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Mastering podcasts with Audacity

By Johnathon Williams on March 27, 2006 (9:00:00 AM)

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Open source software makes podcasting easy -- too easy. Listening to a playlist of first-timer podcasts can leave your ears ringing from sudden changes in playback volume. The problem is audio mastering. Recording sound is simple, but mastering that sound -- compressing volume differences, maintaining a decibel ceiling, and similar operations -- is anything but. Fortunately, an open source tool offers everything you need for mastering podcasts and other spoken-word recordings. Audacity is well-known among podcasters on all platforms for its ability as an editor; here are some tips and tools for mastering and adjusting volume, aimed at podcasters, but they could apply to anyone who needs to produce a spoken-word recording under less-than-perfect conditions.

Audacity measures volume levels in decibels, or db for short. A level of 0db is the ceiling; anything above that will sound distorted. When recording a podcast, shoot for a level between -7db and -14db. That range provides a signal that is relatively loud, while leaving a comfortable padding at the top. Moreover, the difference between the loudest and quietest signal is only 7db, making it easy on the ears. If your recording falls below -14db, don't worry. Using the techniques presented in this article, I've been able to salvage recordings with levels as low as -25db.

Audacity displays sound as a waveform. Volume is represented vertically, with loud sections showing higher peaks than quiet ones. The waveform, however, provides only a general sense of volume levels. To see an exact decibel reading, highlight a small section of the waveform and select Analyze -> Plot Spectrum. Audacity will display a graph that shows the decibel reading at different points on the frequency spectrum. (Decibels are displayed on the left side of the graph.) Running the cursor over the graph will display the peak decibel reading at each point.

Just as important as the decibel level of a sound is the frequency where it occurs. Frequency is represented by the numbers on the bottom of the graph. Lower numbers represent sounds with lower pitch (think James Earl Jones), and higher numbers represent sounds with higher pitch (think Mickey Mouse). In the Plot Spectrum window, the human voice appears strongest on the left side of the graph, between 86 hertz and 3 kilohertz. When gauging decibel readings for a spoken-word recording, that is the range that matters.

Get familiar with the Compressor

Audacity's Compressor is one of its most useful and least understood effects. With the proper settings, it can automatically remove volume differences across an entire recording. In essence, it's an automated version of Audacity's Envelope tool (more on that later). The difference is that applying the Envelope tool over an entire recording can take hours, whereas running the Compressor takes only minutes.

The Compressor is confusing because it's a two-stage effect. First, it reduces all audio that exceeds a given volume. Second, it boosts the volume of the entire selection. Think of the first step as a leveler; it gets rid of unwanted peaks. Then, with the loud portions gone, the second stage raises the volume of everything, eliminating quiet sections. Because of this two-stage process, how you set the effect is crucial to getting effective results from it.

To get started, highlight your recording, and select Effect -> Compressor. The effect window presents you with three settings: Threshold, Ratio, and Attack Time. The first is the most important. Threshold is the number that acts as the decibel ceiling for the Compressor. Anything above this number will be reduced. Anything below will be left alone. You want to choose a number that's low enough to bring your loudest sections within 7db or so of your quietest sections. (If you run the Compressor and nothing in your waveform changes, then you set the Threshold too high.)

Find and remove noise in your recording environment

Among podcasting sins, too much background noise is near the top of the list. To test for noise in your environment, use Audacity to record a sample of dead air. Then, highlight the sample and click Analyze -> Plot Spectrum. The graph you see reveals the decibel level of the ambient noise in the room.

My home office, where I record all of my podcasts, shows a noise level of about -60db. This has been fine for my purposes. A bare minimum is -50db, but even that can be audible. If your environment shows a higher reading, shush any noise makers and try another sample. Common culprits include fans, central air, and, unfortunately, desktop computers.

In choosing a Threshold, the first thing I do is use the Plot Spectrum tool to identify the decibel levels throughout my recording. First, I highlight what appear to be the loudest sections (the tallest peaks in the waveform) and take a reading of those. Then, I highlight what appear to be the quietest sections (the shortest peaks in the waveform), and do the same. If the loudest section peaks at -10db, and the quietest at -20db, then I choose a Threshold of -15db, or half the difference. If, however, the loudest section peaks at -5db and the quietest at -25db, then I choose a much lower Threshold of -20db.

Again, the key to getting good use from the compressor is setting the Threshold low enough to bring the loud sections closer to the quiet ones. Don't worry about making it too quiet, because the second-stage gain boost will bring everything up. This is especially true for recordings that mix very quiet sections with very loud sections. (Podcasters who conduct interviews using VoIP software run into this all the time. The local microphone is often recorded louder than the remote signal.)

The next setting, Ratio, determines how severely the Compressor will reduce signals that exceed the Threshold. A ratio of 2:1 is very gentle. It cuts any signal above the threshold in half. In general, it's best to use a low ratio when possible, because high ratios sound clipped. But remember, if your recording contains sections that are much louder than your Threshold, then you need a higher ratio, say 6:1.

Attack time determines how quickly the Compressor activates when the Threshold is exceeded. I like a fast setting of 0.2. (Some recommend a slower setting of 0.5.)

Finally, check the box next to "Normalize to 0db." This activates the second-stage volume boost. Click OK, and the Compressor will go to work. If the results are disappointing, just click Edit -> Undo, and try again with different settings. Getting a feel for the Compressor takes some trial and error, but the results are worth it.

Clean up with the Envelope tool

Run the Compressor enough times, and you'll notice an annoying limitation. Normalizing to 0db increases the volume of your recording, but the amount of the increase is limited by the loudest section in your recording. This means that a single leftover spike of -2db will limit the overall increase to only 2db, since the Normalize effect won't let any section exceed 0db. Obviously, this can be real problem if the rest of your recording is down around -18db.

Theoretically, the compressor should remove any such spikes before it applies the volume increase. (Removing those peaks is, after all, its job.) Unfortunately, it sometimes doesn't. In my experience, it can miss very sharp and sudden peaks. The solution is to find any remaining spikes and eliminate them yourself.

Select the Envelope tool (the hourglass-shaped icon in the top left corner). In the waveform, find the first sudden peak that stands out. Click to create control points on either side of the peak. Then, click and drag the points to reduce the volume of the peak. Dragging up increases volume; dragging down decreases it. You can create as many control points as you need to fine-tune the adjustment. When the peak sits level with the rest of the waveform, move to the next and repeat the process.

Amplify to set a new ceiling

With those leftover peaks out of the way, it's time to increase the volume one final time. Select your entire recording, and click Effect -> Amplify. The Amplify window will automatically calculate how much the volume can be increased without breaking 0db and distorting. This is an important final step for most amateur podcasters, since our recordings tend to be on the quiet side.

If the result isn't loud enough, it's probably because the recording still contains unwanted peaks. If you don't have time to smooth them all out with the Envelope tool, you can try running the Amplify effect again, but this time uncheck the box next to "Don't allow clipping." This will allow you to drag the decibel slider in the Amplify window as high as you like. Existing peaks will be pushed above 0db and badly distorted. I recommend this only if those peaks are extremely brief in duration or if you're too hurried to manually remove peaks with the Envelope tool. Also, don't stray too far above the slider's recommended setting or the entire recording will distort.

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on Mastering podcasts with Audacity

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Recording with Audacity

Posted by: Anonymous Coward on March 28, 2006 04:08 AM
I use audacity on a weekly basis for doing voice recording. My first rule is to record the audio as loud as possible without clipping. I try to peak between -9db and -3db in most sentences or phrases of the person speaking. My reasoning is that you will have some constant noise introduced from your microphone, mixer, sound card, or whatever equipment you are using.

Let's say my sound card is always producing noise of -60db.
If I am peaking at -12db and amplify by +12db to get that to zero, I will raise that noise level to -48db. However if I am peaking at -3db and amplify the recording by +3db that noise level will only be -57db.

Many times I find just using amplify is easier than the envelope tool. If the short peaks are due to p's and t's in the speaking, I just highlight a peak, select Effect > Amplifiy and decrease it by -1db. I then just hit Ctrl+R to repeat the last effect until I get it to the desired level. Now I can highlight the next peak and just hit Ctrl+R as many times as I need.

I would strongly discourage anyone from unchecking the "Don't allow clipping" in the Amplify effect. If you have too many peaks, lower the volume of each one with amplify first; don't just raise the volume and clip those peaks. It may take a little more time, but it is definitely worth it. Otherwise, how would you know how high to run the volume past the 0db ceiling?

Just My $0.02.
Josh

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Re:Recording with Audacity

Posted by: Anonymous Coward on March 28, 2006 05:28 AM
/me fully agrees with josh.

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I really like Audacity but...

Posted by: Anonymous Coward on March 28, 2006 08:05 AM
Although you can create a rough master with it, as a professional sound engineer, working with audio for a living, It sadly lacks real time processing capabilities (awesome app though, very good for audio editing tasks!).

When mastering, metering is very important, you need to be able to keep an eye on the overall level and the level of gain reduction being applied by a dynamics processor, so you can be sure that you're not clipping or squashing the signal too much by making it too 'loud' (like most music these days, sadly). Making sure the overall level does not exceed -0.2dBFS is generally regarded as a sort of standard in the audio world as some CD players and other gear will clip/distort at -0.1 or 0dBFS.

Now if we could pipe the output of Audacity into Jamin.... we'd have a superb mastering solution! When using an acoustically treated room and the right monitoring setup of course...<nobr> <wbr></nobr>;-)

Warm regards.

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Re:I really like Audacity but...

Posted by: Anonymous Coward on March 30, 2006 12:55 AM
For metering and analyzing I use Elemental Audio's Inspector XL. <a href="http://www.elementalaudio.com/" title="elementalaudio.com">http://www.elementalaudio.com/</a elementalaudio.com>
It's been a fantastic tool, even when I use Bias Peak Pro 5, Cubase SX3, or what have you (yes, I'm on a PowerBook and love using Audacity -- my home box is a Debian PC).

I produce a national show on public radio and use Audacity each and every week to record the show to then be cleaned up and squashed down to formats for Audible.com and for a podcast of the show.

It's been a great tool. I haven't done extensive mastering of music with Audacity, but for broadcast, it's surprisingly effective (and affordable).

All good things...

cp

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Re: I really like Audacity but...

Posted by: Anonymous [ip: 24.22.229.188] on December 30, 2007 07:13 PM
That's about what I was going to say. Not only that but the compressor has no release?!

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Regarding my post: I really like Audacity but...

Posted by: Anonymous Coward on March 28, 2006 08:11 AM
I should also mention that applying multiple processes to a digital audio signal will degrade it significantly, introducing 'rounding error', where each recalculated sample is rounded off to the nearest 16bit (or 24bit or whatever bit depth you're working at) number. This is more of a problem at lower bit depths as rounding will be less accurate due to the lower 'resolution', if you will.

The more processes you apply, the more rounding error you introduce, so as a general guideline, it's good to keep the number of processes a signal goes through to a minimum.

Regards.<nobr> <wbr></nobr>:-)

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Re:Regarding my post: I really like Audacity but..

Posted by: Anonymous Coward on March 29, 2006 12:05 PM
I agree with this comment completely.

I've been podcasting for some time and I try to keep my "effecting" to one step before converting to Mp3; Normalize. Yup that's it.

For the skype issue talked about in the article, do what we do, use Gizmo.<nobr> <wbr></nobr>;-) Seriously, we do 1 to 2 test recordings to get the levels right so no one is lower then another. This saves alot of post production time.

BTW, we use Gizmo because it allows one button recording without the need of a paid for plug in with Skype.

My $0.02.

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Podcasting even easier

Posted by: Anonymous Coward on March 28, 2006 11:43 PM
Audacity is a great tool of course, but I still think podcasting is too much of a hassle when it's recorded offline, and then uploading, then putting it in your blog or cms is required.

There are possibilities to record and publish online. I made a forum for spoken word for educational ends, and so far it's working fine, and users are craving about it. It includes a level meter to maintain a healthy overall volume, and microphone input levels can be paramaterized through java/flash.

This really creates a seamless and almost effortless way of podcasting online. Mail me if interested at p.j.van.rees_AT_rug.nl.

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Re: Podcasting even easier

Posted by: Anonymous [ip: 24.232.240.112] on August 24, 2007 05:01 AM
It still can be done offline, the trick is in combining recording, uploading and integration with webpage in a single tool. Check out http://sayandpost.com for example - simply brilliant!

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Fixing podcasts with Audacity... and sox and lame

Posted by: raindog on March 29, 2006 07:22 AM
I'm not doing my own podcast yet, but hearing the newbie podcasts the author refers to has given me lots of ideas on things to avoid when I do. They've also inspired me to first brush up on my mastering skills in Audacity (using techniques largely the same as what he describes in the article, but maybe more severe) and then my sox manpage-reading skills.

I ended up writing a script that goes through my whole podcast media directory, and automatically compresses and re-encodes (mono and with a lower bitrate) all the podcasts. In addition to making them a lot less painful, it lets me fit like 25 or 30 hours of podcasts on one CD to listen to on an MP3 CD player or my Nintendo DS in the car.

The whole script as it currently exists is here:

<a href="http://www.kudla.org/podcastcarencode" title="kudla.org">http://www.kudla.org/podcastcarencode</a kudla.org>

But the sox command line took the longest to figure out, and here it is:

sox -V -v $maxfac $mp3 -c 1 -r 16000 -w<nobr> <wbr></nobr>/tmp/mp3carencode-$$.wav resample compand 0.1,0.5 -120,-20,-60,-15,-40,-15,-20,-9,0,-8 0 -15 0.5

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editors fell asleep on this one

Posted by: Anonymous Coward on March 29, 2006 09:09 AM
Hello? Could we have something to break up the monotonous paragraph after paragraph? It makes my eyes hurt!

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Don't forget headphones

Posted by: h23 on March 29, 2006 11:06 AM

One very basic thing that I think a lot of podcasters fail to do is to wear headphones when they talk into a mic.

I don't know exactly why it works, but if you can hear yourself loud and clear you sound better and are able to control your voice better. DJ's and people who speak on the air have known this forever-- and they don't have to do post processing to sound decent!

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Try limiter instead of compressor

Posted by: Anonymous Coward on March 29, 2006 03:28 PM
Theoretically, the compressor should remove any such spikes before it applies the volume increase. (Removing those peaks is, after all, its job.) Unfortunately, it sometimes doesn't. In my experience, it can miss very sharp and sudden peaks. The solution is to find any remaining spikes and eliminate them yourself.

Removing all peaks above a certain threshold is the is the job of the limiter rather than the compressor, try that instead. A compressor is much more flexible than a limiter and it's job is more difficult to summarise.

The 'attack' setting of a compressor determines how quickly the compressor begins attenuating the signal after the signal has exceeded the threshold.. ie. depending on how fast the compressor reacts, you will still end up with material whose level is above the threshold after processing. cbit.

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Re:Try limiter instead of compressor

Posted by: Anonymous Coward on March 29, 2006 08:51 PM
Yep, I always use a limiter on my output to get the overall level up without clipping, setting it to limit at -0.2dBFS and boosting the overall signal by a few dB's. I also compress the vocal tracks individually to even out the dynamics within them, also giving them that classic 'squashed' radio sound. You can check out some of my recordings here if you like: <a href="http://www.radio2012.org/" title="radio2012.org">http://www.radio2012.org/</a radio2012.org> . This also ensures that the recording will be compatable with a wide range of playback gear, such as cheap in built monitor speakers that have a very narrow dynamic range. One goal of mastering is to make the recording 'sound good' on as many different playback devices as possible. Regarding the vocal tracks, I also EQ them sometimes, especially if I'm mixing an on-site recording with one made in the studio as they will have different tonal qualities due to the acoustic environment they are recorded in as well as differences inherant in using different equipment to record both pieces. It also helps a lot when you've recorded someone with a very deep voice close to the microphone, so you can roll off some of that bottom end 'mud'.

By the way, a limiter is basically a compressor with a ratio setting of infinity:1! Used most commonly to squash those transients that human hearing can't perceive in order to bring the overall level up.

Regards.

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Got to love open source!

Posted by: Anonymous Coward on March 29, 2006 10:17 PM
Open source is such a great thing. It's programs like Audacity that make Linux so great. I recently bought my first Linux desktop from Frye's Electronics with Linspire 5. I've tried a few distributions but I kept getting stuck with compatibility problems or trying to install software. This computer from Frye's is awesome though because it's not just a Linux distribution but an entire solution. It was easy enough to install Microsoft Windows ME on Linux with the virtual emulator program called win4lin because of CNR. For those who don't know CNR is a subscription service/program that makes installing software a one-click process on Linspire computers. They don't charge anything for open source and have a pronominal selection of commercial software. For those who have been frustrated by other Linux distributions like me I suggest you try this. I found some coupon codes for discounted Linspire 5 copies at <a href="http://tryoutlinux.com/" title="tryoutlinux.com">http://tryoutlinux.com/</a tryoutlinux.com> although I think purchasing a Linspire computer is easier if you can afford it. I haven't actually used them since my computer came with it. I've been Microsoft free for several months! wee... I'm going to go try this with Audacity now.

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Cooledit was better

Posted by: Anonymous Coward on April 01, 2006 09:08 AM
Cooledit was better than Audacity. Unfortunately, my key was mangled and the company is now defunct. Fortunately Audacity does most of what Cooledit could do. Particular shortcomings of Audacity: 1) Inability to select a noise signature for reduction, 2) Many of the effects such as fade are presets and nonconfigurable, 3) No filter for opening non-wave files.

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