February 23rd, 2011 – Lately, I’ve been working on mixing a small musical that will debut this Friday. It’s definitely a step outside of my comfort zone as mixing live is very different from mixing in a studio, and mixing a musical is very different from mixing a band. Here’s a compilation of some of the things I’ve learned and things that are different from studio work.

For one, there’s a lot of waiting around, but when the show is live, it’s very stressful. In fact, I don’t sit down for the entire performance, constantly monitoring levels and riding the faders. I am communicating constantly with the stage manager, a partner that cues music, and another that cues multimedia portions. Great communication between all of us makes the job flow a lot smoother.

One of the things that makes mixing theatre particularly difficult is different actors coming on and off scene at different times in each act which requires muting and un-muting their wireless microphones on the fly. This requires learning the scripts very well to know who needs what at any given time. Additionally, the actors have very different levels when singing versus speaking. This can be partially combated by subtle compression, but to remain dynamically interesting I prefer to manually adjust levels as the situation requires.

Occasionally all twelve of the main actors sing together. We have the use of stage monitors but not in-ear monitors. This means that for the actors to hear enough of themselves to work out their sometimes complex harmonies, the monitor mix must be loud enough for them to hear, but quiet enough to not feedback into their sensitive wireless microphones. Because they’re dancing and moving around quite a bit, there is no ideal position for the monitors. I must EQ out the natural feedback tones of the space to reduce the occurrences of feedback, but I cannot eliminate it.

This musical moves pretty quickly between acts and scenes with no intermission. This means that I have very little time to set up audio for the next scene. Luckily, the venue I’m mixing in has a digital mixer which allows me to save mix settings. I’ve found that saving presets for each significantly different scene helps a great deal. I can set up default mutes for actors that aren’t onstage at the beginning of a scene, and I can also set mutes and un-mutes for the monitors for when the actors are just speaking and not singing. The downside of this is that if I want to adjust say the pre-amp setting of an actor’s microphone, I must do it across all preset scenes which can’t exactly be done on the fly. If I need to adjust more than one actor’s settings, that multiplies the work. Take the amount of changes you need to do across the board, and multiply it by the number of presets. You can see how one small change could be a big problem.

For this particular musical which lasts just an hour and a half, I have 19 different scene presets. Even with the presets, I have a copy of the script beside me that I’ve written all over that includes my own scripted movements. For example, when one actor goes into a whisper, I have a note to myself to raise the gain on his channel and bring it back down again after that short section. As well as I think I know to make certain changes, I write everything down because when it’s showtime, I can get caught up in the moment and forget. I never want my mistakes to kill the actors’ flow.

One of the biggest things I’ve learned is to let things out of my control go. In a studio setting, you can control almost everything in the environment down to the performances that are recorded for posterity. In live situations, anything can and does happen. Sometimes an actor will position their microphone in a position that pops when they move in a certain situation. There’s nothing you can do during the performance except compress their channel so that pops don’t feedback into their microphones and maybe ride the mutes along with their dialogue.

Coming from a studio background, the “in the moment”-ness of live sound mixing is a refreshing change. It’s definitely been an eye-opening experience. I was a bit apprehensive when I first took on the project, but I’m glad I had a chance to be a part of it. And then there are great moments when I can forget about what needs to be done, and just lose myself in mixing the music; it’s fantastic.

If you’ve mixed live sound, do you have any tips to share?

December 19th, 2010 – With the amount of cool apps coming out for iOS, there hasn’t been a whole lot of good hardware solutions for getting great sound in and out of the iPhone/iPod Touch. Sure there are some for plugging in a guitar, but they aren’t exactly the highest quality. New Signal Process, a boutique shop out of Portland Oregon, has released the NSP BreakOut to remedy that situation.

The BreakOut has an input to get your audio into whatever music apps you’re running in iOS, and an output to send to your outboard gear. There’s a gain knob and an on/bypass switch and that’s pretty much it; no muss, no fuss. You can buy the BreakOut for $115 at the New Signal Process website.

NSP BreakOut

November 11, 2010 – When I first started working with Logic, I immediately maxed out its processing power on my laptop within the first 5 minutes. In the time since then, I’ve done full 36 track mixes with effects, and mixed my own complete productions; all of this involved eking out as much processing and hard drive performance as I could from my laptop.

One of the problems with Apple is that upgrading your machine (in terms of processors) just doesn’t happen. As such, most users who spent all their money on their current computer often don’t have the money to upgrade as often as a PC user might(although I’m sure the rich fashionable set do). I’m sure Apple’s Logic team (consisting of legacy Emagic employees) quickly understood the limitations of Apple’s closed hardware model where a user might not be able to upgrade their hardware, and started to program clever little ways for a user to get the performance they need.

One thing they added was the ability to “freeze” tracks. Freezing a track basically tells the DAW to bounce down an a new audio track that’s a recording of the original with all the effects. At this point, the DAW will disable the original track and all the effects, and play the recording which has the effects “printed” on it. All you play is an audio file, thereby saving processing power, and the process is completely reversible.

The other awesome feature that they added was smart disabling of effects. As anyone who has used Space Designer can tell you, Logic effects are processor hungry. Smart bypassing means that when a track isn’t using an effect, Logic will automatically disable it upon playback; this saves your processing power for when you really need it.

Of course, this relies on Logic to make the decision for you. Because I don’t like to leave anything to chance, I automate the bypassing manually. If you press “A” on the arrange page, you will be able to draw in automation curves. I actually prefer drawing in automation to twiddling fake knobs (that’s what she said), so I will do volume automation almost exclusively here.

When you click down the dropdown for the track you want to automate, you will be given a menu that has a few standard options like Volume, however you will want to navigate further to “Main”. In this section, you will see a number of inserts corresponding to the effects you have inserted on the track. Select the insert number of the effect you want to automate the bypass for (the numbers represent effects top to bottom on your track). A coloured line will appear over the track in the Arrange window. Draw in your bypass automation which will be either on or off.

A quick note, smart disabling may give you a false sense of the max processor load since it will disable the effects in parts where you don’t need it, but if you have a chorus that has a lot of effects, bringing them all in at once may cause a core audio overload. You might not have expected this since the verse didn’t experience the same problem when the effects were smart disabled.

Hope this helps!

November 3rd, 2010 – I’m sure you’ve all heard that Protools, along with going native, now has an SE version that ships with either a USB microphone, an interface similar to the Mbox mini, or a MIDI keyboard. You may also have heard that you don’t have to go to your local Guitar Center to buy one either; you can get them at Best Buy.

Power users may scoff at the baby Protools (after all it has a maximum of 16 tracks) however I wouldn’t be so quick to dismiss it. At prices starting at $99 for the Avid Vocal Studio, $119 for the Avid Recording Studio, and $129 for the KeyStudio, the value is strong.

For the consumer that’s going to be picking up one of these bundles, 16 tracks will probably be more than adequate. Of course, maybe my perspective is a little skewed; when I first started recording, it was on four track tape reels (cassette and ADAT were way out of my budget). I quickly learned how to bounce down tracks to be able to record more.

Even when I tried the first versions of Cakewalk (one of the first consumer DAWs), I was limited in tracks because the computers weren’t fast enough back then. To me, 16 tracks would have been a huge luxury. But even today, I think that it would be adequate. Most of the people who are going to be buying the pack, are likely going to be recording themselves and won’t need tonnes of tracks.

I think Avid is being pretty smart with these low-cost offerings. They are including everything a person needs to get started at a super low pricepoint. When the user is ready to move on, there are plenty of higher priced offerings. I’m sure this is a direct effort against Apple’s DAW strategy: bundle Garageband for free, get users hooked and interested in Logic Express which is also very inexpensive, and then finally up to Logic Studio. Avid is one-upping this strategy by bundling hardware; what used to be a detriment in their professional offerings is now an asset in their lower ones.

When I worked in someone else’s studio, I used a combination of Protools and Logic. Logic to write songs, Protools to record vocals, and back into Logic for mixing. Even with all the improvements Logic has made with their audio editing workflow, I still find Protools faster, and more intuitive.

I am not currently running Protools (although I could get the M-Audio editions since I have FW410), so these low cost versions are enticing. As I would be using it to cut vocals only (16 tracks should be adequate for recording quick and dirty demos), the Vocal Studio is the most interesting to me. A hundred bucks for an M-Audio (on a side note, I’m sad that M-Audio as a separate brand from Avid no longer exists) USB microphone and Protools editing workflow? That seems like a very compact mobile recording solution.

I’m actually curious to hear what the bundled gear sounds like. Obviously the microphone isn’t going to compete with any Neumann, but will the interface included with the Avid Recording Studio hold up to my old FW410, or even an Mbox Mini? Have any of you had a chance to try them out?

Avid Vocal Studio

October 17th, 2010 – Imagine you’re a mix engineer and you get a song to mix recorded by another engineer. Unfortunately, it has a lot of noise: hissing, pops, background noise. When there’s something playing on that track, for example the vocalist is singing, you can barely hear the noise, but if the singer isn’t singing constantly, you can hear it.

Perhaps it will be hidden by other instruments that play during those gaps, but when you’re trying to do the best job possible, you want to take out as much of that noise as possible, so it doesn’t compound upon itself (unless of course, you’re going for a certain stylistic choice).

There are three main ways to deal with this noise: you can cut out the unnecessary spaces in the audio to minimize noise, you can automate the volume to either mute or turn down the audio, or you can use a noise gate.

A noise gate is a device that acts very similar to a compressor, with one huge difference; instead of turning the volume up in quiet parts, a noise gate turns the volume down. In the picture above (I should probably point out that I just opened and screenshotted the noise gate, it’s not set up for the project I’m working on), you can see that the main settings on the device are shared with compressors: threshold, reduction, attack, hold, and release.

Like on a compressor, the threshold is user settable point that triggers the device. On a compressor, when the source goes above the threshold, compression is triggered. On a noise gate, when the source goes below the threshold, the noise gate will clamp down on the audio. Just how much it will lower the volume depends on the amount of reduction that is set. Attack, hold, and release, as with compression, need to be set on a source by source basis.

I should also mention that noise gates aren’t always used as a tool to get rid of noise; sometimes they’re used as an effect. If you listen to a lot of electronic music, you will no doubt recognize stuttering vocals and synths that pulse in rhythm to the song. While sometimes the effect is managed manually through edits and automation, it has been traditionally created through the use of gates. Usually, a noise gate will be put on the desired track, but it will be triggered by something on a different track using the noise gate’s sidechain option. For more information on side-chaining check out this previous post.

October 4th, 2010 – As I slowly install all the music software back onto my hard drive, it occured to me that some of you may not know that it’s a good idea to put all your samples and project files onto a second hard drive. Not only will you save space on your main drive, but offloading some of the reading and writing of audio onto another drive will prevent clicks and pops, or even writing errors when recording. Spreading the heavy lifting will help your system handle the load more effectively.

Of course, your second drive will need to be able to write and read at a quick rate so you’ll need to be running in a second hard drive slot in your PC, or at least a Firewire 400 connection. I personally don’t use USB 2.0 for audio, but I know that some do, so if you need to save some money and don’t have a Firewire connection, you can use that at the very least. Happy recording!

October 1st, 2010 – Today I want to talk a bit about when to make edits, and when to warp audio tracks. While there are some artists that are vehemently opposed to any editing of their performance, most want you as an engineer to make them sound as good as possible. In many cases, this will include fixing little timing issues. Sometimes you won’t be able to re-record a performance (time is money after all), so you will need to make changes to the recorded performance after the artist has left.

There are two choices when it comes to fixing timing: editing and warping. With editing, you are cutting the audio track apart, and pasting it back together in time. Engineers have been doing this as long as they’ve had magnetic tape to work with (they physically cut the tape with a razor blade and used adhesive tape to put it back together). The benefit of this method is that the audio performance is exactly the same, just shifted on the timeline. The downsides are that it’s time consuming to “pocket” all the audio if there is a lot to fix, and some things are impossible (or at least very difficult) to fix. For example, if you want to fix timing on a word that drags out over a long melodic phrase.

The modern way to fix timing is warping. Warping uses a software algorithm to stretch and render new audio to fit user-defined timing. Generally a user will define anchor points and then drag the audio around within anchor points to set timing. The benefits are that it’s extremely easy to do, and you can adjust almost everything. The downside is that it’s a software rendition of the original performance so some fidelity is lost. In situations where audio is stretched an extreme amount, the loss of quality is readily apparent.

So when should you use each? As usual, I’d like to first state that there are no hard rules, you should figure out what works best for you. There are, however, situations that generally call for one method over another. When fixing timing issues with drums, I would suggest editing the audio over warping. Editing will keep the pitch and impact of the original audio, and since the hits are short, it lends itself to clean edits. For working with melodic instruments, such as guitars, bass and vocals, I would edit as much as I could between phrases, and then use warping for detailed pitched audio phrases. For example, a guitar note that is played legato with the note before and rings out would be hard to edit. That’s a good place to warp instead.

On an interesting side note, Reaper seems to have a pretty effective system for pocketing audio. You can move audio portions of a track around as if they are being warped, but they are actually just being edited, with extra space added or removed when needed. Of course, it doesn’t stretch the audio if that’s what you need, but It’s a pretty nifty time saver for edits.

September 18th, 2010 – The NS10, NS10, NS10; to a beginner it must seem that audio engineers have an infatuation with it. Indeed, the signature white cone on black cabinet is so recognizable, people who aren’t audio geeks might think all studio speakers look like that.

Earlier this week, I linked a Gizmodo article on the Yammies through Twitter (btw, add me here!). Later I figured that I might go a step further and provide a succinct review of the venerable speakers.

In what seems forever ago, I worked in a small studio where the producer swore by NS-10s. You know the oft repeated stories about how some engineers experimented with how many sheets of tissue paper (along with what brand, and ply count) were best for monitoring? Yeah, he was one of those guys.

Truthfully though, you really do need to use tissue paper over the tweeter. The speakers have such a pronounced mid-range that the tweeter pierces your ears like hot needles. I’m absolutely serious: the NS10′s are incredibly unpleasant to listen to. Not only that, the low-end is pretty weak (maybe something to do with the completely sealed design, although I’m not an acoustics expert).

That being said, some of my best mixes used the NS10. The mid-range for many engineers is the most difficult frequency range to deal with because almost every instrument has frequencies present there. Because the NS10 accentuates this range, it makes it easy to hear what is masking, clashing, and otherwise muddying up your mix. The bottom line is that the speakers are terrible for enjoying music but great for checking your mix against (you would probably want other monitors as mains, preferably something with bass).

Recently Jon over at AudioGeekZine had me over to his studio to chill out and make some noise, and he has a set of HS50M’s which are supposed to be the spiritual successor to the NS10′s. Unlike the NS10′s though, the HS50M’s possess an active design (the fancy way of saying that you don’t need a separate amp to drive them) which helps reduce the variability the amp brings to the sound. As Jon pointed out, and to which I agreed, the HS50M sounds completely different.

While I have read reviews that the HS50M can somewhat emulate the mid-range push of the NS10, it does so in a way that sounds completely different. The big difference is that the HS50M is a much better designed speaker and therefore sounds a lot better. Does this make it better to mix with? I’m not sure.

The HS50M sounds pretty good, but with a lot of my music requiring bass awareness, a proper set up would require the optional sub woofer. That brings the price up considerably, and puts it in the sights of a lot of competitors. The HS80M has a lot more bass given its 8 inch woofer (and costs less than the HS50M with the subwoofer), but I’ve read that they are not as tight sounding.

The NS10 is a great speaker for checking your mix, but you probably wouldn’t want one as your main set of monitors. That being said, I’d love to have a set again.

September 15th, 2010 – I am not a professional skateboader, snowboarder, or mountain biker, however living in Vancouver, I come to know and hang out with a few athletes. I have never been in contention for pro-level status but even I can tell when an athlete has good flow.

Have you ever watched videos of athletes that seem really smooth and link everything they do into one continuous stream of talent? It’s pretty obvious when watching a snowboarder ride powder, but even basketball players exhibit this. At the highest levels of achievement, that flow translates into a certain confidence and smoothness that helps individuals excel.

In music, I try to capture the same feeling. While there are usually distinct parts (verse, chorus, bridge, etc) in a song, I try to get a good flow going. Everything must make sense, and there shouldn’t be any extraneous “fat” that hinders the song.

It may seem a bit weird to think about music having the same type of flow as top athletes possess but I’m convinced that they’re one and the same. Music with good flow is direct and to the point; there isn’t anything that distracts from the message. It progresses confidently and smoothly. This doesn’t mean that every song has to be minimalist, but if an element doesn’t add anything, it doesn’t need to be there.

The concept of flow can be applied when writing a song, producing it, or recording and mixing it. When recording and producing the song, consider if an instrument really needs to be there, or if a melody is appropriate for that section. When mixing a song, think about whether that horn needs a reverb as an insert and as a send. Conversely, maybe something needs to be added to make the song flow properly. Perhaps a glitchy looped sample, building to a synched rhythmic gate (as I heard in a Wolfgang Gartner song today) is just the buildup that’s needed.

I realize that flow is a pretty esoteric idea to apply musically. I believe, however, that it’s the key to music resonating with as many people as possible. Flow is something that is felt without necessarily being obvious, and it helps make music feel right.

September 6th, 2010 – I’ve had my M-Audio Firewire 410 for a long time. I picked it up in 2004 when I needed a mobile recording solution and I still use it. Lately though, I’ve been having audio drop outs. The software mixer will register audio output, and the card itself will light up its LEDs, but no audio will play. I can unplug and replug the interface back in, and it still won’t register.

What’s especially weird, is that it will be already working and cut out. For example, I’ll be listening to my iTunes library and all of the sudden a song will cut out and I won’t be able to get any audio back. I used to restart my computer, but that was also a crap shoot if the sound would work again.

Apparently, this problem is not localized to me; I’ve read a lot of complaints on the forums about it with others who are on Leopard. M-Audio hasn’t updated the drivers since November of last year and since it’s a discontinued product, I doubt they’re rushing to fix it. I’ve completely removed and re-installed the drivers several times to no avail.

In the meantime, the only way to bring the audio back consistently is to go into the M-Audio software and change the sample rate, and then change it back. Doing so allows you to get the audio back without rebooting, or hotswapping the interface. Are any of you running into the same problems in Snow Leopard?