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Tips For Removing Breaths From Voice Recordings

May 13, 2013 by Ken 2 Comments

Recently, over at the Audio Daily Blog, the voice-over audio production blog at Voices.com, there was a post with suggestions about how to remove breaths from voice recordings. I figured I’d add my primary method for doing this to their list of 6 tips.
I typically record all my audio into Reaper, an outstanding multi-track recording program that rivals Pro Tools for a fraction of the cost. With music, and with move complex voice-over projects, I’ll do all my mixing in Reaper.
For voice-over jobs that require multiple parts – say a conversation between two or more people or a series of phrases that need exactly 3 seconds between them (I have done a lot of both types of jobs), Reaper is the tool I use to get everything in the right order and sounding good.

The Audio Editing Program

But after I create a single file by mixing it all down in Reaper (“rendering”), I open that single file in a different program – an audio editor – to do the final polish. Some folks consider this overkill, since so much can be done from within Reaper. But coming from a music recording background, I’m used to mastering my mixed-down/rendered audio. And to do that I use a separate audio recording software type called and audio editor. In my case, I use Adobe Audition. But there are much more affordable editors out there. I show you how to do this in Audacity (free) below, for example.

Adobe Audition

So once I open the rendered, “almost-finished” audio in my editor, I put the headphones on so I can hear every little breath, click, pop, etc. My primary tool within Adobe Audition for this is called the “Silence” tool (Effects > Silence). This tool removes all audio in a selected area, which is different from “deleting it!” If you delete the audio, you also delete that section of time. Silencing leaves the time alone, simply making it audio-free time.go through the audio, selecting breaths, and then silencing them with that silence tool.
Anyway, I actually silence all the sections between phrases to get rid of extraneous little noises that become audible when there is no vocal, including lots of the breaths. It works really well, and its fast.

Audacity

You can do the same thing in the free (but incredibly powerful) editing software, Audacity. The process is very similar. highlight just the breath sound between phrases and then locate the “Silence Audio Selection” button on the Audacity tool bar. See pic below.

I love that there is a quick, visible button to use for this in Audacity. Adobe Audition (since moving to Creative Cloud) removed the quick button thing, which used to be there in the desktop versions. So you have to use the menu drop-down. That actually makes it easier to do in Audacity! Anyway, once you remove the breath using the button, you can just “Repeat Last Command” (CTRL-R) for the next breath and the next, instead of having to click the button every time, if you think it’s faster.

True Things No Matter What Editing Software You Use

Heed this warning though. A lot of breaths are just part of the way we speak, especially the breaths before a phrase (both my wife and I tend to put a little exhale or quick intake of breath sound at the end of our phrases. Most of these get silenced out. But many of the breaths that come before a phrase get left in, since people are used to hearing people breathe before they talk.
One more warning – if you silence most of the space between phrases, the start of a pre-phrase breath may be cut off. THAT will make the breath sound unnatural. You can fix that problem either by making sure the section you select for silencing does not include the start of the breath. Or you can use the “fade-in” tool if the start of a breath gets cut off. Creating a fade-in makes it sound natural again.
I hope this helps you get much better-sounding voice recordings!
By the way, the Audio Daily Blog post I referred to at the beginning of this article is here: http://blogs.voices.com/audiodaily/2013/05/6_tips_for_removing_breaths.html

Review of the Audio-Technica AT2035 Microphone (Audio Samples)

December 4, 2012 by Ken 80 Comments

This is an excellent microphone for home recording, especially for vocals/voice-over purposes. It gives professional sounding results for only about $149. This review contains audio samples and comparisons with a more expensive mic. My Star Rating –

Audio-Technica AT2035 Mic — Audio-Technica AT2035

The Audio-Technica AT2035 is a large diaphragm condenser (LDC) microphone with a cardioid pickup pattern. (For a review of what a large diaphragm condenser mic is, see our post: What Are The Different Types of Microphones?)

The AT2035, which normally sells for $149, is an improvement on its less expensive sibling, the AT2020, which sells for $65, which is still a great value.

The capsule in the 2035 is a newer and larger design that gives a flatter response in the low (bass) frequencies and is overall more sensitive than the 2020. In fact technically, the AT2020 is not really a larger diaphragm condenser mic, even though it looks VERY similar. It’s only 16mm compared to the 24mm diaphragm in the 2035.

So What is Different?

For the extra 85 bucks or so you not only get a more sensitive mic with flatter low end and more sensitivity, but you also get a mic with less self noise, a much larger capsule, and some additional features thrown in for good measure. The 2035 gives you more control over the sound by offering two switches: a high pass filter (meaning that it reduces the low frequencies) switch at 80 Hz and a 10 dB pad, which lets you reduce the signal coming from the microphone by 10 decibels when you engage that switch. This function is pretty handy, especially if you’ve done all you can in a recording session to tame loud signals.

And that brings us to another improvement in this mic over the 2020, which is that it can handle a very loud signal (148dB of signal processing level (SPL) without the pad, or 158dB with the pad engaged) before it distorts. For a mic to be both very sensitive (can pick up very quiet sounds) and be able to handle that level of SPL is a good thing.

And finally, this mic comes with the shock mount in the box. This may seem like a small thing, but it gets very difficult to keep noise from traveling up a mic stand with sensitive mics like these if you just use a clip. A shock mount suspends the mic from a web of elastic bands, which keeps out a lot of low frequency noise. If you have ever had an LDC mic without a compatible shock mount, it gets to be a chore to find one and then order it and wait for it to come in, etc. So the fact that this mic comes with a shock mount is awesome and should be standard practice.

So How Does It Sound?

I always recommend that when they can afford it, folks get a large diaphragm condenser mic, along with a decent audio interface, if they want to produce professional audio. If you have less than $100 to spend, the AT2020 is an excellent choice. But if you can produce the extra 85 bucks, I highly recommend the 2035 because it sounds awesome.

I compared the AT to my Rode NT2-A, another LDC mic, one that I use daily. I recorded a short voice passage as well as some acoustic guitar. The first thing I noticed before anything else was that with everything else being equal (audio interface settings), the level of the AT was significantly higher than the Rode, which is saying something. That’s that sensitivity thing coming into play.

First the voice test. I just read some specs from the box for this. The 2035 sounded clear and present, very nice. Compared to the Rode (a mic costing almost 3 times more than the AT, by the way) the sound had a bit of mid-range hype to it. Take a listen to the samples below.

	Rode Voice
	AT Voice

For the guitars I found exactly the same thing, a mid-range bump and less even across frequencies. Take a listen to those below.

Rode Guitar

AT Guitar

Lastly, I recorded silence (well, the sound of my room without any talking, etc.) to measure mic noise, and corrected the Rode recording for the difference in levels. There was no significant difference in noise at all. both were very quiet, registering about -65 RMS average level of amplitude. This is pretty amazing considering the price difference between these two mics.

Overall

For the things I tested it on, voice and acoustic guitar, the AT2035 sounded great. If you weren’t comparing it to a mic that cost nearly 3 times more, the bumps in the mid range wouldn’t be noticeable. My advice is to get one of these mics as your first large diaphragm condenser if you can afford the $149 and if don’t have a really good one already. To find out more of go get one right now, click here.

You cannot go wrong.

What Does Compression Mean In Audio Recording?

September 29, 2012 by Ken 10 Comments

A very common question I see is “should I use compression in audio recording?” The best answer to this question is, as is so often the case, “Do you need compression in your recordings?”
Let’s start with a workable definition of compression in the audio recording sense. When you lower the volume of only SOME of your audio, usually the bits that are clearly louder than most of the rest of the audio you’re working with, you are “compressing” that audio. You would do this in order to allow you to raise the average loudness of the entire audio file.

How Can I Compress My Audio?

Now, this can be done manually, by which I mean you could open your audio in an editor, seek out all the areas where the wave forms (I like to use the term “blobs” instead) are loudest, then turn those bits down. But that can get REALLY tedious and time consuming. So to automate this process, a machine (nowadays done with software) called a “compressor” was invented. This allowed folks who really knew what they were doing to more quickly manipulate volume and loudness dynamics. The dark side of the situation, though, was that it allowed folks who were less expert to mess up their audio, and do it much faster and more efficiently than ever.

Compressor Settings?

There are all kinds of settings on a compressor that are better discussed in other articles. For a good idea of what those are, see my article and video – Vocal Compression Using Reaper’s ReaComp Effect Plugin.
For the moment I’d like to focus on a visual explanation of basic compression, since I strongly believe that those who use compressors to mess up their audio (usually without actually WANTING to mess up their audio) do it because they don’t have a really good grasp on what compression really is. This should help.

Lets Just Focus on What Compression Is

use compression in audio recording — Figure 1 – Uncompressed Audio

Take a look at figure 1, which is a voice narration file. I’ve boxed off the two loudest peaks.
If I tried to turn up the volume of this audio, all of the audio, including the 2 peaks, would “get bigger.” But there is a problem here. Do you see it? Look at the label in the picture pointing to Max Loudness. If any of the audio wave forms (blobs) get pushed beyond that border line the arrow is pointing to, you get nasty, awful digital distortion. Suffice it to say that the border line represents the upper boundary you must not cross.
Knowing that, it should be easier to see why you can’t turn the loudness up very far before the peaks hit that boundary. And when that happens, none of the rest of the audio can get any louder either. See figure 3 to see what would happen if we told our editor to turn the file up until the loudest bit hit the upper boundary. There wasn’t a lot of change was there?

The blobs didn’t get much bigger (louder) did they? Oh no, whatever shall we do?
How about if we turn DOWN the level of JUST those two peaks? Let’s do that. Now take a look at figure 3 to see what our new file looks like. Notice that the

average level of the audio blobs is much more even now? THAT is what compression is for. Now that we don’t have just a couple of pesky peaks preventing us from turning our audio up without it distorting, we can raise the average level much higher, resulting in louder audio across the entire file. If we now tell our editor to raise the level of the loudest peak to maximum loudness (raising the rest of the audio by the same amount), we get figure 4. Compare

the picture in figure 4 with the one in figure 2. You will notice that blobs in figure 4 are much wider/bigger than the ones in figure 2 across the board.
Being able to increase the overall loudness level of your audio is just one benefit of compression, and perhaps its most common goal. when used wisely, this can also add punch and “up-frontness” to your audio. But don’t forget what compression actually accomplished for us BEFORE we were able to turn it up. It evened out the overall levels first, which gives a more consistent listening level to the user.

How Compression Would Help When Watching TV

I’ve often wished I had an audio compressor attached to my television for this very reason. Have you ever been watching a movie where the action scenes were so loud that you had to turn the volume down on the TV, only to find that now the talking parts are too quiet? Then you have to turn the TV back up to hear those parts. You end up turning the volume up and down throughout the show.
What about when the commercials on TV are much louder than the show itself? Don’t you wish you had a compressor to even out the volume so that it automatically turned the commercials down to the same level as your TV show?
I mentioned earlier that people frequently mess up their audio by over-use or incorrect use of compression. The most common problem is with music.
If you compress it too much, say, in order to make your mix louder than everyone else’s, you risk sucking out the dynamic range of the music. Changes in levels are important to the emotion of music. Over-compressing can flatten things so much that there is no more emotional flow to the music. That can also make it sound unnatural.
Also, compressors tend to impart certain sonic strangeness to audio when over-done, such as increasing sibilance (the sort of hissy sounds you hear when someone uses the letter “S”) in vocals, or audio “pumping” which is when the compressor is rapidly clamping down on loud audio and then letting up on quiet parts over and over.

So Should I or Shouldn’t I?

So back to the original question. “Should I use compression in audio recording?” Now that you understand what compression is, you can answer your own question. Do you want to even out the loudness of your audio? Then compression can help you. Do you want to raise the overall loudness of your audio without distorting? Compression can help you. Do you want to add some punch and ‘in-your-faceness” to a voice over? Compression can help. Just make sure you remember that it is really easy to overdo compression if you’re not careful.
Now go forth and squash your audio responsibly.
Ken
ps – Since this article was written, there have been several more posts that have to do with compression in some way. Since you’re now thinking about compression, you might be interested in learning more about it. See our other articles on compression here:
Using Multi-Band Compression
Home Recording: Singing As Loud As You Want
Compression For Punch and Presence

Crossfade Meaning – What Does It Mean To Crossfade Audio?

January 5, 2012 by Ken Leave a Comment

So what exactly does it mean to “crossfade audio”? Well before we talk about crossfading, we should probably first talk about just plain old regular fading. What is a fade?
Happily, this is one of those few audio terms that actually sounds like what it means. Whether you know the term or not, I’m fairly confident that you’ve at least experienced/heard fading going on before.
Pop songs used to end with a fadeout a lot back in the day. You see it on sheet music all the time. At the end it often says “repeat and fade.” It just means that instead of an abrupt or definite ending, the song would continue. The volume would just get lower and lower until it was inaudible, at which point it would end. It fades out.

Fades

Often on TV or movie soundtracks you hear the audio fade in (as opposed to the above example of fading out). The music (usually) or other audio will start out inaudible and rise in volume gradually until you can hear it at full volume.
This is the same idea for audio items in our recording projects. We use it as describe above sometimes, especially when producing voice overs or narrations that have background music.
The music frequently fades in before the narrator starts talking. If the music is not going to continue behind the voice for the duration of the program, as in a podcast or radio show intro perhaps, the music usually fades out just after the narrator starts speaking.
But the most common use of fading when doing multitrack recording projects is simply to allow audio items to start and finish more smoothly. If the edges of an audio item do not fade in or out at least a little bit, you get clicks and pops due to the sudden stops and starts of the item.

Crossfades

Now we know what fading is. So let’s talk the crossfading. It is when one bit of audio fades out at the exact time as another bit of audio is fading in (see the picture at the beginning of the article).
Let’s say you are recording a vocal track, and you mess up. What most of us do is wait a second while the program continues recording, and read/sing the part you messed up a second time, and then proceed with the recording.
Things often run more smoothly this way, as opposed to stopping and starting over all the time. However, doing it like this requires that when you have finished recording, you go back and cut out that messed up part. Crossfades to the rescue!
A common way to deal with mistakes is to simply highlight the screw-up, and cut it out. This leaves a blank space there (unless you have ripple editing turned on – see our post on THAT here: https://www.homebrewaudio.com/what-is-ripple-editing), creating 2 audio items, the part before the cut and the part after it. Now you’ll want to join these two items so you drag the later audio to the left to connect it up like a train section to the end of the earlier audio item (see figure 1).

Crossfades Are About Seamlessness

In order for the audio to sound seamless to the listener, you’ll usually want to slightly overlap the end of the earlier audio and the beginning of the later audio item. But that will still frequently give you an abrupt and unnatural sounding transition. So you want the end part of item 1 to be fading out exactly at the same time as the beginning part of audio 2 is fading in.

So most programs, including Reaper (shown in the pictures in this article), default to automatically crossfading any audio items that overlap. Take a look at Figure 2 where I have connected the same audio items shown in figure 1. I’ve drawn a box around where the item on the left is fading out as the item on the right is fading in. The red lines represent the volume of the items. It sort of looks like an “X.” I did not have to do anything extra because as soon as the items started overlap, Reaper automatically executed the crossfading.
This kind of crossfading is not done as an effect, where you’d hear something slowly fade in or out pleasantly. I had to zoom in pretty far to even see the red fade lines. That’s because the fades happen very quickly. This type of crossfading is used as a tool to help avoid the popping and clicking that frequently accompany the cutting and deleting of audio.

You can edit each fade if you wish. You could change the shape of the curve or make the fade longer or shorter. Do this by hovering your mouse over the edge of the audio item until the “fade” tool icon appears (Figure 3). Then right-mouse-click to change the shape of the fade. Or simply drag the edge of the item to change the length of the fade.

Items Do Not Have To Be On the Same Track To Crossfade Them

One last note – items do not have to be on the same track to use the crossfade idea. In Figure 4 I used the same two audio items, but I put the second one on a different track.
They still overlap in time. The end of the first one still fades out just as the beginning of the second one is fading in. The only difference is that they are on different tracks.
So now you know what crossfading means. And you know how you can use it to smooth out the transitions between audio items. Crossfading is a good tool to give you better audio recordings.

What is Ducking In Audio Recording?

December 30, 2011 by Ken 1 Comment

Ducking in audio recording is a technique for allowing a narrator’s voice to be heard clearly and consistently when there is other audio happening at the same time, such as background music. If nothing else, this is a rare example where the term actually describes, in plain English, what it accomplishes. Plus it’s fun to say.

So what does ducking mean anyway?

The way it works is pretty simple. Let’s say you have some background music for either a voice-over project or a video project that will have a narrator, and that this music will play at the same time as the narrator speaks. When the narrator is not speaking, the music should be clear and present – the main thing to listen to. But when the voice comes in, it should take over as the main audio, the background music becoming, well, background audio.
Sure, you can simply turn the music track volume down, but then when the voice goes away the music won’t be loud enough. You’ll have to turn it up again. This cycle will repeat as many times as the narrator stops speaking, which is a lot of work. Ick. I’m all in favor of reducing work. If only there were a way for the audio (or video) program to sense when the narrator is talking and when he/she is not, followed by the program automatically turning the music up when there is no vocal, and down again when there is. Guess what? There is! That’s what ducking is. The audio you want to be the primary thing, like the narrator’s voice, is made prominent because some other audio is “ducking” underneath it at key moments.

So how can I make my program do ducking?

I’ll explain how to do it in Reaper, though the process is going to be the same in any multi-track audio program or video editor. There usually isn’t a tool called “ducking” in these programs since the technique uses a combination of tools working together. The main tool is compression (see our article Should You Use Compression In Audio Recording?), which is a tool that automatically turns audio down by a certain amount, but only when the volume gets loud enough to trigger it.
Usually you insert a compressor effect onto a track to control the audio on that track. So if you have a voice on track 1, and you put a compressor on track 1, the voice will trigger the compressor when its volume goes over a certain level or threshold that you set. Well the big difference with ducking is that the voice is still going to trigger the volume reduction actions of a compressor, but that compressor will be turning down the volume NOT of the voice, but of something else; in this case, a background music track. That means we have to put the compressor on the music track (say, track 2), but make it listen to track 1 for its instructions on when (and how much) to turn down the music. Pretty nifty huh? No? I lost you? Yeah, this kind of thing is much easier to explain with pictures. Take a look at the picture on the right. So it’s just a matter of having at least two tracks, putting a compressor on the one you want to get ducked, and feeding the output of the main audio (the voice) into that compressor.

Ducking in Reaper

In Reaper the first thing to do is put the compressor on the track to be ducked, in this case, track 2 with the music on. Simply click the FX button and choose the built-in compressor called ReaComp. Then close the FX window. You’ll be able to tell that the compressor is loaded up because the FX button will be green.
Next find the I/O (stands for “in/out”) button on the vocal track. See picture on the left. You’re going to click and drag that down to the I/O button on track 2. An I/O window will open with lots of scary-looking stuff on it. Leave everything the way it is except at the bottom where it says “1/2 => 1/2”. Just click the drop-down arrow next to the second “1/2” and change it to “3/4”. See the picture on the right. You have just set it up so that track 1 will send a signal to track 2. The next thing you need to do is click on the FX button in track 2 to open your compressor control. Now it’s time to tell the compressor NOT to listen to the music track for cues as to when to engage, but instead, listen to the voice on track 1. So in the compressor control window (see picture below and to the left). Find where it says “Detector Input” and use the drop-down arrow to change this from the default “Main Input L+R” to “Auxiliary Input L+R“. Now the compressor will change the volume settings of the music based on the fluctuations of the voice track. Pretty neat huh?
Now all you need to do adjust the compressor settings. You can experiment here by listening as you adjust. Start out with a Ratio setting between 4:1 and 6:1 and put the Threshold slider down to about -20 dB. One thing to be sure of before you do any of this is that the overall volume on the music track is already set to where you could hear the voice over it fairly well even without the ducking compressor. If the average volume of the music is so loud that the voice can’t even be heard to start with, ducking won’t help you much.
To make sure things are working properly, close the FX window and play your audio. Click the “S” (stands for “solo“) button on the music track to solo it. It ought to sound very odd, sort of choppy to you. This means it’s working. Now un-solo the music track and listen to how beautifully clear the voice is while the music is still very audible as well.
Use this knowledge for good.
Cheers!