July 9, 2009
In the last 2 weeks, I’ve recorded several interviews at 2 different trade shows. Those interviews have ended up in episodes of both Shutters Inc and Sine Lanugage.
In the wake of this, I’ve had a couple of enquiries from listeners as to my technique for recording interviews on flash-based field recorders like the Zoom H2.
These listeners have commented that they never seem to achieve the same level of results as I have managed, and have asked for some insights.
So, I shall endeavour to outline the pitfalls as I understand them.
First off, tempting though it may be, do not record to mp3!
Remember, mp3 is a lossy format, and we don’t ever want to save production audio (clips which still need further work before release) in a lossy format.
And for that matter, don’t record at 16 bit wav either.
No, your preferred option is to record 24 bit wav.
Yes, it will chew up your memory cards quicker, but memory cards are really not that expensive these days, so carrying a couple of extras shouldn’t represent too much of a burden, either physically or financially.
If your recorder of choice does not offer 24 bit wav, then fall back to recording 16 bit wav instead.
Secondly, your recorder SHOULD offer a choice of microphone gain sensitivities.
The Zoom H2 offers low, medium and high.
Low will turn the sensitivity down (useful for really loud sources), medium is what it sounds like, and high turns the sensitivity up (for really quiet sound sources).
I have found that the medium setting usually works well for these trade show interviews, but obviously, judge each on a case by case basis.
Remember, we are recording at 24 bit, so we don’t NEED to peg the meters at zero!
Peaks of -20dB to -12dB are just fine!
Third, if your recorder of choice has a headphone output (I don’t imagine there’d be any which do not, but you never know), then absolutely have some form of monitoring with you when you are recording.
This may be a set of lightweight street headphones, or even a decent set of earbuds.
Me? I use my trusty old Sennheiser CX300′s, with just one earbud stuck in one ear.
The reason for that is that through that ear, I can hear what the microphone is picking up, and through my other ear, I’m hearing the world around me.
Now, because you are monitoring (via your earbud) what the microphone of the recorder is hearing, you are able to move the recorder around as necessary througout the interview to make sure the talent stays ‘on mic’.
Now, you might be thinking that people aren’t going to like having a flash recorder stuck in (and moving around in front of) their face.
I would contend that if they have agreed to do an interview, then they are probably going to be ok with it.
My technique is to hold the recorder at chest height between myself and the talent.
That way, you SHOULDN’T get any plosives (pops), but the mic should be able to hear the talent fairly well, while keeping the ambient noise reasonably under control.
If you talent is a very soft speaker, then you may have to move the recorder closer toward them, and that may feel a little uncomfortable at first.
If the talent keeps backing away from the mic, stop the interview, explain to them that you NEED the mic that close in order to hear what they are saying without being drowned by background noise, then recommence the interview.
Thing is, MOST of the time, the person you’ll be interviewing is from the marketing department or the sales team and they generally don’t speak that quietly!!
OK, so now you’re back at your desktop (or in your hotel room working on your laptop) and ready to edit and mix.
Drag the files into your DAW of choice.
DO NOT go and normalise the waveforms!
Remember, they’re 24 bit files, so it’s all good.
In your multitrack (which is also mixing at 24 bits or higher, right? RIGHT??), lay up your interviews where you want them.
Adjust the gain so you’ve got peaks around -20dBFS to -15dBFS off each channel. At this point, you should have NO processing on your master output.
Put in some per channel automation to keep each interview roughly in the bacllpark in terms of output level. You don’t have to get too finicky with it, just ‘in the ballpark’ will be good enough at this stage.
Now, if your final audio piece is going to feature other pieces of audio as well, I’d suggest setting up a submix (buss) for just the interviews to go through.
Then, slap a peak limiter across that buss with an output level set for -15dBFS, and the threshold set so that you’re getting about 4-6dB of gain reduction on that peak limiter.
Then, AFTER the peak limiter, put a compressor with a moderate attack (~20-30ms), moderate release (~100-150ms), a medium ratio (3:1-5:1) and again, enough threshold to give you another 3-6dB of gain reduction.
Your interview submix should now be exhibiting tightly controlled dynmaics, but not sounding squashed.
Go ahead and mix it in with all your other audio bits so that everything sounds roughly equal in apparent volume.
Slap a peak limiter across your master output, and you should be cookin’ with gas!
May 24, 2009
This week, a couple of links from my man in Hollywood:
An article from Sound On Sound magazine on surround sound,
plus this excerpt from the NAB Engineering Handbook on Audio for Digital Television.
Also, another war story for ya!
For anyone who needs help remembering their DAW’s keyboard shotcuts, try Editor’s Keys.
A quick moment of introspection about the quality of my work at ARN,
Jim Weishorn brought the Pleasurize Music Foundation to our attention, including their free Dynamic Range meter.
Then, Felix told us about his home made plate reverb.
He also pointed me to this article on tightenening up your mixes with the aid of a spectrum analyzer.
Which proved a nice link to my final piece…. my newly acquired Genelec 7050B subwoofer.
December 7, 2008
In ep 127, we start (one more time!) working our way through the effects menus of Audition 3, starting with the ‘Amplitude and Compression’ submenu items.
November 23, 2008
This week, Greg Anderson sent in a voice comment of his own “radio war story”,
Jim Weishorn wrote to ask for more info on subtractive eq,
Alexander Williams* (no relation) wrote to ask about processing audio for live streaming (as opposed to pre-producing content),
…which led me to again remark about having VU meters rather than just peak program meters (PPM’s), and a great free VST plugin VU meter is the Modern Meter,
and then Jim wrote again asking about home theatre… setting up speakers and subwoofers and so on.
After I finished mixing this episode, I realised I didn’t really finish answering Jim’s questions, so consider this ‘part 1 of 2′.
And talk about freaky… in the very week I talk about my Energy 10.2 subwoofer, the damn thing decided to die on me! About an hour after I recorded this ep, I realised that the sub wasn’t working. After some investigating, I came to the conclusion that it had died. I took it to 2 different stores to be checked, and they both deemed that it was dead, too. So, I’m taking it to a friend’s place today for him to have a look at it (he MIGHT be able to fix it). But if he has no luck, looks like I’ll be buying a new sub in the next week or so.
* BTW, I love Alexander’s tag line for his streaming show:
“Like a morning show. Only interesting. And at night.”
November 16, 2008
This week, “Big C” asked about combatting air conditioner hum,
Tim Cumings sent in an audio sample of a Skype interview he conducted (pre and post eq),
Greg Anderson wrote to say that he’s launched his Atomic Time Keeping podcast on the world (Onya, Greg!)
He also wondered about mp3 encoding… mono vs stereo.
Then, I heard from Jeremy James, author of the podcast novel “The Veingel”.
Jeremy explained some of his workflow, and I offered some ideas on same… namely, downward expansion vs gating, broadband compression vs multiband compression, and so on.
A large part of what I had to say came back to getting as much done as possible NON-destructively in the multitrack environment.
Also, for those who are looking for a good, low-cost alternative multitrack recorder/editor, try out Reaper. It’s free for non commercial use, although they would like you to pay a small donation! Check it out, nonetheless.
Plus, Bruce’s new toy (which should arrive in the next day or so)… the Safe Sound Audio Dynamics Toolbox.
Can not wait for this little puppy to get here!
November 9, 2008
Believe it or not, we’re finally here!
The building of this video has been an absolute labour of love…. I’d estimate that it’s taken me about 20 hours of work to complete!
Maybe, that’s partly my inexperience at producing video podcasts, but hopefully, when you watch it, you’ll see where those hours went.
In an effort to ease the load on my hosting company’s servers, I will be setting up a torrent feed later tonight.
Check back here later for a link to the torrent file.
There’s also a copy of the file at YouSendIt, plus the copy here at audio2u.com.
To download manually from audio2u:
From YouSendIt here:
If you want to download a copy of the final mix of the song (featuring a couple of extra tweaks I did later), grab that here:
Fear of Holding On – 320 kbit joint stereo (13MB)
Towards the end of the podcast (or is that vidcast?), I mentioned that I would make the individual instrument tracks available to anyone who wanted to have a crack at mixing the song for themselves.
Drop me an e-mail and I’ll send you the link to download the files.
Just be warned… that’s an even bigger download than the mp4!!
The tracks are 32bit mono and total about 1.1GB!
The download will be in .rar format, so you’ll need an archive utility like WinRAR (or similar) which is capable of unzipping a .rar file.
October 12, 2008
This week, an answer for Bomar who thought he heard distortion in episode 98,
another mic comparison, this time between the R84 and my AKG C3000,
Jim Addie wrote in with detailed definitions of DC offset and wave asymmetry,
plus a reminder about a great thread on PSW about digital recording levels.
Also mentioned, the Shure SM81, and Rycote wind socks.
May 25, 2008
Wow! What a big week!
In episode 88, a ton of feedback regarding microphones on noisy trade show floors and the like…
Scott Hess likes the Sennheiser HMD25′s, and the HMD280-XQ’s,
Tokyo Dan suggested checking out this video on YouTube, as well as this video on shooting video on a Nokia N95,
Dave King mentioned these headsets from Eartec, which look like your standard headsets that you’d get from AKG or Sennheiser, but Eartec say they will customise the circuitry for you upon request!
Then, Michael from New Zealand asked about matching RMS levels between his locally produced content, and material coming from other studios,
Jim Weishorn asked about multiband compression,
and we finished off with a discussion on the differences between MP3 and MP3Pro encoding.
March 23, 2008
I know, I know… I told you all that there would be no podcasts for 7 weeks, right?
Yeah well… I got a great e-mail from Meredith Matthews (otherwise known as Mer from Braindouche, but I hadn’t made that connection at the time I recorded this) who asked about the whole “loudness war” thing.
Which of course led me off on a half hour rant about peak limiting, rms, VU, average loudness… all the usual suspects.
Anyway… THIS will be the last podcast for another 6 weeks, ok?
The prosoundweb thread I mentioned is here.
And here’s some 1kHz tone to play with, if you’re interested.
And a late starter… the ModernMeter plugin can be downloaded from here (right click and “save as”).
October 28, 2007
This week, the final episode in the series on “constructing your own promo”… the mixing stage.
October 14, 2007
This week, I read an absolutely amazing thread on prosoundweb about digital levels.
Some of the ideas fly a little bit in the face of some of the things I’ve suggested in the past (particularly with regard to tracking as hot as possible).
Having now finished reading it (I was still half way through it at the time I recorded this episode), the main gist of what was being discussed (and there were some pretty big industry heavyweights in there) was that if you are tracking in 24 bit, there is no need to aim for the hottest possible level to disc. These guys were advocating tracking with average levels around -20dBFS (and peaks around -12dBFS)! Now, if anyone had tried to sell me on that idea a week ago, I probably would have held on to my existing position and disagreed.
But no, the theory makes a whole lot of sense.
In a nutshell, the idea is this:
At 24 bit resolution, we’ve got 144dB of S/N ratio to play with (or you could refer to it as dynamic range if you wanted to).
NOTHING (Shall I repeat that? NOTHING) that you are going to record into your DAW has that kind of dynamic range.
Nothin’, zip, nada, nil.
For podcasters, the widest dynamic you’re likely to deal with will be your own voice, and even if you’re REALLY inconsistent with levels (and don’t own an outboard compressor), the most you’ll have to contend with might be in the vicinity of 50dB (from the quietest passage to the loudest passage)…. but even that is unlikely.
So, if you track with peaks at -12dB, AND you happen to have a soft passage 50dB below that (-62dBFS), you’ve still got 82dB of S/N below ya!
But why would you need to track that low?
Well, the theory (and according to these heavyweights who are tracking and mixing this way, the practise) suggests that if you track too close to full scale, you might introduce clipping when you start running plugins (EQ, compressors, peak limiters, whatever) that don’t operate at higher bitrates.
Look, it’s not a thread for the faint hearted, but if you have an interest in getting the best possible quality out of your digital audio setup, it’s worth the time and effort.
It took me 4 days to get through it because I kept on re-reading lines and paragraphs to make sure I understood it.
Also, the real meat starts about 7 pages in when Paul Frindle weighs in on the discussion.
OK, enough of that.
This week in the podcast, part 4 in the series on constructing your own promo: some tips on track laying.
July 29, 2007
This week, George (at the Eclectic Mix and One Minute How-To podcasts) wrote and asked about workflow, and whether certain audio processing tasks should be conducted in a particular order, according to their “destructiveness”.
July 15, 2007
In ep 66, Bruce answers a couple of listener e-mails.
One about the multiband compression used for FM radio broadcasts, and the other one…. was actually 4 questions!
1. What is normalising about?
2. Why do we do it?
3. Should you do it before or after getting rid of undesirable sounds?
4. What is the difference between Gain and Volume?
July 8, 2007
This week, I saw a news item in one of my tech blog feeds (don’t recall which one) about a new site called Soundsnap, which is kind of like YouTube for audio. Commercial-free music and sound effects to download which can be used in podcasts and the like.
I also received an e-mail from Jim Weishorn… (just between you and me, I think he’s got a calculator fetish, but don’t tell him I said that!) where he calculated the time it takes for a 45 degree splice across analogue quarter inch tape to pass by the heads! Woah. (Honestly though, nice work, Jim!)
He also asked about the idea of using different compression ratios for different frequency ranges within you audio. Not a concept I’d ever heard of, but an interesting one nonetheless.
July 1, 2007
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Will you still need me, will you still feed me, when I’m sixty-four?
In episode 64, a listener query from John Meadows prompts a discussion on how to aviod getting nasty little clicks and pops when you’re editing. Bruce answers not only how to avoid it, but why it happens in the first place.
John also mentions a piece of software called Click Repair, which may come in handy for those wishing to digitise their old LP collections.