it comes to audio. "If it sounds good, it is good." Don't judge solely on meters. The quicker you learn to use your ears, the better your mixes will become. Remember, nobody listening to the final product will see your compression meters (or any meter for that matter), so they won't see how good your mix "looked" :D It's audio, so trust your ears more than your eyes.

Phew......... I need a cup of tea and a cigarette :D

Welcome to the board dude :)
 
Yup, compression to take care of the peaks and then fader riding or volume automation to get the overall levels to be where they need to be at any given time in the piece.

Might want to look at how you're micing the voicalists if there is any tracking left to do that is some huge spikes a compared to the RMS level that is what translates to perceived volume. Either the vocalist is really over emphasing hard consonants or leaning into the mic at the start of certain words or something wierd. At some points it looks like your getting a 400% increase in signal level at the spike/consonant or whatever it is. Also it looks like your close to or even clipping the recording which cannot be fixed after the recording is made

If you have access to a hardware compressor this might be a good time to apply some light compression on the way in if the vocalists have poor control/technique on the mics
 
Hey, Philbagg... That's awesome... Thanks!

Now, the Edison does feature it's own compressor script (the studio features 'live' compressors for mixing) which can be applied to the wave and saved. Thing is, it isn't visual, like the ones used in the mixer. It requires knowing what numbers to punch in and that's above where I am with compressors at the moment.

So, I've had an idea... I'll try re-recording the wave with some live compression, using the info you've kindly posted and see what happens. Gotta be worth a try.

Again - many thanks.

Dr. V
 
If you have access to a hardware compressor this might be a good time to apply some light compression on the way in if the vocalists have poor control/technique on the mics

Yes - as it happens, I do! I have a compressor, included as one of the side-chain effects, built into my mixer. I could actually try using it in future. Great idea, thanks!

Dr. V
 
One more thing - is a compressor really the right tool for dealing with volume variations, like what I have here?

Yes. Compression and volume automation.

Here's what I do:

Volume automation to catch the big ones (if there's only a few) and do a rough leveling of the signal. Then I'll use compression to get an even level throughout.

You mentioned you had a limiter too. That's basically a compressor with an instant attack and a brickwall ceiling (ie. nothing will get past the threshold)

Some people say that a limiter is any compressor with a ratio of over 10:1. I'd have to disagree with that. You can still go over the threshold, so it's not really a "limit".
 
You mentioned you had a limiter too. That's basically a compressor with an instant attack and a brickwall ceiling (ie. nothing will get past the threshold)

Some people say that a limiter is any compressor with a ratio of over 10:1. I'd have to disagree with that. You can still go over the threshold, so it's not really a "limit".

Ouch! I'm afraid I don't understand those two paragraphs. Maybe the bit about the ceiling...

Anyway, I've tried it but it hasn't worked how I thought. I roughly get the gist of this 'threshold'/'ceiling'/'ratio' thing but I don't understand how attack and release come into it.

I exported a .wav of the project, with the compressor/limiter (I'm still struggling to get my head around the difference between the two units) on and when I compared it with the uncompressed waveform, it looked exactly the same.

Basically, I'm not getting volume evenness with it - it just sounds like EQ when I do it.

To get any kind of result, I had to stack 'Limiter', 'Multi-Band Compressor' and 'Maximus' all together on the same send track (common sense tells me it's unnecessary). Yet, even with overkill, the new waveform doesn't look any different.

At this stage, all I'm able to do is fiddle with knobs, aimlessly, until it sounds 'different' but after a while I just find it all confusing and can't tell if I've improved things or just made them worse. They all feature a visual display but that isn't really helping much.

So from what you're saying - it won't limit large volume spikes? Just the little ones? The really big ones I suppose I could lower by using negative amplitude on them, in the sample and re-save the .wav.

I'm gonna stop now and read your article again... with a cup of tea and a cigarette! :)

Then I'll have a play with volume automation.

Dr. V
 
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Result!

Drew out a volume automation curve, using my pen tablet, following the rise and fall of the sample's waveform (in negative). Then found a preset on the limiter, called 'Transient Shaping'. I'm assuming 'transients' meaning: sudden, aberrant spikes, in the wave...?

Anyway, time will tell if this is quicker but it does seem a lot less faff than what I was doing before. Either way, it's certainly an improvement.

So thanks - your answers have inspired a good result.

Kind regards

Dr. V
 
So from what you're saying - it won't limit large volume spikes? Just the little ones?
It does compress all the signals that go above the threshold. He ment that if you for example use a compressor with a certain threshold and a high ratio like 10, the signals shooting over the threshold will be attenuated a lot, but they will anyway go over the threshold (though ten times less than without the compressor), when again, if you use limiter, the output will never exceed that threshold.

The attack and release times ARE hard to understand, so don't worry. If the problem is that you can't get the spikes gone, try first using as fast attack as possible (might be 0,1ms or 1ms), and quite a fast release (say.. 50-100ms?). That should make sure that compression really happens. However, now you probably will get some strange artifacts so you may need to start turning them slower untill it sounds good. (The final values will be probably in range: attack 0,1-50ms and release 50-500ms). It will just need a lot of experimenting. For me being able to hear those small differences is still very hard.
 
Before we get into all of this, there is something you must do: Turn off your computer monitor!

The shape of the wave has no real meaning. If you try and fit every voice into some sort of visual boundary you will not only waste time but you will also end up with boring voices.

So look away from the screen and use your ears. When you hear a wave that sounds wrong, then fix it with the techniques given here...again, without looking.
 
Before we get into all of this, there is something you must do: Turn off your computer monitor!

The shape of the wave has no real meaning. If you try and fit every voice into some sort of visual boundary you will not only waste time but you will also end up with boring voices.

So look away from the screen and use your ears. When you hear a wave that sounds wrong, then fix it with the techniques given here...again, without looking.

As I was waiting for the computer to fire up, I was just thinking: "I bet someone has already said listen more/look less"!

Yes,it hit me again, with a vengeance. The old rule. The one that got me this far in music, to date.

Though the waveform does speak to me. It tells me where the spikes are, in the timeline. I've learned to recognise where consonants; Ss and Ts fall, just by looking at it.

But yes - you are absolutely right, Chibi. I mostly tend to listen with my eyes closed but lately, I've become distracted, while trying to fix these vocal problems. I'm saving up for a general MIDI control surface, which means I'll be able to look away from the monitor entirely and make the important decisions, using my ears.

Dr. V
 
Thing is, it isn't visual, like the ones used in the mixer. It requires knowing what numbers to punch in
No, it requires developping your ears. The above statement and many others in this thread lead me to believe that you're relying on your eyes WAAAYYYY too much. Don't worry about how something looks, we're not painting. If you rely on "formulas" (which don't really exist), visual waveforms, and how your music looks, you'll never learn to use your ears.

Most decisions you make should be done before you even hit the record button. Do you HEAR the peaks that you see? If yes, then why are they being recorded in the first place?
Before we get into all of this, there is something you must do: Turn off your computer monitor!

The shape of the wave has no real meaning. If you try and fit every voice into some sort of visual boundary you will not only waste time but you will also end up with boring voices.

So look away from the screen and use your ears. When you hear a wave that sounds wrong, then fix it with the techniques given here...again, without looking.

Couldn't have put it better.:cool:
 
No, it requires developping your ears. The above statement and many others in this thread lead me to believe that you're relying on your eyes WAAAYYYY too much.

As I said - I've always used my ears far more than my eyes, when editing sound. The misinterpretation of my comments has arisen simply from the fact I have noticed the visual information reflects what I am hearing - and that sometimes, it doesn't tell me everything that my ears can. That's not to say I cannot become distracted by it sometimes - and in a DAW, that's very easy.

Most decisions you make should be done before you even hit the record button. Do you HEAR the peaks that you see? If yes, then why are they being recorded in the first place?

Because of inexperienced actors. With some, a line has to be drawn somewhere, as to what point the recording session stops being a recording session and turns into a voice training lesson. With time restraints being what they are, hence it is often decided to run with the take and fix it later. Not ideal, but that's just the way it is sometimes.

Some things are easier to address before the start. IE: When I meet someone whose Bs and Ps are likely to pop madly, on goes the furry mic jacket and EQ is applied on the way in, after the first dry run.

Dr. V
 
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Varney, just curious, did you really give my long post a good read-through? Go over it a few times, especially the section on parameters, as you seem to still be confused about attack and release. I felt I explained them pretty well. If there's some wording or whatever that you don't get, just ask me about it and I'll be glad to explain it further.
 
Varney, just curious, did you really give my long post a good read-through? Go over it a few times, especially the section on parameters, as you seem to still be confused about attack and release. I felt I explained them pretty well. If there's some wording or whatever that you don't get, just ask me about it and I'll be glad to explain it further.

I would remark that understanding what attack and release are and what they do doesn't yet give the ability to learn to set them right. That will need also good ears and a lot of practise.

I just read an excelent mixing book and I feel that I understand the concepts behind them really well, but when I have to actually set them, I often still feel confused. Mostly because the differences that you hear are quite small, and as you have said, you should do the decicions based on that and not the meters.
 
Dont squash the hell out of it or your spoken word will be sleeeeepy.

I would compress enough to tame but not go overboard. You still want it to breath.
 
I would remark that understanding what attack and release are and what they do doesn't yet give the ability to learn to set them right. That will need also good ears and a lot of practise.

I just read an excelent mixing book and I feel that I understand the concepts behind them really well, but when I have to actually set them, I often still feel confused. Mostly because the differences that you hear are quite small, and as you have said, you should do the decicions based on that and not the meters.

I also gave examples of what the different attack and release times too (The slow attack on the snare drum and the quick attack on the acoustic guitar). Then again, your best bet is to play around and just listen.

In the op's case, he's looking for pure volume control and nothing more (ie. not envelope shaping), so a quick attack and quick release will do just the job.
 
I also gave examples of what the different attack and release times too (The slow attack on the snare drum and the quick attack on the acoustic guitar).

I know, but it is still the same. After all you can't think "this is a snare drum so I need to use slow attack", and even if you would, 'slow' doesn't define how slow (in some context 10ms might be 'slow', and in some other 100ms might be slow, and there is a huge difference between those two).

Your post is very good and after reading it carefully one probably understands which kind of settings he probably should be using. It gives a starting point. But being able to finetune the settings needs much more than that.
 
I know, but it is still the same. After all you can't think "this is a snare drum so I need to use slow attack", and even if you would, 'slow' doesn't define how slow (in some context 10ms might be 'slow', and in some other 100ms might be slow, and there is a huge difference between those two).

Your post is very good and after reading it carefully one probably understands which kind of settings he probably should be using. It gives a starting point. But being able to finetune the settings needs much more than that.

Oh yeah I completely agree. There's nothing you could ever read/watch as a n00b and be able to mix as well as the likes of CLA straight afterwards. It's a starting point. I'd hope it'd be answers to a general FAQ about compression, but you gotta practice aswell. There's no substitute.

You can't learn about mixing without practicing, but you can't really practice without having a rough idea of what you're doing ;)
 
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