Here is a great feature explaining the manufacturing process for making vinyl records.
Understanding the manufacturing process helps us understand the technical aspects needed from an engineering standpoint.
These aspects are handled my the Mastering Engineer, and as producers and mixing engineers we will greatly benefit from knowing this. It is also very interesting.
Part 1:
Part 2:
The term “Polar Pattern” relates to the directionality of a microphone. What is directionality? Sound sources can come from different positions and directions around the microphone. Polar patterns tell us from which direction the microphone is most sensitive to sound, and from where the microphone will be less sensitive, and even completely reject a sound. It is necessary to understand this concept and its implications when discussing microphone techniques. The most basic as well as the most advanced techniques make use of the directional characteristics of a microphone.
The above information tells us that a microphone has a front and a back, and sound sources coming from the front of the microphone will sound different from sounds coming from behind the mic or from the sides. A sound source located in front of the microphone is called an “on axis” sound, and sound sources on the sides or behind the mic are called “off axis.” When looking at a microphone, the first thing you have to understand is where the front of the mic is (where the on-axis sounds will be coming from).
The visual representation of a mic’s polar pattern is presented as a two-dimensional “slice”, where the circle around an axis shows the sensitivity (in dB) at the various angles around the microphone’s diaphragm. The farther away the line is form the center axis, the more sensitive the mic from in that direction (meaning, the sound will be louder coming from that direction) and the closer the line gets, the less sensitive the pickup of the microphone from that direction (meaning, less level from the sounds coming from that direction – also referred to as sound rejection).
In the above example, the diagram on the right shows a microphone that is equally sensitive from every direction. This type of microphone is called an “Omnidirectional” microphone (sometimes also called “Unidirectional”). The left diagram represents a microphone that is most sensitive to on-axis sound (the sound sources that are in front of the microphone), where sound coming from the back of the microphone is completely rejected (this means that in an “ideal” environment an instrument playing behind the mic isn’t picked up at all). This type of microphone is a directional microphone, and because the diagram looks like a heart, is called a “cardiod” microphone.
In reality the way microphones pick up sound is more like a three-dimensional spherical ball around the diaphragm. The illustrations below, found on the Shure website, give a nice representation on how the above polar patterns would translate to a real microphone:
As you can see, the polar pattern diagram is basically a two-dimensional slice from above the microphone (kind of like an MRI shows slices of your body from the head down).
The three common types of microphone polar patterns are “cardioid”, “omnidirectional”, and “bi-directional” (also called “figure-eight”). Some mics have a fixed polar pattern, while others have a switching mechanism where the engineer can decide which polar pattern he wants to utilize on a particular recording.
The bi-directional polar patterns looks like this:
A bi-directional microphone picks up sound equally from the front and the rear, while completely rejecting sounds coming from either side of the diaphragm. There are other types of polar patterns, and they are variations and combinations of these basic types. We will discuss those in future posts.
This was brief introduction to microphone polar patterns are. On the next post I will start to discuss some of the meanings, implications and applications for these different types of mics. Happy reading!
A very big difference between Pro Tools HD and LE (this also includes M-Powered) is the issue of delay compensation. In Pro Tools HD you can activate “automatic delay compensation” and the TDM engine will compensate for latency introduced to each individual track. Pro Tools LE (including M-Powered) does not have this feature.
Crash Course:
TDM stands for Time Division Multiplexing. Pro Tools HD is a TDM system. This refers to the method by which the signal processing is performed on PCI cards rather than by the host computer which is running the program. LE and M-Powered systems are called “Native” systems, where all of the digital signal processing has to be handled by the CPU of the host computer that is running Pro Tools.
Audio Plug-In – a plug-in is a program that runs within a host-program. It is inserted in a digital audio path and performs a mathematical process (known as an algorithm) to the bits that represent the audio. Examples of plug-ins are EQs, compressors, delays, reverbs…
Plug-In Delay or Latency - Some plug-ins have algorithms that introduce a short amount of delay from the time the audio “enters” the plug-in on one end and the processed audio “exits” the plug-in on the other end. This short delay is commonly referred to as latency. An example for such a plug-in is a look-ahead plug-in such as certain peak limiters or a real-time vocal tuner that has to delay the output by a certain amount in order to “look ahead” before it processes the audio.
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Latency can also occur by using sends to bus audio from one track to another, usually for the purpose of effects such as reverb or parallel compression. The short delays introduced in this way can have a very small effect on the outcome (such as reverb sends). Other times, the latency can create artifacts that are very audible, to the point where you cannot even use these effects in native systems (for example when you try to use sends to do parallel compression, which we will get to in a future post). If you have two very similar tracks, the introduction of delay to one of those tracks can create phase cancellations and cause a comb-filtering effect like a flanger.
In cases where these short delays may not be very audible, they still cause your mix to lose clarity and punch. If you were mixing in the box you may not notice this latency too much, but if you heard that same mix without the latency – the mix would instantly gain clarity, definition and punch. I bet you would choose to use the delay-compensated mix 
.
One way to get around this issue is to manually nudge each track to compensate for the delay. In the Pro Tools Mix window, at the bottom of the fader there is a volume readout. If you command-click (or for you Windows users, control-click) on that “vol” readout once, it will change to a “peak” readout (telling you what the highest meter peak was on that track). If you click the same spot a second time, you will get a delay readout labeled “dly” which is labeled in samples. In this case, the sax track, which has several plug-ins inserted on it, has a delay of 64 samples.
We would now have two options: We would either have to go to the edit window, select the region (or all of the regions in the case of edits or multiple takes) and nudge them to the left (earlier) by 64 samples. The second choice would be to insert a “time adjuster” plug-in on all of the other tracks and delay all of them by 64 samples. This is only if the sax is the only track with any delay on it. If more tracks have different delay values (which could very well be the case) it can get complicated to manage. Add to that the scenario where you may want to continue inserting or removing plug-ins on the sax track or any other track, and then you have to go back and nudge the regions again, or change the time adjuster values (for all of the other tracks). As you can see, it can be a real pain.
I have recently learned about a great plug-in that is specifically designed to solve this problem of track delays. It was designed in a smart way that can handle delays caused by just about any reason, and it is fairly simple to get a handle on its operation.
Mellowmuse ATA
The plug-in is made by a company called Mellowmuse and is called ATA, for Auto Time Adjuster. I will provide an explanation here but if you want to watch some video tutorials and read about it directly on their site, you can go to http://www.mellowmuse.com/ATA.html
The basic idea behind this plugin is that you insert it in the first insert slot on each track, before any other plug-in. You have to designate the type of track (audio, aux or master) and you need to have a master fader with the “master” designated ATA plug-in inserted in the session. Once this is set up, you open the “master” plug in and hit the “P” button. This stands for “ping”. This will cause the tracks to “ping,” with each track firing away individually. The master picks up these signals and measures the time it took to reach the master bus. That way it calculates how much latency each track has, and automatically compensates for it. Pretty elaborate, isn’t it? If you added, removed or changed any plug-in, simply hit the “P” again and it will automatically recalculate. Easy!
There are a few things to understand before using this system. First, all of the tracks in the session have to be un-muted before activating the ping. The plug-in does not “understand” when a ping signal never makes it to the master, and you will start hearing some tracks shifting around.
The second thing that is important to understand is the setup hierarchy with aux tracks. The “group” I was talking about earlier – audio, aux1, aux2, aux3, master – refers to the type of track on which you insert the plug-in. Something a bit confusing here is that when you set it on an “aux” track you must understand that ATA refers to “levels” of auxiliaries. Meaning, let’s say you have a plate reverb on Aux 1, a room reverb on Aux 2 and a delay on Aux 3. in Pro Tools, they are called Aux 1, 2, 3… but to the ATA plug-in, these are all seen as the first level of Aux tracks. Let’s say that the session is more complex and you decided to send all of the drums to Aux 11 (let’s say it’s mono, for explanation purposes) and all your guitars to Aux 12. Then you decided to send the signal from Aux 11 (the drum bus) to the room reverb and the signal from aux 12 (the guitars) to the delay. In this case, the drum bus (Aux 11) and the guitar bus (Aux 12) are the first level of Aux tracks and you would set ATA to the group “Aux 1″ on those tracks. The reverb and delay in this case would be considered the second level of Aux tracks, and ATA for these would need to be set to the “Aux 2″ group. Think about the stages of latency here and it will help you understand what the designers of this plug-in meant. When you open a send from an audio track to an aux track there is some delay introduced to that signal. That would be the first level and would refer to the “Aux 1″ latency group. If then you send a signal from that aux track to a second aux track, you are once again delaying the signal a little bit, and that is the second level which would refer to the “Aux 2″ latency group.
There is a small amount of setup to do here, and at first you might need to get used to the concept but once you’re set up all you have to do is hit the “P” (=ping) button and you’re set. Another good thing is that since the plug-in uses ping and not just a latency readout from the plug-ins, it is guaranteed to reflect the true delay value of the track. Also, the plug-in should be able to accurately compensate for any hardware inserts as well. I have not tried this yet, but I will soon try it using outboard EQ when mixing. I will update you and tell you how it worked!
Best of all, they let you download a demo to make sure it works on your system, and for you to learn how to use it. It works with many different plug-in formats, so it is not restricted to Pro Tools – you can use it with almost any audio software. The best part is that it is not an expensive plug-in – the full version costs $49. I highly recommend it! http://www.mellowmuse.com/ATA.html
If you haven’t yet read the first part – LOUDER IS BETTER! (Part 1) Please scroll down and read it first, to better understand this article.
Now that you’ve had a look at the Fletcher-Munson equal loudness contours, you have an understanding of the way we hear the different frequencies. Let’s review the important concepts to understand here.
First- if we want to hear different frequencies across the audible spectrum at the same level, we actually have to play some frequencies louder than others. Our ears are most sensitive to the “presence” frequencies: 2-5 KHz, where out speech is most intelligible.
Second- and perhaps most important, the louder we play these sounds; the less sensitive we get to the differences in frequencies. This means that when we play music very loud, we will hear it with enhanced low end and high end content. When we lower the volumes, it will sound as if the low and high frequencies are being gradually cut off. Do you know those buttons on stereo receivers called “Loudness” ? What this button is actually meant to do is compensate for the loss of frequencies when listening at low levels, for instance at night when you can’t turn the level up to “normal” levels. Many of you, I’m sure, like to use this button even at moderate to loud levels, because it’s fun to listen to music rich in high and low frequency content! However, the true meaning of the loudness button is to maintain the correct frequency response of the system at lower levels, i.e. to compensate for the Fletcher-Munson loudness curve.
Let’s do an experiment. Below is an audio clip of a frequency sweep. All of the frequencies are played at the same level. (This is a tough experiment to conduct in this setting, as there are so many variables because each of you have a different system with different characteristics and abilities. Regardless, let’s try this. I recommend that you listen with headphones. This will hopefully reduce the variables a little bit. Also, there is a greater chance that your headphones will be capable of reproducing a larger range of frequencies.) Play the frequency sweep, and remember to keep the level low. As you have learned, the level at which you listen has a large effect on your perception. Try to see if you can hear how the level you hear changes as the frequency sweeps. Don’t worry if you find it is hard for you to tell the difference. In my opinion, hearing level differences is one of the hardest things to learn. It took me the longest to develop this sensitivity. It’s good to start thinking about this and sharpen your ear’s sensitivity to level differences.
Now for some conclusions:
- Having talked about the Fletcher-Munson Equal Loudness Contour, we can understand that if we mix at excessively loud levels, our mix won’t sound the same when we bring the levels back down. Our mix will become “small” as the high and low frequencies begin to roll off.
- If we mix at moderate to low levels, turning the level up will make things sound better, as we typically like to hear music rich in high and low frequencies.
- We want our mix to translate as best as possible to as many systems out there, and there are somewhat “established” standard for mixing levels. Typically, you should not exceed 85 dB SPL (more on what dB SPL means on a future post). This is mostly a standard when mixing for film. It is much easier to establish standards when you can somewhat control the environment the audio will be played back in (i.e. the movie theater). Generally, the recommendation is to stay within the 65-85 dB SPL range.
This is also good practice to protect your hearing. Every once in a while, I turn my mix up for just a little while, to hear what it sounds like at higher levels. I never leave the music loud for too long, however, as my ears will reach fatigue very fast that way.
As always, I would love to hear what you think in the comments section. If you want to be alerted when new posts are made available, sign up at the top right of the blog (www.EarsandGears.com)
Rock on!
Well, actually, it is – in a way. There is scientific research to prove it. The data from that research draws some conclusions that are important for us to understand when working with sound.
The Fletcher-Munson Equal Loudness Contour
What is this, and why should you care? The Fletcher-Munson Equal Loudness Contour (named after the two researchers) explains how we perceive the sounds we hear. It turns out we hear things differently at different frequencies and at different sound levels.
The Wikipedia entry for “equal-loudness contour” starts off by saying: “An equal-loudness contour is a measure of sound pressure (dB SPL), over the frequency spectrum, for which a listener perceives a constant loudness when presented with pure steady tones.”(1) In plain easy-to-understand language, what this means is that in order for us to hear low frequency at the same level as a tone in the mid-frequency range, it has to actually be a lot louder. The same is true for high frequencies. If we want to hear a 10 KHz tone at the same level as a 4 KHz tone it has to be played louder.
Let’s take a look at the chart (called the Fletcher-Munson curve) and talk about what we can learn from it. You can click on the chart to see it larger:
This chart shows us how much level is needed for each frequency to be perceived as being at the same volume. As you can see in the chart there is a large dip in the 2-5 KHz frequency range, where our ears are most sensitive. The 2-5 KHz frequencies are where our speech is most intelligible. This range is typically referred to as “presence.” Evolution has designed us to be most sensitive to human speech, and it is done through a fascinating system of resonators (our ear canal) and mechanical amplifiers inside our middle ear.
Another important thing that we learn from this chart is that our sensitivity changes depending on how loud the sounds are when they are played. You will notice on the chart that the louder everything becomes, the less difference there is in sensitivity to the different frequencies. Our hearing “flattens out” at louder levels. This is perhaps the most important conclusion for us to remember when dealing with audio.
Take your time to learn this important concept and explore the chart a bit. If there is anything you don’t understand please post a comment or contact me. The next post will deal with the conclusions of these findings and the implications they have for audio production. We will discuss some standards that were established to help us deal with the issue.
Check back soon!
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Once you understand the concepts here, please continue to the next post – LOUDER IS BETTER! (Part 2)
