December 16, 2009
Yay! Got another ep out the door before the eyar escaped from me!
This week, just tidying up some loose ends.
A while back, JR sent us a link about the Schwartz Engineering Laser microphone. That link has now been revised.
Someone also sent in a link to the Waves Vocal Rider, which at the time I received the e-mail was still in development.
It is now a released product, and you can check it on the Waves site here. (Now that the podcast is edited and mixed, I discover that I covered this on the last episode! D’oh!)
Bomar wrote in asking about ID3 tags, artwork and metadata.
I use MP3 Tag Studio almost exclusively.
He also asked about the chipmunk effect and how to avoid it.
Plus he mentioned this article about shockwaves and how they can be photographed.
Also, if you are interested in picking up one of my Bruce Williams Photography 2010 calendars, please check ’em out here!
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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!
December 7, 2008
In ep 103, a continuation of the discussion on home theatre,
Bomar asked about why I use Lame (which I use in conjunction with Speex Multi frontend) for mp3 encoding,
plus he gave us some links for websites concerned with hearling loss:
University of NSW
jimmyr (link 1)
jimmyr (link 2)
and Tim Cumings was after some advice on how to mic up his computer user’s group monthly meetings.
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.
June 8, 2008
This week, my new toy (an AEA R84 ribbon mic),
my lynda.com Reason v4 title is now online,
plus a few e-mails to answer, including using Lame in Audition (thanks Steve Mayfield!),
plus a request… where am I up to?
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.
May 11, 2008
And you thought it would never happen, right?
At last, BTP is back.
This week, a couple of e-mails to answer.
Jim Weishorn asked about the introduction of quantization errors from too much processing.
And Mike Wills asked about encoding your mp3’s from inside Audition, but using Lame instead of the Fraunhofer codec.
Which led to a discussion of the CoolLame filter, available from Rarewares.
Then, a look at the extended capabilities of the Extract Audio From CD tool in Audition v3 (which now has the ability to retrieve album metadata from an internet database so you can rip and tag your CD’s!).
As you’ll hear though, going forward from here, it’s going to be tough for me to maintain 3 weekly podcasts. Time will tell just what I am able to deliver.
February 3, 2008
This week, some e-mail to answer regarding what sampling frequency should we use as a default and why,
utilities to use for ripping and encoding your mp3’s,
how to mix podcasts that prevent your listeners from falling asleep (an area in which I apparently succeed!),
and recording audio on a laptop.
Also, Bruce’s recent home theatre trials and tribulations.
Speek’s multi front end
January 6, 2008
I wasn’t planning on doing another podcast this early in the year, but a couple of e-mails came in, and I figured I may as well just get on answer ’em!
This week, an “end of year” greeting from long-time listener, Gary Lerude. Thanks mate!
John Meadows asked about what effects or processes get over used, and which don’t get the attention they deserve. (NB. The one thing I should have also mentioned which doesn’t receive the attention it should is mic technique!)
And Vassya asked about why 44.1 kHz or 48kHz, dither, and headphones.
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October 9, 2007
So, there I was, catching up on my morning round of tech blogs, when I cam across this great article on Ars Technica about how MP3 encoding works.
Having now read the article, even I have learned some more about how the prcoess works.
It makes interesting reading (if you’re into this kind of thing), and will only take you about 15 minutes (there’s 4 pages to the article).
Interestingly, the Nyquist Theorem actually dates from work done by Harry Nyquist way way back in 1928 (the Ars article says ’27, the Wikipedia article says ’28)…. certainly a lot earlier than I had previously thought.
Anyway, thought I’d share this with you in the absence of a Sine Language for this week.
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