Saturday, February 25, 2012

MTA 980 Console!

Assessments and more assessments. Yeah, here we go!


First, let's start with Impedance. There are 2 kinds of impedance: High and Low. Impedance is the resistance to the flow of current and is measured in ohms. High impedance has high resistance from 10,000 to 20,000 ohms. Low impedance has low resistance about 150 to 1000 ohms. The best way to understand impedance is to think of a water hose. A lot of water being sprayed out of a small garden hose into a drain pipe will have no resistance and the small, high impedance hose is spraying into a big, low impedance drain causing distortion. The other way around causes low signal. 

Here are some typical equipment levels:

  • Microphone - about 0.002 volts to 1 volt depending on the mic.
  • Instrument - about 0.1 volt to 1 volt for passive acoustic guitar pickup and to 1.75 volts for active.
  • Semi-pro, or consumer Line - -10 dBV, or 0.316 volts.
  • Pro Line - +4 dBV, or 1.23 volts. 
To connect a high-impedance instrument to a low-impedance amplifier, you use a direct box, or a line matching transformer, impedance matching transformer, or DI Box (direct injection box).

Before the signal reaches the EQ's, it can toggle between mic and line level, but mic level is at line level due to the mic preamps.

The mic preamp is a small amplifier circuit normally at the top of the channel strip. The preamp level controls how much of the incoming signal is amplified. Other names for the mic preamp are mic gain trim, input preamp, trim, preamp, or gain.

The signal passes through the microphone before it hits the mic preamp. When it does hit the preamp, the signal must be at mic level, or the level that the microphone "hears", which is about 30 to 60 db. Then, the preamp can boost the signal to line level. Because mixers work at line level, the mic level must pass through, and be amplified by the mic preamp before the signal can continue on its course.

Though the mic level must be amplified through the mic preamp, we want to use as little mic pre-amplification as possible to reduce the noise it makes when it feeds the signal back into itself. This noise is also amplified--amplified self-noise.

Frequency ranges within each level of the EQ:

<-------- The High Frequencies range from 1,000 hz to 15,000 hz

<-------- The High Mid Frequencies range from 700 hz to 10,000 hz

<-------- The Low Mid Frequencies range from 150 hz to 2,000 hz

<-------- The Low Frequencies range from 40 hz to 650 hz

<-------- Boost or cut EACH by 15db with the right lower knobs

<-------- High Pass Filter at 50 hz
<-------- Pushbutton allows the EQ to be switched in or out of the circuit

Next, the signal hits the Inserts. Something crucial to understanding the difference between Inserts and Sends is that inserts are NOT variable, there is NO percentage of signal sent to them, they are either ON or OFF, 100% or 0%. This is called dynamic processing which includes:

  • Gates
  • Limiters
  • Expanders
  • Compressors
Next, the signal hits the phase option (little hollow circle with slanted line through it). 

Then, to the Auxiliary Sends all the way down to Aux 1, Aux 2, Aux 3, Aux 4 (Mono), Aux 5-6, and Aux 7-8 (Stereo). 

Next, comes the PFL, or Pre-Fader Listen. The name is pretty explanatory. Then, the Fader - The linear, variable, volume knob on the bottom of the channel strip. 

And on to the Mutes and the Solos. 

To the Direct Out. 

To the Pan rotary knobs. 

Then, to the pointless AFL, After-Fader Listen. Whatever that means. 

On the Post Channel Strip, the signal can go through the Group Bus, the Remix (Mix Output), the Solo Bus, Auxiliary Bus, PFL/AFL, and Auto-Mute Bus. 

This is the Monitor Module. The EQ's are the same as the input modules without the 50 hz HPF. 

<-------------- 4 of the possible 8 Auxiliary Busses are available. Aux 1 and Aux 2 are mono and Aux 5-6 is Stereo. 

<--------------- Monitor Level and Pan are here with a rotary level control and a center detention pan pot routes the modules output to the monitors OR through the remix groups when the MIX button is pushed in. 

The Echo Returns are the same except for the bottom half with the Return Level, Pan, Solo, and Mute. 

The Auxiliary Master Module is a little different from the Monitor and Echo Returns Modules.

Studio playback sends monitored signal (Aux 5-6/Aux 7-8) to studio ------->
Solo Master - Rotary potentiometer adjusts the control room monitor level of the stereo PFL/AFL solo signal ------------------------------------------------------>

Alternative Speaker Switch ----------------------------------------------------------->

<-------------- 8 Auxiliary Master Level controllers, each with solo and master mute.
Control room master functions with 4 stereo sources to the control room speakers, MIX - switches the control room speakers from monitoring to 24 multitrack groups to the stereo remix outputs, DIM - attenuates by 20db, and MUTE - kills the control room monitor signal ------------------------------------>
The Auto-Mutes masters that can be pre-selected on the input modules ----->

Talkback - built-in electret mic and level control to route your voice to the mono Aux 1 or 2, or the stereo Aux 5-6 or 7-8 ------------------------------------>
<-------------- Down at the bottom, there is an oscillator with the options 50 hz, 100 hz, 1,000 hz, and 10,000 hz. 

Here is the layout from the MTA Console of the signal flow for each input module. 

The Assessment on this is going to be hard :( Hopefully, I am grasping this because you never know what is going to go through your head come test time. 

Anyways, I am getting a lot of really great experience working with Kory on some of his Capstone Project and singing for the baseball games!

Thursday, February 16, 2012

Terminology of Sound

Terms, terms, and more terms. Good thing I want to know what they mean hahah! Here we go!

  • Sound-Pressure Waves - When sound arrives at the ear in the form of periodic variations in sound pressure.
  • Compression - When a vibrating mass moves outward from its normal resting state, such as the strings of a guitar or the vocal cords, it squeezes air molecules into a compressed area, away from the sound source. This causes the area to be acted on to have a greater than normal atmospheric pressure. 
  • Rarefraction - When the vibration mass moves inward from its normal resting state and an area with less-than-normal atmospheric pressure is created. 
  • Wave Propegation - Tying compression and rarefraction together, wave propegation is the movement of the soundwaves outward from places of high pressure to places of low pressure like the way molecules will puss outward from a baloon when popped.
  • Waveform - It is essentially the graphic representation of a sound-pressure level or voltage level as it moves through a medium over time. It lets us see and explain wave propegation in our physical enviornment. 
  • Amplitude - The distance above or below the centerline of a waveform, such as a pure sine wave. The greater the distance or displacement from the centerline, the great the intensity.
  • Frequency - The rate at which an acoustic generator, electrical signal, or vibrating mass repeats within a cycle of positve and negative amplitudes. 
  • Velocity - The velocity of a sound wave as it travels though air at 68°F is approximately 1130 ftt. per second. This speed is tempurature dependent and increases at a rate of 1.1 ft. per second for each degree increase in tempurature. 
  • Wavelength - The physical distance in a medium between the beginning and the end of a cycle. 
  • Phase - When two or more waveforms are involved in producing a sound, their realtive amplitudes can, and often are, different at some point in time. Phase varies in degrees from 0 to 360. When out of phase waveforms are anything but 180° out of phase, they add together increasing the amplitude. 
  • Harmonic Content - The pressence of other frequencies relative to the fundamental pitch. 
  • Phase Shift - A term that describes one waveform's lead or lag in time with respect to another. It results from a time delay between two or more waveforms. 
  • Patials - The reason that not every instrument sounds the same like a one frequency sine wave, is because of other frequencies, or partials, that exsist in addition to the fundamental pitch that's being played.  
  • Overtone - Upper partials in the higher frequencies above the fundamental pitch.
  • Harmonics - Overtone frequencies that are whole number multiples of the fundamental.
  • Sine Wave - Named so because its amplitude corresponds to the trigonometric sine function, is considered to begin at 0° with an aplitude of 0. The waveform increases gradually to 90°, 180°, 270°, and back to where it started at 360° completing the circle.  
  • Cycle - One completed excursion of a wave, which is plotted over a 360° axis of a circle. 
  • Hertz - The number of cycles that occur within a second in mesured in hertz. 
  • Wavelength = V/F - The mathmatical equation to calculate the physical length of a wave. 
  • Reflection of Sound - Just as light reflects off surfaces, sound waves reflect off surface boundaries at an angle that is equal to, and in the opposite direction of, its initial angle of incidence. 
  • Diffraction of Sound - When sound bends around an object in a manner that reconstructs the signal back to its original form in both frequency and amplitude.
  • Frequency Response Curve - The charted output of an audio device. This is used to depict how a device will respond to the audio spectrum and how it will affect a signal's overall sound. 
  • Timbre - The harmonics and their relative intensities which determine an instruments characteristic sound. 
  • Envelope - This is another characteristic of an instrument that differentiates it from others. An envelope of a waveform is characteristic variations in level that occur in time over the duration of a note.
  • Attack/Decay/Sustain/Release - Each envelope has these 4 sections that vary in amplitude over time. Attack refers to the time taken for a sound to build up to its full volume when a note is initially started. Decay is how quickly the sound levels off to a sustain level after the attack peak. Sustain is the duration of the ongoing sound that is generated after the attack decay, and release is how quickly the sound will ultimately decay when the note is released. 
  • Decible - The unit used to measure sound-pressure levels, and relative changes in signal levels. It is a logarithmic value that expresses differences in intensities between 2 levels.
  • SPL - SPL stands for sound-pressure level. It is the acoustic pressue that's built up within a defined atmospheric area. Basically, the higher the SPL, the louder the sound.
  • Fletcher-Munson Curve of Equal Loudness - The Fletcher-Munson curve shows an equalloudness contour for pure tones as percieved by humans having an average hearing acuity. The curves indicate the average sensitivity to different frequencies at various levels. 
  • Beats - Two tones that differ only slightly in frequency and have approximately the same amplitude will produce an effect known as beats. This effect sounds like repetitive volume surges that are equal in frequency to the difference between these two tones. 
  • Panning - A placement tecnique acheieved when the engineer changes the relative interaural intensity differences and thus creates the illusion of physical positioning between the speakers by by changing the proportion that is sent to each speaker. 
  • Direct Sound - Because sound travels at a constant speed, sound will take the shortest path to a listeners ear, thus, direct sounds detwermine our perception of a sound source's location and size and timbre. 
  • Early Reflections - Waves that bounce off the surrounding surfaces of the room have to travel further to get to the listeners ear and will arrive later than direct sound. Because of this, early refelctions give us clues as to the reflectivity, size, and general nature of an acoustic space.
  • Acoustics - a science dealing with the production, effects, and transmition of sound waves; the transmission of sound waves through various mediums, including reflection, refraction, diffraction, absorption, and interference; the characteristics of auditoriums, theaters, and studios, as well as their design. 
  • Acoustic Isolation - The prevention of external noise (bleed) transmitting into the studio enviornment through the air, ground, or building structure. Basically, controlling the sound from leaking out to a place you don't want it to go.
  • Vocal Booths - Also called Isolation Booths, or iso-booths, a vocal booth is a small version of an iso-room and are perfect for isolating vocals and a single instrument from the larger studio. 
  • Frequency Balance - The frequency components of a room should not adversely affect the acoustic balance of instruments and/or speakers. The acoustic enviornment should not alter the sound quality of the original or recorded performance. The room should exhibit a relatively flat frequency response over the entire audio range without adding its own particular sound coloration.
  • Near Field Monitors - Monitors that deliver clean sound at high SPLs. These are being used more often because they are more acurate in representing the sound that would ne reproduced by the average home speaker system. These also cost a lot less than farfield monitors. 
  • Far Field Monitors - Often involve large, multidriver loudspeakers that are capable of delivering relatively accurate sound at moxerate to high volume levels. Usually these monitors are built into the control room wall to reduce reflections arouns and behind the enclosure and to increase overall speaker efficiency. 
  • Angle of Incidence/Reflection - A refelcted sound will be equal to and opposite from the angle of incidence. When a sound hits a flat surface at a 45° angle, the reflected sound will be 90° from the angle of the first sound path. 
  • Standing Waves - Also known as Room Modes, occur when sound is reflected off of parallel surfaces and travels back on its own path, thereby causing phase differences to interfere with a room's amplitude response.
  • Room Modes - Expressed as integer multiples of the length, width, and depth of the room and indicate which multiple is being referred to for a particular reflection. 
  • Flutter (Slap) Echo - A condition that occurs when parallel boundaries are spaced far enough apart that the listener is able to discern a numner of discrete echoes. It will often produce a "boingy" hollow sound that greatly affects a room's sound character as well as its frequency response. 
  • Diffusers - Acoustical boundaries that reflect the sound wave back at various anfles that are wider than the original inident angle, which breaks up the energy-destructive standing waves. 
  • Direct Signal - Signal made up of the original, incident sound that travels from the source to the listener. 
  • Reverb - The persistence of a signal in the form of reflected waves within an acoustic space, that continues after the original sound has ceased. 
Source: Huber, David Miles., and Robert E. Runstein. Modern Recording Techniques. 7th ed. Boston: Focal, 2001. Print.


Holy Shit. Whoa

Thursday, February 9, 2012

Microphone Placement: Assignment 1

Lab Assignment #1 - 1.30.12

This is group twos first lab and first assignment. We are working with the AKG414 condenser mic in cardioid.

First, we turned on the computer and started a new ProTools session.

I went to the cable room and grabbed 3 cables and a mic stand while Nick went and got the AKG414 with the shock mount and Collin manned the patch panel. Collin plugged the XLR cable into channel 1.

In the lab, Kevin plugged one patch cable into direct output 1 on row 9 and into the ProTools input channel 1. 2 patch cables are going from ProTools output channel 1and 2 to the 2 track input channel 1 and 2.

In the session labeled "group2_project1", we created a mono audio track with the input at A1. Next Michael and I set up the mic in the stand and once it was all locked in, I turned on phantom power on the board on channel 1, turned the fader up to unity gain, and the mic pre to 12:00.

Once Kevin record enabled the track, we started on my mic assignment experience. First, I recorded in front of the glass window/back door with the mic about a foot away from the glass and my mouth about 6 inches from the AKG 414 in cardioid. Next, I went outside, in between the two solid walls in the back and I could hear the sound reflecting off the two cement walls on my sides and how the sound traveling forward echoed out into the lot. Then, I went into the lecture room and the sound was very reverberant due to the high ceilings, but ultimately, the sound was slightly dulled and all over the place. Then, my personal favorite, against the curtains. I was about 6 inches away from the mic and the mic was practically against the curtains. The sound was totally absorbed into the curtains and it sounded like I was in a treated room. Next, I went out to the middle of the front room where the sound was so reverberant due to the cement walls around me and the brick floor below. Then, I used the mic in front of the cement wall next to the women's bathroom and the sound was muted and dull, not like the curtains though. I was 6 inches from the mic and about a foot from the wall. Then, I went into the bathroom and the sound was so nice and the echoes were really cool. Probably not the best for recording vocals unless you didn't want to but reverb on the track later. Lastly, I went back into the lecture hall and put the mic off-axis and, upon listening back, could tell that it was very quiet and sounded like the mic was pointing the wrong way...

The hardest part was remembering everything we learned in class and everything we took notes on. Mostly, it was hard to not just remember them, but to also apply them. For example, not forgetting about the phantom power being on to get signal or turning it off before you pull the plug. The coolest part was actually hearing how different each take actually sounded. I was pretty surprised when listening back how much each sound was affected by the surrounding areas. My favorite part was in front of the curtains because I liked how crisp the sound was, like I was in a vocal booth or something.

  • Window: Short, sharp reverberation from the glass window in front of me, but still a very raw sounding recording.
  • Lecture Hall: Still short sounding reverb, but when something was dropped in the background, you could hear how reverberant the room really is. At the time of recording, it sounded a lot more spacious.
  • Curtains: Still my favorite spot, very sharp with almost no reverb except the little amount that is from the back of me, but such a clean sound. Damn, this microphone is so amazing... 
  • Outside: More roomy of a sound. You can tell that the 2 walls on either side of my are reflecting the sound waves into the microphone but most of my voice is traveling into the woods. 
  • Front Room:  Very reverberant. At the time, it didn't sound sooo hall-like, but the even the sounds of Nadia's chair at least 8 ft. away from me are clear as day in the recording. Sound doesn't get absorbed very well in the front room.
  • Cement Wall: The sound immediately changes to a short, crisp sound again almost more so than the curtains because the back of the room isn't reflecting my voice at all. 
  • Bathroom:  I was in the middle of the bathroom facing the stalls (I forgot to mention that in the recording...) but it is soo reverberant in the bathroom. Maybe because of all the cement and tile, but the reflections are going in every direction. 
  • Off-Axis: The sound is very quiet, obviously because I'm not speaking directly at the mic, but this time I was facing the audience instead of the side of the room and you can hear the slight echoes in the room coming from the high ceilings and back walls. 

This was a really fun activity. It was our first assignment and group meeting basically and overall, we all worked together pretty well. We are still in the feeling out eachothers' personalities on a collaborative basis at this point, but since this was 2 weeks ago, I can say that we are a fantastic group and I couldn't be happier learning about how much we all think alike and different! 

Last night I was freaking out about having our songs done by Monday, but fortunately, after an hour of running through our last song, I am so much less stressed. Thanks group 2 for understanding my crazy weekend schedule and working with each other so perfectly and graciously! We rock as a team :)

The Tendency to Talkback

So Vagina Monologues opening night is tonight. Oh man, I am starting to feel the pressure from all my classes and activities.

I'm also singing the National Anthem for the basketball games on Friday and Saturday night along with Sunday afternoon at the bowling alley tournament in Livermore. Bring it on.

My jacket says Vagina Warrior..... Awesome!


In class, we needed 1 AKG 414, a shock mount, 3 XLR cables, the headphone box, a quarter inch connection, and headphones. 

The patch cables for the board are actually called TT cables for Tiny Telephone cables. 

With the mic on the stand in the lab, we attached the XLR cable to the adapter and plugged it into the Neve 1 spot, completely bypassing the board. 

<--------------  Here, you can see the setup for the microphone to go to the Neve and a pretty awful drawing of what the Neve looks like on the strip.

It's pretty cool!

Next, the signal goes out from the Neve into ProTools (IN on the BOTTOM) in slot 1 and out to the monitors (OUT on TOP) from ProTools 1 and 2 to 2 TR 1 and 2. Next, we will send reverb through a reverb matrix to the headphones for the singer or performer. 

Create a new stereo audio track and name it Talkback and make the input Interface B 1-2. 

THEN, from the Auxiliary sends (9th row up from the bottom of the gray patch bay), take 2 TT cables and plug them from Aux sends 7 and 8 to ProTools IN 9 and 10. In ProTools, under the sends on the Talkback track, select Interface A 9-10. 

You can talk back by pressing the 7-8 button under the talkback section. 

Next, take another set of TT cables and go OUT from ProTools 9-10 to the Headphones IN 1 and 2. 

Always check to see if the signal is being read on the Sampson (the headphone signal distributor) before continuing to the next step.

Good, I'm getting signal. So, I go OUT from the Headphones and go IN to the ROOM 100 IN A and B. From here, you just have to route the tracks you want in the headphones through interface A 9-10. 

Here is a depiction of what the board looks like with all the cables plugged in and getting signal!

Next create a click track with the Marimba II sound as to not be obnoxious and send it to the headphones like explained above. 

NOTE: The output of all tracks needs to be A 1-2 in order to be heard in the monitors in the studio, and any other outputs desired must be through the sends.
If you don't want to hear the click track in the studio, but want to keep the sound in the headphones, you apply the Prefader to the send window that pops up. A little button at the top of the fader will make this happen. What happens is the pre fader makes the signal precede the main fader and goes to the headphones first. 

Then, create a stereo Auxiliary track for the Reverb. Put a Reverb insert on the track. Then make the input of the track BUS 1-2.  On anything meant to have reverb on it, BUS 1-2 should be selected in the sends. Apply the Prefader to bypass the reverb without affecting it in the headphones. 


I am pretty upset that I missed class last night. I'm upset that I have to miss the lab on top of the missed class. I am grateful, however, that Drew understands. I can't deal with my own disappointment let alone his. Anyway, I am nervous about all the boys seeing the play tonight. I forgot about how completely taboo the things I talk about are. Ugh. 

I'm proud to say that my groups 3 songs just got finished last night as well :) Yay!

Sunday, February 5, 2012

Trum Tracking and the Millenia

So, this class we continued miking and tracking drums, but we treated the room as best we could and used a few more mics than the last. On top of that, we got to track bass and guitar over the drums with the Millenia.


Today, we used the D112 on the inside of the bass drum and the 421 on the outside.
The SM57 on the bottom of the snare and the E22 on the top of the snare.
Last, 2 AKG 4050's are used in the M/S style for the overheads.

Mic    |     Patch Bay/Panel Channel   |    ProTools Channel
D112                      1                                          1
421                         3                                          2
SM57                     4                                          3
E22                        5                                          4
4050 Side               8                                          5
4050 Mid               9                                          6

The reason why we skipped channels 2, 6, and 7 on the patch panel and bay is because something along that line didn't work for those channels.

BUT, the channels in ProTools don't have to correspond to anything other than the brown patch bay on the board, so if row 9, channel 3 goes to ProTools in channel 2, ProTools will register whatever mic going through the patch panel and bay channel 3 as channel 2.

And never forget.... IN on BOTTOM, OUT on TOP!!!!!!

All the faders should b, and pretty much were, at unity gain and the mic pre at 12 o'clock. Both the 4050s and the E22 are condenser mics and need phantom +48v power.

Duplicate the Side 4050 track and Opt. Click Drag the wave file to the duplicate track. Apply the Trim under TDM > Other > Trim then bring up that track's fader a little to compensate for the loss in dbs.

To clarify which mic is the mid and which is the side... the mic with the side facing the kit is the side and the one with the front facing the kit is the mid. Duplicate the side mic track.

To treat the room, we put packing blankets over the two white boards and used the bass drop wall things on the sides. The bass drop walls sit on chairs and two go in between the white boards. The difference in sound was incredible! The drums went from sounding spacey and roomy to crisp and sharp.

Next, tracking with the Millenia..... So we created a mono audio track in ProTools and plugged the bass into the Millenia. This is a drawing of what the left side of the Millenia strip looks like.....

VT = Vacuum Tube
SS = Solid State
POL = Polar Opposite
XFMR = Transformer

The output master is what controls the level of the instrument being recorded into ProTools. So the Millenia doesn't really pass through the board at all. There are 4 patch points for the Millenia on the Brown Patch Bay that go to ProTools.

From the Millenia patch point 1 to ProTools channel 1 will register sound to the new audio track. Then, don't forget to go out from ProTools 1 and 2 to the 2 Track plugs (the monitors). 


This was a really fun class because the possibilities of what we will be able to do is just starting to unfold and it's really exciting for all of us!