What is stereo? One of Wikipedia’s definitions is “stereo usually means 2-channel sound recording and sound reproduction using data for more than one speaker simultaneously.” I think that this is a very good definition because it is simple and does not assume too much. Stereo is not synonymous with panning or just sound coming out of two speakers. Stereo is not a name for a fancy musical playback setup. It is simply two independent sound channels that play back simultaneously. One of the conditions of the above statement is that in order to each channel to be played back independently two speakers are needed. Things really get interesting if the person who made the two channels understands how binaural hearing, the human ear, and these two sound sources can be used together to create some interesting effects and a stereo image.

Part of understanding how to create a good stereo images is to first understand a little bit about how binaural hearing works (humans by default are binaural). Binaural literally means having two ears. If we’ve learned anything about sound it’s that as the number our sound sources and the number our sound receivers (ears and microphones) increases the complexity and possibilities of mixing those sounds increases exponentially. There are three main things to pay attention to when mixing in stereo for a binaural audience (that’s all the time these days right).

1. Inter aural amplitude differences

2. Inter aural timing differences

3. The head related transfer function

(When I say inter aural I’m basically referring to how the sound is different in the left and right ears.)

Inter Aural Timing Differences
Note that the list above works from the low end of the frequency spectrum to the high end, I’ve started with timing differences because they are the most commonly manipulated by engineers. Inter aural timing differences refer to the difference in time that it takes the same sound to reach either ear. Image you are mixing and you’ve panned the rhythm guitar hard left. If you take a minute to think about how your brain figures out that the rhythm guitar is now coming from the left it’s because your left ear actually hears it before the right hear (ever so slightly). Depending on how much later your right ear hears the sound helps your brain actually place the sound in 2 dimensional space. This is how you know often were to look and verify the source of any sound, be it a car approaching, a bee on a flower, or where the phone is when it rings. Your brain does this well with frequencies roughly between 150Hz and 1.5 KHz beyond those ranges the waveforms of the sounds become either too big or too small for the brain to decipher in this way.

If we take a step back in the frequency spectrum and look at how our brain handles frequencies below 150 Hz we’ll find some interesting things. The waveform of 150 Hz is 2.25m or 7.4 feet in length. The size of these low frequency waveforms make it seem like the low frequencies are coming from everywhere, if you think about it one or two cycle could easily fill a room. This is why subwoofer placement in a surround sound system is not as critical as the other speakers. This is also why by default the bass is typically panned up the middle. If you try panning the bass guitar to the hard left or right in a mix it’s hard to close your eyes and place it, it will likely sound like it’s still in the middle. The lower the frequency the less control you have over it in the stereo image. There is a technique that can work for lower frequencies but begins to lose effectiveness around 80-100 Hz.

You can fabricate the timing difference between speakers by manually delaying one of the signals rather than panning. This technique is sometimes referred to at the Haas Panner Technique. If you delay the sound in the right channel somewhere between 0.2 and 1.12 ms (9-49 samples at 44.1KHz sampling rate), you can effectively make the sound seem to come from the left speaker. Note that DAW without plug-in delay compensation can yield effect when it is unintended or unwanted.

The Head Related Transfer Function
The Head Related Transfer Function has to do with the complex way that all the junk between our ears affects the way sound waves hit our ears. Due to the size, density and shape of the human head higher frequencies or smaller wavelengths will exhibit more pronounced effects. Typically frequencies above 2.5 KHz are most effected by the head. The head transfer function produces both phase and frequency effects, so the best way to counteract or control these effects are through EQ. Here is another one of those places in record where the term “There is no such thing as a free lunch” really applies. EQ used as a tool to add and blend layers to a mix, but it can also have an effect on the stereo image. Unfortunately there are no hard fast rules for EQing for a good mix and EQ for a good stereo image, every mix will be different, the process to marry the two will undoubtedly be careful listening and trail and error tweaking.

Inter Aural Amplitude Differences
Inter Aural Amplitude Differences are possibly the easiest to achieve. They occur when a sound is simply louder in one ear than the other. The easiest way to manipulate this is panning, but you could also accomplish this by processing the left channel differently from the right, though you will undoubtedly add some of the other aforementioned effects as well.

Another thing to mention about stereo image is which mic techniques are most conducive to stereo imaging and which techniques can reproduce the effects above. Below is a chart of which techniques will capture which effects.

End

Getting drums to sound big, full, and all of the other words we use for excellent sounding drums has got to be one of the toughest jobs in the business. There are mic choices, placement choices, room choices, damping choices, drummer choices, too many choices all of which add variables to the X+Y*Z = Excellent sounding drums equation. In this article I intend to discuss only a couple of these topics, namely mic choice and mic placement.

One of the first decisions you must make when beginning to setup to record drums is how important is a stereo image of the kit, and how do you plan on achieve that image. Spaced pair overheads are a popular choice as well as the M+S setup. I tend to prefer the M+S setup myself as long as a few conditions are met: 1) the room is not too reflective (which can ruin any stereo image), and 2) do I have a nice omni and figure of 8 mic on hand. Given these two conditions I nearly always choose an M+S setup for the ambient mics on a drum kit.

I prefer this setup for numerous reasons. The first is that omni-directional mic have the best bass response of all the polar patterns. Step one of capturing a sound is having a mic that can capture the whole sound. (Note: though these techniques can be used alone, I typically use them in conjunction with close mics as well) That said a good kick mic like a D112 or Beta 52A can capture much of the sound that we are used to hearing from a kick drum on a recording. The thing to consider here is when was the last time you put your head into a kick drum and listened to the drummer play. Hopefully your answer was never because doing so could be very detrimental to you health. Needless to say most sounds need both time and space to mature, and because we don’t stick our heads in bass drums we are naturally used to hearing a mature kick sound. Now the trick becomes finding where this mature sound is most pronounced and then place a microphone that is capable of capturing it there.

This will often make people point out something like “Hey, if we put the omni there we have to put the bi-directional there as well.” and I will respond with something like “So?” Most people don’t like my answer and try to explain to me why it is a horrible decision to place the figure of 8 that low and most of their arguments stem from some kind of “Well what about the cymbals” thing. More often than not the drummer does not have a separate set of cymbals for the studio, this means that the same loud and ringy cymbals that he uses live are the ones he/she is using in the studio. When was the last time you reached to turn up the cymbals in the mix. Most times I’m struggling to get the cymbals out of the some of the close mics because they are too loud. Case and point you don’t put your ears over the cymbals like traditional overheads, and when seeing a live show the drum kit is almost always elevated so the mic is actually closer to where your ears would be at a live show down there.

Here is the problem anytime you use close micing and distant micing at the same time you will encounter the natural phenomenon sometime know as “undesirable number one” in recording, Phase. In the past phase was a much bigger pain in the ass than it is to day with the non-linear editing capabilities of digital audio. By moving the the distant mics ever so slightly earlier in the timeline of your DAW you can eliminate some or much of you phase problem. Notice I said much and some, in a 2D world this would be much less complicated, but that third dimension adds all kinds of new reflections to our recording environment, meaning you never get perfect in-phase signals from two mics. You can get a rough estimate by measuring the distance from the source (kick drum) and knowing the fact in the typical earth environment sound travels at (1124 ft/s, or 344 m/s) so your time would be your distance divided by 1124 or 344 depending on your measurement system. Always good to back this up with a careful listen, you could even solo or route the two signals you are trying to match through a phase meter and bump/nudge the distant mic earlier in the timeline until the best phase is achieved.

This technique can be used to on any distant mic in any situation close and distant micing a guitar amp for example, to try an improve phase results. You could even use the inside kick mic as the reference for your entire kit and move all of the other tracks earlier in time according to their distances from the kick mic.

I was recently glancing over some of the articles popping up on home recording blogs and came across this so called “Recorderman” overhead drum mic placement technique. A good technique built on sound principles. This brings to light some interesting thoughts. This technique can be expanded and applied to any multiple microphone situations. The process of keeping both mics equal distances from the main sound source (in the given example the kick and snare) eliminates what is commonly referred to as phasing.

Phasing occurs when the same sound arrives at two different mics at different times. Sound travels at a particular speed, but always at the same speed (in the same medium, in most cases in the studio: Air). Therefore if the microphones are the same distance from the source and point at generally the same part of the source, they should yield a similar sound (especially if they are a matched pair).

When used in this way multiple microphone techniques can yield good “images” of a sound source with minimal phase issues. The “image”, referring to the stereo placement of sounds, is the sound coming from the left or right side. This also is the logic behind coincident mic techniques i.e. M+S and XY where the mic capsules are placed as close together as possible (which in turn makes them equal distances from the source).

The downside of phasing and the use of multiple microphones is a phenomenon known as comb filtering. When out of phase signals are summed (mixed together) parts of each wave form will cancel, resulting in silence. This is not a general silence but the elimination of specific frequency depending on how out of phase the signals are. Frequencies very near the canceled frequency will also be affected to less and less of a degree as you move away from the canceled frequency. This is very much like applying a notch filter to a sound to eliminate a particular frequency. To make matters worse this notch effect will also occur at each harmonic of the canceled frequency. For example, if 800 Hz is canceled, then 1600, 3200, 6400, and 12800 Hz will also be affected. If you can imagine an EQ with a notch filter on each of the above frequencies you should get an idea of why they call it “comb” filtering. This effect can be simulated and sometimes overlooked when using a delay plug-in. Sweeping the delay time of a delay plug-in will sound similar to sweeping a notch filter up and down the frequency spectrum. Try it!

In conclusion, may I offer two additonal pearls wisdom? 1. Don’t use more mics than necessary. 2. If you want to use more than one mic to create a stereo image be very intentional about how you do it. (A tailors cloth tape measure can be very good investment in a studio.)