Hi! I recently purchased a digi-003r with protools and have been making considerable progress learning the software but am now trying to understand and implement sends and busing.

Can you point me in the right direction with suggested reading material, I would be most grateful.

Thanks Leagh

Leagh, thanks for the question. A good resource for general recording techniques is:

Modern Recording Techniques, Sixth Edition
For ProTools specifically:

Pro Tools 101

Pro Tools 101 Official Courseware, Version 7.4

Pro Tools 7 for Macintosh and Windows (Visual QuickStart Guide)
for a quick reference.

I’m not sure if you have a grasp of how sends and busses are implemented in general so I apologize if some of this post tells you what you already know.

In brief sends allow you to make copies of channels and busses let you collect those copies and sum (add) them together. The master fader in Pro Tools (and on any console virtual or real) is the last bus in the chain and adds all things sent to it together. Some other common uses for sends and busses are to send copies of multiple channels to the same effect (on a single bus) on a DAW this can help save precious RAM and CPU resources, they are also sometimes used to make groups and stems when mixing.

In ProTools it is important to understand the difference between and insert and a send. Inserts grab the recorded audio send it to the plug-in and return the result right back to the channel (often to the next insert in the channel). Inserts work from top to bottom so the order does matter. This lets you decide for instance if you would like to compress a signal first and then EQ or vise-versa.

Sends on the other hand offer some more functionality. First of all they come in two flavors Pre-Fader, and Post-Fader. This means you can take your copy before the inserts and the fader (meaning the send level is independent of the channel fader), this is called Pre-Fader. A post fader send takes a copy from after the fader meaning that the level sent depends on the channel fader. Both flavors also offer a send level and panning if appropriate.

Pre-Fader sends are helpful when you are trying to mix wet and dry components of a signal. The signal with no processing (dry) is available pre-fader, you could then send that signal to say a chorus effect on a bus and return that bus to the master fader or mix bus.

Busses act much like and audio channel in Pro Tools except their input comes from sends rather than reading from an audio file. They are useful to collect groups of sends (or single sends) and apply processing across all of the signals sent to the bus. Note you can send the output of channels to a bus as well, but in that case the signal is only going to the bus and not the designated output (typically the master fader) and the bus.

Implementation in Pro Tools is quite simple. Create an AUX channel and set it’s input to bus or busses (say 15-16). Then use the sends from the desired audio channels (Pre or Post Fader) to send to the same bus/busses (15-16). You can actually rename the busses in the I/O setup of ProTools to make life easier. Instead of seeing busses 15-16 on the drop down menu you could see something useful like “Long Reverb.”

Hopefully this answers you question and in a timely fashion.

Cheers-
Ian

I was sent this material to review with emphasis that it was recorded completely analog (sadly I was delivered mp3s so there are some digital artifacts that I’m listening to.) That said this blog deals primarily with DIY, Home, and project studio work and in most cases today that also means digital recording. With our further ado, the review for “Beneath & Beyond” by Standfast.
standfast unknown video

Let’s get one thing out of the way, in my opinion drums always sound exceptionally better coming off of tape. There is a fullness in the kick and a smoothness of the highs and cymbals that complements a drum kit very well. I’m glad to hear that they left most of the drum processing to the tape.

In most cases the the guitar is bland, but don’t think the album ever really aims to showcase the guitar, so all in all it sits well and serves a purpose. (Beginning of “Getting Off” show a bit of the potential guitar sounds).

I like the various keyed instruments very much, and though this is an all analog recording I think it exudes a lot for freshness by good multipart music writing and interesting instrumentation.

The vocal layering is also a highlight. Sounds like good levels were brought to tape as the hiss is very low in the recording (hi tech mastering may also get the credit for this).

All in all though what I would suggest most to listeners/readers of this blog should listen to is the sound of the drums. I think this recording it good example of what we want to try and get our drums to sound like using plug-ins like Analog Channel by McDSP and other plug-ins used to add analog warmth.

A very nice recording from Junk Musik, and Standfast.

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.

Continued from The Basics of Mixing (Part 2)

Special Effects/Spot Effects
Special Effects and Spot Effects are like the engineer’s solo in the song. These are the parts that the band may not have been able to do without the help of the engineer. The trick here is to use the effect sparingly and tastefully. Again just because you can do it does not mean that it supports and enhances the song. As the engineer you can take pride in adding these effects, but you also must listen to the band if they do not think that it matches their artistic vision. This is job security territory so that is about all you’re going to get out of me, but keep a copy of all the work you do and experiment when you have time on a slow an rainy day.

Finishing Touches
The finishing touches include any last minute edits or mix tweaks. Remember to apply a dither if you are bouncing down for CD (you did track at 44.1kHz or 48kHz with a 24 bit bit-depth right, there is no good reason not too anymore). Dither is one of the hardest things to describe, everyone understands that it makes a 24bit file 16bits but what does it really do? The best analogy that I’ve heard is this: Imagine a painting in the distance, you are viewing the painting through a window that you are holding. The window isn’t quite big enough to allow you to see the whole picture at once, but if you move it slightly in any direction you can see that edge of the picture. Now imagine that you could shake that picture very fast, like inhumanly fast. If you shook the window frame fast enough you would no longer see the frame and you would be able to see the whole picture (like looking through a fan). This basically what dither does to sound, but most of us can see a loss or resolution when looking through a fan, and some of us can hear the loss of resolution due to dither. Dither does effect the resolution of the sound so it’s a good idea to apply master fades after dithering the mix. I don’t think we will use dither too long as the only drawbacks of publishing audio as 24bit files is slightly more storage space and the fact that most hardware does not yet accept it, but things will change, the always do.


If you are going to have you track mastered don’t dither anything, just bounce files that are the same data type as the rest of your session. A mastering engineer (especially one that masters digitally) can do a lot more with a 24bit file than a 16bit file because there is more head-room. Chances are the mastering engineer will have a nicer dither algorithm as well. Another note when prepping for a mastering engineer. Leave them some headroom to work with, this will let him or her do what they are best at, instead of trying to apply work-arounds to a mix that has no head-room left. Usually 2-3dB is a nice amount for them to work with. The higher the quality of file you can bring them the better, I’ve even heard of people brining the computer and interface they recorded with and bounced the mix directly into the mastering engineers hardware. (check to see if your mastering house offers this and if the charge more/less for that kind of thing)

Test Monitoring
This possibly should have gone before for the later step, but many find that this is where the mix is really completed. When you have bounce your mix be sure and demo it on as may practical sources as possible. Play it in your car, on you iPod in headphones, on your home stereo, on a boombox, through your cheesy computer speakers (or nice ones if you have them). Make sure that your mix has consistency from player to player and adjust accordingly. One of the olds sayings is that if you can get a mix sounding good on a pair of Yamaha NS-10s (studio monitors that are very unforgiving) then your mix will sound good anywhere. If you get a chance listen to some of your favorite CDs thourgh NS-10s you’ll be surprised at how hard they are to please (or get anything pleasing out of). That’s it easy as pie right. Good luck and have fun.