Archive for March, 2008

Aperture 2 On a MDD G4

Saturday, March 29th, 2008

Thanks to these hacks I’m test driving Aperture 2 on a MDD G4 (2 X 1.25GHz.) It took it nearly 24 hours to pull in my 60,000+ library, and another 12 hrs or so to build previews for them all. The MDD isn’t officially supported nor will A2 install (without help) onto it. I have to assume that the only thing the PowerBook that is supported has over it is built-in USB2 and higher end video, all of which cal easily be added. I’ve spent minimal time working with the tools but I could see where this CPU upgrade and this video upgrade would make it usable for average use by the hobbyist. Yes, for about that amount I could buy a 1.83GHz Mini. But, I have a ton of stuff in my MDD, and the storage options are more robust on the MDD. To make a fair comparison one would have to compare the upgrade package to at least the iMac to get in the same league, now you’re talking almost twice the money. I think a maxed out MDD would be a good legacy machine to keep for a link back to the world of OS9 and such.

Make A Difference In The Life Of A Child

Friday, March 21st, 2008

Kids Against Hunger locally is headed up by my friend Tim Stromer.

KAH

Audio Batch Processing With Amadeus Pro

Thursday, March 20th, 2008

I was on the warpath today for batch processing options, as I have been many times before. Amadeus Pro ($40) is not an app that I remember getting a hit on before when doing these searches, but I hit it today. I already had Amadeus Pro in my toolbelt, thanks to Geoff Hankerson, so I thought I’d check out these features (honestly since I have Peak I just didn’t do much exploring to start with.) As I mentioned in this post, I have a workflow I follow to generate the two different quality Mp3 files for my churches sermon downloads. I’ve been looking for a way to cut Peak out of the picture, just for the sake of doing it. Today I think I hit on another option, all things being equal. I discovered the batch processing options in Amadeus Pro. Here are some screen shots of the setup I used, the -a arg in the LAME window forces mono (even though my source files are mono I had to do this to get Amadeus to recognize and output as such, there is no option in the main encoder section for number of channels.) This also means that the bitrate selection appears to be of the output file, after channel conversion, whereas in my example with iTunes, all bitrates are stated as stereo, you just do the math to pick the one for the desired mono bitrate. Translation; I wanted 24k mono, so rather than setting it to 48k (and divide by 2) like I would in iTunes, I set it to 24k.

Batch processing in Amadeus

Encoding options showing the LAME command line args.

Batch processing in Amadeus

Adding an AudioUnits plugin to the action list, my parametric EQ setup as a high pass.

Batch processing in Amadeus

The parameters in the Parametric EQ.

There are a ton of things you can do in batch mode. After I researched the command line args for LAME I added the -a for mono conversion so I didn’t have to do it as a stereo to mono conversion though you can do that as an action item too. I also experimented with normalizing, there are a couple options there including RMS normalize and fixed percentage (or dB) normalize. You can retain the input file format, however if you want to change the bitrate of a source MP3 file, you have to specify MP3 as the output format and set your encoder options there as shown. There are more expensive options ($70) as well for the hardcore production user, unfortunately the demo does not allow batch processing so I couldn’t “let the machines speak” for themselves. Comparing it to my Peak & iTunes workflow it took 1:22 to process my test file – open in Peak, apply Parametric EQ (as low cut), export as AIFF, re-encode with iTunes. Using Amadeus Pro it took 2:38 for the same source file and the same operations. This is on the same MDD 2 X 1.25GHz G4, 10.4.11, 1.75GB RAM machine. Based on sheer speed, Amadeus was not a runaway winner. But for a large group of files and a persons sanity (or schedule), Amadeus Pro is a viable option. Granted this is an extremely narrow test (I didn’t set out to do a review, just to find a better mousetrap), other functions may be faster, and if there were more gyrations to be done that had to be done serially in Peak I wonder if Amadeus wouldn’t show a gain there as well. Tests were done on the latest version 1.2.1. Bottom line; if you’re on a budget and need to do production-like processes with audio files, Amadeus Pro has a lot to offer, especially for the money.
Some additional how-to pages on Amadeus.

Of Audio Encoders and Bits

Thursday, March 20th, 2008

LameBrain has been my MP3 encoder of choice. Today I discovered Max, yet another audio encoder, but more than that a batch converter and CD ripper. It’s free, like LameBrain, but has multiple encoders including Ogg, FLAC, WAV and many others. It allows you to queue up multiple encoding options for a group of files – which is great for production type workflows where you need multiple versions of a file, though I haven’t yet discovered a way to do multiple variants of the same encoder type i.e. two different bitrate MP3 files in the same batch (it overwrites the previous version with the last version.) It can do tagging, add the output files to your iTunes library, and embed album artwork. I found it via Google for batch encoding, and that it does, like LameBrain. I did a quick speed comparison, only one file, one format and came up with LameBrain in the lead with 29 seconds vs. 34 seconds for a 2:31 stereo AIFF to 192k high quality MP3 (on a MDD dual 1.25, 10.4.11, 1.75GB RAM.) And on the same machine encoding to AAC (m4a) iTunes 7.6.1 bested it by 2 seconds on the same source track, 17 secs vs. 15 secs but this was with Max set to highest quality for AAC where iTunes gives you no choices. In Max at lowest quality AAC 128k it managed it in 8 seconds, yet for some reason iTunes says the same for all three versions – “Low Complexity”, can somebody show me a way to do high complexity?! And it’s not all about speed, there are some nice batch processing features in here for the production-minded, for free!

(Some) Church Sound System Essentials Part 2

Monday, March 10th, 2008

Topics for this post include Subgroups and AUX busses on mixing consoles. Subgroups can really simplify managing the mix, especially if your services are very dynamic. AUX busses are a necessity as well if you are running your monitor mixes from the same console as the FOH, and I will also elaborate on how I use AUX busses for my record mix.

Current Version of SR32.4

Subgroups are smaller (sub – below a full mix) mixes within the main mix generally. Here’s how I use mine (on a Mackie SR32.4) and why: I assign my vocal mics (for singing) to subgroup 1 by first depressing the button on each vocal mic channel labeled 1-2, then turning the pan control to the left which narrows the scope of the signal down to subgroup 1. I assign my two speech mics to subgroup 2 by again depressing the 1-2 button and turning the pan control all the way to the right. I assign my instruments to subgroup 3 by depressing the 3-4 button and panning all the way left, and lastly I assign my six drum mics to subgroup 4 by depressing the 3-4 button and panning to the right. What this allows me to do is run most of the service with 5 or 6 fingers. Once I have established the balance of the vocal mics (by pressing the solo button on the Subgroup 1 master channel strip) I can adjust vocals as a whole with a single fader. Same with my instruments on subgroup 3, the speech mics (for speaking, a wireless Shure LX series handheld with SM87 head, and an old AT W series with 831c lav), and the drums on subgroup 4. Because of the way I have my record mix setup (or should I say I setup my record mix this way because) I can tweak the speech mic level going to the record mix by adjusting the channel fader for that mic, and while watching the level meters on the PMD570 recorder simultaneously trim subgroup 2 to maintain the appropriate level at FOH. If the voice (person) leading a given song changes I can always trim up that channel’s mic to bring it out in the mix, and I may pull back subgroup 2 as well, depending on what kind of level we’re already at and if the other vocalists will be coming in at all. But you get the general idea: make your fine mix adjustments with the individual channels (by using the solo button on the associated subgroup master channel strip), then make the coarse adjustments with the subgroup. And again, note that this allows my record mix through the AUX busses to work out the way I want which I’ll explain. One thing that I know generates confusion about AUX busses is the whole “pre-fade” and “post-fade” thing. What this means is that the signal feed to a given channel’s AUX 1 control (being pre-fade, for instance) is pulled from the channel before the channel fader. Which means what? You can jack the channel fader around all you want and the level of that channel’s signal going into the AUX 1 mix (presumably a monitor mix) will not change. Which is how we want it 99% of the time; we don’t want to mess up the person’s monitor mix when we make a change to the FOH mix. Post-fade is the opposite, the feed to that channel’s AUX buss 3-6 (on the Mackie, and 3-4 is switchable) comes after the channel fader. Which now means that it doesn’t matter how high you hack the AUX 3-6 send on that channel, if the fader is all the way down, the most you’re going to get is maybe a smidge of crosstalk. Post-fade sends (tech speak for AUX buss) are used for effects such as reverb and delay, and also for record mixes (at least in my setup.) My theory on this is that if I need to trim a vocal channel or the speech mic for the house, then I would also want to trim it for the record. The trick is getting the AUX level set for each channel so that that relationship works for both mixes within tolerance. But that’s also the beauty of it – I can setup a different mix for recording than I use for live, which means I can use ambiance mics for the record and just not assign them to the FOH mix. And I feed the PMD570 into the tape in on the console so that I can monitor the record mix from the console as well – including the processing I’m doing between the console and the recorder, a Behringer Ultramizer (which I wouldn’t recommend to anyone.) It was a good idea in theory, but the hard compression that this unit applies means all it’s good for is a fancy limiter. Being 2 bands it does allow you to reduce some of the brutality by setting the crossover point just high enough that thumps and bumps and heavy low end don’t bring the entire mix to its knees. And why am I doing all of this? So that when I get home with the flash card I can have a good solid recording that only requires some simple edits in Fission, final MP3 encode and it’s ready to upload. See what you think of the results, our sermon downloads are here and here. Low bitrate versions are encoded with iTunes using this workflow. So, back on the main highway here, a very general guide is that you use pre-fade AUX busses for monitor mixes and post-fade busses for effects (but not processing devices such as compressors since in general they’re meant to be inline devices, not mixed in in a parallel fashion (as you typically would with reverb or delay), and to feed recording devices and/or even distributed sound speakers in the nursery or bathrooms. Our distributed feed is tapped off the output of the Ultramizer, which is feed by AUX 5 and 6. Which brings me to the last thing I want to cover, using a pair of AUX busses to generate a stereo mix for your recording device. I’d love to run a multitrack recording of our services, at least in theory, but that’s not a real need now and just another way to burn up 8 more hours of my precious time. So, for a time I was using AUX 5 and 6 for Left and Right channels, respectively, to generate a stereo mix for recording. What you do to “pan” a voice or an instrument is adjust the AUX 5 and 6 controls differently, for instance; to pan hard left you only turn up the AUX 5 control for this channel, and leave AUX 6 all the way down (and then feed AUX 5 to the left channel of your recorder, and AUX 6 to the right.) To pan in the middle just adjust 5 and 6 to be the same. And for somewhere in-between adjust one AUX slightly higher than the other. That’s the down and dirty way to pan. When we regularly had 2 guitarists playing it really added some nice width to the soundstage of the recording to pan each one fairly hard to one side and the other, and of course panning the cymbal mics and toms, and the vocals.

Living in Church Tech With Spirit-Lead Worship

Thursday, March 6th, 2008

Mike Sessler started an interesting thread with his post titled “Worship Leaders – You’re Killing Us!”. Rather than burn up bandwidth on his blog I decided to post more of my thoughts here. The church (and circle of churches) I’m a part of are strongly convinced that every detail of how we spend our time together on Sunday morning (primarily) is entirely in Gods’ hands. Leadership is faithful to prepare and is ready to do what they feel God wants said and done, but that does not preclude them from completely changing things up if God is obviously going another direction when things get rolling for real. To us the idea of scripting in advance what Gods’ intentions are for a service doesn’t give Him complete Lordship and latitude to do as He wills in the moment. There is a general flow that’s a backdrop, it’s not a free-for-all, but we’re committed to listening for any subtle nuances or direction at any time. You have to ask yourself what the objective of the service is, is it to build ministry, or to build family? Are we performing for God, or are we sitting around Abba Father’s living room spending time together with Him. Assuming we don’t do 3 new songs every service, how badly do people really need lyrics that extra 10 seconds it might take to find the right song if the worship team changes things up? And if that screen is the only thing keeping them engaged with Jesus, can you call that heart-felt worship? Worship isn’t the song coming out of our mouth only, it’s where our heart and mind are as well. That worship should still be active even if we temporarily lose our visual aid while worshiping – our heart should still be corresponding. Are we going to sit back and watch worship, like a movie, or are we going to take part and be the active participants in what happens during worship? Our worship team strives to operate as a team, so when they deviate from the list they gave us in the booth, they deviate together, it just means the person running lyrics projection needs to be attentive. And that’s the primary thing about Oratorio I’ll plug right here is that it’s able to keep up with us in that setting. It’s easy to search the lyrics if they go to a song that’s not in your playlist already, and you can get some good sized thumbnails on a 17″ screen so it’s not hard to figure out which slide to present. My point with this post isn’t to argue one method over another, not to say that it can’t be done in other ways, but to simply state that there are church tech folks out there working very hard every Sunday to give God what they feel He deserves in a way that may be uncommon, and that what some may find technically outrageous is exactly where some of us live. And it really is a matter of what your leadership’s philosophy is – what is important, why do we do what we do? As our pastor often says, “we don’t have a lot of protocol or rules here, but our one imperative is that we meet with Jesus Christ today, that we find Him and engage Him.” I just have to raise my hand and say that I’m doing tech in an environment where it works.